Merge from Chromium at DEPS revision 267aeeb8d85c
This commit was generated by merge_to_master.py.
Change-Id: I3cccc8f04ad0036aecdb7eefe316a059ebcefaf9
diff --git a/OWNERS b/OWNERS
index 570953a..a0b7b4b 100644
--- a/OWNERS
+++ b/OWNERS
@@ -1,14 +1,13 @@
set noparent
-fischman@webrtc.org
henrike@webrtc.org
hta@webrtc.org
+jiayl@webrtc.org
juberti@webrtc.org
mallinath@webrtc.org
perkj@webrtc.org
pthatcher@webrtc.org
sergeyu@chromium.org
tommi@webrtc.org
-wu@webrtc.org
per-file *.isolate=kjellander@webrtc.org
diff --git a/PRESUBMIT.py b/PRESUBMIT.py
index d447947..a7b4851 100644
--- a/PRESUBMIT.py
+++ b/PRESUBMIT.py
@@ -24,11 +24,7 @@
# ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
# List of files that should not be committed to
-DO_NOT_SUBMIT_FILES = [
- "talk/media/webrtc/webrtcmediaengine.h",
- "talk/media/webrtc/webrtcvideoengine.cc",
- "talk/media/webrtc/webrtcvideoengine.h",
- "talk/media/webrtc/webrtcvideoengine_unittest.cc"]
+DO_NOT_SUBMIT_FILES = ["talk/media/webrtc/webrtcvideoengine.cc"]
def _LicenseHeader(input_api):
"""Returns the license header regexp."""
diff --git a/app/webrtc/java/android/org/webrtc/VideoRendererGui.java b/app/webrtc/java/android/org/webrtc/VideoRendererGui.java
index 439f942..af625c0 100644
--- a/app/webrtc/java/android/org/webrtc/VideoRendererGui.java
+++ b/app/webrtc/java/android/org/webrtc/VideoRendererGui.java
@@ -37,6 +37,10 @@
import javax.microedition.khronos.egl.EGLConfig;
import javax.microedition.khronos.opengles.GL10;
+import android.graphics.SurfaceTexture;
+import android.opengl.EGL14;
+import android.opengl.EGLContext;
+import android.opengl.GLES11Ext;
import android.opengl.GLES20;
import android.opengl.GLSurfaceView;
import android.util.Log;
@@ -54,14 +58,28 @@
private static VideoRendererGui instance = null;
private static final String TAG = "VideoRendererGui";
private GLSurfaceView surface;
+ private static EGLContext eglContext = null;
// Indicates if SurfaceView.Renderer.onSurfaceCreated was called.
// If true then for every newly created yuv image renderer createTexture()
// should be called. The variable is accessed on multiple threads and
// all accesses are synchronized on yuvImageRenderers' object lock.
private boolean onSurfaceCreatedCalled;
+ private int screenWidth;
+ private int screenHeight;
// List of yuv renderers.
private ArrayList<YuvImageRenderer> yuvImageRenderers;
- private int program;
+ private int yuvProgram;
+ private int oesProgram;
+ // Types of video scaling:
+ // SCALE_ASPECT_FIT - video frame is scaled to fit the size of the view by
+ // maintaining the aspect ratio (black borders may be displayed).
+ // SCALE_ASPECT_FILL - video frame is scaled to fill the size of the view by
+ // maintaining the aspect ratio. Some portion of the video frame may be
+ // clipped.
+ // SCALE_FILL - video frame is scaled to to fill the size of the view. Video
+ // aspect ratio is changed if necessary.
+ private static enum ScalingType
+ { SCALE_ASPECT_FIT, SCALE_ASPECT_FILL, SCALE_FILL };
private final String VERTEX_SHADER_STRING =
"varying vec2 interp_tc;\n" +
@@ -73,7 +91,7 @@
" interp_tc = in_tc;\n" +
"}\n";
- private final String FRAGMENT_SHADER_STRING =
+ private final String YUV_FRAGMENT_SHADER_STRING =
"precision mediump float;\n" +
"varying vec2 interp_tc;\n" +
"\n" +
@@ -91,6 +109,19 @@
" y + 1.77 * u, 1);\n" +
"}\n";
+
+ private static final String OES_FRAGMENT_SHADER_STRING =
+ "#extension GL_OES_EGL_image_external : require\n" +
+ "precision mediump float;\n" +
+ "varying vec2 interp_tc;\n" +
+ "\n" +
+ "uniform samplerExternalOES oes_tex;\n" +
+ "\n" +
+ "void main() {\n" +
+ " gl_FragColor = texture2D(oes_tex, interp_tc);\n" +
+ "}\n";
+
+
private VideoRendererGui(GLSurfaceView surface) {
this.surface = surface;
// Create an OpenGL ES 2.0 context.
@@ -124,23 +155,46 @@
return buffer;
}
- // Compile & attach a |type| shader specified by |source| to |program|.
- private static void addShaderTo(
- int type, String source, int program) {
+ private int loadShader(int shaderType, String source) {
int[] result = new int[] {
GLES20.GL_FALSE
};
- int shader = GLES20.glCreateShader(type);
+ int shader = GLES20.glCreateShader(shaderType);
GLES20.glShaderSource(shader, source);
GLES20.glCompileShader(shader);
GLES20.glGetShaderiv(shader, GLES20.GL_COMPILE_STATUS, result, 0);
- abortUnless(result[0] == GLES20.GL_TRUE,
- GLES20.glGetShaderInfoLog(shader) + ", source: " + source);
- GLES20.glAttachShader(program, shader);
- GLES20.glDeleteShader(shader);
-
+ if (result[0] != GLES20.GL_TRUE) {
+ Log.e(TAG, "Could not compile shader " + shaderType + ":" +
+ GLES20.glGetShaderInfoLog(shader));
+ throw new RuntimeException(GLES20.glGetShaderInfoLog(shader));
+ }
checkNoGLES2Error();
- }
+ return shader;
+}
+
+
+ private int createProgram(String vertexSource, String fragmentSource) {
+ int vertexShader = loadShader(GLES20.GL_VERTEX_SHADER, vertexSource);
+ int fragmentShader = loadShader(GLES20.GL_FRAGMENT_SHADER, fragmentSource);
+ int program = GLES20.glCreateProgram();
+ if (program == 0) {
+ throw new RuntimeException("Could not create program");
+ }
+ GLES20.glAttachShader(program, vertexShader);
+ GLES20.glAttachShader(program, fragmentShader);
+ GLES20.glLinkProgram(program);
+ int[] linkStatus = new int[] {
+ GLES20.GL_FALSE
+ };
+ GLES20.glGetProgramiv(program, GLES20.GL_LINK_STATUS, linkStatus, 0);
+ if (linkStatus[0] != GLES20.GL_TRUE) {
+ Log.e(TAG, "Could not link program: " +
+ GLES20.glGetProgramInfoLog(program));
+ throw new RuntimeException(GLES20.glGetProgramInfoLog(program));
+ }
+ checkNoGLES2Error();
+ return program;
+}
/**
* Class used to display stream of YUV420 frames at particular location
@@ -149,9 +203,12 @@
*/
private static class YuvImageRenderer implements VideoRenderer.Callbacks {
private GLSurfaceView surface;
- private int program;
- private FloatBuffer textureVertices;
+ private int id;
+ private int yuvProgram;
+ private int oesProgram;
private int[] yuvTextures = { -1, -1, -1 };
+ private int oesTexture = -1;
+ private float[] stMatrix = new float[16];
// Render frame queue - accessed by two threads. renderFrame() call does
// an offer (writing I420Frame to render) and early-returns (recording
@@ -159,8 +216,13 @@
// copies frame to texture and then removes it from a queue using poll().
LinkedBlockingQueue<I420Frame> frameToRenderQueue;
// Local copy of incoming video frame.
- private I420Frame frameToRender;
- // Flag if renderFrame() was ever called
+ private I420Frame yuvFrameToRender;
+ private I420Frame textureFrameToRender;
+ // Type of video frame used for recent frame rendering.
+ private static enum RendererType { RENDERER_YUV, RENDERER_TEXTURE };
+ private RendererType rendererType;
+ private ScalingType scalingType;
+ // Flag if renderFrame() was ever called.
boolean seenFrame;
// Total number of video frames received in renderFrame() call.
private int framesReceived;
@@ -174,40 +236,68 @@
// Time in ns spent in draw() function.
private long drawTimeNs;
// Time in ns spent in renderFrame() function - including copying frame
- // data to rendering planes
+ // data to rendering planes.
private long copyTimeNs;
-
- // Texture Coordinates mapping the entire texture.
- private final FloatBuffer textureCoords = directNativeFloatBuffer(
- new float[] {
- 0, 0, 0, 1, 1, 0, 1, 1
- });
+ // Texture vertices.
+ private float texLeft;
+ private float texRight;
+ private float texTop;
+ private float texBottom;
+ private FloatBuffer textureVertices;
+ // Texture UV coordinates offsets.
+ private float texOffsetU;
+ private float texOffsetV;
+ private FloatBuffer textureCoords;
+ // Flag if texture vertices or coordinates update is needed.
+ private boolean updateTextureProperties;
+ // Viewport dimensions.
+ private int screenWidth;
+ private int screenHeight;
+ // Video dimension.
+ private int videoWidth;
+ private int videoHeight;
private YuvImageRenderer(
- GLSurfaceView surface,
- int x, int y, int width, int height) {
- Log.v(TAG, "YuvImageRenderer.Create");
+ GLSurfaceView surface, int id,
+ int x, int y, int width, int height,
+ ScalingType scalingType) {
+ Log.d(TAG, "YuvImageRenderer.Create id: " + id);
this.surface = surface;
+ this.id = id;
+ this.scalingType = scalingType;
frameToRenderQueue = new LinkedBlockingQueue<I420Frame>(1);
// Create texture vertices.
- float xLeft = (x - 50) / 50.0f;
- float yTop = (50 - y) / 50.0f;
- float xRight = Math.min(1.0f, (x + width - 50) / 50.0f);
- float yBottom = Math.max(-1.0f, (50 - y - height) / 50.0f);
+ texLeft = (x - 50) / 50.0f;
+ texTop = (50 - y) / 50.0f;
+ texRight = Math.min(1.0f, (x + width - 50) / 50.0f);
+ texBottom = Math.max(-1.0f, (50 - y - height) / 50.0f);
float textureVeticesFloat[] = new float[] {
- xLeft, yTop,
- xLeft, yBottom,
- xRight, yTop,
- xRight, yBottom
+ texLeft, texTop,
+ texLeft, texBottom,
+ texRight, texTop,
+ texRight, texBottom
};
textureVertices = directNativeFloatBuffer(textureVeticesFloat);
+ // Create texture UV coordinates.
+ texOffsetU = 0;
+ texOffsetV = 0;
+ float textureCoordinatesFloat[] = new float[] {
+ texOffsetU, texOffsetV, // left top
+ texOffsetU, 1.0f - texOffsetV, // left bottom
+ 1.0f - texOffsetU, texOffsetV, // right top
+ 1.0f - texOffsetU, 1.0f - texOffsetV // right bottom
+ };
+ textureCoords = directNativeFloatBuffer(textureCoordinatesFloat);
+ updateTextureProperties = false;
}
- private void createTextures(int program) {
- Log.v(TAG, " YuvImageRenderer.createTextures");
- this.program = program;
+ private void createTextures(int yuvProgram, int oesProgram) {
+ Log.d(TAG, " YuvImageRenderer.createTextures " + id + " on GL thread:" +
+ Thread.currentThread().getId());
+ this.yuvProgram = yuvProgram;
+ this.oesProgram = oesProgram;
- // Generate 3 texture ids for Y/U/V and place them into |textures|.
+ // Generate 3 texture ids for Y/U/V and place them into |yuvTextures|.
GLES20.glGenTextures(3, yuvTextures, 0);
for (int i = 0; i < 3; i++) {
GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
@@ -226,39 +316,139 @@
checkNoGLES2Error();
}
+ private void checkAdjustTextureCoords() {
+ if (!updateTextureProperties ||
+ scalingType == ScalingType.SCALE_FILL) {
+ return;
+ }
+ // Re - calculate texture vertices to preserve video aspect ratio.
+ float texRight = this.texRight;
+ float texLeft = this.texLeft;
+ float texTop = this.texTop;
+ float texBottom = this.texBottom;
+ float displayWidth = (texRight - texLeft) * screenWidth / 2;
+ float displayHeight = (texTop - texBottom) * screenHeight / 2;
+ if (displayWidth > 1 && displayHeight > 1 &&
+ videoWidth > 1 && videoHeight > 1) {
+ float displayAspectRatio = displayWidth / displayHeight;
+ float videoAspectRatio = (float)videoWidth / videoHeight;
+ if (scalingType == ScalingType.SCALE_ASPECT_FIT) {
+ // Need to re-adjust vertices width or height to match video AR.
+ if (displayAspectRatio > videoAspectRatio) {
+ float deltaX = (displayWidth - videoAspectRatio * displayHeight) /
+ instance.screenWidth;
+ texRight -= deltaX;
+ texLeft += deltaX;
+ } else {
+ float deltaY = (displayHeight - displayWidth / videoAspectRatio) /
+ instance.screenHeight;
+ texTop -= deltaY;
+ texBottom += deltaY;
+ }
+ // Re-allocate vertices buffer to adjust to video aspect ratio.
+ float textureVeticesFloat[] = new float[] {
+ texLeft, texTop,
+ texLeft, texBottom,
+ texRight, texTop,
+ texRight, texBottom
+ };
+ textureVertices = directNativeFloatBuffer(textureVeticesFloat);
+ }
+ if (scalingType == ScalingType.SCALE_ASPECT_FILL) {
+ // Need to re-adjust UV coordinates to match display AR.
+ if (displayAspectRatio > videoAspectRatio) {
+ texOffsetV = (1.0f - videoAspectRatio / displayAspectRatio) / 2.0f;
+ } else {
+ texOffsetU = (1.0f - displayAspectRatio / videoAspectRatio) / 2.0f;
+ }
+ // Re-allocate coordinates buffer to adjust to display aspect ratio.
+ float textureCoordinatesFloat[] = new float[] {
+ texOffsetU, texOffsetV, // left top
+ texOffsetU, 1.0f - texOffsetV, // left bottom
+ 1.0f - texOffsetU, texOffsetV, // right top
+ 1.0f - texOffsetU, 1.0f - texOffsetV // right bottom
+ };
+ textureCoords = directNativeFloatBuffer(textureCoordinatesFloat);
+ }
+ }
+ updateTextureProperties = false;
+ }
+
private void draw() {
- long now = System.nanoTime();
if (!seenFrame) {
// No frame received yet - nothing to render.
return;
}
+ // Check if texture vertices/coordinates adjustment is required when
+ // screen orientation changes or video frame size changes.
+ checkAdjustTextureCoords();
+
+ long now = System.nanoTime();
+
I420Frame frameFromQueue;
synchronized (frameToRenderQueue) {
frameFromQueue = frameToRenderQueue.peek();
if (frameFromQueue != null && startTimeNs == -1) {
startTimeNs = now;
}
- for (int i = 0; i < 3; ++i) {
- int w = (i == 0) ? frameToRender.width : frameToRender.width / 2;
- int h = (i == 0) ? frameToRender.height : frameToRender.height / 2;
- GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
- GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, yuvTextures[i]);
- if (frameFromQueue != null) {
- GLES20.glTexImage2D(GLES20.GL_TEXTURE_2D, 0, GLES20.GL_LUMINANCE,
- w, h, 0, GLES20.GL_LUMINANCE, GLES20.GL_UNSIGNED_BYTE,
- frameFromQueue.yuvPlanes[i]);
+
+ if (rendererType == RendererType.RENDERER_YUV) {
+ // YUV textures rendering.
+ GLES20.glUseProgram(yuvProgram);
+
+ for (int i = 0; i < 3; ++i) {
+ GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
+ GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, yuvTextures[i]);
+ if (frameFromQueue != null) {
+ int w = (i == 0) ?
+ frameFromQueue.width : frameFromQueue.width / 2;
+ int h = (i == 0) ?
+ frameFromQueue.height : frameFromQueue.height / 2;
+ GLES20.glTexImage2D(GLES20.GL_TEXTURE_2D, 0, GLES20.GL_LUMINANCE,
+ w, h, 0, GLES20.GL_LUMINANCE, GLES20.GL_UNSIGNED_BYTE,
+ frameFromQueue.yuvPlanes[i]);
+ }
}
+ } else {
+ // External texture rendering.
+ GLES20.glUseProgram(oesProgram);
+
+ if (frameFromQueue != null) {
+ oesTexture = frameFromQueue.textureId;
+ if (frameFromQueue.textureObject instanceof SurfaceTexture) {
+ SurfaceTexture surfaceTexture =
+ (SurfaceTexture) frameFromQueue.textureObject;
+ surfaceTexture.updateTexImage();
+ surfaceTexture.getTransformMatrix(stMatrix);
+ }
+ }
+ GLES20.glActiveTexture(GLES20.GL_TEXTURE0);
+ GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES, oesTexture);
}
+
if (frameFromQueue != null) {
frameToRenderQueue.poll();
}
}
- int posLocation = GLES20.glGetAttribLocation(program, "in_pos");
+
+ if (rendererType == RendererType.RENDERER_YUV) {
+ GLES20.glUniform1i(GLES20.glGetUniformLocation(yuvProgram, "y_tex"), 0);
+ GLES20.glUniform1i(GLES20.glGetUniformLocation(yuvProgram, "u_tex"), 1);
+ GLES20.glUniform1i(GLES20.glGetUniformLocation(yuvProgram, "v_tex"), 2);
+ }
+
+ int posLocation = GLES20.glGetAttribLocation(yuvProgram, "in_pos");
+ if (posLocation == -1) {
+ throw new RuntimeException("Could not get attrib location for in_pos");
+ }
GLES20.glEnableVertexAttribArray(posLocation);
GLES20.glVertexAttribPointer(
posLocation, 2, GLES20.GL_FLOAT, false, 0, textureVertices);
- int texLocation = GLES20.glGetAttribLocation(program, "in_tc");
+ int texLocation = GLES20.glGetAttribLocation(yuvProgram, "in_tc");
+ if (texLocation == -1) {
+ throw new RuntimeException("Could not get attrib location for in_tc");
+ }
GLES20.glEnableVertexAttribArray(texLocation);
GLES20.glVertexAttribPointer(
texLocation, 2, GLES20.GL_FLOAT, false, 0, textureCoords);
@@ -281,29 +471,41 @@
private void logStatistics() {
long timeSinceFirstFrameNs = System.nanoTime() - startTimeNs;
- Log.v(TAG, "Frames received: " + framesReceived + ". Dropped: " +
- framesDropped + ". Rendered: " + framesRendered);
+ Log.d(TAG, "ID: " + id + ". Type: " + rendererType +
+ ". Frames received: " + framesReceived +
+ ". Dropped: " + framesDropped + ". Rendered: " + framesRendered);
if (framesReceived > 0 && framesRendered > 0) {
- Log.v(TAG, "Duration: " + (int)(timeSinceFirstFrameNs / 1e6) +
+ Log.d(TAG, "Duration: " + (int)(timeSinceFirstFrameNs / 1e6) +
" ms. FPS: " + (float)framesRendered * 1e9 / timeSinceFirstFrameNs);
- Log.v(TAG, "Draw time: " +
+ Log.d(TAG, "Draw time: " +
(int) (drawTimeNs / (1000 * framesRendered)) + " us. Copy time: " +
(int) (copyTimeNs / (1000 * framesReceived)) + " us");
}
}
+ public void setScreenSize(final int screenWidth, final int screenHeight) {
+ this.screenWidth = screenWidth;
+ this.screenHeight = screenHeight;
+ updateTextureProperties = true;
+ }
+
@Override
public void setSize(final int width, final int height) {
- Log.v(TAG, "YuvImageRenderer.setSize: " + width + " x " + height);
+ Log.d(TAG, "ID: " + id + ". YuvImageRenderer.setSize: " +
+ width + " x " + height);
+ videoWidth = width;
+ videoHeight = height;
int[] strides = { width, width / 2, width / 2 };
// Frame re-allocation need to be synchronized with copying
// frame to textures in draw() function to avoid re-allocating
// the frame while it is being copied.
synchronized (frameToRenderQueue) {
- // Clear rendering queue
+ // Clear rendering queue.
frameToRenderQueue.poll();
- // Re-allocate / allocate the frame
- frameToRender = new I420Frame(width, height, strides, null);
+ // Re-allocate / allocate the frame.
+ yuvFrameToRender = new I420Frame(width, height, strides, null);
+ textureFrameToRender = new I420Frame(width, height, null, -1);
+ updateTextureProperties = true;
}
}
@@ -311,24 +513,26 @@
public synchronized void renderFrame(I420Frame frame) {
long now = System.nanoTime();
framesReceived++;
- // Check input frame parameters.
- if (!(frame.yuvStrides[0] == frame.width &&
- frame.yuvStrides[1] == frame.width / 2 &&
- frame.yuvStrides[2] == frame.width / 2)) {
- Log.e(TAG, "Incorrect strides " + frame.yuvStrides[0] + ", " +
- frame.yuvStrides[1] + ", " + frame.yuvStrides[2]);
- return;
- }
// Skip rendering of this frame if setSize() was not called.
- if (frameToRender == null) {
+ if (yuvFrameToRender == null || textureFrameToRender == null) {
framesDropped++;
return;
}
- // Check incoming frame dimensions
- if (frame.width != frameToRender.width ||
- frame.height != frameToRender.height) {
- throw new RuntimeException("Wrong frame size " +
- frame.width + " x " + frame.height);
+ // Check input frame parameters.
+ if (frame.yuvFrame) {
+ if (!(frame.yuvStrides[0] == frame.width &&
+ frame.yuvStrides[1] == frame.width / 2 &&
+ frame.yuvStrides[2] == frame.width / 2)) {
+ Log.e(TAG, "Incorrect strides " + frame.yuvStrides[0] + ", " +
+ frame.yuvStrides[1] + ", " + frame.yuvStrides[2]);
+ return;
+ }
+ // Check incoming frame dimensions.
+ if (frame.width != yuvFrameToRender.width ||
+ frame.height != yuvFrameToRender.height) {
+ throw new RuntimeException("Wrong frame size " +
+ frame.width + " x " + frame.height);
+ }
}
if (frameToRenderQueue.size() > 0) {
@@ -336,20 +540,36 @@
framesDropped++;
return;
}
- frameToRender.copyFrom(frame);
+
+ // Create a local copy of the frame.
+ if (frame.yuvFrame) {
+ yuvFrameToRender.copyFrom(frame);
+ rendererType = RendererType.RENDERER_YUV;
+ frameToRenderQueue.offer(yuvFrameToRender);
+ } else {
+ textureFrameToRender.copyFrom(frame);
+ rendererType = RendererType.RENDERER_TEXTURE;
+ frameToRenderQueue.offer(textureFrameToRender);
+ }
copyTimeNs += (System.nanoTime() - now);
- frameToRenderQueue.offer(frameToRender);
seenFrame = true;
+
+ // Request rendering.
surface.requestRender();
}
+
}
/** Passes GLSurfaceView to video renderer. */
public static void setView(GLSurfaceView surface) {
- Log.v(TAG, "VideoRendererGui.setView");
+ Log.d(TAG, "VideoRendererGui.setView");
instance = new VideoRendererGui(surface);
}
+ public static EGLContext getEGLContext() {
+ return eglContext;
+ }
+
/**
* Creates VideoRenderer with top left corner at (x, y) and resolution
* (width, height). All parameters are in percentage of screen resolution.
@@ -360,6 +580,11 @@
return new VideoRenderer(javaGuiRenderer);
}
+ public static VideoRenderer.Callbacks createGuiRenderer(
+ int x, int y, int width, int height) {
+ return create(x, y, width, height);
+ }
+
/**
* Creates VideoRenderer.Callbacks with top left corner at (x, y) and
* resolution (width, height). All parameters are in percentage of
@@ -379,7 +604,8 @@
"Attempt to create yuv renderer before setting GLSurfaceView");
}
final YuvImageRenderer yuvImageRenderer = new YuvImageRenderer(
- instance.surface, x, y, width, height);
+ instance.surface, instance.yuvImageRenderers.size(),
+ x, y, width, height, ScalingType.SCALE_ASPECT_FIT);
synchronized (instance.yuvImageRenderers) {
if (instance.onSurfaceCreatedCalled) {
// onSurfaceCreated has already been called for VideoRendererGui -
@@ -388,7 +614,10 @@
final CountDownLatch countDownLatch = new CountDownLatch(1);
instance.surface.queueEvent(new Runnable() {
public void run() {
- yuvImageRenderer.createTextures(instance.program);
+ yuvImageRenderer.createTextures(
+ instance.yuvProgram, instance.oesProgram);
+ yuvImageRenderer.setScreenSize(
+ instance.screenWidth, instance.screenHeight);
countDownLatch.countDown();
}
});
@@ -407,43 +636,40 @@
@Override
public void onSurfaceCreated(GL10 unused, EGLConfig config) {
- Log.v(TAG, "VideoRendererGui.onSurfaceCreated");
+ Log.d(TAG, "VideoRendererGui.onSurfaceCreated");
+ // Store render EGL context
+ eglContext = EGL14.eglGetCurrentContext();
+ Log.d(TAG, "VideoRendererGui EGL Context: " + eglContext);
- // Create program.
- program = GLES20.glCreateProgram();
- addShaderTo(GLES20.GL_VERTEX_SHADER, VERTEX_SHADER_STRING, program);
- addShaderTo(GLES20.GL_FRAGMENT_SHADER, FRAGMENT_SHADER_STRING, program);
-
- GLES20.glLinkProgram(program);
- int[] result = new int[] {
- GLES20.GL_FALSE
- };
- result[0] = GLES20.GL_FALSE;
- GLES20.glGetProgramiv(program, GLES20.GL_LINK_STATUS, result, 0);
- abortUnless(result[0] == GLES20.GL_TRUE,
- GLES20.glGetProgramInfoLog(program));
- GLES20.glUseProgram(program);
-
- GLES20.glUniform1i(GLES20.glGetUniformLocation(program, "y_tex"), 0);
- GLES20.glUniform1i(GLES20.glGetUniformLocation(program, "u_tex"), 1);
- GLES20.glUniform1i(GLES20.glGetUniformLocation(program, "v_tex"), 2);
+ // Create YUV and OES programs.
+ yuvProgram = createProgram(VERTEX_SHADER_STRING,
+ YUV_FRAGMENT_SHADER_STRING);
+ oesProgram = createProgram(VERTEX_SHADER_STRING,
+ OES_FRAGMENT_SHADER_STRING);
synchronized (yuvImageRenderers) {
// Create textures for all images.
for (YuvImageRenderer yuvImageRenderer : yuvImageRenderers) {
- yuvImageRenderer.createTextures(program);
+ yuvImageRenderer.createTextures(yuvProgram, oesProgram);
}
onSurfaceCreatedCalled = true;
}
checkNoGLES2Error();
- GLES20.glClearColor(0.0f, 0.0f, 0.3f, 1.0f);
+ GLES20.glClearColor(0.0f, 0.0f, 0.1f, 1.0f);
}
@Override
public void onSurfaceChanged(GL10 unused, int width, int height) {
- Log.v(TAG, "VideoRendererGui.onSurfaceChanged: " +
+ Log.d(TAG, "VideoRendererGui.onSurfaceChanged: " +
width + " x " + height + " ");
+ screenWidth = width;
+ screenHeight = height;
GLES20.glViewport(0, 0, width, height);
+ synchronized (yuvImageRenderers) {
+ for (YuvImageRenderer yuvImageRenderer : yuvImageRenderers) {
+ yuvImageRenderer.setScreenSize(screenWidth, screenHeight);
+ }
+ }
}
@Override
diff --git a/app/webrtc/java/jni/peerconnection_jni.cc b/app/webrtc/java/jni/peerconnection_jni.cc
index 9b34fb5..27f69e4 100644
--- a/app/webrtc/java/jni/peerconnection_jni.cc
+++ b/app/webrtc/java/jni/peerconnection_jni.cc
@@ -77,23 +77,31 @@
#include "third_party/libyuv/include/libyuv/convert_from.h"
#include "third_party/libyuv/include/libyuv/video_common.h"
#include "webrtc/base/bind.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/messagequeue.h"
#include "webrtc/base/ssladapter.h"
+#include "webrtc/common_video/interface/texture_video_frame.h"
#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
#include "webrtc/system_wrappers/interface/compile_assert.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/video_engine/include/vie_base.h"
#include "webrtc/voice_engine/include/voe_base.h"
-#ifdef ANDROID
+#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
+#include <android/log.h>
+#include "webrtc/modules/video_capture/video_capture_internal.h"
+#include "webrtc/modules/video_render/video_render_internal.h"
#include "webrtc/system_wrappers/interface/logcat_trace_context.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
using webrtc::CodecSpecificInfo;
using webrtc::DecodedImageCallback;
using webrtc::EncodedImage;
using webrtc::I420VideoFrame;
using webrtc::LogcatTraceContext;
using webrtc::RTPFragmentationHeader;
+using webrtc::TextureVideoFrame;
+using webrtc::TickTime;
using webrtc::VideoCodec;
#endif
@@ -111,6 +119,7 @@
using webrtc::DataChannelInterface;
using webrtc::DataChannelObserver;
using webrtc::IceCandidateInterface;
+using webrtc::NativeHandle;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaSourceInterface;
using webrtc::MediaStreamInterface;
@@ -128,33 +137,18 @@
using webrtc::VideoTrackVector;
using webrtc::kVideoCodecVP8;
-// Abort the process if |x| is false, emitting |msg|.
-#define CHECK(x, msg) \
- if (x) {} else { \
- LOG(LS_ERROR) << __FILE__ << ":" << __LINE__ << ": " << msg; \
- abort(); \
- }
-// Abort the process if |jni| has a Java exception pending, emitting |msg|.
-#define CHECK_EXCEPTION(jni, msg) \
- if (0) {} else { \
- if (jni->ExceptionCheck()) { \
- jni->ExceptionDescribe(); \
- jni->ExceptionClear(); \
- CHECK(0, msg); \
- } \
- }
+// Abort the process if |jni| has a Java exception pending.
+// This macros uses the comma operator to execute ExceptionDescribe
+// and ExceptionClear ignoring their return values and sending ""
+// to the error stream.
+#define CHECK_EXCEPTION(jni) \
+ CHECK(!jni->ExceptionCheck()) \
+ << (jni->ExceptionDescribe(), jni->ExceptionClear(), "")
-// Helper that calls ptr->Release() and logs a useful message if that didn't
-// actually delete *ptr because of extra refcounts.
-#define CHECK_RELEASE(ptr) \
- do { \
- int count = (ptr)->Release(); \
- if (count != 0) { \
- LOG(LS_ERROR) << "Refcount unexpectedly not 0: " << (ptr) \
- << ": " << count; \
- } \
- CHECK(!count, "Unexpected refcount"); \
- } while (0)
+// Helper that calls ptr->Release() and aborts the process with a useful
+// message if that didn't actually delete *ptr because of extra refcounts.
+#define CHECK_RELEASE(ptr) \
+ CHECK_EQ(0, (ptr)->Release()) << "Unexpected refcount."
namespace {
@@ -166,18 +160,25 @@
// were attached by the JVM because of a Java->native call.
static pthread_key_t g_jni_ptr;
+#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
+// Set in PeerConnectionFactory_initializeAndroidGlobals().
+static bool factory_static_initialized = false;
+#endif
+
+
// Return thread ID as a string.
static std::string GetThreadId() {
char buf[21]; // Big enough to hold a kuint64max plus terminating NULL.
- CHECK(snprintf(buf, sizeof(buf), "%llu", syscall(__NR_gettid)) <= sizeof(buf),
- "Thread id is bigger than uint64??");
+ CHECK_LT(snprintf(buf, sizeof(buf), "%llu", syscall(__NR_gettid)),
+ sizeof(buf))
+ << "Thread id is bigger than uint64??";
return std::string(buf);
}
// Return the current thread's name.
static std::string GetThreadName() {
char name[17];
- CHECK(prctl(PR_GET_NAME, name) == 0, "prctl(PR_GET_NAME) failed");
+ CHECK_EQ(0, prctl(PR_GET_NAME, name)) << "prctl(PR_GET_NAME) failed";
name[16] = '\0';
return std::string(name);
}
@@ -187,8 +188,8 @@
void* env = NULL;
jint status = g_jvm->GetEnv(&env, JNI_VERSION_1_6);
CHECK(((env != NULL) && (status == JNI_OK)) ||
- ((env == NULL) && (status == JNI_EDETACHED)),
- "Unexpected GetEnv return: " << status << ":" << env);
+ ((env == NULL) && (status == JNI_EDETACHED)))
+ << "Unexpected GetEnv return: " << status << ":" << env;
return reinterpret_cast<JNIEnv*>(env);
}
@@ -203,16 +204,16 @@
if (!GetEnv())
return;
- CHECK(GetEnv() == prev_jni_ptr,
- "Detaching from another thread: " << prev_jni_ptr << ":" << GetEnv());
+ CHECK(GetEnv() == prev_jni_ptr)
+ << "Detaching from another thread: " << prev_jni_ptr << ":" << GetEnv();
jint status = g_jvm->DetachCurrentThread();
- CHECK(status == JNI_OK, "Failed to detach thread: " << status);
- CHECK(!GetEnv(), "Detaching was a successful no-op???");
+ CHECK(status == JNI_OK) << "Failed to detach thread: " << status;
+ CHECK(!GetEnv()) << "Detaching was a successful no-op???";
}
static void CreateJNIPtrKey() {
- CHECK(!pthread_key_create(&g_jni_ptr, &ThreadDestructor),
- "pthread_key_create");
+ CHECK(!pthread_key_create(&g_jni_ptr, &ThreadDestructor))
+ << "pthread_key_create";
}
// Return a |JNIEnv*| usable on this thread. Attaches to |g_jvm| if necessary.
@@ -220,7 +221,8 @@
JNIEnv* jni = GetEnv();
if (jni)
return jni;
- CHECK(!pthread_getspecific(g_jni_ptr), "TLS has a JNIEnv* but not attached?");
+ CHECK(!pthread_getspecific(g_jni_ptr))
+ << "TLS has a JNIEnv* but not attached?";
char* name = strdup((GetThreadName() + " - " + GetThreadId()).c_str());
JavaVMAttachArgs args;
@@ -233,11 +235,11 @@
#else
JNIEnv* env = NULL;
#endif
- CHECK(!g_jvm->AttachCurrentThread(&env, &args), "Failed to attach thread");
+ CHECK(!g_jvm->AttachCurrentThread(&env, &args)) << "Failed to attach thread";
free(name);
- CHECK(env, "AttachCurrentThread handed back NULL!");
+ CHECK(env) << "AttachCurrentThread handed back NULL!";
jni = reinterpret_cast<JNIEnv*>(env);
- CHECK(!pthread_setspecific(g_jni_ptr, jni), "pthread_setspecific");
+ CHECK(!pthread_setspecific(g_jni_ptr, jni)) << "pthread_setspecific";
return jni;
}
@@ -269,10 +271,13 @@
LoadClass(jni, "org/webrtc/DataChannel$Init");
LoadClass(jni, "org/webrtc/DataChannel$State");
LoadClass(jni, "org/webrtc/IceCandidate");
-#ifdef ANDROID
+#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
+ LoadClass(jni, "android/graphics/SurfaceTexture");
+ LoadClass(jni, "android/opengl/EGLContext");
LoadClass(jni, "org/webrtc/MediaCodecVideoEncoder");
LoadClass(jni, "org/webrtc/MediaCodecVideoEncoder$OutputBufferInfo");
LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder");
+ LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder$DecoderOutputBufferInfo");
#endif
LoadClass(jni, "org/webrtc/MediaSource$State");
LoadClass(jni, "org/webrtc/MediaStream");
@@ -289,7 +294,7 @@
}
~ClassReferenceHolder() {
- CHECK(classes_.empty(), "Must call FreeReferences() before dtor!");
+ CHECK(classes_.empty()) << "Must call FreeReferences() before dtor!";
}
void FreeReferences(JNIEnv* jni) {
@@ -302,20 +307,20 @@
jclass GetClass(const std::string& name) {
std::map<std::string, jclass>::iterator it = classes_.find(name);
- CHECK(it != classes_.end(), "Unexpected GetClass() call for: " << name);
+ CHECK(it != classes_.end()) << "Unexpected GetClass() call for: " << name;
return it->second;
}
private:
void LoadClass(JNIEnv* jni, const std::string& name) {
jclass localRef = jni->FindClass(name.c_str());
- CHECK_EXCEPTION(jni, "error during FindClass: " << name);
- CHECK(localRef, name);
+ CHECK_EXCEPTION(jni) << "error during FindClass: " << name;
+ CHECK(localRef) << name;
jclass globalRef = reinterpret_cast<jclass>(jni->NewGlobalRef(localRef));
- CHECK_EXCEPTION(jni, "error during NewGlobalRef: " << name);
- CHECK(globalRef, name);
+ CHECK_EXCEPTION(jni) << "error during NewGlobalRef: " << name;
+ CHECK(globalRef) << name;
bool inserted = classes_.insert(std::make_pair(name, globalRef)).second;
- CHECK(inserted, "Duplicate class name: " << name);
+ CHECK(inserted) << "Duplicate class name: " << name;
}
std::map<std::string, jclass> classes_;
@@ -329,27 +334,26 @@
jmethodID GetMethodID(
JNIEnv* jni, jclass c, const std::string& name, const char* signature) {
jmethodID m = jni->GetMethodID(c, name.c_str(), signature);
- CHECK_EXCEPTION(jni,
- "error during GetMethodID: " << name << ", " << signature);
- CHECK(m, name << ", " << signature);
+ CHECK_EXCEPTION(jni) << "error during GetMethodID: " << name << ", "
+ << signature;
+ CHECK(m) << name << ", " << signature;
return m;
}
jmethodID GetStaticMethodID(
JNIEnv* jni, jclass c, const char* name, const char* signature) {
jmethodID m = jni->GetStaticMethodID(c, name, signature);
- CHECK_EXCEPTION(jni,
- "error during GetStaticMethodID: "
- << name << ", " << signature);
- CHECK(m, name << ", " << signature);
+ CHECK_EXCEPTION(jni) << "error during GetStaticMethodID: " << name << ", "
+ << signature;
+ CHECK(m) << name << ", " << signature;
return m;
}
jfieldID GetFieldID(
JNIEnv* jni, jclass c, const char* name, const char* signature) {
jfieldID f = jni->GetFieldID(c, name, signature);
- CHECK_EXCEPTION(jni, "error during GetFieldID");
- CHECK(f, name << ", " << signature);
+ CHECK_EXCEPTION(jni) << "error during GetFieldID";
+ CHECK(f) << name << ", " << signature;
return f;
}
@@ -361,15 +365,15 @@
jclass GetObjectClass(JNIEnv* jni, jobject object) {
jclass c = jni->GetObjectClass(object);
- CHECK_EXCEPTION(jni, "error during GetObjectClass");
- CHECK(c, "");
+ CHECK_EXCEPTION(jni) << "error during GetObjectClass";
+ CHECK(c) << "GetObjectClass returned NULL";
return c;
}
jobject GetObjectField(JNIEnv* jni, jobject object, jfieldID id) {
jobject o = jni->GetObjectField(object, id);
- CHECK_EXCEPTION(jni, "error during GetObjectField");
- CHECK(o, "");
+ CHECK_EXCEPTION(jni) << "error during GetObjectField";
+ CHECK(o) << "GetObjectField returned NULL";
return o;
}
@@ -379,32 +383,32 @@
jlong GetLongField(JNIEnv* jni, jobject object, jfieldID id) {
jlong l = jni->GetLongField(object, id);
- CHECK_EXCEPTION(jni, "error during GetLongField");
+ CHECK_EXCEPTION(jni) << "error during GetLongField";
return l;
}
jint GetIntField(JNIEnv* jni, jobject object, jfieldID id) {
jint i = jni->GetIntField(object, id);
- CHECK_EXCEPTION(jni, "error during GetIntField");
+ CHECK_EXCEPTION(jni) << "error during GetIntField";
return i;
}
bool GetBooleanField(JNIEnv* jni, jobject object, jfieldID id) {
jboolean b = jni->GetBooleanField(object, id);
- CHECK_EXCEPTION(jni, "error during GetBooleanField");
+ CHECK_EXCEPTION(jni) << "error during GetBooleanField";
return b;
}
jobject NewGlobalRef(JNIEnv* jni, jobject o) {
jobject ret = jni->NewGlobalRef(o);
- CHECK_EXCEPTION(jni, "error during NewGlobalRef");
- CHECK(ret, "");
+ CHECK_EXCEPTION(jni) << "error during NewGlobalRef";
+ CHECK(ret);
return ret;
}
void DeleteGlobalRef(JNIEnv* jni, jobject o) {
jni->DeleteGlobalRef(o);
- CHECK_EXCEPTION(jni, "error during DeleteGlobalRef");
+ CHECK_EXCEPTION(jni) << "error during DeleteGlobalRef";
}
// Given a jweak reference, allocate a (strong) local reference scoped to the
@@ -414,12 +418,12 @@
public:
WeakRef(JNIEnv* jni, jweak ref)
: jni_(jni), obj_(jni_->NewLocalRef(ref)) {
- CHECK_EXCEPTION(jni, "error during NewLocalRef");
+ CHECK_EXCEPTION(jni) << "error during NewLocalRef";
}
~WeakRef() {
if (obj_) {
jni_->DeleteLocalRef(obj_);
- CHECK_EXCEPTION(jni_, "error during DeleteLocalRef");
+ CHECK_EXCEPTION(jni_) << "error during DeleteLocalRef";
}
}
jobject obj() { return obj_; }
@@ -435,7 +439,7 @@
class ScopedLocalRefFrame {
public:
explicit ScopedLocalRefFrame(JNIEnv* jni) : jni_(jni) {
- CHECK(!jni_->PushLocalFrame(0), "Failed to PushLocalFrame");
+ CHECK(!jni_->PushLocalFrame(0)) << "Failed to PushLocalFrame";
}
~ScopedLocalRefFrame() {
jni_->PopLocalFrame(NULL);
@@ -478,9 +482,9 @@
jni, state_class, "values", ("()[L" + state_class_name + ";").c_str());
jobjectArray state_values = static_cast<jobjectArray>(
jni->CallStaticObjectMethod(state_class, state_values_id));
- CHECK_EXCEPTION(jni, "error during CallStaticObjectMethod");
+ CHECK_EXCEPTION(jni) << "error during CallStaticObjectMethod";
jobject ret = jni->GetObjectArrayElement(state_values, index);
- CHECK_EXCEPTION(jni, "error during GetObjectArrayElement");
+ CHECK_EXCEPTION(jni) << "error during GetObjectArrayElement";
return ret;
}
@@ -488,18 +492,18 @@
static jstring JavaStringFromStdString(JNIEnv* jni, const std::string& native) {
UnicodeString ustr(UnicodeString::fromUTF8(native));
jstring jstr = jni->NewString(ustr.getBuffer(), ustr.length());
- CHECK_EXCEPTION(jni, "error during NewString");
+ CHECK_EXCEPTION(jni) << "error during NewString";
return jstr;
}
// Given a (UTF-16) jstring return a new UTF-8 native string.
static std::string JavaToStdString(JNIEnv* jni, const jstring& j_string) {
const jchar* jchars = jni->GetStringChars(j_string, NULL);
- CHECK_EXCEPTION(jni, "Error during GetStringChars");
+ CHECK_EXCEPTION(jni) << "Error during GetStringChars";
UnicodeString ustr(jchars, jni->GetStringLength(j_string));
- CHECK_EXCEPTION(jni, "Error during GetStringLength");
+ CHECK_EXCEPTION(jni) << "Error during GetStringLength";
jni->ReleaseStringChars(j_string, jchars);
- CHECK_EXCEPTION(jni, "Error during ReleaseStringChars");
+ CHECK_EXCEPTION(jni) << "Error during ReleaseStringChars";
std::string ret;
return ustr.toUTF8String(ret);
}
@@ -559,7 +563,7 @@
virtual void OnIceCandidate(const IceCandidateInterface* candidate) OVERRIDE {
ScopedLocalRefFrame local_ref_frame(jni());
std::string sdp;
- CHECK(candidate->ToString(&sdp), "got so far: " << sdp);
+ CHECK(candidate->ToString(&sdp)) << "got so far: " << sdp;
jclass candidate_class = FindClass(jni(), "org/webrtc/IceCandidate");
jmethodID ctor = GetMethodID(jni(), candidate_class,
"<init>", "(Ljava/lang/String;ILjava/lang/String;)V");
@@ -567,18 +571,18 @@
jstring j_sdp = JavaStringFromStdString(jni(), sdp);
jobject j_candidate = jni()->NewObject(
candidate_class, ctor, j_mid, candidate->sdp_mline_index(), j_sdp);
- CHECK_EXCEPTION(jni(), "error during NewObject");
+ CHECK_EXCEPTION(jni()) << "error during NewObject";
jmethodID m = GetMethodID(jni(), *j_observer_class_,
"onIceCandidate", "(Lorg/webrtc/IceCandidate;)V");
jni()->CallVoidMethod(*j_observer_global_, m, j_candidate);
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
virtual void OnError() OVERRIDE {
ScopedLocalRefFrame local_ref_frame(jni());
jmethodID m = GetMethodID(jni(), *j_observer_class_, "onError", "()V");
jni()->CallVoidMethod(*j_observer_global_, m);
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
virtual void OnSignalingChange(
@@ -590,7 +594,7 @@
jobject new_state_enum =
JavaEnumFromIndex(jni(), "PeerConnection$SignalingState", new_state);
jni()->CallVoidMethod(*j_observer_global_, m, new_state_enum);
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
virtual void OnIceConnectionChange(
@@ -602,7 +606,7 @@
jobject new_state_enum = JavaEnumFromIndex(
jni(), "PeerConnection$IceConnectionState", new_state);
jni()->CallVoidMethod(*j_observer_global_, m, new_state_enum);
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
virtual void OnIceGatheringChange(
@@ -614,14 +618,14 @@
jobject new_state_enum = JavaEnumFromIndex(
jni(), "PeerConnection$IceGatheringState", new_state);
jni()->CallVoidMethod(*j_observer_global_, m, new_state_enum);
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
virtual void OnAddStream(MediaStreamInterface* stream) OVERRIDE {
ScopedLocalRefFrame local_ref_frame(jni());
jobject j_stream = jni()->NewObject(
*j_media_stream_class_, j_media_stream_ctor_, (jlong)stream);
- CHECK_EXCEPTION(jni(), "error during NewObject");
+ CHECK_EXCEPTION(jni()) << "error during NewObject";
AudioTrackVector audio_tracks = stream->GetAudioTracks();
for (size_t i = 0; i < audio_tracks.size(); ++i) {
@@ -629,7 +633,7 @@
jstring id = JavaStringFromStdString(jni(), track->id());
jobject j_track = jni()->NewObject(
*j_audio_track_class_, j_audio_track_ctor_, (jlong)track, id);
- CHECK_EXCEPTION(jni(), "error during NewObject");
+ CHECK_EXCEPTION(jni()) << "error during NewObject";
jfieldID audio_tracks_id = GetFieldID(jni(),
*j_media_stream_class_,
"audioTracks",
@@ -640,8 +644,8 @@
"add",
"(Ljava/lang/Object;)Z");
jboolean added = jni()->CallBooleanMethod(audio_tracks, add, j_track);
- CHECK_EXCEPTION(jni(), "error during CallBooleanMethod");
- CHECK(added, "");
+ CHECK_EXCEPTION(jni()) << "error during CallBooleanMethod";
+ CHECK(added);
}
VideoTrackVector video_tracks = stream->GetVideoTracks();
@@ -650,7 +654,7 @@
jstring id = JavaStringFromStdString(jni(), track->id());
jobject j_track = jni()->NewObject(
*j_video_track_class_, j_video_track_ctor_, (jlong)track, id);
- CHECK_EXCEPTION(jni(), "error during NewObject");
+ CHECK_EXCEPTION(jni()) << "error during NewObject";
jfieldID video_tracks_id = GetFieldID(jni(),
*j_media_stream_class_,
"videoTracks",
@@ -661,22 +665,22 @@
"add",
"(Ljava/lang/Object;)Z");
jboolean added = jni()->CallBooleanMethod(video_tracks, add, j_track);
- CHECK_EXCEPTION(jni(), "error during CallBooleanMethod");
- CHECK(added, "");
+ CHECK_EXCEPTION(jni()) << "error during CallBooleanMethod";
+ CHECK(added);
}
streams_[stream] = jni()->NewWeakGlobalRef(j_stream);
- CHECK_EXCEPTION(jni(), "error during NewWeakGlobalRef");
+ CHECK_EXCEPTION(jni()) << "error during NewWeakGlobalRef";
jmethodID m = GetMethodID(jni(), *j_observer_class_, "onAddStream",
"(Lorg/webrtc/MediaStream;)V");
jni()->CallVoidMethod(*j_observer_global_, m, j_stream);
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
virtual void OnRemoveStream(MediaStreamInterface* stream) OVERRIDE {
ScopedLocalRefFrame local_ref_frame(jni());
NativeToJavaStreamsMap::iterator it = streams_.find(stream);
- CHECK(it != streams_.end(), "unexpected stream: " << std::hex << stream);
+ CHECK(it != streams_.end()) << "unexpected stream: " << std::hex << stream;
WeakRef s(jni(), it->second);
streams_.erase(it);
@@ -686,14 +690,14 @@
jmethodID m = GetMethodID(jni(), *j_observer_class_, "onRemoveStream",
"(Lorg/webrtc/MediaStream;)V");
jni()->CallVoidMethod(*j_observer_global_, m, s.obj());
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
virtual void OnDataChannel(DataChannelInterface* channel) OVERRIDE {
ScopedLocalRefFrame local_ref_frame(jni());
jobject j_channel = jni()->NewObject(
*j_data_channel_class_, j_data_channel_ctor_, (jlong)channel);
- CHECK_EXCEPTION(jni(), "error during NewObject");
+ CHECK_EXCEPTION(jni()) << "error during NewObject";
jmethodID m = GetMethodID(jni(), *j_observer_class_, "onDataChannel",
"(Lorg/webrtc/DataChannel;)V");
@@ -704,9 +708,9 @@
// CallVoidMethod above as Java code might call back into native code and be
// surprised to see a refcount of 2.
int bumped_count = channel->AddRef();
- CHECK(bumped_count == 2, "Unexpected refcount OnDataChannel");
+ CHECK(bumped_count == 2) << "Unexpected refcount OnDataChannel";
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
virtual void OnRenegotiationNeeded() OVERRIDE {
@@ -714,11 +718,11 @@
jmethodID m =
GetMethodID(jni(), *j_observer_class_, "onRenegotiationNeeded", "()V");
jni()->CallVoidMethod(*j_observer_global_, m);
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
void SetConstraints(ConstraintsWrapper* constraints) {
- CHECK(!constraints_.get(), "constraints already set!");
+ CHECK(!constraints_.get()) << "constraints already set!";
constraints_.reset(constraints);
}
@@ -777,29 +781,29 @@
jmethodID j_iterator_id = GetMethodID(jni,
GetObjectClass(jni, j_list), "iterator", "()Ljava/util/Iterator;");
jobject j_iterator = jni->CallObjectMethod(j_list, j_iterator_id);
- CHECK_EXCEPTION(jni, "error during CallObjectMethod");
+ CHECK_EXCEPTION(jni) << "error during CallObjectMethod";
jmethodID j_has_next = GetMethodID(jni,
GetObjectClass(jni, j_iterator), "hasNext", "()Z");
jmethodID j_next = GetMethodID(jni,
GetObjectClass(jni, j_iterator), "next", "()Ljava/lang/Object;");
while (jni->CallBooleanMethod(j_iterator, j_has_next)) {
- CHECK_EXCEPTION(jni, "error during CallBooleanMethod");
+ CHECK_EXCEPTION(jni) << "error during CallBooleanMethod";
jobject entry = jni->CallObjectMethod(j_iterator, j_next);
- CHECK_EXCEPTION(jni, "error during CallObjectMethod");
+ CHECK_EXCEPTION(jni) << "error during CallObjectMethod";
jmethodID get_key = GetMethodID(jni,
GetObjectClass(jni, entry), "getKey", "()Ljava/lang/String;");
jstring j_key = reinterpret_cast<jstring>(
jni->CallObjectMethod(entry, get_key));
- CHECK_EXCEPTION(jni, "error during CallObjectMethod");
+ CHECK_EXCEPTION(jni) << "error during CallObjectMethod";
jmethodID get_value = GetMethodID(jni,
GetObjectClass(jni, entry), "getValue", "()Ljava/lang/String;");
jstring j_value = reinterpret_cast<jstring>(
jni->CallObjectMethod(entry, get_value));
- CHECK_EXCEPTION(jni, "error during CallObjectMethod");
+ CHECK_EXCEPTION(jni) << "error during CallObjectMethod";
field->push_back(Constraint(JavaToStdString(jni, j_key),
JavaToStdString(jni, j_value)));
}
- CHECK_EXCEPTION(jni, "error during CallBooleanMethod");
+ CHECK_EXCEPTION(jni) << "error during CallBooleanMethod";
}
Constraints mandatory_;
@@ -809,7 +813,7 @@
static jobject JavaSdpFromNativeSdp(
JNIEnv* jni, const SessionDescriptionInterface* desc) {
std::string sdp;
- CHECK(desc->ToString(&sdp), "got so far: " << sdp);
+ CHECK(desc->ToString(&sdp)) << "got so far: " << sdp;
jstring j_description = JavaStringFromStdString(jni, sdp);
jclass j_type_class = FindClass(
@@ -820,7 +824,7 @@
jstring j_type_string = JavaStringFromStdString(jni, desc->type());
jobject j_type = jni->CallStaticObjectMethod(
j_type_class, j_type_from_canonical, j_type_string);
- CHECK_EXCEPTION(jni, "error during CallObjectMethod");
+ CHECK_EXCEPTION(jni) << "error during CallObjectMethod";
jclass j_sdp_class = FindClass(jni, "org/webrtc/SessionDescription");
jmethodID j_sdp_ctor = GetMethodID(
@@ -828,7 +832,7 @@
"(Lorg/webrtc/SessionDescription$Type;Ljava/lang/String;)V");
jobject j_sdp = jni->NewObject(
j_sdp_class, j_sdp_ctor, j_type, j_description);
- CHECK_EXCEPTION(jni, "error during NewObject");
+ CHECK_EXCEPTION(jni) << "error during NewObject";
return j_sdp;
}
@@ -849,7 +853,7 @@
ScopedLocalRefFrame local_ref_frame(jni());
jmethodID m = GetMethodID(jni(), *j_observer_class_, "onSetSuccess", "()V");
jni()->CallVoidMethod(*j_observer_global_, m);
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
// Can't mark OVERRIDE because of templating.
@@ -860,7 +864,7 @@
"(Lorg/webrtc/SessionDescription;)V");
jobject j_sdp = JavaSdpFromNativeSdp(jni(), desc);
jni()->CallVoidMethod(*j_observer_global_, m, j_sdp);
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
protected:
@@ -871,7 +875,7 @@
"(Ljava/lang/String;)V");
jstring j_error_string = JavaStringFromStdString(jni(), error);
jni()->CallVoidMethod(*j_observer_global_, m, j_error_string);
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
JNIEnv* jni() {
@@ -917,11 +921,11 @@
DataChannelObserverWrapper(JNIEnv* jni, jobject j_observer)
: j_observer_global_(jni, j_observer),
j_observer_class_(jni, GetObjectClass(jni, j_observer)),
+ j_buffer_class_(jni, FindClass(jni, "org/webrtc/DataChannel$Buffer")),
j_on_state_change_mid_(GetMethodID(jni, *j_observer_class_,
"onStateChange", "()V")),
j_on_message_mid_(GetMethodID(jni, *j_observer_class_, "onMessage",
"(Lorg/webrtc/DataChannel$Buffer;)V")),
- j_buffer_class_(jni, FindClass(jni, "org/webrtc/DataChannel$Buffer")),
j_buffer_ctor_(GetMethodID(jni, *j_buffer_class_,
"<init>", "(Ljava/nio/ByteBuffer;Z)V")) {
}
@@ -931,7 +935,7 @@
virtual void OnStateChange() OVERRIDE {
ScopedLocalRefFrame local_ref_frame(jni());
jni()->CallVoidMethod(*j_observer_global_, j_on_state_change_mid_);
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
virtual void OnMessage(const DataBuffer& buffer) OVERRIDE {
@@ -942,7 +946,7 @@
jobject j_buffer = jni()->NewObject(*j_buffer_class_, j_buffer_ctor_,
byte_buffer, buffer.binary);
jni()->CallVoidMethod(*j_observer_global_, j_on_message_mid_, j_buffer);
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
private:
@@ -985,7 +989,7 @@
jmethodID m = GetMethodID(jni(), *j_observer_class_, "onComplete",
"([Lorg/webrtc/StatsReport;)V");
jni()->CallVoidMethod(*j_observer_global_, m, j_reports);
- CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
+ CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
}
private:
@@ -1067,6 +1071,38 @@
scoped_ptr<cricket::VideoRenderer> renderer_;
};
+// Wrapper for texture object in TextureVideoFrame.
+class NativeHandleImpl : public NativeHandle {
+ public:
+ NativeHandleImpl() :
+ ref_count_(0), texture_object_(NULL), texture_id_(-1) {}
+ virtual ~NativeHandleImpl() {}
+ virtual int32_t AddRef() {
+ return ++ref_count_;
+ }
+ virtual int32_t Release() {
+ return --ref_count_;
+ }
+ virtual void* GetHandle() {
+ return texture_object_;
+ }
+ int GetTextureId() {
+ return texture_id_;
+ }
+ void SetTextureObject(void *texture_object, int texture_id) {
+ texture_object_ = reinterpret_cast<jobject>(texture_object);
+ texture_id_ = texture_id;
+ }
+ int32_t ref_count() {
+ return ref_count_;
+ }
+
+ private:
+ int32_t ref_count_;
+ jobject texture_object_;
+ int32_t texture_id_;
+};
+
// Wrapper dispatching webrtc::VideoRendererInterface to a Java VideoRenderer
// instance.
class JavaVideoRendererWrapper : public VideoRendererInterface {
@@ -1080,10 +1116,13 @@
"(Lorg/webrtc/VideoRenderer$I420Frame;)V")),
j_frame_class_(jni,
FindClass(jni, "org/webrtc/VideoRenderer$I420Frame")),
- j_frame_ctor_id_(GetMethodID(
+ j_i420_frame_ctor_id_(GetMethodID(
jni, *j_frame_class_, "<init>", "(II[I[Ljava/nio/ByteBuffer;)V")),
+ j_texture_frame_ctor_id_(GetMethodID(
+ jni, *j_frame_class_, "<init>",
+ "(IILjava/lang/Object;I)V")),
j_byte_buffer_class_(jni, FindClass(jni, "java/nio/ByteBuffer")) {
- CHECK_EXCEPTION(jni, "");
+ CHECK_EXCEPTION(jni);
}
virtual ~JavaVideoRendererWrapper() {}
@@ -1091,19 +1130,25 @@
virtual void SetSize(int width, int height) OVERRIDE {
ScopedLocalRefFrame local_ref_frame(jni());
jni()->CallVoidMethod(*j_callbacks_, j_set_size_id_, width, height);
- CHECK_EXCEPTION(jni(), "");
+ CHECK_EXCEPTION(jni());
}
virtual void RenderFrame(const cricket::VideoFrame* frame) OVERRIDE {
ScopedLocalRefFrame local_ref_frame(jni());
- jobject j_frame = CricketToJavaFrame(frame);
- jni()->CallVoidMethod(*j_callbacks_, j_render_frame_id_, j_frame);
- CHECK_EXCEPTION(jni(), "");
+ if (frame->GetNativeHandle() != NULL) {
+ jobject j_frame = CricketToJavaTextureFrame(frame);
+ jni()->CallVoidMethod(*j_callbacks_, j_render_frame_id_, j_frame);
+ CHECK_EXCEPTION(jni());
+ } else {
+ jobject j_frame = CricketToJavaI420Frame(frame);
+ jni()->CallVoidMethod(*j_callbacks_, j_render_frame_id_, j_frame);
+ CHECK_EXCEPTION(jni());
+ }
}
private:
// Return a VideoRenderer.I420Frame referring to the data in |frame|.
- jobject CricketToJavaFrame(const cricket::VideoFrame* frame) {
+ jobject CricketToJavaI420Frame(const cricket::VideoFrame* frame) {
jintArray strides = jni()->NewIntArray(3);
jint* strides_array = jni()->GetIntArrayElements(strides, NULL);
strides_array[0] = frame->GetYPitch();
@@ -1122,10 +1167,21 @@
jni()->SetObjectArrayElement(planes, 1, u_buffer);
jni()->SetObjectArrayElement(planes, 2, v_buffer);
return jni()->NewObject(
- *j_frame_class_, j_frame_ctor_id_,
+ *j_frame_class_, j_i420_frame_ctor_id_,
frame->GetWidth(), frame->GetHeight(), strides, planes);
}
+ // Return a VideoRenderer.I420Frame referring texture object in |frame|.
+ jobject CricketToJavaTextureFrame(const cricket::VideoFrame* frame) {
+ NativeHandleImpl* handle =
+ reinterpret_cast<NativeHandleImpl*>(frame->GetNativeHandle());
+ jobject texture_object = reinterpret_cast<jobject>(handle->GetHandle());
+ int texture_id = handle->GetTextureId();
+ return jni()->NewObject(
+ *j_frame_class_, j_texture_frame_ctor_id_,
+ frame->GetWidth(), frame->GetHeight(), texture_object, texture_id);
+ }
+
JNIEnv* jni() {
return AttachCurrentThreadIfNeeded();
}
@@ -1134,16 +1190,16 @@
jmethodID j_set_size_id_;
jmethodID j_render_frame_id_;
ScopedGlobalRef<jclass> j_frame_class_;
- jmethodID j_frame_ctor_id_;
+ jmethodID j_i420_frame_ctor_id_;
+ jmethodID j_texture_frame_ctor_id_;
ScopedGlobalRef<jclass> j_byte_buffer_class_;
};
-#ifdef ANDROID
+#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
// TODO(fischman): consider pulling MediaCodecVideoEncoder out of this file and
// into its own .h/.cc pair, if/when the JNI helper stuff above is extracted
// from this file.
-#include <android/log.h>
//#define TRACK_BUFFER_TIMING
#define TAG "MediaCodecVideo"
#ifdef TRACK_BUFFER_TIMING
@@ -1169,6 +1225,14 @@
// Arbitrary interval to poll the codec for new outputs.
enum { kMediaCodecPollMs = 10 };
+// Media codec maximum output buffer ready timeout.
+enum { kMediaCodecTimeoutMs = 500 };
+// Interval to print codec statistics (bitrate, fps, encoding/decoding time).
+enum { kMediaCodecStatisticsIntervalMs = 3000 };
+
+static int64_t GetCurrentTimeMs() {
+ return TickTime::Now().Ticks() / 1000000LL;
+}
// MediaCodecVideoEncoder is a webrtc::VideoEncoder implementation that uses
// Android's MediaCodec SDK API behind the scenes to implement (hopefully)
@@ -1224,9 +1288,6 @@
int32_t ReleaseOnCodecThread();
int32_t SetRatesOnCodecThread(uint32_t new_bit_rate, uint32_t frame_rate);
- // Reset parameters valid between InitEncode() & Release() (see below).
- void ResetParameters(JNIEnv* jni);
-
// Helper accessors for MediaCodecVideoEncoder$OutputBufferInfo members.
int GetOutputBufferInfoIndex(JNIEnv* jni, jobject j_output_buffer_info);
jobject GetOutputBufferInfoBuffer(JNIEnv* jni, jobject j_output_buffer_info);
@@ -1269,11 +1330,21 @@
enum libyuv::FourCC encoder_fourcc_; // Encoder color space format.
int last_set_bitrate_kbps_; // Last-requested bitrate in kbps.
int last_set_fps_; // Last-requested frame rate.
- int frames_received_; // Number of frames received by encoder.
- int frames_dropped_; // Number of frames dropped by encoder.
- int frames_in_queue_; // Number of frames in encoder queue.
- int64_t last_input_timestamp_ms_; // Timestamp of last received yuv frame.
- int64_t last_output_timestamp_ms_; // Timestamp of last encoded frame.
+ int64_t current_timestamp_us_; // Current frame timestamps in us.
+ int frames_received_; // Number of frames received by encoder.
+ int frames_dropped_; // Number of frames dropped by encoder.
+ int frames_resolution_update_; // Number of frames with new codec resolution.
+ int frames_in_queue_; // Number of frames in encoder queue.
+ int64_t start_time_ms_; // Start time for statistics.
+ int current_frames_; // Number of frames in the current statistics interval.
+ int current_bytes_; // Encoded bytes in the current statistics interval.
+ int current_encoding_time_ms_; // Overall encoding time in the current second
+ int64_t last_input_timestamp_ms_; // Timestamp of last received yuv frame.
+ int64_t last_output_timestamp_ms_; // Timestamp of last encoded frame.
+ std::vector<int32_t> timestamps_; // Video frames timestamp queue.
+ std::vector<int64_t> render_times_ms_; // Video frames render time queue.
+ std::vector<int64_t> frame_rtc_times_ms_; // Time when video frame is sent to
+ // encoder input.
// Frame size in bytes fed to MediaCodec.
int yuv_size_;
// True only when between a callback_->Encoded() call return a positive value
@@ -1284,24 +1355,24 @@
};
MediaCodecVideoEncoder::~MediaCodecVideoEncoder() {
- // We depend on ResetParameters() to ensure no more callbacks to us after we
- // are deleted, so assert it here.
- CHECK(width_ == 0, "Release() should have been called");
+ // Call Release() to ensure no more callbacks to us after we are deleted.
+ Release();
}
MediaCodecVideoEncoder::MediaCodecVideoEncoder(JNIEnv* jni)
- : callback_(NULL),
- codec_thread_(new Thread()),
- j_media_codec_video_encoder_class_(
- jni,
- FindClass(jni, "org/webrtc/MediaCodecVideoEncoder")),
- j_media_codec_video_encoder_(
- jni,
- jni->NewObject(*j_media_codec_video_encoder_class_,
- GetMethodID(jni,
- *j_media_codec_video_encoder_class_,
- "<init>",
- "()V"))) {
+ : callback_(NULL),
+ inited_(false),
+ codec_thread_(new Thread()),
+ j_media_codec_video_encoder_class_(
+ jni,
+ FindClass(jni, "org/webrtc/MediaCodecVideoEncoder")),
+ j_media_codec_video_encoder_(
+ jni,
+ jni->NewObject(*j_media_codec_video_encoder_class_,
+ GetMethodID(jni,
+ *j_media_codec_video_encoder_class_,
+ "<init>",
+ "()V"))) {
ScopedLocalRefFrame local_ref_frame(jni);
// It would be nice to avoid spinning up a new thread per MediaCodec, and
// instead re-use e.g. the PeerConnectionFactory's |worker_thread_|, but bug
@@ -1311,9 +1382,7 @@
// in the bug, we have a problem. For now work around that with a dedicated
// thread.
codec_thread_->SetName("MediaCodecVideoEncoder", NULL);
- CHECK(codec_thread_->Start(), "Failed to start MediaCodecVideoEncoder");
-
- ResetParameters(jni);
+ CHECK(codec_thread_->Start()) << "Failed to start MediaCodecVideoEncoder";
jclass j_output_buffer_info_class =
FindClass(jni, "org/webrtc/MediaCodecVideoEncoder$OutputBufferInfo");
@@ -1347,7 +1416,7 @@
GetFieldID(jni, j_output_buffer_info_class, "isKeyFrame", "Z");
j_info_presentation_timestamp_us_field_ = GetFieldID(
jni, j_output_buffer_info_class, "presentationTimestampUs", "J");
- CHECK_EXCEPTION(jni, "MediaCodecVideoEncoder ctor failed");
+ CHECK_EXCEPTION(jni) << "MediaCodecVideoEncoder ctor failed";
}
int32_t MediaCodecVideoEncoder::InitEncode(
@@ -1355,7 +1424,7 @@
int32_t /* number_of_cores */,
uint32_t /* max_payload_size */) {
// Factory should guard against other codecs being used with us.
- CHECK(codec_settings->codecType == kVideoCodecVP8, "Unsupported codec");
+ CHECK(codec_settings->codecType == kVideoCodecVP8) << "Unsupported codec";
return codec_thread_->Invoke<int32_t>(
Bind(&MediaCodecVideoEncoder::InitEncodeOnCodecThread,
@@ -1407,9 +1476,12 @@
// We only ever send one message to |this| directly (not through a Bind()'d
// functor), so expect no ID/data.
- CHECK(!msg->message_id, "Unexpected message!");
- CHECK(!msg->pdata, "Unexpected message!");
+ CHECK(!msg->message_id) << "Unexpected message!";
+ CHECK(!msg->pdata) << "Unexpected message!";
CheckOnCodecThread();
+ if (!inited_) {
+ return;
+ }
// It would be nice to recover from a failure here if one happened, but it's
// unclear how to signal such a failure to the app, so instead we stay silent
@@ -1419,16 +1491,16 @@
}
void MediaCodecVideoEncoder::CheckOnCodecThread() {
- CHECK(codec_thread_ == ThreadManager::Instance()->CurrentThread(),
- "Running on wrong thread!");
+ CHECK(codec_thread_ == ThreadManager::Instance()->CurrentThread())
+ << "Running on wrong thread!";
}
void MediaCodecVideoEncoder::ResetCodec() {
ALOGE("ResetCodec");
if (Release() != WEBRTC_VIDEO_CODEC_OK ||
codec_thread_->Invoke<int32_t>(Bind(
- &MediaCodecVideoEncoder::InitEncodeOnCodecThread, this, 0, 0, 0, 0))
- != WEBRTC_VIDEO_CODEC_OK) {
+ &MediaCodecVideoEncoder::InitEncodeOnCodecThread, this,
+ width_, height_, 0, 0)) != WEBRTC_VIDEO_CODEC_OK) {
// TODO(fischman): wouldn't it be nice if there was a way to gracefully
// degrade to a SW encoder at this point? There isn't one AFAICT :(
// https://code.google.com/p/webrtc/issues/detail?id=2920
@@ -1440,12 +1512,13 @@
CheckOnCodecThread();
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedLocalRefFrame local_ref_frame(jni);
- ALOGD("InitEncodeOnCodecThread %d x %d", width, height);
- if (width == 0) {
- width = width_;
- height = height_;
+ ALOGD("InitEncodeOnCodecThread %d x %d. Bitrate: %d kbps. Fps: %d",
+ width, height, kbps, fps);
+ if (kbps == 0) {
kbps = last_set_bitrate_kbps_;
+ }
+ if (fps == 0) {
fps = last_set_fps_;
}
@@ -1456,9 +1529,19 @@
yuv_size_ = width_ * height_ * 3 / 2;
frames_received_ = 0;
frames_dropped_ = 0;
+ frames_resolution_update_ = 0;
frames_in_queue_ = 0;
+ current_timestamp_us_ = 0;
+ start_time_ms_ = GetCurrentTimeMs();
+ current_frames_ = 0;
+ current_bytes_ = 0;
+ current_encoding_time_ms_ = 0;
last_input_timestamp_ms_ = -1;
last_output_timestamp_ms_ = -1;
+ timestamps_.clear();
+ render_times_ms_.clear();
+ frame_rtc_times_ms_.clear();
+ drop_next_input_frame_ = false;
// We enforce no extra stride/padding in the format creation step.
jobjectArray input_buffers = reinterpret_cast<jobjectArray>(
jni->CallObjectMethod(*j_media_codec_video_encoder_,
@@ -1467,7 +1550,7 @@
height_,
kbps,
fps));
- CHECK_EXCEPTION(jni, "");
+ CHECK_EXCEPTION(jni);
if (IsNull(jni, input_buffers))
return WEBRTC_VIDEO_CODEC_ERROR;
@@ -1487,17 +1570,18 @@
return WEBRTC_VIDEO_CODEC_ERROR;
}
size_t num_input_buffers = jni->GetArrayLength(input_buffers);
- CHECK(input_buffers_.empty(), "Unexpected double InitEncode without Release");
+ CHECK(input_buffers_.empty())
+ << "Unexpected double InitEncode without Release";
input_buffers_.resize(num_input_buffers);
for (size_t i = 0; i < num_input_buffers; ++i) {
input_buffers_[i] =
jni->NewGlobalRef(jni->GetObjectArrayElement(input_buffers, i));
int64 yuv_buffer_capacity =
jni->GetDirectBufferCapacity(input_buffers_[i]);
- CHECK_EXCEPTION(jni, "");
- CHECK(yuv_buffer_capacity >= yuv_size_, "Insufficient capacity");
+ CHECK_EXCEPTION(jni);
+ CHECK(yuv_buffer_capacity >= yuv_size_) << "Insufficient capacity";
}
- CHECK_EXCEPTION(jni, "");
+ CHECK_EXCEPTION(jni);
codec_thread_->PostDelayed(kMediaCodecPollMs, this);
return WEBRTC_VIDEO_CODEC_OK;
@@ -1510,6 +1594,9 @@
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedLocalRefFrame local_ref_frame(jni);
+ if (!inited_) {
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
frames_received_++;
if (!DeliverPendingOutputs(jni)) {
ResetCodec();
@@ -1517,23 +1604,35 @@
}
if (drop_next_input_frame_) {
+ ALOGV("Encoder drop frame - failed callback.");
drop_next_input_frame_ = false;
return WEBRTC_VIDEO_CODEC_OK;
}
- CHECK(frame_types->size() == 1, "Unexpected stream count");
+ CHECK(frame_types->size() == 1) << "Unexpected stream count";
+ if (frame.width() != width_ || frame.height() != height_) {
+ frames_resolution_update_++;
+ ALOGD("Unexpected frame resolution change from %d x %d to %d x %d",
+ width_, height_, frame.width(), frame.height());
+ if (frames_resolution_update_ > 3) {
+ // Reset codec if we received more than 3 frames with new resolution.
+ width_ = frame.width();
+ height_ = frame.height();
+ frames_resolution_update_ = 0;
+ ResetCodec();
+ }
+ return WEBRTC_VIDEO_CODEC_OK;
+ }
+ frames_resolution_update_ = 0;
+
bool key_frame = frame_types->front() != webrtc::kDeltaFrame;
- CHECK(frame.width() == width_, "Unexpected resolution change");
- CHECK(frame.height() == height_, "Unexpected resolution change");
-
// Check if we accumulated too many frames in encoder input buffers
- // so the encoder latency exceeds 100ms and drop frame if so.
- if (frames_in_queue_ > 0 && last_input_timestamp_ms_ > 0 &&
- last_output_timestamp_ms_ > 0) {
+ // or the encoder latency exceeds 70 ms and drop frame if so.
+ if (frames_in_queue_ > 0 && last_input_timestamp_ms_ >= 0) {
int encoder_latency_ms = last_input_timestamp_ms_ -
last_output_timestamp_ms_;
- if (encoder_latency_ms > 100) {
+ if (frames_in_queue_ > 2 || encoder_latency_ms > 70) {
ALOGV("Drop frame - encoder is behind by %d ms. Q size: %d",
encoder_latency_ms, frames_in_queue_);
frames_dropped_++;
@@ -1543,10 +1642,10 @@
int j_input_buffer_index = jni->CallIntMethod(*j_media_codec_video_encoder_,
j_dequeue_input_buffer_method_);
- CHECK_EXCEPTION(jni, "");
+ CHECK_EXCEPTION(jni);
if (j_input_buffer_index == -1) {
// Video codec falls behind - no input buffer available.
- ALOGV("Drop frame - no input buffers available");
+ ALOGV("Encoder drop frame - no input buffers available");
frames_dropped_++;
return WEBRTC_VIDEO_CODEC_OK; // TODO(fischman): see webrtc bug 2887.
}
@@ -1556,31 +1655,38 @@
}
ALOGV("Encode frame # %d. Buffer # %d. TS: %lld.",
- frames_received_, j_input_buffer_index, frame.render_time_ms());
+ frames_received_, j_input_buffer_index, current_timestamp_us_ / 1000);
jobject j_input_buffer = input_buffers_[j_input_buffer_index];
uint8* yuv_buffer =
reinterpret_cast<uint8*>(jni->GetDirectBufferAddress(j_input_buffer));
- CHECK_EXCEPTION(jni, "");
- CHECK(yuv_buffer, "Indirect buffer??");
+ CHECK_EXCEPTION(jni);
+ CHECK(yuv_buffer) << "Indirect buffer??";
CHECK(!libyuv::ConvertFromI420(
frame.buffer(webrtc::kYPlane), frame.stride(webrtc::kYPlane),
frame.buffer(webrtc::kUPlane), frame.stride(webrtc::kUPlane),
frame.buffer(webrtc::kVPlane), frame.stride(webrtc::kVPlane),
yuv_buffer, width_,
width_, height_,
- encoder_fourcc_),
- "ConvertFromI420 failed");
- jlong timestamp_us = frame.render_time_ms() * 1000;
- last_input_timestamp_ms_ = frame.render_time_ms();
+ encoder_fourcc_))
+ << "ConvertFromI420 failed";
+ last_input_timestamp_ms_ = current_timestamp_us_ / 1000;
frames_in_queue_++;
+
+ // Save input image timestamps for later output
+ timestamps_.push_back(frame.timestamp());
+ render_times_ms_.push_back(frame.render_time_ms());
+ frame_rtc_times_ms_.push_back(GetCurrentTimeMs());
+
bool encode_status = jni->CallBooleanMethod(*j_media_codec_video_encoder_,
j_encode_method_,
key_frame,
j_input_buffer_index,
yuv_size_,
- timestamp_us);
- CHECK_EXCEPTION(jni, "");
+ current_timestamp_us_);
+ CHECK_EXCEPTION(jni);
+ current_timestamp_us_ += 1000000 / last_set_fps_;
+
if (!encode_status || !DeliverPendingOutputs(jni)) {
ResetCodec();
return WEBRTC_VIDEO_CODEC_ERROR;
@@ -1599,8 +1705,9 @@
}
int32_t MediaCodecVideoEncoder::ReleaseOnCodecThread() {
- if (!inited_)
+ if (!inited_) {
return WEBRTC_VIDEO_CODEC_OK;
+ }
CheckOnCodecThread();
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ALOGD("EncoderRelease: Frames received: %d. Frames dropped: %d.",
@@ -1610,8 +1717,9 @@
jni->DeleteGlobalRef(input_buffers_[i]);
input_buffers_.clear();
jni->CallVoidMethod(*j_media_codec_video_encoder_, j_release_method_);
- ResetParameters(jni);
- CHECK_EXCEPTION(jni, "");
+ CHECK_EXCEPTION(jni);
+ rtc::MessageQueueManager::Clear(this);
+ inited_ = false;
return WEBRTC_VIDEO_CODEC_OK;
}
@@ -1624,13 +1732,17 @@
}
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedLocalRefFrame local_ref_frame(jni);
- last_set_bitrate_kbps_ = new_bit_rate;
- last_set_fps_ = frame_rate;
+ if (new_bit_rate > 0) {
+ last_set_bitrate_kbps_ = new_bit_rate;
+ }
+ if (frame_rate > 0) {
+ last_set_fps_ = frame_rate;
+ }
bool ret = jni->CallBooleanMethod(*j_media_codec_video_encoder_,
j_set_rates_method_,
- new_bit_rate,
- frame_rate);
- CHECK_EXCEPTION(jni, "");
+ last_set_bitrate_kbps_,
+ last_set_fps_);
+ CHECK_EXCEPTION(jni);
if (!ret) {
ResetCodec();
return WEBRTC_VIDEO_CODEC_ERROR;
@@ -1638,17 +1750,6 @@
return WEBRTC_VIDEO_CODEC_OK;
}
-void MediaCodecVideoEncoder::ResetParameters(JNIEnv* jni) {
- rtc::MessageQueueManager::Clear(this);
- width_ = 0;
- height_ = 0;
- yuv_size_ = 0;
- drop_next_input_frame_ = false;
- inited_ = false;
- CHECK(input_buffers_.empty(),
- "ResetParameters called while holding input_buffers_!");
-}
-
int MediaCodecVideoEncoder::GetOutputBufferInfoIndex(
JNIEnv* jni,
jobject j_output_buffer_info) {
@@ -1678,9 +1779,10 @@
while (true) {
jobject j_output_buffer_info = jni->CallObjectMethod(
*j_media_codec_video_encoder_, j_dequeue_output_buffer_method_);
- CHECK_EXCEPTION(jni, "");
- if (IsNull(jni, j_output_buffer_info))
+ CHECK_EXCEPTION(jni);
+ if (IsNull(jni, j_output_buffer_info)) {
break;
+ }
int output_buffer_index =
GetOutputBufferInfoIndex(jni, j_output_buffer_info);
@@ -1689,31 +1791,62 @@
return false;
}
- jlong capture_time_ms =
+ // Get frame timestamps from a queue.
+ last_output_timestamp_ms_ =
GetOutputBufferInfoPresentationTimestampUs(jni, j_output_buffer_info) /
1000;
- last_output_timestamp_ms_ = capture_time_ms;
+ int32_t timestamp = timestamps_.front();
+ timestamps_.erase(timestamps_.begin());
+ int64_t render_time_ms = render_times_ms_.front();
+ render_times_ms_.erase(render_times_ms_.begin());
+ int64_t frame_encoding_time_ms = GetCurrentTimeMs() -
+ frame_rtc_times_ms_.front();
+ frame_rtc_times_ms_.erase(frame_rtc_times_ms_.begin());
frames_in_queue_--;
- ALOGV("Encoder got output buffer # %d. TS: %lld. Latency: %lld",
- output_buffer_index, last_output_timestamp_ms_,
- last_input_timestamp_ms_ - last_output_timestamp_ms_);
+ // Extract payload and key frame flag.
int32_t callback_status = 0;
+ jobject j_output_buffer =
+ GetOutputBufferInfoBuffer(jni, j_output_buffer_info);
+ bool key_frame = GetOutputBufferInfoIsKeyFrame(jni, j_output_buffer_info);
+ size_t payload_size = jni->GetDirectBufferCapacity(j_output_buffer);
+ uint8* payload = reinterpret_cast<uint8_t*>(
+ jni->GetDirectBufferAddress(j_output_buffer));
+ CHECK_EXCEPTION(jni);
+
+ ALOGV("Encoder got output buffer # %d. Size: %d. TS: %lld. Latency: %lld."
+ " EncTime: %lld",
+ output_buffer_index, payload_size, last_output_timestamp_ms_,
+ last_input_timestamp_ms_ - last_output_timestamp_ms_,
+ frame_encoding_time_ms);
+
+ // Calculate and print encoding statistics - every 3 seconds.
+ current_frames_++;
+ current_bytes_ += payload_size;
+ current_encoding_time_ms_ += frame_encoding_time_ms;
+ int statistic_time_ms = GetCurrentTimeMs() - start_time_ms_;
+ if (statistic_time_ms >= kMediaCodecStatisticsIntervalMs &&
+ current_frames_ > 0) {
+ ALOGD("Encoder bitrate: %d, target: %d kbps, fps: %d,"
+ " encTime: %d for last %d ms",
+ current_bytes_ * 8 / statistic_time_ms,
+ last_set_bitrate_kbps_,
+ (current_frames_ * 1000 + statistic_time_ms / 2) / statistic_time_ms,
+ current_encoding_time_ms_ / current_frames_, statistic_time_ms);
+ start_time_ms_ = GetCurrentTimeMs();
+ current_frames_ = 0;
+ current_bytes_= 0;
+ current_encoding_time_ms_ = 0;
+ }
+
+ // Callback - return encoded frame.
if (callback_) {
- jobject j_output_buffer =
- GetOutputBufferInfoBuffer(jni, j_output_buffer_info);
- bool key_frame = GetOutputBufferInfoIsKeyFrame(jni, j_output_buffer_info);
- size_t payload_size = jni->GetDirectBufferCapacity(j_output_buffer);
- uint8* payload = reinterpret_cast<uint8_t*>(
- jni->GetDirectBufferAddress(j_output_buffer));
- CHECK_EXCEPTION(jni, "");
scoped_ptr<webrtc::EncodedImage> image(
new webrtc::EncodedImage(payload, payload_size, payload_size));
image->_encodedWidth = width_;
image->_encodedHeight = height_;
- // Convert capture time to 90 kHz RTP timestamp.
- image->_timeStamp = static_cast<uint32_t>(90 * capture_time_ms);
- image->capture_time_ms_ = capture_time_ms;
+ image->_timeStamp = timestamp;
+ image->capture_time_ms_ = render_time_ms;
image->_frameType = (key_frame ? webrtc::kKeyFrame : webrtc::kDeltaFrame);
image->_completeFrame = true;
@@ -1736,19 +1869,21 @@
callback_status = callback_->Encoded(*image, &info, &header);
}
+ // Return output buffer back to the encoder.
bool success = jni->CallBooleanMethod(*j_media_codec_video_encoder_,
j_release_output_buffer_method_,
output_buffer_index);
- CHECK_EXCEPTION(jni, "");
+ CHECK_EXCEPTION(jni);
if (!success) {
ResetCodec();
return false;
}
- if (callback_status > 0)
+ if (callback_status > 0) {
drop_next_input_frame_ = true;
// Theoretically could handle callback_status<0 here, but unclear what that
// would mean for us.
+ }
}
return true;
@@ -1782,7 +1917,7 @@
bool is_platform_supported = jni->CallStaticBooleanMethod(
j_encoder_class,
GetStaticMethodID(jni, j_encoder_class, "isPlatformSupported", "()Z"));
- CHECK_EXCEPTION(jni, "");
+ CHECK_EXCEPTION(jni);
if (!is_platform_supported)
return;
@@ -1823,6 +1958,8 @@
explicit MediaCodecVideoDecoder(JNIEnv* jni);
virtual ~MediaCodecVideoDecoder();
+ static int SetAndroidObjects(JNIEnv* jni, jobject render_egl_context);
+
virtual int32_t InitDecode(const VideoCodec* codecSettings,
int32_t numberOfCores) OVERRIDE;
@@ -1848,13 +1985,29 @@
int32_t InitDecodeOnCodecThread();
int32_t ReleaseOnCodecThread();
int32_t DecodeOnCodecThread(const EncodedImage& inputImage);
+ // Deliver any outputs pending in the MediaCodec to our |callback_| and return
+ // true on success.
+ bool DeliverPendingOutputs(JNIEnv* jni, int dequeue_timeout_us);
+
bool key_frame_required_;
bool inited_;
+ bool use_surface_;
VideoCodec codec_;
I420VideoFrame decoded_image_;
+ NativeHandleImpl native_handle_;
DecodedImageCallback* callback_;
- int frames_received_; // Number of frames received by decoder.
+ int frames_received_; // Number of frames received by decoder.
+ int frames_decoded_; // Number of frames decoded by decoder
+ int64_t start_time_ms_; // Start time for statistics.
+ int current_frames_; // Number of frames in the current statistics interval.
+ int current_bytes_; // Encoded bytes in the current statistics interval.
+ int current_decoding_time_ms_; // Overall decoding time in the current second
+ uint32_t max_pending_frames_; // Maximum number of pending input frames
+ std::vector<int32_t> timestamps_;
+ std::vector<int64_t> ntp_times_ms_;
+ std::vector<int64_t> frame_rtc_times_ms_; // Time when video frame is sent to
+ // decoder input.
// State that is constant for the lifetime of this object once the ctor
// returns.
@@ -1867,6 +2020,7 @@
jmethodID j_queue_input_buffer_method_;
jmethodID j_dequeue_output_buffer_method_;
jmethodID j_release_output_buffer_method_;
+ // MediaCodecVideoDecoder fields.
jfieldID j_input_buffers_field_;
jfieldID j_output_buffers_field_;
jfieldID j_color_format_field_;
@@ -1874,32 +2028,59 @@
jfieldID j_height_field_;
jfieldID j_stride_field_;
jfieldID j_slice_height_field_;
+ jfieldID j_surface_texture_field_;
+ jfieldID j_textureID_field_;
+ // MediaCodecVideoDecoder.DecoderOutputBufferInfo fields.
+ jfieldID j_info_index_field_;
+ jfieldID j_info_offset_field_;
+ jfieldID j_info_size_field_;
+ jfieldID j_info_presentation_timestamp_us_field_;
// Global references; must be deleted in Release().
std::vector<jobject> input_buffers_;
+ jobject surface_texture_;
+
+ // Render EGL context.
+ static jobject render_egl_context_;
};
-MediaCodecVideoDecoder::MediaCodecVideoDecoder(JNIEnv* jni) :
- key_frame_required_(true),
- inited_(false),
- codec_thread_(new Thread()),
- j_media_codec_video_decoder_class_(
- jni,
- FindClass(jni, "org/webrtc/MediaCodecVideoDecoder")),
- j_media_codec_video_decoder_(
- jni,
- jni->NewObject(*j_media_codec_video_decoder_class_,
- GetMethodID(jni,
- *j_media_codec_video_decoder_class_,
- "<init>",
- "()V"))) {
+jobject MediaCodecVideoDecoder::render_egl_context_ = NULL;
+
+int MediaCodecVideoDecoder::SetAndroidObjects(JNIEnv* jni,
+ jobject render_egl_context) {
+ if (render_egl_context_) {
+ jni->DeleteGlobalRef(render_egl_context_);
+ }
+ if (IsNull(jni, render_egl_context)) {
+ render_egl_context_ = NULL;
+ } else {
+ render_egl_context_ = jni->NewGlobalRef(render_egl_context);
+ }
+ ALOGD("VideoDecoder EGL context set.");
+ return 0;
+}
+
+MediaCodecVideoDecoder::MediaCodecVideoDecoder(JNIEnv* jni)
+ : key_frame_required_(true),
+ inited_(false),
+ codec_thread_(new Thread()),
+ j_media_codec_video_decoder_class_(
+ jni,
+ FindClass(jni, "org/webrtc/MediaCodecVideoDecoder")),
+ j_media_codec_video_decoder_(
+ jni,
+ jni->NewObject(*j_media_codec_video_decoder_class_,
+ GetMethodID(jni,
+ *j_media_codec_video_decoder_class_,
+ "<init>",
+ "()V"))) {
ScopedLocalRefFrame local_ref_frame(jni);
codec_thread_->SetName("MediaCodecVideoDecoder", NULL);
- CHECK(codec_thread_->Start(), "Failed to start MediaCodecVideoDecoder");
+ CHECK(codec_thread_->Start()) << "Failed to start MediaCodecVideoDecoder";
- j_init_decode_method_ = GetMethodID(jni,
- *j_media_codec_video_decoder_class_,
- "initDecode", "(II)Z");
+ j_init_decode_method_ = GetMethodID(
+ jni, *j_media_codec_video_decoder_class_, "initDecode",
+ "(IIZLandroid/opengl/EGLContext;)Z");
j_release_method_ =
GetMethodID(jni, *j_media_codec_video_decoder_class_, "release", "()V");
j_dequeue_input_buffer_method_ = GetMethodID(
@@ -1907,9 +2088,10 @@
j_queue_input_buffer_method_ = GetMethodID(
jni, *j_media_codec_video_decoder_class_, "queueInputBuffer", "(IIJ)Z");
j_dequeue_output_buffer_method_ = GetMethodID(
- jni, *j_media_codec_video_decoder_class_, "dequeueOutputBuffer", "()I");
+ jni, *j_media_codec_video_decoder_class_, "dequeueOutputBuffer",
+ "(I)Lorg/webrtc/MediaCodecVideoDecoder$DecoderOutputBufferInfo;");
j_release_output_buffer_method_ = GetMethodID(
- jni, *j_media_codec_video_decoder_class_, "releaseOutputBuffer", "(I)Z");
+ jni, *j_media_codec_video_decoder_class_, "releaseOutputBuffer", "(IZ)Z");
j_input_buffers_field_ = GetFieldID(
jni, *j_media_codec_video_decoder_class_,
@@ -1927,12 +2109,32 @@
jni, *j_media_codec_video_decoder_class_, "stride", "I");
j_slice_height_field_ = GetFieldID(
jni, *j_media_codec_video_decoder_class_, "sliceHeight", "I");
+ j_textureID_field_ = GetFieldID(
+ jni, *j_media_codec_video_decoder_class_, "textureID", "I");
+ j_surface_texture_field_ = GetFieldID(
+ jni, *j_media_codec_video_decoder_class_, "surfaceTexture",
+ "Landroid/graphics/SurfaceTexture;");
- CHECK_EXCEPTION(jni, "MediaCodecVideoDecoder ctor failed");
+ jclass j_decoder_output_buffer_info_class = FindClass(jni,
+ "org/webrtc/MediaCodecVideoDecoder$DecoderOutputBufferInfo");
+ j_info_index_field_ = GetFieldID(
+ jni, j_decoder_output_buffer_info_class, "index", "I");
+ j_info_offset_field_ = GetFieldID(
+ jni, j_decoder_output_buffer_info_class, "offset", "I");
+ j_info_size_field_ = GetFieldID(
+ jni, j_decoder_output_buffer_info_class, "size", "I");
+ j_info_presentation_timestamp_us_field_ = GetFieldID(
+ jni, j_decoder_output_buffer_info_class, "presentationTimestampUs", "J");
+
+ CHECK_EXCEPTION(jni) << "MediaCodecVideoDecoder ctor failed";
+ use_surface_ = true;
+ if (render_egl_context_ == NULL)
+ use_surface_ = false;
memset(&codec_, 0, sizeof(codec_));
}
MediaCodecVideoDecoder::~MediaCodecVideoDecoder() {
+ // Call Release() to ensure no more callbacks to us after we are deleted.
Release();
}
@@ -1954,6 +2156,7 @@
// Always start with a complete key frame.
key_frame_required_ = true;
frames_received_ = 0;
+ frames_decoded_ = 0;
// Call Java init.
return codec_thread_->Invoke<int32_t>(
@@ -1964,28 +2167,50 @@
CheckOnCodecThread();
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedLocalRefFrame local_ref_frame(jni);
- ALOGD("InitDecodeOnCodecThread: %d x %d. FPS: %d",
+ ALOGD("InitDecodeOnCodecThread: %d x %d. fps: %d",
codec_.width, codec_.height, codec_.maxFramerate);
bool success = jni->CallBooleanMethod(*j_media_codec_video_decoder_,
j_init_decode_method_,
codec_.width,
- codec_.height);
- CHECK_EXCEPTION(jni, "");
- if (!success)
+ codec_.height,
+ use_surface_,
+ render_egl_context_);
+ CHECK_EXCEPTION(jni);
+ if (!success) {
return WEBRTC_VIDEO_CODEC_ERROR;
+ }
inited_ = true;
+ max_pending_frames_ = 0;
+ if (use_surface_) {
+ max_pending_frames_ = 1;
+ }
+ start_time_ms_ = GetCurrentTimeMs();
+ current_frames_ = 0;
+ current_bytes_ = 0;
+ current_decoding_time_ms_ = 0;
+ timestamps_.clear();
+ ntp_times_ms_.clear();
+ frame_rtc_times_ms_.clear();
+
jobjectArray input_buffers = (jobjectArray)GetObjectField(
jni, *j_media_codec_video_decoder_, j_input_buffers_field_);
size_t num_input_buffers = jni->GetArrayLength(input_buffers);
-
input_buffers_.resize(num_input_buffers);
for (size_t i = 0; i < num_input_buffers; ++i) {
input_buffers_[i] =
jni->NewGlobalRef(jni->GetObjectArrayElement(input_buffers, i));
- CHECK_EXCEPTION(jni, "");
+ CHECK_EXCEPTION(jni);
}
+
+ if (use_surface_) {
+ jobject surface_texture = GetObjectField(
+ jni, *j_media_codec_video_decoder_, j_surface_texture_field_);
+ surface_texture_ = jni->NewGlobalRef(surface_texture);
+ }
+ codec_thread_->PostDelayed(kMediaCodecPollMs, this);
+
return WEBRTC_VIDEO_CODEC_OK;
}
@@ -1995,25 +2220,40 @@
}
int32_t MediaCodecVideoDecoder::ReleaseOnCodecThread() {
- if (!inited_)
+ if (!inited_) {
return WEBRTC_VIDEO_CODEC_OK;
+ }
CheckOnCodecThread();
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ALOGD("DecoderRelease: Frames received: %d.", frames_received_);
ScopedLocalRefFrame local_ref_frame(jni);
- for (size_t i = 0; i < input_buffers_.size(); ++i)
+ for (size_t i = 0; i < input_buffers_.size(); i++) {
jni->DeleteGlobalRef(input_buffers_[i]);
+ }
input_buffers_.clear();
+ if (use_surface_) {
+ // Before deleting texture object make sure it is no longer referenced
+ // by any TextureVideoFrame.
+ int32_t waitTimeoutUs = 3000000; // 3 second wait
+ while (waitTimeoutUs > 0 && native_handle_.ref_count() > 0) {
+ ALOGD("Current Texture RefCnt: %d", native_handle_.ref_count());
+ usleep(30000);
+ waitTimeoutUs -= 30000;
+ }
+ ALOGD("TextureRefCnt: %d", native_handle_.ref_count());
+ jni->DeleteGlobalRef(surface_texture_);
+ }
jni->CallVoidMethod(*j_media_codec_video_decoder_, j_release_method_);
- CHECK_EXCEPTION(jni, "");
+ CHECK_EXCEPTION(jni);
+ rtc::MessageQueueManager::Clear(this);
inited_ = false;
return WEBRTC_VIDEO_CODEC_OK;
}
void MediaCodecVideoDecoder::CheckOnCodecThread() {
- CHECK(codec_thread_ == ThreadManager::Instance()->CurrentThread(),
- "Running on wrong thread!");
+ CHECK(codec_thread_ == ThreadManager::Instance()->CurrentThread())
+ << "Running on wrong thread!";
}
int32_t MediaCodecVideoDecoder::Decode(
@@ -2066,10 +2306,25 @@
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedLocalRefFrame local_ref_frame(jni);
+ // Try to drain the decoder and wait until output is not too
+ // much behind the input.
+ if (frames_received_ > frames_decoded_ + max_pending_frames_) {
+ ALOGV("Wait for output...");
+ if (!DeliverPendingOutputs(jni, kMediaCodecTimeoutMs * 1000)) {
+ Reset();
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ if (frames_received_ > frames_decoded_ + max_pending_frames_) {
+ ALOGE("Output buffer dequeue timeout");
+ Reset();
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ }
+
// Get input buffer.
int j_input_buffer_index = jni->CallIntMethod(*j_media_codec_video_decoder_,
j_dequeue_input_buffer_method_);
- CHECK_EXCEPTION(jni, "");
+ CHECK_EXCEPTION(jni);
if (j_input_buffer_index < 0) {
ALOGE("dequeueInputBuffer error");
Reset();
@@ -2080,19 +2335,26 @@
jobject j_input_buffer = input_buffers_[j_input_buffer_index];
uint8* buffer =
reinterpret_cast<uint8*>(jni->GetDirectBufferAddress(j_input_buffer));
- CHECK(buffer, "Indirect buffer??");
+ CHECK(buffer) << "Indirect buffer??";
int64 buffer_capacity = jni->GetDirectBufferCapacity(j_input_buffer);
- CHECK_EXCEPTION(jni, "");
+ CHECK_EXCEPTION(jni);
if (buffer_capacity < inputImage._length) {
ALOGE("Input frame size %d is bigger than buffer size %d.",
inputImage._length, buffer_capacity);
Reset();
return WEBRTC_VIDEO_CODEC_ERROR;
}
- ALOGV("Decode frame # %d. Buffer # %d. Size: %d",
+ ALOGV("Decoder frame in # %d. Buffer # %d. Size: %d",
frames_received_, j_input_buffer_index, inputImage._length);
memcpy(buffer, inputImage._buffer, inputImage._length);
+ // Save input image timestamps for later output.
+ frames_received_++;
+ current_bytes_ += inputImage._length;
+ timestamps_.push_back(inputImage._timeStamp);
+ ntp_times_ms_.push_back(inputImage.ntp_time_ms_);
+ frame_rtc_times_ms_.push_back(GetCurrentTimeMs());
+
// Feed input to decoder.
jlong timestamp_us = (frames_received_ * 1000000) / codec_.maxFramerate;
bool success = jni->CallBooleanMethod(*j_media_codec_video_decoder_,
@@ -2100,33 +2362,55 @@
j_input_buffer_index,
inputImage._length,
timestamp_us);
- CHECK_EXCEPTION(jni, "");
+ CHECK_EXCEPTION(jni);
if (!success) {
ALOGE("queueInputBuffer error");
Reset();
return WEBRTC_VIDEO_CODEC_ERROR;
}
- // Get output index.
- int j_output_buffer_index =
- jni->CallIntMethod(*j_media_codec_video_decoder_,
- j_dequeue_output_buffer_method_);
- CHECK_EXCEPTION(jni, "");
- if (j_output_buffer_index < 0) {
- ALOGE("dequeueOutputBuffer error");
+ // Try to drain the decoder
+ if (!DeliverPendingOutputs(jni, 0)) {
+ ALOGE("DeliverPendingOutputs error");
Reset();
return WEBRTC_VIDEO_CODEC_ERROR;
}
- // Extract data from Java ByteBuffer.
- jobjectArray output_buffers = reinterpret_cast<jobjectArray>(GetObjectField(
- jni, *j_media_codec_video_decoder_, j_output_buffers_field_));
- jobject output_buffer =
- jni->GetObjectArrayElement(output_buffers, j_output_buffer_index);
- buffer_capacity = jni->GetDirectBufferCapacity(output_buffer);
- uint8_t* payload =
- reinterpret_cast<uint8_t*>(jni->GetDirectBufferAddress(output_buffer));
- CHECK_EXCEPTION(jni, "");
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+bool MediaCodecVideoDecoder::DeliverPendingOutputs(
+ JNIEnv* jni, int dequeue_timeout_us) {
+ if (frames_received_ <= frames_decoded_) {
+ // No need to query for output buffers - decoder is drained.
+ return true;
+ }
+ // Get decoder output.
+ jobject j_decoder_output_buffer_info = jni->CallObjectMethod(
+ *j_media_codec_video_decoder_,
+ j_dequeue_output_buffer_method_,
+ dequeue_timeout_us);
+
+ CHECK_EXCEPTION(jni);
+ if (IsNull(jni, j_decoder_output_buffer_info)) {
+ return true;
+ }
+
+ // Extract output buffer info from Java DecoderOutputBufferInfo.
+ int output_buffer_index =
+ GetIntField(jni, j_decoder_output_buffer_info, j_info_index_field_);
+ if (output_buffer_index < 0) {
+ ALOGE("dequeueOutputBuffer error : %d", output_buffer_index);
+ Reset();
+ return false;
+ }
+ int output_buffer_offset =
+ GetIntField(jni, j_decoder_output_buffer_info, j_info_offset_field_);
+ int output_buffer_size =
+ GetIntField(jni, j_decoder_output_buffer_info, j_info_size_field_);
+ CHECK_EXCEPTION(jni);
+
+ // Get decoded video frame properties.
int color_format = GetIntField(jni, *j_media_codec_video_decoder_,
j_color_format_field_);
int width = GetIntField(jni, *j_media_codec_video_decoder_, j_width_field_);
@@ -2134,52 +2418,112 @@
int stride = GetIntField(jni, *j_media_codec_video_decoder_, j_stride_field_);
int slice_height = GetIntField(jni, *j_media_codec_video_decoder_,
j_slice_height_field_);
- if (buffer_capacity < width * height * 3 / 2) {
- ALOGE("Insufficient output buffer capacity: %d", buffer_capacity);
- Reset();
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
- ALOGV("Decoder got output buffer %d x %d. %d x %d. Color: 0x%x. Size: %d",
- width, height, stride, slice_height, color_format, buffer_capacity);
+ int texture_id = GetIntField(jni, *j_media_codec_video_decoder_,
+ j_textureID_field_);
- if (color_format == COLOR_FormatYUV420Planar) {
- decoded_image_.CreateFrame(
- stride * slice_height, payload,
- (stride * slice_height) / 4, payload + (stride * slice_height),
- (stride * slice_height) / 4, payload + (5 * stride * slice_height / 4),
- width, height,
- stride, stride / 2, stride / 2);
- } else {
- // All other supported formats are nv12.
- decoded_image_.CreateEmptyFrame(width, height, width, width / 2, width / 2);
- libyuv::NV12ToI420(
- payload, stride,
- payload + stride * slice_height, stride,
- decoded_image_.buffer(webrtc::kYPlane),
- decoded_image_.stride(webrtc::kYPlane),
- decoded_image_.buffer(webrtc::kUPlane),
- decoded_image_.stride(webrtc::kUPlane),
- decoded_image_.buffer(webrtc::kVPlane),
- decoded_image_.stride(webrtc::kVPlane),
- width, height);
+ // Extract data from Java ByteBuffer and create output yuv420 frame -
+ // for non surface decoding only.
+ if (!use_surface_) {
+ if (output_buffer_size < width * height * 3 / 2) {
+ ALOGE("Insufficient output buffer size: %d", output_buffer_size);
+ Reset();
+ return false;
+ }
+ jobjectArray output_buffers = reinterpret_cast<jobjectArray>(GetObjectField(
+ jni, *j_media_codec_video_decoder_, j_output_buffers_field_));
+ jobject output_buffer =
+ jni->GetObjectArrayElement(output_buffers, output_buffer_index);
+ uint8_t* payload = reinterpret_cast<uint8_t*>(jni->GetDirectBufferAddress(
+ output_buffer));
+ CHECK_EXCEPTION(jni);
+ payload += output_buffer_offset;
+
+ // Create yuv420 frame.
+ if (color_format == COLOR_FormatYUV420Planar) {
+ decoded_image_.CreateFrame(
+ stride * slice_height, payload,
+ (stride * slice_height) / 4, payload + (stride * slice_height),
+ (stride * slice_height) / 4, payload + (5 * stride * slice_height / 4),
+ width, height,
+ stride, stride / 2, stride / 2);
+ } else {
+ // All other supported formats are nv12.
+ decoded_image_.CreateEmptyFrame(width, height, width,
+ width / 2, width / 2);
+ libyuv::NV12ToI420(
+ payload, stride,
+ payload + stride * slice_height, stride,
+ decoded_image_.buffer(webrtc::kYPlane),
+ decoded_image_.stride(webrtc::kYPlane),
+ decoded_image_.buffer(webrtc::kUPlane),
+ decoded_image_.stride(webrtc::kUPlane),
+ decoded_image_.buffer(webrtc::kVPlane),
+ decoded_image_.stride(webrtc::kVPlane),
+ width, height);
+ }
}
+ // Get frame timestamps from a queue.
+ int32_t timestamp = timestamps_.front();
+ timestamps_.erase(timestamps_.begin());
+ int64_t ntp_time_ms = ntp_times_ms_.front();
+ ntp_times_ms_.erase(ntp_times_ms_.begin());
+ int64_t frame_decoding_time_ms = GetCurrentTimeMs() -
+ frame_rtc_times_ms_.front();
+ frame_rtc_times_ms_.erase(frame_rtc_times_ms_.begin());
+
+ ALOGV("Decoder frame out # %d. %d x %d. %d x %d. Color: 0x%x. Size: %d."
+ " DecTime: %lld", frames_decoded_, width, height, stride, slice_height,
+ color_format, output_buffer_size, frame_decoding_time_ms);
+
// Return output buffer back to codec.
- success = jni->CallBooleanMethod(*j_media_codec_video_decoder_,
- j_release_output_buffer_method_,
- j_output_buffer_index);
- CHECK_EXCEPTION(jni, "");
+ bool success = jni->CallBooleanMethod(
+ *j_media_codec_video_decoder_,
+ j_release_output_buffer_method_,
+ output_buffer_index,
+ use_surface_);
+ CHECK_EXCEPTION(jni);
if (!success) {
ALOGE("releaseOutputBuffer error");
Reset();
- return WEBRTC_VIDEO_CODEC_ERROR;
+ return false;
}
- // Callback.
- decoded_image_.set_timestamp(inputImage._timeStamp);
- decoded_image_.set_ntp_time_ms(inputImage.ntp_time_ms_);
- frames_received_++;
- return callback_->Decoded(decoded_image_);
+ // Calculate and print decoding statistics - every 3 seconds.
+ frames_decoded_++;
+ current_frames_++;
+ current_decoding_time_ms_ += frame_decoding_time_ms;
+ int statistic_time_ms = GetCurrentTimeMs() - start_time_ms_;
+ if (statistic_time_ms >= kMediaCodecStatisticsIntervalMs &&
+ current_frames_ > 0) {
+ ALOGD("Decoder bitrate: %d kbps, fps: %d, decTime: %d for last %d ms",
+ current_bytes_ * 8 / statistic_time_ms,
+ (current_frames_ * 1000 + statistic_time_ms / 2) / statistic_time_ms,
+ current_decoding_time_ms_ / current_frames_, statistic_time_ms);
+ start_time_ms_ = GetCurrentTimeMs();
+ current_frames_ = 0;
+ current_bytes_= 0;
+ current_decoding_time_ms_ = 0;
+ }
+
+ // Callback - output decoded frame.
+ int32_t callback_status = WEBRTC_VIDEO_CODEC_OK;
+ if (use_surface_) {
+ native_handle_.SetTextureObject(surface_texture_, texture_id);
+ TextureVideoFrame texture_image(
+ &native_handle_, width, height, timestamp, 0);
+ texture_image.set_ntp_time_ms(ntp_time_ms);
+ callback_status = callback_->Decoded(texture_image);
+ } else {
+ decoded_image_.set_timestamp(timestamp);
+ decoded_image_.set_ntp_time_ms(ntp_time_ms);
+ callback_status = callback_->Decoded(decoded_image_);
+ }
+ if (callback_status > 0) {
+ ALOGE("callback error");
+ }
+
+ return true;
}
int32_t MediaCodecVideoDecoder::RegisterDecodeCompleteCallback(
@@ -2197,6 +2541,19 @@
}
void MediaCodecVideoDecoder::OnMessage(rtc::Message* msg) {
+ JNIEnv* jni = AttachCurrentThreadIfNeeded();
+ ScopedLocalRefFrame local_ref_frame(jni);
+ if (!inited_) {
+ return;
+ }
+ // We only ever send one message to |this| directly (not through a Bind()'d
+ // functor), so expect no ID/data.
+ CHECK(!msg->message_id) << "Unexpected message!";
+ CHECK(!msg->pdata) << "Unexpected message!";
+ CheckOnCodecThread();
+
+ DeliverPendingOutputs(jni, 0);
+ codec_thread_->PostDelayed(kMediaCodecPollMs, this);
}
class MediaCodecVideoDecoderFactory
@@ -2221,7 +2578,7 @@
is_platform_supported_ = jni->CallStaticBooleanMethod(
j_decoder_class,
GetStaticMethodID(jni, j_decoder_class, "isPlatformSupported", "()Z"));
- CHECK_EXCEPTION(jni, "");
+ CHECK_EXCEPTION(jni);
}
MediaCodecVideoDecoderFactory::~MediaCodecVideoDecoderFactory() {}
@@ -2240,7 +2597,7 @@
delete decoder;
}
-#endif // ANDROID
+#endif // #if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
} // anonymous namespace
@@ -2250,13 +2607,13 @@
Java_org_webrtc_##name
extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM *jvm, void *reserved) {
- CHECK(!g_jvm, "JNI_OnLoad called more than once!");
+ CHECK(!g_jvm) << "JNI_OnLoad called more than once!";
g_jvm = jvm;
- CHECK(g_jvm, "JNI_OnLoad handed NULL?");
+ CHECK(g_jvm) << "JNI_OnLoad handed NULL?";
- CHECK(!pthread_once(&g_jni_ptr_once, &CreateJNIPtrKey), "pthread_once");
+ CHECK(!pthread_once(&g_jni_ptr_once, &CreateJNIPtrKey)) << "pthread_once";
- CHECK(rtc::InitializeSSL(), "Failed to InitializeSSL()");
+ CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
JNIEnv* jni;
if (jvm->GetEnv(reinterpret_cast<void**>(&jni), JNI_VERSION_1_6) != JNI_OK)
@@ -2270,7 +2627,7 @@
g_class_reference_holder->FreeReferences(AttachCurrentThreadIfNeeded());
delete g_class_reference_holder;
g_class_reference_holder = NULL;
- CHECK(rtc::CleanupSSL(), "Failed to CleanupSSL()");
+ CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()";
g_jvm = NULL;
}
@@ -2306,8 +2663,8 @@
JOW(jlong, DataChannel_bufferedAmount)(JNIEnv* jni, jobject j_dc) {
uint64 buffered_amount = ExtractNativeDC(jni, j_dc)->buffered_amount();
- CHECK(buffered_amount <= std::numeric_limits<int64>::max(),
- "buffered_amount overflowed jlong!");
+ CHECK_LE(buffered_amount, std::numeric_limits<int64>::max())
+ << "buffered_amount overflowed jlong!";
return static_cast<jlong>(buffered_amount);
}
@@ -2335,12 +2692,12 @@
std::string path = JavaToStdString(jni, j_path);
if (nativeLevels != webrtc::kTraceNone) {
webrtc::Trace::set_level_filter(nativeLevels);
-#ifdef ANDROID
+#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
if (path != "logcat:") {
#endif
- CHECK(webrtc::Trace::SetTraceFile(path.c_str(), false) == 0,
- "SetTraceFile failed");
-#ifdef ANDROID
+ CHECK_EQ(0, webrtc::Trace::SetTraceFile(path.c_str(), false))
+ << "SetTraceFile failed";
+#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
} else {
// Intentionally leak this to avoid needing to reason about its lifecycle.
// It keeps no state and functions only as a dispatch point.
@@ -2368,10 +2725,14 @@
delete reinterpret_cast<cricket::VideoCapturer*>(j_p);
}
-JOW(void, VideoRenderer_free)(JNIEnv*, jclass, jlong j_p) {
+JOW(void, VideoRenderer_freeGuiVideoRenderer)(JNIEnv*, jclass, jlong j_p) {
delete reinterpret_cast<VideoRendererWrapper*>(j_p);
}
+JOW(void, VideoRenderer_freeWrappedVideoRenderer)(JNIEnv*, jclass, jlong j_p) {
+ delete reinterpret_cast<JavaVideoRendererWrapper*>(j_p);
+}
+
JOW(void, MediaStreamTrack_free)(JNIEnv*, jclass, jlong j_p) {
CHECK_RELEASE(reinterpret_cast<MediaStreamTrackInterface*>(j_p));
}
@@ -2414,19 +2775,28 @@
return (jlong)new PCOJava(jni, j_observer);
}
-#ifdef ANDROID
+#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
JOW(jboolean, PeerConnectionFactory_initializeAndroidGlobals)(
JNIEnv* jni, jclass, jobject context,
- jboolean initialize_audio, jboolean initialize_video) {
- CHECK(g_jvm, "JNI_OnLoad failed to run?");
+ jboolean initialize_audio, jboolean initialize_video,
+ jobject render_egl_context) {
+ CHECK(g_jvm) << "JNI_OnLoad failed to run?";
bool failure = false;
+ if (!factory_static_initialized) {
+ if (initialize_video) {
+ failure |= webrtc::SetCaptureAndroidVM(g_jvm, context);
+ failure |= webrtc::SetRenderAndroidVM(g_jvm);
+ }
+ if (initialize_audio)
+ failure |= webrtc::VoiceEngine::SetAndroidObjects(g_jvm, jni, context);
+ factory_static_initialized = true;
+ }
if (initialize_video)
- failure |= webrtc::VideoEngine::SetAndroidObjects(g_jvm, context);
- if (initialize_audio)
- failure |= webrtc::VoiceEngine::SetAndroidObjects(g_jvm, jni, context);
+ failure |= MediaCodecVideoDecoder::SetAndroidObjects(jni,
+ render_egl_context);
return !failure;
}
-#endif // ANDROID
+#endif // defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
// Helper struct for working around the fact that CreatePeerConnectionFactory()
// comes in two flavors: either entirely automagical (constructing its own
@@ -2466,11 +2836,11 @@
worker_thread->SetName("worker_thread", NULL);
Thread* signaling_thread = new Thread();
signaling_thread->SetName("signaling_thread", NULL);
- CHECK(worker_thread->Start() && signaling_thread->Start(),
- "Failed to start threads");
+ CHECK(worker_thread->Start() && signaling_thread->Start())
+ << "Failed to start threads";
scoped_ptr<cricket::WebRtcVideoEncoderFactory> encoder_factory;
scoped_ptr<cricket::WebRtcVideoDecoderFactory> decoder_factory;
-#ifdef ANDROID
+#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
encoder_factory.reset(new MediaCodecVideoEncoderFactory());
decoder_factory.reset(new MediaCodecVideoDecoderFactory());
#endif
@@ -2558,15 +2928,15 @@
jmethodID iterator_id = GetMethodID(
jni, list_class, "iterator", "()Ljava/util/Iterator;");
jobject iterator = jni->CallObjectMethod(j_ice_servers, iterator_id);
- CHECK_EXCEPTION(jni, "error during CallObjectMethod");
+ CHECK_EXCEPTION(jni) << "error during CallObjectMethod";
jmethodID iterator_has_next = GetMethodID(
jni, GetObjectClass(jni, iterator), "hasNext", "()Z");
jmethodID iterator_next = GetMethodID(
jni, GetObjectClass(jni, iterator), "next", "()Ljava/lang/Object;");
while (jni->CallBooleanMethod(iterator, iterator_has_next)) {
- CHECK_EXCEPTION(jni, "error during CallBooleanMethod");
+ CHECK_EXCEPTION(jni) << "error during CallBooleanMethod";
jobject j_ice_server = jni->CallObjectMethod(iterator, iterator_next);
- CHECK_EXCEPTION(jni, "error during CallObjectMethod");
+ CHECK_EXCEPTION(jni) << "error during CallObjectMethod";
jclass j_ice_server_class = GetObjectClass(jni, j_ice_server);
jfieldID j_ice_server_uri_id =
GetFieldID(jni, j_ice_server_class, "uri", "Ljava/lang/String;");
@@ -2586,7 +2956,7 @@
server.password = JavaToStdString(jni, password);
ice_servers->push_back(server);
}
- CHECK_EXCEPTION(jni, "error during CallBooleanMethod");
+ CHECK_EXCEPTION(jni) << "error during CallBooleanMethod";
}
JOW(jlong, PeerConnectionFactory_nativeCreatePeerConnection)(
@@ -2635,16 +3005,16 @@
// vararg parameter as 64-bit and reading memory that doesn't belong to the
// 32-bit parameter.
jlong nativeChannelPtr = jlongFromPointer(channel.get());
- CHECK(nativeChannelPtr, "Failed to create DataChannel");
+ CHECK(nativeChannelPtr) << "Failed to create DataChannel";
jclass j_data_channel_class = FindClass(jni, "org/webrtc/DataChannel");
jmethodID j_data_channel_ctor = GetMethodID(
jni, j_data_channel_class, "<init>", "(J)V");
jobject j_channel = jni->NewObject(
j_data_channel_class, j_data_channel_ctor, nativeChannelPtr);
- CHECK_EXCEPTION(jni, "error during NewObject");
+ CHECK_EXCEPTION(jni) << "error during NewObject";
// Channel is now owned by Java object, and will be freed from there.
int bumped_count = channel->AddRef();
- CHECK(bumped_count == 2, "Unexpected refcount");
+ CHECK(bumped_count == 2) << "Unexpected refcount";
return j_channel;
}
@@ -2680,7 +3050,7 @@
"()Ljava/lang/String;");
jstring j_type_string = (jstring)jni->CallObjectMethod(
j_type, j_canonical_form_id);
- CHECK_EXCEPTION(jni, "error during CallObjectMethod");
+ CHECK_EXCEPTION(jni) << "error during CallObjectMethod";
std::string std_type = JavaToStdString(jni, j_type_string);
jfieldID j_description_id = GetFieldID(
@@ -2790,7 +3160,7 @@
std::string device_name = JavaToStdString(jni, j_device_name);
scoped_ptr<cricket::DeviceManagerInterface> device_manager(
cricket::DeviceManagerFactory::Create());
- CHECK(device_manager->Init(), "DeviceManager::Init() failed");
+ CHECK(device_manager->Init()) << "DeviceManager::Init() failed";
cricket::Device device;
if (!device_manager->GetVideoCaptureDevice(device_name, &device)) {
LOG(LS_ERROR) << "GetVideoCaptureDevice failed for " << device_name;
@@ -2826,8 +3196,8 @@
JOW(void, VideoSource_restart)(
JNIEnv* jni, jclass, jlong j_p_source, jlong j_p_format) {
- CHECK(j_p_source, "");
- CHECK(j_p_format, "");
+ CHECK(j_p_source);
+ CHECK(j_p_format);
scoped_ptr<cricket::VideoFormatPod> format(
reinterpret_cast<cricket::VideoFormatPod*>(j_p_format));
reinterpret_cast<VideoSourceInterface*>(j_p_source)->GetVideoCapturer()->
diff --git a/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java b/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
index a6a059e..9280743 100644
--- a/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
+++ b/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
@@ -27,14 +27,24 @@
package org.webrtc;
+import android.graphics.SurfaceTexture;
import android.media.MediaCodec;
-import android.media.MediaCodecInfo.CodecCapabilities;
import android.media.MediaCodecInfo;
+import android.media.MediaCodecInfo.CodecCapabilities;
import android.media.MediaCodecList;
import android.media.MediaFormat;
+import android.opengl.EGL14;
+import android.opengl.EGLConfig;
+import android.opengl.EGLContext;
+import android.opengl.EGLDisplay;
+import android.opengl.EGLSurface;
+import android.opengl.GLES11Ext;
+import android.opengl.GLES20;
import android.os.Build;
import android.os.Bundle;
import android.util.Log;
+import android.view.Surface;
+
import java.nio.ByteBuffer;
// Java-side of peerconnection_jni.cc:MediaCodecVideoDecoder.
@@ -49,7 +59,7 @@
private static final String TAG = "MediaCodecVideoDecoder";
- private static final int DEQUEUE_TIMEOUT = 1000000; // 1 sec timeout.
+ private static final int DEQUEUE_INPUT_TIMEOUT = 500000; // 500 ms timeout.
private Thread mediaCodecThread;
private MediaCodec mediaCodec;
private ByteBuffer[] inputBuffers;
@@ -57,7 +67,7 @@
private static final String VP8_MIME_TYPE = "video/x-vnd.on2.vp8";
// List of supported HW VP8 decoders.
private static final String[] supportedHwCodecPrefixes =
- {"OMX.Nvidia."};
+ {"OMX.qcom.", "OMX.Nvidia." };
// NV12 color format supported by QCOM codec, but not declared in MediaCodec -
// see /hardware/qcom/media/mm-core/inc/OMX_QCOMExtns.h
private static final int
@@ -74,12 +84,21 @@
private int height;
private int stride;
private int sliceHeight;
+ private boolean useSurface;
+ private int textureID = -1;
+ private SurfaceTexture surfaceTexture = null;
+ private Surface surface = null;
+ private float[] stMatrix = new float[16];
+ private EGLDisplay eglDisplay = EGL14.EGL_NO_DISPLAY;
+ private EGLContext eglContext = EGL14.EGL_NO_CONTEXT;
+ private EGLSurface eglSurface = EGL14.EGL_NO_SURFACE;
+
private MediaCodecVideoDecoder() { }
// Helper struct for findVp8HwDecoder() below.
private static class DecoderProperties {
- DecoderProperties(String codecName, int colorFormat) {
+ public DecoderProperties(String codecName, int colorFormat) {
this.codecName = codecName;
this.colorFormat = colorFormat;
}
@@ -107,26 +126,32 @@
continue; // No VP8 support in this codec; try the next one.
}
Log.d(TAG, "Found candidate decoder " + name);
+
+ // Check if this is supported HW decoder.
+ boolean supportedCodec = false;
+ for (String hwCodecPrefix : supportedHwCodecPrefixes) {
+ if (name.startsWith(hwCodecPrefix)) {
+ supportedCodec = true;
+ break;
+ }
+ }
+ if (!supportedCodec) {
+ continue;
+ }
+
+ // Check if codec supports either yuv420 or nv12.
CodecCapabilities capabilities =
info.getCapabilitiesForType(VP8_MIME_TYPE);
for (int colorFormat : capabilities.colorFormats) {
Log.d(TAG, " Color: 0x" + Integer.toHexString(colorFormat));
}
-
- // Check if this is supported HW decoder
- for (String hwCodecPrefix : supportedHwCodecPrefixes) {
- if (!name.startsWith(hwCodecPrefix)) {
- continue;
- }
- // Check if codec supports either yuv420 or nv12
- for (int supportedColorFormat : supportedColorList) {
- for (int codecColorFormat : capabilities.colorFormats) {
- if (codecColorFormat == supportedColorFormat) {
- // Found supported HW VP8 decoder
- Log.d(TAG, "Found target decoder " + name +
- ". Color: 0x" + Integer.toHexString(codecColorFormat));
- return new DecoderProperties(name, codecColorFormat);
- }
+ for (int supportedColorFormat : supportedColorList) {
+ for (int codecColorFormat : capabilities.colorFormats) {
+ if (codecColorFormat == supportedColorFormat) {
+ // Found supported HW VP8 decoder.
+ Log.d(TAG, "Found target decoder " + name +
+ ". Color: 0x" + Integer.toHexString(codecColorFormat));
+ return new DecoderProperties(name, codecColorFormat);
}
}
}
@@ -146,31 +171,166 @@
}
}
- private boolean initDecode(int width, int height) {
+ private void checkEglError(String msg) {
+ int error;
+ if ((error = EGL14.eglGetError()) != EGL14.EGL_SUCCESS) {
+ Log.e(TAG, msg + ": EGL Error: 0x" + Integer.toHexString(error));
+ throw new RuntimeException(
+ msg + ": EGL error: 0x" + Integer.toHexString(error));
+ }
+ }
+
+ private void checkGlError(String msg) {
+ int error;
+ if ((error = GLES20.glGetError()) != GLES20.GL_NO_ERROR) {
+ Log.e(TAG, msg + ": GL Error: 0x" + Integer.toHexString(error));
+ throw new RuntimeException(
+ msg + ": GL Error: 0x " + Integer.toHexString(error));
+ }
+ }
+
+ private void eglSetup(EGLContext sharedContext, int width, int height) {
+ Log.d(TAG, "EGL setup");
+ if (sharedContext == null) {
+ sharedContext = EGL14.EGL_NO_CONTEXT;
+ }
+ eglDisplay = EGL14.eglGetDisplay(EGL14.EGL_DEFAULT_DISPLAY);
+ if (eglDisplay == EGL14.EGL_NO_DISPLAY) {
+ throw new RuntimeException("Unable to get EGL14 display");
+ }
+ int[] version = new int[2];
+ if (!EGL14.eglInitialize(eglDisplay, version, 0, version, 1)) {
+ throw new RuntimeException("Unable to initialize EGL14");
+ }
+
+ // Configure EGL for pbuffer and OpenGL ES 2.0.
+ int[] attribList = {
+ EGL14.EGL_RED_SIZE, 8,
+ EGL14.EGL_GREEN_SIZE, 8,
+ EGL14.EGL_BLUE_SIZE, 8,
+ EGL14.EGL_RENDERABLE_TYPE, EGL14.EGL_OPENGL_ES2_BIT,
+ EGL14.EGL_SURFACE_TYPE, EGL14.EGL_PBUFFER_BIT,
+ EGL14.EGL_NONE
+ };
+ EGLConfig[] configs = new EGLConfig[1];
+ int[] numConfigs = new int[1];
+ if (!EGL14.eglChooseConfig(eglDisplay, attribList, 0, configs, 0,
+ configs.length, numConfigs, 0)) {
+ throw new RuntimeException("Unable to find RGB888 EGL config");
+ }
+
+ // Configure context for OpenGL ES 2.0.
+ int[] attrib_list = {
+ EGL14.EGL_CONTEXT_CLIENT_VERSION, 2,
+ EGL14.EGL_NONE
+ };
+ eglContext = EGL14.eglCreateContext(eglDisplay, configs[0], sharedContext,
+ attrib_list, 0);
+ checkEglError("eglCreateContext");
+ if (eglContext == null) {
+ throw new RuntimeException("Null EGL context");
+ }
+
+ // Create a pbuffer surface.
+ int[] surfaceAttribs = {
+ EGL14.EGL_WIDTH, width,
+ EGL14.EGL_HEIGHT, height,
+ EGL14.EGL_NONE
+ };
+ eglSurface = EGL14.eglCreatePbufferSurface(eglDisplay, configs[0],
+ surfaceAttribs, 0);
+ checkEglError("eglCreatePbufferSurface");
+ if (eglSurface == null) {
+ throw new RuntimeException("EGL surface was null");
+ }
+ }
+
+ private void eglRelease() {
+ Log.d(TAG, "EGL release");
+ if (eglDisplay != EGL14.EGL_NO_DISPLAY) {
+ EGL14.eglDestroySurface(eglDisplay, eglSurface);
+ EGL14.eglDestroyContext(eglDisplay, eglContext);
+ EGL14.eglReleaseThread();
+ EGL14.eglTerminate(eglDisplay);
+ }
+ eglDisplay = EGL14.EGL_NO_DISPLAY;
+ eglContext = EGL14.EGL_NO_CONTEXT;
+ eglSurface = EGL14.EGL_NO_SURFACE;
+ }
+
+
+ private void makeCurrent() {
+ if (!EGL14.eglMakeCurrent(eglDisplay, eglSurface, eglSurface, eglContext)) {
+ throw new RuntimeException("eglMakeCurrent failed");
+ }
+ }
+
+ private boolean initDecode(int width, int height, boolean useSurface,
+ EGLContext sharedContext) {
if (mediaCodecThread != null) {
throw new RuntimeException("Forgot to release()?");
}
+ if (useSurface && sharedContext == null) {
+ throw new RuntimeException("No shared EGL context.");
+ }
DecoderProperties properties = findVp8HwDecoder();
if (properties == null) {
throw new RuntimeException("Cannot find HW VP8 decoder");
}
Log.d(TAG, "Java initDecode: " + width + " x " + height +
- ". Color: 0x" + Integer.toHexString(properties.colorFormat));
+ ". Color: 0x" + Integer.toHexString(properties.colorFormat) +
+ ". Use Surface: " + useSurface );
+ if (sharedContext != null) {
+ Log.d(TAG, "Decoder shared EGL Context: " + sharedContext);
+ }
mediaCodecThread = Thread.currentThread();
try {
+ Surface decodeSurface = null;
this.width = width;
this.height = height;
+ this.useSurface = useSurface;
stride = width;
sliceHeight = height;
+
+ if (useSurface) {
+ // Create shared EGL context.
+ eglSetup(sharedContext, width, height);
+ makeCurrent();
+
+ // Create output surface
+ int[] textures = new int[1];
+ GLES20.glGenTextures(1, textures, 0);
+ checkGlError("glGenTextures");
+ textureID = textures[0];
+ GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES, textureID);
+ checkGlError("glBindTexture mTextureID");
+
+ GLES20.glTexParameterf(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
+ GLES20.GL_TEXTURE_MIN_FILTER, GLES20.GL_NEAREST);
+ GLES20.glTexParameterf(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
+ GLES20.GL_TEXTURE_MAG_FILTER, GLES20.GL_LINEAR);
+ GLES20.glTexParameteri(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
+ GLES20.GL_TEXTURE_WRAP_S, GLES20.GL_CLAMP_TO_EDGE);
+ GLES20.glTexParameteri(GLES11Ext.GL_TEXTURE_EXTERNAL_OES,
+ GLES20.GL_TEXTURE_WRAP_T, GLES20.GL_CLAMP_TO_EDGE);
+ checkGlError("glTexParameter");
+ Log.d(TAG, "Video decoder TextureID = " + textureID);
+ surfaceTexture = new SurfaceTexture(textureID);
+ surface = new Surface(surfaceTexture);
+ decodeSurface = surface;
+ }
+
MediaFormat format =
MediaFormat.createVideoFormat(VP8_MIME_TYPE, width, height);
- format.setInteger(MediaFormat.KEY_COLOR_FORMAT, properties.colorFormat);
+ if (!useSurface) {
+ format.setInteger(MediaFormat.KEY_COLOR_FORMAT, properties.colorFormat);
+ }
Log.d(TAG, " Format: " + format);
mediaCodec = MediaCodec.createByCodecName(properties.codecName);
if (mediaCodec == null) {
return false;
}
- mediaCodec.configure(format, null, null, 0);
+ mediaCodec.configure(format, decodeSurface, null, 0);
mediaCodec.start();
colorFormat = properties.colorFormat;
outputBuffers = mediaCodec.getOutputBuffers();
@@ -195,6 +355,19 @@
}
mediaCodec = null;
mediaCodecThread = null;
+ if (useSurface) {
+ surface.release();
+ surface = null;
+ surfaceTexture = null;
+ if (textureID >= 0) {
+ int[] textures = new int[1];
+ textures[0] = textureID;
+ Log.d(TAG, "Delete video decoder TextureID " + textureID);
+ GLES20.glDeleteTextures(1, textures, 0);
+ checkGlError("glDeleteTextures");
+ }
+ eglRelease();
+ }
}
// Dequeue an input buffer and return its index, -1 if no input buffer is
@@ -202,7 +375,7 @@
private int dequeueInputBuffer() {
checkOnMediaCodecThread();
try {
- return mediaCodec.dequeueInputBuffer(DEQUEUE_TIMEOUT);
+ return mediaCodec.dequeueInputBuffer(DEQUEUE_INPUT_TIMEOUT);
} catch (IllegalStateException e) {
Log.e(TAG, "dequeueIntputBuffer failed", e);
return -2;
@@ -224,23 +397,40 @@
}
}
+ // Helper struct for dequeueOutputBuffer() below.
+ private static class DecoderOutputBufferInfo {
+ public DecoderOutputBufferInfo(
+ int index, int offset, int size, long presentationTimestampUs) {
+ this.index = index;
+ this.offset = offset;
+ this.size = size;
+ this.presentationTimestampUs = presentationTimestampUs;
+ }
+
+ private final int index;
+ private final int offset;
+ private final int size;
+ private final long presentationTimestampUs;
+ }
+
// Dequeue and return an output buffer index, -1 if no output
// buffer available or -2 if error happened.
- private int dequeueOutputBuffer() {
+ private DecoderOutputBufferInfo dequeueOutputBuffer(int dequeueTimeoutUs) {
checkOnMediaCodecThread();
try {
MediaCodec.BufferInfo info = new MediaCodec.BufferInfo();
- int result = mediaCodec.dequeueOutputBuffer(info, DEQUEUE_TIMEOUT);
+ int result = mediaCodec.dequeueOutputBuffer(info, dequeueTimeoutUs);
while (result == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED ||
result == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
if (result == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
outputBuffers = mediaCodec.getOutputBuffers();
+ Log.d(TAG, "Output buffers changed: " + outputBuffers.length);
} else if (result == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
MediaFormat format = mediaCodec.getOutputFormat();
Log.d(TAG, "Format changed: " + format.toString());
width = format.getInteger(MediaFormat.KEY_WIDTH);
height = format.getInteger(MediaFormat.KEY_HEIGHT);
- if (format.containsKey(MediaFormat.KEY_COLOR_FORMAT)) {
+ if (!useSurface && format.containsKey(MediaFormat.KEY_COLOR_FORMAT)) {
colorFormat = format.getInteger(MediaFormat.KEY_COLOR_FORMAT);
Log.d(TAG, "Color: 0x" + Integer.toHexString(colorFormat));
// Check if new color space is supported.
@@ -253,7 +443,7 @@
}
if (!validColorFormat) {
Log.e(TAG, "Non supported color format");
- return -2;
+ return new DecoderOutputBufferInfo(-1, 0, 0, -1);
}
}
if (format.containsKey("stride")) {
@@ -267,21 +457,28 @@
stride = Math.max(width, stride);
sliceHeight = Math.max(height, sliceHeight);
}
- result = mediaCodec.dequeueOutputBuffer(info, DEQUEUE_TIMEOUT);
+ result = mediaCodec.dequeueOutputBuffer(info, dequeueTimeoutUs);
}
- return result;
+ if (result >= 0) {
+ return new DecoderOutputBufferInfo(result, info.offset, info.size,
+ info.presentationTimeUs);
+ }
+ return null;
} catch (IllegalStateException e) {
Log.e(TAG, "dequeueOutputBuffer failed", e);
- return -2;
+ return new DecoderOutputBufferInfo(-1, 0, 0, -1);
}
}
// Release a dequeued output buffer back to the codec for re-use. Return
// false if the codec is no longer operable.
- private boolean releaseOutputBuffer(int index) {
+ private boolean releaseOutputBuffer(int index, boolean render) {
checkOnMediaCodecThread();
try {
- mediaCodec.releaseOutputBuffer(index, false);
+ if (!useSurface) {
+ render = false;
+ }
+ mediaCodec.releaseOutputBuffer(index, render);
return true;
} catch (IllegalStateException e) {
Log.e(TAG, "releaseOutputBuffer failed", e);
diff --git a/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java b/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
index 45b8d6a..cf11573 100644
--- a/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
+++ b/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
@@ -78,7 +78,7 @@
// Helper struct for findVp8HwEncoder() below.
private static class EncoderProperties {
- EncoderProperties(String codecName, int colorFormat) {
+ public EncoderProperties(String codecName, int colorFormat) {
this.codecName = codecName;
this.colorFormat = colorFormat;
}
@@ -106,26 +106,33 @@
continue; // No VP8 support in this codec; try the next one.
}
Log.d(TAG, "Found candidate encoder " + name);
+
+ // Check if this is supported HW encoder.
+ boolean supportedCodec = false;
+ for (String hwCodecPrefix : supportedHwCodecPrefixes) {
+ if (name.startsWith(hwCodecPrefix)) {
+ supportedCodec = true;
+ break;
+ }
+ }
+ if (!supportedCodec) {
+ continue;
+ }
+
CodecCapabilities capabilities =
info.getCapabilitiesForType(VP8_MIME_TYPE);
for (int colorFormat : capabilities.colorFormats) {
Log.d(TAG, " Color: 0x" + Integer.toHexString(colorFormat));
}
- // Check if this is supported HW encoder
- for (String hwCodecPrefix : supportedHwCodecPrefixes) {
- if (!name.startsWith(hwCodecPrefix)) {
- continue;
- }
- // Check if codec supports either yuv420 or nv12
- for (int supportedColorFormat : supportedColorList) {
- for (int codecColorFormat : capabilities.colorFormats) {
- if (codecColorFormat == supportedColorFormat) {
- // Found supported HW VP8 encoder
- Log.d(TAG, "Found target encoder " + name +
- ". Color: 0x" + Integer.toHexString(codecColorFormat));
- return new EncoderProperties(name, codecColorFormat);
- }
+ // Check if codec supports either yuv420 or nv12.
+ for (int supportedColorFormat : supportedColorList) {
+ for (int codecColorFormat : capabilities.colorFormats) {
+ if (codecColorFormat == supportedColorFormat) {
+ // Found supported HW VP8 encoder.
+ Log.d(TAG, "Found target encoder " + name +
+ ". Color: 0x" + Integer.toHexString(codecColorFormat));
+ return new EncoderProperties(name, codecColorFormat);
}
}
}
@@ -137,15 +144,6 @@
return findVp8HwEncoder() != null;
}
- private static int bitRate(int kbps) {
- // webrtc "kilo" means 1000, not 1024. Apparently.
- // (and the price for overshooting is frame-dropping as webrtc enforces its
- // bandwidth estimation, which is unpleasant).
- // Since the HW encoder in the N5 overshoots, we clamp to a bit less than
- // the requested rate. Sad but true. Bug 3194.
- return kbps * 950;
- }
-
private void checkOnMediaCodecThread() {
if (mediaCodecThread.getId() != Thread.currentThread().getId()) {
throw new RuntimeException(
@@ -170,7 +168,7 @@
try {
MediaFormat format =
MediaFormat.createVideoFormat(VP8_MIME_TYPE, width, height);
- format.setInteger(MediaFormat.KEY_BIT_RATE, bitRate(kbps));
+ format.setInteger(MediaFormat.KEY_BIT_RATE, 1000 * kbps);
format.setInteger("bitrate-mode", VIDEO_ControlRateConstant);
format.setInteger(MediaFormat.KEY_COLOR_FORMAT, properties.colorFormat);
// Default WebRTC settings
@@ -241,7 +239,7 @@
Log.v(TAG, "setRates: " + kbps + " kbps. Fps: " + frameRateIgnored);
try {
Bundle params = new Bundle();
- params.putInt(MediaCodec.PARAMETER_KEY_VIDEO_BITRATE, bitRate(kbps));
+ params.putInt(MediaCodec.PARAMETER_KEY_VIDEO_BITRATE, 1000 * kbps);
mediaCodec.setParameters(params);
return true;
} catch (IllegalStateException e) {
diff --git a/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java b/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
index 441f37b..f9f96e7 100644
--- a/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
+++ b/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
@@ -46,8 +46,12 @@
// Callers may specify either |initializeAudio| or |initializeVideo| as false
// to skip initializing the respective engine (and avoid the need for the
// respective permissions).
+ // |renderEGLContext| can be provided to suport HW video decoding to
+ // texture and will be used to create a shared EGL context on video
+ // decoding thread.
public static native boolean initializeAndroidGlobals(
- Object context, boolean initializeAudio, boolean initializeVideo);
+ Object context, boolean initializeAudio, boolean initializeVideo,
+ Object renderEGLContext);
public PeerConnectionFactory() {
nativeFactory = nativeCreatePeerConnectionFactory();
diff --git a/app/webrtc/java/src/org/webrtc/VideoRenderer.java b/app/webrtc/java/src/org/webrtc/VideoRenderer.java
index 4cc341a..19df137 100644
--- a/app/webrtc/java/src/org/webrtc/VideoRenderer.java
+++ b/app/webrtc/java/src/org/webrtc/VideoRenderer.java
@@ -44,6 +44,9 @@
public final int height;
public final int[] yuvStrides;
public final ByteBuffer[] yuvPlanes;
+ public final boolean yuvFrame;
+ public Object textureObject;
+ public int textureId;
/**
* Construct a frame of the given dimensions with the specified planar
@@ -62,25 +65,72 @@
yuvPlanes[2] = ByteBuffer.allocateDirect(yuvStrides[2] * height);
}
this.yuvPlanes = yuvPlanes;
+ this.yuvFrame = true;
+ }
+
+ /**
+ * Construct a texture frame of the given dimensions with data in SurfaceTexture
+ */
+ public I420Frame(
+ int width, int height, Object textureObject, int textureId) {
+ this.width = width;
+ this.height = height;
+ this.yuvStrides = null;
+ this.yuvPlanes = null;
+ this.textureObject = textureObject;
+ this.textureId = textureId;
+ this.yuvFrame = false;
}
/**
* Copy the planes out of |source| into |this| and return |this|. Calling
- * this with mismatched frame dimensions is a programming error and will
- * likely crash.
+ * this with mismatched frame dimensions or frame type is a programming
+ * error and will likely crash.
*/
public I420Frame copyFrom(I420Frame source) {
- if (!Arrays.equals(yuvStrides, source.yuvStrides) ||
- width != source.width || height != source.height) {
- throw new RuntimeException("Mismatched dimensions! Source: " +
+ if (source.yuvFrame && yuvFrame) {
+ if (!Arrays.equals(yuvStrides, source.yuvStrides) ||
+ width != source.width || height != source.height) {
+ throw new RuntimeException("Mismatched dimensions! Source: " +
+ source.toString() + ", destination: " + toString());
+ }
+ copyPlane(source.yuvPlanes[0], yuvPlanes[0]);
+ copyPlane(source.yuvPlanes[1], yuvPlanes[1]);
+ copyPlane(source.yuvPlanes[2], yuvPlanes[2]);
+ return this;
+ } else if (!source.yuvFrame && !yuvFrame) {
+ textureObject = source.textureObject;
+ textureId = source.textureId;
+ return this;
+ } else {
+ throw new RuntimeException("Mismatched frame types! Source: " +
source.toString() + ", destination: " + toString());
}
- copyPlane(source.yuvPlanes[0], yuvPlanes[0]);
- copyPlane(source.yuvPlanes[1], yuvPlanes[1]);
- copyPlane(source.yuvPlanes[2], yuvPlanes[2]);
- return this;
}
+ public I420Frame copyFrom(byte[] yuvData) {
+ if (yuvData.length < width * height * 3 / 2) {
+ throw new RuntimeException("Wrong arrays size: " + yuvData.length);
+ }
+ if (!yuvFrame) {
+ throw new RuntimeException("Can not feed yuv data to texture frame");
+ }
+ int planeSize = width * height;
+ ByteBuffer[] planes = new ByteBuffer[3];
+ planes[0] = ByteBuffer.wrap(yuvData, 0, planeSize);
+ planes[1] = ByteBuffer.wrap(yuvData, planeSize, planeSize / 4);
+ planes[2] = ByteBuffer.wrap(yuvData, planeSize + planeSize / 4,
+ planeSize / 4);
+ for (int i = 0; i < 3; i++) {
+ yuvPlanes[i].position(0);
+ yuvPlanes[i].put(planes[i]);
+ yuvPlanes[i].position(0);
+ yuvPlanes[i].limit(yuvPlanes[i].capacity());
+ }
+ return this;
+ }
+
+
@Override
public String toString() {
return width + "x" + height + ":" + yuvStrides[0] + ":" + yuvStrides[1] +
@@ -126,11 +176,16 @@
}
public void dispose() {
- free(nativeVideoRenderer);
+ if (callbacks == null) {
+ freeGuiVideoRenderer(nativeVideoRenderer);
+ } else {
+ freeWrappedVideoRenderer(nativeVideoRenderer);
+ }
}
private static native long nativeCreateGuiVideoRenderer(int x, int y);
private static native long nativeWrapVideoRenderer(Callbacks callbacks);
- private static native void free(long nativeVideoRenderer);
+ private static native void freeGuiVideoRenderer(long nativeVideoRenderer);
+ private static native void freeWrappedVideoRenderer(long nativeVideoRenderer);
}
diff --git a/app/webrtc/jsepsessiondescription.cc b/app/webrtc/jsepsessiondescription.cc
index 23a7cdd..c318d91 100644
--- a/app/webrtc/jsepsessiondescription.cc
+++ b/app/webrtc/jsepsessiondescription.cc
@@ -57,11 +57,19 @@
const char SessionDescriptionInterface::kAnswer[] = "answer";
const int JsepSessionDescription::kDefaultVideoCodecId = 100;
-const int JsepSessionDescription::kDefaultVideoCodecFramerate = 30;
+// This is effectively a max value of the frame rate. 30 is default from camera.
+const int JsepSessionDescription::kDefaultVideoCodecFramerate = 60;
const char JsepSessionDescription::kDefaultVideoCodecName[] = "VP8";
// Used as default max video codec size before we have it in signaling.
-const int JsepSessionDescription::kMaxVideoCodecWidth = 3840;
-const int JsepSessionDescription::kMaxVideoCodecHeight = 2160;
+#if defined(ANDROID)
+// Limit default max video codec size for Android to avoid
+// HW VP8 codec initialization failure for resolution higher than 720p.
+const int JsepSessionDescription::kMaxVideoCodecWidth = 1280;
+const int JsepSessionDescription::kMaxVideoCodecHeight = 720;
+#else
+const int JsepSessionDescription::kMaxVideoCodecWidth = 1920;
+const int JsepSessionDescription::kMaxVideoCodecHeight = 1080;
+#endif
const int JsepSessionDescription::kDefaultVideoCodecPreference = 1;
SessionDescriptionInterface* CreateSessionDescription(const std::string& type,
diff --git a/app/webrtc/mediaconstraintsinterface.cc b/app/webrtc/mediaconstraintsinterface.cc
index d59716e..a7b88aa 100644
--- a/app/webrtc/mediaconstraintsinterface.cc
+++ b/app/webrtc/mediaconstraintsinterface.cc
@@ -95,6 +95,8 @@
"googSuspendBelowMinBitrate";
const char MediaConstraintsInterface::kNumUnsignalledRecvStreams[] =
"googNumUnsignalledRecvStreams";
+const char MediaConstraintsInterface::kCombinedAudioVideoBwe[] =
+ "googCombinedAudioVideoBwe";
const char MediaConstraintsInterface::kScreencastMinBitrate[] =
"googScreencastMinBitrate";
// TODO(ronghuawu): Remove once cpu overuse detection is stable.
diff --git a/app/webrtc/mediaconstraintsinterface.h b/app/webrtc/mediaconstraintsinterface.h
index d74ffab..045da79 100644
--- a/app/webrtc/mediaconstraintsinterface.h
+++ b/app/webrtc/mediaconstraintsinterface.h
@@ -117,6 +117,8 @@
// googSuspendBelowMinBitrate
static const char kNumUnsignalledRecvStreams[];
// googNumUnsignalledRecvStreams
+ // Constraint to enable combined audio+video bandwidth estimation.
+ static const char kCombinedAudioVideoBwe[]; // googCombinedAudioVideoBwe
static const char kScreencastMinBitrate[]; // googScreencastMinBitrate
static const char kCpuOveruseDetection[]; // googCpuOveruseDetection
static const char kCpuUnderuseThreshold[]; // googCpuUnderuseThreshold
diff --git a/app/webrtc/mediastreamsignaling.cc b/app/webrtc/mediastreamsignaling.cc
index c4a8281..df51ba1 100644
--- a/app/webrtc/mediastreamsignaling.cc
+++ b/app/webrtc/mediastreamsignaling.cc
@@ -51,9 +51,9 @@
using rtc::scoped_ptr;
using rtc::scoped_refptr;
-static bool ParseConstraints(
+static bool ParseConstraintsForAnswer(
const MediaConstraintsInterface* constraints,
- cricket::MediaSessionOptions* options, bool is_answer) {
+ cricket::MediaSessionOptions* options) {
bool value;
size_t mandatory_constraints_satisfied = 0;
@@ -82,7 +82,7 @@
// kOfferToReceiveVideo defaults to false according to spec. But
// if it is an answer and video is offered, we should still accept video
// per default.
- options->has_video |= is_answer;
+ options->has_video = true;
}
if (FindConstraint(constraints,
@@ -133,6 +133,55 @@
(value <= Options::kMaxOfferToReceiveMedia);
}
+// Add the stream and RTP data channel info to |session_options|.
+static void SetStreams(
+ cricket::MediaSessionOptions* session_options,
+ rtc::scoped_refptr<StreamCollection> streams,
+ const MediaStreamSignaling::RtpDataChannels& rtp_data_channels) {
+ session_options->streams.clear();
+ if (streams != NULL) {
+ for (size_t i = 0; i < streams->count(); ++i) {
+ MediaStreamInterface* stream = streams->at(i);
+
+ AudioTrackVector audio_tracks(stream->GetAudioTracks());
+
+ // For each audio track in the stream, add it to the MediaSessionOptions.
+ for (size_t j = 0; j < audio_tracks.size(); ++j) {
+ scoped_refptr<MediaStreamTrackInterface> track(audio_tracks[j]);
+ session_options->AddStream(
+ cricket::MEDIA_TYPE_AUDIO, track->id(), stream->label());
+ }
+
+ VideoTrackVector video_tracks(stream->GetVideoTracks());
+
+ // For each video track in the stream, add it to the MediaSessionOptions.
+ for (size_t j = 0; j < video_tracks.size(); ++j) {
+ scoped_refptr<MediaStreamTrackInterface> track(video_tracks[j]);
+ session_options->AddStream(
+ cricket::MEDIA_TYPE_VIDEO, track->id(), stream->label());
+ }
+ }
+ }
+
+ // Check for data channels.
+ MediaStreamSignaling::RtpDataChannels::const_iterator data_channel_it =
+ rtp_data_channels.begin();
+ for (; data_channel_it != rtp_data_channels.end(); ++data_channel_it) {
+ const DataChannel* channel = data_channel_it->second;
+ if (channel->state() == DataChannel::kConnecting ||
+ channel->state() == DataChannel::kOpen) {
+ // |streamid| and |sync_label| are both set to the DataChannel label
+ // here so they can be signaled the same way as MediaStreams and Tracks.
+ // For MediaStreams, the sync_label is the MediaStream label and the
+ // track label is the same as |streamid|.
+ const std::string& streamid = channel->label();
+ const std::string& sync_label = channel->label();
+ session_options->AddStream(
+ cricket::MEDIA_TYPE_DATA, streamid, sync_label);
+ }
+ }
+}
+
// Factory class for creating remote MediaStreams and MediaStreamTracks.
class RemoteMediaStreamFactory {
public:
@@ -192,8 +241,6 @@
channel_manager)),
last_allocated_sctp_even_sid_(-2),
last_allocated_sctp_odd_sid_(-1) {
- options_.has_video = false;
- options_.has_audio = false;
}
MediaStreamSignaling::~MediaStreamSignaling() {
@@ -279,12 +326,13 @@
LOG(LS_ERROR) << "Failed to create DataChannel from the OPEN message.";
return false;
}
- sctp_data_channels_.push_back(channel);
+
stream_observer_->OnAddDataChannel(channel);
return true;
}
void MediaStreamSignaling::RemoveSctpDataChannel(int sid) {
+ ASSERT(sid >= 0);
for (SctpDataChannels::iterator iter = sctp_data_channels_.begin();
iter != sctp_data_channels_.end();
++iter) {
@@ -377,40 +425,38 @@
return false;
}
- UpdateSessionOptions();
+ session_options->has_audio = false;
+ session_options->has_video = false;
+ SetStreams(session_options, local_streams_, rtp_data_channels_);
- // |options.has_audio| and |options.has_video| can only change from false to
- // true, but never change from true to false. This is to make sure
- // CreateOffer / CreateAnswer doesn't remove a media content
- // description that has been created.
- if (rtc_options.offer_to_receive_audio > 0) {
- options_.has_audio = true;
+ // If |offer_to_receive_[audio/video]| is undefined, respect the flags set
+ // from SetStreams. Otherwise, overwrite it based on |rtc_options|.
+ if (rtc_options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
+ session_options->has_audio = rtc_options.offer_to_receive_audio > 0;
}
- if (rtc_options.offer_to_receive_video > 0) {
- options_.has_video = true;
+ if (rtc_options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
+ session_options->has_video = rtc_options.offer_to_receive_video > 0;
}
- options_.vad_enabled = rtc_options.voice_activity_detection;
- options_.transport_options.ice_restart = rtc_options.ice_restart;
- options_.bundle_enabled = rtc_options.use_rtp_mux;
- options_.bundle_enabled = EvaluateNeedForBundle(options_);
- *session_options = options_;
+ session_options->vad_enabled = rtc_options.voice_activity_detection;
+ session_options->transport_options.ice_restart = rtc_options.ice_restart;
+ session_options->bundle_enabled = rtc_options.use_rtp_mux;
+
+ session_options->bundle_enabled = EvaluateNeedForBundle(*session_options);
return true;
}
bool MediaStreamSignaling::GetOptionsForAnswer(
const MediaConstraintsInterface* constraints,
cricket::MediaSessionOptions* options) {
- UpdateSessionOptions();
+ options->has_audio = false;
+ options->has_video = false;
+ SetStreams(options, local_streams_, rtp_data_channels_);
- // Copy the |options_| to not let the flag MediaSessionOptions::has_audio and
- // MediaSessionOptions::has_video affect subsequent offers.
- cricket::MediaSessionOptions current_options = options_;
- if (!ParseConstraints(constraints, ¤t_options, true)) {
+ if (!ParseConstraintsForAnswer(constraints, options)) {
return false;
}
- current_options.bundle_enabled = EvaluateNeedForBundle(current_options);
- *options = current_options;
+ options->bundle_enabled = EvaluateNeedForBundle(*options);
return true;
}
@@ -545,54 +591,6 @@
}
}
-void MediaStreamSignaling::UpdateSessionOptions() {
- options_.streams.clear();
- if (local_streams_ != NULL) {
- for (size_t i = 0; i < local_streams_->count(); ++i) {
- MediaStreamInterface* stream = local_streams_->at(i);
-
- AudioTrackVector audio_tracks(stream->GetAudioTracks());
- if (!audio_tracks.empty()) {
- options_.has_audio = true;
- }
-
- // For each audio track in the stream, add it to the MediaSessionOptions.
- for (size_t j = 0; j < audio_tracks.size(); ++j) {
- scoped_refptr<MediaStreamTrackInterface> track(audio_tracks[j]);
- options_.AddStream(cricket::MEDIA_TYPE_AUDIO, track->id(),
- stream->label());
- }
-
- VideoTrackVector video_tracks(stream->GetVideoTracks());
- if (!video_tracks.empty()) {
- options_.has_video = true;
- }
- // For each video track in the stream, add it to the MediaSessionOptions.
- for (size_t j = 0; j < video_tracks.size(); ++j) {
- scoped_refptr<MediaStreamTrackInterface> track(video_tracks[j]);
- options_.AddStream(cricket::MEDIA_TYPE_VIDEO, track->id(),
- stream->label());
- }
- }
- }
-
- // Check for data channels.
- RtpDataChannels::const_iterator data_channel_it = rtp_data_channels_.begin();
- for (; data_channel_it != rtp_data_channels_.end(); ++data_channel_it) {
- const DataChannel* channel = data_channel_it->second;
- if (channel->state() == DataChannel::kConnecting ||
- channel->state() == DataChannel::kOpen) {
- // |streamid| and |sync_label| are both set to the DataChannel label
- // here so they can be signaled the same way as MediaStreams and Tracks.
- // For MediaStreams, the sync_label is the MediaStream label and the
- // track label is the same as |streamid|.
- const std::string& streamid = channel->label();
- const std::string& sync_label = channel->label();
- options_.AddStream(cricket::MEDIA_TYPE_DATA, streamid, sync_label);
- }
- }
-}
-
void MediaStreamSignaling::UpdateRemoteStreamsList(
const cricket::StreamParamsVec& streams,
cricket::MediaType media_type,
diff --git a/app/webrtc/mediastreamsignaling.h b/app/webrtc/mediastreamsignaling.h
index 7f17971..d4b1be8 100644
--- a/app/webrtc/mediastreamsignaling.h
+++ b/app/webrtc/mediastreamsignaling.h
@@ -160,6 +160,9 @@
class MediaStreamSignaling : public sigslot::has_slots<> {
public:
+ typedef std::map<std::string, rtc::scoped_refptr<DataChannel> >
+ RtpDataChannels;
+
MediaStreamSignaling(rtc::Thread* signaling_thread,
MediaStreamSignalingObserver* stream_observer,
cricket::ChannelManager* channel_manager);
@@ -289,8 +292,6 @@
};
typedef std::vector<TrackInfo> TrackInfos;
- void UpdateSessionOptions();
-
// Makes sure a MediaStream Track is created for each StreamParam in
// |streams|. |media_type| is the type of the |streams| and can be either
// audio or video.
@@ -378,7 +379,6 @@
RemotePeerInfo remote_info_;
rtc::Thread* signaling_thread_;
DataChannelFactory* data_channel_factory_;
- cricket::MediaSessionOptions options_;
MediaStreamSignalingObserver* stream_observer_;
rtc::scoped_refptr<StreamCollection> local_streams_;
rtc::scoped_refptr<StreamCollection> remote_streams_;
@@ -392,8 +392,6 @@
int last_allocated_sctp_even_sid_;
int last_allocated_sctp_odd_sid_;
- typedef std::map<std::string, rtc::scoped_refptr<DataChannel> >
- RtpDataChannels;
typedef std::vector<rtc::scoped_refptr<DataChannel> > SctpDataChannels;
RtpDataChannels rtp_data_channels_;
diff --git a/app/webrtc/mediastreamsignaling_unittest.cc b/app/webrtc/mediastreamsignaling_unittest.cc
index fa83646..84f67b9 100644
--- a/app/webrtc/mediastreamsignaling_unittest.cc
+++ b/app/webrtc/mediastreamsignaling_unittest.cc
@@ -261,14 +261,20 @@
class FakeDataChannelFactory : public webrtc::DataChannelFactory {
public:
FakeDataChannelFactory(FakeDataChannelProvider* provider,
- cricket::DataChannelType dct)
- : provider_(provider), type_(dct) {}
+ cricket::DataChannelType dct,
+ webrtc::MediaStreamSignaling* media_stream_signaling)
+ : provider_(provider),
+ type_(dct),
+ media_stream_signaling_(media_stream_signaling) {}
virtual rtc::scoped_refptr<webrtc::DataChannel> CreateDataChannel(
const std::string& label,
const webrtc::InternalDataChannelInit* config) {
last_init_ = *config;
- return webrtc::DataChannel::Create(provider_, type_, label, *config);
+ rtc::scoped_refptr<webrtc::DataChannel> data_channel =
+ webrtc::DataChannel::Create(provider_, type_, label, *config);
+ media_stream_signaling_->AddDataChannel(data_channel);
+ return data_channel;
}
const webrtc::InternalDataChannelInit& last_init() const {
@@ -278,6 +284,7 @@
private:
FakeDataChannelProvider* provider_;
cricket::DataChannelType type_;
+ webrtc::MediaStreamSignaling* media_stream_signaling_;
webrtc::InternalDataChannelInit last_init_;
};
@@ -775,8 +782,10 @@
RTCOfferAnswerOptions default_rtc_options;
EXPECT_TRUE(signaling_->GetOptionsForOffer(default_rtc_options,
&updated_offer_options));
- EXPECT_TRUE(updated_offer_options.has_audio);
- EXPECT_TRUE(updated_offer_options.has_video);
+ // By default, |has_audio| or |has_video| are false if there is no media
+ // track.
+ EXPECT_FALSE(updated_offer_options.has_audio);
+ EXPECT_FALSE(updated_offer_options.has_video);
}
// This test verifies that the remote MediaStreams corresponding to a received
@@ -1263,7 +1272,8 @@
// message.
TEST_F(MediaStreamSignalingTest, CreateDataChannelFromOpenMessage) {
FakeDataChannelFactory fake_factory(data_channel_provider_.get(),
- cricket::DCT_SCTP);
+ cricket::DCT_SCTP,
+ signaling_.get());
signaling_->SetDataChannelFactory(&fake_factory);
webrtc::DataChannelInit config;
config.id = 1;
@@ -1283,7 +1293,8 @@
AddDataChannel(cricket::DCT_SCTP, "a", -1);
FakeDataChannelFactory fake_factory(data_channel_provider_.get(),
- cricket::DCT_SCTP);
+ cricket::DCT_SCTP,
+ signaling_.get());
signaling_->SetDataChannelFactory(&fake_factory);
webrtc::DataChannelInit config;
config.id = 0;
@@ -1312,3 +1323,24 @@
signaling_->OnRemoteSctpDataChannelClosed(config.id);
EXPECT_EQ(webrtc::DataChannelInterface::kClosed, data_channel->state());
}
+
+// Verifies that DataChannel added from OPEN message is added to
+// MediaStreamSignaling only once (webrtc issue 3778).
+TEST_F(MediaStreamSignalingTest, DataChannelFromOpenMessageAddedOnce) {
+ FakeDataChannelFactory fake_factory(data_channel_provider_.get(),
+ cricket::DCT_SCTP,
+ signaling_.get());
+ signaling_->SetDataChannelFactory(&fake_factory);
+ webrtc::DataChannelInit config;
+ config.id = 1;
+ rtc::Buffer payload;
+ webrtc::WriteDataChannelOpenMessage("a", config, &payload);
+ cricket::ReceiveDataParams params;
+ params.ssrc = config.id;
+ EXPECT_TRUE(signaling_->AddDataChannelFromOpenMessage(params, payload));
+ EXPECT_TRUE(signaling_->HasDataChannels());
+
+ // Removes the DataChannel and verifies that no DataChannel is left.
+ signaling_->RemoveSctpDataChannel(config.id);
+ EXPECT_FALSE(signaling_->HasDataChannels());
+}
diff --git a/app/webrtc/objc/RTCEAGLVideoView.m b/app/webrtc/objc/RTCEAGLVideoView.m
index 5365d98..faacef6 100644
--- a/app/webrtc/objc/RTCEAGLVideoView.m
+++ b/app/webrtc/objc/RTCEAGLVideoView.m
@@ -173,6 +173,7 @@
return;
}
[_videoTrack removeRenderer:_videoRenderer];
+ self.i420Frame = nil;
_videoTrack = videoTrack;
[_videoTrack addRenderer:_videoRenderer];
// TODO(tkchin): potentially handle changes in track state - e.g. render
@@ -191,11 +192,9 @@
// This method is called when the GLKView's content is dirty and needs to be
// redrawn. This occurs on main thread.
- (void)glkView:(GLKView*)view drawInRect:(CGRect)rect {
- if (self.i420Frame) {
- // The renderer will draw the frame to the framebuffer corresponding to the
- // one used by |view|.
- [_glRenderer drawFrame:self.i420Frame];
- }
+ // The renderer will draw the frame to the framebuffer corresponding to the
+ // one used by |view|.
+ [_glRenderer drawFrame:self.i420Frame];
}
#pragma mark - Private
diff --git a/app/webrtc/objc/RTCNSGLVideoView.m b/app/webrtc/objc/RTCNSGLVideoView.m
index 39f3678..292e792 100644
--- a/app/webrtc/objc/RTCNSGLVideoView.m
+++ b/app/webrtc/objc/RTCNSGLVideoView.m
@@ -116,6 +116,9 @@
if (_videoTrack) {
[_videoTrack removeRenderer:_videoRenderer];
CVDisplayLinkStop(_displayLink);
+ // Clear contents.
+ self.i420Frame = nil;
+ [self drawFrame];
}
_videoTrack = videoTrack;
if (_videoTrack) {
@@ -144,7 +147,7 @@
- (void)drawFrame {
RTCI420Frame* i420Frame = self.i420Frame;
- if (i420Frame && self.glRenderer.lastDrawnFrame != i420Frame) {
+ if (self.glRenderer.lastDrawnFrame != i420Frame) {
// This method may be called from CVDisplayLink callback which isn't on the
// main thread so we have to lock the GL context before drawing.
CGLLockContext([[self openGLContext] CGLContextObj]);
diff --git a/app/webrtc/objc/RTCOpenGLVideoRenderer.mm b/app/webrtc/objc/RTCOpenGLVideoRenderer.mm
index 9ee0216..5a24cf0 100644
--- a/app/webrtc/objc/RTCOpenGLVideoRenderer.mm
+++ b/app/webrtc/objc/RTCOpenGLVideoRenderer.mm
@@ -205,16 +205,18 @@
return NO;
}
[self ensureGLContext];
- if (![self updateTextureSizesForFrame:frame] ||
- ![self updateTextureDataForFrame:frame]) {
- return NO;
- }
glClear(GL_COLOR_BUFFER_BIT);
+ if (frame) {
+ if (![self updateTextureSizesForFrame:frame] ||
+ ![self updateTextureDataForFrame:frame]) {
+ return NO;
+ }
#if !TARGET_OS_IPHONE
- glBindVertexArray(_vertexArray);
+ glBindVertexArray(_vertexArray);
#endif
- glBindBuffer(GL_ARRAY_BUFFER, _vertexBuffer);
- glDrawArrays(GL_TRIANGLE_FAN, 0, 4);
+ glBindBuffer(GL_ARRAY_BUFFER, _vertexBuffer);
+ glDrawArrays(GL_TRIANGLE_FAN, 0, 4);
+ }
#if !TARGET_OS_IPHONE
[_context flushBuffer];
#endif
@@ -238,7 +240,6 @@
}
glUseProgram(_program);
glPixelStorei(GL_UNPACK_ALIGNMENT, 1);
- glClearColor(0, 0, 0, 1);
_isInitialized = YES;
}
diff --git a/app/webrtc/objc/RTCVideoRenderer.mm b/app/webrtc/objc/RTCVideoRenderer.mm
index de03a1e..4cfe43a 100644
--- a/app/webrtc/objc/RTCVideoRenderer.mm
+++ b/app/webrtc/objc/RTCVideoRenderer.mm
@@ -30,10 +30,6 @@
#endif
#import "RTCVideoRenderer+Internal.h"
-
-#if TARGET_OS_IPHONE
-#import "RTCEAGLVideoView+Internal.h"
-#endif
#import "RTCI420Frame+Internal.h"
namespace webrtc {
@@ -62,9 +58,6 @@
@implementation RTCVideoRenderer {
rtc::scoped_ptr<webrtc::RTCVideoRendererAdapter> _adapter;
-#if TARGET_OS_IPHONE
- RTCEAGLVideoView* _videoView;
-#endif
}
- (instancetype)initWithDelegate:(id<RTCVideoRendererDelegate>)delegate {
@@ -75,22 +68,6 @@
return self;
}
-#if TARGET_OS_IPHONE
-// TODO(tkchin): remove shim for deprecated method.
-- (instancetype)initWithView:(UIView*)view {
- if (self = [super init]) {
- _videoView = [[RTCEAGLVideoView alloc] initWithFrame:view.bounds];
- _videoView.autoresizingMask =
- UIViewAutoresizingFlexibleHeight | UIViewAutoresizingFlexibleWidth;
- _videoView.translatesAutoresizingMaskIntoConstraints = YES;
- [view addSubview:_videoView];
- self.delegate = _videoView;
- _adapter.reset(new webrtc::RTCVideoRendererAdapter(self));
- }
- return self;
-}
-#endif
-
@end
@implementation RTCVideoRenderer (Internal)
diff --git a/app/webrtc/objc/public/RTCVideoRenderer.h b/app/webrtc/objc/public/RTCVideoRenderer.h
index f78746c..37977ce 100644
--- a/app/webrtc/objc/public/RTCVideoRenderer.h
+++ b/app/webrtc/objc/public/RTCVideoRenderer.h
@@ -55,12 +55,6 @@
// of frames.
- (instancetype)initWithDelegate:(id<RTCVideoRendererDelegate>)delegate;
-#if TARGET_OS_IPHONE
-// DEPRECATED. See https://code.google.com/p/webrtc/issues/detail?id=3341 for
-// details.
-- (instancetype)initWithView:(UIView*)view;
-#endif
-
#ifndef DOXYGEN_SHOULD_SKIP_THIS
// Disallow init and don't add to documentation
- (id)init __attribute__((
diff --git a/app/webrtc/peerconnection.cc b/app/webrtc/peerconnection.cc
index 201269a..d939f98 100644
--- a/app/webrtc/peerconnection.cc
+++ b/app/webrtc/peerconnection.cc
@@ -39,6 +39,7 @@
#include "talk/session/media/channelmanager.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
+#include "webrtc/system_wrappers/interface/field_trial.h"
namespace {
@@ -353,10 +354,14 @@
cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET;
bool value;
+ // If IPv6 flag was specified, we'll not override it by experiment.
if (FindConstraint(
- constraints,
- MediaConstraintsInterface::kEnableIPv6,
- &value, NULL) && value) {
+ constraints, MediaConstraintsInterface::kEnableIPv6, &value, NULL)) {
+ if (value) {
+ portallocator_flags |= cricket::PORTALLOCATOR_ENABLE_IPV6;
+ }
+ } else if (webrtc::field_trial::FindFullName("WebRTC-IPv6Default") ==
+ "Enabled") {
portallocator_flags |= cricket::PORTALLOCATOR_ENABLE_IPV6;
}
@@ -508,10 +513,6 @@
return;
}
RTCOfferAnswerOptions options;
- // Defaults to receiving audio and not receiving video.
- options.offer_to_receive_audio =
- RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
- options.offer_to_receive_video = 0;
bool value;
size_t mandatory_constraints = 0;
@@ -672,7 +673,7 @@
}
}
}
- return session_->UpdateIce(config.type);
+ return session_->SetIceTransports(config.type);
}
bool PeerConnection::AddIceCandidate(
diff --git a/app/webrtc/peerconnection_unittest.cc b/app/webrtc/peerconnection_unittest.cc
index ca022c6..977fc11 100644
--- a/app/webrtc/peerconnection_unittest.cc
+++ b/app/webrtc/peerconnection_unittest.cc
@@ -90,6 +90,7 @@
// warnings.
#if !defined(THREAD_SANITIZER)
static const int kMaxWaitForStatsMs = 3000;
+static const int kMaxWaitForRembMs = 5000;
#endif
static const int kMaxWaitForFramesMs = 10000;
static const int kEndAudioFrameCount = 3;
@@ -154,11 +155,11 @@
}
void AddMediaStream(bool audio, bool video) {
- std::string label = kStreamLabelBase +
+ std::string stream_label = kStreamLabelBase +
rtc::ToString<int>(
static_cast<int>(peer_connection_->local_streams()->count()));
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
- peer_connection_factory_->CreateLocalMediaStream(label);
+ peer_connection_factory_->CreateLocalMediaStream(stream_label);
if (audio && can_receive_audio()) {
FakeConstraints constraints;
@@ -169,13 +170,13 @@
peer_connection_factory_->CreateAudioSource(&constraints);
// TODO(perkj): Test audio source when it is implemented. Currently audio
// always use the default input.
+ std::string label = stream_label + kAudioTrackLabelBase;
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
- peer_connection_factory_->CreateAudioTrack(kAudioTrackLabelBase,
- source));
+ peer_connection_factory_->CreateAudioTrack(label, source));
stream->AddTrack(audio_track);
}
if (video && can_receive_video()) {
- stream->AddTrack(CreateLocalVideoTrack(label));
+ stream->AddTrack(CreateLocalVideoTrack(stream_label));
}
EXPECT_TRUE(peer_connection_->AddStream(stream, NULL));
@@ -368,6 +369,16 @@
return observer->BytesSent();
}
+ int GetAvailableReceivedBandwidthStats() {
+ rtc::scoped_refptr<MockStatsObserver>
+ observer(new rtc::RefCountedObject<MockStatsObserver>());
+ EXPECT_TRUE(peer_connection_->GetStats(
+ observer, NULL, PeerConnectionInterface::kStatsOutputLevelStandard));
+ EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
+ int bw = observer->AvailableReceiveBandwidth();
+ return bw;
+ }
+
int rendered_width() {
EXPECT_FALSE(fake_video_renderers_.empty());
return fake_video_renderers_.empty() ? 1 :
@@ -447,6 +458,12 @@
webrtc::PeerConnectionInterface* pc() {
return peer_connection_.get();
}
+ void StopVideoCapturers() {
+ for (std::vector<cricket::VideoCapturer*>::iterator it =
+ video_capturers_.begin(); it != video_capturers_.end(); ++it) {
+ (*it)->Stop();
+ }
+ }
protected:
explicit PeerConnectionTestClientBase(const std::string& id)
@@ -464,9 +481,8 @@
if (!allocator_factory_) {
return false;
}
- audio_thread_.Start();
fake_audio_capture_module_ = FakeAudioCaptureModule::Create(
- &audio_thread_);
+ rtc::Thread::Current());
if (fake_audio_capture_module_ == NULL) {
return false;
@@ -529,21 +545,17 @@
FakeConstraints source_constraints = video_constraints_;
source_constraints.SetMandatoryMaxFrameRate(10);
+ cricket::FakeVideoCapturer* fake_capturer =
+ new webrtc::FakePeriodicVideoCapturer();
+ video_capturers_.push_back(fake_capturer);
rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
peer_connection_factory_->CreateVideoSource(
- new webrtc::FakePeriodicVideoCapturer(),
- &source_constraints);
+ fake_capturer, &source_constraints);
std::string label = stream_label + kVideoTrackLabelBase;
return peer_connection_factory_->CreateVideoTrack(label, source);
}
std::string id_;
- // Separate thread for executing |fake_audio_capture_module_| tasks. Audio
- // processing must not be performed on the same thread as signaling due to
- // signaling time constraints and relative complexity of the audio pipeline.
- // This is consistent with the video pipeline that us a a separate thread for
- // encoding and decoding.
- rtc::Thread audio_thread_;
rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
allocator_factory_;
@@ -569,6 +581,10 @@
// For remote peer communication.
MessageReceiver* signaling_message_receiver_;
+
+ // Store references to the video capturers we've created, so that we can stop
+ // them, if required.
+ std::vector<cricket::VideoCapturer*> video_capturers_;
};
class JsepTestClient
@@ -591,7 +607,7 @@
}
virtual void Negotiate(bool audio, bool video) {
rtc::scoped_ptr<SessionDescriptionInterface> offer;
- EXPECT_TRUE(DoCreateOffer(offer.use()));
+ ASSERT_TRUE(DoCreateOffer(offer.use()));
if (offer->description()->GetContentByName("audio")) {
offer->description()->GetContentByName("audio")->rejected = !audio;
@@ -1019,6 +1035,30 @@
}
}
+ // Wait until 'size' bytes of audio has been seen by the receiver, on the
+ // first audio stream.
+ void WaitForAudioData(int size) {
+ const int kMaxWaitForAudioDataMs = 10000;
+
+ StreamCollectionInterface* local_streams =
+ initializing_client()->local_streams();
+ ASSERT_GT(local_streams->count(), 0u);
+ ASSERT_GT(local_streams->at(0)->GetAudioTracks().size(), 0u);
+ MediaStreamTrackInterface* local_audio_track =
+ local_streams->at(0)->GetAudioTracks()[0];
+
+ // Wait until *any* audio has been received.
+ EXPECT_TRUE_WAIT(
+ receiving_client()->GetBytesReceivedStats(local_audio_track) > 0,
+ kMaxWaitForAudioDataMs);
+
+ // Wait until 'size' number of bytes have been received.
+ size += receiving_client()->GetBytesReceivedStats(local_audio_track);
+ EXPECT_TRUE_WAIT(
+ receiving_client()->GetBytesReceivedStats(local_audio_track) > size,
+ kMaxWaitForAudioDataMs);
+ }
+
SignalingClass* initializing_client() { return initiating_client_.get(); }
SignalingClass* receiving_client() { return receiving_client_.get(); }
@@ -1314,6 +1354,7 @@
// Unregister the existing observer.
receiving_client()->data_channel()->UnregisterObserver();
+
std::string data = "hello world";
SendRtpData(initializing_client()->data_channel(), data);
@@ -1437,4 +1478,69 @@
EnableVideoDecoderFactory();
LocalP2PTest();
}
+
+// Test receive bandwidth stats with only audio enabled at receiver.
+TEST_F(JsepPeerConnectionP2PTestClient, ReceivedBweStatsAudio) {
+ ASSERT_TRUE(CreateTestClients());
+ receiving_client()->SetReceiveAudioVideo(true, false);
+ LocalP2PTest();
+
+ // Wait until we have received some audio data. Following REMB shoud be zero.
+ WaitForAudioData(10000);
+ EXPECT_EQ_WAIT(
+ receiving_client()->GetAvailableReceivedBandwidthStats(), 0,
+ kMaxWaitForRembMs);
+}
+
+// Test receive bandwidth stats with combined BWE.
+TEST_F(JsepPeerConnectionP2PTestClient, ReceivedBweStatsCombined) {
+ FakeConstraints setup_constraints;
+ setup_constraints.AddOptional(
+ MediaConstraintsInterface::kCombinedAudioVideoBwe, true);
+ ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
+ initializing_client()->AddMediaStream(true, true);
+ initializing_client()->AddMediaStream(false, true);
+ initializing_client()->AddMediaStream(false, true);
+ initializing_client()->AddMediaStream(false, true);
+ LocalP2PTest();
+
+ // Run until a non-zero bw is reported.
+ EXPECT_TRUE_WAIT(receiving_client()->GetAvailableReceivedBandwidthStats() > 0,
+ kMaxWaitForRembMs);
+
+ // Halt video capturers, then run until we have gotten some audio. Following
+ // REMB should be non-zero.
+ initializing_client()->StopVideoCapturers();
+ WaitForAudioData(10000);
+ EXPECT_TRUE_WAIT(
+ receiving_client()->GetAvailableReceivedBandwidthStats() > 0,
+ kMaxWaitForRembMs);
+}
+
+// Test receive bandwidth stats with 1 video, 3 audio streams but no combined
+// BWE.
+TEST_F(JsepPeerConnectionP2PTestClient, ReceivedBweStatsNotCombined) {
+ FakeConstraints setup_constraints;
+ setup_constraints.AddOptional(
+ MediaConstraintsInterface::kCombinedAudioVideoBwe, false);
+ ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
+ initializing_client()->AddMediaStream(true, true);
+ initializing_client()->AddMediaStream(false, true);
+ initializing_client()->AddMediaStream(false, true);
+ initializing_client()->AddMediaStream(false, true);
+ LocalP2PTest();
+
+ // Run until a non-zero bw is reported.
+ EXPECT_TRUE_WAIT(receiving_client()->GetAvailableReceivedBandwidthStats() > 0,
+ kMaxWaitForRembMs);
+
+ // Halt video capturers, then run until we have gotten some audio. Following
+ // REMB should be zero.
+ initializing_client()->StopVideoCapturers();
+ WaitForAudioData(10000);
+ EXPECT_EQ_WAIT(
+ receiving_client()->GetAvailableReceivedBandwidthStats(), 0,
+ kMaxWaitForRembMs);
+}
+
#endif // if !defined(THREAD_SANITIZER)
diff --git a/app/webrtc/peerconnectionfactory.cc b/app/webrtc/peerconnectionfactory.cc
index 5dccba8..862ceda 100644
--- a/app/webrtc/peerconnectionfactory.cc
+++ b/app/webrtc/peerconnectionfactory.cc
@@ -41,6 +41,7 @@
#include "talk/media/webrtc/webrtcmediaengine.h"
#include "talk/media/webrtc/webrtcvideodecoderfactory.h"
#include "talk/media/webrtc/webrtcvideoencoderfactory.h"
+#include "webrtc/base/bind.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
using rtc::scoped_refptr;
diff --git a/app/webrtc/statstypes.h b/app/webrtc/statstypes.h
index 8eae1ad..656b83c 100644
--- a/app/webrtc/statstypes.h
+++ b/app/webrtc/statstypes.h
@@ -36,6 +36,7 @@
#include <vector>
#include "webrtc/base/basictypes.h"
+#include "webrtc/base/common.h"
#include "webrtc/base/stringencode.h"
namespace webrtc {
diff --git a/app/webrtc/test/fakeaudiocapturemodule.cc b/app/webrtc/test/fakeaudiocapturemodule.cc
index ff45f14..2ad3f0f 100644
--- a/app/webrtc/test/fakeaudiocapturemodule.cc
+++ b/app/webrtc/test/fakeaudiocapturemodule.cc
@@ -94,13 +94,6 @@
return frames_received_;
}
-int32_t FakeAudioCaptureModule::Version(char* /*version*/,
- uint32_t& /*remaining_buffer_in_bytes*/,
- uint32_t& /*position*/) const {
- ASSERT(false);
- return 0;
-}
-
int32_t FakeAudioCaptureModule::TimeUntilNextProcess() {
const uint32 current_time = rtc::Time();
if (current_time < last_process_time_ms_) {
@@ -325,12 +318,6 @@
return 0;
}
-int32_t FakeAudioCaptureModule::SpeakerIsAvailable(bool* available) {
- // No speaker, just dropping audio. Return success.
- *available = true;
- return 0;
-}
-
int32_t FakeAudioCaptureModule::InitSpeaker() {
// No speaker, just playing from file. Return success.
return 0;
@@ -341,12 +328,6 @@
return 0;
}
-int32_t FakeAudioCaptureModule::MicrophoneIsAvailable(bool* available) {
- // No microphone, just playing from file. Return success.
- *available = true;
- return 0;
-}
-
int32_t FakeAudioCaptureModule::InitMicrophone() {
// No microphone, just playing from file. Return success.
return 0;
diff --git a/app/webrtc/test/fakeaudiocapturemodule.h b/app/webrtc/test/fakeaudiocapturemodule.h
index aec3e5e..79b72b6 100644
--- a/app/webrtc/test/fakeaudiocapturemodule.h
+++ b/app/webrtc/test/fakeaudiocapturemodule.h
@@ -76,133 +76,132 @@
// Only functions called by PeerConnection are implemented, the rest do
// nothing and return success. If a function is not expected to be called by
// PeerConnection an assertion is triggered if it is in fact called.
- virtual int32_t Version(char* version,
- uint32_t& remaining_buffer_in_bytes,
- uint32_t& position) const;
- virtual int32_t TimeUntilNextProcess();
- virtual int32_t Process();
- virtual int32_t ChangeUniqueId(const int32_t id);
+ virtual int32_t TimeUntilNextProcess() OVERRIDE;
+ virtual int32_t Process() OVERRIDE;
+ virtual int32_t ChangeUniqueId(const int32_t id) OVERRIDE;
- virtual int32_t ActiveAudioLayer(AudioLayer* audio_layer) const;
+ virtual int32_t ActiveAudioLayer(AudioLayer* audio_layer) const OVERRIDE;
- virtual ErrorCode LastError() const;
+ virtual ErrorCode LastError() const OVERRIDE;
virtual int32_t RegisterEventObserver(
- webrtc::AudioDeviceObserver* event_callback);
+ webrtc::AudioDeviceObserver* event_callback) OVERRIDE;
// Note: Calling this method from a callback may result in deadlock.
- virtual int32_t RegisterAudioCallback(webrtc::AudioTransport* audio_callback);
+ virtual int32_t RegisterAudioCallback(
+ webrtc::AudioTransport* audio_callback) OVERRIDE;
- virtual int32_t Init();
- virtual int32_t Terminate();
- virtual bool Initialized() const;
+ virtual int32_t Init() OVERRIDE;
+ virtual int32_t Terminate() OVERRIDE;
+ virtual bool Initialized() const OVERRIDE;
- virtual int16_t PlayoutDevices();
- virtual int16_t RecordingDevices();
- virtual int32_t PlayoutDeviceName(uint16_t index,
- char name[webrtc::kAdmMaxDeviceNameSize],
- char guid[webrtc::kAdmMaxGuidSize]);
- virtual int32_t RecordingDeviceName(uint16_t index,
- char name[webrtc::kAdmMaxDeviceNameSize],
- char guid[webrtc::kAdmMaxGuidSize]);
+ virtual int16_t PlayoutDevices() OVERRIDE;
+ virtual int16_t RecordingDevices() OVERRIDE;
+ virtual int32_t PlayoutDeviceName(
+ uint16_t index,
+ char name[webrtc::kAdmMaxDeviceNameSize],
+ char guid[webrtc::kAdmMaxGuidSize]) OVERRIDE;
+ virtual int32_t RecordingDeviceName(
+ uint16_t index,
+ char name[webrtc::kAdmMaxDeviceNameSize],
+ char guid[webrtc::kAdmMaxGuidSize]) OVERRIDE;
- virtual int32_t SetPlayoutDevice(uint16_t index);
- virtual int32_t SetPlayoutDevice(WindowsDeviceType device);
- virtual int32_t SetRecordingDevice(uint16_t index);
- virtual int32_t SetRecordingDevice(WindowsDeviceType device);
+ virtual int32_t SetPlayoutDevice(uint16_t index) OVERRIDE;
+ virtual int32_t SetPlayoutDevice(WindowsDeviceType device) OVERRIDE;
+ virtual int32_t SetRecordingDevice(uint16_t index) OVERRIDE;
+ virtual int32_t SetRecordingDevice(WindowsDeviceType device) OVERRIDE;
- virtual int32_t PlayoutIsAvailable(bool* available);
- virtual int32_t InitPlayout();
- virtual bool PlayoutIsInitialized() const;
- virtual int32_t RecordingIsAvailable(bool* available);
- virtual int32_t InitRecording();
- virtual bool RecordingIsInitialized() const;
+ virtual int32_t PlayoutIsAvailable(bool* available) OVERRIDE;
+ virtual int32_t InitPlayout() OVERRIDE;
+ virtual bool PlayoutIsInitialized() const OVERRIDE;
+ virtual int32_t RecordingIsAvailable(bool* available) OVERRIDE;
+ virtual int32_t InitRecording() OVERRIDE;
+ virtual bool RecordingIsInitialized() const OVERRIDE;
- virtual int32_t StartPlayout();
- virtual int32_t StopPlayout();
- virtual bool Playing() const;
- virtual int32_t StartRecording();
- virtual int32_t StopRecording();
- virtual bool Recording() const;
+ virtual int32_t StartPlayout() OVERRIDE;
+ virtual int32_t StopPlayout() OVERRIDE;
+ virtual bool Playing() const OVERRIDE;
+ virtual int32_t StartRecording() OVERRIDE;
+ virtual int32_t StopRecording() OVERRIDE;
+ virtual bool Recording() const OVERRIDE;
- virtual int32_t SetAGC(bool enable);
- virtual bool AGC() const;
+ virtual int32_t SetAGC(bool enable) OVERRIDE;
+ virtual bool AGC() const OVERRIDE;
virtual int32_t SetWaveOutVolume(uint16_t volume_left,
- uint16_t volume_right);
+ uint16_t volume_right) OVERRIDE;
virtual int32_t WaveOutVolume(uint16_t* volume_left,
- uint16_t* volume_right) const;
+ uint16_t* volume_right) const OVERRIDE;
- virtual int32_t SpeakerIsAvailable(bool* available);
- virtual int32_t InitSpeaker();
- virtual bool SpeakerIsInitialized() const;
- virtual int32_t MicrophoneIsAvailable(bool* available);
- virtual int32_t InitMicrophone();
- virtual bool MicrophoneIsInitialized() const;
+ virtual int32_t InitSpeaker() OVERRIDE;
+ virtual bool SpeakerIsInitialized() const OVERRIDE;
+ virtual int32_t InitMicrophone() OVERRIDE;
+ virtual bool MicrophoneIsInitialized() const OVERRIDE;
- virtual int32_t SpeakerVolumeIsAvailable(bool* available);
- virtual int32_t SetSpeakerVolume(uint32_t volume);
- virtual int32_t SpeakerVolume(uint32_t* volume) const;
- virtual int32_t MaxSpeakerVolume(uint32_t* max_volume) const;
- virtual int32_t MinSpeakerVolume(uint32_t* min_volume) const;
- virtual int32_t SpeakerVolumeStepSize(uint16_t* step_size) const;
+ virtual int32_t SpeakerVolumeIsAvailable(bool* available) OVERRIDE;
+ virtual int32_t SetSpeakerVolume(uint32_t volume) OVERRIDE;
+ virtual int32_t SpeakerVolume(uint32_t* volume) const OVERRIDE;
+ virtual int32_t MaxSpeakerVolume(uint32_t* max_volume) const OVERRIDE;
+ virtual int32_t MinSpeakerVolume(uint32_t* min_volume) const OVERRIDE;
+ virtual int32_t SpeakerVolumeStepSize(uint16_t* step_size) const OVERRIDE;
- virtual int32_t MicrophoneVolumeIsAvailable(bool* available);
- virtual int32_t SetMicrophoneVolume(uint32_t volume);
- virtual int32_t MicrophoneVolume(uint32_t* volume) const;
- virtual int32_t MaxMicrophoneVolume(uint32_t* max_volume) const;
+ virtual int32_t MicrophoneVolumeIsAvailable(bool* available) OVERRIDE;
+ virtual int32_t SetMicrophoneVolume(uint32_t volume) OVERRIDE;
+ virtual int32_t MicrophoneVolume(uint32_t* volume) const OVERRIDE;
+ virtual int32_t MaxMicrophoneVolume(uint32_t* max_volume) const OVERRIDE;
- virtual int32_t MinMicrophoneVolume(uint32_t* min_volume) const;
- virtual int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const;
+ virtual int32_t MinMicrophoneVolume(uint32_t* min_volume) const OVERRIDE;
+ virtual int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const OVERRIDE;
- virtual int32_t SpeakerMuteIsAvailable(bool* available);
- virtual int32_t SetSpeakerMute(bool enable);
- virtual int32_t SpeakerMute(bool* enabled) const;
+ virtual int32_t SpeakerMuteIsAvailable(bool* available) OVERRIDE;
+ virtual int32_t SetSpeakerMute(bool enable) OVERRIDE;
+ virtual int32_t SpeakerMute(bool* enabled) const OVERRIDE;
- virtual int32_t MicrophoneMuteIsAvailable(bool* available);
- virtual int32_t SetMicrophoneMute(bool enable);
- virtual int32_t MicrophoneMute(bool* enabled) const;
+ virtual int32_t MicrophoneMuteIsAvailable(bool* available) OVERRIDE;
+ virtual int32_t SetMicrophoneMute(bool enable) OVERRIDE;
+ virtual int32_t MicrophoneMute(bool* enabled) const OVERRIDE;
- virtual int32_t MicrophoneBoostIsAvailable(bool* available);
- virtual int32_t SetMicrophoneBoost(bool enable);
- virtual int32_t MicrophoneBoost(bool* enabled) const;
+ virtual int32_t MicrophoneBoostIsAvailable(bool* available) OVERRIDE;
+ virtual int32_t SetMicrophoneBoost(bool enable) OVERRIDE;
+ virtual int32_t MicrophoneBoost(bool* enabled) const OVERRIDE;
- virtual int32_t StereoPlayoutIsAvailable(bool* available) const;
- virtual int32_t SetStereoPlayout(bool enable);
- virtual int32_t StereoPlayout(bool* enabled) const;
- virtual int32_t StereoRecordingIsAvailable(bool* available) const;
- virtual int32_t SetStereoRecording(bool enable);
- virtual int32_t StereoRecording(bool* enabled) const;
- virtual int32_t SetRecordingChannel(const ChannelType channel);
- virtual int32_t RecordingChannel(ChannelType* channel) const;
+ virtual int32_t StereoPlayoutIsAvailable(bool* available) const OVERRIDE;
+ virtual int32_t SetStereoPlayout(bool enable) OVERRIDE;
+ virtual int32_t StereoPlayout(bool* enabled) const OVERRIDE;
+ virtual int32_t StereoRecordingIsAvailable(bool* available) const OVERRIDE;
+ virtual int32_t SetStereoRecording(bool enable) OVERRIDE;
+ virtual int32_t StereoRecording(bool* enabled) const OVERRIDE;
+ virtual int32_t SetRecordingChannel(const ChannelType channel) OVERRIDE;
+ virtual int32_t RecordingChannel(ChannelType* channel) const OVERRIDE;
virtual int32_t SetPlayoutBuffer(const BufferType type,
- uint16_t size_ms = 0);
+ uint16_t size_ms = 0) OVERRIDE;
virtual int32_t PlayoutBuffer(BufferType* type,
- uint16_t* size_ms) const;
- virtual int32_t PlayoutDelay(uint16_t* delay_ms) const;
- virtual int32_t RecordingDelay(uint16_t* delay_ms) const;
+ uint16_t* size_ms) const OVERRIDE;
+ virtual int32_t PlayoutDelay(uint16_t* delay_ms) const OVERRIDE;
+ virtual int32_t RecordingDelay(uint16_t* delay_ms) const OVERRIDE;
- virtual int32_t CPULoad(uint16_t* load) const;
+ virtual int32_t CPULoad(uint16_t* load) const OVERRIDE;
virtual int32_t StartRawOutputFileRecording(
- const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]);
- virtual int32_t StopRawOutputFileRecording();
+ const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) OVERRIDE;
+ virtual int32_t StopRawOutputFileRecording() OVERRIDE;
virtual int32_t StartRawInputFileRecording(
- const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]);
- virtual int32_t StopRawInputFileRecording();
+ const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) OVERRIDE;
+ virtual int32_t StopRawInputFileRecording() OVERRIDE;
- virtual int32_t SetRecordingSampleRate(const uint32_t samples_per_sec);
- virtual int32_t RecordingSampleRate(uint32_t* samples_per_sec) const;
- virtual int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec);
- virtual int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const;
+ virtual int32_t SetRecordingSampleRate(
+ const uint32_t samples_per_sec) OVERRIDE;
+ virtual int32_t RecordingSampleRate(uint32_t* samples_per_sec) const OVERRIDE;
+ virtual int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) OVERRIDE;
+ virtual int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const OVERRIDE;
- virtual int32_t ResetAudioDevice();
- virtual int32_t SetLoudspeakerStatus(bool enable);
- virtual int32_t GetLoudspeakerStatus(bool* enabled) const;
+ virtual int32_t ResetAudioDevice() OVERRIDE;
+ virtual int32_t SetLoudspeakerStatus(bool enable) OVERRIDE;
+ virtual int32_t GetLoudspeakerStatus(bool* enabled) const OVERRIDE;
// End of functions inherited from webrtc::AudioDeviceModule.
// The following function is inherited from rtc::MessageHandler.
- virtual void OnMessage(rtc::Message* msg);
+ virtual void OnMessage(rtc::Message* msg) OVERRIDE;
protected:
// The constructor is protected because the class needs to be created as a
diff --git a/app/webrtc/test/fakeaudiocapturemodule_unittest.cc b/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
index 9e63c1c..ddacc38 100644
--- a/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
+++ b/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
@@ -145,11 +145,6 @@
TEST_F(FakeAdmTest, PlayoutTest) {
EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
- bool speaker_available = false;
- EXPECT_EQ(0, fake_audio_capture_module_->SpeakerIsAvailable(
- &speaker_available));
- EXPECT_TRUE(speaker_available);
-
bool stereo_available = false;
EXPECT_EQ(0,
fake_audio_capture_module_->StereoPlayoutIsAvailable(
@@ -182,11 +177,6 @@
TEST_F(FakeAdmTest, RecordTest) {
EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
- bool microphone_available = false;
- EXPECT_EQ(0, fake_audio_capture_module_->MicrophoneIsAvailable(
- µphone_available));
- EXPECT_TRUE(microphone_available);
-
bool stereo_available = false;
EXPECT_EQ(0, fake_audio_capture_module_->StereoRecordingIsAvailable(
&stereo_available));
diff --git a/app/webrtc/test/mockpeerconnectionobservers.h b/app/webrtc/test/mockpeerconnectionobservers.h
index 174b80b..0570d40 100644
--- a/app/webrtc/test/mockpeerconnectionobservers.h
+++ b/app/webrtc/test/mockpeerconnectionobservers.h
@@ -133,31 +133,37 @@
size_t number_of_reports() const { return reports_.size(); }
int AudioOutputLevel() {
- return GetSsrcStatsValue(
- webrtc::StatsReport::kStatsValueNameAudioOutputLevel);
+ return GetStatsValue(StatsReport::kStatsReportTypeSsrc,
+ StatsReport::kStatsValueNameAudioOutputLevel);
}
int AudioInputLevel() {
- return GetSsrcStatsValue(
- webrtc::StatsReport::kStatsValueNameAudioInputLevel);
+ return GetStatsValue(StatsReport::kStatsReportTypeSsrc,
+ StatsReport::kStatsValueNameAudioInputLevel);
}
int BytesReceived() {
- return GetSsrcStatsValue(
- webrtc::StatsReport::kStatsValueNameBytesReceived);
+ return GetStatsValue(StatsReport::kStatsReportTypeSsrc,
+ StatsReport::kStatsValueNameBytesReceived);
}
int BytesSent() {
- return GetSsrcStatsValue(webrtc::StatsReport::kStatsValueNameBytesSent);
+ return GetStatsValue(StatsReport::kStatsReportTypeSsrc,
+ StatsReport::kStatsValueNameBytesSent);
+ }
+
+ int AvailableReceiveBandwidth() {
+ return GetStatsValue(StatsReport::kStatsReportTypeBwe,
+ StatsReport::kStatsValueNameAvailableReceiveBandwidth);
}
private:
- int GetSsrcStatsValue(StatsReport::StatsValueName name) {
+ int GetStatsValue(const std::string& type, StatsReport::StatsValueName name) {
if (reports_.empty()) {
return 0;
}
for (size_t i = 0; i < reports_.size(); ++i) {
- if (reports_[i].type != StatsReport::kStatsReportTypeSsrc)
+ if (reports_[i].type != type)
continue;
webrtc::StatsReport::Values::const_iterator it =
reports_[i].values.begin();
diff --git a/app/webrtc/test/peerconnectiontestwrapper.cc b/app/webrtc/test/peerconnectiontestwrapper.cc
index 8a4f45c..24932b8 100644
--- a/app/webrtc/test/peerconnectiontestwrapper.cc
+++ b/app/webrtc/test/peerconnectiontestwrapper.cc
@@ -75,9 +75,8 @@
return false;
}
- audio_thread_.Start();
fake_audio_capture_module_ = FakeAudioCaptureModule::Create(
- &audio_thread_);
+ rtc::Thread::Current());
if (fake_audio_capture_module_ == NULL) {
return false;
}
diff --git a/app/webrtc/test/peerconnectiontestwrapper.h b/app/webrtc/test/peerconnectiontestwrapper.h
index f3477ce..d4a0e4e 100644
--- a/app/webrtc/test/peerconnectiontestwrapper.h
+++ b/app/webrtc/test/peerconnectiontestwrapper.h
@@ -111,7 +111,6 @@
bool video, const webrtc::FakeConstraints& video_constraints);
std::string name_;
- rtc::Thread audio_thread_;
rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
allocator_factory_;
rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
diff --git a/app/webrtc/videosource.cc b/app/webrtc/videosource.cc
index 8770e6d..589341d 100644
--- a/app/webrtc/videosource.cc
+++ b/app/webrtc/videosource.cc
@@ -28,6 +28,7 @@
#include "talk/app/webrtc/videosource.h"
#include <vector>
+#include <cstdlib>
#include "talk/app/webrtc/mediaconstraintsinterface.h"
#include "talk/session/media/channelmanager.h"
@@ -254,11 +255,15 @@
std::vector<cricket::VideoFormat>::const_iterator it = formats.begin();
std::vector<cricket::VideoFormat>::const_iterator best_it = formats.begin();
- int best_diff = abs(default_area - it->width* it->height);
+ int best_diff_area = std::abs(default_area - it->width * it->height);
+ int64 best_diff_interval = kDefaultFormat.interval;
for (; it != formats.end(); ++it) {
- int diff = abs(default_area - it->width* it->height);
- if (diff < best_diff) {
- best_diff = diff;
+ int diff_area = std::abs(default_area - it->width * it->height);
+ int64 diff_interval = std::abs(kDefaultFormat.interval - it->interval);
+ if (diff_area < best_diff_area ||
+ (diff_area == best_diff_area && diff_interval < best_diff_interval)) {
+ best_diff_area = diff_area;
+ best_diff_interval = diff_interval;
best_it = it;
}
}
diff --git a/app/webrtc/webrtcsdp.cc b/app/webrtc/webrtcsdp.cc
index 792a091..5887409 100644
--- a/app/webrtc/webrtcsdp.cc
+++ b/app/webrtc/webrtcsdp.cc
@@ -71,6 +71,7 @@
using cricket::kCodecParamSctpProtocol;
using cricket::kCodecParamSctpStreams;
using cricket::kCodecParamMaxAverageBitrate;
+using cricket::kCodecParamMaxPlaybackRate;
using cricket::kCodecParamAssociatedPayloadType;
using cricket::kWildcardPayloadType;
using cricket::MediaContentDescription;
@@ -155,6 +156,9 @@
static const char kAttributeRtcpFb[] = "rtcp-fb";
static const char kAttributeSendRecv[] = "sendrecv";
static const char kAttributeInactive[] = "inactive";
+// draft-ietf-mmusic-sctp-sdp-07
+// a=sctp-port
+static const char kAttributeSctpPort[] = "sctp-port";
// Experimental flags
static const char kAttributeXGoogleFlag[] = "x-google-flag";
@@ -1100,6 +1104,26 @@
return true;
}
+bool ParseSctpPort(const std::string& line,
+ int* sctp_port,
+ SdpParseError* error) {
+ // draft-ietf-mmusic-sctp-sdp-07
+ // a=sctp-port
+ std::vector<std::string> fields;
+ rtc::split(line.substr(kLinePrefixLength),
+ kSdpDelimiterSpace, &fields);
+ const size_t expected_min_fields = 2;
+ if (fields.size() < expected_min_fields) {
+ return ParseFailedExpectMinFieldNum(line, expected_min_fields, error);
+ }
+ if (!rtc::FromString(fields[1], sctp_port)) {
+ return ParseFailed(line,
+ "Invalid sctp port value.",
+ error);
+ }
+ return true;
+}
+
bool ParseExtmap(const std::string& line, RtpHeaderExtension* extmap,
SdpParseError* error) {
// RFC 5285
@@ -1511,7 +1535,8 @@
kCodecParamStereo, kCodecParamUseInbandFec, kCodecParamStartBitrate,
kCodecParamMaxBitrate, kCodecParamMinBitrate, kCodecParamMaxQuantization,
kCodecParamSctpProtocol, kCodecParamSctpStreams,
- kCodecParamMaxAverageBitrate, kCodecParamAssociatedPayloadType
+ kCodecParamMaxAverageBitrate, kCodecParamMaxPlaybackRate,
+ kCodecParamAssociatedPayloadType
};
for (size_t i = 0; i < ARRAY_SIZE(kFmtpParams); ++i) {
if (_stricmp(name.c_str(), kFmtpParams[i]) == 0) {
@@ -1563,6 +1588,24 @@
}
}
+bool AddSctpDataCodec(DataContentDescription* media_desc,
+ int sctp_port) {
+ if (media_desc->HasCodec(cricket::kGoogleSctpDataCodecId)) {
+ return ParseFailed("",
+ "Can't have multiple sctp port attributes.",
+ NULL);
+ }
+ // Add the SCTP Port number as a pseudo-codec "port" parameter
+ cricket::DataCodec codec_port(
+ cricket::kGoogleSctpDataCodecId, cricket::kGoogleSctpDataCodecName,
+ 0);
+ codec_port.SetParam(cricket::kCodecParamPort, sctp_port);
+ LOG(INFO) << "AddSctpDataCodec: Got SCTP Port Number "
+ << sctp_port;
+ media_desc->AddCodec(codec_port);
+ return true;
+}
+
bool GetMinValue(const std::vector<int>& values, int* value) {
if (values.empty()) {
return false;
@@ -2129,18 +2172,20 @@
// <fmt>
std::vector<int> codec_preference;
- for (size_t j = 3 ; j < fields.size(); ++j) {
- // TODO(wu): Remove when below bug is fixed.
- // https://bugzilla.mozilla.org/show_bug.cgi?id=996329
- if (fields[j] == "" && j == fields.size() - 1) {
- continue;
- }
+ if (!is_sctp) {
+ for (size_t j = 3 ; j < fields.size(); ++j) {
+ // TODO(wu): Remove when below bug is fixed.
+ // https://bugzilla.mozilla.org/show_bug.cgi?id=996329
+ if (fields[j] == "" && j == fields.size() - 1) {
+ continue;
+ }
- int pl = 0;
- if (!GetValueFromString(line, fields[j], &pl, error)) {
- return false;
+ int pl = 0;
+ if (!GetValueFromString(line, fields[j], &pl, error)) {
+ return false;
+ }
+ codec_preference.push_back(pl);
}
- codec_preference.push_back(pl);
}
// Make a temporary TransportDescription based on |session_td|.
@@ -2167,26 +2212,20 @@
codec_preference, pos, &content_name,
&transport, candidates, error));
} else if (HasAttribute(line, kMediaTypeData)) {
- DataContentDescription* desc =
+ DataContentDescription* data_desc =
ParseContentDescription<DataContentDescription>(
message, cricket::MEDIA_TYPE_DATA, mline_index, protocol,
codec_preference, pos, &content_name,
&transport, candidates, error);
+ content.reset(data_desc);
- if (desc && protocol == cricket::kMediaProtocolDtlsSctp) {
- // Add the SCTP Port number as a pseudo-codec "port" parameter
- cricket::DataCodec codec_port(
- cricket::kGoogleSctpDataCodecId, cricket::kGoogleSctpDataCodecName,
- 0);
- codec_port.SetParam(cricket::kCodecParamPort, fields[3]);
- LOG(INFO) << "ParseMediaDescription: Got SCTP Port Number "
- << fields[3];
- ASSERT(!desc->HasCodec(cricket::kGoogleSctpDataCodecId));
- desc->AddCodec(codec_port);
+ int p;
+ if (data_desc && protocol == cricket::kMediaProtocolDtlsSctp &&
+ rtc::FromString(fields[3], &p)) {
+ if (!AddSctpDataCodec(data_desc, p))
+ return false;
}
- content.reset(desc);
-
// We should always use the default bandwidth for RTP-based data
// channels. Don't allow SDP to set the bandwidth, because that
// would give JS the opportunity to "break the Internet".
@@ -2518,6 +2557,15 @@
if (!ParseDtlsSetup(line, &(transport->connection_role), error)) {
return false;
}
+ } else if (HasAttribute(line, kAttributeSctpPort)) {
+ int sctp_port;
+ if (!ParseSctpPort(line, &sctp_port, error)) {
+ return false;
+ }
+ if (!AddSctpDataCodec(static_cast<DataContentDescription*>(media_desc),
+ sctp_port)) {
+ return false;
+ }
} else if (is_rtp) {
//
// RTP specific attrubtes
diff --git a/app/webrtc/webrtcsdp_unittest.cc b/app/webrtc/webrtcsdp_unittest.cc
index 6a22e38..560d5da 100644
--- a/app/webrtc/webrtcsdp_unittest.cc
+++ b/app/webrtc/webrtcsdp_unittest.cc
@@ -284,6 +284,16 @@
"a=mid:data_content_name\r\n"
"a=sctpmap:5000 webrtc-datachannel 1024\r\n";
+// draft-ietf-mmusic-sctp-sdp-07
+static const char kSdpSctpDataChannelStringWithSctpPort[] =
+ "m=application 1 DTLS/SCTP webrtc-datachannel\r\n"
+ "a=fmtp:webrtc-datachannel max-message-size=100000\r\n"
+ "a=sctp-port 5000\r\n"
+ "c=IN IP4 0.0.0.0\r\n"
+ "a=ice-ufrag:ufrag_data\r\n"
+ "a=ice-pwd:pwd_data\r\n"
+ "a=mid:data_content_name\r\n";
+
static const char kSdpSctpDataChannelWithCandidatesString[] =
"m=application 2345 DTLS/SCTP 5000\r\n"
"c=IN IP4 74.125.127.126\r\n"
@@ -2023,6 +2033,36 @@
EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
}
+TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannelsWithSctpPort) {
+ AddSctpDataChannel();
+ JsepSessionDescription jdesc(kDummyString);
+ ASSERT_TRUE(jdesc.Initialize(desc_.Copy(), kSessionId, kSessionVersion));
+
+ std::string sdp_with_data = kSdpString;
+ sdp_with_data.append(kSdpSctpDataChannelStringWithSctpPort);
+ JsepSessionDescription jdesc_output(kDummyString);
+
+ EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output));
+ EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
+}
+
+// Test to check the behaviour if sctp-port is specified
+// on the m= line and in a=sctp-port.
+TEST_F(WebRtcSdpTest, DeserializeSdpWithMultiSctpPort) {
+ AddSctpDataChannel();
+ JsepSessionDescription jdesc(kDummyString);
+ ASSERT_TRUE(jdesc.Initialize(desc_.Copy(), kSessionId, kSessionVersion));
+
+ std::string sdp_with_data = kSdpString;
+ // Append m= attributes
+ sdp_with_data.append(kSdpSctpDataChannelString);
+ // Append a=sctp-port attribute
+ sdp_with_data.append("a=sctp-port 5000\r\n");
+ JsepSessionDescription jdesc_output(kDummyString);
+
+ EXPECT_FALSE(SdpDeserialize(sdp_with_data, &jdesc_output));
+}
+
// For crbug/344475.
TEST_F(WebRtcSdpTest, DeserializeSdpWithCorruptedSctpDataChannels) {
std::string sdp_with_data = kSdpString;
@@ -2071,6 +2111,19 @@
EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output));
EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
+
+ // We need to test the deserialized JsepSessionDescription from
+ // kSdpSctpDataChannelStringWithSctpPort for
+ // draft-ietf-mmusic-sctp-sdp-07
+ // a=sctp-port
+ sdp_with_data = kSdpString;
+ sdp_with_data.append(kSdpSctpDataChannelStringWithSctpPort);
+ rtc::replace_substrs(default_portstr, strlen(default_portstr),
+ unusual_portstr, strlen(unusual_portstr),
+ &sdp_with_data);
+
+ EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output));
+ EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
}
TEST_F(WebRtcSdpTest, DeserializeSdpWithRtpDataChannelsAndBandwidth) {
diff --git a/app/webrtc/webrtcsession.cc b/app/webrtc/webrtcsession.cc
index 6d57afb..13e2128 100644
--- a/app/webrtc/webrtcsession.cc
+++ b/app/webrtc/webrtcsession.cc
@@ -224,30 +224,28 @@
cricket::StreamParams stream_out;
const cricket::ContentInfo* audio_info =
cricket::GetFirstAudioContent(session_description);
- if (!audio_info) {
- return false;
- }
- const cricket::MediaContentDescription* audio_content =
- static_cast<const cricket::MediaContentDescription*>(
- audio_info->description);
+ if (audio_info) {
+ const cricket::MediaContentDescription* audio_content =
+ static_cast<const cricket::MediaContentDescription*>(
+ audio_info->description);
- if (cricket::GetStreamBySsrc(audio_content->streams(), ssrc, &stream_out)) {
- *track_id = stream_out.id;
- return true;
+ if (cricket::GetStreamBySsrc(audio_content->streams(), ssrc, &stream_out)) {
+ *track_id = stream_out.id;
+ return true;
+ }
}
const cricket::ContentInfo* video_info =
cricket::GetFirstVideoContent(session_description);
- if (!video_info) {
- return false;
- }
- const cricket::MediaContentDescription* video_content =
- static_cast<const cricket::MediaContentDescription*>(
- video_info->description);
+ if (video_info) {
+ const cricket::MediaContentDescription* video_content =
+ static_cast<const cricket::MediaContentDescription*>(
+ video_info->description);
- if (cricket::GetStreamBySsrc(video_content->streams(), ssrc, &stream_out)) {
- *track_id = stream_out.id;
- return true;
+ if (cricket::GetStreamBySsrc(video_content->streams(), ssrc, &stream_out)) {
+ *track_id = stream_out.id;
+ return true;
+ }
}
return false;
}
@@ -389,6 +387,22 @@
}
}
+uint32 ConvertIceTransportTypeToCandidateFilter(
+ PeerConnectionInterface::IceTransportsType type) {
+ switch (type) {
+ case PeerConnectionInterface::kNone:
+ return cricket::CF_NONE;
+ case PeerConnectionInterface::kRelay:
+ return cricket::CF_RELAY;
+ case PeerConnectionInterface::kNoHost:
+ return (cricket::CF_ALL & ~cricket::CF_HOST);
+ case PeerConnectionInterface::kAll:
+ return cricket::CF_ALL;
+ default: ASSERT(false);
+ }
+ return cricket::CF_NONE;
+}
+
// Help class used to remember if a a remote peer has requested ice restart by
// by sending a description with new ice ufrag and password.
class IceRestartAnswerLatch {
@@ -471,14 +485,16 @@
}
WebRtcSession::~WebRtcSession() {
- if (voice_channel_.get()) {
- SignalVoiceChannelDestroyed();
- channel_manager_->DestroyVoiceChannel(voice_channel_.release());
- }
+ // Destroy video_channel_ first since it may have a pointer to the
+ // voice_channel_.
if (video_channel_.get()) {
SignalVideoChannelDestroyed();
channel_manager_->DestroyVideoChannel(video_channel_.release());
}
+ if (voice_channel_.get()) {
+ SignalVoiceChannelDestroyed();
+ channel_manager_->DestroyVoiceChannel(voice_channel_.release());
+ }
if (data_channel_.get()) {
SignalDataChannelDestroyed();
channel_manager_->DestroyDataChannel(data_channel_.release());
@@ -610,6 +626,10 @@
cricket::VideoOptions::HIGH);
}
+ SetOptionFromOptionalConstraint(constraints,
+ MediaConstraintsInterface::kCombinedAudioVideoBwe,
+ &audio_options_.combined_audio_video_bwe);
+
const cricket::VideoCodec default_codec(
JsepSessionDescription::kDefaultVideoCodecId,
JsepSessionDescription::kDefaultVideoCodecName,
@@ -636,7 +656,8 @@
if (options.disable_encryption) {
webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
}
-
+ port_allocator()->set_candidate_filter(
+ ConvertIceTransportTypeToCandidateFilter(ice_transport));
return true;
}
@@ -742,6 +763,7 @@
if (!UpdateSessionState(action, cricket::CS_LOCAL, err_desc)) {
return false;
}
+
// Kick starting the ice candidates allocation.
StartCandidatesAllocation();
@@ -903,8 +925,10 @@
return UseCandidate(candidate);
}
-bool WebRtcSession::UpdateIce(PeerConnectionInterface::IceTransportsType type) {
- return false;
+bool WebRtcSession::SetIceTransports(
+ PeerConnectionInterface::IceTransportsType type) {
+ return port_allocator()->set_candidate_filter(
+ ConvertIceTransportTypeToCandidateFilter(type));
}
bool WebRtcSession::GetLocalTrackIdBySsrc(uint32 ssrc, std::string* track_id) {
@@ -1425,16 +1449,8 @@
void WebRtcSession::RemoveUnusedChannelsAndTransports(
const SessionDescription* desc) {
- const cricket::ContentInfo* voice_info =
- cricket::GetFirstAudioContent(desc);
- if ((!voice_info || voice_info->rejected) && voice_channel_) {
- mediastream_signaling_->OnAudioChannelClose();
- SignalVoiceChannelDestroyed();
- const std::string content_name = voice_channel_->content_name();
- channel_manager_->DestroyVoiceChannel(voice_channel_.release());
- DestroyTransportProxy(content_name);
- }
-
+ // Destroy video_channel_ first since it may have a pointer to the
+ // voice_channel_.
const cricket::ContentInfo* video_info =
cricket::GetFirstVideoContent(desc);
if ((!video_info || video_info->rejected) && video_channel_) {
@@ -1445,6 +1461,16 @@
DestroyTransportProxy(content_name);
}
+ const cricket::ContentInfo* voice_info =
+ cricket::GetFirstAudioContent(desc);
+ if ((!voice_info || voice_info->rejected) && voice_channel_) {
+ mediastream_signaling_->OnAudioChannelClose();
+ SignalVoiceChannelDestroyed();
+ const std::string content_name = voice_channel_->content_name();
+ channel_manager_->DestroyVoiceChannel(voice_channel_.release());
+ DestroyTransportProxy(content_name);
+ }
+
const cricket::ContentInfo* data_info =
cricket::GetFirstDataContent(desc);
if ((!data_info || data_info->rejected) && data_channel_) {
diff --git a/app/webrtc/webrtcsession.h b/app/webrtc/webrtcsession.h
index d1e5645..86ae435 100644
--- a/app/webrtc/webrtcsession.h
+++ b/app/webrtc/webrtcsession.h
@@ -160,7 +160,7 @@
std::string* err_desc);
bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
- bool UpdateIce(PeerConnectionInterface::IceTransportsType type);
+ bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
const SessionDescriptionInterface* local_description() const {
return local_desc_.get();
diff --git a/app/webrtc/webrtcsession_unittest.cc b/app/webrtc/webrtcsession_unittest.cc
index 206f320..7aa87fb 100644
--- a/app/webrtc/webrtcsession_unittest.cc
+++ b/app/webrtc/webrtcsession_unittest.cc
@@ -108,6 +108,8 @@
static const char kStunAddrHost[] = "99.99.99.1";
static const SocketAddress kTurnUdpIntAddr("99.99.99.4", 3478);
static const SocketAddress kTurnUdpExtAddr("99.99.99.6", 0);
+static const char kTurnUsername[] = "test";
+static const char kTurnPassword[] = "test";
static const char kSessionVersion[] = "1";
@@ -510,7 +512,7 @@
}
void VerifyAnswerFromNonCryptoOffer() {
- // Create a SDP without Crypto.
+ // Create an SDP without Crypto.
cricket::MediaSessionOptions options;
options.has_video = true;
JsepSessionDescription* offer(
@@ -1083,6 +1085,18 @@
}
}
+ void ConfigureAllocatorWithTurn() {
+ cricket::RelayServerConfig relay_server(cricket::RELAY_TURN);
+ cricket::RelayCredentials credentials(kTurnUsername, kTurnPassword);
+ relay_server.credentials = credentials;
+ relay_server.ports.push_back(cricket::ProtocolAddress(
+ kTurnUdpIntAddr, cricket::PROTO_UDP, false));
+ allocator_->AddRelay(relay_server);
+ allocator_->set_step_delay(cricket::kMinimumStepDelay);
+ allocator_->set_flags(cricket::PORTALLOCATOR_DISABLE_TCP |
+ cricket::PORTALLOCATOR_ENABLE_BUNDLE);
+ }
+
cricket::FakeMediaEngine* media_engine_;
cricket::FakeDataEngine* data_engine_;
cricket::FakeDeviceManager* device_manager_;
@@ -1162,6 +1176,53 @@
EXPECT_EQ(6u, observer_.mline_1_candidates_.size());
}
+// Test session delivers no candidates gathered when constraint set to "none".
+TEST_F(WebRtcSessionTest, TestIceTransportsNone) {
+ AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
+ SetIceTransportType(PeerConnectionInterface::kNone);
+ Init(NULL);
+ mediastream_signaling_.SendAudioVideoStream1();
+ InitiateCall();
+ EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
+ EXPECT_EQ(0u, observer_.mline_0_candidates_.size());
+ EXPECT_EQ(0u, observer_.mline_1_candidates_.size());
+}
+
+// Test session delivers only relay candidates gathered when constaint set to
+// "relay".
+TEST_F(WebRtcSessionTest, TestIceTransportsRelay) {
+ AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
+ ConfigureAllocatorWithTurn();
+ SetIceTransportType(PeerConnectionInterface::kRelay);
+ Init(NULL);
+ mediastream_signaling_.SendAudioVideoStream1();
+ InitiateCall();
+ EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
+ EXPECT_EQ(2u, observer_.mline_0_candidates_.size());
+ EXPECT_EQ(2u, observer_.mline_1_candidates_.size());
+ for (size_t i = 0; i < observer_.mline_0_candidates_.size(); ++i) {
+ EXPECT_EQ(cricket::RELAY_PORT_TYPE,
+ observer_.mline_0_candidates_[i].type());
+ }
+ for (size_t i = 0; i < observer_.mline_1_candidates_.size(); ++i) {
+ EXPECT_EQ(cricket::RELAY_PORT_TYPE,
+ observer_.mline_1_candidates_[i].type());
+ }
+}
+
+// Test session delivers all candidates gathered when constaint set to "all".
+TEST_F(WebRtcSessionTest, TestIceTransportsAll) {
+ AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
+ SetIceTransportType(PeerConnectionInterface::kAll);
+ Init(NULL);
+ mediastream_signaling_.SendAudioVideoStream1();
+ InitiateCall();
+ EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
+ // Host + STUN. By default allocator is disabled to gather relay candidates.
+ EXPECT_EQ(4u, observer_.mline_0_candidates_.size());
+ EXPECT_EQ(4u, observer_.mline_1_candidates_.size());
+}
+
TEST_F(WebRtcSessionTest, SetSdpFailedOnInvalidSdp) {
Init(NULL);
SessionDescriptionInterface* offer = NULL;
@@ -1211,14 +1272,13 @@
rtc::FromString<uint64>(offer->session_version()));
SetLocalDescriptionWithoutError(offer);
+ EXPECT_EQ(0u, video_channel_->send_streams().size());
+ EXPECT_EQ(0u, voice_channel_->send_streams().size());
mediastream_signaling_.SendAudioVideoStream2();
answer = CreateRemoteAnswer(session_->local_description());
SetRemoteDescriptionWithoutError(answer);
- EXPECT_EQ(0u, video_channel_->send_streams().size());
- EXPECT_EQ(0u, voice_channel_->send_streams().size());
-
// Make sure the receive streams have not changed.
ASSERT_EQ(1u, video_channel_->recv_streams().size());
EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[0].id);
@@ -1992,13 +2052,21 @@
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(offer->description());
-
EXPECT_TRUE(content != NULL);
+
content = cricket::GetFirstVideoContent(offer->description());
EXPECT_TRUE(content != NULL);
- // TODO(perkj): Should the direction be set to SEND_ONLY if
- // The constraints is set to not receive audio or video but a track is added?
+ // Sets constraints to false and verifies that audio/video contents are
+ // removed.
+ options.offer_to_receive_audio = 0;
+ options.offer_to_receive_video = 0;
+ offer.reset(CreateOffer(options));
+
+ content = cricket::GetFirstAudioContent(offer->description());
+ EXPECT_TRUE(content == NULL);
+ content = cricket::GetFirstVideoContent(offer->description());
+ EXPECT_TRUE(content == NULL);
}
// Test that an answer can not be created if the last remote description is not
@@ -2037,8 +2105,7 @@
Init(NULL);
// Create a remote offer with audio only.
cricket::MediaSessionOptions options;
- options.has_audio = true;
- options.has_video = false;
+
rtc::scoped_ptr<JsepSessionDescription> offer(
CreateRemoteOffer(options));
ASSERT_TRUE(cricket::GetFirstVideoContent(offer->description()) == NULL);
@@ -2174,7 +2241,6 @@
SessionDescriptionInterface* offer = CreateOffer();
cricket::MediaSessionOptions options;
- options.has_video = false;
SessionDescriptionInterface* answer = CreateRemoteAnswer(offer, options);
// SetLocalDescription and SetRemoteDescriptions takes ownership of offer
@@ -2887,7 +2953,6 @@
TEST_F(WebRtcSessionTest, TestCreateAnswerWithNewUfragAndPassword) {
Init(NULL);
cricket::MediaSessionOptions options;
- options.has_audio = true;
options.has_video = true;
rtc::scoped_ptr<JsepSessionDescription> offer(
CreateRemoteOffer(options));
@@ -2919,7 +2984,6 @@
TEST_F(WebRtcSessionTest, TestCreateAnswerWithOldUfragAndPassword) {
Init(NULL);
cricket::MediaSessionOptions options;
- options.has_audio = true;
options.has_video = true;
rtc::scoped_ptr<JsepSessionDescription> offer(
CreateRemoteOffer(options));
@@ -2985,7 +3049,6 @@
TEST_F(WebRtcSessionTest, SetSdpFailedOnSessionError) {
Init(NULL);
cricket::MediaSessionOptions options;
- options.has_audio = true;
options.has_video = true;
cricket::BaseSession::Error error_code = cricket::BaseSession::ERROR_CONTENT;
@@ -3311,6 +3374,26 @@
SetAndVerifyNumUnsignalledRecvStreams(-1, 0);
}
+TEST_F(WebRtcSessionTest, TestCombinedAudioVideoBweConstraint) {
+ constraints_.reset(new FakeConstraints());
+ constraints_->AddOptional(
+ webrtc::MediaConstraintsInterface::kCombinedAudioVideoBwe,
+ true);
+ Init(NULL);
+ mediastream_signaling_.SendAudioVideoStream1();
+ SessionDescriptionInterface* offer = CreateOffer();
+
+ SetLocalDescriptionWithoutError(offer);
+
+ voice_channel_ = media_engine_->GetVoiceChannel(0);
+
+ ASSERT_TRUE(voice_channel_ != NULL);
+ cricket::AudioOptions audio_options;
+ EXPECT_TRUE(voice_channel_->GetOptions(&audio_options));
+ EXPECT_TRUE(
+ audio_options.combined_audio_video_bwe.GetWithDefaultIfUnset(false));
+}
+
// Tests that we can renegotiate new media content with ICE candidates in the
// new remote SDP.
TEST_F(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesInSdp) {
diff --git a/build/common.gypi b/build/common.gypi
index 57b21af..7ee4224 100644
--- a/build/common.gypi
+++ b/build/common.gypi
@@ -91,6 +91,7 @@
# LateBindingSymbolTable::TableInfo from
# latebindingsymboltable.cc.def and remove below flag.
'-Wno-address-of-array-temporary',
+ '-Wthread-safety',
],
}],
],
diff --git a/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java b/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
index 213da7b..468ce22 100644
--- a/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
+++ b/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
@@ -77,7 +77,6 @@
public class AppRTCDemoActivity extends Activity
implements AppRTCClient.IceServersObserver {
private static final String TAG = "AppRTCDemoActivity";
- private static boolean factoryStaticInitialized;
private PeerConnectionFactory factory;
private VideoSource videoSource;
private boolean videoSourceStopped;
@@ -133,13 +132,6 @@
hudView.setVisibility(View.INVISIBLE);
addContentView(hudView, hudLayout);
- if (!factoryStaticInitialized) {
- abortUnless(PeerConnectionFactory.initializeAndroidGlobals(
- this, true, true),
- "Failed to initializeAndroidGlobals");
- factoryStaticInitialized = true;
- }
-
AudioManager audioManager =
((AudioManager) getSystemService(AUDIO_SERVICE));
// TODO(fischman): figure out how to do this Right(tm) and remove the
@@ -282,6 +274,9 @@
@Override
public void onIceServers(List<PeerConnection.IceServer> iceServers) {
+ abortUnless(PeerConnectionFactory.initializeAndroidGlobals(
+ this, true, true, VideoRendererGui.getEGLContext()),
+ "Failed to initializeAndroidGlobals");
factory = new PeerConnectionFactory();
MediaConstraints pcConstraints = appRtcClient.pcConstraints();
diff --git a/examples/call/callclient.cc b/examples/call/callclient.cc
index 2c8a6bc..b31e24a 100644
--- a/examples/call/callclient.cc
+++ b/examples/call/callclient.cc
@@ -889,10 +889,8 @@
AddSession(call_->InitiateSession(jid, media_client_->jid(), options));
}
media_client_->SetFocus(call_);
- if (call_->has_video() && render_) {
- if (!options.is_muc) {
- call_->SetLocalRenderer(local_renderer_);
- }
+ if (call_->has_video() && render_ && !options.is_muc) {
+ // TODO(pthatcher): Hookup local_render_ to the local capturer.
}
if (options.is_muc) {
const std::string& nick = mucs_[jid]->local_jid().resource();
@@ -1086,7 +1084,7 @@
call_->AcceptSession(session, options);
media_client_->SetFocus(call_);
if (call_->has_video() && render_) {
- call_->SetLocalRenderer(local_renderer_);
+ // TODO(pthatcher): Hookup local_render_ to the local capturer.
RenderAllStreams(call_, session, true);
}
SetupAcceptedCall();
diff --git a/examples/objc/AppRTCDemo/ios/APPRTCViewController.m b/examples/objc/AppRTCDemo/ios/APPRTCViewController.m
index a4a0bd3..8042762 100644
--- a/examples/objc/AppRTCDemo/ios/APPRTCViewController.m
+++ b/examples/objc/AppRTCDemo/ios/APPRTCViewController.m
@@ -63,6 +63,18 @@
- (void)viewDidLoad {
[super viewDidLoad];
+
+ self.remoteVideoView =
+ [[RTCEAGLVideoView alloc] initWithFrame:self.blackView.bounds];
+ self.remoteVideoView.delegate = self;
+ self.remoteVideoView.transform = CGAffineTransformMakeScale(-1, 1);
+ [self.blackView addSubview:self.remoteVideoView];
+
+ self.localVideoView =
+ [[RTCEAGLVideoView alloc] initWithFrame:self.blackView.bounds];
+ self.localVideoView.delegate = self;
+ [self.blackView addSubview:self.localVideoView];
+
self.statusBarOrientation =
[UIApplication sharedApplication].statusBarOrientation;
self.roomInput.delegate = self;
@@ -181,25 +193,13 @@
self.instructionsView.hidden = NO;
self.logView.hidden = YES;
self.logView.text = nil;
+ self.localVideoView.videoTrack = nil;
+ self.remoteVideoView.videoTrack = nil;
self.blackView.hidden = YES;
- [self.remoteVideoView removeFromSuperview];
- self.remoteVideoView = nil;
- [self.localVideoView removeFromSuperview];
- self.localVideoView = nil;
}
- (void)setupCaptureSession {
self.blackView.hidden = NO;
- self.remoteVideoView =
- [[RTCEAGLVideoView alloc] initWithFrame:self.blackView.bounds];
- self.remoteVideoView.delegate = self;
- self.remoteVideoView.transform = CGAffineTransformMakeScale(-1, 1);
- [self.blackView addSubview:self.remoteVideoView];
-
- self.localVideoView =
- [[RTCEAGLVideoView alloc] initWithFrame:self.blackView.bounds];
- self.localVideoView.delegate = self;
- [self.blackView addSubview:self.localVideoView];
[self updateVideoViewLayout];
}
diff --git a/examples/peerconnection/server/utils.h b/examples/peerconnection/server/utils.h
index d05a2c3..5320d5a 100644
--- a/examples/peerconnection/server/utils.h
+++ b/examples/peerconnection/server/utils.h
@@ -29,18 +29,7 @@
#define TALK_EXAMPLES_PEERCONNECTION_SERVER_UTILS_H_
#pragma once
-#ifndef assert
-#ifndef WIN32
#include <assert.h>
-#else
-#ifndef NDEBUG
-#define assert(expr) ((void)((expr) ? true : __debugbreak()))
-#else
-#define assert(expr) ((void)0)
-#endif // NDEBUG
-#endif // WIN32
-#endif // assert
-
#include <string>
#ifndef ARRAYSIZE
diff --git a/libjingle.gyp b/libjingle.gyp
index f4f9bf9..335a788 100755
--- a/libjingle.gyp
+++ b/libjingle.gyp
@@ -303,27 +303,13 @@
'dependencies': [
'<(DEPTH)/third_party/expat/expat.gyp:expat',
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
- '<(webrtc_root)/base/base.gyp:webrtc_base',
+ '<(webrtc_root)/base/base.gyp:rtc_base',
],
'export_dependent_settings': [
'<(DEPTH)/third_party/expat/expat.gyp:expat',
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
],
'sources': [
- 'xmllite/qname.cc',
- 'xmllite/qname.h',
- 'xmllite/xmlbuilder.cc',
- 'xmllite/xmlbuilder.h',
- 'xmllite/xmlconstants.cc',
- 'xmllite/xmlconstants.h',
- 'xmllite/xmlelement.cc',
- 'xmllite/xmlelement.h',
- 'xmllite/xmlnsstack.cc',
- 'xmllite/xmlnsstack.h',
- 'xmllite/xmlparser.cc',
- 'xmllite/xmlparser.h',
- 'xmllite/xmlprinter.cc',
- 'xmllite/xmlprinter.h',
'xmpp/asyncsocket.h',
'xmpp/chatroommodule.h',
'xmpp/chatroommoduleimpl.cc',
@@ -400,48 +386,6 @@
],
}, # target libjingle
{
- 'target_name': 'libjingle_sound',
- 'type': 'static_library',
- 'dependencies': [
- 'libjingle',
- ],
- 'sources': [
- 'sound/automaticallychosensoundsystem.h',
- 'sound/nullsoundsystem.cc',
- 'sound/nullsoundsystem.h',
- 'sound/nullsoundsystemfactory.cc',
- 'sound/nullsoundsystemfactory.h',
- 'sound/platformsoundsystem.cc',
- 'sound/platformsoundsystem.h',
- 'sound/platformsoundsystemfactory.cc',
- 'sound/platformsoundsystemfactory.h',
- 'sound/sounddevicelocator.h',
- 'sound/soundinputstreaminterface.h',
- 'sound/soundoutputstreaminterface.h',
- 'sound/soundsystemfactory.h',
- 'sound/soundsysteminterface.cc',
- 'sound/soundsysteminterface.h',
- 'sound/soundsystemproxy.cc',
- 'sound/soundsystemproxy.h',
- ],
- 'conditions': [
- ['OS=="linux"', {
- 'sources': [
- 'sound/alsasoundsystem.cc',
- 'sound/alsasoundsystem.h',
- 'sound/alsasymboltable.cc',
- 'sound/alsasymboltable.h',
- 'sound/linuxsoundsystem.cc',
- 'sound/linuxsoundsystem.h',
- 'sound/pulseaudiosoundsystem.cc',
- 'sound/pulseaudiosoundsystem.h',
- 'sound/pulseaudiosymboltable.cc',
- 'sound/pulseaudiosymboltable.h',
- ],
- }],
- ],
- }, # target libjingle_sound
- {
'target_name': 'libjingle_media',
'type': 'static_library',
'include_dirs': [
@@ -452,14 +396,14 @@
'dependencies': [
'<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',
'<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib',
- '<(webrtc_root)/modules/modules.gyp:video_capture_module',
'<(webrtc_root)/modules/modules.gyp:video_render_module',
'<(webrtc_root)/webrtc.gyp:webrtc',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
+ '<(webrtc_root)/sound/sound.gyp:rtc_sound',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:field_trial_default',
+ '<(webrtc_root)/libjingle/xmllite/xmllite.gyp:rtc_xmllite',
'libjingle',
- 'libjingle_sound',
],
'direct_dependent_settings': {
'include_dirs': [
@@ -485,8 +429,6 @@
'media/base/filemediaengine.cc',
'media/base/filemediaengine.h',
'media/base/hybriddataengine.h',
- 'media/base/hybridvideoengine.cc',
- 'media/base/hybridvideoengine.h',
'media/base/mediachannel.h',
'media/base/mediacommon.h',
'media/base/mediaengine.cc',
@@ -556,6 +498,17 @@
'media/webrtc/webrtcvoiceengine.h',
],
'conditions': [
+ ['build_with_chromium==1', {
+ 'dependencies': [
+ '<(webrtc_root)/modules/modules.gyp:video_capture_module_impl',
+ '<(webrtc_root)/modules/modules.gyp:video_render_module_impl',
+ ],
+ }, {
+ 'dependencies': [
+ '<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
+ '<(webrtc_root)/modules/modules.gyp:video_render_module_internal_impl',
+ ],
+ }],
['OS=="linux"', {
'sources': [
'media/devices/gtkvideorenderer.cc',
@@ -631,6 +584,7 @@
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
+ '-weak_framework AVFoundation',
'-framework Cocoa',
'-framework CoreAudio',
'-framework CoreVideo',
diff --git a/libjingle_tests.gyp b/libjingle_tests.gyp
index 76f22ea..44adec3 100755
--- a/libjingle_tests.gyp
+++ b/libjingle_tests.gyp
@@ -33,7 +33,7 @@
'type': 'static_library',
'dependencies': [
'<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',
- '<(webrtc_root)/base/base_tests.gyp:webrtc_base_tests_utils',
+ '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<@(libjingle_tests_additional_deps)',
],
'direct_dependent_settings': {
@@ -74,18 +74,13 @@
'type': 'executable',
'includes': [ 'build/ios_tests.gypi', ],
'dependencies': [
- '<(webrtc_root)/base/base.gyp:webrtc_base',
- '<(webrtc_root)/base/base_tests.gyp:webrtc_base_tests_utils',
+ '<(webrtc_root)/base/base.gyp:rtc_base',
+ '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
+ '<(webrtc_root)/libjingle/xmllite/xmllite.gyp:rtc_xmllite',
'libjingle.gyp:libjingle',
'libjingle_unittest_main',
],
'sources': [
- 'xmllite/qname_unittest.cc',
- 'xmllite/xmlbuilder_unittest.cc',
- 'xmllite/xmlelement_unittest.cc',
- 'xmllite/xmlnsstack_unittest.cc',
- 'xmllite/xmlparser_unittest.cc',
- 'xmllite/xmlprinter_unittest.cc',
'xmpp/fakexmppclient.h',
'xmpp/hangoutpubsubclient_unittest.cc',
'xmpp/jid_unittest.cc',
@@ -104,22 +99,10 @@
], # sources
}, # target libjingle_unittest
{
- 'target_name': 'libjingle_sound_unittest',
- 'type': 'executable',
- 'dependencies': [
- '<(webrtc_root)/base/base_tests.gyp:webrtc_base_tests_utils',
- 'libjingle.gyp:libjingle_sound',
- 'libjingle_unittest_main',
- ],
- 'sources': [
- 'sound/automaticallychosensoundsystem_unittest.cc',
- ],
- }, # target libjingle_sound_unittest
- {
'target_name': 'libjingle_media_unittest',
'type': 'executable',
'dependencies': [
- '<(webrtc_root)/base/base_tests.gyp:webrtc_base_tests_utils',
+ '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'libjingle.gyp:libjingle_media',
'libjingle_unittest_main',
],
@@ -157,9 +140,8 @@
'media/sctp/sctpdataengine_unittest.cc',
'media/webrtc/webrtcpassthroughrender_unittest.cc',
'media/webrtc/webrtcvideocapturer_unittest.cc',
- # Omitted because depends on non-open-source testdata files.
- # 'media/base/videoframe_unittest.h',
- # 'media/webrtc/webrtcvideoframe_unittest.cc',
+ 'media/base/videoframe_unittest.h',
+ 'media/webrtc/webrtcvideoframe_unittest.cc',
# Disabled because some tests fail.
# TODO(ronghuawu): Reenable these tests.
@@ -171,6 +153,11 @@
],
'conditions': [
['OS=="win"', {
+ 'dependencies': [
+ '<(DEPTH)/net/third_party/nss/ssl.gyp:libssl',
+ '<(DEPTH)/third_party/nss/nss.gyp:nspr',
+ '<(DEPTH)/third_party/nss/nss.gyp:nss',
+ ],
'msvs_settings': {
'VCLinkerTool': {
'AdditionalDependencies': [
@@ -194,7 +181,7 @@
'type': 'executable',
'dependencies': [
'<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
- '<(webrtc_root)/base/base_tests.gyp:webrtc_base_tests_utils',
+ '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'libjingle.gyp:libjingle',
'libjingle.gyp:libjingle_p2p',
'libjingle_unittest_main',
@@ -253,7 +240,7 @@
'type': 'executable',
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
- '<(webrtc_root)/base/base_tests.gyp:webrtc_base_tests_utils',
+ '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'libjingle.gyp:libjingle',
'libjingle.gyp:libjingle_p2p',
'libjingle.gyp:libjingle_peerconnection',
@@ -386,7 +373,7 @@
'type': 'executable',
'includes': [ 'build/ios_tests.gypi', ],
'dependencies': [
- '<(webrtc_root)/base/base_tests.gyp:webrtc_base_tests_utils',
+ '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'libjingle.gyp:libjingle_peerconnection_objc',
],
'sources': [
@@ -486,20 +473,6 @@
],
},
{
- 'target_name': 'libjingle_sound_unittest_run',
- 'type': 'none',
- 'dependencies': [
- 'libjingle_sound_unittest',
- ],
- 'includes': [
- 'build/isolate.gypi',
- 'libjingle_sound_unittest.isolate',
- ],
- 'sources': [
- 'libjingle_sound_unittest.isolate',
- ],
- },
- {
'target_name': 'libjingle_unittest_run',
'type': 'none',
'dependencies': [
diff --git a/media/base/constants.cc b/media/base/constants.cc
index cd10ef7..19a960f 100644
--- a/media/base/constants.cc
+++ b/media/base/constants.cc
@@ -59,6 +59,7 @@
const char kCodecParamStereo[] = "stereo";
const char kCodecParamUseInbandFec[] = "useinbandfec";
const char kCodecParamMaxAverageBitrate[] = "maxaveragebitrate";
+const char kCodecParamMaxPlaybackRate[] = "maxplaybackrate";
const char kCodecParamSctpProtocol[] = "protocol";
const char kCodecParamSctpStreams[] = "streams";
@@ -72,6 +73,7 @@
const int kOpusDefaultSPropStereo = 0;
const int kOpusDefaultStereo = 0;
const int kOpusDefaultUseInbandFec = 0;
+const int kOpusDefaultMaxPlaybackRate = 48000;
const int kPreferredMaxPTime = 60;
const int kPreferredMinPTime = 10;
diff --git a/media/base/constants.h b/media/base/constants.h
index 5ac1be2..5168acb 100644
--- a/media/base/constants.h
+++ b/media/base/constants.h
@@ -62,6 +62,7 @@
extern const char kCodecParamStereo[];
extern const char kCodecParamUseInbandFec[];
extern const char kCodecParamMaxAverageBitrate[];
+extern const char kCodecParamMaxPlaybackRate[];
extern const char kCodecParamSctpProtocol[];
extern const char kCodecParamSctpStreams[];
@@ -79,6 +80,8 @@
extern const int kOpusDefaultSPropStereo;
extern const int kOpusDefaultStereo;
extern const int kOpusDefaultUseInbandFec;
+extern const int kOpusDefaultMaxPlaybackRate;
+
// Prefered values in this code base. Note that they may differ from the default
// values in http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03
// Only frames larger or equal to 10 ms are currently supported in this code
diff --git a/media/base/executablehelpers.h b/media/base/executablehelpers.h
new file mode 100644
index 0000000..2dde010
--- /dev/null
+++ b/media/base/executablehelpers.h
@@ -0,0 +1,100 @@
+/*
+ * libjingle
+ * Copyright 2014 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef TALK_MEDIA_BASE_EXECUTABLEHELPERS_H_
+#define TALK_MEDIA_BASE_EXECUTABLEHELPERS_H_
+
+#ifdef OSX
+#include <mach-o/dyld.h>
+#endif
+
+#include <string>
+
+#include "webrtc/base/logging.h"
+#include "webrtc/base/pathutils.h"
+
+namespace rtc {
+
+// Returns the path to the running executable or an empty path.
+// TODO(thorcarpenter): Consolidate with FluteClient::get_executable_dir.
+inline Pathname GetExecutablePath() {
+ const int32 kMaxExePathSize = 255;
+#ifdef WIN32
+ TCHAR exe_path_buffer[kMaxExePathSize];
+ DWORD copied_length = GetModuleFileName(NULL, // NULL = Current process
+ exe_path_buffer, kMaxExePathSize);
+ if (0 == copied_length) {
+ LOG(LS_ERROR) << "Copied length is zero";
+ return rtc::Pathname();
+ }
+ if (kMaxExePathSize == copied_length) {
+ LOG(LS_ERROR) << "Buffer too small";
+ return rtc::Pathname();
+ }
+#ifdef UNICODE
+ std::wstring wdir(exe_path_buffer);
+ std::string dir_tmp(wdir.begin(), wdir.end());
+ rtc::Pathname path(dir_tmp);
+#else // UNICODE
+ rtc::Pathname path(exe_path_buffer);
+#endif // UNICODE
+#elif defined(OSX) || defined(LINUX)
+ char exe_path_buffer[kMaxExePathSize];
+#ifdef OSX
+ uint32_t copied_length = kMaxExePathSize - 1;
+ if (_NSGetExecutablePath(exe_path_buffer, &copied_length) == -1) {
+ LOG(LS_ERROR) << "Buffer too small";
+ return rtc::Pathname();
+ }
+#elif defined LINUX
+ int32 copied_length = kMaxExePathSize - 1;
+ const char* kProcExeFmt = "/proc/%d/exe";
+ char proc_exe_link[40];
+ snprintf(proc_exe_link, sizeof(proc_exe_link), kProcExeFmt, getpid());
+ copied_length = readlink(proc_exe_link, exe_path_buffer, copied_length);
+ if (copied_length == -1) {
+ LOG_ERR(LS_ERROR) << "Error reading link " << proc_exe_link;
+ return rtc::Pathname();
+ }
+ if (copied_length == kMaxExePathSize - 1) {
+ LOG(LS_ERROR) << "Probably truncated result when reading link "
+ << proc_exe_link;
+ return rtc::Pathname();
+ }
+ exe_path_buffer[copied_length] = '\0';
+#endif // LINUX
+ rtc::Pathname path(exe_path_buffer);
+#else // Android || IOS
+ rtc::Pathname path;
+#endif // OSX || LINUX
+ return path;
+}
+
+} // namespace rtc
+
+#endif // TALK_MEDIA_BASE_EXECUTABLEHELPERS_H_
+
diff --git a/media/base/fakemediaengine.h b/media/base/fakemediaengine.h
index 7bc3958..a6eabef 100644
--- a/media/base/fakemediaengine.h
+++ b/media/base/fakemediaengine.h
@@ -868,7 +868,7 @@
class FakeVideoEngine : public FakeBaseEngine {
public:
- FakeVideoEngine() : renderer_(NULL), capture_(false), processor_(NULL) {
+ FakeVideoEngine() : capture_(false), processor_(NULL) {
// Add a fake video codec. Note that the name must not be "" as there are
// sanity checks against that.
codecs_.push_back(VideoCodec(0, "fake_video_codec", 0, 0, 0, 0));
@@ -926,10 +926,6 @@
options_changed_ = true;
return true;
}
- bool SetLocalRenderer(VideoRenderer* r) {
- renderer_ = r;
- return true;
- }
bool SetCapture(bool capture) {
capture_ = capture;
return true;
@@ -946,7 +942,6 @@
std::vector<VideoCodec> codecs_;
VideoEncoderConfig default_encoder_config_;
std::string in_device_;
- VideoRenderer* renderer_;
bool capture_;
VideoProcessor* processor_;
VideoOptions options_;
@@ -994,7 +989,6 @@
}
const std::string& audio_in_device() const { return voice_.in_device_; }
const std::string& audio_out_device() const { return voice_.out_device_; }
- VideoRenderer* local_renderer() { return video_.renderer_; }
int voice_loglevel() const { return voice_.loglevel_; }
const std::string& voice_logfilter() const { return voice_.logfilter_; }
int video_loglevel() const { return video_.loglevel_; }
diff --git a/media/base/filemediaengine.h b/media/base/filemediaengine.h
index e546328..d3e99a8 100644
--- a/media/base/filemediaengine.h
+++ b/media/base/filemediaengine.h
@@ -88,7 +88,6 @@
virtual SoundclipMedia* CreateSoundclip() { return NULL; }
virtual AudioOptions GetAudioOptions() const { return AudioOptions(); }
virtual bool SetAudioOptions(const AudioOptions& options) { return true; }
- virtual bool SetVideoOptions(const VideoOptions& options) { return true; }
virtual bool SetAudioDelayOffset(int offset) { return true; }
virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config) {
return true;
@@ -113,7 +112,6 @@
virtual bool SetOutputVolume(int level) { return true; }
virtual int GetInputLevel() { return 0; }
virtual bool SetLocalMonitor(bool enable) { return true; }
- virtual bool SetLocalRenderer(VideoRenderer* renderer) { return true; }
// TODO(whyuan): control channel send?
virtual bool SetVideoCapture(bool capture) { return true; }
virtual const std::vector<AudioCodec>& audio_codecs() {
diff --git a/media/base/filemediaengine_unittest.cc b/media/base/filemediaengine_unittest.cc
index 1f7405d..c542baf 100644
--- a/media/base/filemediaengine_unittest.cc
+++ b/media/base/filemediaengine_unittest.cc
@@ -223,8 +223,6 @@
EXPECT_TRUE(NULL == engine_->CreateSoundclip());
cricket::AudioOptions audio_options;
EXPECT_TRUE(engine_->SetAudioOptions(audio_options));
- cricket::VideoOptions video_options;
- EXPECT_TRUE(engine_->SetVideoOptions(video_options));
VideoEncoderConfig video_encoder_config;
EXPECT_TRUE(engine_->SetDefaultVideoEncoderConfig(video_encoder_config));
EXPECT_TRUE(engine_->SetSoundDevices(NULL, NULL));
@@ -232,7 +230,6 @@
EXPECT_TRUE(engine_->SetOutputVolume(0));
EXPECT_EQ(0, engine_->GetInputLevel());
EXPECT_TRUE(engine_->SetLocalMonitor(true));
- EXPECT_TRUE(engine_->SetLocalRenderer(NULL));
EXPECT_TRUE(engine_->SetVideoCapture(true));
EXPECT_EQ(0U, engine_->audio_codecs().size());
EXPECT_EQ(0U, engine_->video_codecs().size());
diff --git a/media/base/hybridvideoengine.cc b/media/base/hybridvideoengine.cc
deleted file mode 100644
index 289c4fe..0000000
--- a/media/base/hybridvideoengine.cc
+++ /dev/null
@@ -1,356 +0,0 @@
-/*
- * libjingle
- * Copyright 2004 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/media/base/hybridvideoengine.h"
-
-#include "webrtc/base/logging.h"
-
-namespace cricket {
-
-HybridVideoMediaChannel::HybridVideoMediaChannel(
- HybridVideoEngineInterface* engine,
- VideoMediaChannel* channel1,
- VideoMediaChannel* channel2)
- : engine_(engine),
- channel1_(channel1),
- channel2_(channel2),
- active_channel_(NULL),
- sending_(false) {
-}
-
-HybridVideoMediaChannel::~HybridVideoMediaChannel() {
-}
-
-void HybridVideoMediaChannel::SetInterface(NetworkInterface* iface) {
- if (channel1_) {
- channel1_->SetInterface(iface);
- }
- if (channel2_) {
- channel2_->SetInterface(iface);
- }
-}
-
-bool HybridVideoMediaChannel::SetOptions(const VideoOptions &options) {
- bool ret = true;
- if (channel1_) {
- ret = channel1_->SetOptions(options);
- }
- if (channel2_ && ret) {
- ret = channel2_->SetOptions(options);
- }
- return ret;
-}
-
-bool HybridVideoMediaChannel::GetOptions(VideoOptions *options) const {
- if (active_channel_) {
- return active_channel_->GetOptions(options);
- }
- if (channel1_) {
- return channel1_->GetOptions(options);
- }
- if (channel2_) {
- return channel2_->GetOptions(options);
- }
- return false;
-}
-
-bool HybridVideoMediaChannel::SetRecvCodecs(
- const std::vector<VideoCodec>& codecs) {
- // Only give each channel the codecs it knows about.
- bool ret = true;
- std::vector<VideoCodec> codecs1, codecs2;
- SplitCodecs(codecs, &codecs1, &codecs2);
- if (channel1_) {
- ret = channel1_->SetRecvCodecs(codecs1);
- }
- if (channel2_ && ret) {
- ret = channel2_->SetRecvCodecs(codecs2);
- }
- return ret;
-}
-
-bool HybridVideoMediaChannel::SetRecvRtpHeaderExtensions(
- const std::vector<RtpHeaderExtension>& extensions) {
- bool ret = true;
- if (channel1_) {
- ret = channel1_->SetRecvRtpHeaderExtensions(extensions);
- }
- if (channel2_ && ret) {
- ret = channel2_->SetRecvRtpHeaderExtensions(extensions);
- }
- return ret;
-}
-
-bool HybridVideoMediaChannel::SetRenderer(uint32 ssrc,
- VideoRenderer* renderer) {
- bool ret = true;
- if (channel1_) {
- ret = channel1_->SetRenderer(ssrc, renderer);
- }
- if (channel2_ && ret) {
- ret = channel2_->SetRenderer(ssrc, renderer);
- }
- return ret;
-}
-
-bool HybridVideoMediaChannel::SetRender(bool render) {
- bool ret = true;
- if (channel1_) {
- ret = channel1_->SetRender(render);
- }
- if (channel2_ && ret) {
- ret = channel2_->SetRender(render);
- }
- return ret;
-}
-
-bool HybridVideoMediaChannel::MuteStream(uint32 ssrc, bool muted) {
- bool ret = true;
- if (channel1_) {
- ret = channel1_->MuteStream(ssrc, muted);
- }
- if (channel2_ && ret) {
- ret = channel2_->MuteStream(ssrc, muted);
- }
- return ret;
-}
-
-bool HybridVideoMediaChannel::SetSendCodecs(
- const std::vector<VideoCodec>& codecs) {
- // Use the input to this function to decide what impl we're going to use.
- if (!active_channel_ && !SelectActiveChannel(codecs)) {
- LOG(LS_WARNING) << "Failed to select active channel";
- return false;
- }
- // Only give the active channel the codecs it knows about.
- std::vector<VideoCodec> codecs1, codecs2;
- SplitCodecs(codecs, &codecs1, &codecs2);
- const std::vector<VideoCodec>& codecs_to_set =
- (active_channel_ == channel1_.get()) ? codecs1 : codecs2;
- bool return_value = active_channel_->SetSendCodecs(codecs_to_set);
- if (!return_value) {
- return false;
- }
- VideoCodec send_codec;
- return_value = active_channel_->GetSendCodec(&send_codec);
- if (!return_value) {
- return false;
- }
- engine_->OnNewSendResolution(send_codec.width, send_codec.height);
- active_channel_->UpdateAspectRatio(send_codec.width, send_codec.height);
- return true;
-}
-
-bool HybridVideoMediaChannel::GetSendCodec(VideoCodec* send_codec) {
- if (!active_channel_) {
- return false;
- }
- return active_channel_->GetSendCodec(send_codec);
-}
-
-bool HybridVideoMediaChannel::SetSendStreamFormat(uint32 ssrc,
- const VideoFormat& format) {
- return active_channel_ && active_channel_->SetSendStreamFormat(ssrc, format);
-}
-
-bool HybridVideoMediaChannel::SetSendRtpHeaderExtensions(
- const std::vector<RtpHeaderExtension>& extensions) {
- return active_channel_ &&
- active_channel_->SetSendRtpHeaderExtensions(extensions);
-}
-
-bool HybridVideoMediaChannel::SetStartSendBandwidth(int bps) {
- return active_channel_ && active_channel_->SetStartSendBandwidth(bps);
-}
-
-bool HybridVideoMediaChannel::SetMaxSendBandwidth(int bps) {
- return active_channel_ && active_channel_->SetMaxSendBandwidth(bps);
-}
-
-bool HybridVideoMediaChannel::SetSend(bool send) {
- if (send == sending()) {
- return true; // no action required if already set.
- }
-
- bool ret = active_channel_ &&
- active_channel_->SetSend(send);
-
- // Returns error and don't connect the signal if starting up.
- // Disconnects the signal anyway if shutting down.
- if (ret || !send) {
- // TODO(juberti): Remove this hack that connects the WebRTC channel
- // to the capturer.
- if (active_channel_ == channel1_.get()) {
- engine_->OnSendChange1(channel1_.get(), send);
- } else {
- engine_->OnSendChange2(channel2_.get(), send);
- }
- // If succeeded, remember the state as is.
- // If failed to open, sending_ should be false.
- // If failed to stop, sending_ should also be false, as we disconnect the
- // capture anyway.
- // The failure on SetSend(false) is a known issue in webrtc.
- sending_ = send;
- }
- return ret;
-}
-
-bool HybridVideoMediaChannel::SetCapturer(uint32 ssrc,
- VideoCapturer* capturer) {
- bool ret = true;
- if (channel1_.get()) {
- ret = channel1_->SetCapturer(ssrc, capturer);
- }
- if (channel2_.get() && ret) {
- ret = channel2_->SetCapturer(ssrc, capturer);
- }
- return ret;
-}
-
-bool HybridVideoMediaChannel::AddSendStream(const StreamParams& sp) {
- bool ret = true;
- if (channel1_) {
- ret = channel1_->AddSendStream(sp);
- }
- if (channel2_ && ret) {
- ret = channel2_->AddSendStream(sp);
- }
- return ret;
-}
-
-bool HybridVideoMediaChannel::RemoveSendStream(uint32 ssrc) {
- bool ret = true;
- if (channel1_) {
- ret = channel1_->RemoveSendStream(ssrc);
- }
- if (channel2_ && ret) {
- ret = channel2_->RemoveSendStream(ssrc);
- }
- return ret;
-}
-
-bool HybridVideoMediaChannel::AddRecvStream(const StreamParams& sp) {
- return active_channel_ &&
- active_channel_->AddRecvStream(sp);
-}
-
-bool HybridVideoMediaChannel::RemoveRecvStream(uint32 ssrc) {
- return active_channel_ &&
- active_channel_->RemoveRecvStream(ssrc);
-}
-
-bool HybridVideoMediaChannel::SendIntraFrame() {
- return active_channel_ &&
- active_channel_->SendIntraFrame();
-}
-
-bool HybridVideoMediaChannel::RequestIntraFrame() {
- return active_channel_ &&
- active_channel_->RequestIntraFrame();
-}
-
-bool HybridVideoMediaChannel::GetStats(
- const StatsOptions& options, VideoMediaInfo* info) {
- // TODO(juberti): Ensure that returning no stats until SetSendCodecs is OK.
- return active_channel_ &&
- active_channel_->GetStats(options, info);
-}
-
-void HybridVideoMediaChannel::OnPacketReceived(
- rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
- // Eat packets until we have an active channel;
- if (active_channel_) {
- active_channel_->OnPacketReceived(packet, packet_time);
- } else {
- LOG(LS_INFO) << "HybridVideoChannel: Eating early RTP packet";
- }
-}
-
-void HybridVideoMediaChannel::OnRtcpReceived(
- rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
- // Eat packets until we have an active channel;
- if (active_channel_) {
- active_channel_->OnRtcpReceived(packet, packet_time);
- } else {
- LOG(LS_INFO) << "HybridVideoChannel: Eating early RTCP packet";
- }
-}
-
-void HybridVideoMediaChannel::OnReadyToSend(bool ready) {
- if (channel1_) {
- channel1_->OnReadyToSend(ready);
- }
- if (channel2_) {
- channel2_->OnReadyToSend(ready);
- }
-}
-
-void HybridVideoMediaChannel::UpdateAspectRatio(int ratio_w, int ratio_h) {
- if (active_channel_) active_channel_->UpdateAspectRatio(ratio_w, ratio_h);
-}
-
-bool HybridVideoMediaChannel::SelectActiveChannel(
- const std::vector<VideoCodec>& codecs) {
- if (!active_channel_ && !codecs.empty()) {
- if (engine_->HasCodec1(codecs[0])) {
- channel2_.reset();
- active_channel_ = channel1_.get();
- } else if (engine_->HasCodec2(codecs[0])) {
- channel1_.reset();
- active_channel_ = channel2_.get();
- }
- }
- if (NULL == active_channel_) {
- return false;
- }
- // Connect signals from the active channel.
- active_channel_->SignalMediaError.connect(
- this,
- &HybridVideoMediaChannel::OnMediaError);
- return true;
-}
-
-void HybridVideoMediaChannel::SplitCodecs(
- const std::vector<VideoCodec>& codecs,
- std::vector<VideoCodec>* codecs1, std::vector<VideoCodec>* codecs2) {
- codecs1->clear();
- codecs2->clear();
- for (size_t i = 0; i < codecs.size(); ++i) {
- if (engine_->HasCodec1(codecs[i])) {
- codecs1->push_back(codecs[i]);
- }
- if (engine_->HasCodec2(codecs[i])) {
- codecs2->push_back(codecs[i]);
- }
- }
-}
-
-void HybridVideoMediaChannel::OnMediaError(uint32 ssrc, Error error) {
- SignalMediaError(ssrc, error);
-}
-
-} // namespace cricket
diff --git a/media/base/hybridvideoengine.h b/media/base/hybridvideoengine.h
deleted file mode 100644
index 004d3cf..0000000
--- a/media/base/hybridvideoengine.h
+++ /dev/null
@@ -1,286 +0,0 @@
-/*
- * libjingle
- * Copyright 2004 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_MEDIA_BASE_HYBRIDVIDEOENGINE_H_
-#define TALK_MEDIA_BASE_HYBRIDVIDEOENGINE_H_
-
-#include <string>
-#include <vector>
-
-#include "talk/media/base/codec.h"
-#include "talk/media/base/mediachannel.h"
-#include "talk/media/base/videocapturer.h"
-#include "talk/media/base/videocommon.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/sigslotrepeater.h"
-
-namespace cricket {
-
-struct Device;
-struct VideoFormat;
-class HybridVideoEngineInterface;
-class VideoCapturer;
-class VideoFrame;
-class VideoRenderer;
-
-// HybridVideoMediaChannels work with a HybridVideoEngine to combine
-// two unrelated VideoMediaChannel implementations into a single class.
-class HybridVideoMediaChannel : public VideoMediaChannel {
- public:
- HybridVideoMediaChannel(HybridVideoEngineInterface* engine,
- VideoMediaChannel* channel1,
- VideoMediaChannel* channel2);
- virtual ~HybridVideoMediaChannel();
-
- // VideoMediaChannel methods
- virtual void SetInterface(NetworkInterface* iface);
- virtual bool SetOptions(const VideoOptions& options);
- virtual bool GetOptions(VideoOptions* options) const;
- virtual bool AddSendStream(const StreamParams& sp);
- virtual bool RemoveSendStream(uint32 ssrc);
- virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer);
- virtual bool SetRender(bool render);
- virtual bool MuteStream(uint32 ssrc, bool muted);
-
- virtual bool SetRecvCodecs(const std::vector<VideoCodec>& codecs);
- virtual bool SetRecvRtpHeaderExtensions(
- const std::vector<RtpHeaderExtension>& extensions);
-
- virtual bool SetSendCodecs(const std::vector<VideoCodec>& codecs);
- virtual bool GetSendCodec(VideoCodec* codec);
- virtual bool SetSendStreamFormat(uint32 ssrc, const VideoFormat& format);
- virtual bool SetSendRtpHeaderExtensions(
- const std::vector<RtpHeaderExtension>& extensions);
- virtual bool SetStartSendBandwidth(int bps);
- virtual bool SetMaxSendBandwidth(int bps);
- virtual bool SetSend(bool send);
-
- virtual bool AddRecvStream(const StreamParams& sp);
- virtual bool RemoveRecvStream(uint32 ssrc);
- virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer);
-
- virtual bool SendIntraFrame();
- virtual bool RequestIntraFrame();
-
- virtual bool GetStats(const StatsOptions& options, VideoMediaInfo* info);
-
- virtual void OnPacketReceived(rtc::Buffer* packet,
- const rtc::PacketTime& packet_time);
- virtual void OnRtcpReceived(rtc::Buffer* packet,
- const rtc::PacketTime& packet_time);
- virtual void OnReadyToSend(bool ready);
-
- virtual void UpdateAspectRatio(int ratio_w, int ratio_h);
-
- void OnLocalFrame(VideoCapturer*, const VideoFrame*);
- void OnLocalFrameFormat(VideoCapturer*, const VideoFormat*);
-
- bool sending() const { return sending_; }
-
- private:
- bool SelectActiveChannel(const std::vector<VideoCodec>& codecs);
- void SplitCodecs(const std::vector<VideoCodec>& codecs,
- std::vector<VideoCodec>* codecs1,
- std::vector<VideoCodec>* codecs2);
-
- void OnMediaError(uint32 ssrc, Error error);
-
- HybridVideoEngineInterface* engine_;
- rtc::scoped_ptr<VideoMediaChannel> channel1_;
- rtc::scoped_ptr<VideoMediaChannel> channel2_;
- VideoMediaChannel* active_channel_;
- bool sending_;
-};
-
-// Interface class for HybridVideoChannels to talk to the engine.
-class HybridVideoEngineInterface {
- public:
- virtual ~HybridVideoEngineInterface() {}
- virtual bool HasCodec1(const VideoCodec& codec) = 0;
- virtual bool HasCodec2(const VideoCodec& codec) = 0;
- virtual void OnSendChange1(VideoMediaChannel* channel1, bool send) = 0;
- virtual void OnSendChange2(VideoMediaChannel* channel1, bool send) = 0;
- virtual void OnNewSendResolution(int width, int height) = 0;
-};
-
-// The HybridVideoEngine class combines two unrelated VideoEngine impls
-// into a single class. It creates HybridVideoMediaChannels that also contain
-// a VideoMediaChannel implementation from each engine. Policy is then used
-// during call setup to determine which VideoMediaChannel should be used.
-// Currently, this policy is based on what codec the remote side wants to use.
-template<class VIDEO1, class VIDEO2>
-class HybridVideoEngine : public HybridVideoEngineInterface {
- public:
- HybridVideoEngine() {
- // Unify the codec lists.
- codecs_ = video1_.codecs();
- codecs_.insert(codecs_.end(), video2_.codecs().begin(),
- video2_.codecs().end());
-
- rtp_header_extensions_ = video1_.rtp_header_extensions();
- rtp_header_extensions_.insert(rtp_header_extensions_.end(),
- video2_.rtp_header_extensions().begin(),
- video2_.rtp_header_extensions().end());
-
- SignalCaptureStateChange.repeat(video2_.SignalCaptureStateChange);
- }
-
- bool Init(rtc::Thread* worker_thread) {
- if (!video1_.Init(worker_thread)) {
- LOG(LS_ERROR) << "Failed to init VideoEngine1";
- return false;
- }
- if (!video2_.Init(worker_thread)) {
- LOG(LS_ERROR) << "Failed to init VideoEngine2";
- video1_.Terminate();
- return false;
- }
- return true;
- }
- void Terminate() {
- video1_.Terminate();
- video2_.Terminate();
- }
-
- int GetCapabilities() {
- return (video1_.GetCapabilities() | video2_.GetCapabilities());
- }
- HybridVideoMediaChannel* CreateChannel(VoiceMediaChannel* channel) {
- rtc::scoped_ptr<VideoMediaChannel> channel1(
- video1_.CreateChannel(channel));
- if (!channel1) {
- LOG(LS_ERROR) << "Failed to create VideoMediaChannel1";
- return NULL;
- }
- rtc::scoped_ptr<VideoMediaChannel> channel2(
- video2_.CreateChannel(channel));
- if (!channel2) {
- LOG(LS_ERROR) << "Failed to create VideoMediaChannel2";
- return NULL;
- }
- return new HybridVideoMediaChannel(this,
- channel1.release(), channel2.release());
- }
-
- bool SetOptions(const VideoOptions& options) {
- return video1_.SetOptions(options) && video2_.SetOptions(options);
- }
- bool SetDefaultEncoderConfig(const VideoEncoderConfig& config) {
- VideoEncoderConfig conf = config;
- if (video1_.codecs().size() > 0) {
- conf.max_codec.name = video1_.codecs()[0].name;
- if (!video1_.SetDefaultEncoderConfig(conf)) {
- LOG(LS_ERROR) << "Failed to SetDefaultEncoderConfig for video1";
- return false;
- }
- }
- if (video2_.codecs().size() > 0) {
- conf.max_codec.name = video2_.codecs()[0].name;
- if (!video2_.SetDefaultEncoderConfig(conf)) {
- LOG(LS_ERROR) << "Failed to SetDefaultEncoderConfig for video2";
- return false;
- }
- }
- return true;
- }
- VideoEncoderConfig GetDefaultEncoderConfig() const {
- // This looks pretty strange, but, in practice, it'll do sane things if
- // GetDefaultEncoderConfig is only called after SetDefaultEncoderConfig,
- // since both engines should be essentially equivalent at that point. If it
- // hasn't been called, though, we'll use the first meaningful encoder
- // config, or the config from the second video engine if neither are
- // meaningful.
- VideoEncoderConfig config = video1_.GetDefaultEncoderConfig();
- if (config.max_codec.width != 0) {
- return config;
- } else {
- return video2_.GetDefaultEncoderConfig();
- }
- }
- const std::vector<VideoCodec>& codecs() const {
- return codecs_;
- }
- const std::vector<RtpHeaderExtension>& rtp_header_extensions() const {
- return rtp_header_extensions_;
- }
- void SetLogging(int min_sev, const char* filter) {
- video1_.SetLogging(min_sev, filter);
- video2_.SetLogging(min_sev, filter);
- }
-
- VideoFormat GetStartCaptureFormat() const {
- return video2_.GetStartCaptureFormat();
- }
-
- // TODO(juberti): Remove these functions after we do the capturer refactoring.
- // For now they are set to always use the second engine for capturing, which
- // is convenient given our intended use case.
- bool SetCaptureDevice(const Device* device) {
- return video2_.SetCaptureDevice(device);
- }
- VideoCapturer* GetVideoCapturer() const {
- return video2_.GetVideoCapturer();
- }
- bool SetLocalRenderer(VideoRenderer* renderer) {
- return video2_.SetLocalRenderer(renderer);
- }
- sigslot::repeater2<VideoCapturer*, CaptureState> SignalCaptureStateChange;
-
- virtual bool HasCodec1(const VideoCodec& codec) {
- return HasCodec(video1_, codec);
- }
- virtual bool HasCodec2(const VideoCodec& codec) {
- return HasCodec(video2_, codec);
- }
- template<typename VIDEO>
- bool HasCodec(const VIDEO& engine, const VideoCodec& codec) const {
- for (std::vector<VideoCodec>::const_iterator i = engine.codecs().begin();
- i != engine.codecs().end();
- ++i) {
- if (i->Matches(codec)) {
- return true;
- }
- }
- return false;
- }
- virtual void OnSendChange1(VideoMediaChannel* channel1, bool send) {
- }
- virtual void OnSendChange2(VideoMediaChannel* channel2, bool send) {
- }
- virtual void OnNewSendResolution(int width, int height) {
- }
-
- protected:
- VIDEO1 video1_;
- VIDEO2 video2_;
- std::vector<VideoCodec> codecs_;
- std::vector<RtpHeaderExtension> rtp_header_extensions_;
-};
-
-} // namespace cricket
-
-#endif // TALK_MEDIA_BASE_HYBRIDVIDEOENGINE_H_
diff --git a/media/base/hybridvideoengine_unittest.cc b/media/base/hybridvideoengine_unittest.cc
deleted file mode 100644
index 7b409ea..0000000
--- a/media/base/hybridvideoengine_unittest.cc
+++ /dev/null
@@ -1,486 +0,0 @@
-/*
- * libjingle
- * Copyright 2004 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/media/base/fakemediaengine.h"
-#include "talk/media/base/fakenetworkinterface.h"
-#include "talk/media/base/fakevideocapturer.h"
-#include "talk/media/base/hybridvideoengine.h"
-#include "talk/media/base/mediachannel.h"
-#include "talk/media/base/testutils.h"
-#include "webrtc/base/gunit.h"
-
-static const cricket::VideoCodec kGenericCodec(97, "Generic", 640, 360, 30, 0);
-static const cricket::VideoCodec kVp8Codec(100, "VP8", 640, 360, 30, 0);
-static const cricket::VideoCodec kCodecsVp8Only[] = { kVp8Codec };
-static const cricket::VideoCodec kCodecsGenericOnly[] = { kGenericCodec };
-static const cricket::VideoCodec kCodecsVp8First[] = { kVp8Codec,
- kGenericCodec };
-static const cricket::VideoCodec kCodecsGenericFirst[] = { kGenericCodec,
- kVp8Codec };
-
-using cricket::StreamParams;
-
-class FakeVp8VideoEngine : public cricket::FakeVideoEngine {
- public:
- FakeVp8VideoEngine() {
- SetCodecs(MAKE_VECTOR(kCodecsVp8Only));
- }
-};
-class FakeGenericVideoEngine : public cricket::FakeVideoEngine {
- public:
- FakeGenericVideoEngine() {
- SetCodecs(MAKE_VECTOR(kCodecsGenericOnly));
- }
-
- // For testing purposes, mimic the behavior of a media engine that throws out
- // resolutions that don't match the codec list. A width or height of 0
- // trivially will never match the codec list, so this is sufficient for
- // testing the case we want (0x0).
- virtual bool FindCodec(const cricket::VideoCodec& codec) {
- if (codec.width == 0 || codec.height == 0) {
- return false;
- } else {
- return cricket::FakeVideoEngine::FindCodec(codec);
- }
- }
-};
-class HybridVideoEngineForTest : public cricket::HybridVideoEngine<
- FakeVp8VideoEngine, FakeGenericVideoEngine> {
- public:
- HybridVideoEngineForTest()
- :
- num_ch1_send_on_(0),
- num_ch1_send_off_(0),
- send_width_(0),
- send_height_(0) { }
- cricket::FakeVideoEngine* sub_engine1() { return &video1_; }
- cricket::FakeVideoEngine* sub_engine2() { return &video2_; }
-
- // From base class HybridVideoEngine.
- void OnSendChange1(cricket::VideoMediaChannel* channel1, bool send) {
- if (send) {
- ++num_ch1_send_on_;
- } else {
- ++num_ch1_send_off_;
- }
- }
- // From base class HybridVideoEngine
- void OnNewSendResolution(int width, int height) {
- send_width_ = width;
- send_height_ = height;
- }
-
- int num_ch1_send_on() const { return num_ch1_send_on_; }
- int num_ch1_send_off() const { return num_ch1_send_off_; }
-
- int send_width() const { return send_width_; }
- int send_height() const { return send_height_; }
-
- private:
- int num_ch1_send_on_;
- int num_ch1_send_off_;
-
- int send_width_;
- int send_height_;
-};
-
-class HybridVideoEngineTest : public testing::Test {
- public:
- HybridVideoEngineTest() : sub_channel1_(NULL), sub_channel2_(NULL) {
- }
- ~HybridVideoEngineTest() {
- engine_.Terminate();
- }
- bool SetupEngine() {
- bool result = engine_.Init(rtc::Thread::Current());
- if (result) {
- channel_.reset(engine_.CreateChannel(NULL));
- result = (channel_.get() != NULL);
- sub_channel1_ = engine_.sub_engine1()->GetChannel(0);
- sub_channel2_ = engine_.sub_engine2()->GetChannel(0);
- }
- return result;
- }
- bool SetupRenderAndAddStream(const StreamParams& sp) {
- if (!SetupEngine())
- return false;
- channel_->SetInterface(transport_.get());
- return channel_->SetRecvCodecs(engine_.codecs()) &&
- channel_->AddSendStream(sp) &&
- channel_->SetRender(true);
- }
- void DeliverPacket(const void* data, int len) {
- rtc::Buffer packet(data, len);
- channel_->OnPacketReceived(&packet, rtc::CreatePacketTime(0));
- }
- void DeliverRtcp(const void* data, int len) {
- rtc::Buffer packet(data, len);
- channel_->OnRtcpReceived(&packet, rtc::CreatePacketTime(0));
- }
-
- protected:
- void TestSetSendCodecs(cricket::FakeVideoEngine* sub_engine,
- const std::vector<cricket::VideoCodec>& codecs) {
- EXPECT_TRUE(SetupRenderAndAddStream(StreamParams::CreateLegacy(1234)));
- EXPECT_TRUE(channel_->SetSendCodecs(codecs));
- cricket::FakeVideoMediaChannel* sub_channel = sub_engine->GetChannel(0);
- ASSERT_EQ(1U, sub_channel->send_codecs().size());
- EXPECT_EQ(codecs[0], sub_channel->send_codecs()[0]);
- EXPECT_TRUE(channel_->SetSend(true));
- EXPECT_TRUE(sub_channel->sending());
- }
- void TestSetSendBandwidth(cricket::FakeVideoEngine* sub_engine,
- const std::vector<cricket::VideoCodec>& codecs,
- int start_bitrate,
- int max_bitrate) {
- EXPECT_TRUE(SetupRenderAndAddStream(StreamParams::CreateLegacy(1234)));
- EXPECT_TRUE(channel_->SetSendCodecs(codecs));
- EXPECT_TRUE(channel_->SetStartSendBandwidth(start_bitrate));
- EXPECT_TRUE(channel_->SetMaxSendBandwidth(max_bitrate));
- cricket::FakeVideoMediaChannel* sub_channel = sub_engine->GetChannel(0);
- EXPECT_EQ(start_bitrate, sub_channel->start_bps());
- EXPECT_EQ(max_bitrate, sub_channel->max_bps());
- }
- HybridVideoEngineForTest engine_;
- rtc::scoped_ptr<cricket::HybridVideoMediaChannel> channel_;
- rtc::scoped_ptr<cricket::FakeNetworkInterface> transport_;
- cricket::FakeVideoMediaChannel* sub_channel1_;
- cricket::FakeVideoMediaChannel* sub_channel2_;
-};
-
-TEST_F(HybridVideoEngineTest, StartupShutdown) {
- EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
- engine_.Terminate();
-}
-
-// Tests that SetDefaultVideoEncoderConfig passes down to both engines.
-TEST_F(HybridVideoEngineTest, SetDefaultVideoEncoderConfig) {
- cricket::VideoEncoderConfig config(
- cricket::VideoCodec(105, "", 640, 400, 30, 0), 1, 2);
- EXPECT_TRUE(engine_.SetDefaultEncoderConfig(config));
-
- cricket::VideoEncoderConfig config_1 = config;
- config_1.max_codec.name = kCodecsVp8Only[0].name;
- EXPECT_EQ(config_1, engine_.sub_engine1()->default_encoder_config());
-
- cricket::VideoEncoderConfig config_2 = config;
- config_2.max_codec.name = kCodecsGenericOnly[0].name;
- EXPECT_EQ(config_2, engine_.sub_engine2()->default_encoder_config());
-}
-
-// Tests that GetDefaultVideoEncoderConfig picks a meaningful encoder config
-// based on the underlying engine config and then after a call to
-// SetDefaultEncoderConfig on the hybrid engine.
-TEST_F(HybridVideoEngineTest, SetDefaultVideoEncoderConfigDefaultValue) {
- cricket::VideoEncoderConfig blank_config;
- cricket::VideoEncoderConfig meaningful_config1(
- cricket::VideoCodec(111, "abcd", 320, 240, 30, 0), 1, 2);
- cricket::VideoEncoderConfig meaningful_config2(
- cricket::VideoCodec(111, "abcd", 1280, 720, 30, 0), 1, 2);
- cricket::VideoEncoderConfig meaningful_config3(
- cricket::VideoCodec(111, "abcd", 640, 360, 30, 0), 1, 2);
- engine_.sub_engine1()->SetDefaultEncoderConfig(blank_config);
- engine_.sub_engine2()->SetDefaultEncoderConfig(blank_config);
- EXPECT_EQ(blank_config, engine_.GetDefaultEncoderConfig());
-
- engine_.sub_engine2()->SetDefaultEncoderConfig(meaningful_config2);
- EXPECT_EQ(meaningful_config2, engine_.GetDefaultEncoderConfig());
-
- engine_.sub_engine1()->SetDefaultEncoderConfig(meaningful_config1);
- EXPECT_EQ(meaningful_config1, engine_.GetDefaultEncoderConfig());
-
- EXPECT_TRUE(engine_.SetDefaultEncoderConfig(meaningful_config3));
- // The overall config should now match, though the codec name will have been
- // rewritten for the first media engine.
- meaningful_config3.max_codec.name = kCodecsVp8Only[0].name;
- EXPECT_EQ(meaningful_config3, engine_.GetDefaultEncoderConfig());
-}
-
-// Tests that our engine has the right codecs in the right order.
-TEST_F(HybridVideoEngineTest, CheckCodecs) {
- const std::vector<cricket::VideoCodec>& c = engine_.codecs();
- ASSERT_EQ(2U, c.size());
- EXPECT_EQ(kVp8Codec, c[0]);
- EXPECT_EQ(kGenericCodec, c[1]);
-}
-
-// Tests that our engine has the right caps.
-TEST_F(HybridVideoEngineTest, CheckCaps) {
- EXPECT_EQ(cricket::VIDEO_SEND | cricket::VIDEO_RECV,
- engine_.GetCapabilities());
-}
-
-// Tests that we can create and destroy a channel.
-TEST_F(HybridVideoEngineTest, CreateChannel) {
- EXPECT_TRUE(SetupEngine());
- EXPECT_TRUE(sub_channel1_ != NULL);
- EXPECT_TRUE(sub_channel2_ != NULL);
-}
-
-// Tests that we properly handle failures in CreateChannel.
-TEST_F(HybridVideoEngineTest, CreateChannelFail) {
- engine_.sub_engine1()->set_fail_create_channel(true);
- EXPECT_FALSE(SetupEngine());
- EXPECT_TRUE(channel_.get() == NULL);
- EXPECT_TRUE(sub_channel1_ == NULL);
- EXPECT_TRUE(sub_channel2_ == NULL);
- engine_.sub_engine1()->set_fail_create_channel(false);
- engine_.sub_engine2()->set_fail_create_channel(true);
- EXPECT_FALSE(SetupEngine());
- EXPECT_TRUE(channel_.get() == NULL);
- EXPECT_TRUE(sub_channel1_ == NULL);
- EXPECT_TRUE(sub_channel2_ == NULL);
-}
-
-// Test that we set our inbound codecs and settings properly.
-TEST_F(HybridVideoEngineTest, SetLocalDescription) {
- EXPECT_TRUE(SetupEngine());
- channel_->SetInterface(transport_.get());
- EXPECT_TRUE(channel_->SetRecvCodecs(engine_.codecs()));
- ASSERT_EQ(1U, sub_channel1_->recv_codecs().size());
- ASSERT_EQ(1U, sub_channel2_->recv_codecs().size());
- EXPECT_EQ(kVp8Codec, sub_channel1_->recv_codecs()[0]);
- EXPECT_EQ(kGenericCodec, sub_channel2_->recv_codecs()[0]);
- StreamParams stream;
- stream.id = "TestStream";
- stream.ssrcs.push_back(1234);
- stream.cname = "5678";
- EXPECT_TRUE(channel_->AddSendStream(stream));
- EXPECT_EQ(1234U, sub_channel1_->send_ssrc());
- EXPECT_EQ(1234U, sub_channel2_->send_ssrc());
- EXPECT_EQ("5678", sub_channel1_->rtcp_cname());
- EXPECT_EQ("5678", sub_channel2_->rtcp_cname());
- EXPECT_TRUE(channel_->SetRender(true));
- // We've called SetRender, so we should be playing out, but not yet sending.
- EXPECT_TRUE(sub_channel1_->playout());
- EXPECT_TRUE(sub_channel2_->playout());
- EXPECT_FALSE(sub_channel1_->sending());
- EXPECT_FALSE(sub_channel2_->sending());
- // We may get SetSend(false) calls during call setup.
- // Since this causes no change in state, they should no-op and return true.
- EXPECT_TRUE(channel_->SetSend(false));
- EXPECT_FALSE(sub_channel1_->sending());
- EXPECT_FALSE(sub_channel2_->sending());
-}
-
-TEST_F(HybridVideoEngineTest, OnNewSendResolution) {
- EXPECT_TRUE(SetupEngine());
- EXPECT_TRUE(channel_->SetSendCodecs(MAKE_VECTOR(kCodecsVp8First)));
- EXPECT_EQ(640, engine_.send_width());
- EXPECT_EQ(360, engine_.send_height());
-}
-
-// Test that we converge to the active channel for engine 1.
-TEST_F(HybridVideoEngineTest, SetSendCodecs1) {
- // This will nuke the object that sub_channel2_ points to.
- TestSetSendCodecs(engine_.sub_engine1(), MAKE_VECTOR(kCodecsVp8First));
- EXPECT_TRUE(engine_.sub_engine2()->GetChannel(0) == NULL);
-}
-
-// Test that we converge to the active channel for engine 2.
-TEST_F(HybridVideoEngineTest, SetSendCodecs2) {
- // This will nuke the object that sub_channel1_ points to.
- TestSetSendCodecs(engine_.sub_engine2(), MAKE_VECTOR(kCodecsGenericFirst));
- EXPECT_TRUE(engine_.sub_engine1()->GetChannel(0) == NULL);
-}
-
-// Test that we don't accidentally eat 0x0 in SetSendCodecs
-TEST_F(HybridVideoEngineTest, SetSendCodecs0x0) {
- EXPECT_TRUE(SetupRenderAndAddStream(StreamParams::CreateLegacy(1234)));
- // Send using generic codec, but with 0x0 resolution.
- std::vector<cricket::VideoCodec> codecs(MAKE_VECTOR(kCodecsGenericFirst));
- codecs.resize(1);
- codecs[0].width = 0;
- codecs[0].height = 0;
- EXPECT_TRUE(channel_->SetSendCodecs(codecs));
-}
-
-// Test setting the send bandwidth for VP8.
-TEST_F(HybridVideoEngineTest, SetSendBandwidth1) {
- TestSetSendBandwidth(engine_.sub_engine1(),
- MAKE_VECTOR(kCodecsVp8First),
- 100000,
- 384000);
-}
-
-// Test setting the send bandwidth for a generic codec.
-TEST_F(HybridVideoEngineTest, SetSendBandwidth2) {
- TestSetSendBandwidth(engine_.sub_engine2(),
- MAKE_VECTOR(kCodecsGenericFirst),
- 100001,
- 384002);
-}
-
-// Test that we dump RTP packets that arrive early.
-TEST_F(HybridVideoEngineTest, HandleEarlyRtp) {
- static const uint8 kPacket[1024] = { 0 };
- static const uint8 kRtcp[1024] = { 1 };
- EXPECT_TRUE(SetupRenderAndAddStream(StreamParams::CreateLegacy(1234)));
- DeliverPacket(kPacket, sizeof(kPacket));
- DeliverRtcp(kRtcp, sizeof(kRtcp));
- EXPECT_TRUE(sub_channel1_->CheckNoRtp());
- EXPECT_TRUE(sub_channel2_->CheckNoRtp());
- EXPECT_TRUE(sub_channel1_->CheckNoRtcp());
- EXPECT_TRUE(sub_channel2_->CheckNoRtcp());
-}
-
-// Test that we properly pass on normal RTP packets.
-TEST_F(HybridVideoEngineTest, HandleRtp) {
- static const uint8 kPacket[1024] = { 0 };
- static const uint8 kRtcp[1024] = { 1 };
- EXPECT_TRUE(SetupRenderAndAddStream(StreamParams::CreateLegacy(1234)));
- EXPECT_TRUE(channel_->SetSendCodecs(MAKE_VECTOR(kCodecsVp8First)));
- EXPECT_TRUE(channel_->SetSend(true));
- DeliverPacket(kPacket, sizeof(kPacket));
- DeliverRtcp(kRtcp, sizeof(kRtcp));
- EXPECT_TRUE(sub_channel1_->CheckRtp(kPacket, sizeof(kPacket)));
- EXPECT_TRUE(sub_channel1_->CheckRtcp(kRtcp, sizeof(kRtcp)));
-}
-
-// Test that we properly connect media error signal.
-TEST_F(HybridVideoEngineTest, MediaErrorSignal) {
- cricket::VideoMediaErrorCatcher catcher;
-
- // Verify no signal from either channel before the active channel is set.
- EXPECT_TRUE(SetupEngine());
- channel_->SignalMediaError.connect(&catcher,
- &cricket::VideoMediaErrorCatcher::OnError);
- sub_channel1_->SignalMediaError(1, cricket::VideoMediaChannel::ERROR_OTHER);
- EXPECT_EQ(0U, catcher.ssrc());
- sub_channel2_->SignalMediaError(2,
- cricket::VideoMediaChannel::ERROR_REC_DEVICE_OPEN_FAILED);
- EXPECT_EQ(0U, catcher.ssrc());
-
- // Set vp8 as active channel and verify that a signal comes from it.
- EXPECT_TRUE(channel_->SetSendCodecs(MAKE_VECTOR(kCodecsVp8First)));
- sub_channel1_->SignalMediaError(1, cricket::VideoMediaChannel::ERROR_OTHER);
- EXPECT_EQ(cricket::VideoMediaChannel::ERROR_OTHER, catcher.error());
- EXPECT_EQ(1U, catcher.ssrc());
-
- // Set generic codec as active channel and verify that a signal comes from it.
- EXPECT_TRUE(SetupEngine());
- channel_->SignalMediaError.connect(&catcher,
- &cricket::VideoMediaErrorCatcher::OnError);
- EXPECT_TRUE(channel_->SetSendCodecs(MAKE_VECTOR(kCodecsGenericFirst)));
- sub_channel2_->SignalMediaError(2,
- cricket::VideoMediaChannel::ERROR_REC_DEVICE_OPEN_FAILED);
- EXPECT_EQ(cricket::VideoMediaChannel::ERROR_REC_DEVICE_OPEN_FAILED,
- catcher.error());
- EXPECT_EQ(2U, catcher.ssrc());
-}
-
-// Test that SetSend doesn't re-enter.
-TEST_F(HybridVideoEngineTest, RepeatSetSend) {
- EXPECT_TRUE(SetupEngine());
- EXPECT_TRUE(channel_->SetSendCodecs(MAKE_VECTOR(kCodecsVp8First)));
-
- // Verify initial status.
- EXPECT_FALSE(channel_->sending());
- EXPECT_FALSE(sub_channel1_->sending());
- EXPECT_EQ(0, engine_.num_ch1_send_on());
- EXPECT_EQ(0, engine_.num_ch1_send_off());
-
- // Verfiy SetSend(true) works correctly.
- EXPECT_TRUE(channel_->SetSend(true));
- EXPECT_TRUE(channel_->sending());
- EXPECT_TRUE(sub_channel1_->sending());
- EXPECT_EQ(1, engine_.num_ch1_send_on());
- EXPECT_EQ(0, engine_.num_ch1_send_off());
-
- // SetSend(true) again and verify nothing changes.
- EXPECT_TRUE(channel_->SetSend(true));
- EXPECT_TRUE(channel_->sending());
- EXPECT_TRUE(sub_channel1_->sending());
- EXPECT_EQ(1, engine_.num_ch1_send_on());
- EXPECT_EQ(0, engine_.num_ch1_send_off());
-
- // Verify SetSend(false) works correctly.
- EXPECT_TRUE(channel_->SetSend(false));
- EXPECT_FALSE(channel_->sending());
- EXPECT_FALSE(sub_channel1_->sending());
- EXPECT_EQ(1, engine_.num_ch1_send_on());
- EXPECT_EQ(1, engine_.num_ch1_send_off());
-
- // SetSend(false) again and verfiy nothing changes.
- EXPECT_TRUE(channel_->SetSend(false));
- EXPECT_FALSE(channel_->sending());
- EXPECT_FALSE(sub_channel1_->sending());
- EXPECT_EQ(1, engine_.num_ch1_send_on());
- EXPECT_EQ(1, engine_.num_ch1_send_off());
-}
-
-// Test that SetOptions.
-TEST_F(HybridVideoEngineTest, SetOptions) {
- cricket::VideoOptions vmo;
- vmo.video_high_bitrate.Set(true);
- vmo.system_low_adaptation_threshhold.Set(0.10f);
- EXPECT_TRUE(SetupEngine());
- EXPECT_TRUE(channel_->SetOptions(vmo));
-
- bool high_bitrate;
- float low;
- EXPECT_TRUE(sub_channel1_->GetOptions(&vmo));
- EXPECT_TRUE(vmo.video_high_bitrate.Get(&high_bitrate));
- EXPECT_TRUE(high_bitrate);
- EXPECT_TRUE(vmo.system_low_adaptation_threshhold.Get(&low));
- EXPECT_EQ(0.10f, low);
- EXPECT_TRUE(sub_channel2_->GetOptions(&vmo));
- EXPECT_TRUE(vmo.video_high_bitrate.Get(&high_bitrate));
- EXPECT_TRUE(high_bitrate);
- EXPECT_TRUE(vmo.system_low_adaptation_threshhold.Get(&low));
- EXPECT_EQ(0.10f, low);
-
- vmo.video_high_bitrate.Set(false);
- vmo.system_low_adaptation_threshhold.Set(0.50f);
-
- EXPECT_TRUE(channel_->SetOptions(vmo));
- EXPECT_TRUE(sub_channel1_->GetOptions(&vmo));
- EXPECT_TRUE(vmo.video_high_bitrate.Get(&high_bitrate));
- EXPECT_FALSE(high_bitrate);
- EXPECT_TRUE(vmo.system_low_adaptation_threshhold.Get(&low));
- EXPECT_EQ(0.50f, low);
- EXPECT_TRUE(sub_channel2_->GetOptions(&vmo));
- EXPECT_TRUE(vmo.video_high_bitrate.Get(&high_bitrate));
- EXPECT_FALSE(high_bitrate);
- EXPECT_TRUE(vmo.system_low_adaptation_threshhold.Get(&low));
- EXPECT_EQ(0.50f, low);
-}
-
-TEST_F(HybridVideoEngineTest, SetCapturer) {
- EXPECT_TRUE(SetupEngine());
- // Set vp8 as active channel and verify that capturer can be set.
- EXPECT_TRUE(channel_->SetSendCodecs(MAKE_VECTOR(kCodecsVp8First)));
- cricket::FakeVideoCapturer fake_video_capturer;
- EXPECT_TRUE(channel_->SetCapturer(0, &fake_video_capturer));
- EXPECT_TRUE(channel_->SetCapturer(0, NULL));
-
- // Set generic codec active channel and verify that capturer can be set.
- EXPECT_TRUE(SetupEngine());
- EXPECT_TRUE(channel_->SetSendCodecs(MAKE_VECTOR(kCodecsGenericFirst)));
- EXPECT_TRUE(channel_->SetCapturer(0, &fake_video_capturer));
- EXPECT_TRUE(channel_->SetCapturer(0, NULL));
-}
diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
index 62d6b61..5232e5d 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -182,6 +182,7 @@
recording_sample_rate.SetFrom(change.recording_sample_rate);
playout_sample_rate.SetFrom(change.playout_sample_rate);
dscp.SetFrom(change.dscp);
+ combined_audio_video_bwe.SetFrom(change.combined_audio_video_bwe);
}
bool operator==(const AudioOptions& o) const {
@@ -207,7 +208,8 @@
rx_agc_limiter == o.rx_agc_limiter &&
recording_sample_rate == o.recording_sample_rate &&
playout_sample_rate == o.playout_sample_rate &&
- dscp == o.dscp;
+ dscp == o.dscp &&
+ combined_audio_video_bwe == o.combined_audio_video_bwe;
}
std::string ToString() const {
@@ -238,6 +240,7 @@
ost << ToStringIfSet("recording_sample_rate", recording_sample_rate);
ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
ost << ToStringIfSet("dscp", dscp);
+ ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
ost << "}";
return ost.str();
}
@@ -275,6 +278,8 @@
Settable<uint32> playout_sample_rate;
// Set DSCP value for packet sent from audio channel.
Settable<bool> dscp;
+ // Enable combined audio+bandwidth BWE.
+ Settable<bool> combined_audio_video_bwe;
};
// Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
diff --git a/media/base/mediaengine.h b/media/base/mediaengine.h
index 6c4b740..f30e3b1 100644
--- a/media/base/mediaengine.h
+++ b/media/base/mediaengine.h
@@ -91,8 +91,6 @@
virtual AudioOptions GetAudioOptions() const = 0;
// Sets global audio options. "options" are from AudioOptions, above.
virtual bool SetAudioOptions(const AudioOptions& options) = 0;
- // Sets global video options. "options" are from VideoOptions, above.
- virtual bool SetVideoOptions(const VideoOptions& options) = 0;
// Sets the value used by the echo canceller to offset delay values obtained
// from the OS.
virtual bool SetAudioDelayOffset(int offset) = 0;
@@ -124,7 +122,6 @@
// when a VoiceMediaChannel starts sending.
virtual bool SetLocalMonitor(bool enable) = 0;
// Installs a callback for raw frames from the local camera.
- virtual bool SetLocalRenderer(VideoRenderer* renderer) = 0;
virtual const std::vector<AudioCodec>& audio_codecs() = 0;
virtual const std::vector<RtpHeaderExtension>&
@@ -214,9 +211,6 @@
virtual bool SetAudioOptions(const AudioOptions& options) {
return voice_.SetOptions(options);
}
- virtual bool SetVideoOptions(const VideoOptions& options) {
- return video_.SetOptions(options);
- }
virtual bool SetAudioDelayOffset(int offset) {
return voice_.SetDelayOffset(offset);
}
@@ -245,10 +239,6 @@
virtual bool SetLocalMonitor(bool enable) {
return voice_.SetLocalMonitor(enable);
}
- virtual bool SetLocalRenderer(VideoRenderer* renderer) {
- return video_.SetLocalRenderer(renderer);
- }
-
virtual const std::vector<AudioCodec>& audio_codecs() {
return voice_.codecs();
}
@@ -361,7 +351,6 @@
bool SetDefaultEncoderConfig(const VideoEncoderConfig& config) {
return true;
}
- bool SetLocalRenderer(VideoRenderer* renderer) { return true; }
const std::vector<VideoCodec>& codecs() { return codecs_; }
const std::vector<RtpHeaderExtension>& rtp_header_extensions() {
return rtp_header_extensions_;
diff --git a/media/base/testutils.cc b/media/base/testutils.cc
index 8b79df4..84fd05c 100644
--- a/media/base/testutils.cc
+++ b/media/base/testutils.cc
@@ -29,6 +29,7 @@
#include <math.h>
+#include "talk/media/base/executablehelpers.h"
#include "talk/media/base/rtpdump.h"
#include "talk/media/base/videocapturer.h"
#include "talk/media/base/videoframe.h"
@@ -255,10 +256,15 @@
// Returns the absolute path to a file in the testdata/ directory.
std::string GetTestFilePath(const std::string& filename) {
// Locate test data directory.
+#ifdef ENABLE_WEBRTC
+ rtc::Pathname path = rtc::GetExecutablePath();
+ EXPECT_FALSE(path.empty());
+ path.AppendPathname("../../talk/");
+#else
rtc::Pathname path = testing::GetTalkDirectory();
EXPECT_FALSE(path.empty()); // must be run from inside "talk"
- path.AppendFolder("media");
- path.AppendFolder("testdata");
+#endif
+ path.AppendFolder("media/testdata/");
path.SetFilename(filename);
return path.pathname();
}
diff --git a/media/base/videoadapter_unittest.cc b/media/base/videoadapter_unittest.cc
index af374e0..04bf3d1 100755
--- a/media/base/videoadapter_unittest.cc
+++ b/media/base/videoadapter_unittest.cc
@@ -69,17 +69,24 @@
listener_.get(), &VideoCapturerListener::OnFrameCaptured);
}
- void VerifyAdaptedResolution(int width, int height) {
- EXPECT_TRUE(NULL != listener_->adapted_frame());
- EXPECT_EQ(static_cast<size_t>(width),
- listener_->adapted_frame()->GetWidth());
- EXPECT_EQ(static_cast<size_t>(height),
- listener_->adapted_frame()->GetHeight());
+ virtual void TearDown() {
+ // Explicitly disconnect the VideoCapturer before to avoid data races
+ // (frames delivered to VideoCapturerListener while it's being destructed).
+ capturer_->SignalFrameCaptured.disconnect_all();
}
protected:
class VideoCapturerListener: public sigslot::has_slots<> {
public:
+ struct Stats {
+ int captured_frames;
+ int dropped_frames;
+ bool last_adapt_was_no_op;
+
+ int adapted_width;
+ int adapted_height;
+ };
+
explicit VideoCapturerListener(VideoAdapter* adapter)
: video_adapter_(adapter),
adapted_frame_(NULL),
@@ -95,6 +102,7 @@
EXPECT_TRUE(temp_i420.Init(captured_frame,
captured_frame->width, abs(captured_frame->height)));
VideoFrame* out_frame = NULL;
+ rtc::CritScope lock(&crit_);
EXPECT_TRUE(video_adapter_->AdaptFrame(&temp_i420, &out_frame));
if (out_frame) {
if (out_frame == &temp_i420) {
@@ -112,12 +120,32 @@
++captured_frames_;
}
- const VideoFrame* adapted_frame() const { return adapted_frame_; }
- int captured_frames() const { return captured_frames_; }
- int dropped_frames() const { return dropped_frames_; }
- bool last_adapt_was_no_op() const { return last_adapt_was_no_op_; }
+ Stats GetStats() {
+ rtc::CritScope lock(&crit_);
+ Stats stats;
+ stats.captured_frames = captured_frames_;
+ stats.dropped_frames = dropped_frames_;
+ stats.last_adapt_was_no_op = last_adapt_was_no_op_;
+ if (adapted_frame_ != NULL) {
+ stats.adapted_width = static_cast<int>(adapted_frame_->GetWidth());
+ stats.adapted_height = static_cast<int>(adapted_frame_->GetHeight());
+ } else {
+ stats.adapted_width = stats.adapted_height = -1;
+ }
+
+ return stats;
+ }
+
+ VideoFrame* CopyAdaptedFrame() {
+ rtc::CritScope lock(&crit_);
+ if (adapted_frame_ == NULL) {
+ return NULL;
+ }
+ return adapted_frame_->Copy();
+ }
private:
+ rtc::CriticalSection crit_;
VideoAdapter* video_adapter_;
const VideoFrame* adapted_frame_;
rtc::scoped_ptr<VideoFrame> copied_output_frame_;
@@ -135,6 +163,13 @@
bool received_cpu_signal_;
};
+ void VerifyAdaptedResolution(const VideoCapturerListener::Stats& stats,
+ int width,
+ int height) {
+ EXPECT_EQ(width, stats.adapted_width);
+ EXPECT_EQ(height, stats.adapted_height);
+ }
+
rtc::scoped_ptr<FileVideoCapturer> capturer_;
rtc::scoped_ptr<VideoAdapter> adapter_;
rtc::scoped_ptr<VideoCapturerListener> listener_;
@@ -157,12 +192,13 @@
// Call Adapter with some frames.
EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 10, kWaitTimeout);
+ listener_->GetStats().captured_frames >= 10, kWaitTimeout);
// Verify no frame drop and no resolution change.
- EXPECT_GE(listener_->captured_frames(), 10);
- EXPECT_EQ(0, listener_->dropped_frames());
- VerifyAdaptedResolution(capture_format_.width, capture_format_.height);
+ VideoCapturerListener::Stats stats = listener_->GetStats();
+ EXPECT_GE(stats.captured_frames, 10);
+ EXPECT_EQ(0, stats.dropped_frames);
+ VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height);
}
// Do not adapt the frame rate or the resolution. Expect no frame drop and no
@@ -171,13 +207,14 @@
adapter_->SetOutputFormat(capture_format_);
EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 10, kWaitTimeout);
+ listener_->GetStats().captured_frames >= 10, kWaitTimeout);
// Verify no frame drop and no resolution change.
- EXPECT_GE(listener_->captured_frames(), 10);
- EXPECT_EQ(0, listener_->dropped_frames());
- VerifyAdaptedResolution(capture_format_.width, capture_format_.height);
- EXPECT_TRUE(listener_->last_adapt_was_no_op());
+ VideoCapturerListener::Stats stats = listener_->GetStats();
+ EXPECT_GE(stats.captured_frames, 10);
+ EXPECT_EQ(0, stats.dropped_frames);
+ VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height);
+ EXPECT_TRUE(stats.last_adapt_was_no_op);
}
TEST_F(VideoAdapterTest, AdaptZeroInterval) {
@@ -187,12 +224,13 @@
adapter_->SetOutputFormat(format);
EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 10, kWaitTimeout);
+ listener_->GetStats().captured_frames >= 10, kWaitTimeout);
// Verify no crash and that frames aren't dropped.
- EXPECT_GE(listener_->captured_frames(), 10);
- EXPECT_EQ(0, listener_->dropped_frames());
- VerifyAdaptedResolution(capture_format_.width, capture_format_.height);
+ VideoCapturerListener::Stats stats = listener_->GetStats();
+ EXPECT_GE(stats.captured_frames, 10);
+ EXPECT_EQ(0, stats.dropped_frames);
+ VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height);
}
// Adapt the frame rate to be half of the capture rate at the beginning. Expect
@@ -203,12 +241,13 @@
adapter_->SetOutputFormat(request_format);
EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 10, kWaitTimeout);
+ listener_->GetStats().captured_frames >= 10, kWaitTimeout);
// Verify frame drop and no resolution change.
- EXPECT_GE(listener_->captured_frames(), 10);
- EXPECT_EQ(listener_->captured_frames() / 2, listener_->dropped_frames());
- VerifyAdaptedResolution(capture_format_.width, capture_format_.height);
+ VideoCapturerListener::Stats stats = listener_->GetStats();
+ EXPECT_GE(stats.captured_frames, 10);
+ EXPECT_EQ(stats.captured_frames / 2, stats.dropped_frames);
+ VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height);
}
// Adapt the frame rate to be half of the capture rate at the beginning. Expect
@@ -219,13 +258,14 @@
adapter_->SetOutputFormat(request_format);
EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 30, kWaitTimeout);
+ listener_->GetStats().captured_frames >= 30, kWaitTimeout);
// Verify frame drop and no resolution change.
- EXPECT_GE(listener_->captured_frames(), 30);
+ VideoCapturerListener::Stats stats = listener_->GetStats();
+ EXPECT_GE(stats.captured_frames, 30);
// Verify 2 / 3 kept (20) and 1 / 3 dropped (10).
- EXPECT_EQ(listener_->captured_frames() * 1 / 3, listener_->dropped_frames());
- VerifyAdaptedResolution(capture_format_.width, capture_format_.height);
+ EXPECT_EQ(stats.captured_frames * 1 / 3, stats.dropped_frames);
+ VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height);
}
// Adapt the frame rate to be half of the capture rate after capturing no less
@@ -236,20 +276,20 @@
adapter_->SetOutputFormat(request_format);
EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 10, kWaitTimeout);
+ listener_->GetStats().captured_frames >= 10, kWaitTimeout);
// Verify no frame drop before adaptation.
- EXPECT_EQ(0, listener_->dropped_frames());
+ EXPECT_EQ(0, listener_->GetStats().dropped_frames);
// Adapat the frame rate.
request_format.interval *= 2;
adapter_->SetOutputFormat(request_format);
EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 20, kWaitTimeout);
+ listener_->GetStats().captured_frames >= 20, kWaitTimeout);
// Verify frame drop after adaptation.
- EXPECT_GT(listener_->dropped_frames(), 0);
+ EXPECT_GT(listener_->GetStats().dropped_frames, 0);
}
// Adapt the frame resolution to be a quarter of the capture resolution at the
@@ -261,11 +301,12 @@
adapter_->SetOutputFormat(request_format);
EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 10, kWaitTimeout);
+ listener_->GetStats().captured_frames >= 10, kWaitTimeout);
// Verify no frame drop and resolution change.
- EXPECT_EQ(0, listener_->dropped_frames());
- VerifyAdaptedResolution(request_format.width, request_format.height);
+ VideoCapturerListener::Stats stats = listener_->GetStats();
+ EXPECT_EQ(0, stats.dropped_frames);
+ VerifyAdaptedResolution(stats, request_format.width, request_format.height);
}
// Adapt the frame resolution to half width. Expect resolution change.
@@ -276,10 +317,10 @@
adapter_->SetOutputFormat(request_format);
EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 10, kWaitTimeout);
+ listener_->GetStats().captured_frames >= 10, kWaitTimeout);
// Verify resolution change.
- VerifyAdaptedResolution(213, 160);
+ VerifyAdaptedResolution(listener_->GetStats(), 213, 160);
}
// Adapt the frame resolution to half height. Expect resolution change.
@@ -290,10 +331,10 @@
adapter_->SetOutputFormat(request_format);
EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 10, kWaitTimeout);
+ listener_->GetStats().captured_frames >= 10, kWaitTimeout);
// Verify resolution change.
- VerifyAdaptedResolution(213, 160);
+ VerifyAdaptedResolution(listener_->GetStats(), 213, 160);
}
// Adapt the frame resolution to be a quarter of the capture resolution after
@@ -304,21 +345,25 @@
adapter_->SetOutputFormat(request_format);
EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 10, kWaitTimeout);
+ listener_->GetStats().captured_frames >= 10, kWaitTimeout);
// Verify no resolution change before adaptation.
- VerifyAdaptedResolution(request_format.width, request_format.height);
+ VerifyAdaptedResolution(
+ listener_->GetStats(), request_format.width, request_format.height);
// Adapt the frame resolution.
request_format.width /= 2;
request_format.height /= 2;
adapter_->SetOutputFormat(request_format);
- EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 20, kWaitTimeout);
-
+ int captured_frames = listener_->GetStats().captured_frames;
+ EXPECT_TRUE_WAIT(
+ !capturer_->IsRunning() ||
+ listener_->GetStats().captured_frames >= captured_frames + 10,
+ kWaitTimeout);
// Verify resolution change after adaptation.
- VerifyAdaptedResolution(request_format.width, request_format.height);
+ VerifyAdaptedResolution(
+ listener_->GetStats(), request_format.width, request_format.height);
}
// Black the output frame.
@@ -326,42 +371,57 @@
adapter_->SetOutputFormat(capture_format_);
EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 10, kWaitTimeout);
+ listener_->GetStats().captured_frames >= 10, kWaitTimeout);
// Verify that the output frame is not black.
- EXPECT_NE(16, *listener_->adapted_frame()->GetYPlane());
- EXPECT_NE(128, *listener_->adapted_frame()->GetUPlane());
- EXPECT_NE(128, *listener_->adapted_frame()->GetVPlane());
+ rtc::scoped_ptr<VideoFrame> adapted_frame(listener_->CopyAdaptedFrame());
+ EXPECT_NE(16, *adapted_frame->GetYPlane());
+ EXPECT_NE(128, *adapted_frame->GetUPlane());
+ EXPECT_NE(128, *adapted_frame->GetVPlane());
adapter_->SetBlackOutput(true);
- EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 20, kWaitTimeout);
+ int captured_frames = listener_->GetStats().captured_frames;
+ EXPECT_TRUE_WAIT(
+ !capturer_->IsRunning() ||
+ listener_->GetStats().captured_frames >= captured_frames + 10,
+ kWaitTimeout);
// Verify that the output frame is black.
- EXPECT_EQ(16, *listener_->adapted_frame()->GetYPlane());
- EXPECT_EQ(128, *listener_->adapted_frame()->GetUPlane());
- EXPECT_EQ(128, *listener_->adapted_frame()->GetVPlane());
+ adapted_frame.reset(listener_->CopyAdaptedFrame());
+ EXPECT_EQ(16, *adapted_frame->GetYPlane());
+ EXPECT_EQ(128, *adapted_frame->GetUPlane());
+ EXPECT_EQ(128, *adapted_frame->GetVPlane());
// Verify that the elapsed time and timestamp of the black frame increase.
- int64 elapsed_time = listener_->adapted_frame()->GetElapsedTime();
- int64 timestamp = listener_->adapted_frame()->GetTimeStamp();
- EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 22, kWaitTimeout);
- EXPECT_GT(listener_->adapted_frame()->GetElapsedTime(), elapsed_time);
- EXPECT_GT(listener_->adapted_frame()->GetTimeStamp(), timestamp);
+ int64 elapsed_time = adapted_frame->GetElapsedTime();
+ int64 timestamp = adapted_frame->GetTimeStamp();
+ captured_frames = listener_->GetStats().captured_frames;
+ EXPECT_TRUE_WAIT(
+ !capturer_->IsRunning() ||
+ listener_->GetStats().captured_frames >= captured_frames + 10,
+ kWaitTimeout);
+
+ adapted_frame.reset(listener_->CopyAdaptedFrame());
+ EXPECT_GT(adapted_frame->GetElapsedTime(), elapsed_time);
+ EXPECT_GT(adapted_frame->GetTimeStamp(), timestamp);
// Change the output size
VideoFormat request_format = capture_format_;
request_format.width /= 2;
request_format.height /= 2;
adapter_->SetOutputFormat(request_format);
+ captured_frames = listener_->GetStats().captured_frames;
+ EXPECT_TRUE_WAIT(
+ !capturer_->IsRunning() ||
+ listener_->GetStats().captured_frames >= captured_frames + 10,
+ kWaitTimeout);
- EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 40, kWaitTimeout);
// Verify resolution change after adaptation.
- VerifyAdaptedResolution(request_format.width, request_format.height);
+ VerifyAdaptedResolution(
+ listener_->GetStats(), request_format.width, request_format.height);
// Verify that the output frame is black.
- EXPECT_EQ(16, *listener_->adapted_frame()->GetYPlane());
- EXPECT_EQ(128, *listener_->adapted_frame()->GetUPlane());
- EXPECT_EQ(128, *listener_->adapted_frame()->GetVPlane());
+ adapted_frame.reset(listener_->CopyAdaptedFrame());
+ EXPECT_EQ(16, *adapted_frame->GetYPlane());
+ EXPECT_EQ(128, *adapted_frame->GetUPlane());
+ EXPECT_EQ(128, *adapted_frame->GetVPlane());
}
// Drop all frames.
@@ -370,11 +430,12 @@
adapter_->SetOutputFormat(format);
EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
EXPECT_TRUE_WAIT(!capturer_->IsRunning() ||
- listener_->captured_frames() >= 10, kWaitTimeout);
+ listener_->GetStats().captured_frames >= 10, kWaitTimeout);
// Verify all frames are dropped.
- EXPECT_GE(listener_->captured_frames(), 10);
- EXPECT_EQ(listener_->captured_frames(), listener_->dropped_frames());
+ VideoCapturerListener::Stats stats = listener_->GetStats();
+ EXPECT_GE(stats.captured_frames, 10);
+ EXPECT_EQ(stats.captured_frames, stats.dropped_frames);
}
TEST(CoordinatedVideoAdapterTest, TestCoordinatedWithoutCpuAdaptation) {
diff --git a/media/base/videoengine_unittest.h b/media/base/videoengine_unittest.h
index 0f03c7b..8eab347 100644
--- a/media/base/videoengine_unittest.h
+++ b/media/base/videoengine_unittest.h
@@ -1593,6 +1593,25 @@
frame_count += 2;
EXPECT_EQ_WAIT(frame_count, renderer_.num_rendered_frames(), kTimeout);
}
+ // Tests that adapted frames won't be upscaled to a higher resolution.
+ void SendsLowerResolutionOnSmallerFrames() {
+ cricket::VideoCodec codec = DefaultCodec();
+ codec.width = 320;
+ codec.height = 240;
+ EXPECT_TRUE(SetOneCodec(codec));
+ EXPECT_TRUE(SetSend(true));
+ EXPECT_TRUE(channel_->SetRender(true));
+ EXPECT_TRUE(channel_->SetRenderer(kDefaultReceiveSsrc, &renderer_));
+ EXPECT_EQ(0, renderer_.num_rendered_frames());
+ EXPECT_TRUE(SendFrame());
+ EXPECT_FRAME_WAIT(1, codec.width, codec.height, kTimeout);
+
+ // Check that we send smaller frames at the new resolution.
+ EXPECT_TRUE(rtc::Thread::Current()->ProcessMessages(33));
+ EXPECT_TRUE(video_capturer_->CaptureCustomFrame(
+ codec.width / 2, codec.height / 2, cricket::FOURCC_I420));
+ EXPECT_FRAME_WAIT(2, codec.width / 2, codec.height / 2, kTimeout);
+ }
// Tests that we can set the send stream format properly.
void SetSendStreamFormat() {
cricket::VideoCodec codec(DefaultCodec());
diff --git a/media/base/videoframe.cc b/media/base/videoframe.cc
index 1c5cfd8..018d065 100644
--- a/media/base/videoframe.cc
+++ b/media/base/videoframe.cc
@@ -235,7 +235,7 @@
}
static const size_t kMaxSampleSize = 1000000000u;
-// Returns whether a sample is valid
+// Returns whether a sample is valid.
bool VideoFrame::Validate(uint32 fourcc, int w, int h,
const uint8 *sample, size_t sample_size) {
if (h < 0) {
@@ -311,6 +311,11 @@
<< " " << sample_size;
return false;
}
+ // TODO(fbarchard): Make function to dump information about frames.
+ uint8 four_samples[4] = { 0, 0, 0, 0 };
+ for (size_t i = 0; i < ARRAY_SIZE(four_samples) && i < sample_size; ++i) {
+ four_samples[i] = sample[i];
+ }
if (sample_size < expected_size) {
LOG(LS_ERROR) << "Size field is too small."
<< " format: " << GetFourccName(format)
@@ -318,10 +323,10 @@
<< " size: " << w << "x" << h
<< " " << sample_size
<< " expected: " << expected_size
- << " sample[0..3]: " << static_cast<int>(sample[0])
- << ", " << static_cast<int>(sample[1])
- << ", " << static_cast<int>(sample[2])
- << ", " << static_cast<int>(sample[3]);
+ << " sample[0..3]: " << static_cast<int>(four_samples[0])
+ << ", " << static_cast<int>(four_samples[1])
+ << ", " << static_cast<int>(four_samples[2])
+ << ", " << static_cast<int>(four_samples[3]);
return false;
}
if (sample_size > kMaxSampleSize) {
@@ -331,13 +336,14 @@
<< " size: " << w << "x" << h
<< " " << sample_size
<< " expected: " << 2 * expected_size
- << " sample[0..3]: " << static_cast<int>(sample[0])
- << ", " << static_cast<int>(sample[1])
- << ", " << static_cast<int>(sample[2])
- << ", " << static_cast<int>(sample[3]);
+ << " sample[0..3]: " << static_cast<int>(four_samples[0])
+ << ", " << static_cast<int>(four_samples[1])
+ << ", " << static_cast<int>(four_samples[2])
+ << ", " << static_cast<int>(four_samples[3]);
return false;
}
// Show large size warning once every 100 frames.
+ // TODO(fbarchard): Make frame counter atomic for thread safety.
static int large_warn100 = 0;
size_t large_expected_size = expected_size * 2;
if (expected_bpp >= 8 &&
@@ -350,27 +356,14 @@
<< " size: " << w << "x" << h
<< " bytes: " << sample_size
<< " expected: " << large_expected_size
- << " sample[0..3]: " << static_cast<int>(sample[0])
- << ", " << static_cast<int>(sample[1])
- << ", " << static_cast<int>(sample[2])
- << ", " << static_cast<int>(sample[3]);
- }
- // Scan pages to ensure they are there and don't contain a single value and
- // to generate an error.
- if (!memcmp(sample + sample_size - 8, sample + sample_size - 4, 4) &&
- !memcmp(sample, sample + 4, sample_size - 4)) {
- LOG(LS_WARNING) << "Duplicate value for all pixels."
- << " format: " << GetFourccName(format)
- << " bpp: " << expected_bpp
- << " size: " << w << "x" << h
- << " bytes: " << sample_size
- << " expected: " << expected_size
- << " sample[0..3]: " << static_cast<int>(sample[0])
- << ", " << static_cast<int>(sample[1])
- << ", " << static_cast<int>(sample[2])
- << ", " << static_cast<int>(sample[3]);
+ << " sample[0..3]: " << static_cast<int>(four_samples[0])
+ << ", " << static_cast<int>(four_samples[1])
+ << ", " << static_cast<int>(four_samples[2])
+ << ", " << static_cast<int>(four_samples[3]);
}
+ // TODO(fbarchard): Add duplicate pixel check.
+ // TODO(fbarchard): Use frame counter atomic for thread safety.
static bool valid_once = true;
if (valid_once) {
valid_once = false;
@@ -380,10 +373,10 @@
<< " size: " << w << "x" << h
<< " bytes: " << sample_size
<< " expected: " << expected_size
- << " sample[0..3]: " << static_cast<int>(sample[0])
- << ", " << static_cast<int>(sample[1])
- << ", " << static_cast<int>(sample[2])
- << ", " << static_cast<int>(sample[3]);
+ << " sample[0..3]: " << static_cast<int>(four_samples[0])
+ << ", " << static_cast<int>(four_samples[1])
+ << ", " << static_cast<int>(four_samples[2])
+ << ", " << static_cast<int>(four_samples[3]);
}
return true;
}
diff --git a/media/base/videoframe_unittest.h b/media/base/videoframe_unittest.h
index c4a7a8c..483fc34 100644
--- a/media/base/videoframe_unittest.h
+++ b/media/base/videoframe_unittest.h
@@ -135,6 +135,9 @@
rtc::scoped_ptr<rtc::FileStream> fs(
rtc::Filesystem::OpenFile(path, "rb"));
if (!fs.get()) {
+ LOG(LS_ERROR) << "Could not open test file path: " << path.pathname()
+ << " from current dir "
+ << rtc::Filesystem::GetCurrentDirectory().pathname();
return NULL;
}
@@ -143,6 +146,7 @@
new rtc::MemoryStream());
rtc::StreamResult res = Flow(fs.get(), buf, sizeof(buf), ms.get());
if (res != rtc::SR_SUCCESS) {
+ LOG(LS_ERROR) << "Could not load test file path: " << path.pathname();
return NULL;
}
@@ -419,17 +423,22 @@
const uint8* u, uint32 upitch,
const uint8* v, uint32 vpitch,
int max_error) {
- return IsSize(frame, width, height) &&
+ return IsSize(frame,
+ static_cast<uint32>(width),
+ static_cast<uint32>(height)) &&
frame.GetPixelWidth() == pixel_width &&
frame.GetPixelHeight() == pixel_height &&
frame.GetElapsedTime() == elapsed_time &&
frame.GetTimeStamp() == time_stamp &&
IsPlaneEqual("y", frame.GetYPlane(), frame.GetYPitch(), y, ypitch,
- width, height, max_error) &&
+ static_cast<uint32>(width),
+ static_cast<uint32>(height), max_error) &&
IsPlaneEqual("u", frame.GetUPlane(), frame.GetUPitch(), u, upitch,
- (width + 1) / 2, (height + 1) / 2, max_error) &&
+ static_cast<uint32>((width + 1) / 2),
+ static_cast<uint32>((height + 1) / 2), max_error) &&
IsPlaneEqual("v", frame.GetVPlane(), frame.GetVPitch(), v, vpitch,
- (width + 1) / 2, (height + 1) / 2, max_error);
+ static_cast<uint32>((width + 1) / 2),
+ static_cast<uint32>((height + 1) / 2), max_error);
}
static bool IsEqual(const cricket::VideoFrame& frame1,
@@ -719,7 +728,7 @@
T frame1, frame2;
size_t out_size = kWidth * kHeight * 2;
rtc::scoped_ptr<uint8[]> outbuf(new uint8[out_size + kAlignment]);
- uint8 *out = ALIGNP(outbuf.get(), kAlignment);
+ uint8* out = ALIGNP(outbuf.get(), kAlignment);
T frame;
ASSERT_TRUE(LoadFrameNoRepeat(&frame1));
EXPECT_EQ(out_size, frame1.ConvertToRgbBuffer(cricket::FOURCC_RGBP,
@@ -735,7 +744,7 @@
T frame1, frame2;
size_t out_size = kWidth * kHeight * 2;
rtc::scoped_ptr<uint8[]> outbuf(new uint8[out_size + kAlignment]);
- uint8 *out = ALIGNP(outbuf.get(), kAlignment);
+ uint8* out = ALIGNP(outbuf.get(), kAlignment);
T frame;
ASSERT_TRUE(LoadFrameNoRepeat(&frame1));
EXPECT_EQ(out_size, frame1.ConvertToRgbBuffer(cricket::FOURCC_RGBO,
@@ -751,7 +760,7 @@
T frame1, frame2;
size_t out_size = kWidth * kHeight * 2;
rtc::scoped_ptr<uint8[]> outbuf(new uint8[out_size + kAlignment]);
- uint8 *out = ALIGNP(outbuf.get(), kAlignment);
+ uint8* out = ALIGNP(outbuf.get(), kAlignment);
T frame;
ASSERT_TRUE(LoadFrameNoRepeat(&frame1));
EXPECT_EQ(out_size, frame1.ConvertToRgbBuffer(cricket::FOURCC_R444,
@@ -771,12 +780,12 @@
size_t bayer_size = kWidth * kHeight; \
rtc::scoped_ptr<uint8[]> bayerbuf(new uint8[ \
bayer_size + kAlignment]); \
- uint8 *bayer = ALIGNP(bayerbuf.get(), kAlignment); \
+ uint8* bayer = ALIGNP(bayerbuf.get(), kAlignment); \
T frame1, frame2; \
rtc::scoped_ptr<rtc::MemoryStream> ms( \
CreateRgbSample(cricket::FOURCC_ARGB, kWidth, kHeight)); \
ASSERT_TRUE(ms.get() != NULL); \
- libyuv::ARGBToBayer##BAYER(reinterpret_cast<uint8 *>(ms->GetBuffer()), \
+ libyuv::ARGBToBayer##BAYER(reinterpret_cast<uint8* >(ms->GetBuffer()), \
kWidth * 4, \
bayer, kWidth, \
kWidth, kHeight); \
@@ -812,8 +821,8 @@
reinterpret_cast<uint8*>(ms->GetBuffer()), \
data_size, \
1, 1, 0, 0, 0)); \
- int width_rotate = frame1.GetWidth(); \
- int height_rotate = frame1.GetHeight(); \
+ int width_rotate = static_cast<int>(frame1.GetWidth()); \
+ int height_rotate = static_cast<int>(frame1.GetHeight()); \
EXPECT_TRUE(frame3.InitToBlack(width_rotate, height_rotate, 1, 1, 0, 0)); \
libyuv::I420Mirror(frame2.GetYPlane(), frame2.GetYPitch(), \
frame2.GetUPlane(), frame2.GetUPitch(), \
@@ -845,8 +854,8 @@
reinterpret_cast<uint8*>(ms->GetBuffer()), \
data_size, \
1, 1, 0, 0, 0)); \
- int width_rotate = frame1.GetWidth(); \
- int height_rotate = frame1.GetHeight(); \
+ int width_rotate = static_cast<int>(frame1.GetWidth()); \
+ int height_rotate = static_cast<int>(frame1.GetHeight()); \
EXPECT_TRUE(frame3.InitToBlack(width_rotate, height_rotate, 1, 1, 0, 0)); \
libyuv::I420Rotate(frame2.GetYPlane(), frame2.GetYPitch(), \
frame2.GetUPlane(), frame2.GetUPitch(), \
@@ -995,7 +1004,7 @@
// Convert back to ARGB.
size_t out_size = 4;
rtc::scoped_ptr<uint8[]> outbuf(new uint8[out_size + kAlignment]);
- uint8 *out = ALIGNP(outbuf.get(), kAlignment);
+ uint8* out = ALIGNP(outbuf.get(), kAlignment);
EXPECT_EQ(out_size, frame.ConvertToRgbBuffer(cricket::FOURCC_ARGB,
out,
@@ -1032,7 +1041,7 @@
// Convert back to ARGB
size_t out_size = 10 * 4;
rtc::scoped_ptr<uint8[]> outbuf(new uint8[out_size + kAlignment]);
- uint8 *out = ALIGNP(outbuf.get(), kAlignment);
+ uint8* out = ALIGNP(outbuf.get(), kAlignment);
EXPECT_EQ(out_size, frame.ConvertToRgbBuffer(cricket::FOURCC_ARGB,
out,
@@ -1431,8 +1440,8 @@
size_t out_size = astride * kHeight;
rtc::scoped_ptr<uint8[]> outbuf(new uint8[out_size + kAlignment + 1]);
memset(outbuf.get(), 0, out_size + kAlignment + 1);
- uint8 *outtop = ALIGNP(outbuf.get(), kAlignment);
- uint8 *out = outtop;
+ uint8* outtop = ALIGNP(outbuf.get(), kAlignment);
+ uint8* out = outtop;
int stride = astride;
if (invert) {
out += (kHeight - 1) * stride; // Point to last row.
@@ -1869,7 +1878,7 @@
size_t bayer_size = kWidth * kHeight; \
rtc::scoped_ptr<uint8[]> bayerbuf(new uint8[ \
bayer_size + kAlignment]); \
- uint8 *bayer = ALIGNP(bayerbuf.get(), kAlignment); \
+ uint8* bayer = ALIGNP(bayerbuf.get(), kAlignment); \
T frame; \
rtc::scoped_ptr<rtc::MemoryStream> ms( \
CreateRgbSample(cricket::FOURCC_ARGB, kWidth, kHeight)); \
@@ -1898,7 +1907,7 @@
size_t bayer_size = kWidth * kHeight; \
rtc::scoped_ptr<uint8[]> bayerbuf(new uint8[ \
bayer_size + 1 + kAlignment]); \
- uint8 *bayer = ALIGNP(bayerbuf.get(), kAlignment) + 1; \
+ uint8* bayer = ALIGNP(bayerbuf.get(), kAlignment) + 1; \
T frame; \
rtc::scoped_ptr<rtc::MemoryStream> ms( \
CreateRgbSample(cricket::FOURCC_ARGB, kWidth, kHeight)); \
@@ -1935,7 +1944,7 @@
size_t bayer_size = kWidth * kHeight; \
rtc::scoped_ptr<uint8[]> bayerbuf(new uint8[ \
bayer_size + kAlignment]); \
- uint8 *bayer1 = ALIGNP(bayerbuf.get(), kAlignment); \
+ uint8* bayer1 = ALIGNP(bayerbuf.get(), kAlignment); \
for (int i = 0; i < kWidth * kHeight; ++i) { \
bayer1[i] = static_cast<uint8>(i * 33u + 183u); \
} \
@@ -1951,7 +1960,7 @@
} \
rtc::scoped_ptr<uint8[]> bayer2buf(new uint8[ \
bayer_size + kAlignment]); \
- uint8 *bayer2 = ALIGNP(bayer2buf.get(), kAlignment); \
+ uint8* bayer2 = ALIGNP(bayer2buf.get(), kAlignment); \
libyuv::ARGBToBayer##BAYER(reinterpret_cast<uint8*>(ms->GetBuffer()), \
kWidth * 4, \
bayer2, kWidth, \
diff --git a/media/devices/linuxdevicemanager.cc b/media/devices/linuxdevicemanager.cc
index a79e226..d122169 100644
--- a/media/devices/linuxdevicemanager.cc
+++ b/media/devices/linuxdevicemanager.cc
@@ -31,10 +31,10 @@
#include "talk/media/base/mediacommon.h"
#include "talk/media/devices/libudevsymboltable.h"
#include "talk/media/devices/v4llookup.h"
-#include "talk/sound/platformsoundsystem.h"
-#include "talk/sound/platformsoundsystemfactory.h"
-#include "talk/sound/sounddevicelocator.h"
-#include "talk/sound/soundsysteminterface.h"
+#include "webrtc/sound/platformsoundsystem.h"
+#include "webrtc/sound/platformsoundsystemfactory.h"
+#include "webrtc/sound/sounddevicelocator.h"
+#include "webrtc/sound/soundsysteminterface.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/linux.h"
#include "webrtc/base/logging.h"
@@ -89,7 +89,7 @@
};
LinuxDeviceManager::LinuxDeviceManager()
- : sound_system_(new PlatformSoundSystemFactory()) {
+ : sound_system_(new rtc::PlatformSoundSystemFactory()) {
set_watcher(new LinuxDeviceWatcher(this));
}
@@ -102,7 +102,7 @@
if (!sound_system_.get()) {
return false;
}
- SoundSystemInterface::SoundDeviceLocatorList list;
+ rtc::SoundSystemInterface::SoundDeviceLocatorList list;
bool success;
if (input) {
success = sound_system_->EnumerateCaptureDevices(&list);
@@ -118,12 +118,12 @@
// device at index 0, but Enumerate(Capture|Playback)Devices does not include
// a locator for the default device.
int index = 1;
- for (SoundSystemInterface::SoundDeviceLocatorList::iterator i = list.begin();
+ for (rtc::SoundSystemInterface::SoundDeviceLocatorList::iterator i = list.begin();
i != list.end();
++i, ++index) {
devs->push_back(Device((*i)->name(), index));
}
- SoundSystemInterface::ClearSoundDeviceLocatorList(&list);
+ rtc::SoundSystemInterface::ClearSoundDeviceLocatorList(&list);
sound_system_.release();
return FilterDevices(devs, kFilteredAudioDevicesName);
}
diff --git a/media/devices/linuxdevicemanager.h b/media/devices/linuxdevicemanager.h
index 88aee4e..1eb648f 100644
--- a/media/devices/linuxdevicemanager.h
+++ b/media/devices/linuxdevicemanager.h
@@ -32,7 +32,7 @@
#include <vector>
#include "talk/media/devices/devicemanager.h"
-#include "talk/sound/soundsystemfactory.h"
+#include "webrtc/sound/soundsystemfactory.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/stringencode.h"
@@ -47,7 +47,7 @@
private:
virtual bool GetAudioDevices(bool input, std::vector<Device>* devs);
- SoundSystemHandle sound_system_;
+ rtc::SoundSystemHandle sound_system_;
};
} // namespace cricket
diff --git a/media/devices/macdevicemanager.cc b/media/devices/macdevicemanager.cc
index 568ee53..8f777b4 100644
--- a/media/devices/macdevicemanager.cc
+++ b/media/devices/macdevicemanager.cc
@@ -71,7 +71,7 @@
extern DeviceWatcherImpl* CreateDeviceWatcherCallback(
DeviceManagerInterface* dm);
extern void ReleaseDeviceWatcherCallback(DeviceWatcherImpl* impl);
-extern bool GetQTKitVideoDevices(std::vector<Device>* out);
+extern bool GetAVFoundationVideoDevices(std::vector<Device>* out);
static bool GetAudioDeviceIDs(bool inputs, std::vector<AudioDeviceID>* out);
static bool GetAudioDeviceName(AudioDeviceID id, bool input, std::string* out);
@@ -84,7 +84,7 @@
bool MacDeviceManager::GetVideoCaptureDevices(std::vector<Device>* devices) {
devices->clear();
- if (!GetQTKitVideoDevices(devices)) {
+ if (!GetAVFoundationVideoDevices(devices)) {
return false;
}
return FilterDevices(devices, kFilteredVideoDevicesName);
diff --git a/media/devices/macdevicemanagermm.mm b/media/devices/macdevicemanagermm.mm
index 3091ec4..cfcf5a4 100644
--- a/media/devices/macdevicemanagermm.mm
+++ b/media/devices/macdevicemanagermm.mm
@@ -33,6 +33,11 @@
#include "talk/media/devices/devicemanager.h"
#import <assert.h>
+#ifdef __MAC_OS_X_VERSION_MAX_ALLOWED
+#if __MAC_OS_X_VERSION_MAX_ALLOWED >= 1070
+ #import <AVFoundation/AVFoundation.h>
+#endif
+#endif
#import <QTKit/QTKit.h>
#include "webrtc/base/logging.h"
@@ -136,4 +141,52 @@
return true;
}
+bool GetAVFoundationVideoDevices(std::vector<Device>* devices) {
+#ifdef __MAC_OS_X_VERSION_MAX_ALLOWED
+#if __MAC_OS_X_VERSION_MAX_ALLOWED >=1070
+ if (![AVCaptureDevice class]) {
+ // Fallback to using QTKit if AVFoundation is not available
+ return GetQTKitVideoDevices(devices);
+ }
+#if !__has_feature(objc_arc)
+ NSAutoreleasePool* pool = [[NSAutoreleasePool alloc] init];
+#else
+ @autoreleasepool
+#endif
+ {
+ NSArray* capture_devices = [AVCaptureDevice devices];
+ LOG(LS_INFO) << [capture_devices count] << " capture device(s) found:";
+ for (AVCaptureDevice* capture_device in capture_devices) {
+ if ([capture_device hasMediaType:AVMediaTypeVideo] ||
+ [capture_device hasMediaType:AVMediaTypeMuxed]) {
+ static NSString* const kFormat = @"localizedName: \"%@\", "
+ @"modelID: \"%@\", uniqueID \"%@\", isConnected: %d, "
+ @"isInUseByAnotherApplication: %d";
+ NSString* info = [NSString
+ stringWithFormat:kFormat,
+ [capture_device localizedName],
+ [capture_device modelID],
+ [capture_device uniqueID],
+ [capture_device isConnected],
+ [capture_device isInUseByAnotherApplication]];
+ LOG(LS_INFO) << [info UTF8String];
+
+ std::string name([[capture_device localizedName] UTF8String]);
+ devices->push_back(
+ Device(name, [[capture_device uniqueID] UTF8String]));
+ }
+ }
+ }
+#if !__has_feature(objc_arc)
+ [pool drain];
+#endif
+ return true;
+#else // __MAC_OS_X_VERSION_MAX_ALLOWED >=1070
+ return GetQTKitVideoDevices(devices);
+#endif // __MAC_OS_X_VERSION_MAX_ALLOWED >=1070
+#else // __MAC_OS_X_VERSION_MAX_ALLOWED
+ return GetQTKitVideoDevices(devices);
+#endif // __MAC_OS_X_VERSION_MAX_ALLOWED
+}
+
} // namespace cricket
diff --git a/media/other/linphonemediaengine.h b/media/other/linphonemediaengine.h
index e4ba345..b4950de 100644
--- a/media/other/linphonemediaengine.h
+++ b/media/other/linphonemediaengine.h
@@ -70,7 +70,6 @@
virtual VideoMediaChannel* CreateVideoChannel(VoiceMediaChannel* voice_ch);
virtual SoundclipMedia* CreateSoundclip() { return NULL; }
virtual bool SetAudioOptions(int options) { return true; }
- virtual bool SetVideoOptions(int options) { return true; }
virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config) {
return true;
}
@@ -81,7 +80,6 @@
virtual bool SetOutputVolume(int level) { return true; }
virtual int GetInputLevel() { return 0; }
virtual bool SetLocalMonitor(bool enable) { return true; }
- virtual bool SetLocalRenderer(VideoRenderer* renderer) { return true; }
// TODO: control channel send?
virtual bool SetVideoCapture(bool capture) { return true; }
virtual const std::vector<AudioCodec>& audio_codecs() {
diff --git a/media/sctp/sctpdataengine_unittest.cc b/media/sctp/sctpdataengine_unittest.cc
index 0fb9b9d..fb00650 100644
--- a/media/sctp/sctpdataengine_unittest.cc
+++ b/media/sctp/sctpdataengine_unittest.cc
@@ -45,6 +45,11 @@
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/thread.h"
+#ifdef HAVE_NSS_SSL_H
+// TODO(thorcarpenter): Remove after webrtc switches over to BoringSSL.
+#include "webrtc/base/nssstreamadapter.h"
+#endif // HAVE_NSS_SSL_H
+
enum {
MSG_PACKET = 1,
};
@@ -69,8 +74,7 @@
// TODO(ldixon): Can/should we use Buffer.TransferTo here?
// Note: this assignment does a deep copy of data from packet.
- rtc::Buffer* buffer = new rtc::Buffer(packet->data(),
- packet->length());
+ rtc::Buffer* buffer = new rtc::Buffer(packet->data(), packet->length());
thread_->Post(this, MSG_PACKET, rtc::WrapMessageData(buffer));
LOG(LS_VERBOSE) << "SctpFakeNetworkInterface::SendPacket, Posted message.";
return true;
@@ -219,6 +223,12 @@
// usrsctp uses the NSS random number generator on non-Android platforms,
// so we need to initialize SSL.
static void SetUpTestCase() {
+#ifdef HAVE_NSS_SSL_H
+ // TODO(thorcarpenter): Remove after webrtc switches over to BoringSSL.
+ if (!rtc::NSSContext::InitializeSSL(NULL)) {
+ LOG(LS_WARNING) << "Unabled to initialize NSS.";
+ }
+#endif // HAVE_NSS_SSL_H
rtc::InitializeSSL();
}
@@ -266,6 +276,10 @@
virtual void TearDown() {
channel1()->SetSend(false);
channel2()->SetSend(false);
+
+ // Process messages until idle to prevent a sent packet from being dropped
+ // and causing memory leaks (not being deleted by the receiver).
+ ProcessMessagesUntilIdle();
}
void AddStream(int ssrc) {
diff --git a/media/testdata/faces.1280x720_P420.yuv b/media/testdata/faces.1280x720_P420.yuv
new file mode 100644
index 0000000..f4844ee
--- /dev/null
+++ b/media/testdata/faces.1280x720_P420.yuv
Binary files differ
diff --git a/media/testdata/faces_I400.jpg b/media/testdata/faces_I400.jpg
new file mode 100644
index 0000000..c928f00
--- /dev/null
+++ b/media/testdata/faces_I400.jpg
Binary files differ
diff --git a/media/testdata/faces_I411.jpg b/media/testdata/faces_I411.jpg
new file mode 100644
index 0000000..600c122
--- /dev/null
+++ b/media/testdata/faces_I411.jpg
Binary files differ
diff --git a/media/testdata/faces_I420.jpg b/media/testdata/faces_I420.jpg
new file mode 100644
index 0000000..10c5332
--- /dev/null
+++ b/media/testdata/faces_I420.jpg
Binary files differ
diff --git a/media/testdata/faces_I422.jpg b/media/testdata/faces_I422.jpg
new file mode 100644
index 0000000..9907aa1
--- /dev/null
+++ b/media/testdata/faces_I422.jpg
Binary files differ
diff --git a/media/testdata/faces_I444.jpg b/media/testdata/faces_I444.jpg
new file mode 100644
index 0000000..3422755
--- /dev/null
+++ b/media/testdata/faces_I444.jpg
Binary files differ
diff --git a/media/webrtc/OWNERS b/media/webrtc/OWNERS
index 9a3546e..aef5939 100644
--- a/media/webrtc/OWNERS
+++ b/media/webrtc/OWNERS
@@ -1,3 +1,3 @@
mflodman@webrtc.org
+pbos@webrtc.org
pthatcher@webrtc.org
-wu@webrtc.org
diff --git a/media/webrtc/fakewebrtcvideocapturemodule.h b/media/webrtc/fakewebrtcvideocapturemodule.h
index 347e4b7..82b5cdd 100644
--- a/media/webrtc/fakewebrtcvideocapturemodule.h
+++ b/media/webrtc/fakewebrtcvideocapturemodule.h
@@ -44,76 +44,72 @@
running_(false),
delay_(0) {
}
- virtual int32_t Version(char* version,
- uint32_t& remaining_buffer_in_bytes,
- uint32_t& position) const {
+ virtual int32_t TimeUntilNextProcess() OVERRIDE {
return 0;
}
- virtual int32_t TimeUntilNextProcess() {
+ virtual int32_t Process() OVERRIDE {
return 0;
}
- virtual int32_t Process() {
- return 0;
- }
- virtual int32_t ChangeUniqueId(const int32_t id) {
+ virtual int32_t ChangeUniqueId(const int32_t id) OVERRIDE {
id_ = id;
return 0;
}
virtual void RegisterCaptureDataCallback(
- webrtc::VideoCaptureDataCallback& callback) {
+ webrtc::VideoCaptureDataCallback& callback) OVERRIDE {
callback_ = &callback;
}
- virtual void DeRegisterCaptureDataCallback() { callback_ = NULL; }
- virtual void RegisterCaptureCallback(webrtc::VideoCaptureFeedBack& callback) {
+ virtual void DeRegisterCaptureDataCallback() OVERRIDE { callback_ = NULL; }
+ virtual void RegisterCaptureCallback(
+ webrtc::VideoCaptureFeedBack& callback) OVERRIDE {
// Not implemented.
}
- virtual void DeRegisterCaptureCallback() {
+ virtual void DeRegisterCaptureCallback() OVERRIDE {
// Not implemented.
}
- virtual void SetCaptureDelay(int32_t delay) { delay_ = delay; }
- virtual int32_t CaptureDelay() { return delay_; }
- virtual void EnableFrameRateCallback(const bool enable) {
+ virtual void SetCaptureDelay(int32_t delay) OVERRIDE { delay_ = delay; }
+ virtual int32_t CaptureDelay() OVERRIDE { return delay_; }
+ virtual void EnableFrameRateCallback(const bool enable) OVERRIDE {
// not implemented
}
- virtual void EnableNoPictureAlarm(const bool enable) {
+ virtual void EnableNoPictureAlarm(const bool enable) OVERRIDE {
// not implemented
}
virtual int32_t StartCapture(
- const webrtc::VideoCaptureCapability& cap) {
+ const webrtc::VideoCaptureCapability& cap) OVERRIDE {
if (running_) return -1;
cap_ = cap;
running_ = true;
return 0;
}
- virtual int32_t StopCapture() {
+ virtual int32_t StopCapture() OVERRIDE {
running_ = false;
return 0;
}
- virtual const char* CurrentDeviceName() const {
+ virtual const char* CurrentDeviceName() const OVERRIDE {
return NULL; // not implemented
}
- virtual bool CaptureStarted() {
+ virtual bool CaptureStarted() OVERRIDE {
return running_;
}
virtual int32_t CaptureSettings(
- webrtc::VideoCaptureCapability& settings) {
+ webrtc::VideoCaptureCapability& settings) OVERRIDE {
if (!running_) return -1;
settings = cap_;
return 0;
}
virtual int32_t SetCaptureRotation(
- webrtc::VideoCaptureRotation rotation) {
+ webrtc::VideoCaptureRotation rotation) OVERRIDE {
return -1; // not implemented
}
virtual VideoCaptureEncodeInterface* GetEncodeInterface(
- const webrtc::VideoCodec& codec) {
+ const webrtc::VideoCodec& codec) OVERRIDE {
return NULL; // not implemented
}
- virtual int32_t AddRef() {
+ virtual int32_t AddRef() OVERRIDE {
return 0;
}
- virtual int32_t Release() {
+ virtual int32_t Release() OVERRIDE {
delete this;
return 0;
}
diff --git a/media/webrtc/fakewebrtcvideoengine.h b/media/webrtc/fakewebrtcvideoengine.h
index 5cba380..ec4b9c6 100644
--- a/media/webrtc/fakewebrtcvideoengine.h
+++ b/media/webrtc/fakewebrtcvideoengine.h
@@ -335,6 +335,7 @@
unsigned int send_nack_bitrate_;
unsigned int send_bandwidth_;
unsigned int receive_bandwidth_;
+ // Bandwidth to deduct from estimated uplink capacity.
unsigned int reserved_transmit_bitrate_bps_;
bool suspend_below_min_bitrate_;
webrtc::CpuOveruseObserver* overuse_observer_;
diff --git a/media/webrtc/fakewebrtcvoiceengine.h b/media/webrtc/fakewebrtcvoiceengine.h
index f731b8d..52a50ff 100644
--- a/media/webrtc/fakewebrtcvoiceengine.h
+++ b/media/webrtc/fakewebrtcvoiceengine.h
@@ -43,10 +43,7 @@
#ifdef USE_WEBRTC_DEV_BRANCH
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#endif
-
-namespace webrtc {
-class ViENetwork;
-}
+#include "webrtc/video_engine/include/vie_network.h"
namespace cricket {
@@ -64,6 +61,12 @@
static const int kFakeDeviceId = 1;
#endif
+static const int kOpusBandwidthNb = 4000;
+static const int kOpusBandwidthMb = 6000;
+static const int kOpusBandwidthWb = 8000;
+static const int kOpusBandwidthSwb = 12000;
+static const int kOpusBandwidthFb = 20000;
+
// Verify the header extension ID, if enabled, is within the bounds specified in
// [RFC5285]: 1-14 inclusive.
#define WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id) \
@@ -183,6 +186,7 @@
file(false),
vad(false),
codec_fec(false),
+ max_encoding_bandwidth(0),
red(false),
nack(false),
media_processor_registered(false),
@@ -212,6 +216,7 @@
bool file;
bool vad;
bool codec_fec;
+ int max_encoding_bandwidth;
bool red;
bool nack;
bool media_processor_registered;
@@ -308,6 +313,9 @@
bool GetCodecFEC(int channel) {
return channels_[channel]->codec_fec;
}
+ int GetMaxEncodingBandwidth(int channel) {
+ return channels_[channel]->max_encoding_bandwidth;
+ }
bool GetNACK(int channel) {
return channels_[channel]->nack;
}
@@ -316,6 +324,8 @@
}
webrtc::ViENetwork* GetViENetwork(int channel) {
WEBRTC_ASSERT_CHANNEL(channel);
+ // WARNING: This pointer is for verification purposes only. Calling
+ // functions on it may result in undefined behavior!
return channels_[channel]->vie_network;
}
int GetVideoChannel(int channel) {
@@ -488,8 +498,6 @@
WEBRTC_STUB(LastError, ());
WEBRTC_STUB(SetOnHoldStatus, (int, bool, webrtc::OnHoldModes));
WEBRTC_STUB(GetOnHoldStatus, (int, bool&, webrtc::OnHoldModes&));
- WEBRTC_STUB(SetNetEQPlayoutMode, (int, webrtc::NetEqModes));
- WEBRTC_STUB(GetNetEQPlayoutMode, (int, webrtc::NetEqModes&));
// webrtc::VoECodec
WEBRTC_FUNC(NumOfCodecs, ()) {
@@ -625,10 +633,11 @@
}
WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled,
webrtc::VadModes& mode, bool& disabledDTX));
+
#ifdef USE_WEBRTC_DEV_BRANCH
WEBRTC_FUNC(SetFECStatus, (int channel, bool enable)) {
WEBRTC_CHECK_CHANNEL(channel);
- if (strcmp(channels_[channel]->send_codec.plname, "opus")) {
+ if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
// Return -1 if current send codec is not Opus.
// TODO(minyue): Excludes other codecs if they support inband FEC.
return -1;
@@ -641,6 +650,25 @@
enable = channels_[channel]->codec_fec;
return 0;
}
+
+ WEBRTC_FUNC(SetOpusMaxPlaybackRate, (int channel, int frequency_hz)) {
+ WEBRTC_CHECK_CHANNEL(channel);
+ if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
+ // Return -1 if current send codec is not Opus.
+ return -1;
+ }
+ if (frequency_hz <= 8000)
+ channels_[channel]->max_encoding_bandwidth = kOpusBandwidthNb;
+ else if (frequency_hz <= 12000)
+ channels_[channel]->max_encoding_bandwidth = kOpusBandwidthMb;
+ else if (frequency_hz <= 16000)
+ channels_[channel]->max_encoding_bandwidth = kOpusBandwidthWb;
+ else if (frequency_hz <= 24000)
+ channels_[channel]->max_encoding_bandwidth = kOpusBandwidthSwb;
+ else
+ channels_[channel]->max_encoding_bandwidth = kOpusBandwidthFb;
+ return 0;
+ }
#endif // USE_WEBRTC_DEV_BRANCH
// webrtc::VoEDtmf
@@ -999,6 +1027,11 @@
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->vie_network = vie_network;
channels_[channel]->video_channel = video_channel;
+ if (vie_network) {
+ // The interface is released here to avoid leaks. A test should not
+ // attempt to call functions on the interface stored in the channel.
+ vie_network->Release();
+ }
return 0;
}
diff --git a/media/webrtc/webrtcmediaengine.cc b/media/webrtc/webrtcmediaengine.cc
index 252b4e6..cf0fcdf 100644
--- a/media/webrtc/webrtcmediaengine.cc
+++ b/media/webrtc/webrtcmediaengine.cc
@@ -50,7 +50,6 @@
WebRtcVideoDecoderFactory* decoder_factory) {
voice_.SetAudioDeviceModule(adm, adm_sc);
video_.SetVoiceEngine(&voice_);
- video_.EnableTimedRender();
video_.SetExternalEncoderFactory(encoder_factory);
video_.SetExternalDecoderFactory(decoder_factory);
}
@@ -65,8 +64,9 @@
WebRtcVideoEncoderFactory* encoder_factory,
WebRtcVideoDecoderFactory* decoder_factory) {
voice_.SetAudioDeviceModule(adm, adm_sc);
+ video_.SetExternalDecoderFactory(decoder_factory);
+ video_.SetExternalEncoderFactory(encoder_factory);
video_.SetVoiceEngine(&voice_);
- video_.EnableTimedRender();
}
};
#endif // WEBRTC_CHROMIUM_BUILD
@@ -85,7 +85,6 @@
adm, adm_sc, encoder_factory, decoder_factory);
}
#endif // WEBRTC_CHROMIUM_BUILD
- // This is just to get a diff to run pulse.
return new cricket::WebRtcMediaEngine(
adm, adm_sc, encoder_factory, decoder_factory);
}
diff --git a/media/webrtc/webrtcmediaengine.h b/media/webrtc/webrtcmediaengine.h
index b906f5d..df517ee 100644
--- a/media/webrtc/webrtcmediaengine.h
+++ b/media/webrtc/webrtcmediaengine.h
@@ -122,9 +122,6 @@
virtual bool SetAudioOptions(const AudioOptions& options) OVERRIDE {
return delegate_->SetAudioOptions(options);
}
- virtual bool SetVideoOptions(const VideoOptions& options) OVERRIDE {
- return delegate_->SetVideoOptions(options);
- }
virtual bool SetAudioDelayOffset(int offset) OVERRIDE {
return delegate_->SetAudioDelayOffset(offset);
}
@@ -151,9 +148,6 @@
virtual bool SetLocalMonitor(bool enable) OVERRIDE {
return delegate_->SetLocalMonitor(enable);
}
- virtual bool SetLocalRenderer(VideoRenderer* renderer) OVERRIDE {
- return delegate_->SetLocalRenderer(renderer);
- }
virtual const std::vector<AudioCodec>& audio_codecs() OVERRIDE {
return delegate_->audio_codecs();
}
diff --git a/media/webrtc/webrtcpassthroughrender.h b/media/webrtc/webrtcpassthroughrender.h
index a432776..8d8c488 100644
--- a/media/webrtc/webrtcpassthroughrender.h
+++ b/media/webrtc/webrtcpassthroughrender.h
@@ -41,26 +41,20 @@
WebRtcPassthroughRender();
virtual ~WebRtcPassthroughRender();
- virtual int32_t Version(int8_t* version,
- uint32_t& remainingBufferInBytes,
- uint32_t& position) const {
+ virtual int32_t ChangeUniqueId(const int32_t id) OVERRIDE {
return 0;
}
- virtual int32_t ChangeUniqueId(const int32_t id) {
- return 0;
- }
+ virtual int32_t TimeUntilNextProcess() OVERRIDE { return 0; }
- virtual int32_t TimeUntilNextProcess() { return 0; }
+ virtual int32_t Process() OVERRIDE { return 0; }
- virtual int32_t Process() { return 0; }
-
- virtual void* Window() {
+ virtual void* Window() OVERRIDE {
rtc::CritScope cs(&render_critical_);
return window_;
}
- virtual int32_t ChangeWindow(void* window) {
+ virtual int32_t ChangeWindow(void* window) OVERRIDE {
rtc::CritScope cs(&render_critical_);
window_ = window;
return 0;
@@ -70,64 +64,60 @@
const uint32_t stream_id,
const uint32_t zOrder,
const float left, const float top,
- const float right, const float bottom);
+ const float right, const float bottom) OVERRIDE;
- virtual int32_t DeleteIncomingRenderStream(const uint32_t stream_id);
+ virtual int32_t DeleteIncomingRenderStream(const uint32_t stream_id) OVERRIDE;
virtual int32_t AddExternalRenderCallback(
const uint32_t stream_id,
- webrtc::VideoRenderCallback* render_object);
+ webrtc::VideoRenderCallback* render_object) OVERRIDE;
virtual int32_t GetIncomingRenderStreamProperties(
const uint32_t stream_id,
uint32_t& zOrder,
float& left, float& top,
- float& right, float& bottom) const {
+ float& right, float& bottom) const OVERRIDE {
return -1;
}
- virtual uint32_t GetIncomingFrameRate(
- const uint32_t stream_id) {
+ virtual uint32_t GetIncomingFrameRate(const uint32_t stream_id) OVERRIDE {
return 0;
}
- virtual uint32_t GetNumIncomingRenderStreams() const {
+ virtual uint32_t GetNumIncomingRenderStreams() const OVERRIDE {
return static_cast<uint32_t>(stream_render_map_.size());
}
- virtual bool HasIncomingRenderStream(const uint32_t stream_id) const;
+ virtual bool HasIncomingRenderStream(const uint32_t stream_id) const OVERRIDE;
virtual int32_t RegisterRawFrameCallback(
const uint32_t stream_id,
- webrtc::VideoRenderCallback* callback_obj) {
+ webrtc::VideoRenderCallback* callback_obj) OVERRIDE {
return -1;
}
virtual int32_t GetLastRenderedFrame(
const uint32_t stream_id,
- webrtc::I420VideoFrame &frame) const {
+ webrtc::I420VideoFrame &frame) const OVERRIDE {
return -1;
}
- virtual int32_t StartRender(
- const uint32_t stream_id);
+ virtual int32_t StartRender(const uint32_t stream_id) OVERRIDE;
- virtual int32_t StopRender(
- const uint32_t stream_id);
+ virtual int32_t StopRender(const uint32_t stream_id) OVERRIDE;
- virtual int32_t ResetRender() { return 0; }
+ virtual int32_t ResetRender() OVERRIDE { return 0; }
- virtual webrtc::RawVideoType PreferredVideoType() const;
+ virtual webrtc::RawVideoType PreferredVideoType() const OVERRIDE;
- virtual bool IsFullScreen() { return false; }
+ virtual bool IsFullScreen() OVERRIDE { return false; }
virtual int32_t GetScreenResolution(uint32_t& screenWidth,
- uint32_t& screenHeight) const {
+ uint32_t& screenHeight) const OVERRIDE {
return -1;
}
- virtual uint32_t RenderFrameRate(
- const uint32_t stream_id) {
+ virtual uint32_t RenderFrameRate(const uint32_t stream_id) OVERRIDE {
return 0;
}
@@ -135,11 +125,12 @@
const uint32_t stream_id,
const float left, const float top,
const float right,
- const float bottom) {
+ const float bottom) OVERRIDE {
return -1;
}
- virtual int32_t SetExpectedRenderDelay(uint32_t stream_id, int32_t delay_ms) {
+ virtual int32_t SetExpectedRenderDelay(uint32_t stream_id,
+ int32_t delay_ms) OVERRIDE {
return -1;
}
@@ -148,22 +139,22 @@
const unsigned int zOrder,
const float left, const float top,
const float right,
- const float bottom) {
+ const float bottom) OVERRIDE {
return -1;
}
- virtual int32_t SetTransparentBackground(const bool enable) {
+ virtual int32_t SetTransparentBackground(const bool enable) OVERRIDE {
return -1;
}
- virtual int32_t FullScreenRender(void* window, const bool enable) {
+ virtual int32_t FullScreenRender(void* window, const bool enable) OVERRIDE {
return -1;
}
virtual int32_t SetBitmap(const void* bitMap,
const uint8_t pictureId, const void* colorKey,
const float left, const float top,
- const float right, const float bottom) {
+ const float right, const float bottom) OVERRIDE {
return -1;
}
@@ -173,27 +164,27 @@
const uint32_t textColorRef,
const uint32_t backgroundColorRef,
const float left, const float top,
- const float right, const float bottom) {
+ const float right, const float bottom) OVERRIDE {
return -1;
}
virtual int32_t SetStartImage(
const uint32_t stream_id,
- const webrtc::I420VideoFrame& videoFrame) {
+ const webrtc::I420VideoFrame& videoFrame) OVERRIDE {
return -1;
}
virtual int32_t SetTimeoutImage(
const uint32_t stream_id,
const webrtc::I420VideoFrame& videoFrame,
- const uint32_t timeout) {
+ const uint32_t timeout) OVERRIDE {
return -1;
}
virtual int32_t MirrorRenderStream(const int renderId,
const bool enable,
const bool mirrorXAxis,
- const bool mirrorYAxis) {
+ const bool mirrorYAxis) OVERRIDE {
return -1;
}
diff --git a/media/webrtc/webrtcvideoengine.cc b/media/webrtc/webrtcvideoengine.cc
index 83b1177..b2533b3 100644
--- a/media/webrtc/webrtcvideoengine.cc
+++ b/media/webrtc/webrtcvideoengine.cc
@@ -64,6 +64,22 @@
#include "webrtc/experiments.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+namespace {
+
+template <class T>
+bool Changed(cricket::Settable<T> proposed,
+ cricket::Settable<T> original) {
+ return proposed.IsSet() && proposed != original;
+}
+
+template <class T>
+bool Changed(cricket::Settable<T> proposed,
+ cricket::Settable<T> original,
+ T* value) {
+ return proposed.Get(value) && proposed != original;
+}
+
+} // namespace
namespace cricket {
@@ -81,6 +97,9 @@
const int kCpuMonitorPeriodMs = 2000; // 2 seconds.
+// TODO(pthatcher): Figure out what the proper value here is, or if we
+// can just remove this altogether.
+static const int kDefaultRenderDelayMs = 100;
static const int kDefaultLogSeverity = rtc::LS_WARNING;
@@ -926,8 +945,6 @@
initialized_ = false;
SetTraceFilter(SeverityToFilter(kDefaultLogSeverity));
render_module_.reset(new WebRtcPassthroughRender());
- local_renderer_w_ = local_renderer_h_ = 0;
- local_renderer_ = NULL;
capture_started_ = false;
decoder_factory_ = NULL;
encoder_factory_ = NULL;
@@ -953,6 +970,9 @@
LOG(LS_ERROR) << "Failed to initialize list of supported codec types";
}
+ // Consider jitter, packet loss, etc when rendering. This will
+ // theoretically make rendering more smooth.
+ EnableTimedRender();
// Load our RTP Header extensions.
rtp_header_extensions_.push_back(
@@ -1059,10 +1079,6 @@
return VIDEO_RECV | VIDEO_SEND;
}
-bool WebRtcVideoEngine::SetOptions(const VideoOptions &options) {
- return true;
-}
-
bool WebRtcVideoEngine::SetDefaultEncoderConfig(
const VideoEncoderConfig& config) {
return SetDefaultCodec(config.max_codec);
@@ -1109,12 +1125,6 @@
return channel;
}
-bool WebRtcVideoEngine::SetLocalRenderer(VideoRenderer* renderer) {
- local_renderer_w_ = local_renderer_h_ = 0;
- local_renderer_ = renderer;
- return true;
-}
-
const std::vector<VideoCodec>& WebRtcVideoEngine::codecs() const {
return video_codecs_;
}
@@ -1568,6 +1578,7 @@
remb_enabled_(false),
render_started_(false),
first_receive_ssrc_(kSsrcUnset),
+ receiver_report_ssrc_(kSsrcUnset),
num_unsignalled_recv_channels_(0),
send_rtx_type_(-1),
send_red_type_(-1),
@@ -1580,7 +1591,19 @@
bool WebRtcVideoMediaChannel::Init() {
const uint32 ssrc_key = 0;
- return CreateChannel(ssrc_key, MD_SENDRECV, &default_channel_id_);
+ bool result = CreateChannel(ssrc_key, MD_SENDRECV, &default_channel_id_);
+ if (!result) {
+ return false;
+ }
+ if (voice_channel_) {
+ WebRtcVoiceMediaChannel* voice_channel =
+ static_cast<WebRtcVoiceMediaChannel*>(voice_channel_);
+ if (!voice_channel->SetupSharedBandwidthEstimation(
+ engine()->vie()->engine(), default_channel_id_)) {
+ return false;
+ }
+ }
+ return true;
}
WebRtcVideoMediaChannel::~WebRtcVideoMediaChannel() {
@@ -1752,6 +1775,35 @@
return true;
}
+bool WebRtcVideoMediaChannel::MaybeRegisterExternalEncoder(
+ WebRtcVideoChannelSendInfo* send_channel,
+ const webrtc::VideoCodec& codec) {
+ // Codec type not supported or encoder already registered, so
+ // nothing to do.
+ if (!engine()->IsExternalEncoderCodecType(codec.codecType)
+ || send_channel->IsEncoderRegistered(codec.plType)) {
+ return true;
+ }
+
+ webrtc::VideoEncoder* encoder =
+ engine()->CreateExternalEncoder(codec.codecType);
+ if (!encoder) {
+ // No encoder factor, so nothing to do.
+ return true;
+ }
+
+ const int channel_id = send_channel->channel_id();
+ if (engine()->vie()->ext_codec()->RegisterExternalSendCodec(
+ channel_id, codec.plType, encoder, false) != 0) {
+ LOG_RTCERR2(RegisterExternalSendCodec, channel_id, codec.plName);
+ engine()->DestroyExternalEncoder(encoder);
+ return false;
+ }
+
+ send_channel->RegisterEncoder(codec.plType, encoder);
+ return true;
+}
+
bool WebRtcVideoMediaChannel::GetSendCodec(VideoCodec* send_codec) {
if (!send_codec_) {
return false;
@@ -1861,18 +1913,9 @@
}
WebRtcVideoChannelSendInfo* send_channel = GetSendChannelBySsrcKey(ssrc_key);
- // Set the send (local) SSRC.
// If there are multiple send SSRCs, we can only set the first one here, and
// the rest of the SSRC(s) need to be set after SetSendCodec has been called
- // (with a codec requires multiple SSRC(s)).
- if (engine()->vie()->rtp()->SetLocalSSRC(channel_id,
- sp.first_ssrc()) != 0) {
- LOG_RTCERR2(SetLocalSSRC, channel_id, sp.first_ssrc());
- return false;
- }
-
- // Set the corresponding RTX SSRC.
- if (!SetLocalRtxSsrc(channel_id, sp, sp.first_ssrc(), 0)) {
+ if (!SetLimitedNumberOfSendSsrcs(channel_id, sp, 1)) {
return false;
}
@@ -1883,21 +1926,9 @@
return false;
}
- // At this point the channel's local SSRC has been updated. If the channel is
- // the default channel make sure that all the receive channels are updated as
- // well. Receive channels have to have the same SSRC as the default channel in
- // order to send receiver reports with this SSRC.
+ // Use the SSRC of the default channel in the RTCP receiver reports.
if (IsDefaultChannelId(channel_id)) {
- for (RecvChannelMap::const_iterator it = recv_channels_.begin();
- it != recv_channels_.end(); ++it) {
- WebRtcVideoChannelRecvInfo* info = it->second;
- int channel_id = info->channel_id();
- if (engine()->vie()->rtp()->SetLocalSSRC(channel_id,
- sp.first_ssrc()) != 0) {
- LOG_RTCERR1(SetLocalSSRC, it->first);
- return false;
- }
- }
+ SetReceiverReportSsrc(sp.first_ssrc());
}
send_channel->set_stream_params(sp);
@@ -2018,25 +2049,6 @@
return false;
}
- // Get the default renderer.
- VideoRenderer* default_renderer = NULL;
- if (InConferenceMode()) {
- // The recv_channels_ size start out being 1, so if it is two here
- // this is the first receive channel created (default_channel_id_
- // is not used for receiving in a conference call). This means
- // that the renderer stored inside default_channel_id_ should be
- // used for the just created channel.
- if (recv_channels_.size() == 2 && GetDefaultRecvChannel()) {
- GetDefaultRenderer(&default_renderer);
- }
- }
-
- // The first recv stream reuses the default renderer (if a default renderer
- // has been set).
- if (default_renderer) {
- SetRenderer(sp.first_ssrc(), default_renderer);
- }
-
LOG(LS_INFO) << "New video stream " << sp.first_ssrc()
<< " registered to VideoEngine channel #"
<< channel_id << " and connected to channel #"
@@ -2228,6 +2240,9 @@
return true;
}
if (!GetSendChannelBySsrcKey(local_ssrc)) {
+ // If a stream has multiple ssrcs, the local_ssrc could be any of
+ // them, but we use the first one (StreamParams::first_ssrc()) as
+ // the key.
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
@@ -2898,7 +2913,7 @@
}
// On success, SetSendCodec() will reset |send_start_bitrate_| to |bps/1000|,
- // by calling MaybeChangeBitrates. That method will also clamp the
+ // by calling SanitizeBitrates. That method will also clamp the
// start bitrate between min and max, consistent with the override behavior
// in SetMaxSendBandwidth.
webrtc::VideoCodec new_codec = *send_codec_;
@@ -2934,39 +2949,10 @@
return true;
}
- // Trigger SetSendCodec to set correct noise reduction state if the option has
- // changed.
- bool denoiser_changed = options.video_noise_reduction.IsSet() &&
- (options_.video_noise_reduction != options.video_noise_reduction);
-
- bool leaky_bucket_changed = options.video_leaky_bucket.IsSet() &&
- (options_.video_leaky_bucket != options.video_leaky_bucket);
-
- bool buffer_latency_changed = options.buffered_mode_latency.IsSet() &&
- (options_.buffered_mode_latency != options.buffered_mode_latency);
-
- bool dscp_option_changed = (options_.dscp != options.dscp);
-
- bool suspend_below_min_bitrate_changed =
- options.suspend_below_min_bitrate.IsSet() &&
- (options_.suspend_below_min_bitrate != options.suspend_below_min_bitrate);
-
- bool conference_mode_turned_off = false;
- if (options_.conference_mode.IsSet() && options.conference_mode.IsSet() &&
- options_.conference_mode.GetWithDefaultIfUnset(false) &&
- !options.conference_mode.GetWithDefaultIfUnset(false)) {
- conference_mode_turned_off = true;
- }
-
-#ifdef USE_WEBRTC_DEV_BRANCH
- bool payload_padding_changed = options.use_payload_padding.IsSet() &&
- options_.use_payload_padding != options.use_payload_padding;
-#endif
-
-
// Save the options, to be interpreted where appropriate.
// Use options_.SetAll() instead of assignment so that unset value in options
// will not overwrite the previous option value.
+ VideoOptions original = options_;
options_.SetAll(options);
// Set CPU options for all send channels.
@@ -2977,38 +2963,36 @@
}
if (send_codec_) {
- bool reset_send_codec_needed = denoiser_changed;
webrtc::VideoCodec new_codec = *send_codec_;
+ bool conference_mode_turned_off = (
+ original.conference_mode.IsSet() &&
+ options.conference_mode.IsSet() &&
+ original.conference_mode.GetWithDefaultIfUnset(false) &&
+ !options.conference_mode.GetWithDefaultIfUnset(false));
if (conference_mode_turned_off) {
// This is a special case for turning conference mode off.
// Max bitrate should go back to the default maximum value instead
// of the current maximum.
new_codec.maxBitrate = kAutoBandwidth;
- reset_send_codec_needed = true;
}
// TODO(pthatcher): Remove this. We don't need 4 ways to set bitrates.
int new_start_bitrate;
if (options.video_start_bitrate.Get(&new_start_bitrate)) {
new_codec.startBitrate = new_start_bitrate;
- reset_send_codec_needed = true;
}
-
- LOG(LS_INFO) << "Reset send codec needed is enabled? "
- << reset_send_codec_needed;
- if (reset_send_codec_needed) {
- if (!SetSendCodec(new_codec)) {
- return false;
- }
- LogSendCodecChange("SetOptions()");
+ if (!SetSendCodec(new_codec)) {
+ return false;
}
+ LogSendCodecChange("SetOptions()");
}
- if (leaky_bucket_changed) {
- bool enable_leaky_bucket =
- options_.video_leaky_bucket.GetWithDefaultIfUnset(true);
+ bool enable_leaky_bucket;
+ if (Changed(options.video_leaky_bucket,
+ original.video_leaky_bucket,
+ &enable_leaky_bucket)) {
LOG(LS_INFO) << "Leaky bucket is enabled? " << enable_leaky_bucket;
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
@@ -3022,10 +3006,11 @@
}
}
}
- if (buffer_latency_changed) {
- int buffer_latency =
- options_.buffered_mode_latency.GetWithDefaultIfUnset(
- cricket::kBufferedModeDisabled);
+
+ int buffer_latency;
+ if (Changed(options.buffered_mode_latency,
+ original.buffered_mode_latency,
+ &buffer_latency)) {
LOG(LS_INFO) << "Buffer latency is " << buffer_latency;
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
@@ -3044,17 +3029,24 @@
}
}
}
- if (dscp_option_changed) {
+
+ bool dscp_enabled;
+ if (Changed(options.dscp, original.dscp, &dscp_enabled)) {
rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
- if (options_.dscp.GetWithDefaultIfUnset(false))
+ if (dscp_enabled) {
dscp = kVideoDscpValue;
+ }
LOG(LS_INFO) << "DSCP is " << dscp;
if (MediaChannel::SetDscp(dscp) != 0) {
LOG(LS_WARNING) << "Failed to set DSCP settings for video channel";
}
}
- if (suspend_below_min_bitrate_changed) {
- if (options_.suspend_below_min_bitrate.GetWithDefaultIfUnset(false)) {
+
+ bool suspend_below_min_bitrate;
+ if (Changed(options.suspend_below_min_bitrate,
+ original.suspend_below_min_bitrate,
+ &suspend_below_min_bitrate)) {
+ if (suspend_below_min_bitrate) {
LOG(LS_INFO) << "Suspend below min bitrate enabled.";
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
@@ -3065,14 +3057,17 @@
LOG(LS_WARNING) << "Cannot disable video suspension once it is enabled";
}
}
+
#ifdef USE_WEBRTC_DEV_BRANCH
- if (payload_padding_changed) {
+ bool use_payload_padding;
+ if (Changed(options.use_payload_padding,
+ original.use_payload_padding,
+ &use_payload_padding)) {
LOG(LS_INFO) << "Payload-based padding called.";
for (SendChannelMap::iterator it = send_channels_.begin();
it != send_channels_.end(); ++it) {
engine()->vie()->rtp()->SetPadWithRedundantPayloads(
- it->second->channel_id(),
- options_.use_payload_padding.GetWithDefaultIfUnset(false));
+ it->second->channel_id(), use_payload_padding);
}
}
#endif
@@ -3139,10 +3134,6 @@
return true;
}
-bool WebRtcVideoMediaChannel::GetDefaultRenderer(VideoRenderer** renderer) {
- return GetRenderer(kDefaultChannelSsrcKey, renderer);
-}
-
bool WebRtcVideoMediaChannel::GetVideoAdapter(
uint32 ssrc, CoordinatedVideoAdapter** video_adapter) {
WebRtcVideoChannelSendInfo* send_channel = GetSendChannelBySsrc(ssrc);
@@ -3413,6 +3404,11 @@
return false;
}
+ if (engine()->vie()->render()->SetExpectedRenderDelay(
+ channel_id, kDefaultRenderDelayMs)) {
+ LOG_RTCERR2(SetExpectedRenderDelay,
+ channel_id, kDefaultRenderDelayMs);
+ }
if (engine_->vie()->rtp()->SetRembStatus(channel_id,
kNotSending,
@@ -3431,20 +3427,13 @@
return false;
}
- if (remote_ssrc != kDefaultChannelSsrcKey) {
- // Use the same SSRC as our default channel
- // (so the RTCP reports are correct).
- unsigned int send_ssrc = 0;
- webrtc::ViERTP_RTCP* rtp = engine()->vie()->rtp();
- if (rtp->GetLocalSSRC(default_channel_id_, send_ssrc) == -1) {
- LOG_RTCERR2(GetLocalSSRC, default_channel_id_, send_ssrc);
+ if (receiver_report_ssrc_ != kSsrcUnset) {
+ if (engine()->vie()->rtp()->SetLocalSSRC(
+ channel_id, receiver_report_ssrc_) == -1) {
+ LOG_RTCERR2(SetLocalSSRC, channel_id, receiver_report_ssrc_);
return false;
}
- if (rtp->SetLocalSSRC(channel_id, send_ssrc) == -1) {
- LOG_RTCERR2(SetLocalSSRC, channel_id, send_ssrc);
- return false;
- }
- } // Else this is the the default channel and we don't change the SSRC.
+ }
// Disable color enhancement since it is a bit too aggressive.
if (engine()->vie()->image()->EnableColorEnhancement(channel_id,
@@ -3673,21 +3662,7 @@
target_codec.codecSpecific.VP8.denoisingOn = enable_denoising;
}
- // Register external encoder if codec type is supported by encoder factory.
- if (engine()->IsExternalEncoderCodecType(codec.codecType) &&
- !send_channel->IsEncoderRegistered(target_codec.plType)) {
- webrtc::VideoEncoder* encoder =
- engine()->CreateExternalEncoder(codec.codecType);
- if (encoder) {
- if (engine()->vie()->ext_codec()->RegisterExternalSendCodec(
- channel_id, target_codec.plType, encoder, false) == 0) {
- send_channel->RegisterEncoder(target_codec.plType, encoder);
- } else {
- LOG_RTCERR2(RegisterExternalSendCodec, channel_id, target_codec.plName);
- engine()->DestroyExternalEncoder(encoder);
- }
- }
- }
+ MaybeRegisterExternalEncoder(send_channel, target_codec);
// Resolution and framerate may vary for different send channels.
const VideoFormat& video_format = send_channel->video_format();
@@ -3698,7 +3673,8 @@
LOG(LS_INFO) << "0x0 resolution selected. Captured frames will be dropped "
<< "for ssrc: " << ssrc << ".";
} else {
- MaybeChangeBitrates(channel_id, &target_codec);
+ StreamParams* send_params = send_channel->stream_params();
+ SanitizeBitrates(channel_id, &target_codec);
webrtc::VideoCodec current_codec;
if (!engine()->vie()->codec()->GetSendCodec(channel_id, current_codec)) {
// Compare against existing configured send codec.
@@ -3713,6 +3689,11 @@
return false;
}
+ if (send_params) {
+ if (!SetSendSsrcs(channel_id, *send_params, target_codec)) {
+ return false;
+ }
+ }
// NOTE: SetRtxSendPayloadType must be called after all simulcast SSRCs
// are configured. Otherwise ssrc's configured after this point will use
// the primary PT for RTX.
@@ -3956,6 +3937,7 @@
int screencast_min_bitrate =
options_.screencast_min_bitrate.GetWithDefaultIfUnset(0);
bool leaky_bucket = options_.video_leaky_bucket.GetWithDefaultIfUnset(true);
+ StreamParams* send_params = send_channel->stream_params();
bool reset_send_codec =
target_width != cur_width || target_height != cur_height;
if (vie_codec.codecType == webrtc::kVideoCodecVP8) {
@@ -3979,7 +3961,7 @@
vie_codec.codecSpecific.VP8.denoisingOn = enable_denoising;
vie_codec.codecSpecific.VP8.frameDroppingOn = vp8_frame_dropping;
}
- MaybeChangeBitrates(channel_id, &vie_codec);
+ SanitizeBitrates(channel_id, &vie_codec);
if (engine()->vie()->codec()->SetSendCodec(channel_id, vie_codec) != 0) {
LOG_RTCERR1(SetSendCodec, channel_id);
@@ -4001,6 +3983,13 @@
engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id,
leaky_bucket);
}
+ // TODO(sriniv): SetSendCodec already sets ssrc's like below.
+ // Consider removing.
+ if (send_params) {
+ if (!SetSendSsrcs(channel_id, *send_params, target_codec)) {
+ return false;
+ }
+ }
if (reset) {
*reset = true;
}
@@ -4010,7 +3999,7 @@
return true;
}
-void WebRtcVideoMediaChannel::MaybeChangeBitrates(
+void WebRtcVideoMediaChannel::SanitizeBitrates(
int channel_id, webrtc::VideoCodec* codec) {
codec->minBitrate = GetBitrate(codec->minBitrate, kMinVideoBitrate);
codec->startBitrate = GetBitrate(codec->startBitrate, kStartVideoBitrate);
@@ -4047,7 +4036,6 @@
codec->startBitrate = current_target_bitrate;
}
}
-
}
void WebRtcVideoMediaChannel::OnMessage(rtc::Message* msg) {
@@ -4156,21 +4144,55 @@
return SetHeaderExtension(setter, channel_id, extension);
}
-bool WebRtcVideoMediaChannel::SetLocalRtxSsrc(int channel_id,
- const StreamParams& send_params,
- uint32 primary_ssrc,
- int stream_idx) {
- uint32 rtx_ssrc = 0;
- bool has_rtx = send_params.GetFidSsrc(primary_ssrc, &rtx_ssrc);
- if (has_rtx && engine()->vie()->rtp()->SetLocalSSRC(
- channel_id, rtx_ssrc, webrtc::kViEStreamTypeRtx, stream_idx) != 0) {
- LOG_RTCERR4(SetLocalSSRC, channel_id, rtx_ssrc,
- webrtc::kViEStreamTypeRtx, stream_idx);
+bool WebRtcVideoMediaChannel::SetPrimaryAndRtxSsrcs(
+ int channel_id, int idx, uint32 primary_ssrc,
+ const StreamParams& send_params) {
+ LOG(LS_INFO) << "Set primary ssrc " << primary_ssrc
+ << " on channel " << channel_id << " idx " << idx;
+ if (engine()->vie()->rtp()->SetLocalSSRC(
+ channel_id, primary_ssrc, webrtc::kViEStreamTypeNormal, idx) != 0) {
+ LOG_RTCERR4(SetLocalSSRC,
+ channel_id, primary_ssrc, webrtc::kViEStreamTypeNormal, idx);
return false;
}
+
+ uint32 rtx_ssrc = 0;
+ if (send_params.GetFidSsrc(primary_ssrc, &rtx_ssrc)) {
+ LOG(LS_INFO) << "Set rtx ssrc " << rtx_ssrc
+ << " on channel " << channel_id << " idx " << idx;
+ if (engine()->vie()->rtp()->SetLocalSSRC(
+ channel_id, rtx_ssrc, webrtc::kViEStreamTypeRtx, idx) != 0) {
+ LOG_RTCERR4(SetLocalSSRC,
+ channel_id, rtx_ssrc, webrtc::kViEStreamTypeRtx, idx);
+ return false;
+ }
+ }
return true;
}
+bool WebRtcVideoMediaChannel::SetLimitedNumberOfSendSsrcs(
+ int channel_id, const StreamParams& sp, size_t limit) {
+ const SsrcGroup* sim_group = sp.get_ssrc_group(kSimSsrcGroupSemantics);
+ if (!sim_group || limit == 1) {
+ return SetPrimaryAndRtxSsrcs(channel_id, 0, sp.first_ssrc(), sp);
+ }
+
+ std::vector<uint32> ssrcs = sim_group->ssrcs;
+ for (size_t i = 0; i < ssrcs.size() && i < limit; ++i) {
+ if (!SetPrimaryAndRtxSsrcs(channel_id, static_cast<int>(i), ssrcs[i], sp)) {
+ return false;
+ }
+ }
+ return true;
+}
+
+bool WebRtcVideoMediaChannel::SetSendSsrcs(
+ int channel_id, const StreamParams& sp,
+ const webrtc::VideoCodec& codec) {
+ // TODO(pthatcher): Support more than one primary SSRC per stream.
+ return SetLimitedNumberOfSendSsrcs(channel_id, sp, 1);
+}
+
void WebRtcVideoMediaChannel::MaybeConnectCapturer(VideoCapturer* capturer) {
if (capturer && GetSendChannelNum(capturer) == 1) {
capturer->SignalVideoFrame.connect(this,
@@ -4184,6 +4206,18 @@
}
}
+void WebRtcVideoMediaChannel::SetReceiverReportSsrc(uint32 ssrc) {
+ for (RecvChannelMap::const_iterator it = recv_channels_.begin();
+ it != recv_channels_.end(); ++it) {
+ int channel_id = it->second->channel_id();
+ if (engine()->vie()->rtp()->SetLocalSSRC(channel_id, ssrc) != 0) {
+ LOG_RTCERR2(SetLocalSSRC, channel_id, ssrc);
+ ASSERT(false);
+ }
+ }
+ receiver_report_ssrc_ = ssrc;
+}
+
} // namespace cricket
#endif // HAVE_WEBRTC_VIDEO
diff --git a/media/webrtc/webrtcvideoengine.h b/media/webrtc/webrtcvideoengine.h
index c31f547..6f939d2 100644
--- a/media/webrtc/webrtcvideoengine.h
+++ b/media/webrtc/webrtcvideoengine.h
@@ -42,6 +42,9 @@
#if !defined(LIBPEERCONNECTION_LIB) && \
!defined(LIBPEERCONNECTION_IMPLEMENTATION)
+// If you hit this, then you've tried to include this header from outside
+// a shared library. An instance of this class must only be created from
+// within the library that actually implements it.
#error "Bogus include."
#endif
@@ -98,24 +101,23 @@
ViEWrapper* vie_wrapper,
ViETraceWrapper* tracing,
rtc::CpuMonitor* cpu_monitor);
- ~WebRtcVideoEngine();
+ virtual ~WebRtcVideoEngine();
// Basic video engine implementation.
bool Init(rtc::Thread* worker_thread);
void Terminate();
int GetCapabilities();
- bool SetOptions(const VideoOptions &options);
bool SetDefaultEncoderConfig(const VideoEncoderConfig& config);
VideoEncoderConfig GetDefaultEncoderConfig() const;
- WebRtcVideoMediaChannel* CreateChannel(VoiceMediaChannel* voice_channel);
+ virtual WebRtcVideoMediaChannel* CreateChannel(
+ VoiceMediaChannel* voice_channel);
const std::vector<VideoCodec>& codecs() const;
const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
void SetLogging(int min_sev, const char* filter);
- bool SetLocalRenderer(VideoRenderer* renderer);
sigslot::repeater2<VideoCapturer*, CaptureState> SignalCaptureStateChange;
// Set the VoiceEngine for A/V sync. This can only be called before Init.
@@ -127,7 +129,8 @@
// Set a WebRtcVideoEncoderFactory for external encoding. Video engine does
// not take the ownership of |encoder_factory|. The caller needs to make sure
// that |encoder_factory| outlives the video engine.
- void SetExternalEncoderFactory(WebRtcVideoEncoderFactory* encoder_factory);
+ virtual void SetExternalEncoderFactory(
+ WebRtcVideoEncoderFactory* encoder_factory);
// Enable the render module with timing control.
bool EnableTimedRender();
@@ -225,9 +228,6 @@
VideoChannels channels_;
bool capture_started_;
- int local_renderer_w_;
- int local_renderer_h_;
- VideoRenderer* local_renderer_;
rtc::scoped_ptr<rtc::CpuMonitor> cpu_monitor_;
};
@@ -238,7 +238,7 @@
public:
WebRtcVideoMediaChannel(WebRtcVideoEngine* engine,
VoiceMediaChannel* voice_channel);
- ~WebRtcVideoMediaChannel();
+ virtual ~WebRtcVideoMediaChannel();
bool Init();
WebRtcVideoEngine* engine() { return engine_; }
@@ -310,6 +310,20 @@
virtual int SendPacket(int channel, const void* data, int len);
virtual int SendRTCPPacket(int channel, const void* data, int len);
+ // Checks the current bitrate estimate and modifies the bitrates
+ // accordingly, including converting kAutoBandwidth to the correct defaults.
+ virtual void SanitizeBitrates(
+ int channel_id, webrtc::VideoCodec* video_codec);
+ virtual void LogSendCodecChange(const std::string& reason);
+ bool SetPrimaryAndRtxSsrcs(
+ int channel_id, int idx, uint32 primary_ssrc,
+ const StreamParams& send_params);
+ bool SetLimitedNumberOfSendSsrcs(
+ int channel_id, const StreamParams& send_params, size_t limit);
+ virtual bool SetSendSsrcs(
+ int channel_id, const StreamParams& send_params,
+ const webrtc::VideoCodec& codec);
+
private:
typedef std::map<uint32, WebRtcVideoChannelRecvInfo*> RecvChannelMap;
typedef std::map<uint32, WebRtcVideoChannelSendInfo*> SendChannelMap;
@@ -339,22 +353,23 @@
bool SetSendCodec(const webrtc::VideoCodec& codec);
bool SetSendCodec(WebRtcVideoChannelSendInfo* send_channel,
const webrtc::VideoCodec& codec);
- void LogSendCodecChange(const std::string& reason);
// Prepares the channel with channel id |info->channel_id()| to receive all
// codecs in |receive_codecs_| and start receive packets.
bool SetReceiveCodecs(WebRtcVideoChannelRecvInfo* info);
// Returns the channel ID that receives the stream with SSRC |ssrc|.
int GetRecvChannelId(uint32 ssrc);
bool MaybeSetRtxSsrc(const StreamParams& sp, int channel_id);
+ // Create and register an external endcoder if it's possible to do
+ // so and one isn't already registered.
+ bool MaybeRegisterExternalEncoder(
+ WebRtcVideoChannelSendInfo* send_channel,
+ const webrtc::VideoCodec& codec);
// Given captured video frame size, checks if we need to reset vie send codec.
// |reset| is set to whether resetting has happened on vie or not.
// Returns false on error.
bool MaybeResetVieSendCodec(WebRtcVideoChannelSendInfo* send_channel,
int new_width, int new_height, bool is_screencast,
bool* reset);
- // Checks the current bitrate estimate and modifies the bitrates
- // accordingly, including converting kAutoBandwidth to the correct defaults.
- void MaybeChangeBitrates(int channel_id, webrtc::VideoCodec* video_codec);
// Helper function for starting the sending of media on all channels or
// |channel_id|. Note that these two function do not change |sending_|.
bool StartSend();
@@ -371,9 +386,10 @@
// Returns the ssrc key corresponding to the provided local SSRC in
// |ssrc_key|. The return value is true upon success. If the local
// ssrc correspond to that of the default channel the key is
- // kDefaultChannelSsrcKey.
- // For all other channels the returned ssrc key will be the same as
- // the local ssrc.
+ // kDefaultChannelSsrcKey. For all other channels the returned ssrc
+ // key will be the same as the local ssrc. If a stream has more
+ // than one ssrc, the first (corresponding to
+ // StreamParams::first_ssrc()) is used as the key.
bool GetSendChannelSsrcKey(uint32 local_ssrc, uint32* ssrc_key);
WebRtcVideoChannelSendInfo* GetDefaultSendChannel();
WebRtcVideoChannelSendInfo* GetSendChannelBySsrcKey(uint32 ssrc_key);
@@ -387,8 +403,6 @@
bool IsDefaultChannelId(int channel_id) const {
return channel_id == default_channel_id_;
}
- bool GetDefaultRenderer(VideoRenderer** renderer);
-
bool DeleteSendChannel(uint32 ssrc_key);
WebRtcVideoChannelRecvInfo* GetDefaultRecvChannel();
@@ -414,10 +428,6 @@
// Signal when cpu adaptation has no further scope to adapt.
void OnCpuAdaptationUnable();
- // Set the local (send-side) RTX SSRC corresponding to primary_ssrc.
- bool SetLocalRtxSsrc(int channel_id, const StreamParams& send_params,
- uint32 primary_ssrc, int stream_idx);
-
// Connect |capturer| to WebRtcVideoMediaChannel if it is only registered
// to one send channel, i.e. the first send channel.
void MaybeConnectCapturer(VideoCapturer* capturer);
@@ -427,6 +437,9 @@
bool RemoveRecvStreamInternal(uint32 ssrc);
+ // Set the ssrc to use for RTCP receiver reports.
+ void SetReceiverReportSsrc(uint32 ssrc);
+
// Global state.
WebRtcVideoEngine* engine_;
VoiceMediaChannel* voice_channel_;
@@ -454,6 +467,7 @@
std::map<int, int> associated_payload_types_;
bool render_started_;
uint32 first_receive_ssrc_;
+ uint32 receiver_report_ssrc_;
std::vector<RtpHeaderExtension> receive_extensions_;
int num_unsignalled_recv_channels_;
diff --git a/media/webrtc/webrtcvideoengine2.cc b/media/webrtc/webrtcvideoengine2.cc
index ea53596..26e3079 100644
--- a/media/webrtc/webrtcvideoengine2.cc
+++ b/media/webrtc/webrtcvideoengine2.cc
@@ -42,8 +42,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/call.h"
-// TODO(pbos): Move codecs out of modules (webrtc:3070).
-#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
+#include "webrtc/video_encoder.h"
#define UNIMPLEMENTED \
LOG(LS_ERROR) << "Call to unimplemented function " << __FUNCTION__; \
@@ -55,6 +54,8 @@
// duration hasn't been implemented.
static const int kNackHistoryMs = 1000;
+static const int kDefaultQpMax = 56;
+
static const int kDefaultRtcpReceiverReportSsrc = 1;
struct VideoCodecPref {
@@ -187,7 +188,7 @@
stream.min_bitrate_bps = min_bitrate * 1000;
stream.target_bitrate_bps = stream.max_bitrate_bps = max_bitrate * 1000;
- int max_qp = 56;
+ int max_qp = kDefaultQpMax;
codec.GetParam(kCodecParamMaxQuantization, &max_qp);
stream.max_qp = max_qp;
std::vector<webrtc::VideoStream> streams;
@@ -200,7 +201,7 @@
const VideoOptions& options) {
assert(SupportsCodec(codec));
if (_stricmp(codec.name.c_str(), kVp8CodecName) == 0) {
- return webrtc::VP8Encoder::Create();
+ return webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp8);
}
// This shouldn't happen, we should be able to create encoders for all codecs
// we support.
@@ -213,14 +214,9 @@
const VideoOptions& options) {
assert(SupportsCodec(codec));
if (_stricmp(codec.name.c_str(), kVp8CodecName) == 0) {
- webrtc::VideoCodecVP8* settings = new webrtc::VideoCodecVP8();
- settings->resilience = webrtc::kResilientStream;
- settings->numberOfTemporalLayers = 1;
+ webrtc::VideoCodecVP8* settings = new webrtc::VideoCodecVP8(
+ webrtc::VideoEncoder::GetDefaultVp8Settings());
options.video_noise_reduction.Get(&settings->denoisingOn);
- settings->errorConcealmentOn = false;
- settings->automaticResizeOn = false;
- settings->frameDroppingOn = true;
- settings->keyFrameInterval = 3000;
return settings;
}
return NULL;
@@ -233,13 +229,9 @@
if (encoder_settings == NULL) {
return;
}
-
if (_stricmp(codec.name.c_str(), kVp8CodecName) == 0) {
delete reinterpret_cast<webrtc::VideoCodecVP8*>(encoder_settings);
- return;
}
- // We should be able to destroy all encoder settings we've allocated.
- assert(false);
}
bool WebRtcVideoEncoderFactory2::SupportsCodec(const VideoCodec& codec) {
@@ -283,37 +275,19 @@
}
WebRtcVideoEngine2::WebRtcVideoEngine2()
- : default_codec_format_(kDefaultVideoCodecPref.width,
+ : worker_thread_(NULL),
+ voice_engine_(NULL),
+ video_codecs_(DefaultVideoCodecs()),
+ default_codec_format_(kDefaultVideoCodecPref.width,
kDefaultVideoCodecPref.height,
FPS_TO_INTERVAL(kDefaultFramerate),
- FOURCC_ANY) {
- // Construct without a factory or voice engine.
- Construct(NULL, NULL, new rtc::CpuMonitor(NULL));
-}
-
-WebRtcVideoEngine2::WebRtcVideoEngine2(
- WebRtcVideoChannelFactory* channel_factory)
- : default_codec_format_(kDefaultVideoCodecPref.width,
- kDefaultVideoCodecPref.height,
- FPS_TO_INTERVAL(kDefaultFramerate),
- FOURCC_ANY) {
- // Construct without a voice engine.
- Construct(channel_factory, NULL, new rtc::CpuMonitor(NULL));
-}
-
-void WebRtcVideoEngine2::Construct(WebRtcVideoChannelFactory* channel_factory,
- WebRtcVoiceEngine* voice_engine,
- rtc::CpuMonitor* cpu_monitor) {
- LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2";
- worker_thread_ = NULL;
- voice_engine_ = voice_engine;
- initialized_ = false;
- capture_started_ = false;
- cpu_monitor_.reset(cpu_monitor);
- channel_factory_ = channel_factory;
-
- video_codecs_ = DefaultVideoCodecs();
-
+ FOURCC_ANY),
+ initialized_(false),
+ cpu_monitor_(new rtc::CpuMonitor(NULL)),
+ channel_factory_(NULL),
+ external_decoder_factory_(NULL),
+ external_encoder_factory_(NULL) {
+ LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2()";
rtp_header_extensions_.push_back(
RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
kRtpTimestampOffsetHeaderExtensionDefaultId));
@@ -322,6 +296,11 @@
kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
}
+void WebRtcVideoEngine2::SetChannelFactory(
+ WebRtcVideoChannelFactory* channel_factory) {
+ channel_factory_ = channel_factory;
+}
+
WebRtcVideoEngine2::~WebRtcVideoEngine2() {
LOG(LS_INFO) << "WebRtcVideoEngine2::~WebRtcVideoEngine2";
@@ -355,14 +334,6 @@
int WebRtcVideoEngine2::GetCapabilities() { return VIDEO_RECV | VIDEO_SEND; }
-bool WebRtcVideoEngine2::SetOptions(const VideoOptions& options) {
- // TODO(pbos): Do we need this? This is a no-op in the existing
- // WebRtcVideoEngine implementation.
- LOG(LS_VERBOSE) << "SetOptions: " << options.ToString();
- // options_ = options;
- return true;
-}
-
bool WebRtcVideoEngine2::SetDefaultEncoderConfig(
const VideoEncoderConfig& config) {
const VideoCodec& codec = config.max_codec;
@@ -424,14 +395,32 @@
}
}
-bool WebRtcVideoEngine2::EnableTimedRender() {
- // TODO(pbos): Figure out whether this can be removed.
- return true;
+void WebRtcVideoEngine2::SetExternalDecoderFactory(
+ WebRtcVideoDecoderFactory* decoder_factory) {
+ external_decoder_factory_ = decoder_factory;
}
-bool WebRtcVideoEngine2::SetLocalRenderer(VideoRenderer* renderer) {
- // TODO(pbos): Implement or remove. Unclear which stream should be rendered
- // locally even.
+void WebRtcVideoEngine2::SetExternalEncoderFactory(
+ WebRtcVideoEncoderFactory* encoder_factory) {
+ if (external_encoder_factory_ == encoder_factory) {
+ return;
+ }
+ if (external_encoder_factory_) {
+ external_encoder_factory_->RemoveObserver(this);
+ }
+ external_encoder_factory_ = encoder_factory;
+ if (external_encoder_factory_) {
+ external_encoder_factory_->AddObserver(this);
+ }
+
+ // Invoke OnCodecAvailable() here in case the list of codecs is already
+ // available when the encoder factory is installed. If not the encoder
+ // factory will invoke the callback later when the codecs become available.
+ OnCodecsAvailable();
+}
+
+bool WebRtcVideoEngine2::EnableTimedRender() {
+ // TODO(pbos): Figure out whether this can be removed.
return true;
}
@@ -517,6 +506,9 @@
return &default_video_encoder_factory_;
}
+void WebRtcVideoEngine2::OnCodecsAvailable() {
+ // TODO(pbos): Implement.
+}
// Thin map between VideoFrame and an existing webrtc::I420VideoFrame
// to avoid having to copy the rendered VideoFrame prematurely.
// This implementation is only safe to use in a const context and should never
@@ -679,8 +671,8 @@
WebRtcVideoEngine2* engine,
VoiceMediaChannel* voice_channel,
WebRtcVideoEncoderFactory2* encoder_factory)
- : encoder_factory_(encoder_factory),
- unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_) {
+ : unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
+ encoder_factory_(encoder_factory) {
// TODO(pbos): Connect the video and audio with |voice_channel|.
webrtc::Call::Config config(this);
Construct(webrtc::Call::Create(config), engine);
@@ -690,8 +682,8 @@
webrtc::Call* call,
WebRtcVideoEngine2* engine,
WebRtcVideoEncoderFactory2* encoder_factory)
- : encoder_factory_(encoder_factory),
- unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_) {
+ : unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
+ encoder_factory_(encoder_factory) {
Construct(call, engine);
}
@@ -1174,7 +1166,9 @@
}
void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
- LOG(LS_VERBOSE) << "OnReadySend: " << (ready ? "Ready." : "Not ready.");
+ LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
+ call_->SignalNetworkState(ready ? webrtc::Call::kNetworkUp
+ : webrtc::Call::kNetworkDown);
}
bool WebRtcVideoChannel2::MuteStream(uint32 ssrc, bool mute) {
@@ -1315,10 +1309,10 @@
const StreamParams& sp,
const std::vector<webrtc::RtpExtension>& rtp_extensions)
: call_(call),
- parameters_(webrtc::VideoSendStream::Config(), options, codec_settings),
encoder_factory_(encoder_factory),
- capturer_(NULL),
stream_(NULL),
+ parameters_(webrtc::VideoSendStream::Config(), options, codec_settings),
+ capturer_(NULL),
sending_(false),
muted_(false) {
parameters_.config.rtp.max_packet_size = kVideoMtu;
@@ -1385,16 +1379,10 @@
const VideoFrame* frame) {
LOG(LS_VERBOSE) << "InputFrame: " << frame->GetWidth() << "x"
<< frame->GetHeight();
- bool is_screencast = capturer->IsScreencast();
// Lock before copying, can be called concurrently when swapping input source.
rtc::CritScope frame_cs(&frame_lock_);
- if (!muted_) {
- ConvertToI420VideoFrame(*frame, &video_frame_);
- } else {
- // Create a tiny black frame to transmit instead.
- CreateBlackFrame(&video_frame_, 1, 1);
- is_screencast = false;
- }
+ ConvertToI420VideoFrame(*frame, &video_frame_);
+
rtc::CritScope cs(&lock_);
if (stream_ == NULL) {
LOG(LS_WARNING) << "Capturer inputting frames before send codecs are "
@@ -1406,14 +1394,20 @@
LOG(LS_VERBOSE) << "VideoFormat 0x0 set, Dropping frame.";
return;
}
- // Reconfigure codec if necessary.
- if (is_screencast) {
- SetDimensions(video_frame_.width(), video_frame_.height());
+ if (muted_) {
+ // Create a black frame to transmit instead.
+ CreateBlackFrame(&video_frame_,
+ static_cast<int>(frame->GetWidth()),
+ static_cast<int>(frame->GetHeight()));
}
+ // Reconfigure codec if necessary.
+ SetDimensions(
+ video_frame_.width(), video_frame_.height(), capturer->IsScreencast());
+
LOG(LS_VERBOSE) << "SwapFrame: " << video_frame_.width() << "x"
<< video_frame_.height() << " -> (codec) "
- << parameters_.video_streams.back().width << "x"
- << parameters_.video_streams.back().height;
+ << parameters_.encoder_config.streams.back().width << "x"
+ << parameters_.encoder_config.streams.back().height;
stream_->Input()->SwapFrame(&video_frame_);
}
@@ -1437,7 +1431,7 @@
black_frame.CreateEmptyFrame(
width, height, width, half_width, half_width);
SetWebRtcFrameToBlack(&black_frame);
- SetDimensions(width, height);
+ SetDimensions(width, height, false);
stream_->Input()->SwapFrame(&black_frame);
}
@@ -1469,9 +1463,9 @@
<< parameters_.config.rtp.ssrcs[0] << ".";
} else {
// TODO(pbos): Fix me, this only affects the last stream!
- parameters_.video_streams.back().max_framerate =
+ parameters_.encoder_config.streams.back().max_framerate =
VideoFormat::IntervalToFps(format.interval);
- SetDimensions(format.width, format.height);
+ SetDimensions(format.width, format.height, false);
}
format_ = format;
@@ -1517,7 +1511,7 @@
if (video_streams.empty()) {
return;
}
- parameters_.video_streams = video_streams;
+ parameters_.encoder_config.streams = video_streams;
format_ = VideoFormat(codec_settings.codec.width,
codec_settings.codec.height,
VideoFormat::FpsToInterval(30),
@@ -1560,35 +1554,55 @@
RecreateWebRtcStream();
}
-void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions(int width,
- int height) {
- assert(!parameters_.video_streams.empty());
+void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions(
+ int width,
+ int height,
+ bool override_max) {
+ assert(!parameters_.encoder_config.streams.empty());
LOG(LS_VERBOSE) << "SetDimensions: " << width << "x" << height;
- if (parameters_.video_streams.back().width == width &&
- parameters_.video_streams.back().height == height) {
- return;
- }
-
- // TODO(pbos): Fix me, this only affects the last stream!
- parameters_.video_streams.back().width = width;
- parameters_.video_streams.back().height = height;
VideoCodecSettings codec_settings;
parameters_.codec_settings.Get(&codec_settings);
- void* encoder_settings = encoder_factory_->CreateVideoEncoderSettings(
- codec_settings.codec, parameters_.options);
+ // Restrict dimensions according to codec max.
+ if (!override_max) {
+ if (codec_settings.codec.width < width)
+ width = codec_settings.codec.width;
+ if (codec_settings.codec.height < height)
+ height = codec_settings.codec.height;
+ }
- bool stream_reconfigured = stream_->ReconfigureVideoEncoder(
- parameters_.video_streams, encoder_settings);
+ if (parameters_.encoder_config.streams.back().width == width &&
+ parameters_.encoder_config.streams.back().height == height) {
+ return;
+ }
- encoder_factory_->DestroyVideoEncoderSettings(codec_settings.codec,
- encoder_settings);
+ webrtc::VideoEncoderConfig encoder_config = parameters_.encoder_config;
+ encoder_config.encoder_specific_settings =
+ encoder_factory_->CreateVideoEncoderSettings(codec_settings.codec,
+ parameters_.options);
+
+ VideoCodec codec = codec_settings.codec;
+ codec.width = width;
+ codec.height = height;
+
+ encoder_config.streams = encoder_factory_->CreateVideoStreams(
+ codec, parameters_.options, parameters_.config.rtp.ssrcs.size());
+
+ bool stream_reconfigured = stream_->ReconfigureVideoEncoder(encoder_config);
+
+ encoder_factory_->DestroyVideoEncoderSettings(
+ codec_settings.codec,
+ encoder_config.encoder_specific_settings);
+
+ encoder_config.encoder_specific_settings = NULL;
if (!stream_reconfigured) {
LOG(LS_WARNING) << "Failed to reconfigure video encoder for dimensions: "
<< width << "x" << height;
return;
}
+
+ parameters_.encoder_config = encoder_config;
}
void WebRtcVideoChannel2::WebRtcVideoSendStream::Start() {
@@ -1652,9 +1666,9 @@
info.input_frame_width = last_captured_frame_format.width;
info.input_frame_height = last_captured_frame_format.height;
info.send_frame_width =
- static_cast<int>(parameters_.video_streams.front().width);
+ static_cast<int>(parameters_.encoder_config.streams.front().width);
info.send_frame_height =
- static_cast<int>(parameters_.video_streams.front().height);
+ static_cast<int>(parameters_.encoder_config.streams.front().height);
}
// TODO(pbos): Support or remove the following stats.
@@ -1671,14 +1685,18 @@
VideoCodecSettings codec_settings;
parameters_.codec_settings.Get(&codec_settings);
- void* encoder_settings = encoder_factory_->CreateVideoEncoderSettings(
- codec_settings.codec, parameters_.options);
+ parameters_.encoder_config.encoder_specific_settings =
+ encoder_factory_->CreateVideoEncoderSettings(codec_settings.codec,
+ parameters_.options);
- stream_ = call_->CreateVideoSendStream(
- parameters_.config, parameters_.video_streams, encoder_settings);
+ stream_ = call_->CreateVideoSendStream(parameters_.config,
+ parameters_.encoder_config);
- encoder_factory_->DestroyVideoEncoderSettings(codec_settings.codec,
- encoder_settings);
+ encoder_factory_->DestroyVideoEncoderSettings(
+ codec_settings.codec,
+ parameters_.encoder_config.encoder_specific_settings);
+
+ parameters_.encoder_config.encoder_specific_settings = NULL;
if (sending_) {
stream_->Start();
@@ -1690,11 +1708,11 @@
const webrtc::VideoReceiveStream::Config& config,
const std::vector<VideoCodecSettings>& recv_codecs)
: call_(call),
- config_(config),
stream_(NULL),
+ config_(config),
+ renderer_(NULL),
last_width_(-1),
- last_height_(-1),
- renderer_(NULL) {
+ last_height_(-1) {
config_.renderer = this;
// SetRecvCodecs will also reset (start) the VideoReceiveStream.
SetRecvCodecs(recv_codecs);
diff --git a/media/webrtc/webrtcvideoengine2.h b/media/webrtc/webrtcvideoengine2.h
index c7eb6a4..18a80d7 100644
--- a/media/webrtc/webrtcvideoengine2.h
+++ b/media/webrtc/webrtcvideoengine2.h
@@ -34,10 +34,12 @@
#include "talk/media/base/mediaengine.h"
#include "talk/media/webrtc/webrtcvideochannelfactory.h"
+#include "talk/media/webrtc/webrtcvideodecoderfactory.h"
+#include "talk/media/webrtc/webrtcvideoencoderfactory.h"
#include "webrtc/base/cpumonitor.h"
#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_video/interface/i420_video_frame.h"
-#include "webrtc/system_wrappers/interface/thread_annotations.h"
#include "webrtc/transport.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_renderer.h"
@@ -65,14 +67,13 @@
class VideoProcessor;
class VideoRenderer;
class VoiceMediaChannel;
-class WebRtcVideoChannel2;
class WebRtcDecoderObserver;
class WebRtcEncoderObserver;
class WebRtcLocalStreamInfo;
class WebRtcRenderAdapter;
+class WebRtcVideoChannel2;
class WebRtcVideoChannelRecvInfo;
class WebRtcVideoChannelSendInfo;
-class WebRtcVideoDecoderFactory;
class WebRtcVoiceEngine;
struct CapturedFrame;
@@ -119,9 +120,8 @@
const VideoCodec& codec,
const VideoOptions& options);
- virtual void* CreateVideoEncoderSettings(
- const VideoCodec& codec,
- const VideoOptions& options);
+ virtual void* CreateVideoEncoderSettings(const VideoCodec& codec,
+ const VideoOptions& options);
virtual void DestroyVideoEncoderSettings(const VideoCodec& codec,
void* encoder_settings);
@@ -130,20 +130,21 @@
};
// WebRtcVideoEngine2 is used for the new native WebRTC Video API (webrtc:1667).
-class WebRtcVideoEngine2 : public sigslot::has_slots<> {
+class WebRtcVideoEngine2 : public sigslot::has_slots<>,
+ public WebRtcVideoEncoderFactory::Observer {
public:
// Creates the WebRtcVideoEngine2 with internal VideoCaptureModule.
WebRtcVideoEngine2();
- // Custom WebRtcVideoChannelFactory for testing purposes.
- explicit WebRtcVideoEngine2(WebRtcVideoChannelFactory* channel_factory);
- ~WebRtcVideoEngine2();
+ virtual ~WebRtcVideoEngine2();
+
+ // Use a custom WebRtcVideoChannelFactory (for testing purposes).
+ void SetChannelFactory(WebRtcVideoChannelFactory* channel_factory);
// Basic video engine implementation.
bool Init(rtc::Thread* worker_thread);
void Terminate();
int GetCapabilities();
- bool SetOptions(const VideoOptions& options);
bool SetDefaultEncoderConfig(const VideoEncoderConfig& config);
VideoEncoderConfig GetDefaultEncoderConfig() const;
@@ -153,20 +154,23 @@
const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
void SetLogging(int min_sev, const char* filter);
+ // Set a WebRtcVideoDecoderFactory for external decoding. Video engine does
+ // not take the ownership of |decoder_factory|. The caller needs to make sure
+ // that |decoder_factory| outlives the video engine.
+ void SetExternalDecoderFactory(WebRtcVideoDecoderFactory* decoder_factory);
+ // Set a WebRtcVideoEncoderFactory for external encoding. Video engine does
+ // not take the ownership of |encoder_factory|. The caller needs to make sure
+ // that |encoder_factory| outlives the video engine.
+ virtual void SetExternalEncoderFactory(
+ WebRtcVideoEncoderFactory* encoder_factory);
+
bool EnableTimedRender();
- // No-op, never used.
- bool SetLocalRenderer(VideoRenderer* renderer);
// This is currently ignored.
sigslot::repeater2<VideoCapturer*, CaptureState> SignalCaptureStateChange;
// Set the VoiceEngine for A/V sync. This can only be called before Init.
bool SetVoiceEngine(WebRtcVoiceEngine* voice_engine);
- // Functions called by WebRtcVideoChannel2.
- const VideoFormat& default_codec_format() const {
- return default_codec_format_;
- }
-
bool FindCodec(const VideoCodec& in);
bool CanSendCodec(const VideoCodec& in,
const VideoCodec& current,
@@ -181,9 +185,7 @@
virtual WebRtcVideoEncoderFactory2* GetVideoEncoderFactory();
private:
- void Construct(WebRtcVideoChannelFactory* channel_factory,
- WebRtcVoiceEngine* voice_engine,
- rtc::CpuMonitor* cpu_monitor);
+ virtual void OnCodecsAvailable() OVERRIDE;
rtc::Thread* worker_thread_;
WebRtcVoiceEngine* voice_engine_;
@@ -193,8 +195,6 @@
bool initialized_;
- bool capture_started_;
-
// Critical section to protect the media processor register/unregister
// while processing a frame
rtc::CriticalSection signal_media_critical_;
@@ -202,6 +202,9 @@
rtc::scoped_ptr<rtc::CpuMonitor> cpu_monitor_;
WebRtcVideoChannelFactory* channel_factory_;
WebRtcVideoEncoderFactory2 default_video_encoder_factory_;
+
+ WebRtcVideoDecoderFactory* external_decoder_factory_;
+ WebRtcVideoEncoderFactory* external_encoder_factory_;
};
class WebRtcVideoChannel2 : public rtc::MessageHandler,
@@ -328,13 +331,16 @@
// Sent resolutions + bitrates etc. by the underlying VideoSendStream,
// typically changes when setting a new resolution or reconfiguring
// bitrates.
- std::vector<webrtc::VideoStream> video_streams;
+ webrtc::VideoEncoderConfig encoder_config;
};
void SetCodecAndOptions(const VideoCodecSettings& codec,
- const VideoOptions& options);
- void RecreateWebRtcStream();
- void SetDimensions(int width, int height);
+ const VideoOptions& options)
+ EXCLUSIVE_LOCKS_REQUIRED(lock_);
+ void RecreateWebRtcStream() EXCLUSIVE_LOCKS_REQUIRED(lock_);
+ // When |override_max| is false constrain width/height to codec dimensions.
+ void SetDimensions(int width, int height, bool override_max)
+ EXCLUSIVE_LOCKS_REQUIRED(lock_);
webrtc::Call* const call_;
WebRtcVideoEncoderFactory2* const encoder_factory_;
diff --git a/media/webrtc/webrtcvideoengine2_unittest.cc b/media/webrtc/webrtcvideoengine2_unittest.cc
index 9c313e6..6112c50 100644
--- a/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -68,13 +68,10 @@
namespace cricket {
FakeVideoSendStream::FakeVideoSendStream(
const webrtc::VideoSendStream::Config& config,
- const std::vector<webrtc::VideoStream>& video_streams,
- const void* encoder_settings)
- : sending_(false),
- config_(config),
- codec_settings_set_(false) {
+ const webrtc::VideoEncoderConfig& encoder_config)
+ : sending_(false), config_(config), codec_settings_set_(false) {
assert(config.encoder_settings.encoder != NULL);
- ReconfigureVideoEncoder(video_streams, encoder_settings);
+ ReconfigureVideoEncoder(encoder_config);
}
webrtc::VideoSendStream::Config FakeVideoSendStream::GetConfig() {
@@ -82,7 +79,7 @@
}
std::vector<webrtc::VideoStream> FakeVideoSendStream::GetVideoStreams() {
- return video_streams_;
+ return encoder_config_.streams;
}
bool FakeVideoSendStream::IsSending() const {
@@ -104,15 +101,14 @@
}
bool FakeVideoSendStream::ReconfigureVideoEncoder(
- const std::vector<webrtc::VideoStream>& streams,
- const void* encoder_specific) {
- video_streams_ = streams;
- if (encoder_specific != NULL) {
+ const webrtc::VideoEncoderConfig& config) {
+ encoder_config_ = config;
+ if (config.encoder_specific_settings != NULL) {
assert(config_.encoder_settings.payload_name == "VP8");
- vp8_settings_ =
- *reinterpret_cast<const webrtc::VideoCodecVP8*>(encoder_specific);
+ vp8_settings_ = *reinterpret_cast<const webrtc::VideoCodecVP8*>(
+ config.encoder_specific_settings);
}
- codec_settings_set_ = encoder_specific != NULL;
+ codec_settings_set_ = config.encoder_specific_settings != NULL;
return true;
}
@@ -157,7 +153,9 @@
void FakeVideoReceiveStream::GetCurrentReceiveCodec(webrtc::VideoCodec* codec) {
}
-FakeCall::FakeCall() { SetVideoCodecs(GetDefaultVideoCodecs()); }
+FakeCall::FakeCall() : network_state_(kNetworkUp) {
+ SetVideoCodecs(GetDefaultVideoCodecs());
+}
FakeCall::~FakeCall() {
EXPECT_EQ(0u, video_send_streams_.size());
@@ -218,12 +216,15 @@
return codecs;
}
+webrtc::Call::NetworkState FakeCall::GetNetworkState() const {
+ return network_state_;
+}
+
webrtc::VideoSendStream* FakeCall::CreateVideoSendStream(
const webrtc::VideoSendStream::Config& config,
- const std::vector<webrtc::VideoStream>& video_streams,
- const void* encoder_settings) {
+ const webrtc::VideoEncoderConfig& encoder_config) {
FakeVideoSendStream* fake_stream =
- new FakeVideoSendStream(config, video_streams, encoder_settings);
+ new FakeVideoSendStream(config, encoder_config);
video_send_streams_.push_back(fake_stream);
return fake_stream;
}
@@ -274,6 +275,10 @@
return 0;
}
+void FakeCall::SignalNetworkState(webrtc::Call::NetworkState state) {
+ network_state_ = state;
+}
+
FakeWebRtcVideoChannel2::FakeWebRtcVideoChannel2(
FakeCall* call,
WebRtcVideoEngine2* engine,
@@ -289,6 +294,7 @@
VoiceMediaChannel* FakeWebRtcVideoChannel2::GetVoiceChannel() {
return voice_channel_;
}
+
FakeCall* FakeWebRtcVideoChannel2::GetFakeCall() {
return fake_call_;
}
@@ -309,7 +315,8 @@
class WebRtcVideoEngine2Test : public testing::Test {
public:
- WebRtcVideoEngine2Test() : engine_(&factory_) {
+ WebRtcVideoEngine2Test() {
+ engine_.SetChannelFactory(&factory_);
std::vector<VideoCodec> engine_codecs = engine_.codecs();
assert(!engine_codecs.empty());
bool codec_set = false;
@@ -535,15 +542,17 @@
WEBRTC_BASE_TEST(AdaptResolution4x3);
-WEBRTC_BASE_TEST(MuteStream);
-
-WEBRTC_BASE_TEST(MultipleSendStreams);
-
// TODO(juberti): Restore this test once we support sending 0 fps.
WEBRTC_DISABLED_BASE_TEST(AdaptDropAllFrames);
// TODO(juberti): Understand why we get decode errors on this test.
WEBRTC_DISABLED_BASE_TEST(AdaptFramerate);
+WEBRTC_BASE_TEST(SendsLowerResolutionOnSmallerFrames);
+
+WEBRTC_BASE_TEST(MuteStream);
+
+WEBRTC_BASE_TEST(MultipleSendStreams);
+
WEBRTC_BASE_TEST(SetSendStreamFormat0x0);
// TODO(zhurunz): Fix the flakey test.
@@ -1611,8 +1620,17 @@
FAIL() << "Not implemented."; // TODO(pbos): Implement.
}
-TEST_F(WebRtcVideoChannel2Test, DISABLED_OnReadyToSend) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
+TEST_F(WebRtcVideoChannel2Test, OnReadyToSendSignalsNetworkState) {
+ EXPECT_EQ(webrtc::Call::kNetworkUp,
+ fake_channel_->GetFakeCall()->GetNetworkState());
+
+ channel_->OnReadyToSend(false);
+ EXPECT_EQ(webrtc::Call::kNetworkDown,
+ fake_channel_->GetFakeCall()->GetNetworkState());
+
+ channel_->OnReadyToSend(true);
+ EXPECT_EQ(webrtc::Call::kNetworkUp,
+ fake_channel_->GetFakeCall()->GetNetworkState());
}
TEST_F(WebRtcVideoChannel2Test, DISABLED_CaptureFrameTimestampToNtpTimestamp) {
diff --git a/media/webrtc/webrtcvideoengine2_unittest.h b/media/webrtc/webrtcvideoengine2_unittest.h
index 54e6f06..30f1efb 100644
--- a/media/webrtc/webrtcvideoengine2_unittest.h
+++ b/media/webrtc/webrtcvideoengine2_unittest.h
@@ -39,8 +39,7 @@
class FakeVideoSendStream : public webrtc::VideoSendStream {
public:
FakeVideoSendStream(const webrtc::VideoSendStream::Config& config,
- const std::vector<webrtc::VideoStream>& video_streams,
- const void* encoder_settings);
+ const webrtc::VideoEncoderConfig& encoder_config);
webrtc::VideoSendStream::Config GetConfig();
std::vector<webrtc::VideoStream> GetVideoStreams();
@@ -51,8 +50,7 @@
virtual webrtc::VideoSendStream::Stats GetStats() const OVERRIDE;
virtual bool ReconfigureVideoEncoder(
- const std::vector<webrtc::VideoStream>& streams,
- const void* encoder_specific);
+ const webrtc::VideoEncoderConfig& config) OVERRIDE;
virtual webrtc::VideoSendStreamInput* Input() OVERRIDE;
@@ -61,7 +59,7 @@
bool sending_;
webrtc::VideoSendStream::Config config_;
- std::vector<webrtc::VideoStream> video_streams_;
+ webrtc::VideoEncoderConfig encoder_config_;
bool codec_settings_set_;
webrtc::VideoCodecVP8 vp8_settings_;
};
@@ -103,11 +101,12 @@
std::vector<webrtc::VideoCodec> GetDefaultVideoCodecs();
+ webrtc::Call::NetworkState GetNetworkState() const;
+
private:
virtual webrtc::VideoSendStream* CreateVideoSendStream(
const webrtc::VideoSendStream::Config& config,
- const std::vector<webrtc::VideoStream>& video_streams,
- const void* encoder_settings) OVERRIDE;
+ const webrtc::VideoEncoderConfig& encoder_config) OVERRIDE;
virtual void DestroyVideoSendStream(
webrtc::VideoSendStream* send_stream) OVERRIDE;
@@ -122,6 +121,9 @@
virtual uint32_t SendBitrateEstimate() OVERRIDE;
virtual uint32_t ReceiveBitrateEstimate() OVERRIDE;
+ virtual void SignalNetworkState(webrtc::Call::NetworkState state) OVERRIDE;
+
+ webrtc::Call::NetworkState network_state_;
std::vector<webrtc::VideoCodec> codecs_;
std::vector<FakeVideoSendStream*> video_send_streams_;
std::vector<FakeVideoReceiveStream*> video_receive_streams_;
diff --git a/media/webrtc/webrtcvideoengine_unittest.cc b/media/webrtc/webrtcvideoengine_unittest.cc
index 11edd05..35f05e1 100644
--- a/media/webrtc/webrtcvideoengine_unittest.cc
+++ b/media/webrtc/webrtcvideoengine_unittest.cc
@@ -26,22 +26,16 @@
*/
#include "talk/media/base/constants.h"
-#include "talk/media/base/fakemediaprocessor.h"
#include "talk/media/base/fakenetworkinterface.h"
-#include "talk/media/base/fakevideorenderer.h"
#include "talk/media/base/mediachannel.h"
#include "talk/media/base/testutils.h"
-#include "talk/media/base/videoadapter.h"
#include "talk/media/base/videoengine_unittest.h"
-#include "talk/media/webrtc/fakewebrtcvideocapturemodule.h"
#include "talk/media/webrtc/fakewebrtcvideoengine.h"
-#include "talk/media/webrtc/fakewebrtcvoiceengine.h"
#include "webrtc/base/fakecpumonitor.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/stream.h"
-#include "talk/media/webrtc/webrtcvideocapturer.h"
#include "talk/media/webrtc/webrtcvideoengine.h"
#include "talk/media/webrtc/webrtcvideoframe.h"
#include "talk/media/webrtc/webrtcvoiceengine.h"
@@ -54,9 +48,6 @@
using cricket::kRtpAbsoluteSenderTimeHeaderExtension;
static const cricket::VideoCodec kVP8Codec720p(100, "VP8", 1280, 720, 30, 0);
-static const cricket::VideoCodec kVP8Codec360p(100, "VP8", 640, 360, 30, 0);
-static const cricket::VideoCodec kVP8Codec270p(100, "VP8", 480, 270, 30, 0);
-static const cricket::VideoCodec kVP8Codec180p(100, "VP8", 320, 180, 30, 0);
static const cricket::VideoCodec kVP8Codec(100, "VP8", 640, 400, 30, 0);
static const cricket::VideoCodec kH264Codec(127, "H264", 640, 400, 30, 0);
@@ -78,7 +69,6 @@
static const uint32 kRtxSsrcs1[] = {4};
static const uint32 kRtxSsrcs3[] = {4, 5, 6};
-
class FakeViEWrapper : public cricket::ViEWrapper {
public:
explicit FakeViEWrapper(cricket::FakeWebRtcVideoEngine* engine)
@@ -1556,7 +1546,6 @@
768, kMinBandwidthKbps, kStartBandwidthKbps);
}
-
// Test that sending screencast frames doesn't change bitrate.
TEST_F(WebRtcVideoEngineTestFake, SetBandwidthScreencast) {
EXPECT_TRUE(SetupEngine());
@@ -1576,7 +1565,6 @@
VerifyVP8SendCodec(channel_num, kVP8Codec.width, kVP8Codec.height, 0, 111);
}
-
// Test SetSendSsrc.
TEST_F(WebRtcVideoEngineTestFake, SetSendSsrcAndCname) {
EXPECT_TRUE(SetupEngine());
@@ -1597,7 +1585,6 @@
EXPECT_STREQ("cname", rtcp_cname);
}
-
// Test that the local SSRC is the same on sending and receiving channels if the
// receive channel is created before the send channel.
TEST_F(WebRtcVideoEngineTestFake, SetSendSsrcAfterCreatingReceiveChannel) {
@@ -1618,7 +1605,6 @@
EXPECT_EQ(1, vie_.GetNumSsrcs(receive_channel_num));
}
-
// Test SetOptions with denoising flag.
TEST_F(WebRtcVideoEngineTestFake, SetOptionsWithDenoising) {
EXPECT_TRUE(SetupEngine());
@@ -1707,7 +1693,6 @@
EXPECT_EQ(-1, vie_.GetIncomingFrameNum(channel1));
}
-
TEST_F(WebRtcVideoEngineTestFake, SendReceiveBitratesStats) {
EXPECT_TRUE(SetupEngine());
cricket::VideoOptions options;
@@ -1839,7 +1824,6 @@
EXPECT_EQ(high, 1.0f);
}
-
TEST_F(WebRtcVideoEngineTestFake, ResetCodecOnScreencast) {
EXPECT_TRUE(SetupEngine());
cricket::VideoOptions options;
@@ -1870,7 +1854,6 @@
EXPECT_FALSE(gcodec.codecSpecific.VP8.denoisingOn);
}
-
TEST_F(WebRtcVideoEngineTestFake, DontRegisterDecoderIfFactoryIsNotGiven) {
engine_.SetExternalDecoderFactory(NULL);
EXPECT_TRUE(SetupEngine());
@@ -2397,9 +2380,6 @@
EXPECT_FALSE(channel_->SetRecvCodecs(codecs));
}
-// Disable for TSan v2, see
-// https://code.google.com/p/webrtc/issues/detail?id=3671 for details.
-#if !defined(THREAD_SANITIZER)
TEST_F(WebRtcVideoMediaChannelTest, GetRtpSendTimeExtension) {
// Enable RTP timestamp extension.
const int id = 12;
@@ -2411,7 +2391,6 @@
EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
EXPECT_EQ(id, channel_->GetRtpSendTimeExtnId());
}
-#endif // if !defined(THREAD_SANITIZER)
TEST_F(WebRtcVideoMediaChannelTest, SetSend) {
Base::SetSend();
@@ -2574,7 +2553,6 @@
Base::RejectEmptyStreamParams();
}
-
TEST_F(WebRtcVideoMediaChannelTest, AdaptResolution16x10) {
Base::AdaptResolution16x10();
}
diff --git a/media/webrtc/webrtcvideoframe_unittest.cc b/media/webrtc/webrtcvideoframe_unittest.cc
index 4cfe7c0..5f65c58 100644
--- a/media/webrtc/webrtcvideoframe_unittest.cc
+++ b/media/webrtc/webrtcvideoframe_unittest.cc
@@ -27,14 +27,10 @@
#include "talk/media/base/videoframe_unittest.h"
#include "talk/media/webrtc/webrtcvideoframe.h"
-#include "webrtc/base/flags.h"
-
-extern int FLAG_yuvconverter_repeat; // From lmivideoframe_unittest.cc.
class WebRtcVideoFrameTest : public VideoFrameTest<cricket::WebRtcVideoFrame> {
public:
WebRtcVideoFrameTest() {
- repeat_ = FLAG_yuvconverter_repeat;
}
void TestInit(int cropped_width, int cropped_height) {
@@ -136,7 +132,7 @@
TEST_WEBRTCVIDEOFRAME(ConstructBlack)
// TODO(fbarchard): Implement Jpeg
// TEST_WEBRTCVIDEOFRAME(ConstructMjpgI420)
-// TEST_WEBRTCVIDEOFRAME(ConstructMjpgI422)
+TEST_WEBRTCVIDEOFRAME(ConstructMjpgI422)
// TEST_WEBRTCVIDEOFRAME(ConstructMjpgI444)
// TEST_WEBRTCVIDEOFRAME(ConstructMjpgI411)
// TEST_WEBRTCVIDEOFRAME(ConstructMjpgI400)
diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc
index 5cf53e4..a524bad 100644
--- a/media/webrtc/webrtcvoiceengine.cc
+++ b/media/webrtc/webrtcvoiceengine.cc
@@ -52,6 +52,7 @@
#include "webrtc/base/stringutils.h"
#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/video_engine/include/vie_network.h"
#ifdef WIN32
#include <objbase.h> // NOLINT
@@ -119,6 +120,7 @@
// Opus bitrate should be in the range between 6000 and 510000.
static const int kOpusMinBitrate = 6000;
static const int kOpusMaxBitrate = 510000;
+
// Default audio dscp value.
// See http://tools.ietf.org/html/rfc2474 for details.
// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
@@ -403,6 +405,7 @@
return codec.GetParam(kCodecParamStereo, &value) && value == 1;
}
+// TODO(minyue): Clamp bitrate when invalid.
static bool IsValidOpusBitrate(int bitrate) {
return (bitrate >= kOpusMinBitrate && bitrate <= kOpusMaxBitrate);
}
@@ -429,6 +432,59 @@
return codec.GetParam(kCodecParamUseInbandFec, &value) && value == 1;
}
+// Returns kOpusDefaultPlaybackRate if params[kCodecParamMaxPlaybackRate] is not
+// defined. Returns the value of params[kCodecParamMaxPlaybackRate] otherwise.
+static int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
+ int value;
+ if (codec.GetParam(kCodecParamMaxPlaybackRate, &value)) {
+ return value;
+ }
+ return kOpusDefaultMaxPlaybackRate;
+}
+
+static void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
+ bool* enable_codec_fec, int* max_playback_rate) {
+ *enable_codec_fec = IsOpusFecEnabled(codec);
+ *max_playback_rate = GetOpusMaxPlaybackRate(codec);
+
+ // If OPUS, change what we send according to the "stereo" codec
+ // parameter, and not the "channels" parameter. We set
+ // voe_codec.channels to 2 if "stereo=1" and 1 otherwise. If
+ // the bitrate is not specified, i.e. is zero, we set it to the
+ // appropriate default value for mono or stereo Opus.
+
+ // TODO(minyue): The determination of bit rate might take the maximum playback
+ // rate into account.
+
+ if (IsOpusStereoEnabled(codec)) {
+ voe_codec->channels = 2;
+ if (!IsValidOpusBitrate(codec.bitrate)) {
+ if (codec.bitrate != 0) {
+ LOG(LS_WARNING) << "Overrides the invalid supplied bitrate("
+ << codec.bitrate
+ << ") with default opus stereo bitrate: "
+ << kOpusStereoBitrate;
+ }
+ voe_codec->rate = kOpusStereoBitrate;
+ }
+ } else {
+ voe_codec->channels = 1;
+ if (!IsValidOpusBitrate(codec.bitrate)) {
+ if (codec.bitrate != 0) {
+ LOG(LS_WARNING) << "Overrides the invalid supplied bitrate("
+ << codec.bitrate
+ << ") with default opus mono bitrate: "
+ << kOpusMonoBitrate;
+ }
+ voe_codec->rate = kOpusMonoBitrate;
+ }
+ }
+ int bitrate_from_params = GetOpusBitrateFromParams(codec);
+ if (bitrate_from_params != 0) {
+ voe_codec->rate = bitrate_from_params;
+ }
+}
+
void WebRtcVoiceEngine::ConstructCodecs() {
LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
@@ -805,30 +861,6 @@
}
}
- bool experimental_ns;
- if (options.experimental_ns.Get(&experimental_ns)) {
- webrtc::AudioProcessing* audioproc =
- voe_wrapper_->base()->audio_processing();
-#ifdef USE_WEBRTC_DEV_BRANCH
- webrtc::Config config;
- config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(
- experimental_ns));
- audioproc->SetExtraOptions(config);
-#else
- // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
- // returns NULL on audio_processing().
- if (audioproc) {
- if (audioproc->EnableExperimentalNs(experimental_ns) == -1) {
- LOG_RTCERR1(EnableExperimentalNs, experimental_ns);
- return false;
- }
- } else {
- LOG(LS_VERBOSE) << "Experimental noise suppression set to "
- << experimental_ns;
- }
-#endif
- }
-
bool highpass_filter;
if (options.highpass_filter.Get(&highpass_filter)) {
LOG(LS_INFO) << "High pass filter enabled? " << highpass_filter;
@@ -874,20 +906,50 @@
StopAecDump();
}
+ webrtc::Config config;
+
+ experimental_aec_.SetFrom(options.experimental_aec);
bool experimental_aec;
- if (options.experimental_aec.Get(&experimental_aec)) {
- LOG(LS_INFO) << "Experimental aec is " << experimental_aec;
- webrtc::AudioProcessing* audioproc =
- voe_wrapper_->base()->audio_processing();
+ if (experimental_aec_.Get(&experimental_aec)) {
+ LOG(LS_INFO) << "Experimental aec is enabled? " << experimental_aec;
+ config.Set<webrtc::DelayCorrection>(
+ new webrtc::DelayCorrection(experimental_aec));
+ }
+
+#ifdef USE_WEBRTC_DEV_BRANCH
+ experimental_ns_.SetFrom(options.experimental_ns);
+ bool experimental_ns;
+ if (experimental_ns_.Get(&experimental_ns)) {
+ LOG(LS_INFO) << "Experimental ns is enabled? " << experimental_ns;
+ config.Set<webrtc::ExperimentalNs>(
+ new webrtc::ExperimentalNs(experimental_ns));
+ }
+#endif
+
+ // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
+ // returns NULL on audio_processing().
+ webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing();
+ if (audioproc) {
+ audioproc->SetExtraOptions(config);
+ }
+
+#ifndef USE_WEBRTC_DEV_BRANCH
+ bool experimental_ns;
+ if (options.experimental_ns.Get(&experimental_ns)) {
+ LOG(LS_INFO) << "Experimental ns is enabled? " << experimental_ns;
// We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
// returns NULL on audio_processing().
if (audioproc) {
- webrtc::Config config;
- config.Set<webrtc::DelayCorrection>(
- new webrtc::DelayCorrection(experimental_aec));
- audioproc->SetExtraOptions(config);
+ if (audioproc->EnableExperimentalNs(experimental_ns) == -1) {
+ LOG_RTCERR1(EnableExperimentalNs, experimental_ns);
+ return false;
+ }
+ } else {
+ LOG(LS_VERBOSE) << "Experimental noise suppression set to "
+ << experimental_ns;
}
}
+#endif
uint32 recording_sample_rate;
if (options.recording_sample_rate.Get(&recording_sample_rate)) {
@@ -1759,6 +1821,8 @@
typing_noise_detected_(false),
desired_send_(SEND_NOTHING),
send_(SEND_NOTHING),
+ shared_bwe_vie_(NULL),
+ shared_bwe_vie_channel_(-1),
default_receive_ssrc_(0) {
engine->RegisterChannel(this);
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel "
@@ -1770,6 +1834,7 @@
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel "
<< voe_channel();
+ SetupSharedBandwidthEstimation(NULL, -1);
// Remove any remaining send streams, the default channel will be deleted
// later.
@@ -1870,6 +1935,12 @@
}
}
+ // Force update of Video Engine BWE forwarding to reflect experiment setting.
+ if (!SetupSharedBandwidthEstimation(shared_bwe_vie_,
+ shared_bwe_vie_channel_)) {
+ return false;
+ }
+
LOG(LS_INFO) << "Set voice channel options. Current options: "
<< options_.ToString();
return true;
@@ -1976,6 +2047,10 @@
bool nack_enabled = nack_enabled_;
bool enable_codec_fec = false;
+ // max_playback_rate <= 0 will not trigger setting of maximum encoding
+ // bandwidth.
+ int max_playback_rate = 0;
+
// Set send codec (the first non-telephone-event/CN codec)
for (std::vector<AudioCodec>::const_iterator it = codecs.begin();
it != codecs.end(); ++it) {
@@ -1992,40 +2067,6 @@
continue;
}
- // If OPUS, change what we send according to the "stereo" codec
- // parameter, and not the "channels" parameter. We set
- // voe_codec.channels to 2 if "stereo=1" and 1 otherwise. If
- // the bitrate is not specified, i.e. is zero, we set it to the
- // appropriate default value for mono or stereo Opus.
- if (IsOpus(*it)) {
- if (IsOpusStereoEnabled(*it)) {
- voe_codec.channels = 2;
- if (!IsValidOpusBitrate(it->bitrate)) {
- if (it->bitrate != 0) {
- LOG(LS_WARNING) << "Overrides the invalid supplied bitrate("
- << it->bitrate
- << ") with default opus stereo bitrate: "
- << kOpusStereoBitrate;
- }
- voe_codec.rate = kOpusStereoBitrate;
- }
- } else {
- voe_codec.channels = 1;
- if (!IsValidOpusBitrate(it->bitrate)) {
- if (it->bitrate != 0) {
- LOG(LS_WARNING) << "Overrides the invalid supplied bitrate("
- << it->bitrate
- << ") with default opus mono bitrate: "
- << kOpusMonoBitrate;
- }
- voe_codec.rate = kOpusMonoBitrate;
- }
- }
- int bitrate_from_params = GetOpusBitrateFromParams(*it);
- if (bitrate_from_params != 0) {
- voe_codec.rate = bitrate_from_params;
- }
- }
// We'll use the first codec in the list to actually send audio data.
// Be sure to use the payload type requested by the remote side.
@@ -2055,8 +2096,11 @@
} else {
send_codec = voe_codec;
nack_enabled = IsNackEnabled(*it);
- // For Opus as the send codec, we enable inband FEC if requested.
- enable_codec_fec = IsOpus(*it) && IsOpusFecEnabled(*it);
+ // For Opus as the send codec, we are to enable inband FEC if requested
+ // and set maximum playback rate.
+ if (IsOpus(*it)) {
+ GetOpusConfig(*it, &send_codec, &enable_codec_fec, &max_playback_rate);
+ }
}
found_send_codec = true;
break;
@@ -2090,6 +2134,21 @@
#endif // USE_WEBRTC_DEV_BRANCH
}
+ // maxplaybackrate should be set after SetSendCodec.
+ if (max_playback_rate > 0) {
+ LOG(LS_INFO) << "Attempt to set maximum playback rate to "
+ << max_playback_rate
+ << " Hz on channel "
+ << channel;
+#ifdef USE_WEBRTC_DEV_BRANCH
+ // (max_playback_rate + 1) >> 1 is to obtain ceil(max_playback_rate / 2.0).
+ if (engine()->voe()->codec()->SetOpusMaxPlaybackRate(
+ channel, max_playback_rate) == -1) {
+ LOG(LS_WARNING) << "Could not set maximum playback rate.";
+ }
+#endif
+ }
+
// Always update the |send_codec_| to the currently set send codec.
send_codec_.reset(new webrtc::CodecInst(send_codec));
@@ -2531,8 +2590,8 @@
}
if (engine()->voe()->rtp()->SetRTCP_CNAME(channel, sp.cname.c_str()) == -1) {
- LOG_RTCERR2(SetRTCP_CNAME, channel, sp.cname);
- return false;
+ LOG_RTCERR2(SetRTCP_CNAME, channel, sp.cname);
+ return false;
}
// Set the current codecs to be used for the new channel.
@@ -2604,6 +2663,9 @@
receive_channels_.insert(std::make_pair(
default_receive_ssrc_,
new WebRtcVoiceChannelRenderer(voe_channel(), audio_transport)));
+ if (!SetupSharedBweOnChannel(voe_channel())) {
+ return false;
+ }
return SetPlayout(voe_channel(), playout_);
}
@@ -2691,6 +2753,11 @@
return false;
}
+ // Set up channel to be able to forward incoming packets to video engine BWE.
+ if (!SetupSharedBweOnChannel(channel)) {
+ return false;
+ }
+
return SetPlayout(channel, playout_);
}
@@ -3060,7 +3127,8 @@
engine()->voe()->network()->ReceivedRTPPacket(
which_channel,
packet->data(),
- static_cast<unsigned int>(packet->length()));
+ static_cast<unsigned int>(packet->length()),
+ webrtc::PacketTime(packet_time.timestamp, packet_time.not_before));
}
void WebRtcVoiceMediaChannel::OnRtcpReceived(
@@ -3454,6 +3522,23 @@
return -1;
}
+bool WebRtcVoiceMediaChannel::SetupSharedBandwidthEstimation(
+ webrtc::VideoEngine* vie, int vie_channel) {
+ shared_bwe_vie_ = vie;
+ shared_bwe_vie_channel_ = vie_channel;
+
+ if (!SetupSharedBweOnChannel(voe_channel())) {
+ return false;
+ }
+ for (ChannelMap::iterator it = receive_channels_.begin();
+ it != receive_channels_.end(); ++it) {
+ if (!SetupSharedBweOnChannel(it->second->channel())) {
+ return false;
+ }
+ }
+ return true;
+}
+
bool WebRtcVoiceMediaChannel::GetRedSendCodec(const AudioCodec& red_codec,
const std::vector<AudioCodec>& all_codecs, webrtc::CodecInst* send_codec) {
// Get the RED encodings from the parameter with no name. This may
@@ -3603,6 +3688,25 @@
return true;
}
+bool WebRtcVoiceMediaChannel::SetupSharedBweOnChannel(int voe_channel) {
+ webrtc::ViENetwork* vie_network = NULL;
+ int vie_channel = -1;
+ if (options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false) &&
+ shared_bwe_vie_ != NULL && shared_bwe_vie_channel_ != -1) {
+ vie_network = webrtc::ViENetwork::GetInterface(shared_bwe_vie_);
+ vie_channel = shared_bwe_vie_channel_;
+ }
+ if (engine()->voe()->rtp()->SetVideoEngineBWETarget(voe_channel, vie_network,
+ vie_channel) == -1) {
+ LOG_RTCERR3(SetVideoEngineBWETarget, voe_channel, vie_network, vie_channel);
+ if (vie_network != NULL) {
+ // Don't fail if we're tearing down.
+ return false;
+ }
+ }
+ return true;
+}
+
int WebRtcSoundclipStream::Read(void *buf, int len) {
size_t res = 0;
mem_.Read(buf, len, &res, NULL);
diff --git a/media/webrtc/webrtcvoiceengine.h b/media/webrtc/webrtcvoiceengine.h
index 69705cc..5557af0 100644
--- a/media/webrtc/webrtcvoiceengine.h
+++ b/media/webrtc/webrtcvoiceengine.h
@@ -54,6 +54,10 @@
#error "Bogus include."
#endif
+namespace webrtc {
+class VideoEngine;
+}
+
namespace cricket {
// WebRtcSoundclipStream is an adapter object that allows a memory stream to be
@@ -280,6 +284,13 @@
uint32 rx_processor_ssrc_;
rtc::CriticalSection signal_media_critical_;
+
+ // Cache received experimental_aec and experimental_ns values, and apply them
+ // in case they are missing in the audio options. We need to do this because
+ // SetExtraOptions() will revert to defaults for options which are not
+ // provided.
+ Settable<bool> experimental_aec_;
+ Settable<bool> experimental_ns_;
};
// WebRtcMediaChannel is a class that implements the common WebRtc channel
@@ -377,6 +388,8 @@
int GetReceiveChannelNum(uint32 ssrc);
int GetSendChannelNum(uint32 ssrc);
+ bool SetupSharedBandwidthEstimation(webrtc::VideoEngine* vie,
+ int vie_channel);
protected:
int GetLastEngineError() { return engine()->GetLastEngineError(); }
int GetOutputLevel(int channel);
@@ -419,6 +432,7 @@
bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
const RtpHeaderExtension* extension);
+ bool SetupSharedBweOnChannel(int voe_channel);
bool SetChannelRecvRtpHeaderExtensions(
int channel_id,
@@ -442,6 +456,11 @@
bool typing_noise_detected_;
SendFlags desired_send_;
SendFlags send_;
+ // shared_bwe_vie_ and shared_bwe_vie_channel_ together identifies a WebRTC
+ // VideoEngine channel that this voice channel should forward incoming packets
+ // to for Bandwidth Estimation purposes.
+ webrtc::VideoEngine* shared_bwe_vie_;
+ int shared_bwe_vie_channel_;
// send_channels_ contains the channels which are being used for sending.
// When the default channel (voe_channel) is used for sending, it is
diff --git a/media/webrtc/webrtcvoiceengine_unittest.cc b/media/webrtc/webrtcvoiceengine_unittest.cc
index 89d4c4d..b044e92 100644
--- a/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -38,6 +38,7 @@
#include "talk/media/base/fakenetworkinterface.h"
#include "talk/media/base/fakertp.h"
#include "talk/media/webrtc/fakewebrtcvoiceengine.h"
+#include "talk/media/webrtc/webrtcvie.h"
#include "talk/media/webrtc/webrtcvoiceengine.h"
#include "talk/p2p/base/fakesession.h"
#include "talk/session/media/channel.h"
@@ -1151,7 +1152,6 @@
int channel_num = voe_.GetLastChannel();
std::vector<cricket::AudioCodec> codecs;
codecs.push_back(kOpusCodec);
- codecs[0].bitrate = 0;
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
EXPECT_FALSE(voe_.GetCodecFEC(channel_num));
}
@@ -1228,6 +1228,159 @@
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
EXPECT_FALSE(voe_.GetCodecFEC(channel_num));
}
+
+// Test that Opus FEC status can be changed.
+TEST_F(WebRtcVoiceEngineTestFake, ChangeOpusFecStatus) {
+ EXPECT_TRUE(SetupEngine());
+ int channel_num = voe_.GetLastChannel();
+ std::vector<cricket::AudioCodec> codecs;
+ codecs.push_back(kOpusCodec);
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_FALSE(voe_.GetCodecFEC(channel_num));
+ codecs[0].params["useinbandfec"] = "1";
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_TRUE(voe_.GetCodecFEC(channel_num));
+}
+
+// Test maxplaybackrate <= 8000 triggers Opus narrow band mode.
+TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateNb) {
+ EXPECT_TRUE(SetupEngine());
+ int channel_num = voe_.GetLastChannel();
+ std::vector<cricket::AudioCodec> codecs;
+ codecs.push_back(kOpusCodec);
+ codecs[0].bitrate = 0;
+ codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 8000);
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(cricket::kOpusBandwidthNb,
+ voe_.GetMaxEncodingBandwidth(channel_num));
+ webrtc::CodecInst gcodec;
+ EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+ EXPECT_STREQ("opus", gcodec.plname);
+ // TODO(minyue): Default bit rate is not but can in future be affected by
+ // kCodecParamMaxPlaybackRate.
+ EXPECT_EQ(32000, gcodec.rate);
+}
+
+// Test 8000 < maxplaybackrate <= 12000 triggers Opus medium band mode.
+TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateMb) {
+ EXPECT_TRUE(SetupEngine());
+ int channel_num = voe_.GetLastChannel();
+ std::vector<cricket::AudioCodec> codecs;
+ codecs.push_back(kOpusCodec);
+ codecs[0].bitrate = 0;
+ codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 8001);
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(cricket::kOpusBandwidthMb,
+ voe_.GetMaxEncodingBandwidth(channel_num));
+ webrtc::CodecInst gcodec;
+ EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+ EXPECT_STREQ("opus", gcodec.plname);
+ // TODO(minyue): Default bit rate is not but can in future be affected by
+ // kCodecParamMaxPlaybackRate.
+ EXPECT_EQ(32000, gcodec.rate);
+}
+
+// Test 12000 < maxplaybackrate <= 16000 triggers Opus wide band mode.
+TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateWb) {
+ EXPECT_TRUE(SetupEngine());
+ int channel_num = voe_.GetLastChannel();
+ std::vector<cricket::AudioCodec> codecs;
+ codecs.push_back(kOpusCodec);
+ codecs[0].bitrate = 0;
+ codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 12001);
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(cricket::kOpusBandwidthWb,
+ voe_.GetMaxEncodingBandwidth(channel_num));
+ webrtc::CodecInst gcodec;
+ EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+ EXPECT_STREQ("opus", gcodec.plname);
+ // TODO(minyue): Default bit rate is not but can in future be affected by
+ // kCodecParamMaxPlaybackRate.
+ EXPECT_EQ(32000, gcodec.rate);
+}
+
+// Test 16000 < maxplaybackrate <= 24000 triggers Opus super wide band mode.
+TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateSwb) {
+ EXPECT_TRUE(SetupEngine());
+ int channel_num = voe_.GetLastChannel();
+ std::vector<cricket::AudioCodec> codecs;
+ codecs.push_back(kOpusCodec);
+ codecs[0].bitrate = 0;
+ codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 16001);
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(cricket::kOpusBandwidthSwb,
+ voe_.GetMaxEncodingBandwidth(channel_num));
+ webrtc::CodecInst gcodec;
+ EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+ EXPECT_STREQ("opus", gcodec.plname);
+ // TODO(minyue): Default bit rate is not but can in future be affected by
+ // kCodecParamMaxPlaybackRate.
+ EXPECT_EQ(32000, gcodec.rate);
+}
+
+// Test 24000 < maxplaybackrate triggers Opus full band mode.
+TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateFb) {
+ EXPECT_TRUE(SetupEngine());
+ int channel_num = voe_.GetLastChannel();
+ std::vector<cricket::AudioCodec> codecs;
+ codecs.push_back(kOpusCodec);
+ codecs[0].bitrate = 0;
+ codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 24001);
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(cricket::kOpusBandwidthFb,
+ voe_.GetMaxEncodingBandwidth(channel_num));
+ webrtc::CodecInst gcodec;
+ EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+ EXPECT_STREQ("opus", gcodec.plname);
+ // TODO(minyue): Default bit rate is not but can in future be affected by
+ // kCodecParamMaxPlaybackRate.
+ EXPECT_EQ(32000, gcodec.rate);
+}
+
+// Test Opus that without maxplaybackrate, default playback rate is used.
+TEST_F(WebRtcVoiceEngineTestFake, DefaultOpusMaxPlaybackRate) {
+ EXPECT_TRUE(SetupEngine());
+ int channel_num = voe_.GetLastChannel();
+ std::vector<cricket::AudioCodec> codecs;
+ codecs.push_back(kOpusCodec);
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(cricket::kOpusBandwidthFb,
+ voe_.GetMaxEncodingBandwidth(channel_num));
+}
+
+// Test the with non-Opus, maxplaybackrate has no effect.
+TEST_F(WebRtcVoiceEngineTestFake, SetNonOpusMaxPlaybackRate) {
+ EXPECT_TRUE(SetupEngine());
+ int channel_num = voe_.GetLastChannel();
+ std::vector<cricket::AudioCodec> codecs;
+ codecs.push_back(kIsacCodec);
+ codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 32000);
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(0, voe_.GetMaxEncodingBandwidth(channel_num));
+}
+
+// Test maxplaybackrate can be set on two streams.
+TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateOnTwoStreams) {
+ EXPECT_TRUE(SetupEngine());
+ int channel_num = voe_.GetLastChannel();
+ std::vector<cricket::AudioCodec> codecs;
+ codecs.push_back(kOpusCodec);
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ // Default bandwidth is 24000.
+ EXPECT_EQ(cricket::kOpusBandwidthFb,
+ voe_.GetMaxEncodingBandwidth(channel_num));
+
+ codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 8000);
+
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(cricket::kOpusBandwidthNb,
+ voe_.GetMaxEncodingBandwidth(channel_num));
+
+ channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc2));
+ channel_num = voe_.GetLastChannel();
+ EXPECT_EQ(cricket::kOpusBandwidthNb,
+ voe_.GetMaxEncodingBandwidth(channel_num));
+}
#endif // USE_WEBRTC_DEV_BRANCH
// Test that we can apply CELT with stereo mode but fail with mono mode.
@@ -3160,3 +3313,113 @@
CoUninitialize();
}
#endif
+
+TEST_F(WebRtcVoiceEngineTestFake, ChangeCombinedAudioVideoBweOption) {
+ // Test that changing the combined_audio_video_bwe option results in the
+ // expected state changes in VoiceEngine.
+ cricket::ViEWrapper vie;
+ const int kVieCh = 667;
+
+ EXPECT_TRUE(SetupEngine());
+ cricket::WebRtcVoiceMediaChannel* media_channel =
+ static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
+ EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(vie.engine(),
+ kVieCh));
+ EXPECT_TRUE(media_channel->AddRecvStream(
+ cricket::StreamParams::CreateLegacy(2)));
+ int recv_ch = voe_.GetLastChannel();
+
+ // Combined BWE should not be set up yet.
+ EXPECT_EQ(NULL, voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(-1, voe_.GetVideoChannel(recv_ch));
+
+ // Enable combined BWE option - now it should be set up.
+ cricket::AudioOptions options;
+ options.combined_audio_video_bwe.Set(true);
+ EXPECT_TRUE(media_channel->SetOptions(options));
+ EXPECT_EQ(vie.network(), voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(kVieCh, voe_.GetVideoChannel(recv_ch));
+
+ // Disable combined BWE option - should be disabled again.
+ options.combined_audio_video_bwe.Set(false);
+ EXPECT_TRUE(media_channel->SetOptions(options));
+ EXPECT_EQ(NULL, voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(-1, voe_.GetVideoChannel(recv_ch));
+
+ EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(NULL, -1));
+}
+
+TEST_F(WebRtcVoiceEngineTestFake, SetupSharedBandwidthEstimation) {
+ // Test that calling SetupSharedBandwidthEstimation() on the voice media
+ // channel results in the expected state changes in VoiceEngine.
+ cricket::ViEWrapper vie1;
+ cricket::ViEWrapper vie2;
+ const int kVieCh1 = 667;
+ const int kVieCh2 = 70;
+
+ EXPECT_TRUE(SetupEngine());
+ cricket::WebRtcVoiceMediaChannel* media_channel =
+ static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
+ cricket::AudioOptions options;
+ options.combined_audio_video_bwe.Set(true);
+ EXPECT_TRUE(media_channel->SetOptions(options));
+ EXPECT_TRUE(media_channel->AddRecvStream(
+ cricket::StreamParams::CreateLegacy(2)));
+ int recv_ch = voe_.GetLastChannel();
+
+ // Combined BWE should not be set up yet.
+ EXPECT_EQ(NULL, voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(-1, voe_.GetVideoChannel(recv_ch));
+
+ // Register - should be enabled.
+ EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(vie1.engine(),
+ kVieCh1));
+ EXPECT_EQ(vie1.network(), voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(kVieCh1, voe_.GetVideoChannel(recv_ch));
+
+ // Re-register - should still be enabled.
+ EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(vie2.engine(),
+ kVieCh2));
+ EXPECT_EQ(vie2.network(), voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(kVieCh2, voe_.GetVideoChannel(recv_ch));
+
+ // Unregister - should be disabled again.
+ EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(NULL, -1));
+ EXPECT_EQ(NULL, voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(-1, voe_.GetVideoChannel(recv_ch));
+}
+
+TEST_F(WebRtcVoiceEngineTestFake, ConfigureCombinedBweForNewRecvStreams) {
+ // Test that adding receive streams after enabling combined bandwidth
+ // estimation will correctly configure each channel.
+ cricket::ViEWrapper vie;
+ const int kVieCh = 667;
+
+ EXPECT_TRUE(SetupEngine());
+ cricket::WebRtcVoiceMediaChannel* media_channel =
+ static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
+ EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(vie.engine(),
+ kVieCh));
+ cricket::AudioOptions options;
+ options.combined_audio_video_bwe.Set(true);
+ EXPECT_TRUE(media_channel->SetOptions(options));
+
+ static const uint32 kSsrcs[] = {1, 2, 3, 4};
+ int voe_channels[ARRAY_SIZE(kSsrcs)] = {0};
+ for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
+ EXPECT_TRUE(media_channel->AddRecvStream(
+ cricket::StreamParams::CreateLegacy(kSsrcs[i])));
+ int recv_ch = media_channel->GetReceiveChannelNum(kSsrcs[i]);
+ EXPECT_NE(-1, recv_ch);
+ voe_channels[i] = recv_ch;
+ EXPECT_EQ(vie.network(), voe_.GetViENetwork(recv_ch));
+ EXPECT_EQ(kVieCh, voe_.GetVideoChannel(recv_ch));
+ }
+
+ EXPECT_TRUE(media_channel->SetupSharedBandwidthEstimation(NULL, -1));
+
+ for (unsigned int i = 0; i < ARRAY_SIZE(voe_channels); ++i) {
+ EXPECT_EQ(NULL, voe_.GetViENetwork(voe_channels[i]));
+ EXPECT_EQ(-1, voe_.GetVideoChannel(voe_channels[i]));
+ }
+}
diff --git a/p2p/base/asyncstuntcpsocket.h b/p2p/base/asyncstuntcpsocket.h
index b63c0b5..136b4df 100644
--- a/p2p/base/asyncstuntcpsocket.h
+++ b/p2p/base/asyncstuntcpsocket.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_BASE_ASYNCSTUNTCPSOCKET_H_
-#define TALK_BASE_ASYNCSTUNTCPSOCKET_H_
+#ifndef TALK_P2P_BASE_ASYNCSTUNTCPSOCKET_H_
+#define TALK_P2P_BASE_ASYNCSTUNTCPSOCKET_H_
#include "webrtc/base/asynctcpsocket.h"
#include "webrtc/base/scoped_ptr.h"
@@ -64,4 +64,4 @@
} // namespace cricket
-#endif // TALK_BASE_ASYNCSTUNTCPSOCKET_H_
+#endif // TALK_P2P_BASE_ASYNCSTUNTCPSOCKET_H_
diff --git a/p2p/base/basicpacketsocketfactory.h b/p2p/base/basicpacketsocketfactory.h
index b1bae35..77b1652 100644
--- a/p2p/base/basicpacketsocketfactory.h
+++ b/p2p/base/basicpacketsocketfactory.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_BASE_BASICPACKETSOCKETFACTORY_H_
-#define TALK_BASE_BASICPACKETSOCKETFACTORY_H_
+#ifndef TALK_P2P_BASE_BASICPACKETSOCKETFACTORY_H_
+#define TALK_P2P_BASE_BASICPACKETSOCKETFACTORY_H_
#include "talk/p2p/base/packetsocketfactory.h"
@@ -65,4 +65,4 @@
} // namespace rtc
-#endif // TALK_BASE_BASICPACKETSOCKETFACTORY_H_
+#endif // TALK_P2P_BASE_BASICPACKETSOCKETFACTORY_H_
diff --git a/p2p/base/constants.cc b/p2p/base/constants.cc
index 2e57d9d..278a615 100644
--- a/p2p/base/constants.cc
+++ b/p2p/base/constants.cc
@@ -29,7 +29,7 @@
#include <string>
-#include "talk/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/qname.h"
namespace cricket {
diff --git a/p2p/base/constants.h b/p2p/base/constants.h
index 61dd815..4cd1166 100644
--- a/p2p/base/constants.h
+++ b/p2p/base/constants.h
@@ -29,7 +29,7 @@
#define TALK_P2P_BASE_CONSTANTS_H_
#include <string>
-#include "talk/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/qname.h"
// This file contains constants related to signaling that are used in various
// classes in this directory.
diff --git a/p2p/base/p2ptransport.cc b/p2p/base/p2ptransport.cc
index b992cc0..06941ac 100644
--- a/p2p/base/p2ptransport.cc
+++ b/p2p/base/p2ptransport.cc
@@ -35,8 +35,8 @@
#include "talk/p2p/base/parsing.h"
#include "talk/p2p/base/sessionmanager.h"
#include "talk/p2p/base/sessionmessages.h"
-#include "talk/xmllite/qname.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/common.h"
diff --git a/p2p/base/packetsocketfactory.h b/p2p/base/packetsocketfactory.h
index 6b82682..46767c2 100644
--- a/p2p/base/packetsocketfactory.h
+++ b/p2p/base/packetsocketfactory.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_BASE_PACKETSOCKETFACTORY_H_
-#define TALK_BASE_PACKETSOCKETFACTORY_H_
+#ifndef TALK_P2P_BASE_PACKETSOCKETFACTORY_H_
+#define TALK_P2P_BASE_PACKETSOCKETFACTORY_H_
#include "webrtc/base/proxyinfo.h"
@@ -66,4 +66,4 @@
} // namespace rtc
-#endif // TALK_BASE_PACKETSOCKETFACTORY_H_
+#endif // TALK_P2P_BASE_PACKETSOCKETFACTORY_H_
diff --git a/p2p/base/parsing.h b/p2p/base/parsing.h
index 635b122..2aab2f8 100644
--- a/p2p/base/parsing.h
+++ b/p2p/base/parsing.h
@@ -30,7 +30,7 @@
#include <string>
#include <vector>
-#include "talk/xmllite/xmlelement.h" // Needed to delete ParseError.extra.
+#include "webrtc/libjingle/xmllite/xmlelement.h" // Needed to delete ParseError.extra.
#include "webrtc/base/basictypes.h"
#include "webrtc/base/stringencode.h"
diff --git a/p2p/base/port.cc b/p2p/base/port.cc
index 3820154..0cf46eb 100644
--- a/p2p/base/port.cc
+++ b/p2p/base/port.cc
@@ -1212,7 +1212,7 @@
<< ":" << local.type() << ":" << local.protocol()
<< ":" << local.address().ToSensitiveString()
<< "->" << remote.id() << ":" << remote.component()
- << ":" << remote.preference()
+ << ":" << remote.priority()
<< ":" << remote.type() << ":"
<< remote.protocol() << ":" << remote.address().ToSensitiveString() << "|"
<< CONNECT_STATE_ABBREV[connected()]
diff --git a/p2p/base/port.h b/p2p/base/port.h
index cccfdad..4893586 100644
--- a/p2p/base/port.h
+++ b/p2p/base/port.h
@@ -155,6 +155,7 @@
uint64 IceTiebreaker() const { return tiebreaker_; }
virtual bool SharedSocket() const { return shared_socket_; }
+ void ResetSharedSocket() { shared_socket_ = false; }
// The thread on which this port performs its I/O.
rtc::Thread* thread() { return thread_; }
diff --git a/p2p/base/portallocator.h b/p2p/base/portallocator.h
index 84e5fea..5bc389e 100644
--- a/p2p/base/portallocator.h
+++ b/p2p/base/portallocator.h
@@ -44,17 +44,18 @@
// Clients can override this class to control port allocation, including
// what kinds of ports are allocated.
-const uint32 PORTALLOCATOR_DISABLE_UDP = 0x01;
-const uint32 PORTALLOCATOR_DISABLE_STUN = 0x02;
-const uint32 PORTALLOCATOR_DISABLE_RELAY = 0x04;
-const uint32 PORTALLOCATOR_DISABLE_TCP = 0x08;
-const uint32 PORTALLOCATOR_ENABLE_SHAKER = 0x10;
-const uint32 PORTALLOCATOR_ENABLE_BUNDLE = 0x20;
-const uint32 PORTALLOCATOR_ENABLE_IPV6 = 0x40;
-const uint32 PORTALLOCATOR_ENABLE_SHARED_UFRAG = 0x80;
-const uint32 PORTALLOCATOR_ENABLE_SHARED_SOCKET = 0x100;
-const uint32 PORTALLOCATOR_ENABLE_STUN_RETRANSMIT_ATTRIBUTE = 0x200;
-const uint32 PORTALLOCATOR_ENABLE_TURN_SHARED_SOCKET = 0x400;
+enum {
+ PORTALLOCATOR_DISABLE_UDP = 0x01,
+ PORTALLOCATOR_DISABLE_STUN = 0x02,
+ PORTALLOCATOR_DISABLE_RELAY = 0x04,
+ PORTALLOCATOR_DISABLE_TCP = 0x08,
+ PORTALLOCATOR_ENABLE_SHAKER = 0x10,
+ PORTALLOCATOR_ENABLE_BUNDLE = 0x20,
+ PORTALLOCATOR_ENABLE_IPV6 = 0x40,
+ PORTALLOCATOR_ENABLE_SHARED_UFRAG = 0x80,
+ PORTALLOCATOR_ENABLE_SHARED_SOCKET = 0x100,
+ PORTALLOCATOR_ENABLE_STUN_RETRANSMIT_ATTRIBUTE = 0x200,
+};
const uint32 kDefaultPortAllocatorFlags = 0;
@@ -63,6 +64,15 @@
// internal. Less than 20ms is not acceptable. We choose 50ms as our default.
const uint32 kMinimumStepDelay = 50;
+// CF = CANDIDATE FILTER
+enum {
+ CF_NONE = 0x0,
+ CF_HOST = 0x1,
+ CF_REFLEXIVE = 0x2,
+ CF_RELAY = 0x4,
+ CF_ALL = 0x7,
+};
+
class PortAllocatorSessionMuxer;
class PortAllocatorSession : public sigslot::has_slots<> {
@@ -118,7 +128,8 @@
min_port_(0),
max_port_(0),
step_delay_(kDefaultStepDelay),
- allow_tcp_listen_(true) {
+ allow_tcp_listen_(true),
+ candidate_filter_(CF_ALL) {
// This will allow us to have old behavior on non webrtc clients.
}
virtual ~PortAllocator();
@@ -158,7 +169,6 @@
uint32 step_delay() const { return step_delay_; }
void set_step_delay(uint32 delay) {
- ASSERT(delay >= kMinimumStepDelay);
step_delay_ = delay;
}
@@ -167,6 +177,13 @@
allow_tcp_listen_ = allow_tcp_listen;
}
+ uint32 candidate_filter() { return candidate_filter_; }
+ bool set_candidate_filter(uint32 filter) {
+ // TODO(mallinath) - Do transition check?
+ candidate_filter_ = filter;
+ return true;
+ }
+
protected:
virtual PortAllocatorSession* CreateSessionInternal(
const std::string& content_name,
@@ -184,6 +201,7 @@
uint32 step_delay_;
SessionMuxerMap muxers_;
bool allow_tcp_listen_;
+ uint32 candidate_filter_;
};
} // namespace cricket
diff --git a/p2p/base/rawtransport.cc b/p2p/base/rawtransport.cc
index 5913177..2af1864 100644
--- a/p2p/base/rawtransport.cc
+++ b/p2p/base/rawtransport.cc
@@ -32,8 +32,8 @@
#include "talk/p2p/base/rawtransport.h"
#include "talk/p2p/base/rawtransportchannel.h"
#include "talk/p2p/base/sessionmanager.h"
-#include "talk/xmllite/qname.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "webrtc/base/common.h"
diff --git a/p2p/base/rawtransportchannel.cc b/p2p/base/rawtransportchannel.cc
index ef0a532..ae268e3 100644
--- a/p2p/base/rawtransportchannel.cc
+++ b/p2p/base/rawtransportchannel.cc
@@ -36,8 +36,8 @@
#include "talk/p2p/base/relayport.h"
#include "talk/p2p/base/sessionmanager.h"
#include "talk/p2p/base/stunport.h"
-#include "talk/xmllite/qname.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "webrtc/base/common.h"
diff --git a/p2p/base/session.h b/p2p/base/session.h
index 756bbe0..e06cf00 100644
--- a/p2p/base/session.h
+++ b/p2p/base/session.h
@@ -39,7 +39,7 @@
#include "talk/p2p/base/sessionmanager.h"
#include "talk/p2p/base/sessionmessages.h"
#include "talk/p2p/base/transport.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ptr.h"
diff --git a/p2p/base/session_unittest.cc b/p2p/base/session_unittest.cc
index 90cd06d..4674d2c 100644
--- a/p2p/base/session_unittest.cc
+++ b/p2p/base/session_unittest.cc
@@ -601,8 +601,7 @@
network_("network", "unittest",
rtc::IPAddress(INADDR_LOOPBACK), 8),
socket_factory_(rtc::Thread::Current()),
- running_(false),
- port_(28653) {
+ running_(false) {
network_.AddIP(address_.ipaddr());
}
@@ -655,7 +654,6 @@
rtc::Network network_;
rtc::BasicPacketSocketFactory socket_factory_;
bool running_;
- int port_;
};
class TestPortAllocator : public cricket::PortAllocator {
diff --git a/p2p/base/sessiondescription.cc b/p2p/base/sessiondescription.cc
index 7009aa8..7ad3d48 100644
--- a/p2p/base/sessiondescription.cc
+++ b/p2p/base/sessiondescription.cc
@@ -27,7 +27,7 @@
#include "talk/p2p/base/sessiondescription.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
namespace cricket {
diff --git a/p2p/base/sessionmessages.cc b/p2p/base/sessionmessages.cc
index bfba273..b2fd9d6 100644
--- a/p2p/base/sessionmessages.cc
+++ b/p2p/base/sessionmessages.cc
@@ -36,7 +36,7 @@
#include "talk/p2p/base/sessionclient.h"
#include "talk/p2p/base/sessiondescription.h"
#include "talk/p2p/base/transport.h"
-#include "talk/xmllite/xmlconstants.h"
+#include "webrtc/libjingle/xmllite/xmlconstants.h"
#include "talk/xmpp/constants.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
diff --git a/p2p/base/sessionmessages.h b/p2p/base/sessionmessages.h
index 38fca98..7e1f8ac 100644
--- a/p2p/base/sessionmessages.h
+++ b/p2p/base/sessionmessages.h
@@ -36,7 +36,7 @@
#include "talk/p2p/base/parsing.h"
#include "talk/p2p/base/sessiondescription.h" // Needed to delete contents.
#include "talk/p2p/base/transportinfo.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "webrtc/base/basictypes.h"
namespace cricket {
diff --git a/p2p/base/stun.cc b/p2p/base/stun.cc
index be96b76..061fd9a 100644
--- a/p2p/base/stun.cc
+++ b/p2p/base/stun.cc
@@ -41,6 +41,7 @@
namespace cricket {
+const char STUN_ERROR_REASON_TRY_ALTERNATE_SERVER[] = "Try Alternate Server";
const char STUN_ERROR_REASON_BAD_REQUEST[] = "Bad Request";
const char STUN_ERROR_REASON_UNAUTHORIZED[] = "Unauthorized";
const char STUN_ERROR_REASON_FORBIDDEN[] = "Forbidden";
@@ -401,7 +402,7 @@
case STUN_ATTR_NONCE: return STUN_VALUE_BYTE_STRING;
case STUN_ATTR_XOR_MAPPED_ADDRESS: return STUN_VALUE_XOR_ADDRESS;
case STUN_ATTR_SOFTWARE: return STUN_VALUE_BYTE_STRING;
- case STUN_ATTR_ALTERNATE_SERVER: return STUN_VALUE_BYTE_STRING;
+ case STUN_ATTR_ALTERNATE_SERVER: return STUN_VALUE_ADDRESS;
case STUN_ATTR_FINGERPRINT: return STUN_VALUE_UINT32;
case STUN_ATTR_RETRANSMIT_COUNT: return STUN_VALUE_UINT32;
default: return STUN_VALUE_UNKNOWN;
diff --git a/p2p/base/stun.h b/p2p/base/stun.h
index b22b51e..c4f522b 100644
--- a/p2p/base/stun.h
+++ b/p2p/base/stun.h
@@ -63,7 +63,7 @@
STUN_ATTR_NONCE = 0x0015, // ByteString
STUN_ATTR_XOR_MAPPED_ADDRESS = 0x0020, // XorAddress
STUN_ATTR_SOFTWARE = 0x8022, // ByteString
- STUN_ATTR_ALTERNATE_SERVER = 0x8023, // ByteString
+ STUN_ATTR_ALTERNATE_SERVER = 0x8023, // Address
STUN_ATTR_FINGERPRINT = 0x8028, // UInt32
STUN_ATTR_RETRANSMIT_COUNT = 0xFF00 // UInt32
};
@@ -104,6 +104,7 @@
};
// Strings for the error codes above.
+extern const char STUN_ERROR_REASON_TRY_ALTERNATE_SERVER[];
extern const char STUN_ERROR_REASON_BAD_REQUEST[];
extern const char STUN_ERROR_REASON_UNAUTHORIZED[];
extern const char STUN_ERROR_REASON_UNKNOWN_ATTRIBUTE[];
diff --git a/p2p/base/stunport.h b/p2p/base/stunport.h
index 0a49b67..b3b6d5b 100644
--- a/p2p/base/stunport.h
+++ b/p2p/base/stunport.h
@@ -82,7 +82,7 @@
return socket_->GetLocalAddress();
}
- const ServerAddresses server_addresses() const {
+ const ServerAddresses& server_addresses() const {
return server_addresses_;
}
void
diff --git a/p2p/base/testturnserver.h b/p2p/base/testturnserver.h
index e2c0ccb..6c30afe 100644
--- a/p2p/base/testturnserver.h
+++ b/p2p/base/testturnserver.h
@@ -29,6 +29,7 @@
#define TALK_P2P_BASE_TESTTURNSERVER_H_
#include <string>
+#include <vector>
#include "talk/p2p/base/basicpacketsocketfactory.h"
#include "talk/p2p/base/stun.h"
@@ -41,6 +42,27 @@
static const char kTestRealm[] = "example.org";
static const char kTestSoftware[] = "TestTurnServer";
+class TestTurnRedirector : public TurnRedirectInterface {
+ public:
+ explicit TestTurnRedirector(const std::vector<rtc::SocketAddress>& addresses)
+ : alternate_server_addresses_(addresses),
+ iter_(alternate_server_addresses_.begin()) {
+ }
+
+ virtual bool ShouldRedirect(const rtc::SocketAddress&,
+ rtc::SocketAddress* out) {
+ if (!out || iter_ == alternate_server_addresses_.end()) {
+ return false;
+ }
+ *out = *iter_++;
+ return true;
+ }
+
+ private:
+ const std::vector<rtc::SocketAddress>& alternate_server_addresses_;
+ std::vector<rtc::SocketAddress>::const_iterator iter_;
+};
+
class TestTurnServer : public TurnAuthInterface {
public:
TestTurnServer(rtc::Thread* thread,
@@ -61,6 +83,10 @@
TurnServer* server() { return &server_; }
+ void set_redirect_hook(TurnRedirectInterface* redirect_hook) {
+ server_.set_redirect_hook(redirect_hook);
+ }
+
void AddInternalSocket(const rtc::SocketAddress& int_addr,
ProtocolType proto) {
rtc::Thread* thread = rtc::Thread::Current();
diff --git a/p2p/base/transport.cc b/p2p/base/transport.cc
index 6ee7b2a..d88f5e7 100644
--- a/p2p/base/transport.cc
+++ b/p2p/base/transport.cc
@@ -33,7 +33,7 @@
#include "talk/p2p/base/port.h"
#include "talk/p2p/base/sessionmanager.h"
#include "talk/p2p/base/transportchannelimpl.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/common.h"
diff --git a/p2p/base/transport_unittest.cc b/p2p/base/transport_unittest.cc
index fa2c116..8f7dae2 100644
--- a/p2p/base/transport_unittest.cc
+++ b/p2p/base/transport_unittest.cc
@@ -31,7 +31,7 @@
#include "talk/p2p/base/parsing.h"
#include "talk/p2p/base/rawtransport.h"
#include "talk/p2p/base/sessionmessages.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "webrtc/base/fakesslidentity.h"
#include "webrtc/base/gunit.h"
diff --git a/p2p/base/turnport.cc b/p2p/base/turnport.cc
index 3faacd1..2908d71 100644
--- a/p2p/base/turnport.cc
+++ b/p2p/base/turnport.cc
@@ -51,6 +51,10 @@
static const size_t TURN_CHANNEL_HEADER_SIZE = 4U;
+// Retry at most twice (i.e. three different ALLOCATE requests) on
+// STUN_ERROR_ALLOCATION_MISMATCH error per rfc5766.
+static const size_t MAX_ALLOCATE_MISMATCH_RETRIES = 2;
+
inline bool IsTurnChannelData(uint16 msg_type) {
return ((msg_type & 0xC000) == 0x4000); // MSB are 0b01
}
@@ -78,6 +82,7 @@
private:
// Handles authentication challenge from the server.
void OnAuthChallenge(StunMessage* response, int code);
+ void OnTryAlternate(StunMessage* response, int code);
void OnUnknownAttribute(StunMessage* response);
TurnPort* port_;
@@ -187,7 +192,8 @@
request_manager_(thread),
next_channel_number_(TURN_CHANNEL_NUMBER_START),
connected_(false),
- server_priority_(server_priority) {
+ server_priority_(server_priority),
+ allocate_mismatch_retries_(0) {
request_manager_.SignalSendPacket.connect(this, &TurnPort::OnSendStunPacket);
}
@@ -211,7 +217,8 @@
request_manager_(thread),
next_channel_number_(TURN_CHANNEL_NUMBER_START),
connected_(false),
- server_priority_(server_priority) {
+ server_priority_(server_priority),
+ allocate_mismatch_retries_(0) {
request_manager_.SignalSendPacket.connect(this, &TurnPort::OnSendStunPacket);
}
@@ -253,6 +260,9 @@
return;
}
+ // Insert the current address to prevent redirection pingpong.
+ attempted_server_addresses_.insert(server_address_.address);
+
LOG_J(LS_INFO, this) << "Trying to connect to TURN server via "
<< ProtoToString(server_address_.proto) << " @ "
<< server_address_.address.ToSensitiveString();
@@ -267,6 +277,8 @@
}
bool TurnPort::CreateTurnClientSocket() {
+ ASSERT(!socket_ || SharedSocket());
+
if (server_address_.proto == PROTO_UDP && !SharedSocket()) {
socket_ = socket_factory()->CreateUdpSocket(
rtc::SocketAddress(ip(), 0), min_port(), max_port());
@@ -336,6 +348,29 @@
}
}
+void TurnPort::OnAllocateMismatch() {
+ if (allocate_mismatch_retries_ >= MAX_ALLOCATE_MISMATCH_RETRIES) {
+ LOG_J(LS_WARNING, this) << "Giving up on the port after "
+ << allocate_mismatch_retries_
+ << " retries for STUN_ERROR_ALLOCATION_MISMATCH";
+ OnAllocateError();
+ return;
+ }
+
+ LOG_J(LS_INFO, this) << "Allocating a new socket after "
+ << "STUN_ERROR_ALLOCATION_MISMATCH, retry = "
+ << allocate_mismatch_retries_ + 1;
+ if (SharedSocket()) {
+ ResetSharedSocket();
+ } else {
+ delete socket_;
+ }
+ socket_ = NULL;
+
+ PrepareAddress();
+ ++allocate_mismatch_retries_;
+}
+
Connection* TurnPort::CreateConnection(const Candidate& address,
CandidateOrigin origin) {
// TURN-UDP can only connect to UDP candidates.
@@ -458,6 +493,38 @@
}
}
+
+// Update current server address port with the alternate server address port.
+bool TurnPort::SetAlternateServer(const rtc::SocketAddress& address) {
+ // Check if we have seen this address before and reject if we did.
+ AttemptedServerSet::iterator iter = attempted_server_addresses_.find(address);
+ if (iter != attempted_server_addresses_.end()) {
+ LOG_J(LS_WARNING, this) << "Redirection to ["
+ << address.ToSensitiveString()
+ << "] ignored, allocation failed.";
+ return false;
+ }
+
+ // If protocol family of server address doesn't match with local, return.
+ if (!IsCompatibleAddress(address)) {
+ LOG(LS_WARNING) << "Server IP address family does not match with "
+ << "local host address family type";
+ return false;
+ }
+
+ LOG_J(LS_INFO, this) << "Redirecting from TURN server ["
+ << server_address_.address.ToSensitiveString()
+ << "] to TURN server ["
+ << address.ToSensitiveString()
+ << "]";
+ server_address_ = ProtocolAddress(address, server_address_.proto,
+ server_address_.secure);
+
+ // Insert the current address to prevent redirection pingpong.
+ attempted_server_addresses_.insert(server_address_.address);
+ return true;
+}
+
void TurnPort::ResolveTurnAddress(const rtc::SocketAddress& address) {
if (resolver_)
return;
@@ -544,6 +611,9 @@
if (message->message_id == MSG_ERROR) {
SignalPortError(this);
return;
+ } else if (message->message_id == MSG_ALLOCATE_MISMATCH) {
+ OnAllocateMismatch();
+ return;
}
Port::OnMessage(message);
@@ -805,6 +875,14 @@
case STUN_ERROR_UNAUTHORIZED: // Unauthrorized.
OnAuthChallenge(response, error_code->code());
break;
+ case STUN_ERROR_TRY_ALTERNATE:
+ OnTryAlternate(response, error_code->code());
+ break;
+ case STUN_ERROR_ALLOCATION_MISMATCH:
+ // We must handle this error async because trying to delete the socket in
+ // OnErrorResponse will cause a deadlock on the socket.
+ port_->thread()->Post(port_, TurnPort::MSG_ALLOCATE_MISMATCH);
+ break;
default:
LOG_J(LS_WARNING, port_) << "Allocate response error, code="
<< error_code->code();
@@ -849,6 +927,57 @@
port_->SendRequest(new TurnAllocateRequest(port_), 0);
}
+void TurnAllocateRequest::OnTryAlternate(StunMessage* response, int code) {
+ // TODO(guoweis): Currently, we only support UDP redirect
+ if (port_->server_address().proto != PROTO_UDP) {
+ LOG_J(LS_WARNING, port_) << "Receiving 300 Alternate Server on non-UDP "
+ << "allocating request from ["
+ << port_->server_address().address.ToSensitiveString()
+ << "], failed as currently not supported";
+ port_->OnAllocateError();
+ return;
+ }
+
+ // According to RFC 5389 section 11, there are use cases where
+ // authentication of response is not possible, we're not validating
+ // message integrity.
+
+ // Get the alternate server address attribute value.
+ const StunAddressAttribute* alternate_server_attr =
+ response->GetAddress(STUN_ATTR_ALTERNATE_SERVER);
+ if (!alternate_server_attr) {
+ LOG_J(LS_WARNING, port_) << "Missing STUN_ATTR_ALTERNATE_SERVER "
+ << "attribute in try alternate error response";
+ port_->OnAllocateError();
+ return;
+ }
+ if (!port_->SetAlternateServer(alternate_server_attr->GetAddress())) {
+ port_->OnAllocateError();
+ return;
+ }
+
+ // Check the attributes.
+ const StunByteStringAttribute* realm_attr =
+ response->GetByteString(STUN_ATTR_REALM);
+ if (realm_attr) {
+ LOG_J(LS_INFO, port_) << "Applying STUN_ATTR_REALM attribute in "
+ << "try alternate error response.";
+ port_->set_realm(realm_attr->GetString());
+ }
+
+ const StunByteStringAttribute* nonce_attr =
+ response->GetByteString(STUN_ATTR_NONCE);
+ if (nonce_attr) {
+ LOG_J(LS_INFO, port_) << "Applying STUN_ATTR_NONCE attribute in "
+ << "try alternate error response.";
+ port_->set_nonce(nonce_attr->GetString());
+ }
+
+ // Send another allocate request to alternate server,
+ // with the received realm and nonce values.
+ port_->SendRequest(new TurnAllocateRequest(port_), 0);
+}
+
TurnRefreshRequest::TurnRefreshRequest(TurnPort* port)
: StunRequest(new TurnMessage()),
port_(port) {
@@ -876,7 +1005,6 @@
}
void TurnRefreshRequest::OnErrorResponse(StunMessage* response) {
- // TODO(juberti): Handle 437 error response as a success.
const StunErrorCodeAttribute* error_code = response->GetErrorCode();
LOG_J(LS_WARNING, port_) << "Refresh response error, code="
<< error_code->code();
diff --git a/p2p/base/turnport.h b/p2p/base/turnport.h
index d73b11d..ab7d4e7 100644
--- a/p2p/base/turnport.h
+++ b/p2p/base/turnport.h
@@ -30,6 +30,7 @@
#include <stdio.h>
#include <list>
+#include <set>
#include <string>
#include "talk/p2p/base/port.h"
@@ -119,6 +120,12 @@
int error() const { return error_; }
+ void OnAllocateMismatch();
+
+ rtc::AsyncPacketSocket* socket() const {
+ return socket_;
+ }
+
// Signal with resolved server address.
// Parameters are port, server address and resolved server address.
// This signal will be sent only if server address is resolved successfully.
@@ -153,10 +160,14 @@
int server_priority);
private:
- enum { MSG_ERROR = MSG_FIRST_AVAILABLE };
+ enum {
+ MSG_ERROR = MSG_FIRST_AVAILABLE,
+ MSG_ALLOCATE_MISMATCH
+ };
typedef std::list<TurnEntry*> EntryList;
typedef std::map<rtc::Socket::Option, int> SocketOptionsMap;
+ typedef std::set<rtc::SocketAddress> AttemptedServerSet;
virtual void OnMessage(rtc::Message* pmsg);
@@ -170,6 +181,7 @@
}
}
+ bool SetAlternateServer(const rtc::SocketAddress& address);
void ResolveTurnAddress(const rtc::SocketAddress& address);
void OnResolveResult(rtc::AsyncResolverInterface* resolver);
@@ -207,6 +219,7 @@
ProtocolAddress server_address_;
RelayCredentials credentials_;
+ AttemptedServerSet attempted_server_addresses_;
rtc::AsyncPacketSocket* socket_;
SocketOptionsMap socket_options_;
@@ -226,6 +239,9 @@
// calculating the candidate priority.
int server_priority_;
+ // The number of retries made due to allocate mismatch error.
+ size_t allocate_mismatch_retries_;
+
friend class TurnEntry;
friend class TurnAllocateRequest;
friend class TurnRefreshRequest;
diff --git a/p2p/base/turnport_unittest.cc b/p2p/base/turnport_unittest.cc
index 44dc64f..d895cbd 100644
--- a/p2p/base/turnport_unittest.cc
+++ b/p2p/base/turnport_unittest.cc
@@ -64,6 +64,8 @@
static const SocketAddress kTurnTcpIntAddr("99.99.99.4",
cricket::TURN_SERVER_PORT);
static const SocketAddress kTurnUdpExtAddr("99.99.99.5", 0);
+static const SocketAddress kTurnAlternateUdpIntAddr(
+ "99.99.99.6", cricket::TURN_SERVER_PORT);
static const SocketAddress kTurnUdpIPv6IntAddr(
"2400:4030:1:2c00:be30:abcd:efab:cdef", cricket::TURN_SERVER_PORT);
static const SocketAddress kTurnUdpIPv6ExtAddr(
@@ -208,10 +210,13 @@
const cricket::ProtocolAddress& server_address) {
ASSERT(server_address.proto == cricket::PROTO_UDP);
- socket_.reset(socket_factory_.CreateUdpSocket(
- rtc::SocketAddress(kLocalAddr1.ipaddr(), 0), 0, 0));
- ASSERT_TRUE(socket_ != NULL);
- socket_->SignalReadPacket.connect(this, &TurnPortTest::OnSocketReadPacket);
+ if (!socket_) {
+ socket_.reset(socket_factory_.CreateUdpSocket(
+ rtc::SocketAddress(kLocalAddr1.ipaddr(), 0), 0, 0));
+ ASSERT_TRUE(socket_ != NULL);
+ socket_->SignalReadPacket.connect(
+ this, &TurnPortTest::OnSocketReadPacket);
+ }
cricket::RelayCredentials credentials(username, password);
turn_port_.reset(cricket::TurnPort::Create(
@@ -411,6 +416,80 @@
ASSERT_EQ(0U, turn_port_->Candidates().size());
}
+// Tests that a new local address is created after
+// STUN_ERROR_ALLOCATION_MISMATCH.
+TEST_F(TurnPortTest, TestTurnAllocateMismatch) {
+ // Do a normal allocation first.
+ CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
+ turn_port_->PrepareAddress();
+ EXPECT_TRUE_WAIT(turn_ready_, kTimeout);
+ rtc::SocketAddress first_addr(turn_port_->socket()->GetLocalAddress());
+
+ // Forces the socket server to assign the same port.
+ ss_->SetNextPortForTesting(first_addr.port());
+
+ turn_ready_ = false;
+ CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
+ turn_port_->PrepareAddress();
+
+ // Verifies that the new port has the same address.
+ EXPECT_EQ(first_addr, turn_port_->socket()->GetLocalAddress());
+
+ EXPECT_TRUE_WAIT(turn_ready_, kTimeout);
+
+ // Verifies that the new port has a different address now.
+ EXPECT_NE(first_addr, turn_port_->socket()->GetLocalAddress());
+}
+
+// Tests that a shared-socket-TurnPort creates its own socket after
+// STUN_ERROR_ALLOCATION_MISMATCH.
+TEST_F(TurnPortTest, TestSharedSocketAllocateMismatch) {
+ // Do a normal allocation first.
+ CreateSharedTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
+ turn_port_->PrepareAddress();
+ EXPECT_TRUE_WAIT(turn_ready_, kTimeout);
+ rtc::SocketAddress first_addr(turn_port_->socket()->GetLocalAddress());
+
+ turn_ready_ = false;
+ CreateSharedTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
+
+ // Verifies that the new port has the same address.
+ EXPECT_EQ(first_addr, turn_port_->socket()->GetLocalAddress());
+ EXPECT_TRUE(turn_port_->SharedSocket());
+
+ turn_port_->PrepareAddress();
+ EXPECT_TRUE_WAIT(turn_ready_, kTimeout);
+
+ // Verifies that the new port has a different address now.
+ EXPECT_NE(first_addr, turn_port_->socket()->GetLocalAddress());
+ EXPECT_FALSE(turn_port_->SharedSocket());
+}
+
+TEST_F(TurnPortTest, TestTurnTcpAllocateMismatch) {
+ turn_server_.AddInternalSocket(kTurnTcpIntAddr, cricket::PROTO_TCP);
+ CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr);
+
+ // Do a normal allocation first.
+ turn_port_->PrepareAddress();
+ EXPECT_TRUE_WAIT(turn_ready_, kTimeout);
+ rtc::SocketAddress first_addr(turn_port_->socket()->GetLocalAddress());
+
+ // Forces the socket server to assign the same port.
+ ss_->SetNextPortForTesting(first_addr.port());
+
+ turn_ready_ = false;
+ CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr);
+ turn_port_->PrepareAddress();
+
+ // Verifies that the new port has the same address.
+ EXPECT_EQ(first_addr, turn_port_->socket()->GetLocalAddress());
+
+ EXPECT_TRUE_WAIT(turn_ready_, kTimeout);
+
+ // Verifies that the new port has a different address now.
+ EXPECT_NE(first_addr, turn_port_->socket()->GetLocalAddress());
+}
+
// Do a TURN allocation and try to send a packet to it from the outside.
// The packet should be dropped. Then, try to send a packet from TURN to the
// outside. It should reach its destination. Finally, try again from the
@@ -445,6 +524,103 @@
ASSERT_EQ(0U, turn_port_->Candidates().size());
}
+// Test try-alternate-server feature.
+TEST_F(TurnPortTest, TestTurnAlternateServer) {
+ std::vector<rtc::SocketAddress> redirect_addresses;
+ redirect_addresses.push_back(kTurnAlternateUdpIntAddr);
+
+ cricket::TestTurnRedirector redirector(redirect_addresses);
+ turn_server_.AddInternalSocket(kTurnAlternateUdpIntAddr,
+ cricket::PROTO_UDP);
+ turn_server_.set_redirect_hook(&redirector);
+ CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
+
+ // Retrieve the address before we run the state machine.
+ const SocketAddress old_addr = turn_port_->server_address().address;
+
+ turn_port_->PrepareAddress();
+ EXPECT_TRUE_WAIT(turn_ready_, kTimeout);
+ // Retrieve the address again, the turn port's address should be
+ // changed.
+ const SocketAddress new_addr = turn_port_->server_address().address;
+ EXPECT_NE(old_addr, new_addr);
+ ASSERT_EQ(1U, turn_port_->Candidates().size());
+ EXPECT_EQ(kTurnUdpExtAddr.ipaddr(),
+ turn_port_->Candidates()[0].address().ipaddr());
+ EXPECT_NE(0, turn_port_->Candidates()[0].address().port());
+}
+
+// Test that we fail when we redirect to an address different from
+// current IP family.
+TEST_F(TurnPortTest, TestTurnAlternateServerV4toV6) {
+ std::vector<rtc::SocketAddress> redirect_addresses;
+ redirect_addresses.push_back(kTurnUdpIPv6IntAddr);
+
+ cricket::TestTurnRedirector redirector(redirect_addresses);
+ turn_server_.AddInternalSocket(kTurnAlternateUdpIntAddr,
+ cricket::PROTO_UDP);
+ turn_server_.set_redirect_hook(&redirector);
+ CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
+ turn_port_->PrepareAddress();
+ EXPECT_TRUE_WAIT(turn_error_, kTimeout);
+}
+
+// Test that we fail to handle alternate-server response over TCP protocol.
+TEST_F(TurnPortTest, TestTurnAlternateServerTcp) {
+ std::vector<rtc::SocketAddress> redirect_addresses;
+ redirect_addresses.push_back(kTurnAlternateUdpIntAddr);
+
+ cricket::TestTurnRedirector redirector(redirect_addresses);
+ turn_server_.set_redirect_hook(&redirector);
+ turn_server_.AddInternalSocket(kTurnTcpIntAddr, cricket::PROTO_TCP);
+ CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr);
+
+ turn_server_.AddInternalSocket(kTurnAlternateUdpIntAddr, cricket::PROTO_TCP);
+ turn_port_->PrepareAddress();
+ EXPECT_TRUE_WAIT(turn_error_, kTimeout);
+}
+
+// Test try-alternate-server catches the case of pingpong.
+TEST_F(TurnPortTest, TestTurnAlternateServerPingPong) {
+ std::vector<rtc::SocketAddress> redirect_addresses;
+ redirect_addresses.push_back(kTurnAlternateUdpIntAddr);
+ redirect_addresses.push_back(kTurnUdpIntAddr);
+
+ cricket::TestTurnRedirector redirector(redirect_addresses);
+
+ turn_server_.AddInternalSocket(kTurnAlternateUdpIntAddr,
+ cricket::PROTO_UDP);
+ turn_server_.set_redirect_hook(&redirector);
+ CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
+
+ turn_port_->PrepareAddress();
+ EXPECT_TRUE_WAIT(turn_error_, kTimeout);
+ ASSERT_EQ(0U, turn_port_->Candidates().size());
+ rtc::SocketAddress address;
+ // Verify that we have exhausted all alternate servers instead of
+ // failure caused by other errors.
+ EXPECT_FALSE(redirector.ShouldRedirect(address, &address));
+}
+
+// Test try-alternate-server catch the case of repeated server.
+TEST_F(TurnPortTest, TestTurnAlternateServerDetectRepetition) {
+ std::vector<rtc::SocketAddress> redirect_addresses;
+ redirect_addresses.push_back(kTurnAlternateUdpIntAddr);
+ redirect_addresses.push_back(kTurnAlternateUdpIntAddr);
+
+ cricket::TestTurnRedirector redirector(redirect_addresses);
+
+ turn_server_.AddInternalSocket(kTurnAlternateUdpIntAddr,
+ cricket::PROTO_UDP);
+ turn_server_.set_redirect_hook(&redirector);
+ CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
+
+ turn_port_->PrepareAddress();
+ EXPECT_TRUE_WAIT(turn_error_, kTimeout);
+ ASSERT_EQ(0U, turn_port_->Candidates().size());
+}
+
+
// Run TurnConnectionTest with one-time-use nonce feature.
// Here server will send a 438 STALE_NONCE error message for
// every TURN transaction.
@@ -515,4 +691,3 @@
EXPECT_EQ(last_fd_count, GetFDCount());
}
#endif
-
diff --git a/p2p/base/turnserver.cc b/p2p/base/turnserver.cc
index abc065a..dbcbcd4 100644
--- a/p2p/base/turnserver.cc
+++ b/p2p/base/turnserver.cc
@@ -62,9 +62,9 @@
return ((msg_type & 0xC000) == 0x4000);
}
-// IDs used for posted messages.
+// IDs used for posted messages for TurnServer::Allocation.
enum {
- MSG_TIMEOUT,
+ MSG_ALLOCATION_TIMEOUT,
};
// Encapsulates a TURN allocation.
@@ -208,6 +208,7 @@
: thread_(thread),
nonce_key_(rtc::CreateRandomString(kNonceKeySize)),
auth_hook_(NULL),
+ redirect_hook_(NULL),
enable_otu_nonce_(false) {
}
@@ -316,6 +317,15 @@
return;
}
+ if (redirect_hook_ != NULL && msg.type() == STUN_ALLOCATE_REQUEST) {
+ rtc::SocketAddress address;
+ if (redirect_hook_->ShouldRedirect(conn->src(), &address)) {
+ SendErrorResponseWithAlternateServer(
+ conn, &msg, address);
+ return;
+ }
+ }
+
// Look up the key that we'll use to validate the M-I. If we have an
// existing allocation, the key will already be cached.
Allocation* allocation = FindAllocation(conn);
@@ -334,7 +344,6 @@
}
if (!allocation && msg.type() == STUN_ALLOCATE_REQUEST) {
- // This is a new allocate request.
HandleAllocateRequest(conn, &msg, key);
} else if (allocation &&
(msg.type() != STUN_ALLOCATE_REQUEST ||
@@ -551,6 +560,17 @@
SendStun(conn, &resp);
}
+void TurnServer::SendErrorResponseWithAlternateServer(
+ Connection* conn, const StunMessage* msg,
+ const rtc::SocketAddress& addr) {
+ TurnMessage resp;
+ InitErrorResponse(msg, STUN_ERROR_TRY_ALTERNATE,
+ STUN_ERROR_REASON_TRY_ALTERNATE_SERVER, &resp);
+ VERIFY(resp.AddAttribute(new StunAddressAttribute(
+ STUN_ATTR_ALTERNATE_SERVER, addr)));
+ SendStun(conn, &resp);
+}
+
void TurnServer::SendStun(Connection* conn, StunMessage* msg) {
rtc::ByteBuffer buf;
// Add a SOFTWARE attribute if one is set.
@@ -588,7 +608,9 @@
InternalSocketMap::iterator iter = server_sockets_.find(socket);
if (iter != server_sockets_.end()) {
rtc::AsyncPacketSocket* socket = iter->first;
- delete socket;
+ // We must destroy the socket async to avoid invalidating the sigslot
+ // callback list iterator inside a sigslot callback.
+ rtc::Thread::Current()->Dispose(socket);
server_sockets_.erase(iter);
}
}
@@ -642,7 +664,7 @@
it != perms_.end(); ++it) {
delete *it;
}
- thread_->Clear(this, MSG_TIMEOUT);
+ thread_->Clear(this, MSG_ALLOCATION_TIMEOUT);
LOG_J(LS_INFO, this) << "Allocation destroyed";
}
@@ -687,7 +709,7 @@
// Figure out the lifetime and start the allocation timer.
int lifetime_secs = ComputeLifetime(msg);
- thread_->PostDelayed(lifetime_secs * 1000, this, MSG_TIMEOUT);
+ thread_->PostDelayed(lifetime_secs * 1000, this, MSG_ALLOCATION_TIMEOUT);
LOG_J(LS_INFO, this) << "Created allocation, lifetime=" << lifetime_secs;
@@ -714,8 +736,8 @@
int lifetime_secs = ComputeLifetime(msg);
// Reset the expiration timer.
- thread_->Clear(this, MSG_TIMEOUT);
- thread_->PostDelayed(lifetime_secs * 1000, this, MSG_TIMEOUT);
+ thread_->Clear(this, MSG_ALLOCATION_TIMEOUT);
+ thread_->PostDelayed(lifetime_secs * 1000, this, MSG_ALLOCATION_TIMEOUT);
LOG_J(LS_INFO, this) << "Refreshed allocation, lifetime=" << lifetime_secs;
@@ -943,7 +965,7 @@
}
void TurnServer::Allocation::OnMessage(rtc::Message* msg) {
- ASSERT(msg->message_id == MSG_TIMEOUT);
+ ASSERT(msg->message_id == MSG_ALLOCATION_TIMEOUT);
SignalDestroyed(this);
delete this;
}
@@ -968,16 +990,16 @@
}
TurnServer::Permission::~Permission() {
- thread_->Clear(this, MSG_TIMEOUT);
+ thread_->Clear(this, MSG_ALLOCATION_TIMEOUT);
}
void TurnServer::Permission::Refresh() {
- thread_->Clear(this, MSG_TIMEOUT);
- thread_->PostDelayed(kPermissionTimeout, this, MSG_TIMEOUT);
+ thread_->Clear(this, MSG_ALLOCATION_TIMEOUT);
+ thread_->PostDelayed(kPermissionTimeout, this, MSG_ALLOCATION_TIMEOUT);
}
void TurnServer::Permission::OnMessage(rtc::Message* msg) {
- ASSERT(msg->message_id == MSG_TIMEOUT);
+ ASSERT(msg->message_id == MSG_ALLOCATION_TIMEOUT);
SignalDestroyed(this);
delete this;
}
@@ -989,16 +1011,16 @@
}
TurnServer::Channel::~Channel() {
- thread_->Clear(this, MSG_TIMEOUT);
+ thread_->Clear(this, MSG_ALLOCATION_TIMEOUT);
}
void TurnServer::Channel::Refresh() {
- thread_->Clear(this, MSG_TIMEOUT);
- thread_->PostDelayed(kChannelTimeout, this, MSG_TIMEOUT);
+ thread_->Clear(this, MSG_ALLOCATION_TIMEOUT);
+ thread_->PostDelayed(kChannelTimeout, this, MSG_ALLOCATION_TIMEOUT);
}
void TurnServer::Channel::OnMessage(rtc::Message* msg) {
- ASSERT(msg->message_id == MSG_TIMEOUT);
+ ASSERT(msg->message_id == MSG_ALLOCATION_TIMEOUT);
SignalDestroyed(this);
delete this;
}
diff --git a/p2p/base/turnserver.h b/p2p/base/turnserver.h
index 4798232..553d00c 100644
--- a/p2p/base/turnserver.h
+++ b/p2p/base/turnserver.h
@@ -63,6 +63,14 @@
std::string* key) = 0;
};
+// An interface enables Turn Server to control redirection behavior.
+class TurnRedirectInterface {
+ public:
+ virtual bool ShouldRedirect(const rtc::SocketAddress& address,
+ rtc::SocketAddress* out) = 0;
+ virtual ~TurnRedirectInterface() {}
+};
+
// The core TURN server class. Give it a socket to listen on via
// AddInternalServerSocket, and a factory to create external sockets via
// SetExternalSocketFactory, and it's ready to go.
@@ -83,6 +91,10 @@
// Sets the authentication callback; does not take ownership.
void set_auth_hook(TurnAuthInterface* auth_hook) { auth_hook_ = auth_hook; }
+ void set_redirect_hook(TurnRedirectInterface* redirect_hook) {
+ redirect_hook_ = redirect_hook;
+ }
+
void set_enable_otu_nonce(bool enable) { enable_otu_nonce_ = enable; }
// Starts listening for packets from internal clients.
@@ -155,6 +167,11 @@
const StunMessage* req,
int code,
const std::string& reason);
+
+ void SendErrorResponseWithAlternateServer(Connection* conn,
+ const StunMessage* req,
+ const rtc::SocketAddress& addr);
+
void SendStun(Connection* conn, StunMessage* msg);
void Send(Connection* conn, const rtc::ByteBuffer& buf);
@@ -171,14 +188,17 @@
std::string realm_;
std::string software_;
TurnAuthInterface* auth_hook_;
+ TurnRedirectInterface* redirect_hook_;
// otu - one-time-use. Server will respond with 438 if it's
// sees the same nonce in next transaction.
bool enable_otu_nonce_;
+
InternalSocketMap server_sockets_;
ServerSocketMap server_listen_sockets_;
rtc::scoped_ptr<rtc::PacketSocketFactory>
external_socket_factory_;
rtc::SocketAddress external_addr_;
+
AllocationMap allocations_;
};
diff --git a/p2p/client/autoportallocator.h b/p2p/client/autoportallocator.h
index 298f829..ed87162 100644
--- a/p2p/client/autoportallocator.h
+++ b/p2p/client/autoportallocator.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_EXAMPLES_LOGIN_AUTOPORTALLOCATOR_H_
-#define TALK_EXAMPLES_LOGIN_AUTOPORTALLOCATOR_H_
+#ifndef TALK_P2P_CLIENT_AUTOPORTALLOCATOR_H_
+#define TALK_P2P_CLIENT_AUTOPORTALLOCATOR_H_
#include <string>
#include <vector>
@@ -66,4 +66,4 @@
}
};
-#endif // TALK_EXAMPLES_LOGIN_AUTOPORTALLOCATOR_H_
+#endif // TALK_P2P_CLIENT_AUTOPORTALLOCATOR_H_
diff --git a/p2p/client/basicportallocator.cc b/p2p/client/basicportallocator.cc
index d39db8f..0ec13e7 100644
--- a/p2p/client/basicportallocator.cc
+++ b/p2p/client/basicportallocator.cc
@@ -47,13 +47,15 @@
namespace {
-const uint32 MSG_CONFIG_START = 1;
-const uint32 MSG_CONFIG_READY = 2;
-const uint32 MSG_ALLOCATE = 3;
-const uint32 MSG_ALLOCATION_PHASE = 4;
-const uint32 MSG_SHAKE = 5;
-const uint32 MSG_SEQUENCEOBJECTS_CREATED = 6;
-const uint32 MSG_CONFIG_STOP = 7;
+enum {
+ MSG_CONFIG_START,
+ MSG_CONFIG_READY,
+ MSG_ALLOCATE,
+ MSG_ALLOCATION_PHASE,
+ MSG_SHAKE,
+ MSG_SEQUENCEOBJECTS_CREATED,
+ MSG_CONFIG_STOP,
+};
const int PHASE_UDP = 0;
const int PHASE_RELAY = 1;
@@ -147,9 +149,6 @@
const rtc::PacketTime& packet_time);
void OnPortDestroyed(PortInterface* port);
- void OnResolvedTurnServerAddress(
- TurnPort* port, const rtc::SocketAddress& server_address,
- const rtc::SocketAddress& resolved_server_address);
BasicPortAllocatorSession* session_;
rtc::Network* network_;
@@ -161,8 +160,7 @@
rtc::scoped_ptr<rtc::AsyncPacketSocket> udp_socket_;
// There will be only one udp port per AllocationSequence.
UDPPort* udp_port_;
- // Keeping a map for turn ports keyed with server addresses.
- std::map<rtc::SocketAddress, Port*> turn_ports_;
+ std::vector<TurnPort*> turn_ports_;
int phase_;
};
@@ -228,10 +226,11 @@
PortAllocatorSession *BasicPortAllocator::CreateSessionInternal(
const std::string& content_name, int component,
const std::string& ice_ufrag, const std::string& ice_pwd) {
- return new BasicPortAllocatorSession(this, content_name, component,
- ice_ufrag, ice_pwd);
+ return new BasicPortAllocatorSession(
+ this, content_name, component, ice_ufrag, ice_pwd);
}
+
// BasicPortAllocatorSession
BasicPortAllocatorSession::BasicPortAllocatorSession(
BasicPortAllocator *allocator,
@@ -432,7 +431,11 @@
}
if (!(sequence_flags & PORTALLOCATOR_ENABLE_IPV6) &&
+#ifdef USE_WEBRTC_DEV_BRANCH
+ networks[i]->GetBestIP().family() == AF_INET6) {
+#else // USE_WEBRTC_DEV_BRANCH
networks[i]->ip().family() == AF_INET6) {
+#endif // USE_WEBRTC_DEV_BRANCH
// Skip IPv6 networks unless the flag's been set.
continue;
}
@@ -530,8 +533,10 @@
// Send candidates whose protocol is enabled.
std::vector<Candidate> candidates;
ProtocolType pvalue;
+ bool candidate_allowed_to_send = CheckCandidateFilter(c);
if (StringToProto(c.protocol().c_str(), &pvalue) &&
- data->sequence()->ProtocolEnabled(pvalue)) {
+ data->sequence()->ProtocolEnabled(pvalue) &&
+ candidate_allowed_to_send) {
candidates.push_back(c);
}
@@ -542,7 +547,9 @@
// Moving to READY state as we have atleast one candidate from the port.
// Since this port has atleast one candidate we should forward this port
// to listners, to allow connections from this port.
- if (!data->ready()) {
+ // Also we should make sure that candidate gathered from this port is allowed
+ // to send outside.
+ if (!data->ready() && candidate_allowed_to_send) {
data->set_ready();
SignalPortReady(this, port);
}
@@ -588,6 +595,8 @@
const std::vector<Candidate>& potentials = it->port()->Candidates();
for (size_t i = 0; i < potentials.size(); ++i) {
+ if (!CheckCandidateFilter(potentials[i]))
+ continue;
ProtocolType pvalue;
if (!StringToProto(potentials[i].protocol().c_str(), &pvalue))
continue;
@@ -602,6 +611,31 @@
}
}
+bool BasicPortAllocatorSession::CheckCandidateFilter(const Candidate& c) {
+ uint32 filter = allocator_->candidate_filter();
+ bool allowed = false;
+ if (filter & CF_RELAY) {
+ allowed |= (c.type() == RELAY_PORT_TYPE);
+ }
+
+ if (filter & CF_REFLEXIVE) {
+ // We allow host candidates if the filter allows server-reflexive candidates
+ // and the candidate is a public IP. Because we don't generate
+ // server-reflexive candidates if they have the same IP as the host
+ // candidate (i.e. when the host candidate is a public IP), filtering to
+ // only server-reflexive candidates won't work right when the host
+ // candidates have public IPs.
+ allowed |= (c.type() == STUN_PORT_TYPE) ||
+ (c.type() == LOCAL_PORT_TYPE && !c.address().IsPrivateIP());
+ }
+
+ if (filter & CF_HOST) {
+ allowed |= (c.type() == LOCAL_PORT_TYPE);
+ }
+
+ return allowed;
+}
+
void BasicPortAllocatorSession::OnPortAllocationComplete(
AllocationSequence* seq) {
// Send candidate allocation complete signal if all ports are done.
@@ -698,7 +732,12 @@
uint32 flags)
: session_(session),
network_(network),
+
+#ifdef USE_WEBRTC_DEV_BRANCH
+ ip_(network->GetBestIP()),
+#else // USE_WEBRTC_DEV_BRANCH
ip_(network->ip()),
+#endif // USE_WEBRTC_DEV_BRANCH
config_(config),
state_(kInit),
flags_(flags),
@@ -741,7 +780,11 @@
void AllocationSequence::DisableEquivalentPhases(rtc::Network* network,
PortConfiguration* config, uint32* flags) {
+#ifdef USE_WEBRTC_DEV_BRANCH
+ if (!((network == network_) && (ip_ == network->GetBestIP()))) {
+#else // USE_WEBRTC_DEV_BRANCH
if (!((network == network_) && (ip_ == network->ip()))) {
+#endif // USE_WEBRTC_DEV_BRANCH
// Different network setup; nothing is equivalent.
return;
}
@@ -1020,26 +1063,15 @@
// don't pass shared socket for ports which will create TCP sockets.
// TODO(mallinath) - Enable shared socket mode for TURN ports. Disabled
// due to webrtc bug https://code.google.com/p/webrtc/issues/detail?id=3537
- if (IsFlagSet(PORTALLOCATOR_ENABLE_TURN_SHARED_SOCKET) &&
+ if (IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_SOCKET) &&
relay_port->proto == PROTO_UDP) {
port = TurnPort::Create(session_->network_thread(),
session_->socket_factory(),
network_, udp_socket_.get(),
session_->username(), session_->password(),
*relay_port, config.credentials, config.priority);
- // If we are using shared socket for TURN and udp ports, we need to
- // find a way to demux the packets to the correct port when received.
- // Mapping against server_address is one way of doing this. When packet
- // is received the remote_address will be checked against the map.
- // If server address is not resolved, a signal will be sent from the port
- // after the address is resolved. The map entry will updated with the
- // resolved address when the signal is received from the port.
- if ((*relay_port).address.IsUnresolved()) {
- // If server address is not resolved then listen for signal from port.
- port->SignalResolvedServerAddress.connect(
- this, &AllocationSequence::OnResolvedTurnServerAddress);
- }
- turn_ports_[(*relay_port).address] = port;
+
+ turn_ports_.push_back(port);
// Listen to the port destroyed signal, to allow AllocationSequence to
// remove entrt from it's map.
port->SignalDestroyed.connect(this, &AllocationSequence::OnPortDestroyed);
@@ -1063,51 +1095,45 @@
const rtc::SocketAddress& remote_addr,
const rtc::PacketTime& packet_time) {
ASSERT(socket == udp_socket_.get());
- // If the packet is received from one of the TURN server in the config, then
- // pass down the packet to that port, otherwise it will be handed down to
- // the local udp port.
- Port* port = NULL;
- std::map<rtc::SocketAddress, Port*>::iterator iter =
- turn_ports_.find(remote_addr);
- if (iter != turn_ports_.end()) {
- port = iter->second;
- } else if (udp_port_) {
- port = udp_port_;
+
+ bool turn_port_found = false;
+
+ // Try to find the TurnPort that matches the remote address. Note that the
+ // message could be a STUN binding response if the TURN server is also used as
+ // a STUN server. We don't want to parse every message here to check if it is
+ // a STUN binding response, so we pass the message to TurnPort regardless of
+ // the message type. The TurnPort will just ignore the message since it will
+ // not find any request by transaction ID.
+ for (std::vector<TurnPort*>::const_iterator it = turn_ports_.begin();
+ it != turn_ports_.end(); ++it) {
+ TurnPort* port = *it;
+ if (port->server_address().address == remote_addr) {
+ port->HandleIncomingPacket(socket, data, size, remote_addr, packet_time);
+ turn_port_found = true;
+ break;
+ }
}
- ASSERT(port != NULL);
- if (port) {
- port->HandleIncomingPacket(socket, data, size, remote_addr, packet_time);
+
+ if (udp_port_) {
+ const ServerAddresses& stun_servers = udp_port_->server_addresses();
+
+ // Pass the packet to the UdpPort if there is no matching TurnPort, or if
+ // the TURN server is also a STUN server.
+ if (!turn_port_found ||
+ stun_servers.find(remote_addr) != stun_servers.end()) {
+ udp_port_->HandleIncomingPacket(
+ socket, data, size, remote_addr, packet_time);
+ }
}
}
void AllocationSequence::OnPortDestroyed(PortInterface* port) {
if (udp_port_ == port) {
udp_port_ = NULL;
- } else {
- std::map<rtc::SocketAddress, Port*>::iterator iter;
- for (iter = turn_ports_.begin(); iter != turn_ports_.end(); ++iter) {
- if (iter->second == port) {
- turn_ports_.erase(iter);
- break;
- }
- }
- }
-}
-
-void AllocationSequence::OnResolvedTurnServerAddress(
- TurnPort* port, const rtc::SocketAddress& server_address,
- const rtc::SocketAddress& resolved_server_address) {
- std::map<rtc::SocketAddress, Port*>::iterator iter;
- iter = turn_ports_.find(server_address);
- if (iter == turn_ports_.end()) {
- LOG(LS_INFO) << "TurnPort entry is not found in the map.";
return;
}
- ASSERT(iter->second == port);
- // Remove old entry and then insert using the resolved address as key.
- turn_ports_.erase(iter);
- turn_ports_[resolved_server_address] = port;
+ turn_ports_.erase(std::find(turn_ports_.begin(), turn_ports_.end(), port));
}
// PortConfiguration
diff --git a/p2p/client/basicportallocator.h b/p2p/client/basicportallocator.h
index 5f43880..d424772 100644
--- a/p2p/client/basicportallocator.h
+++ b/p2p/client/basicportallocator.h
@@ -160,7 +160,6 @@
void set_ready() { ASSERT(state_ == STATE_INIT); state_ = STATE_READY; }
void set_complete() {
- ASSERT(state_ == STATE_READY);
state_ = STATE_COMPLETE;
}
void set_error() {
@@ -201,6 +200,8 @@
void OnPortAllocationComplete(AllocationSequence* seq);
PortData* FindPort(Port* port);
+ bool CheckCandidateFilter(const Candidate& c);
+
BasicPortAllocator* allocator_;
rtc::Thread* network_thread_;
rtc::scoped_ptr<rtc::PacketSocketFactory> owned_socket_factory_;
diff --git a/p2p/client/connectivitychecker.cc b/p2p/client/connectivitychecker.cc
index 06de5e4..723c5a1 100644
--- a/p2p/client/connectivitychecker.cc
+++ b/p2p/client/connectivitychecker.cc
@@ -1,5 +1,29 @@
-// Copyright 2011 Google Inc. All Rights Reserved.
-
+/*
+ * libjingle
+ * Copyright 2011, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
#include <string>
@@ -214,7 +238,12 @@
return;
}
rtc::ProxyInfo proxy_info = request->proxy();
- NicMap::iterator i = nics_.find(NicId(networks[0]->ip(), proxy_info.address));
+ NicMap::iterator i =
+#ifdef USE_WEBRTC_DEV_BRANCH
+ nics_.find(NicId(networks[0]->GetBestIP(), proxy_info.address));
+#else // USE_WEBRTC_DEV_BRANCH
+ nics_.find(NicId(networks[0]->ip(), proxy_info.address));
+#endif // USE_WEBRTC_DEV_BRANCH
if (i != nics_.end()) {
int port = request->port();
uint32 now = rtc::Time();
@@ -247,7 +276,11 @@
ASSERT(worker_ == rtc::Thread::Current());
RelayPort* relay_port = reinterpret_cast<RelayPort*>(port);
const ProtocolAddress* address = relay_port->ServerAddress(0);
+#ifdef USE_WEBRTC_DEV_BRANCH
+ rtc::IPAddress ip = port->Network()->GetBestIP();
+#else // USE_WEBRTC_DEV_BRANCH
rtc::IPAddress ip = port->Network()->ip();
+#endif // USE_WEBRTC_DEV_BRANCH
NicMap::iterator i = nics_.find(NicId(ip, port->proxy().address));
if (i != nics_.end()) {
// We have it already, add the new information.
@@ -281,7 +314,11 @@
ASSERT(worker_ == rtc::Thread::Current());
const std::vector<Candidate> candidates = port->Candidates();
Candidate c = candidates[0];
+#ifdef USE_WEBRTC_DEV_BRANCH
+ rtc::IPAddress ip = port->Network()->GetBestIP();
+#else // USE_WEBRTC_DEV_BRANCH
rtc::IPAddress ip = port->Network()->ip();
+#endif // USE_WEBRTC_DEV_BRANCH
NicMap::iterator i = nics_.find(NicId(ip, port->proxy().address));
if (i != nics_.end()) {
// We have it already, add the new information.
@@ -300,7 +337,11 @@
void ConnectivityChecker::OnStunPortError(Port* port) {
ASSERT(worker_ == rtc::Thread::Current());
LOG(LS_ERROR) << "Stun address error.";
+#ifdef USE_WEBRTC_DEV_BRANCH
+ rtc::IPAddress ip = port->Network()->GetBestIP();
+#else // USE_WEBRTC_DEV_BRANCH
rtc::IPAddress ip = port->Network()->ip();
+#endif // USE_WEBRTC_DEV_BRANCH
NicMap::iterator i = nics_.find(NicId(ip, port->proxy().address));
if (i != nics_.end()) {
// We have it already, add the new information.
@@ -337,19 +378,36 @@
StunPort* ConnectivityChecker::CreateStunPort(
const std::string& username, const std::string& password,
const PortConfiguration* config, rtc::Network* network) {
- return StunPort::Create(worker_, socket_factory_.get(),
- network, network->ip(), 0, 0,
- username, password, config->stun_servers);
+ return StunPort::Create(worker_,
+ socket_factory_.get(),
+ network,
+#ifdef USE_WEBRTC_DEV_BRANCH
+ network->GetBestIP(),
+#else // USE_WEBRTC_DEV_BRANCH
+ network->ip(),
+#endif // USE_WEBRTC_DEV_BRANCH
+ 0,
+ 0,
+ username,
+ password,
+ config->stun_servers);
}
RelayPort* ConnectivityChecker::CreateRelayPort(
const std::string& username, const std::string& password,
const PortConfiguration* config, rtc::Network* network) {
- return RelayPort::Create(worker_, socket_factory_.get(),
- network, network->ip(),
+ return RelayPort::Create(worker_,
+ socket_factory_.get(),
+ network,
+#ifdef USE_WEBRTC_DEV_BRANCH
+ network->GetBestIP(),
+#else // USE_WEBRTC_DEV_BRANCH
+ network->ip(),
+#endif // USE_WEBRTC_DEV_BRANCH
port_allocator_->min_port(),
port_allocator_->max_port(),
- username, password);
+ username,
+ password);
}
void ConnectivityChecker::CreateRelayPorts(
@@ -365,8 +423,12 @@
for (relay = config->relays.begin();
relay != config->relays.end(); ++relay) {
for (uint32 i = 0; i < networks.size(); ++i) {
- NicMap::iterator iter = nics_.find(NicId(networks[i]->ip(),
- proxy_info.address));
+ NicMap::iterator iter =
+#ifdef USE_WEBRTC_DEV_BRANCH
+ nics_.find(NicId(networks[i]->GetBestIP(), proxy_info.address));
+#else // USE_WEBRTC_DEV_BRANCH
+ nics_.find(NicId(networks[i]->ip(), proxy_info.address));
+#endif // USE_WEBRTC_DEV_BRANCH
if (iter != nics_.end()) {
// TODO: Now setting the same start time for all protocols.
// This might affect accuracy, but since we are mainly looking for
@@ -423,7 +485,11 @@
rtc::ProxyInfo proxy_info = GetProxyInfo();
bool allocate_relay_ports = false;
for (uint32 i = 0; i < networks.size(); ++i) {
+#ifdef USE_WEBRTC_DEV_BRANCH
+ if (AddNic(networks[i]->GetBestIP(), proxy_info.address)) {
+#else // USE_WEBRTC_DEV_BRANCH
if (AddNic(networks[i]->ip(), proxy_info.address)) {
+#endif // USE_WEBRTC_DEV_BRANCH
Port* port = CreateStunPort(username, password, &config, networks[i]);
if (port) {
@@ -500,7 +566,12 @@
return;
}
rtc::ProxyInfo proxy_info = GetProxyInfo();
- NicMap::iterator i = nics_.find(NicId(networks[0]->ip(), proxy_info.address));
+ NicMap::iterator i =
+#ifdef USE_WEBRTC_DEV_BRANCH
+ nics_.find(NicId(networks[0]->GetBestIP(), proxy_info.address));
+#else // USE_WEBRTC_DEV_BRANCH
+ nics_.find(NicId(networks[0]->ip(), proxy_info.address));
+#endif // USE_WEBRTC_DEV_BRANCH
if (i != nics_.end()) {
uint32 now = rtc::Time();
NicInfo* nic_info = &i->second;
diff --git a/p2p/client/connectivitychecker_unittest.cc b/p2p/client/connectivitychecker_unittest.cc
index b96cf17..187505a 100644
--- a/p2p/client/connectivitychecker_unittest.cc
+++ b/p2p/client/connectivitychecker_unittest.cc
@@ -1,5 +1,29 @@
-// Copyright 2011 Google Inc. All Rights Reserved.
-
+/*
+ * libjingle
+ * Copyright 2011, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
#include <string>
@@ -211,19 +235,35 @@
virtual StunPort* CreateStunPort(
const std::string& username, const std::string& password,
const PortConfiguration* config, rtc::Network* network) {
- return new FakeStunPort(worker(), socket_factory_,
- network, network->ip(),
- kMinPort, kMaxPort,
- username, password,
+ return new FakeStunPort(worker(),
+ socket_factory_,
+ network,
+#ifdef USE_WEBRTC_DEV_BRANCH
+ network->GetBestIP(),
+#else // USE_WEBRTC_DEV_BRANCH
+ network->ip(),
+#endif // USE_WEBRTC_DEV_BRANCH
+ kMinPort,
+ kMaxPort,
+ username,
+ password,
config->stun_servers);
}
virtual RelayPort* CreateRelayPort(
const std::string& username, const std::string& password,
const PortConfiguration* config, rtc::Network* network) {
- return new FakeRelayPort(worker(), socket_factory_,
- network, network->ip(),
- kMinPort, kMaxPort,
- username, password);
+ return new FakeRelayPort(worker(),
+ socket_factory_,
+ network,
+#ifdef USE_WEBRTC_DEV_BRANCH
+ network->GetBestIP(),
+#else // USE_WEBRTC_DEV_BRANCH
+ network->ip(),
+#endif // USE_WEBRTC_DEV_BRANCH
+ kMinPort,
+ kMaxPort,
+ username,
+ password);
}
virtual void InitiateProxyDetection() {
if (!proxy_initiated_) {
diff --git a/p2p/client/fakeportallocator.h b/p2p/client/fakeportallocator.h
index 6c36c4e..e1a04dd 100644
--- a/p2p/client/fakeportallocator.h
+++ b/p2p/client/fakeportallocator.h
@@ -1,6 +1,29 @@
-// Copyright 2010 Google Inc. All Rights Reserved,
-//
-// Author: Justin Uberti (juberti@google.com)
+/*
+ * libjingle
+ * Copyright 2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
#ifndef TALK_P2P_CLIENT_FAKEPORTALLOCATOR_H_
#define TALK_P2P_CLIENT_FAKEPORTALLOCATOR_H_
@@ -39,10 +62,18 @@
virtual void StartGettingPorts() {
if (!port_) {
- port_.reset(cricket::UDPPort::Create(worker_thread_, factory_,
- &network_, network_.ip(), 0, 0,
- username(),
- password()));
+ port_.reset(cricket::UDPPort::Create(worker_thread_,
+ factory_,
+ &network_,
+#ifdef USE_WEBRTC_DEV_BRANCH
+ network_.GetBestIP(),
+#else // USE_WEBRTC_DEV_BRANCH
+ network_.ip(),
+#endif // USE_WEBRTC_DEV_BRANCH
+ 0,
+ 0,
+ username(),
+ password()));
AddPort(port_.get());
}
++port_config_count_;
diff --git a/p2p/client/portallocator_unittest.cc b/p2p/client/portallocator_unittest.cc
index 57cfbe3..e793064 100644
--- a/p2p/client/portallocator_unittest.cc
+++ b/p2p/client/portallocator_unittest.cc
@@ -53,6 +53,7 @@
using rtc::Thread;
static const SocketAddress kClientAddr("11.11.11.11", 0);
+static const SocketAddress kPrivateAddr("192.168.1.11", 0);
static const SocketAddress kClientIPv6Addr(
"2401:fa00:4:1000:be30:5bff:fee5:c3", 0);
static const SocketAddress kClientAddr2("22.22.22.22", 0);
@@ -132,9 +133,32 @@
bool SetPortRange(int min_port, int max_port) {
return allocator_->SetPortRange(min_port, max_port);
}
- rtc::NATServer* CreateNatServer(const SocketAddress& addr,
- rtc::NATType type) {
- return new rtc::NATServer(type, vss_.get(), addr, vss_.get(), addr);
+ void ResetWithNatServer(const rtc::SocketAddress& stun_server) {
+ nat_server_.reset(new rtc::NATServer(
+ rtc::NAT_OPEN_CONE, vss_.get(), kNatAddr, vss_.get(), kNatAddr));
+
+ ServerAddresses stun_servers;
+ stun_servers.insert(stun_server);
+ allocator_.reset(new cricket::BasicPortAllocator(
+ &network_manager_, &nat_socket_factory_, stun_servers));
+ allocator().set_step_delay(cricket::kMinimumStepDelay);
+ }
+
+ void AddTurnServers(const rtc::SocketAddress& udp_turn,
+ const rtc::SocketAddress& tcp_turn) {
+ cricket::RelayServerConfig relay_server(cricket::RELAY_TURN);
+ cricket::RelayCredentials credentials(kTurnUsername, kTurnPassword);
+ relay_server.credentials = credentials;
+
+ if (!udp_turn.IsNil()) {
+ relay_server.ports.push_back(cricket::ProtocolAddress(
+ kTurnUdpIntAddr, cricket::PROTO_UDP, false));
+ }
+ if (!tcp_turn.IsNil()) {
+ relay_server.ports.push_back(cricket::ProtocolAddress(
+ kTurnTcpIntAddr, cricket::PROTO_TCP, false));
+ }
+ allocator_->AddRelay(relay_server);
}
bool CreateSession(int component) {
@@ -253,6 +277,7 @@
rtc::scoped_ptr<rtc::VirtualSocketServer> vss_;
rtc::scoped_ptr<rtc::FirewallSocketServer> fss_;
rtc::SocketServerScope ss_scope_;
+ rtc::scoped_ptr<rtc::NATServer> nat_server_;
rtc::NATSocketFactory nat_factory_;
rtc::BasicPacketSocketFactory nat_socket_factory_;
cricket::TestStunServer stun_server_;
@@ -538,7 +563,7 @@
// Test to verify ICE restart process.
TEST_F(PortAllocatorTest, TestGetAllPortsRestarts) {
AddInterface(kClientAddr);
- EXPECT_TRUE(CreateSession(1));
+ EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
session_->StartGettingPorts();
EXPECT_EQ_WAIT(7U, candidates_.size(), kDefaultAllocationTimeout);
EXPECT_EQ(4U, ports_.size());
@@ -546,6 +571,73 @@
// TODO - Extend this to verify ICE restart.
}
+// Test ICE candidate filter mechanism with options Relay/Host/Reflexive.
+TEST_F(PortAllocatorTest, TestCandidateFilterWithRelayOnly) {
+ AddInterface(kClientAddr);
+ allocator().set_candidate_filter(cricket::CF_RELAY);
+ EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+ session_->StartGettingPorts();
+ EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
+ // Using GTURN, we will have 4 candidates.
+ EXPECT_EQ(4U, candidates_.size());
+ EXPECT_EQ(1U, ports_.size()); // Only Relay port will be in ready state.
+ for (size_t i = 0; i < candidates_.size(); ++i) {
+ EXPECT_EQ(std::string(cricket::RELAY_PORT_TYPE), candidates_[i].type());
+ }
+}
+
+TEST_F(PortAllocatorTest, TestCandidateFilterWithHostOnly) {
+ AddInterface(kClientAddr);
+ allocator().set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
+ cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
+ allocator().set_candidate_filter(cricket::CF_HOST);
+ EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+ session_->StartGettingPorts();
+ EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
+ EXPECT_EQ(2U, candidates_.size()); // Host UDP/TCP candidates only.
+ EXPECT_EQ(2U, ports_.size()); // UDP/TCP ports only.
+ for (size_t i = 0; i < candidates_.size(); ++i) {
+ EXPECT_EQ(std::string(cricket::LOCAL_PORT_TYPE), candidates_[i].type());
+ }
+}
+
+// Host is behind the NAT.
+TEST_F(PortAllocatorTest, TestCandidateFilterWithReflexiveOnly) {
+ AddInterface(kPrivateAddr);
+ ResetWithNatServer(kStunAddr);
+
+ allocator().set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
+ cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
+ allocator().set_candidate_filter(cricket::CF_REFLEXIVE);
+ EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+ session_->StartGettingPorts();
+ EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
+ // Host is behind NAT, no private address will be exposed. Hence only UDP
+ // port with STUN candidate will be sent outside.
+ EXPECT_EQ(1U, candidates_.size()); // Only STUN candidate.
+ EXPECT_EQ(1U, ports_.size()); // Only UDP port will be in ready state.
+ for (size_t i = 0; i < candidates_.size(); ++i) {
+ EXPECT_EQ(std::string(cricket::STUN_PORT_TYPE), candidates_[i].type());
+ }
+}
+
+// Host is not behind the NAT.
+TEST_F(PortAllocatorTest, TestCandidateFilterWithReflexiveOnlyAndNoNAT) {
+ AddInterface(kClientAddr);
+ allocator().set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
+ cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
+ allocator().set_candidate_filter(cricket::CF_REFLEXIVE);
+ EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+ session_->StartGettingPorts();
+ EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
+ // Host has a public address, both UDP and TCP candidates will be exposed.
+ EXPECT_EQ(2U, candidates_.size()); // Local UDP + TCP candidate.
+ EXPECT_EQ(2U, ports_.size()); // UDP and TCP ports will be in ready state.
+ for (size_t i = 0; i < candidates_.size(); ++i) {
+ EXPECT_EQ(std::string(cricket::LOCAL_PORT_TYPE), candidates_[i].type());
+ }
+}
+
TEST_F(PortAllocatorTest, TestBasicMuxFeatures) {
AddInterface(kClientAddr);
allocator().set_flags(cricket::PORTALLOCATOR_ENABLE_BUNDLE);
@@ -698,13 +790,8 @@
// local candidates as client behind a nat.
TEST_F(PortAllocatorTest, TestSharedSocketWithNat) {
AddInterface(kClientAddr);
- rtc::scoped_ptr<rtc::NATServer> nat_server(
- CreateNatServer(kNatAddr, rtc::NAT_OPEN_CONE));
- ServerAddresses stun_servers;
- stun_servers.insert(kStunAddr);
- allocator_.reset(new cricket::BasicPortAllocator(
- &network_manager_, &nat_socket_factory_, stun_servers));
- allocator_->set_step_delay(cricket::kMinimumStepDelay);
+ ResetWithNatServer(kStunAddr);
+
allocator_->set_flags(allocator().flags() |
cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
@@ -726,14 +813,8 @@
turn_server_.AddInternalSocket(kTurnTcpIntAddr, cricket::PROTO_TCP);
AddInterface(kClientAddr);
allocator_.reset(new cricket::BasicPortAllocator(&network_manager_));
- cricket::RelayServerConfig relay_server(cricket::RELAY_TURN);
- cricket::RelayCredentials credentials(kTurnUsername, kTurnPassword);
- relay_server.credentials = credentials;
- relay_server.ports.push_back(cricket::ProtocolAddress(
- kTurnUdpIntAddr, cricket::PROTO_UDP, false));
- relay_server.ports.push_back(cricket::ProtocolAddress(
- kTurnTcpIntAddr, cricket::PROTO_TCP, false));
- allocator_->AddRelay(relay_server);
+
+ AddTurnServers(kTurnUdpIntAddr, kTurnTcpIntAddr);
allocator_->set_step_delay(cricket::kMinimumStepDelay);
allocator_->set_flags(allocator().flags() |
@@ -790,20 +871,10 @@
// stun and turn candidates.
TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurn) {
AddInterface(kClientAddr);
- rtc::scoped_ptr<rtc::NATServer> nat_server(
- CreateNatServer(kNatAddr, rtc::NAT_OPEN_CONE));
- ServerAddresses stun_servers;
- stun_servers.insert(kStunAddr);
- allocator_.reset(new cricket::BasicPortAllocator(
- &network_manager_, &nat_socket_factory_, stun_servers));
- cricket::RelayServerConfig relay_server(cricket::RELAY_TURN);
- cricket::RelayCredentials credentials(kTurnUsername, kTurnPassword);
- relay_server.credentials = credentials;
- relay_server.ports.push_back(cricket::ProtocolAddress(
- kTurnUdpIntAddr, cricket::PROTO_UDP, false));
- allocator_->AddRelay(relay_server);
+ ResetWithNatServer(kStunAddr);
- allocator_->set_step_delay(cricket::kMinimumStepDelay);
+ AddTurnServers(kTurnUdpIntAddr, rtc::SocketAddress());
+
allocator_->set_flags(allocator().flags() |
cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
@@ -829,6 +900,45 @@
EXPECT_EQ(1U, ports_[1]->Candidates().size());
}
+// Test that when PORTALLOCATOR_ENABLE_SHARED_SOCKET is enabled and the TURN
+// server is also used as the STUN server, we should get 'local', 'stun', and
+// 'relay' candidates.
+TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurnAsStun) {
+ AddInterface(kClientAddr);
+ ResetWithNatServer(kTurnUdpIntAddr);
+ AddTurnServers(kTurnUdpIntAddr, rtc::SocketAddress());
+
+ // Must set the step delay to 0 to make sure the relay allocation phase is
+ // started before the STUN candidates are obtained, so that the STUN binding
+ // response is processed when both StunPort and TurnPort exist to reproduce
+ // webrtc issue 3537.
+ allocator_->set_step_delay(0);
+ allocator_->set_flags(allocator().flags() |
+ cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
+ cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
+ cricket::PORTALLOCATOR_DISABLE_TCP);
+
+ EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+ session_->StartGettingPorts();
+
+ ASSERT_EQ_WAIT(3U, candidates_.size(), kDefaultAllocationTimeout);
+ EXPECT_PRED5(CheckCandidate, candidates_[0],
+ cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp", kClientAddr);
+ EXPECT_PRED5(CheckCandidate, candidates_[1],
+ cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
+ rtc::SocketAddress(kNatAddr.ipaddr(), 0));
+ EXPECT_PRED5(CheckCandidate, candidates_[2],
+ cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+ rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
+ EXPECT_EQ(candidates_[2].related_address(), candidates_[1].address());
+
+ EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
+ EXPECT_EQ(3U, candidates_.size());
+ // Local port will be created first and then TURN port.
+ EXPECT_EQ(2U, ports_[0]->Candidates().size());
+ EXPECT_EQ(1U, ports_[1]->Candidates().size());
+}
+
// This test verifies when PORTALLOCATOR_ENABLE_SHARED_SOCKET flag is enabled
// and fail to generate STUN candidate, local UDP candidate is generated
// properly.
diff --git a/p2p/client/sessionmanagertask.h b/p2p/client/sessionmanagertask.h
index d7d9733..e16d9d6 100644
--- a/p2p/client/sessionmanagertask.h
+++ b/p2p/client/sessionmanagertask.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef _SESSIONMANAGERTASK_H_
-#define _SESSIONMANAGERTASK_H_
+#ifndef TALK_P2P_CLIENT_SESSIONMANAGERTASK_H_
+#define TALK_P2P_CLIENT_SESSIONMANAGERTASK_H_
#include "talk/p2p/base/sessionmanager.h"
#include "talk/p2p/client/sessionsendtask.h"
@@ -90,4 +90,4 @@
} // namespace cricket
-#endif // _SESSIONMANAGERTASK_H_
+#endif // TALK_P2P_CLIENT_SESSIONMANAGERTASK_H_
diff --git a/session/media/call.cc b/session/media/call.cc
index 56d886d..4fbdd0f 100644
--- a/session/media/call.cc
+++ b/session/media/call.cc
@@ -94,7 +94,6 @@
Call::Call(MediaSessionClient* session_client)
: id_(rtc::CreateRandomId()),
session_client_(session_client),
- local_renderer_(NULL),
has_video_(false),
has_data_(false),
muted_(false),
@@ -207,13 +206,6 @@
return session->SendInfoMessage(elems, session->remote_name());
}
-void Call::SetLocalRenderer(VideoRenderer* renderer) {
- local_renderer_ = renderer;
- if (session_client_->GetFocus() == this) {
- session_client_->channel_manager()->SetLocalRenderer(renderer);
- }
-}
-
void Call::SetVideoRenderer(Session* session, uint32 ssrc,
VideoRenderer* renderer) {
VideoChannel* video_channel = GetVideoChannel(session);
@@ -418,8 +410,6 @@
for (it = media_session_map_.begin(); it != media_session_map_.end(); ++it) {
EnableSessionChannels(it->second.session, enable);
}
- session_client_->channel_manager()->SetLocalRenderer(
- (enable) ? local_renderer_ : NULL);
}
void Call::EnableSessionChannels(Session* session, bool enable) {
diff --git a/session/media/call.h b/session/media/call.h
index a81e74a..be9397e 100644
--- a/session/media/call.h
+++ b/session/media/call.h
@@ -96,7 +96,6 @@
void Terminate();
bool SendViewRequest(Session* session,
const ViewRequest& view_request);
- void SetLocalRenderer(VideoRenderer* renderer);
void SetVideoRenderer(Session* session, uint32 ssrc,
VideoRenderer* renderer);
void StartConnectionMonitor(Session* session, int cms);
@@ -284,7 +283,6 @@
MediaSessionMap media_session_map_;
std::map<std::string, CurrentSpeakerMonitor*> speaker_monitor_map_;
- VideoRenderer* local_renderer_;
bool has_video_;
bool has_data_;
bool muted_;
diff --git a/session/media/channelmanager.cc b/session/media/channelmanager.cc
index a1cdcc0..199bc86 100644
--- a/session/media/channelmanager.cc
+++ b/session/media/channelmanager.cc
@@ -137,6 +137,12 @@
this, &ChannelManager::OnVideoCaptureStateChange);
capture_manager_->SignalCapturerStateChange.connect(
this, &ChannelManager::OnVideoCaptureStateChange);
+
+ if (worker_thread_ != rtc::Thread::Current()) {
+ // Do not allow invoking calls to other threads on the worker thread.
+ worker_thread_->Invoke<bool>(
+ rtc::Bind(&rtc::Thread::SetAllowBlockingCalls, worker_thread_, false));
+ }
}
ChannelManager::~ChannelManager() {
@@ -277,10 +283,6 @@
if (default_video_encoder_config_.max_codec.id != 0) {
SetDefaultVideoEncoderConfig(default_video_encoder_config_);
}
- // And the local renderer.
- if (local_renderer_) {
- SetLocalRenderer(local_renderer_);
- }
}
}
return initialized_;
@@ -750,19 +752,6 @@
return ret;
}
-bool ChannelManager::SetLocalRenderer(VideoRenderer* renderer) {
- bool ret = true;
- if (initialized_) {
- ret = worker_thread_->Invoke<bool>(
- Bind(&MediaEngineInterface::SetLocalRenderer,
- media_engine_.get(), renderer));
- }
- if (ret) {
- local_renderer_ = renderer;
- }
- return ret;
-}
-
void ChannelManager::SetVoiceLogging(int level, const char* filter) {
if (initialized_) {
worker_thread_->Invoke<void>(
diff --git a/session/media/channelmanager.h b/session/media/channelmanager.h
index f7a2890..fa79014 100644
--- a/session/media/channelmanager.h
+++ b/session/media/channelmanager.h
@@ -170,8 +170,6 @@
// Starts/stops the local microphone and enables polling of the input level.
bool SetLocalMonitor(bool enable);
bool monitoring() const { return monitoring_; }
- // Sets the local renderer where to renderer the local camera.
- bool SetLocalRenderer(VideoRenderer* renderer);
bool capturing() const { return capturing_; }
// Configures the logging output of the mediaengine(s).
diff --git a/session/media/channelmanager_unittest.cc b/session/media/channelmanager_unittest.cc
index f234732..98f1400 100644
--- a/session/media/channelmanager_unittest.cc
+++ b/session/media/channelmanager_unittest.cc
@@ -509,22 +509,6 @@
EXPECT_EQ(60, level);
}
-// Test that a value set before Init is applied properly.
-TEST_F(ChannelManagerTest, SetLocalRendererBeforeInit) {
- cricket::NullVideoRenderer renderer;
- EXPECT_TRUE(cm_->SetLocalRenderer(&renderer));
- EXPECT_TRUE(cm_->Init());
- EXPECT_EQ(&renderer, fme_->local_renderer());
-}
-
-// Test that a value set after init is passed through properly.
-TEST_F(ChannelManagerTest, SetLocalRenderer) {
- cricket::NullVideoRenderer renderer;
- EXPECT_TRUE(cm_->Init());
- EXPECT_TRUE(cm_->SetLocalRenderer(&renderer));
- EXPECT_EQ(&renderer, fme_->local_renderer());
-}
-
// Test that logging options set before Init are applied properly,
// and retained even after Init.
TEST_F(ChannelManagerTest, SetLoggingBeforeInit) {
diff --git a/session/media/mediamessages.cc b/session/media/mediamessages.cc
index 1a5094e..6c9f681 100644
--- a/session/media/mediamessages.cc
+++ b/session/media/mediamessages.cc
@@ -34,7 +34,7 @@
#include "talk/p2p/base/constants.h"
#include "talk/p2p/base/parsing.h"
#include "talk/session/media/mediasessionclient.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
diff --git a/session/media/mediamessages_unittest.cc b/session/media/mediamessages_unittest.cc
index c95053d..9ebd38c 100644
--- a/session/media/mediamessages_unittest.cc
+++ b/session/media/mediamessages_unittest.cc
@@ -32,7 +32,7 @@
#include "talk/p2p/base/constants.h"
#include "talk/session/media/mediasessionclient.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/scoped_ptr.h"
diff --git a/session/media/mediamonitor.h b/session/media/mediamonitor.h
index d549362..89740a8 100644
--- a/session/media/mediamonitor.h
+++ b/session/media/mediamonitor.h
@@ -34,6 +34,7 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/thread.h"
+#include "webrtc/base/thread_annotations.h"
namespace cricket {
@@ -77,7 +78,7 @@
media_info_.Clear();
media_channel_->GetStats(&media_info_);
}
- virtual void Update() {
+ virtual void Update() EXCLUSIVE_LOCKS_REQUIRED(crit_) {
MI stats(media_info_);
crit_.Leave();
SignalUpdate(media_channel_, stats);
diff --git a/session/media/mediasession.cc b/session/media/mediasession.cc
index 45e321f..92dd257 100644
--- a/session/media/mediasession.cc
+++ b/session/media/mediasession.cc
@@ -1170,28 +1170,28 @@
if (current_description) {
ContentInfos::const_iterator it = current_description->contents().begin();
for (; it != current_description->contents().end(); ++it) {
- if (IsMediaContentOfType(&*it, MEDIA_TYPE_AUDIO) && options.has_audio) {
+ if (IsMediaContentOfType(&*it, MEDIA_TYPE_AUDIO)) {
if (!AddAudioContentForOffer(options, current_description,
audio_rtp_extensions, audio_codecs,
¤t_streams, offer.get())) {
return NULL;
}
audio_added = true;
- } else if (IsMediaContentOfType(&*it, MEDIA_TYPE_VIDEO) &&
- options.has_video) {
+ } else if (IsMediaContentOfType(&*it, MEDIA_TYPE_VIDEO)) {
if (!AddVideoContentForOffer(options, current_description,
video_rtp_extensions, video_codecs,
¤t_streams, offer.get())) {
return NULL;
}
video_added = true;
- } else if (IsMediaContentOfType(&*it, MEDIA_TYPE_DATA) &&
- options.has_data()) {
+ } else if (IsMediaContentOfType(&*it, MEDIA_TYPE_DATA)) {
if (!AddDataContentForOffer(options, current_description, &data_codecs,
¤t_streams, offer.get())) {
return NULL;
}
data_added = true;
+ } else {
+ ASSERT(false);
}
}
}
@@ -1459,6 +1459,7 @@
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
SetMediaProtocol(secure_transport, audio.get());
+
desc->AddContent(CN_AUDIO, NS_JINGLE_RTP, audio.release());
if (!AddTransportOffer(CN_AUDIO, options.transport_options,
current_description, desc)) {
diff --git a/session/media/mediasessionclient.cc b/session/media/mediasessionclient.cc
index a1096b6..826909a 100644
--- a/session/media/mediasessionclient.cc
+++ b/session/media/mediasessionclient.cc
@@ -36,8 +36,8 @@
#include "talk/p2p/base/parsing.h"
#include "talk/session/media/mediamessages.h"
#include "talk/session/media/srtpfilter.h"
-#include "talk/xmllite/qname.h"
-#include "talk/xmllite/xmlconstants.h"
+#include "webrtc/libjingle/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/xmlconstants.h"
#include "talk/xmpp/constants.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
diff --git a/session/media/mediasessionclient_unittest.cc b/session/media/mediasessionclient_unittest.cc
index 3e36b5a..2998228 100644
--- a/session/media/mediasessionclient_unittest.cc
+++ b/session/media/mediasessionclient_unittest.cc
@@ -34,9 +34,9 @@
#include "talk/p2p/base/constants.h"
#include "talk/p2p/client/basicportallocator.h"
#include "talk/session/media/mediasessionclient.h"
-#include "talk/xmllite/xmlbuilder.h"
-#include "talk/xmllite/xmlelement.h"
-#include "talk/xmllite/xmlprinter.h"
+#include "webrtc/libjingle/xmllite/xmlbuilder.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlprinter.h"
#include "talk/xmpp/constants.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
diff --git a/session/tunnel/securetunnelsessionclient.cc b/session/tunnel/securetunnelsessionclient.cc
index f4a3346..cb41c3b 100644
--- a/session/tunnel/securetunnelsessionclient.cc
+++ b/session/tunnel/securetunnelsessionclient.cc
@@ -30,7 +30,7 @@
#include "talk/p2p/base/transportchannel.h"
#include "talk/session/tunnel/pseudotcpchannel.h"
#include "talk/session/tunnel/securetunnelsessionclient.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "webrtc/base/basicdefs.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/common.h"
diff --git a/session/tunnel/tunnelsessionclient.cc b/session/tunnel/tunnelsessionclient.cc
index 12275fb..7221db4 100644
--- a/session/tunnel/tunnelsessionclient.cc
+++ b/session/tunnel/tunnelsessionclient.cc
@@ -28,7 +28,7 @@
#include "pseudotcpchannel.h"
#include "talk/p2p/base/constants.h"
#include "talk/p2p/base/transportchannel.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "tunnelsessionclient.h"
#include "webrtc/base/basicdefs.h"
#include "webrtc/base/basictypes.h"
diff --git a/session/tunnel/tunnelsessionclient.h b/session/tunnel/tunnelsessionclient.h
index ac12faf..d3fa64a 100644
--- a/session/tunnel/tunnelsessionclient.h
+++ b/session/tunnel/tunnelsessionclient.h
@@ -36,7 +36,7 @@
#include "talk/p2p/base/sessionclient.h"
#include "talk/p2p/base/sessiondescription.h"
#include "talk/p2p/base/sessionmanager.h"
-#include "talk/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/qname.h"
#include "talk/xmpp/constants.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/stream.h"
diff --git a/sound/alsasoundsystem.cc b/sound/alsasoundsystem.cc
deleted file mode 100644
index d9960bb..0000000
--- a/sound/alsasoundsystem.cc
+++ /dev/null
@@ -1,761 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/sound/alsasoundsystem.h"
-
-#include "talk/sound/sounddevicelocator.h"
-#include "talk/sound/soundinputstreaminterface.h"
-#include "talk/sound/soundoutputstreaminterface.h"
-#include "webrtc/base/common.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/stringutils.h"
-#include "webrtc/base/timeutils.h"
-#include "webrtc/base/worker.h"
-
-namespace cricket {
-
-// Lookup table from the cricket format enum in soundsysteminterface.h to
-// ALSA's enums.
-static const snd_pcm_format_t kCricketFormatToAlsaFormatTable[] = {
- // The order here must match the order in soundsysteminterface.h
- SND_PCM_FORMAT_S16_LE,
-};
-
-// Lookup table for the size of a single sample of a given format.
-static const size_t kCricketFormatToSampleSizeTable[] = {
- // The order here must match the order in soundsysteminterface.h
- sizeof(int16_t), // 2
-};
-
-// Minimum latency we allow, in microseconds. This is more or less arbitrary,
-// but it has to be at least large enough to be able to buffer data during a
-// missed context switch, and the typical Linux scheduling quantum is 10ms.
-static const int kMinimumLatencyUsecs = 20 * 1000;
-
-// The latency we'll use for kNoLatencyRequirements (chosen arbitrarily).
-static const int kDefaultLatencyUsecs = kMinimumLatencyUsecs * 2;
-
-// We translate newlines in ALSA device descriptions to hyphens.
-static const char kAlsaDescriptionSearch[] = "\n";
-static const char kAlsaDescriptionReplace[] = " - ";
-
-class AlsaDeviceLocator : public SoundDeviceLocator {
- public:
- AlsaDeviceLocator(const std::string &name,
- const std::string &device_name)
- : SoundDeviceLocator(name, device_name) {
- // The ALSA descriptions have newlines in them, which won't show up in
- // a drop-down box. Replace them with hyphens.
- rtc::replace_substrs(kAlsaDescriptionSearch,
- sizeof(kAlsaDescriptionSearch) - 1,
- kAlsaDescriptionReplace,
- sizeof(kAlsaDescriptionReplace) - 1,
- &name_);
- }
-
- virtual SoundDeviceLocator *Copy() const {
- return new AlsaDeviceLocator(*this);
- }
-};
-
-// Functionality that is common to both AlsaInputStream and AlsaOutputStream.
-class AlsaStream {
- public:
- AlsaStream(AlsaSoundSystem *alsa,
- snd_pcm_t *handle,
- size_t frame_size,
- int wait_timeout_ms,
- int flags,
- int freq)
- : alsa_(alsa),
- handle_(handle),
- frame_size_(frame_size),
- wait_timeout_ms_(wait_timeout_ms),
- flags_(flags),
- freq_(freq) {
- }
-
- ~AlsaStream() {
- Close();
- }
-
- // Waits for the stream to be ready to accept/return more data, and returns
- // how much can be written/read, or 0 if we need to Wait() again.
- snd_pcm_uframes_t Wait() {
- snd_pcm_sframes_t frames;
- // Ideally we would not use snd_pcm_wait() and instead hook snd_pcm_poll_*
- // into PhysicalSocketServer, but PhysicalSocketServer is nasty enough
- // already and the current clients of SoundSystemInterface do not run
- // anything else on their worker threads, so snd_pcm_wait() is good enough.
- frames = symbol_table()->snd_pcm_avail_update()(handle_);
- if (frames < 0) {
- LOG(LS_ERROR) << "snd_pcm_avail_update(): " << GetError(frames);
- Recover(frames);
- return 0;
- } else if (frames > 0) {
- // Already ready, so no need to wait.
- return frames;
- }
- // Else no space/data available, so must wait.
- int ready = symbol_table()->snd_pcm_wait()(handle_, wait_timeout_ms_);
- if (ready < 0) {
- LOG(LS_ERROR) << "snd_pcm_wait(): " << GetError(ready);
- Recover(ready);
- return 0;
- } else if (ready == 0) {
- // Timeout, so nothing can be written/read right now.
- // We set the timeout to twice the requested latency, so continuous
- // timeouts are indicative of a problem, so log as a warning.
- LOG(LS_WARNING) << "Timeout while waiting on stream";
- return 0;
- }
- // Else ready > 0 (i.e., 1), so it's ready. Get count.
- frames = symbol_table()->snd_pcm_avail_update()(handle_);
- if (frames < 0) {
- LOG(LS_ERROR) << "snd_pcm_avail_update(): " << GetError(frames);
- Recover(frames);
- return 0;
- } else if (frames == 0) {
- // wait() said we were ready, so this ought to have been positive. Has
- // been observed to happen in practice though.
- LOG(LS_WARNING) << "Spurious wake-up";
- }
- return frames;
- }
-
- int CurrentDelayUsecs() {
- if (!(flags_ & SoundSystemInterface::FLAG_REPORT_LATENCY)) {
- return 0;
- }
-
- snd_pcm_sframes_t delay;
- int err = symbol_table()->snd_pcm_delay()(handle_, &delay);
- if (err != 0) {
- LOG(LS_ERROR) << "snd_pcm_delay(): " << GetError(err);
- Recover(err);
- // We'd rather continue playout/capture with an incorrect delay than stop
- // it altogether, so return a valid value.
- return 0;
- }
- // The delay is in frames. Convert to microseconds.
- return delay * rtc::kNumMicrosecsPerSec / freq_;
- }
-
- // Used to recover from certain recoverable errors, principally buffer overrun
- // or underrun (identified as EPIPE). Without calling this the stream stays
- // in the error state forever.
- bool Recover(int error) {
- int err;
- err = symbol_table()->snd_pcm_recover()(
- handle_,
- error,
- // Silent; i.e., no logging on stderr.
- 1);
- if (err != 0) {
- // Docs say snd_pcm_recover returns the original error if it is not one
- // of the recoverable ones, so this log message will probably contain the
- // same error twice.
- LOG(LS_ERROR) << "Unable to recover from \"" << GetError(error) << "\": "
- << GetError(err);
- return false;
- }
- if (error == -EPIPE && // Buffer underrun/overrun.
- symbol_table()->snd_pcm_stream()(handle_) == SND_PCM_STREAM_CAPTURE) {
- // For capture streams we also have to repeat the explicit start() to get
- // data flowing again.
- err = symbol_table()->snd_pcm_start()(handle_);
- if (err != 0) {
- LOG(LS_ERROR) << "snd_pcm_start(): " << GetError(err);
- return false;
- }
- }
- return true;
- }
-
- bool Close() {
- if (handle_) {
- int err;
- err = symbol_table()->snd_pcm_drop()(handle_);
- if (err != 0) {
- LOG(LS_ERROR) << "snd_pcm_drop(): " << GetError(err);
- // Continue anyways.
- }
- err = symbol_table()->snd_pcm_close()(handle_);
- if (err != 0) {
- LOG(LS_ERROR) << "snd_pcm_close(): " << GetError(err);
- // Continue anyways.
- }
- handle_ = NULL;
- }
- return true;
- }
-
- AlsaSymbolTable *symbol_table() {
- return &alsa_->symbol_table_;
- }
-
- snd_pcm_t *handle() {
- return handle_;
- }
-
- const char *GetError(int err) {
- return alsa_->GetError(err);
- }
-
- size_t frame_size() {
- return frame_size_;
- }
-
- private:
- AlsaSoundSystem *alsa_;
- snd_pcm_t *handle_;
- size_t frame_size_;
- int wait_timeout_ms_;
- int flags_;
- int freq_;
-
- DISALLOW_COPY_AND_ASSIGN(AlsaStream);
-};
-
-// Implementation of an input stream. See soundinputstreaminterface.h regarding
-// thread-safety.
-class AlsaInputStream :
- public SoundInputStreamInterface,
- private rtc::Worker {
- public:
- AlsaInputStream(AlsaSoundSystem *alsa,
- snd_pcm_t *handle,
- size_t frame_size,
- int wait_timeout_ms,
- int flags,
- int freq)
- : stream_(alsa, handle, frame_size, wait_timeout_ms, flags, freq),
- buffer_size_(0) {
- }
-
- virtual ~AlsaInputStream() {
- bool success = StopReading();
- // We need that to live.
- VERIFY(success);
- }
-
- virtual bool StartReading() {
- return StartWork();
- }
-
- virtual bool StopReading() {
- return StopWork();
- }
-
- virtual bool GetVolume(int *volume) {
- // TODO: Implement this.
- return false;
- }
-
- virtual bool SetVolume(int volume) {
- // TODO: Implement this.
- return false;
- }
-
- virtual bool Close() {
- return StopReading() && stream_.Close();
- }
-
- virtual int LatencyUsecs() {
- return stream_.CurrentDelayUsecs();
- }
-
- private:
- // Inherited from Worker.
- virtual void OnStart() {
- HaveWork();
- }
-
- // Inherited from Worker.
- virtual void OnHaveWork() {
- // Block waiting for data.
- snd_pcm_uframes_t avail = stream_.Wait();
- if (avail > 0) {
- // Data is available.
- size_t size = avail * stream_.frame_size();
- if (size > buffer_size_) {
- // Must increase buffer size.
- buffer_.reset(new char[size]);
- buffer_size_ = size;
- }
- // Read all the data.
- snd_pcm_sframes_t read = stream_.symbol_table()->snd_pcm_readi()(
- stream_.handle(),
- buffer_.get(),
- avail);
- if (read < 0) {
- LOG(LS_ERROR) << "snd_pcm_readi(): " << GetError(read);
- stream_.Recover(read);
- } else if (read == 0) {
- // Docs say this shouldn't happen.
- ASSERT(false);
- LOG(LS_ERROR) << "No data?";
- } else {
- // Got data. Pass it off to the app.
- SignalSamplesRead(buffer_.get(),
- read * stream_.frame_size(),
- this);
- }
- }
- // Check for more data with no delay, after any pending messages are
- // dispatched.
- HaveWork();
- }
-
- // Inherited from Worker.
- virtual void OnStop() {
- // Nothing to do.
- }
-
- const char *GetError(int err) {
- return stream_.GetError(err);
- }
-
- AlsaStream stream_;
- rtc::scoped_ptr<char[]> buffer_;
- size_t buffer_size_;
-
- DISALLOW_COPY_AND_ASSIGN(AlsaInputStream);
-};
-
-// Implementation of an output stream. See soundoutputstreaminterface.h
-// regarding thread-safety.
-class AlsaOutputStream :
- public SoundOutputStreamInterface,
- private rtc::Worker {
- public:
- AlsaOutputStream(AlsaSoundSystem *alsa,
- snd_pcm_t *handle,
- size_t frame_size,
- int wait_timeout_ms,
- int flags,
- int freq)
- : stream_(alsa, handle, frame_size, wait_timeout_ms, flags, freq) {
- }
-
- virtual ~AlsaOutputStream() {
- bool success = DisableBufferMonitoring();
- // We need that to live.
- VERIFY(success);
- }
-
- virtual bool EnableBufferMonitoring() {
- return StartWork();
- }
-
- virtual bool DisableBufferMonitoring() {
- return StopWork();
- }
-
- virtual bool WriteSamples(const void *sample_data,
- size_t size) {
- if (size % stream_.frame_size() != 0) {
- // No client of SoundSystemInterface does this, so let's not support it.
- // (If we wanted to support it, we'd basically just buffer the fractional
- // frame until we get more data.)
- ASSERT(false);
- LOG(LS_ERROR) << "Writes with fractional frames are not supported";
- return false;
- }
- snd_pcm_uframes_t frames = size / stream_.frame_size();
- snd_pcm_sframes_t written = stream_.symbol_table()->snd_pcm_writei()(
- stream_.handle(),
- sample_data,
- frames);
- if (written < 0) {
- LOG(LS_ERROR) << "snd_pcm_writei(): " << GetError(written);
- stream_.Recover(written);
- return false;
- } else if (static_cast<snd_pcm_uframes_t>(written) < frames) {
- // Shouldn't happen. Drop the rest of the data.
- LOG(LS_ERROR) << "Stream wrote only " << written << " of " << frames
- << " frames!";
- return false;
- }
- return true;
- }
-
- virtual bool GetVolume(int *volume) {
- // TODO: Implement this.
- return false;
- }
-
- virtual bool SetVolume(int volume) {
- // TODO: Implement this.
- return false;
- }
-
- virtual bool Close() {
- return DisableBufferMonitoring() && stream_.Close();
- }
-
- virtual int LatencyUsecs() {
- return stream_.CurrentDelayUsecs();
- }
-
- private:
- // Inherited from Worker.
- virtual void OnStart() {
- HaveWork();
- }
-
- // Inherited from Worker.
- virtual void OnHaveWork() {
- snd_pcm_uframes_t avail = stream_.Wait();
- if (avail > 0) {
- size_t space = avail * stream_.frame_size();
- SignalBufferSpace(space, this);
- }
- HaveWork();
- }
-
- // Inherited from Worker.
- virtual void OnStop() {
- // Nothing to do.
- }
-
- const char *GetError(int err) {
- return stream_.GetError(err);
- }
-
- AlsaStream stream_;
-
- DISALLOW_COPY_AND_ASSIGN(AlsaOutputStream);
-};
-
-AlsaSoundSystem::AlsaSoundSystem() : initialized_(false) {}
-
-AlsaSoundSystem::~AlsaSoundSystem() {
- // Not really necessary, because Terminate() doesn't really do anything.
- Terminate();
-}
-
-bool AlsaSoundSystem::Init() {
- if (IsInitialized()) {
- return true;
- }
-
- // Load libasound.
- if (!symbol_table_.Load()) {
- // Very odd for a Linux machine to not have a working libasound ...
- LOG(LS_ERROR) << "Failed to load symbol table";
- return false;
- }
-
- initialized_ = true;
-
- return true;
-}
-
-void AlsaSoundSystem::Terminate() {
- if (!IsInitialized()) {
- return;
- }
-
- initialized_ = false;
-
- // We do not unload the symbol table because we may need it again soon if
- // Init() is called again.
-}
-
-bool AlsaSoundSystem::EnumeratePlaybackDevices(
- SoundDeviceLocatorList *devices) {
- return EnumerateDevices(devices, false);
-}
-
-bool AlsaSoundSystem::EnumerateCaptureDevices(
- SoundDeviceLocatorList *devices) {
- return EnumerateDevices(devices, true);
-}
-
-bool AlsaSoundSystem::GetDefaultPlaybackDevice(SoundDeviceLocator **device) {
- return GetDefaultDevice(device);
-}
-
-bool AlsaSoundSystem::GetDefaultCaptureDevice(SoundDeviceLocator **device) {
- return GetDefaultDevice(device);
-}
-
-SoundOutputStreamInterface *AlsaSoundSystem::OpenPlaybackDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms) {
- return OpenDevice<SoundOutputStreamInterface>(
- device,
- params,
- SND_PCM_STREAM_PLAYBACK,
- &AlsaSoundSystem::StartOutputStream);
-}
-
-SoundInputStreamInterface *AlsaSoundSystem::OpenCaptureDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms) {
- return OpenDevice<SoundInputStreamInterface>(
- device,
- params,
- SND_PCM_STREAM_CAPTURE,
- &AlsaSoundSystem::StartInputStream);
-}
-
-const char *AlsaSoundSystem::GetName() const {
- return "ALSA";
-}
-
-bool AlsaSoundSystem::EnumerateDevices(
- SoundDeviceLocatorList *devices,
- bool capture_not_playback) {
- ClearSoundDeviceLocatorList(devices);
-
- if (!IsInitialized()) {
- return false;
- }
-
- const char *type = capture_not_playback ? "Input" : "Output";
- // dmix and dsnoop are only for playback and capture, respectively, but ALSA
- // stupidly includes them in both lists.
- const char *ignore_prefix = capture_not_playback ? "dmix:" : "dsnoop:";
- // (ALSA lists many more "devices" of questionable interest, but we show them
- // just in case the weird devices may actually be desirable for some
- // users/systems.)
- const char *ignore_default = "default";
- const char *ignore_null = "null";
- const char *ignore_pulse = "pulse";
- // The 'pulse' entry has a habit of mysteriously disappearing when you query
- // a second time. Remove it from our list. (GIPS lib did the same thing.)
- int err;
-
- void **hints;
- err = symbol_table_.snd_device_name_hint()(-1, // All cards
- "pcm", // Only PCM devices
- &hints);
- if (err != 0) {
- LOG(LS_ERROR) << "snd_device_name_hint(): " << GetError(err);
- return false;
- }
-
- for (void **list = hints; *list != NULL; ++list) {
- char *actual_type = symbol_table_.snd_device_name_get_hint()(*list, "IOID");
- if (actual_type) { // NULL means it's both.
- bool wrong_type = (strcmp(actual_type, type) != 0);
- free(actual_type);
- if (wrong_type) {
- // Wrong type of device (i.e., input vs. output).
- continue;
- }
- }
-
- char *name = symbol_table_.snd_device_name_get_hint()(*list, "NAME");
- if (!name) {
- LOG(LS_ERROR) << "Device has no name???";
- // Skip it.
- continue;
- }
-
- // Now check if we actually want to show this device.
- if (strcmp(name, ignore_default) != 0 &&
- strcmp(name, ignore_null) != 0 &&
- strcmp(name, ignore_pulse) != 0 &&
- !rtc::starts_with(name, ignore_prefix)) {
-
- // Yes, we do.
- char *desc = symbol_table_.snd_device_name_get_hint()(*list, "DESC");
- if (!desc) {
- // Virtual devices don't necessarily have descriptions. Use their names
- // instead (not pretty!).
- desc = name;
- }
-
- AlsaDeviceLocator *device = new AlsaDeviceLocator(desc, name);
-
- devices->push_back(device);
-
- if (desc != name) {
- free(desc);
- }
- }
-
- free(name);
- }
-
- err = symbol_table_.snd_device_name_free_hint()(hints);
- if (err != 0) {
- LOG(LS_ERROR) << "snd_device_name_free_hint(): " << GetError(err);
- // Continue and return true anyways, since we did get the whole list.
- }
-
- return true;
-}
-
-bool AlsaSoundSystem::GetDefaultDevice(SoundDeviceLocator **device) {
- if (!IsInitialized()) {
- return false;
- }
- *device = new AlsaDeviceLocator("Default device", "default");
- return true;
-}
-
-inline size_t AlsaSoundSystem::FrameSize(const OpenParams ¶ms) {
- ASSERT(static_cast<int>(params.format) <
- ARRAY_SIZE(kCricketFormatToSampleSizeTable));
- return kCricketFormatToSampleSizeTable[params.format] * params.channels;
-}
-
-template <typename StreamInterface>
-StreamInterface *AlsaSoundSystem::OpenDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms,
- snd_pcm_stream_t type,
- StreamInterface *(AlsaSoundSystem::*start_fn)(
- snd_pcm_t *handle,
- size_t frame_size,
- int wait_timeout_ms,
- int flags,
- int freq)) {
-
- if (!IsInitialized()) {
- return NULL;
- }
-
- StreamInterface *stream;
- int err;
-
- const char *dev = static_cast<const AlsaDeviceLocator *>(device)->
- device_name().c_str();
-
- snd_pcm_t *handle = NULL;
- err = symbol_table_.snd_pcm_open()(
- &handle,
- dev,
- type,
- // No flags.
- 0);
- if (err != 0) {
- LOG(LS_ERROR) << "snd_pcm_open(" << dev << "): " << GetError(err);
- return NULL;
- }
- LOG(LS_VERBOSE) << "Opening " << dev;
- ASSERT(handle); // If open succeeded, handle ought to be valid
-
- // Compute requested latency in microseconds.
- int latency;
- if (params.latency == kNoLatencyRequirements) {
- latency = kDefaultLatencyUsecs;
- } else {
- // kLowLatency is 0, so we treat it the same as a request for zero latency.
- // Compute what the user asked for.
- latency = rtc::kNumMicrosecsPerSec *
- params.latency /
- params.freq /
- FrameSize(params);
- // And this is what we'll actually use.
- latency = rtc::_max(latency, kMinimumLatencyUsecs);
- }
-
- ASSERT(static_cast<int>(params.format) <
- ARRAY_SIZE(kCricketFormatToAlsaFormatTable));
-
- err = symbol_table_.snd_pcm_set_params()(
- handle,
- kCricketFormatToAlsaFormatTable[params.format],
- // SoundSystemInterface only supports interleaved audio.
- SND_PCM_ACCESS_RW_INTERLEAVED,
- params.channels,
- params.freq,
- 1, // Allow ALSA to resample.
- latency);
- if (err != 0) {
- LOG(LS_ERROR) << "snd_pcm_set_params(): " << GetError(err);
- goto fail;
- }
-
- err = symbol_table_.snd_pcm_prepare()(handle);
- if (err != 0) {
- LOG(LS_ERROR) << "snd_pcm_prepare(): " << GetError(err);
- goto fail;
- }
-
- stream = (this->*start_fn)(
- handle,
- FrameSize(params),
- // We set the wait time to twice the requested latency, so that wait
- // timeouts should be rare.
- 2 * latency / rtc::kNumMicrosecsPerMillisec,
- params.flags,
- params.freq);
- if (stream) {
- return stream;
- }
- // Else fall through.
-
- fail:
- err = symbol_table_.snd_pcm_close()(handle);
- if (err != 0) {
- LOG(LS_ERROR) << "snd_pcm_close(): " << GetError(err);
- }
- return NULL;
-}
-
-SoundOutputStreamInterface *AlsaSoundSystem::StartOutputStream(
- snd_pcm_t *handle,
- size_t frame_size,
- int wait_timeout_ms,
- int flags,
- int freq) {
- // Nothing to do here but instantiate the stream.
- return new AlsaOutputStream(
- this, handle, frame_size, wait_timeout_ms, flags, freq);
-}
-
-SoundInputStreamInterface *AlsaSoundSystem::StartInputStream(
- snd_pcm_t *handle,
- size_t frame_size,
- int wait_timeout_ms,
- int flags,
- int freq) {
- // Output streams start automatically once enough data has been written, but
- // input streams must be started manually or else snd_pcm_wait() will never
- // return true.
- int err;
- err = symbol_table_.snd_pcm_start()(handle);
- if (err != 0) {
- LOG(LS_ERROR) << "snd_pcm_start(): " << GetError(err);
- return NULL;
- }
- return new AlsaInputStream(
- this, handle, frame_size, wait_timeout_ms, flags, freq);
-}
-
-inline const char *AlsaSoundSystem::GetError(int err) {
- return symbol_table_.snd_strerror()(err);
-}
-
-} // namespace cricket
diff --git a/sound/alsasoundsystem.h b/sound/alsasoundsystem.h
deleted file mode 100644
index b3abfb0..0000000
--- a/sound/alsasoundsystem.h
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_ALSASOUNDSYSTEM_H_
-#define TALK_SOUND_ALSASOUNDSYSTEM_H_
-
-#include "talk/sound/alsasymboltable.h"
-#include "talk/sound/soundsysteminterface.h"
-#include "webrtc/base/constructormagic.h"
-
-namespace cricket {
-
-class AlsaStream;
-class AlsaInputStream;
-class AlsaOutputStream;
-
-// Sound system implementation for ALSA, the predominant sound device API on
-// Linux (but typically not used directly by applications anymore).
-class AlsaSoundSystem : public SoundSystemInterface {
- friend class AlsaStream;
- friend class AlsaInputStream;
- friend class AlsaOutputStream;
- public:
- static SoundSystemInterface *Create() {
- return new AlsaSoundSystem();
- }
-
- AlsaSoundSystem();
-
- virtual ~AlsaSoundSystem();
-
- virtual bool Init();
- virtual void Terminate();
-
- virtual bool EnumeratePlaybackDevices(SoundDeviceLocatorList *devices);
- virtual bool EnumerateCaptureDevices(SoundDeviceLocatorList *devices);
-
- virtual bool GetDefaultPlaybackDevice(SoundDeviceLocator **device);
- virtual bool GetDefaultCaptureDevice(SoundDeviceLocator **device);
-
- virtual SoundOutputStreamInterface *OpenPlaybackDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms);
- virtual SoundInputStreamInterface *OpenCaptureDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms);
-
- virtual const char *GetName() const;
-
- private:
- bool IsInitialized() { return initialized_; }
-
- bool EnumerateDevices(SoundDeviceLocatorList *devices,
- bool capture_not_playback);
-
- bool GetDefaultDevice(SoundDeviceLocator **device);
-
- static size_t FrameSize(const OpenParams ¶ms);
-
- template <typename StreamInterface>
- StreamInterface *OpenDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms,
- snd_pcm_stream_t type,
- StreamInterface *(AlsaSoundSystem::*start_fn)(
- snd_pcm_t *handle,
- size_t frame_size,
- int wait_timeout_ms,
- int flags,
- int freq));
-
- SoundOutputStreamInterface *StartOutputStream(
- snd_pcm_t *handle,
- size_t frame_size,
- int wait_timeout_ms,
- int flags,
- int freq);
-
- SoundInputStreamInterface *StartInputStream(
- snd_pcm_t *handle,
- size_t frame_size,
- int wait_timeout_ms,
- int flags,
- int freq);
-
- const char *GetError(int err);
-
- bool initialized_;
- AlsaSymbolTable symbol_table_;
-
- DISALLOW_COPY_AND_ASSIGN(AlsaSoundSystem);
-};
-
-} // namespace cricket
-
-#endif // TALK_SOUND_ALSASOUNDSYSTEM_H_
diff --git a/sound/alsasymboltable.cc b/sound/alsasymboltable.cc
deleted file mode 100644
index 570b4b4..0000000
--- a/sound/alsasymboltable.cc
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/sound/alsasymboltable.h"
-
-namespace cricket {
-
-#define LATE_BINDING_SYMBOL_TABLE_CLASS_NAME ALSA_SYMBOLS_CLASS_NAME
-#define LATE_BINDING_SYMBOL_TABLE_SYMBOLS_LIST ALSA_SYMBOLS_LIST
-#define LATE_BINDING_SYMBOL_TABLE_DLL_NAME "libasound.so.2"
-#include "webrtc/base/latebindingsymboltable.cc.def"
-
-} // namespace cricket
diff --git a/sound/alsasymboltable.h b/sound/alsasymboltable.h
deleted file mode 100644
index 98f1645..0000000
--- a/sound/alsasymboltable.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_ALSASYMBOLTABLE_H_
-#define TALK_SOUND_ALSASYMBOLTABLE_H_
-
-#include <alsa/asoundlib.h>
-
-#include "webrtc/base/latebindingsymboltable.h"
-
-namespace cricket {
-
-#define ALSA_SYMBOLS_CLASS_NAME AlsaSymbolTable
-// The ALSA symbols we need, as an X-Macro list.
-// This list must contain precisely every libasound function that is used in
-// alsasoundsystem.cc.
-#define ALSA_SYMBOLS_LIST \
- X(snd_device_name_free_hint) \
- X(snd_device_name_get_hint) \
- X(snd_device_name_hint) \
- X(snd_pcm_avail_update) \
- X(snd_pcm_close) \
- X(snd_pcm_delay) \
- X(snd_pcm_drop) \
- X(snd_pcm_open) \
- X(snd_pcm_prepare) \
- X(snd_pcm_readi) \
- X(snd_pcm_recover) \
- X(snd_pcm_set_params) \
- X(snd_pcm_start) \
- X(snd_pcm_stream) \
- X(snd_pcm_wait) \
- X(snd_pcm_writei) \
- X(snd_strerror)
-
-#define LATE_BINDING_SYMBOL_TABLE_CLASS_NAME ALSA_SYMBOLS_CLASS_NAME
-#define LATE_BINDING_SYMBOL_TABLE_SYMBOLS_LIST ALSA_SYMBOLS_LIST
-#include "webrtc/base/latebindingsymboltable.h.def"
-
-} // namespace cricket
-
-#endif // TALK_SOUND_ALSASYMBOLTABLE_H_
diff --git a/sound/automaticallychosensoundsystem.h b/sound/automaticallychosensoundsystem.h
deleted file mode 100644
index 10ea0df..0000000
--- a/sound/automaticallychosensoundsystem.h
+++ /dev/null
@@ -1,105 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_AUTOMATICALLYCHOSENSOUNDSYSTEM_H_
-#define TALK_SOUND_AUTOMATICALLYCHOSENSOUNDSYSTEM_H_
-
-#include "talk/sound/soundsysteminterface.h"
-#include "talk/sound/soundsystemproxy.h"
-#include "webrtc/base/common.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
-
-namespace cricket {
-
-// A function type that creates an instance of a sound system implementation.
-typedef SoundSystemInterface *(*SoundSystemCreator)();
-
-// An AutomaticallyChosenSoundSystem is a sound system proxy that defers to
-// an instance of the first sound system implementation in a list that
-// successfully initializes.
-template <const SoundSystemCreator kSoundSystemCreators[], int kNumSoundSystems>
-class AutomaticallyChosenSoundSystem : public SoundSystemProxy {
- public:
- // Chooses and initializes the underlying sound system.
- virtual bool Init();
- // Terminates the underlying sound system implementation, but caches it for
- // future re-use.
- virtual void Terminate();
-
- virtual const char *GetName() const;
-
- private:
- rtc::scoped_ptr<SoundSystemInterface> sound_systems_[kNumSoundSystems];
-};
-
-template <const SoundSystemCreator kSoundSystemCreators[], int kNumSoundSystems>
-bool AutomaticallyChosenSoundSystem<kSoundSystemCreators,
- kNumSoundSystems>::Init() {
- if (wrapped_) {
- return true;
- }
- for (int i = 0; i < kNumSoundSystems; ++i) {
- if (!sound_systems_[i].get()) {
- sound_systems_[i].reset((*kSoundSystemCreators[i])());
- }
- if (sound_systems_[i]->Init()) {
- // This is the first sound system in the list to successfully
- // initialize, so we're done.
- wrapped_ = sound_systems_[i].get();
- break;
- }
- // Else it failed to initialize, so try the remaining ones.
- }
- if (!wrapped_) {
- LOG(LS_ERROR) << "Failed to find a usable sound system";
- return false;
- }
- LOG(LS_INFO) << "Selected " << wrapped_->GetName() << " sound system";
- return true;
-}
-
-template <const SoundSystemCreator kSoundSystemCreators[], int kNumSoundSystems>
-void AutomaticallyChosenSoundSystem<kSoundSystemCreators,
- kNumSoundSystems>::Terminate() {
- if (!wrapped_) {
- return;
- }
- wrapped_->Terminate();
- wrapped_ = NULL;
- // We do not free the scoped_ptrs because we may be re-init'ed soon.
-}
-
-template <const SoundSystemCreator kSoundSystemCreators[], int kNumSoundSystems>
-const char *AutomaticallyChosenSoundSystem<kSoundSystemCreators,
- kNumSoundSystems>::GetName() const {
- return wrapped_ ? wrapped_->GetName() : "automatic";
-}
-
-} // namespace cricket
-
-#endif // TALK_SOUND_AUTOMATICALLYCHOSENSOUNDSYSTEM_H_
diff --git a/sound/automaticallychosensoundsystem_unittest.cc b/sound/automaticallychosensoundsystem_unittest.cc
deleted file mode 100644
index 813828d..0000000
--- a/sound/automaticallychosensoundsystem_unittest.cc
+++ /dev/null
@@ -1,214 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/sound/automaticallychosensoundsystem.h"
-#include "talk/sound/nullsoundsystem.h"
-#include "webrtc/base/gunit.h"
-
-namespace cricket {
-
-class NeverFailsToFailSoundSystem : public NullSoundSystem {
- public:
- // Overrides superclass.
- virtual bool Init() {
- return false;
- }
-
- static SoundSystemInterface *Create() {
- return new NeverFailsToFailSoundSystem();
- }
-};
-
-class InitCheckingSoundSystem1 : public NullSoundSystem {
- public:
- // Overrides superclass.
- virtual bool Init() {
- created_ = true;
- return true;
- }
-
- static SoundSystemInterface *Create() {
- return new InitCheckingSoundSystem1();
- }
-
- static bool created_;
-};
-
-bool InitCheckingSoundSystem1::created_ = false;
-
-class InitCheckingSoundSystem2 : public NullSoundSystem {
- public:
- // Overrides superclass.
- virtual bool Init() {
- created_ = true;
- return true;
- }
-
- static SoundSystemInterface *Create() {
- return new InitCheckingSoundSystem2();
- }
-
- static bool created_;
-};
-
-bool InitCheckingSoundSystem2::created_ = false;
-
-class DeletionCheckingSoundSystem1 : public NeverFailsToFailSoundSystem {
- public:
- virtual ~DeletionCheckingSoundSystem1() {
- deleted_ = true;
- }
-
- static SoundSystemInterface *Create() {
- return new DeletionCheckingSoundSystem1();
- }
-
- static bool deleted_;
-};
-
-bool DeletionCheckingSoundSystem1::deleted_ = false;
-
-class DeletionCheckingSoundSystem2 : public NeverFailsToFailSoundSystem {
- public:
- virtual ~DeletionCheckingSoundSystem2() {
- deleted_ = true;
- }
-
- static SoundSystemInterface *Create() {
- return new DeletionCheckingSoundSystem2();
- }
-
- static bool deleted_;
-};
-
-bool DeletionCheckingSoundSystem2::deleted_ = false;
-
-class DeletionCheckingSoundSystem3 : public NullSoundSystem {
- public:
- virtual ~DeletionCheckingSoundSystem3() {
- deleted_ = true;
- }
-
- static SoundSystemInterface *Create() {
- return new DeletionCheckingSoundSystem3();
- }
-
- static bool deleted_;
-};
-
-bool DeletionCheckingSoundSystem3::deleted_ = false;
-
-extern const SoundSystemCreator kSingleSystemFailingCreators[] = {
- &NeverFailsToFailSoundSystem::Create,
-};
-
-TEST(AutomaticallyChosenSoundSystem, SingleSystemFailing) {
- AutomaticallyChosenSoundSystem<
- kSingleSystemFailingCreators,
- ARRAY_SIZE(kSingleSystemFailingCreators)> sound_system;
- EXPECT_FALSE(sound_system.Init());
-}
-
-extern const SoundSystemCreator kSingleSystemSucceedingCreators[] = {
- &NullSoundSystem::Create,
-};
-
-TEST(AutomaticallyChosenSoundSystem, SingleSystemSucceeding) {
- AutomaticallyChosenSoundSystem<
- kSingleSystemSucceedingCreators,
- ARRAY_SIZE(kSingleSystemSucceedingCreators)> sound_system;
- EXPECT_TRUE(sound_system.Init());
-}
-
-extern const SoundSystemCreator
- kFailedFirstSystemResultsInUsingSecondCreators[] = {
- &NeverFailsToFailSoundSystem::Create,
- &NullSoundSystem::Create,
-};
-
-TEST(AutomaticallyChosenSoundSystem, FailedFirstSystemResultsInUsingSecond) {
- AutomaticallyChosenSoundSystem<
- kFailedFirstSystemResultsInUsingSecondCreators,
- ARRAY_SIZE(kFailedFirstSystemResultsInUsingSecondCreators)> sound_system;
- EXPECT_TRUE(sound_system.Init());
-}
-
-extern const SoundSystemCreator kEarlierEntriesHavePriorityCreators[] = {
- &InitCheckingSoundSystem1::Create,
- &InitCheckingSoundSystem2::Create,
-};
-
-TEST(AutomaticallyChosenSoundSystem, EarlierEntriesHavePriority) {
- AutomaticallyChosenSoundSystem<
- kEarlierEntriesHavePriorityCreators,
- ARRAY_SIZE(kEarlierEntriesHavePriorityCreators)> sound_system;
- InitCheckingSoundSystem1::created_ = false;
- InitCheckingSoundSystem2::created_ = false;
- EXPECT_TRUE(sound_system.Init());
- EXPECT_TRUE(InitCheckingSoundSystem1::created_);
- EXPECT_FALSE(InitCheckingSoundSystem2::created_);
-}
-
-extern const SoundSystemCreator kManySoundSystemsCreators[] = {
- &NullSoundSystem::Create,
- &NullSoundSystem::Create,
- &NullSoundSystem::Create,
- &NullSoundSystem::Create,
- &NullSoundSystem::Create,
- &NullSoundSystem::Create,
- &NullSoundSystem::Create,
-};
-
-TEST(AutomaticallyChosenSoundSystem, ManySoundSystems) {
- AutomaticallyChosenSoundSystem<
- kManySoundSystemsCreators,
- ARRAY_SIZE(kManySoundSystemsCreators)> sound_system;
- EXPECT_TRUE(sound_system.Init());
-}
-
-extern const SoundSystemCreator kDeletesAllCreatedSoundSystemsCreators[] = {
- &DeletionCheckingSoundSystem1::Create,
- &DeletionCheckingSoundSystem2::Create,
- &DeletionCheckingSoundSystem3::Create,
-};
-
-TEST(AutomaticallyChosenSoundSystem, DeletesAllCreatedSoundSystems) {
- typedef AutomaticallyChosenSoundSystem<
- kDeletesAllCreatedSoundSystemsCreators,
- ARRAY_SIZE(kDeletesAllCreatedSoundSystemsCreators)> TestSoundSystem;
- TestSoundSystem *sound_system = new TestSoundSystem();
- DeletionCheckingSoundSystem1::deleted_ = false;
- DeletionCheckingSoundSystem2::deleted_ = false;
- DeletionCheckingSoundSystem3::deleted_ = false;
- EXPECT_TRUE(sound_system->Init());
- delete sound_system;
- EXPECT_TRUE(DeletionCheckingSoundSystem1::deleted_);
- EXPECT_TRUE(DeletionCheckingSoundSystem2::deleted_);
- EXPECT_TRUE(DeletionCheckingSoundSystem3::deleted_);
-}
-
-} // namespace cricket
diff --git a/sound/linuxsoundsystem.cc b/sound/linuxsoundsystem.cc
deleted file mode 100644
index 7980a15..0000000
--- a/sound/linuxsoundsystem.cc
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/sound/linuxsoundsystem.h"
-
-#include "talk/sound/alsasoundsystem.h"
-#include "talk/sound/pulseaudiosoundsystem.h"
-
-namespace cricket {
-
-const SoundSystemCreator kLinuxSoundSystemCreators[] = {
-#ifdef HAVE_LIBPULSE
- &PulseAudioSoundSystem::Create,
-#endif
- &AlsaSoundSystem::Create,
-};
-
-} // namespace cricket
diff --git a/sound/linuxsoundsystem.h b/sound/linuxsoundsystem.h
deleted file mode 100644
index eb48b88..0000000
--- a/sound/linuxsoundsystem.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_LINUXSOUNDSYSTEM_H_
-#define TALK_SOUND_LINUXSOUNDSYSTEM_H_
-
-#include "talk/sound/automaticallychosensoundsystem.h"
-
-namespace cricket {
-
-extern const SoundSystemCreator kLinuxSoundSystemCreators[
-#ifdef HAVE_LIBPULSE
- 2
-#else
- 1
-#endif
- ];
-
-// The vast majority of Linux systems use ALSA for the device-level sound API,
-// but an increasing number are using PulseAudio for the application API and
-// only using ALSA internally in PulseAudio itself. But like everything on
-// Linux this is user-configurable, so we need to support both and choose the
-// right one at run-time.
-// PulseAudioSoundSystem is designed to only successfully initialize if
-// PulseAudio is installed and running, and if it is running then direct device
-// access using ALSA typically won't work, so if PulseAudioSoundSystem
-// initializes then we choose that. Otherwise we choose ALSA.
-typedef AutomaticallyChosenSoundSystem<
- kLinuxSoundSystemCreators,
- ARRAY_SIZE(kLinuxSoundSystemCreators)> LinuxSoundSystem;
-
-} // namespace cricket
-
-#endif // TALK_SOUND_LINUXSOUNDSYSTEM_H_
diff --git a/sound/nullsoundsystem.cc b/sound/nullsoundsystem.cc
deleted file mode 100644
index 3408d4c..0000000
--- a/sound/nullsoundsystem.cc
+++ /dev/null
@@ -1,174 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/sound/nullsoundsystem.h"
-
-#include "talk/sound/sounddevicelocator.h"
-#include "talk/sound/soundinputstreaminterface.h"
-#include "talk/sound/soundoutputstreaminterface.h"
-#include "webrtc/base/logging.h"
-
-namespace rtc {
-
-class Thread;
-
-}
-
-namespace cricket {
-
-// Name used for the single device and the sound system itself.
-static const char kNullName[] = "null";
-
-class NullSoundDeviceLocator : public SoundDeviceLocator {
- public:
- NullSoundDeviceLocator() : SoundDeviceLocator(kNullName, kNullName) {}
-
- virtual SoundDeviceLocator *Copy() const {
- return new NullSoundDeviceLocator();
- }
-};
-
-class NullSoundInputStream : public SoundInputStreamInterface {
- public:
- virtual bool StartReading() {
- return true;
- }
-
- virtual bool StopReading() {
- return true;
- }
-
- virtual bool GetVolume(int *volume) {
- *volume = SoundSystemInterface::kMinVolume;
- return true;
- }
-
- virtual bool SetVolume(int volume) {
- return false;
- }
-
- virtual bool Close() {
- return true;
- }
-
- virtual int LatencyUsecs() {
- return 0;
- }
-};
-
-class NullSoundOutputStream : public SoundOutputStreamInterface {
- public:
- virtual bool EnableBufferMonitoring() {
- return true;
- }
-
- virtual bool DisableBufferMonitoring() {
- return true;
- }
-
- virtual bool WriteSamples(const void *sample_data,
- size_t size) {
- LOG(LS_VERBOSE) << "Got " << size << " bytes of playback samples";
- return true;
- }
-
- virtual bool GetVolume(int *volume) {
- *volume = SoundSystemInterface::kMinVolume;
- return true;
- }
-
- virtual bool SetVolume(int volume) {
- return false;
- }
-
- virtual bool Close() {
- return true;
- }
-
- virtual int LatencyUsecs() {
- return 0;
- }
-};
-
-NullSoundSystem::~NullSoundSystem() {
-}
-
-bool NullSoundSystem::Init() {
- return true;
-}
-
-void NullSoundSystem::Terminate() {
- // Nothing to do.
-}
-
-bool NullSoundSystem::EnumeratePlaybackDevices(
- SoundSystemInterface::SoundDeviceLocatorList *devices) {
- ClearSoundDeviceLocatorList(devices);
- SoundDeviceLocator *device;
- GetDefaultPlaybackDevice(&device);
- devices->push_back(device);
- return true;
-}
-
-bool NullSoundSystem::EnumerateCaptureDevices(
- SoundSystemInterface::SoundDeviceLocatorList *devices) {
- ClearSoundDeviceLocatorList(devices);
- SoundDeviceLocator *device;
- GetDefaultCaptureDevice(&device);
- devices->push_back(device);
- return true;
-}
-
-bool NullSoundSystem::GetDefaultPlaybackDevice(
- SoundDeviceLocator **device) {
- *device = new NullSoundDeviceLocator();
- return true;
-}
-
-bool NullSoundSystem::GetDefaultCaptureDevice(
- SoundDeviceLocator **device) {
- *device = new NullSoundDeviceLocator();
- return true;
-}
-
-SoundOutputStreamInterface *NullSoundSystem::OpenPlaybackDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms) {
- return new NullSoundOutputStream();
-}
-
-SoundInputStreamInterface *NullSoundSystem::OpenCaptureDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms) {
- return new NullSoundInputStream();
-}
-
-const char *NullSoundSystem::GetName() const {
- return kNullName;
-}
-
-} // namespace cricket
diff --git a/sound/nullsoundsystem.h b/sound/nullsoundsystem.h
deleted file mode 100644
index 3edb4f9..0000000
--- a/sound/nullsoundsystem.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_NULLSOUNDSYSTEM_H_
-#define TALK_SOUND_NULLSOUNDSYSTEM_H_
-
-#include "talk/sound/soundsysteminterface.h"
-
-namespace cricket {
-
-class SoundDeviceLocator;
-class SoundInputStreamInterface;
-class SoundOutputStreamInterface;
-
-// A simple reference sound system that drops output samples and generates
-// no input samples.
-class NullSoundSystem : public SoundSystemInterface {
- public:
- static SoundSystemInterface *Create() {
- return new NullSoundSystem();
- }
-
- virtual ~NullSoundSystem();
-
- virtual bool Init();
- virtual void Terminate();
-
- virtual bool EnumeratePlaybackDevices(SoundDeviceLocatorList *devices);
- virtual bool EnumerateCaptureDevices(SoundDeviceLocatorList *devices);
-
- virtual SoundOutputStreamInterface *OpenPlaybackDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms);
- virtual SoundInputStreamInterface *OpenCaptureDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms);
-
- virtual bool GetDefaultPlaybackDevice(SoundDeviceLocator **device);
- virtual bool GetDefaultCaptureDevice(SoundDeviceLocator **device);
-
- virtual const char *GetName() const;
-};
-
-} // namespace cricket
-
-#endif // TALK_SOUND_NULLSOUNDSYSTEM_H_
diff --git a/sound/nullsoundsystemfactory.cc b/sound/nullsoundsystemfactory.cc
deleted file mode 100644
index 089d51f..0000000
--- a/sound/nullsoundsystemfactory.cc
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/sound/nullsoundsystemfactory.h"
-
-#include "talk/sound/nullsoundsystem.h"
-
-namespace cricket {
-
-NullSoundSystemFactory::NullSoundSystemFactory() {
-}
-
-NullSoundSystemFactory::~NullSoundSystemFactory() {
-}
-
-bool NullSoundSystemFactory::SetupInstance() {
- instance_.reset(new NullSoundSystem());
- return true;
-}
-
-void NullSoundSystemFactory::CleanupInstance() {
- instance_.reset();
-}
-
-} // namespace cricket
diff --git a/sound/nullsoundsystemfactory.h b/sound/nullsoundsystemfactory.h
deleted file mode 100644
index 71ae980..0000000
--- a/sound/nullsoundsystemfactory.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_NULLSOUNDSYSTEMFACTORY_H_
-#define TALK_SOUND_NULLSOUNDSYSTEMFACTORY_H_
-
-#include "talk/sound/soundsystemfactory.h"
-
-namespace cricket {
-
-// A SoundSystemFactory that always returns a NullSoundSystem. Intended for
-// testing.
-class NullSoundSystemFactory : public SoundSystemFactory {
- public:
- NullSoundSystemFactory();
- virtual ~NullSoundSystemFactory();
-
- protected:
- // Inherited from SoundSystemFactory.
- virtual bool SetupInstance();
- virtual void CleanupInstance();
-};
-
-} // namespace cricket
-
-#endif // TALK_SOUND_NULLSOUNDSYSTEMFACTORY_H_
diff --git a/sound/platformsoundsystem.cc b/sound/platformsoundsystem.cc
deleted file mode 100644
index c39fc83..0000000
--- a/sound/platformsoundsystem.cc
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/sound/platformsoundsystem.h"
-
-#include "webrtc/base/common.h"
-#ifdef LINUX
-#include "talk/sound/linuxsoundsystem.h"
-#else
-#include "talk/sound/nullsoundsystem.h"
-#endif
-
-namespace cricket {
-
-SoundSystemInterface *CreatePlatformSoundSystem() {
-#ifdef LINUX
- return new LinuxSoundSystem();
-#else
- ASSERT(false && "Not implemented");
- return new NullSoundSystem();
-#endif
-}
-
-} // namespace cricket
diff --git a/sound/platformsoundsystem.h b/sound/platformsoundsystem.h
deleted file mode 100644
index 1a8d214..0000000
--- a/sound/platformsoundsystem.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_PLATFORMSOUNDSYSTEM_H_
-#define TALK_SOUND_PLATFORMSOUNDSYSTEM_H_
-
-namespace cricket {
-
-class SoundSystemInterface;
-
-// Creates the sound system implementation for this platform.
-SoundSystemInterface *CreatePlatformSoundSystem();
-
-} // namespace cricket
-
-#endif // TALK_SOUND_PLATFORMSOUNDSYSTEM_H_
diff --git a/sound/platformsoundsystemfactory.cc b/sound/platformsoundsystemfactory.cc
deleted file mode 100644
index 6c69954..0000000
--- a/sound/platformsoundsystemfactory.cc
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/sound/platformsoundsystemfactory.h"
-
-#include "talk/sound/platformsoundsystem.h"
-#include "talk/sound/soundsysteminterface.h"
-
-namespace cricket {
-
-PlatformSoundSystemFactory::PlatformSoundSystemFactory() {
-}
-
-PlatformSoundSystemFactory::~PlatformSoundSystemFactory() {
-}
-
-bool PlatformSoundSystemFactory::SetupInstance() {
- if (!instance_.get()) {
- instance_.reset(CreatePlatformSoundSystem());
- }
- if (!instance_->Init()) {
- LOG(LS_ERROR) << "Can't initialize platform's sound system";
- return false;
- }
- return true;
-}
-
-void PlatformSoundSystemFactory::CleanupInstance() {
- instance_->Terminate();
- // We do not delete the sound system because we might be re-initialized soon.
-}
-
-} // namespace cricket
diff --git a/sound/platformsoundsystemfactory.h b/sound/platformsoundsystemfactory.h
deleted file mode 100644
index 63ca863..0000000
--- a/sound/platformsoundsystemfactory.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_PLATFORMSOUNDSYSTEMFACTORY_H_
-#define TALK_SOUND_PLATFORMSOUNDSYSTEMFACTORY_H_
-
-#include "talk/sound/soundsystemfactory.h"
-
-namespace cricket {
-
-// A SoundSystemFactory that returns the platform's native sound system
-// implementation.
-class PlatformSoundSystemFactory : public SoundSystemFactory {
- public:
- PlatformSoundSystemFactory();
- virtual ~PlatformSoundSystemFactory();
-
- protected:
- // Inherited from SoundSystemFactory.
- virtual bool SetupInstance();
- virtual void CleanupInstance();
-};
-
-} // namespace cricket
-
-#endif // TALK_SOUND_PLATFORMSOUNDSYSTEMFACTORY_H_
-
-
diff --git a/sound/pulseaudiosoundsystem.cc b/sound/pulseaudiosoundsystem.cc
deleted file mode 100644
index 24eea5c..0000000
--- a/sound/pulseaudiosoundsystem.cc
+++ /dev/null
@@ -1,1559 +0,0 @@
-/*
- * libjingle
- * Copyright 2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/sound/pulseaudiosoundsystem.h"
-
-#ifdef HAVE_LIBPULSE
-
-#include "talk/sound/sounddevicelocator.h"
-#include "talk/sound/soundinputstreaminterface.h"
-#include "talk/sound/soundoutputstreaminterface.h"
-#include "webrtc/base/common.h"
-#include "webrtc/base/fileutils.h" // for GetApplicationName()
-#include "webrtc/base/logging.h"
-#include "webrtc/base/timeutils.h"
-#include "webrtc/base/worker.h"
-
-namespace cricket {
-
-// First PulseAudio protocol version that supports PA_STREAM_ADJUST_LATENCY.
-static const uint32_t kAdjustLatencyProtocolVersion = 13;
-
-// Lookup table from the cricket format enum in soundsysteminterface.h to
-// Pulse's enums.
-static const pa_sample_format_t kCricketFormatToPulseFormatTable[] = {
- // The order here must match the order in soundsysteminterface.h
- PA_SAMPLE_S16LE,
-};
-
-// Some timing constants for optimal operation. See
-// https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008-January/001170.html
-// for a good explanation of some of the factors that go into this.
-
-// Playback.
-
-// For playback, there is a round-trip delay to fill the server-side playback
-// buffer, so setting too low of a latency is a buffer underflow risk. We will
-// automatically increase the latency if a buffer underflow does occur, but we
-// also enforce a sane minimum at start-up time. Anything lower would be
-// virtually guaranteed to underflow at least once, so there's no point in
-// allowing lower latencies.
-static const int kPlaybackLatencyMinimumMsecs = 20;
-// Every time a playback stream underflows, we will reconfigure it with target
-// latency that is greater by this amount.
-static const int kPlaybackLatencyIncrementMsecs = 20;
-// We also need to configure a suitable request size. Too small and we'd burn
-// CPU from the overhead of transfering small amounts of data at once. Too large
-// and the amount of data remaining in the buffer right before refilling it
-// would be a buffer underflow risk. We set it to half of the buffer size.
-static const int kPlaybackRequestFactor = 2;
-
-// Capture.
-
-// For capture, low latency is not a buffer overflow risk, but it makes us burn
-// CPU from the overhead of transfering small amounts of data at once, so we set
-// a recommended value that we use for the kLowLatency constant (but if the user
-// explicitly requests something lower then we will honour it).
-// 1ms takes about 6-7% CPU. 5ms takes about 5%. 10ms takes about 4.x%.
-static const int kLowCaptureLatencyMsecs = 10;
-// There is a round-trip delay to ack the data to the server, so the
-// server-side buffer needs extra space to prevent buffer overflow. 20ms is
-// sufficient, but there is no penalty to making it bigger, so we make it huge.
-// (750ms is libpulse's default value for the _total_ buffer size in the
-// kNoLatencyRequirements case.)
-static const int kCaptureBufferExtraMsecs = 750;
-
-static void FillPlaybackBufferAttr(int latency,
- pa_buffer_attr *attr) {
- attr->maxlength = latency;
- attr->tlength = latency;
- attr->minreq = latency / kPlaybackRequestFactor;
- attr->prebuf = attr->tlength - attr->minreq;
- LOG(LS_VERBOSE) << "Configuring latency = " << attr->tlength << ", minreq = "
- << attr->minreq << ", minfill = " << attr->prebuf;
-}
-
-static pa_volume_t CricketVolumeToPulseVolume(int volume) {
- // PA's volume space goes from 0% at PA_VOLUME_MUTED (value 0) to 100% at
- // PA_VOLUME_NORM (value 0x10000). It can also go beyond 100% up to
- // PA_VOLUME_MAX (value UINT32_MAX-1), but using that is probably unwise.
- // We just linearly map the 0-255 scale of SoundSystemInterface onto
- // PA_VOLUME_MUTED-PA_VOLUME_NORM. If the programmer exceeds kMaxVolume then
- // they can access the over-100% features of PA.
- return PA_VOLUME_MUTED + (PA_VOLUME_NORM - PA_VOLUME_MUTED) *
- volume / SoundSystemInterface::kMaxVolume;
-}
-
-static int PulseVolumeToCricketVolume(pa_volume_t pa_volume) {
- return SoundSystemInterface::kMinVolume +
- (SoundSystemInterface::kMaxVolume - SoundSystemInterface::kMinVolume) *
- pa_volume / PA_VOLUME_NORM;
-}
-
-static pa_volume_t MaxChannelVolume(pa_cvolume *channel_volumes) {
- pa_volume_t pa_volume = PA_VOLUME_MUTED; // Minimum possible value.
- for (int i = 0; i < channel_volumes->channels; ++i) {
- if (pa_volume < channel_volumes->values[i]) {
- pa_volume = channel_volumes->values[i];
- }
- }
- return pa_volume;
-}
-
-class PulseAudioDeviceLocator : public SoundDeviceLocator {
- public:
- PulseAudioDeviceLocator(const std::string &name,
- const std::string &device_name)
- : SoundDeviceLocator(name, device_name) {
- }
-
- virtual SoundDeviceLocator *Copy() const {
- return new PulseAudioDeviceLocator(*this);
- }
-};
-
-// Functionality that is common to both PulseAudioInputStream and
-// PulseAudioOutputStream.
-class PulseAudioStream {
- public:
- PulseAudioStream(PulseAudioSoundSystem *pulse, pa_stream *stream, int flags)
- : pulse_(pulse), stream_(stream), flags_(flags) {
- }
-
- ~PulseAudioStream() {
- // Close() should have been called during the containing class's destructor.
- ASSERT(stream_ == NULL);
- }
-
- // Must be called with the lock held.
- bool Close() {
- if (!IsClosed()) {
- // Unset this here so that we don't get a TERMINATED callback.
- symbol_table()->pa_stream_set_state_callback()(stream_, NULL, NULL);
- if (symbol_table()->pa_stream_disconnect()(stream_) != 0) {
- LOG(LS_ERROR) << "Can't disconnect stream";
- // Continue and return true anyways.
- }
- symbol_table()->pa_stream_unref()(stream_);
- stream_ = NULL;
- }
- return true;
- }
-
- // Must be called with the lock held.
- int LatencyUsecs() {
- if (!(flags_ & SoundSystemInterface::FLAG_REPORT_LATENCY)) {
- return 0;
- }
-
- pa_usec_t latency;
- int negative;
- Lock();
- int re = symbol_table()->pa_stream_get_latency()(stream_, &latency,
- &negative);
- Unlock();
- if (re != 0) {
- LOG(LS_ERROR) << "Can't query latency";
- // We'd rather continue playout/capture with an incorrect delay than stop
- // it altogether, so return a valid value.
- return 0;
- }
- if (negative) {
- // The delay can be negative for monitoring streams if the captured
- // samples haven't been played yet. In such a case, "latency" contains the
- // magnitude, so we must negate it to get the real value.
- return -latency;
- } else {
- return latency;
- }
- }
-
- PulseAudioSoundSystem *pulse() {
- return pulse_;
- }
-
- PulseAudioSymbolTable *symbol_table() {
- return &pulse()->symbol_table_;
- }
-
- pa_stream *stream() {
- ASSERT(stream_ != NULL);
- return stream_;
- }
-
- bool IsClosed() {
- return stream_ == NULL;
- }
-
- void Lock() {
- pulse()->Lock();
- }
-
- void Unlock() {
- pulse()->Unlock();
- }
-
- private:
- PulseAudioSoundSystem *pulse_;
- pa_stream *stream_;
- int flags_;
-
- DISALLOW_COPY_AND_ASSIGN(PulseAudioStream);
-};
-
-// Implementation of an input stream. See soundinputstreaminterface.h regarding
-// thread-safety.
-class PulseAudioInputStream :
- public SoundInputStreamInterface,
- private rtc::Worker {
-
- struct GetVolumeCallbackData {
- PulseAudioInputStream *instance;
- pa_cvolume *channel_volumes;
- };
-
- struct GetSourceChannelCountCallbackData {
- PulseAudioInputStream *instance;
- uint8_t *channels;
- };
-
- public:
- PulseAudioInputStream(PulseAudioSoundSystem *pulse,
- pa_stream *stream,
- int flags)
- : stream_(pulse, stream, flags),
- temp_sample_data_(NULL),
- temp_sample_data_size_(0) {
- // This callback seems to never be issued, but let's set it anyways.
- symbol_table()->pa_stream_set_overflow_callback()(stream, &OverflowCallback,
- NULL);
- }
-
- virtual ~PulseAudioInputStream() {
- bool success = Close();
- // We need that to live.
- VERIFY(success);
- }
-
- virtual bool StartReading() {
- return StartWork();
- }
-
- virtual bool StopReading() {
- return StopWork();
- }
-
- virtual bool GetVolume(int *volume) {
- bool ret = false;
-
- Lock();
-
- // Unlike output streams, input streams have no concept of a stream volume,
- // only a device volume. So we have to retrieve the volume of the device
- // itself.
-
- pa_cvolume channel_volumes;
-
- GetVolumeCallbackData data;
- data.instance = this;
- data.channel_volumes = &channel_volumes;
-
- pa_operation *op = symbol_table()->pa_context_get_source_info_by_index()(
- stream_.pulse()->context_,
- symbol_table()->pa_stream_get_device_index()(stream_.stream()),
- &GetVolumeCallbackThunk,
- &data);
- if (!stream_.pulse()->FinishOperation(op)) {
- goto done;
- }
-
- if (data.channel_volumes) {
- // This pointer was never unset by the callback, so we must have received
- // an empty list of infos. This probably never happens, but we code for it
- // anyway.
- LOG(LS_ERROR) << "Did not receive GetVolumeCallback";
- goto done;
- }
-
- // We now have the volume for each channel. Each channel could have a
- // different volume if, e.g., the user went and changed the volumes in the
- // PA UI. To get a single volume for SoundSystemInterface we just take the
- // maximum. Ideally we'd do so with pa_cvolume_max, but it doesn't exist in
- // Hardy, so we do it manually.
- pa_volume_t pa_volume;
- pa_volume = MaxChannelVolume(&channel_volumes);
- // Now map onto the SoundSystemInterface range.
- *volume = PulseVolumeToCricketVolume(pa_volume);
-
- ret = true;
- done:
- Unlock();
- return ret;
- }
-
- virtual bool SetVolume(int volume) {
- bool ret = false;
- pa_volume_t pa_volume = CricketVolumeToPulseVolume(volume);
-
- Lock();
-
- // Unlike output streams, input streams have no concept of a stream volume,
- // only a device volume. So we have to change the volume of the device
- // itself.
-
- // The device may have a different number of channels than the stream and
- // their mapping may be different, so we don't want to use the channel count
- // from our sample spec. We could use PA_CHANNELS_MAX to cover our bases,
- // and the server allows that even if the device's channel count is lower,
- // but some buggy PA clients don't like that (the pavucontrol on Hardy dies
- // in an assert if the channel count is different). So instead we look up
- // the actual number of channels that the device has.
-
- uint8_t channels;
-
- GetSourceChannelCountCallbackData data;
- data.instance = this;
- data.channels = &channels;
-
- uint32_t device_index = symbol_table()->pa_stream_get_device_index()(
- stream_.stream());
-
- pa_operation *op = symbol_table()->pa_context_get_source_info_by_index()(
- stream_.pulse()->context_,
- device_index,
- &GetSourceChannelCountCallbackThunk,
- &data);
- if (!stream_.pulse()->FinishOperation(op)) {
- goto done;
- }
-
- if (data.channels) {
- // This pointer was never unset by the callback, so we must have received
- // an empty list of infos. This probably never happens, but we code for it
- // anyway.
- LOG(LS_ERROR) << "Did not receive GetSourceChannelCountCallback";
- goto done;
- }
-
- pa_cvolume channel_volumes;
- symbol_table()->pa_cvolume_set()(&channel_volumes, channels, pa_volume);
-
- op = symbol_table()->pa_context_set_source_volume_by_index()(
- stream_.pulse()->context_,
- device_index,
- &channel_volumes,
- // This callback merely logs errors.
- &SetVolumeCallback,
- NULL);
- if (!op) {
- LOG(LS_ERROR) << "pa_context_set_source_volume_by_index()";
- goto done;
- }
- // Don't need to wait for this to complete.
- symbol_table()->pa_operation_unref()(op);
-
- ret = true;
- done:
- Unlock();
- return ret;
- }
-
- virtual bool Close() {
- if (!StopReading()) {
- return false;
- }
- bool ret = true;
- if (!stream_.IsClosed()) {
- Lock();
- ret = stream_.Close();
- Unlock();
- }
- return ret;
- }
-
- virtual int LatencyUsecs() {
- return stream_.LatencyUsecs();
- }
-
- private:
- void Lock() {
- stream_.Lock();
- }
-
- void Unlock() {
- stream_.Unlock();
- }
-
- PulseAudioSymbolTable *symbol_table() {
- return stream_.symbol_table();
- }
-
- void EnableReadCallback() {
- symbol_table()->pa_stream_set_read_callback()(
- stream_.stream(),
- &ReadCallbackThunk,
- this);
- }
-
- void DisableReadCallback() {
- symbol_table()->pa_stream_set_read_callback()(
- stream_.stream(),
- NULL,
- NULL);
- }
-
- static void ReadCallbackThunk(pa_stream *unused1,
- size_t unused2,
- void *userdata) {
- PulseAudioInputStream *instance =
- static_cast<PulseAudioInputStream *>(userdata);
- instance->OnReadCallback();
- }
-
- void OnReadCallback() {
- // We get the data pointer and size now in order to save one Lock/Unlock
- // on OnMessage.
- if (symbol_table()->pa_stream_peek()(stream_.stream(),
- &temp_sample_data_,
- &temp_sample_data_size_) != 0) {
- LOG(LS_ERROR) << "Can't read data!";
- return;
- }
- // Since we consume the data asynchronously on a different thread, we have
- // to temporarily disable the read callback or else Pulse will call it
- // continuously until we consume the data. We re-enable it below.
- DisableReadCallback();
- HaveWork();
- }
-
- // Inherited from Worker.
- virtual void OnStart() {
- Lock();
- EnableReadCallback();
- Unlock();
- }
-
- // Inherited from Worker.
- virtual void OnHaveWork() {
- ASSERT(temp_sample_data_ && temp_sample_data_size_);
- SignalSamplesRead(temp_sample_data_,
- temp_sample_data_size_,
- this);
- temp_sample_data_ = NULL;
- temp_sample_data_size_ = 0;
-
- Lock();
- for (;;) {
- // Ack the last thing we read.
- if (symbol_table()->pa_stream_drop()(stream_.stream()) != 0) {
- LOG(LS_ERROR) << "Can't ack read data";
- }
-
- if (symbol_table()->pa_stream_readable_size()(stream_.stream()) <= 0) {
- // Then that was all the data.
- break;
- }
-
- // Else more data.
- const void *sample_data;
- size_t sample_data_size;
- if (symbol_table()->pa_stream_peek()(stream_.stream(),
- &sample_data,
- &sample_data_size) != 0) {
- LOG(LS_ERROR) << "Can't read data!";
- break;
- }
-
- // Drop lock for sigslot dispatch, which could take a while.
- Unlock();
- SignalSamplesRead(sample_data, sample_data_size, this);
- Lock();
-
- // Return to top of loop for the ack and the check for more data.
- }
- EnableReadCallback();
- Unlock();
- }
-
- // Inherited from Worker.
- virtual void OnStop() {
- Lock();
- DisableReadCallback();
- Unlock();
- }
-
- static void OverflowCallback(pa_stream *stream,
- void *userdata) {
- LOG(LS_WARNING) << "Buffer overflow on capture stream " << stream;
- }
-
- static void GetVolumeCallbackThunk(pa_context *unused,
- const pa_source_info *info,
- int eol,
- void *userdata) {
- GetVolumeCallbackData *data =
- static_cast<GetVolumeCallbackData *>(userdata);
- data->instance->OnGetVolumeCallback(info, eol, &data->channel_volumes);
- }
-
- void OnGetVolumeCallback(const pa_source_info *info,
- int eol,
- pa_cvolume **channel_volumes) {
- if (eol) {
- // List is over. Wake GetVolume().
- stream_.pulse()->Signal();
- return;
- }
-
- if (*channel_volumes) {
- **channel_volumes = info->volume;
- // Unset the pointer so that we know that we have have already copied the
- // volume.
- *channel_volumes = NULL;
- } else {
- // We have received an additional callback after the first one, which
- // doesn't make sense for a single source. This probably never happens,
- // but we code for it anyway.
- LOG(LS_WARNING) << "Ignoring extra GetVolumeCallback";
- }
- }
-
- static void GetSourceChannelCountCallbackThunk(pa_context *unused,
- const pa_source_info *info,
- int eol,
- void *userdata) {
- GetSourceChannelCountCallbackData *data =
- static_cast<GetSourceChannelCountCallbackData *>(userdata);
- data->instance->OnGetSourceChannelCountCallback(info, eol, &data->channels);
- }
-
- void OnGetSourceChannelCountCallback(const pa_source_info *info,
- int eol,
- uint8_t **channels) {
- if (eol) {
- // List is over. Wake SetVolume().
- stream_.pulse()->Signal();
- return;
- }
-
- if (*channels) {
- **channels = info->channel_map.channels;
- // Unset the pointer so that we know that we have have already copied the
- // channel count.
- *channels = NULL;
- } else {
- // We have received an additional callback after the first one, which
- // doesn't make sense for a single source. This probably never happens,
- // but we code for it anyway.
- LOG(LS_WARNING) << "Ignoring extra GetSourceChannelCountCallback";
- }
- }
-
- static void SetVolumeCallback(pa_context *unused1,
- int success,
- void *unused2) {
- if (!success) {
- LOG(LS_ERROR) << "Failed to change capture volume";
- }
- }
-
- PulseAudioStream stream_;
- // Temporary storage for passing data between threads.
- const void *temp_sample_data_;
- size_t temp_sample_data_size_;
-
- DISALLOW_COPY_AND_ASSIGN(PulseAudioInputStream);
-};
-
-// Implementation of an output stream. See soundoutputstreaminterface.h
-// regarding thread-safety.
-class PulseAudioOutputStream :
- public SoundOutputStreamInterface,
- private rtc::Worker {
-
- struct GetVolumeCallbackData {
- PulseAudioOutputStream *instance;
- pa_cvolume *channel_volumes;
- };
-
- public:
- PulseAudioOutputStream(PulseAudioSoundSystem *pulse,
- pa_stream *stream,
- int flags,
- int latency)
- : stream_(pulse, stream, flags),
- configured_latency_(latency),
- temp_buffer_space_(0) {
- symbol_table()->pa_stream_set_underflow_callback()(stream,
- &UnderflowCallbackThunk,
- this);
- }
-
- virtual ~PulseAudioOutputStream() {
- bool success = Close();
- // We need that to live.
- VERIFY(success);
- }
-
- virtual bool EnableBufferMonitoring() {
- return StartWork();
- }
-
- virtual bool DisableBufferMonitoring() {
- return StopWork();
- }
-
- virtual bool WriteSamples(const void *sample_data,
- size_t size) {
- bool ret = true;
- Lock();
- if (symbol_table()->pa_stream_write()(stream_.stream(),
- sample_data,
- size,
- NULL,
- 0,
- PA_SEEK_RELATIVE) != 0) {
- LOG(LS_ERROR) << "Unable to write";
- ret = false;
- }
- Unlock();
- return ret;
- }
-
- virtual bool GetVolume(int *volume) {
- bool ret = false;
-
- Lock();
-
- pa_cvolume channel_volumes;
-
- GetVolumeCallbackData data;
- data.instance = this;
- data.channel_volumes = &channel_volumes;
-
- pa_operation *op = symbol_table()->pa_context_get_sink_input_info()(
- stream_.pulse()->context_,
- symbol_table()->pa_stream_get_index()(stream_.stream()),
- &GetVolumeCallbackThunk,
- &data);
- if (!stream_.pulse()->FinishOperation(op)) {
- goto done;
- }
-
- if (data.channel_volumes) {
- // This pointer was never unset by the callback, so we must have received
- // an empty list of infos. This probably never happens, but we code for it
- // anyway.
- LOG(LS_ERROR) << "Did not receive GetVolumeCallback";
- goto done;
- }
-
- // We now have the volume for each channel. Each channel could have a
- // different volume if, e.g., the user went and changed the volumes in the
- // PA UI. To get a single volume for SoundSystemInterface we just take the
- // maximum. Ideally we'd do so with pa_cvolume_max, but it doesn't exist in
- // Hardy, so we do it manually.
- pa_volume_t pa_volume;
- pa_volume = MaxChannelVolume(&channel_volumes);
- // Now map onto the SoundSystemInterface range.
- *volume = PulseVolumeToCricketVolume(pa_volume);
-
- ret = true;
- done:
- Unlock();
- return ret;
- }
-
- virtual bool SetVolume(int volume) {
- bool ret = false;
- pa_volume_t pa_volume = CricketVolumeToPulseVolume(volume);
-
- Lock();
-
- const pa_sample_spec *spec = symbol_table()->pa_stream_get_sample_spec()(
- stream_.stream());
- if (!spec) {
- LOG(LS_ERROR) << "pa_stream_get_sample_spec()";
- goto done;
- }
-
- pa_cvolume channel_volumes;
- symbol_table()->pa_cvolume_set()(&channel_volumes, spec->channels,
- pa_volume);
-
- pa_operation *op;
- op = symbol_table()->pa_context_set_sink_input_volume()(
- stream_.pulse()->context_,
- symbol_table()->pa_stream_get_index()(stream_.stream()),
- &channel_volumes,
- // This callback merely logs errors.
- &SetVolumeCallback,
- NULL);
- if (!op) {
- LOG(LS_ERROR) << "pa_context_set_sink_input_volume()";
- goto done;
- }
- // Don't need to wait for this to complete.
- symbol_table()->pa_operation_unref()(op);
-
- ret = true;
- done:
- Unlock();
- return ret;
- }
-
- virtual bool Close() {
- if (!DisableBufferMonitoring()) {
- return false;
- }
- bool ret = true;
- if (!stream_.IsClosed()) {
- Lock();
- symbol_table()->pa_stream_set_underflow_callback()(stream_.stream(),
- NULL,
- NULL);
- ret = stream_.Close();
- Unlock();
- }
- return ret;
- }
-
- virtual int LatencyUsecs() {
- return stream_.LatencyUsecs();
- }
-
-#if 0
- // TODO: Versions 0.9.16 and later of Pulse have a new API for
- // zero-copy writes, but Hardy is not new enough to have that so we can't
- // rely on it. Perhaps auto-detect if it's present or not and use it if we
- // can?
-
- virtual bool GetWriteBuffer(void **buffer, size_t *size) {
- bool ret = true;
- Lock();
- if (symbol_table()->pa_stream_begin_write()(stream_.stream(), buffer, size)
- != 0) {
- LOG(LS_ERROR) << "Can't get write buffer";
- ret = false;
- }
- Unlock();
- return ret;
- }
-
- // Releases the caller's hold on the write buffer. "written" must be the
- // amount of data that was written.
- virtual bool ReleaseWriteBuffer(void *buffer, size_t written) {
- bool ret = true;
- Lock();
- if (written == 0) {
- if (symbol_table()->pa_stream_cancel_write()(stream_.stream()) != 0) {
- LOG(LS_ERROR) << "Can't cancel write";
- ret = false;
- }
- } else {
- if (symbol_table()->pa_stream_write()(stream_.stream(),
- buffer,
- written,
- NULL,
- 0,
- PA_SEEK_RELATIVE) != 0) {
- LOG(LS_ERROR) << "Unable to write";
- ret = false;
- }
- }
- Unlock();
- return ret;
- }
-#endif
-
- private:
- void Lock() {
- stream_.Lock();
- }
-
- void Unlock() {
- stream_.Unlock();
- }
-
- PulseAudioSymbolTable *symbol_table() {
- return stream_.symbol_table();
- }
-
- void EnableWriteCallback() {
- pa_stream_state_t state = symbol_table()->pa_stream_get_state()(
- stream_.stream());
- if (state == PA_STREAM_READY) {
- // May already have available space. Must check.
- temp_buffer_space_ = symbol_table()->pa_stream_writable_size()(
- stream_.stream());
- if (temp_buffer_space_ > 0) {
- // Yup, there is already space available, so if we register a write
- // callback then it will not receive any event. So dispatch one ourself
- // instead.
- HaveWork();
- return;
- }
- }
- symbol_table()->pa_stream_set_write_callback()(
- stream_.stream(),
- &WriteCallbackThunk,
- this);
- }
-
- void DisableWriteCallback() {
- symbol_table()->pa_stream_set_write_callback()(
- stream_.stream(),
- NULL,
- NULL);
- }
-
- static void WriteCallbackThunk(pa_stream *unused,
- size_t buffer_space,
- void *userdata) {
- PulseAudioOutputStream *instance =
- static_cast<PulseAudioOutputStream *>(userdata);
- instance->OnWriteCallback(buffer_space);
- }
-
- void OnWriteCallback(size_t buffer_space) {
- temp_buffer_space_ = buffer_space;
- // Since we write the data asynchronously on a different thread, we have
- // to temporarily disable the write callback or else Pulse will call it
- // continuously until we write the data. We re-enable it below.
- DisableWriteCallback();
- HaveWork();
- }
-
- // Inherited from Worker.
- virtual void OnStart() {
- Lock();
- EnableWriteCallback();
- Unlock();
- }
-
- // Inherited from Worker.
- virtual void OnHaveWork() {
- ASSERT(temp_buffer_space_ > 0);
-
- SignalBufferSpace(temp_buffer_space_, this);
-
- temp_buffer_space_ = 0;
- Lock();
- EnableWriteCallback();
- Unlock();
- }
-
- // Inherited from Worker.
- virtual void OnStop() {
- Lock();
- DisableWriteCallback();
- Unlock();
- }
-
- static void UnderflowCallbackThunk(pa_stream *unused,
- void *userdata) {
- PulseAudioOutputStream *instance =
- static_cast<PulseAudioOutputStream *>(userdata);
- instance->OnUnderflowCallback();
- }
-
- void OnUnderflowCallback() {
- LOG(LS_WARNING) << "Buffer underflow on playback stream "
- << stream_.stream();
-
- if (configured_latency_ == SoundSystemInterface::kNoLatencyRequirements) {
- // We didn't configure a pa_buffer_attr before, so switching to one now
- // would be questionable.
- return;
- }
-
- // Otherwise reconfigure the stream with a higher target latency.
-
- const pa_sample_spec *spec = symbol_table()->pa_stream_get_sample_spec()(
- stream_.stream());
- if (!spec) {
- LOG(LS_ERROR) << "pa_stream_get_sample_spec()";
- return;
- }
-
- size_t bytes_per_sec = symbol_table()->pa_bytes_per_second()(spec);
-
- int new_latency = configured_latency_ +
- bytes_per_sec * kPlaybackLatencyIncrementMsecs /
- rtc::kNumMicrosecsPerSec;
-
- pa_buffer_attr new_attr = {0};
- FillPlaybackBufferAttr(new_latency, &new_attr);
-
- pa_operation *op = symbol_table()->pa_stream_set_buffer_attr()(
- stream_.stream(),
- &new_attr,
- // No callback.
- NULL,
- NULL);
- if (!op) {
- LOG(LS_ERROR) << "pa_stream_set_buffer_attr()";
- return;
- }
- // Don't need to wait for this to complete.
- symbol_table()->pa_operation_unref()(op);
-
- // Save the new latency in case we underflow again.
- configured_latency_ = new_latency;
- }
-
- static void GetVolumeCallbackThunk(pa_context *unused,
- const pa_sink_input_info *info,
- int eol,
- void *userdata) {
- GetVolumeCallbackData *data =
- static_cast<GetVolumeCallbackData *>(userdata);
- data->instance->OnGetVolumeCallback(info, eol, &data->channel_volumes);
- }
-
- void OnGetVolumeCallback(const pa_sink_input_info *info,
- int eol,
- pa_cvolume **channel_volumes) {
- if (eol) {
- // List is over. Wake GetVolume().
- stream_.pulse()->Signal();
- return;
- }
-
- if (*channel_volumes) {
- **channel_volumes = info->volume;
- // Unset the pointer so that we know that we have have already copied the
- // volume.
- *channel_volumes = NULL;
- } else {
- // We have received an additional callback after the first one, which
- // doesn't make sense for a single sink input. This probably never
- // happens, but we code for it anyway.
- LOG(LS_WARNING) << "Ignoring extra GetVolumeCallback";
- }
- }
-
- static void SetVolumeCallback(pa_context *unused1,
- int success,
- void *unused2) {
- if (!success) {
- LOG(LS_ERROR) << "Failed to change playback volume";
- }
- }
-
- PulseAudioStream stream_;
- int configured_latency_;
- // Temporary storage for passing data between threads.
- size_t temp_buffer_space_;
-
- DISALLOW_COPY_AND_ASSIGN(PulseAudioOutputStream);
-};
-
-PulseAudioSoundSystem::PulseAudioSoundSystem()
- : mainloop_(NULL), context_(NULL) {
-}
-
-PulseAudioSoundSystem::~PulseAudioSoundSystem() {
- Terminate();
-}
-
-bool PulseAudioSoundSystem::Init() {
- if (IsInitialized()) {
- return true;
- }
-
- // Load libpulse.
- if (!symbol_table_.Load()) {
- // Most likely the Pulse library and sound server are not installed on
- // this system.
- LOG(LS_WARNING) << "Failed to load symbol table";
- return false;
- }
-
- // Now create and start the Pulse event thread.
- mainloop_ = symbol_table_.pa_threaded_mainloop_new()();
- if (!mainloop_) {
- LOG(LS_ERROR) << "Can't create mainloop";
- goto fail0;
- }
-
- if (symbol_table_.pa_threaded_mainloop_start()(mainloop_) != 0) {
- LOG(LS_ERROR) << "Can't start mainloop";
- goto fail1;
- }
-
- Lock();
- context_ = CreateNewConnection();
- Unlock();
-
- if (!context_) {
- goto fail2;
- }
-
- // Otherwise we're now ready!
- return true;
-
- fail2:
- symbol_table_.pa_threaded_mainloop_stop()(mainloop_);
- fail1:
- symbol_table_.pa_threaded_mainloop_free()(mainloop_);
- mainloop_ = NULL;
- fail0:
- return false;
-}
-
-void PulseAudioSoundSystem::Terminate() {
- if (!IsInitialized()) {
- return;
- }
-
- Lock();
- symbol_table_.pa_context_disconnect()(context_);
- symbol_table_.pa_context_unref()(context_);
- Unlock();
- context_ = NULL;
- symbol_table_.pa_threaded_mainloop_stop()(mainloop_);
- symbol_table_.pa_threaded_mainloop_free()(mainloop_);
- mainloop_ = NULL;
-
- // We do not unload the symbol table because we may need it again soon if
- // Init() is called again.
-}
-
-bool PulseAudioSoundSystem::EnumeratePlaybackDevices(
- SoundDeviceLocatorList *devices) {
- return EnumerateDevices<pa_sink_info>(
- devices,
- symbol_table_.pa_context_get_sink_info_list(),
- &EnumeratePlaybackDevicesCallbackThunk);
-}
-
-bool PulseAudioSoundSystem::EnumerateCaptureDevices(
- SoundDeviceLocatorList *devices) {
- return EnumerateDevices<pa_source_info>(
- devices,
- symbol_table_.pa_context_get_source_info_list(),
- &EnumerateCaptureDevicesCallbackThunk);
-}
-
-bool PulseAudioSoundSystem::GetDefaultPlaybackDevice(
- SoundDeviceLocator **device) {
- return GetDefaultDevice<&pa_server_info::default_sink_name>(device);
-}
-
-bool PulseAudioSoundSystem::GetDefaultCaptureDevice(
- SoundDeviceLocator **device) {
- return GetDefaultDevice<&pa_server_info::default_source_name>(device);
-}
-
-SoundOutputStreamInterface *PulseAudioSoundSystem::OpenPlaybackDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms) {
- return OpenDevice<SoundOutputStreamInterface>(
- device,
- params,
- "Playback",
- &PulseAudioSoundSystem::ConnectOutputStream);
-}
-
-SoundInputStreamInterface *PulseAudioSoundSystem::OpenCaptureDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms) {
- return OpenDevice<SoundInputStreamInterface>(
- device,
- params,
- "Capture",
- &PulseAudioSoundSystem::ConnectInputStream);
-}
-
-const char *PulseAudioSoundSystem::GetName() const {
- return "PulseAudio";
-}
-
-inline bool PulseAudioSoundSystem::IsInitialized() {
- return mainloop_ != NULL;
-}
-
-struct ConnectToPulseCallbackData {
- PulseAudioSoundSystem *instance;
- bool connect_done;
-};
-
-void PulseAudioSoundSystem::ConnectToPulseCallbackThunk(
- pa_context *context, void *userdata) {
- ConnectToPulseCallbackData *data =
- static_cast<ConnectToPulseCallbackData *>(userdata);
- data->instance->OnConnectToPulseCallback(context, &data->connect_done);
-}
-
-void PulseAudioSoundSystem::OnConnectToPulseCallback(
- pa_context *context, bool *connect_done) {
- pa_context_state_t state = symbol_table_.pa_context_get_state()(context);
- if (state == PA_CONTEXT_READY ||
- state == PA_CONTEXT_FAILED ||
- state == PA_CONTEXT_TERMINATED) {
- // Connection process has reached a terminal state. Wake ConnectToPulse().
- *connect_done = true;
- Signal();
- }
-}
-
-// Must be called with the lock held.
-bool PulseAudioSoundSystem::ConnectToPulse(pa_context *context) {
- bool ret = true;
- ConnectToPulseCallbackData data;
- // Have to put this up here to satisfy the compiler.
- pa_context_state_t state;
-
- data.instance = this;
- data.connect_done = false;
-
- symbol_table_.pa_context_set_state_callback()(context,
- &ConnectToPulseCallbackThunk,
- &data);
-
- // Connect to PulseAudio sound server.
- if (symbol_table_.pa_context_connect()(
- context,
- NULL, // Default server
- PA_CONTEXT_NOAUTOSPAWN,
- NULL) != 0) { // No special fork handling needed
- LOG(LS_ERROR) << "Can't start connection to PulseAudio sound server";
- ret = false;
- goto done;
- }
-
- // Wait for the connection state machine to reach a terminal state.
- do {
- Wait();
- } while (!data.connect_done);
-
- // Now check to see what final state we reached.
- state = symbol_table_.pa_context_get_state()(context);
-
- if (state != PA_CONTEXT_READY) {
- if (state == PA_CONTEXT_FAILED) {
- LOG(LS_ERROR) << "Failed to connect to PulseAudio sound server";
- } else if (state == PA_CONTEXT_TERMINATED) {
- LOG(LS_ERROR) << "PulseAudio connection terminated early";
- } else {
- // Shouldn't happen, because we only signal on one of those three states.
- LOG(LS_ERROR) << "Unknown problem connecting to PulseAudio";
- }
- ret = false;
- }
-
- done:
- // We unset our callback for safety just in case the state might somehow
- // change later, because the pointer to "data" will be invalid after return
- // from this function.
- symbol_table_.pa_context_set_state_callback()(context, NULL, NULL);
- return ret;
-}
-
-// Must be called with the lock held.
-pa_context *PulseAudioSoundSystem::CreateNewConnection() {
- // Create connection context.
- std::string app_name;
- // TODO: Pulse etiquette says this name should be localized. Do
- // we care?
- rtc::Filesystem::GetApplicationName(&app_name);
- pa_context *context = symbol_table_.pa_context_new()(
- symbol_table_.pa_threaded_mainloop_get_api()(mainloop_),
- app_name.c_str());
- if (!context) {
- LOG(LS_ERROR) << "Can't create context";
- goto fail0;
- }
-
- // Now connect.
- if (!ConnectToPulse(context)) {
- goto fail1;
- }
-
- // Otherwise the connection succeeded and is ready.
- return context;
-
- fail1:
- symbol_table_.pa_context_unref()(context);
- fail0:
- return NULL;
-}
-
-struct EnumerateDevicesCallbackData {
- PulseAudioSoundSystem *instance;
- SoundSystemInterface::SoundDeviceLocatorList *devices;
-};
-
-void PulseAudioSoundSystem::EnumeratePlaybackDevicesCallbackThunk(
- pa_context *unused,
- const pa_sink_info *info,
- int eol,
- void *userdata) {
- EnumerateDevicesCallbackData *data =
- static_cast<EnumerateDevicesCallbackData *>(userdata);
- data->instance->OnEnumeratePlaybackDevicesCallback(data->devices, info, eol);
-}
-
-void PulseAudioSoundSystem::EnumerateCaptureDevicesCallbackThunk(
- pa_context *unused,
- const pa_source_info *info,
- int eol,
- void *userdata) {
- EnumerateDevicesCallbackData *data =
- static_cast<EnumerateDevicesCallbackData *>(userdata);
- data->instance->OnEnumerateCaptureDevicesCallback(data->devices, info, eol);
-}
-
-void PulseAudioSoundSystem::OnEnumeratePlaybackDevicesCallback(
- SoundDeviceLocatorList *devices,
- const pa_sink_info *info,
- int eol) {
- if (eol) {
- // List is over. Wake EnumerateDevices().
- Signal();
- return;
- }
-
- // Else this is the next device.
- devices->push_back(
- new PulseAudioDeviceLocator(info->description, info->name));
-}
-
-void PulseAudioSoundSystem::OnEnumerateCaptureDevicesCallback(
- SoundDeviceLocatorList *devices,
- const pa_source_info *info,
- int eol) {
- if (eol) {
- // List is over. Wake EnumerateDevices().
- Signal();
- return;
- }
-
- if (info->monitor_of_sink != PA_INVALID_INDEX) {
- // We don't want to list monitor sources, since they are almost certainly
- // not what the user wants for voice conferencing.
- return;
- }
-
- // Else this is the next device.
- devices->push_back(
- new PulseAudioDeviceLocator(info->description, info->name));
-}
-
-template <typename InfoStruct>
-bool PulseAudioSoundSystem::EnumerateDevices(
- SoundDeviceLocatorList *devices,
- pa_operation *(*enumerate_fn)(
- pa_context *c,
- void (*callback_fn)(
- pa_context *c,
- const InfoStruct *i,
- int eol,
- void *userdata),
- void *userdata),
- void (*callback_fn)(
- pa_context *c,
- const InfoStruct *i,
- int eol,
- void *userdata)) {
- ClearSoundDeviceLocatorList(devices);
- if (!IsInitialized()) {
- return false;
- }
-
- EnumerateDevicesCallbackData data;
- data.instance = this;
- data.devices = devices;
-
- Lock();
- pa_operation *op = (*enumerate_fn)(
- context_,
- callback_fn,
- &data);
- bool ret = FinishOperation(op);
- Unlock();
- return ret;
-}
-
-struct GetDefaultDeviceCallbackData {
- PulseAudioSoundSystem *instance;
- SoundDeviceLocator **device;
-};
-
-template <const char *(pa_server_info::*field)>
-void PulseAudioSoundSystem::GetDefaultDeviceCallbackThunk(
- pa_context *unused,
- const pa_server_info *info,
- void *userdata) {
- GetDefaultDeviceCallbackData *data =
- static_cast<GetDefaultDeviceCallbackData *>(userdata);
- data->instance->OnGetDefaultDeviceCallback<field>(info, data->device);
-}
-
-template <const char *(pa_server_info::*field)>
-void PulseAudioSoundSystem::OnGetDefaultDeviceCallback(
- const pa_server_info *info,
- SoundDeviceLocator **device) {
- if (info) {
- const char *dev = info->*field;
- if (dev) {
- *device = new PulseAudioDeviceLocator("Default device", dev);
- }
- }
- Signal();
-}
-
-template <const char *(pa_server_info::*field)>
-bool PulseAudioSoundSystem::GetDefaultDevice(SoundDeviceLocator **device) {
- if (!IsInitialized()) {
- return false;
- }
- bool ret;
- *device = NULL;
- GetDefaultDeviceCallbackData data;
- data.instance = this;
- data.device = device;
- Lock();
- pa_operation *op = symbol_table_.pa_context_get_server_info()(
- context_,
- &GetDefaultDeviceCallbackThunk<field>,
- &data);
- ret = FinishOperation(op);
- Unlock();
- return ret && (*device != NULL);
-}
-
-void PulseAudioSoundSystem::StreamStateChangedCallbackThunk(
- pa_stream *stream,
- void *userdata) {
- PulseAudioSoundSystem *instance =
- static_cast<PulseAudioSoundSystem *>(userdata);
- instance->OnStreamStateChangedCallback(stream);
-}
-
-void PulseAudioSoundSystem::OnStreamStateChangedCallback(pa_stream *stream) {
- pa_stream_state_t state = symbol_table_.pa_stream_get_state()(stream);
- if (state == PA_STREAM_READY) {
- LOG(LS_INFO) << "Pulse stream " << stream << " ready";
- } else if (state == PA_STREAM_FAILED ||
- state == PA_STREAM_TERMINATED ||
- state == PA_STREAM_UNCONNECTED) {
- LOG(LS_ERROR) << "Pulse stream " << stream << " failed to connect: "
- << LastError();
- }
-}
-
-template <typename StreamInterface>
-StreamInterface *PulseAudioSoundSystem::OpenDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms,
- const char *stream_name,
- StreamInterface *(PulseAudioSoundSystem::*connect_fn)(
- pa_stream *stream,
- const char *dev,
- int flags,
- pa_stream_flags_t pa_flags,
- int latency,
- const pa_sample_spec &spec)) {
- if (!IsInitialized()) {
- return NULL;
- }
-
- const char *dev = static_cast<const PulseAudioDeviceLocator *>(device)->
- device_name().c_str();
-
- StreamInterface *stream_interface = NULL;
-
- ASSERT(params.format < ARRAY_SIZE(kCricketFormatToPulseFormatTable));
-
- pa_sample_spec spec;
- spec.format = kCricketFormatToPulseFormatTable[params.format];
- spec.rate = params.freq;
- spec.channels = params.channels;
-
- int pa_flags = 0;
- if (params.flags & FLAG_REPORT_LATENCY) {
- pa_flags |= PA_STREAM_INTERPOLATE_TIMING |
- PA_STREAM_AUTO_TIMING_UPDATE;
- }
-
- if (params.latency != kNoLatencyRequirements) {
- // If configuring a specific latency then we want to specify
- // PA_STREAM_ADJUST_LATENCY to make the server adjust parameters
- // automatically to reach that target latency. However, that flag doesn't
- // exist in Ubuntu 8.04 and many people still use that, so we have to check
- // the protocol version of libpulse.
- if (symbol_table_.pa_context_get_protocol_version()(context_) >=
- kAdjustLatencyProtocolVersion) {
- pa_flags |= PA_STREAM_ADJUST_LATENCY;
- }
- }
-
- Lock();
-
- pa_stream *stream = symbol_table_.pa_stream_new()(context_, stream_name,
- &spec, NULL);
- if (!stream) {
- LOG(LS_ERROR) << "Can't create pa_stream";
- goto done;
- }
-
- // Set a state callback to log errors.
- symbol_table_.pa_stream_set_state_callback()(stream,
- &StreamStateChangedCallbackThunk,
- this);
-
- stream_interface = (this->*connect_fn)(
- stream,
- dev,
- params.flags,
- static_cast<pa_stream_flags_t>(pa_flags),
- params.latency,
- spec);
- if (!stream_interface) {
- LOG(LS_ERROR) << "Can't connect stream to " << dev;
- symbol_table_.pa_stream_unref()(stream);
- }
-
- done:
- Unlock();
- return stream_interface;
-}
-
-// Must be called with the lock held.
-SoundOutputStreamInterface *PulseAudioSoundSystem::ConnectOutputStream(
- pa_stream *stream,
- const char *dev,
- int flags,
- pa_stream_flags_t pa_flags,
- int latency,
- const pa_sample_spec &spec) {
- pa_buffer_attr attr = {0};
- pa_buffer_attr *pattr = NULL;
- if (latency != kNoLatencyRequirements) {
- // kLowLatency is 0, so we treat it the same as a request for zero latency.
- ssize_t bytes_per_sec = symbol_table_.pa_bytes_per_second()(&spec);
- latency = rtc::_max(
- latency,
- static_cast<int>(
- bytes_per_sec * kPlaybackLatencyMinimumMsecs /
- rtc::kNumMicrosecsPerSec));
- FillPlaybackBufferAttr(latency, &attr);
- pattr = &attr;
- }
- if (symbol_table_.pa_stream_connect_playback()(
- stream,
- dev,
- pattr,
- pa_flags,
- // Let server choose volume
- NULL,
- // Not synchronized to any other playout
- NULL) != 0) {
- return NULL;
- }
- return new PulseAudioOutputStream(this, stream, flags, latency);
-}
-
-// Must be called with the lock held.
-SoundInputStreamInterface *PulseAudioSoundSystem::ConnectInputStream(
- pa_stream *stream,
- const char *dev,
- int flags,
- pa_stream_flags_t pa_flags,
- int latency,
- const pa_sample_spec &spec) {
- pa_buffer_attr attr = {0};
- pa_buffer_attr *pattr = NULL;
- if (latency != kNoLatencyRequirements) {
- size_t bytes_per_sec = symbol_table_.pa_bytes_per_second()(&spec);
- if (latency == kLowLatency) {
- latency = bytes_per_sec * kLowCaptureLatencyMsecs /
- rtc::kNumMicrosecsPerSec;
- }
- // Note: fragsize specifies a maximum transfer size, not a minimum, so it is
- // not possible to force a high latency setting, only a low one.
- attr.fragsize = latency;
- attr.maxlength = latency + bytes_per_sec * kCaptureBufferExtraMsecs /
- rtc::kNumMicrosecsPerSec;
- LOG(LS_VERBOSE) << "Configuring latency = " << attr.fragsize
- << ", maxlength = " << attr.maxlength;
- pattr = &attr;
- }
- if (symbol_table_.pa_stream_connect_record()(stream,
- dev,
- pattr,
- pa_flags) != 0) {
- return NULL;
- }
- return new PulseAudioInputStream(this, stream, flags);
-}
-
-// Must be called with the lock held.
-bool PulseAudioSoundSystem::FinishOperation(pa_operation *op) {
- if (!op) {
- LOG(LS_ERROR) << "Failed to start operation";
- return false;
- }
-
- do {
- Wait();
- } while (symbol_table_.pa_operation_get_state()(op) == PA_OPERATION_RUNNING);
-
- symbol_table_.pa_operation_unref()(op);
-
- return true;
-}
-
-inline void PulseAudioSoundSystem::Lock() {
- symbol_table_.pa_threaded_mainloop_lock()(mainloop_);
-}
-
-inline void PulseAudioSoundSystem::Unlock() {
- symbol_table_.pa_threaded_mainloop_unlock()(mainloop_);
-}
-
-// Must be called with the lock held.
-inline void PulseAudioSoundSystem::Wait() {
- symbol_table_.pa_threaded_mainloop_wait()(mainloop_);
-}
-
-// Must be called with the lock held.
-inline void PulseAudioSoundSystem::Signal() {
- symbol_table_.pa_threaded_mainloop_signal()(mainloop_, 0);
-}
-
-// Must be called with the lock held.
-const char *PulseAudioSoundSystem::LastError() {
- return symbol_table_.pa_strerror()(symbol_table_.pa_context_errno()(
- context_));
-}
-
-} // namespace cricket
-
-#endif // HAVE_LIBPULSE
diff --git a/sound/pulseaudiosoundsystem.h b/sound/pulseaudiosoundsystem.h
deleted file mode 100644
index 16938d6..0000000
--- a/sound/pulseaudiosoundsystem.h
+++ /dev/null
@@ -1,194 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_PULSEAUDIOSOUNDSYSTEM_H_
-#define TALK_SOUND_PULSEAUDIOSOUNDSYSTEM_H_
-
-#ifdef HAVE_LIBPULSE
-
-#include "talk/sound/pulseaudiosymboltable.h"
-#include "talk/sound/soundsysteminterface.h"
-#include "webrtc/base/constructormagic.h"
-
-namespace cricket {
-
-class PulseAudioInputStream;
-class PulseAudioOutputStream;
-class PulseAudioStream;
-
-// Sound system implementation for PulseAudio, a cross-platform sound server
-// (but commonly used only on Linux, which is the only platform we support
-// it on).
-// Init(), Terminate(), and the destructor should never be invoked concurrently,
-// but all other methods are thread-safe.
-class PulseAudioSoundSystem : public SoundSystemInterface {
- friend class PulseAudioInputStream;
- friend class PulseAudioOutputStream;
- friend class PulseAudioStream;
- public:
- static SoundSystemInterface *Create() {
- return new PulseAudioSoundSystem();
- }
-
- PulseAudioSoundSystem();
-
- virtual ~PulseAudioSoundSystem();
-
- virtual bool Init();
- virtual void Terminate();
-
- virtual bool EnumeratePlaybackDevices(SoundDeviceLocatorList *devices);
- virtual bool EnumerateCaptureDevices(SoundDeviceLocatorList *devices);
-
- virtual bool GetDefaultPlaybackDevice(SoundDeviceLocator **device);
- virtual bool GetDefaultCaptureDevice(SoundDeviceLocator **device);
-
- virtual SoundOutputStreamInterface *OpenPlaybackDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms);
- virtual SoundInputStreamInterface *OpenCaptureDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms);
-
- virtual const char *GetName() const;
-
- private:
- bool IsInitialized();
-
- static void ConnectToPulseCallbackThunk(pa_context *context, void *userdata);
-
- void OnConnectToPulseCallback(pa_context *context, bool *connect_done);
-
- bool ConnectToPulse(pa_context *context);
-
- pa_context *CreateNewConnection();
-
- template <typename InfoStruct>
- bool EnumerateDevices(SoundDeviceLocatorList *devices,
- pa_operation *(*enumerate_fn)(
- pa_context *c,
- void (*callback_fn)(
- pa_context *c,
- const InfoStruct *i,
- int eol,
- void *userdata),
- void *userdata),
- void (*callback_fn)(
- pa_context *c,
- const InfoStruct *i,
- int eol,
- void *userdata));
-
- static void EnumeratePlaybackDevicesCallbackThunk(pa_context *unused,
- const pa_sink_info *info,
- int eol,
- void *userdata);
-
- static void EnumerateCaptureDevicesCallbackThunk(pa_context *unused,
- const pa_source_info *info,
- int eol,
- void *userdata);
-
- void OnEnumeratePlaybackDevicesCallback(
- SoundDeviceLocatorList *devices,
- const pa_sink_info *info,
- int eol);
-
- void OnEnumerateCaptureDevicesCallback(
- SoundDeviceLocatorList *devices,
- const pa_source_info *info,
- int eol);
-
- template <const char *(pa_server_info::*field)>
- static void GetDefaultDeviceCallbackThunk(
- pa_context *unused,
- const pa_server_info *info,
- void *userdata);
-
- template <const char *(pa_server_info::*field)>
- void OnGetDefaultDeviceCallback(
- const pa_server_info *info,
- SoundDeviceLocator **device);
-
- template <const char *(pa_server_info::*field)>
- bool GetDefaultDevice(SoundDeviceLocator **device);
-
- static void StreamStateChangedCallbackThunk(pa_stream *stream,
- void *userdata);
-
- void OnStreamStateChangedCallback(pa_stream *stream);
-
- template <typename StreamInterface>
- StreamInterface *OpenDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms,
- const char *stream_name,
- StreamInterface *(PulseAudioSoundSystem::*connect_fn)(
- pa_stream *stream,
- const char *dev,
- int flags,
- pa_stream_flags_t pa_flags,
- int latency,
- const pa_sample_spec &spec));
-
- SoundOutputStreamInterface *ConnectOutputStream(
- pa_stream *stream,
- const char *dev,
- int flags,
- pa_stream_flags_t pa_flags,
- int latency,
- const pa_sample_spec &spec);
-
- SoundInputStreamInterface *ConnectInputStream(
- pa_stream *stream,
- const char *dev,
- int flags,
- pa_stream_flags_t pa_flags,
- int latency,
- const pa_sample_spec &spec);
-
- bool FinishOperation(pa_operation *op);
-
- void Lock();
- void Unlock();
- void Wait();
- void Signal();
-
- const char *LastError();
-
- pa_threaded_mainloop *mainloop_;
- pa_context *context_;
- PulseAudioSymbolTable symbol_table_;
-
- DISALLOW_COPY_AND_ASSIGN(PulseAudioSoundSystem);
-};
-
-} // namespace cricket
-
-#endif // HAVE_LIBPULSE
-
-#endif // TALK_SOUND_PULSEAUDIOSOUNDSYSTEM_H_
diff --git a/sound/pulseaudiosymboltable.cc b/sound/pulseaudiosymboltable.cc
deleted file mode 100644
index 344f354..0000000
--- a/sound/pulseaudiosymboltable.cc
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifdef HAVE_LIBPULSE
-
-#include "talk/sound/pulseaudiosymboltable.h"
-
-namespace cricket {
-
-#define LATE_BINDING_SYMBOL_TABLE_CLASS_NAME PULSE_AUDIO_SYMBOLS_CLASS_NAME
-#define LATE_BINDING_SYMBOL_TABLE_SYMBOLS_LIST PULSE_AUDIO_SYMBOLS_LIST
-#define LATE_BINDING_SYMBOL_TABLE_DLL_NAME "libpulse.so.0"
-#include "webrtc/base/latebindingsymboltable.cc.def"
-
-} // namespace cricket
-
-#endif // HAVE_LIBPULSE
diff --git a/sound/pulseaudiosymboltable.h b/sound/pulseaudiosymboltable.h
deleted file mode 100644
index 46bddea..0000000
--- a/sound/pulseaudiosymboltable.h
+++ /dev/null
@@ -1,104 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_PULSEAUDIOSYMBOLTABLE_H_
-#define TALK_SOUND_PULSEAUDIOSYMBOLTABLE_H_
-
-#include <pulse/context.h>
-#include <pulse/def.h>
-#include <pulse/error.h>
-#include <pulse/introspect.h>
-#include <pulse/stream.h>
-#include <pulse/thread-mainloop.h>
-
-#include "webrtc/base/latebindingsymboltable.h"
-
-namespace cricket {
-
-#define PULSE_AUDIO_SYMBOLS_CLASS_NAME PulseAudioSymbolTable
-// The PulseAudio symbols we need, as an X-Macro list.
-// This list must contain precisely every libpulse function that is used in
-// pulseaudiosoundsystem.cc.
-#define PULSE_AUDIO_SYMBOLS_LIST \
- X(pa_bytes_per_second) \
- X(pa_context_connect) \
- X(pa_context_disconnect) \
- X(pa_context_errno) \
- X(pa_context_get_protocol_version) \
- X(pa_context_get_server_info) \
- X(pa_context_get_sink_info_list) \
- X(pa_context_get_sink_input_info) \
- X(pa_context_get_source_info_by_index) \
- X(pa_context_get_source_info_list) \
- X(pa_context_get_state) \
- X(pa_context_new) \
- X(pa_context_set_sink_input_volume) \
- X(pa_context_set_source_volume_by_index) \
- X(pa_context_set_state_callback) \
- X(pa_context_unref) \
- X(pa_cvolume_set) \
- X(pa_operation_get_state) \
- X(pa_operation_unref) \
- X(pa_stream_connect_playback) \
- X(pa_stream_connect_record) \
- X(pa_stream_disconnect) \
- X(pa_stream_drop) \
- X(pa_stream_get_device_index) \
- X(pa_stream_get_index) \
- X(pa_stream_get_latency) \
- X(pa_stream_get_sample_spec) \
- X(pa_stream_get_state) \
- X(pa_stream_new) \
- X(pa_stream_peek) \
- X(pa_stream_readable_size) \
- X(pa_stream_set_buffer_attr) \
- X(pa_stream_set_overflow_callback) \
- X(pa_stream_set_read_callback) \
- X(pa_stream_set_state_callback) \
- X(pa_stream_set_underflow_callback) \
- X(pa_stream_set_write_callback) \
- X(pa_stream_unref) \
- X(pa_stream_writable_size) \
- X(pa_stream_write) \
- X(pa_strerror) \
- X(pa_threaded_mainloop_free) \
- X(pa_threaded_mainloop_get_api) \
- X(pa_threaded_mainloop_lock) \
- X(pa_threaded_mainloop_new) \
- X(pa_threaded_mainloop_signal) \
- X(pa_threaded_mainloop_start) \
- X(pa_threaded_mainloop_stop) \
- X(pa_threaded_mainloop_unlock) \
- X(pa_threaded_mainloop_wait)
-
-#define LATE_BINDING_SYMBOL_TABLE_CLASS_NAME PULSE_AUDIO_SYMBOLS_CLASS_NAME
-#define LATE_BINDING_SYMBOL_TABLE_SYMBOLS_LIST PULSE_AUDIO_SYMBOLS_LIST
-#include "webrtc/base/latebindingsymboltable.h.def"
-
-} // namespace cricket
-
-#endif // TALK_SOUND_PULSEAUDIOSYMBOLTABLE_H_
diff --git a/sound/sounddevicelocator.h b/sound/sounddevicelocator.h
deleted file mode 100644
index 420226f..0000000
--- a/sound/sounddevicelocator.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_SOUNDDEVICELOCATOR_H_
-#define TALK_SOUND_SOUNDDEVICELOCATOR_H_
-
-#include <string>
-
-#include "webrtc/base/constructormagic.h"
-
-namespace cricket {
-
-// A simple container for holding the name of a device and any additional id
-// information needed to locate and open it. Implementations of
-// SoundSystemInterface must subclass this to add any id information that they
-// need.
-class SoundDeviceLocator {
- public:
- virtual ~SoundDeviceLocator() {}
-
- // Human-readable name for the device.
- const std::string &name() const { return name_; }
-
- // Name sound system uses to locate this device.
- const std::string &device_name() const { return device_name_; }
-
- // Makes a duplicate of this locator.
- virtual SoundDeviceLocator *Copy() const = 0;
-
- protected:
- SoundDeviceLocator(const std::string &name,
- const std::string &device_name)
- : name_(name), device_name_(device_name) {}
-
- explicit SoundDeviceLocator(const SoundDeviceLocator &that)
- : name_(that.name_), device_name_(that.device_name_) {}
-
- std::string name_;
- std::string device_name_;
-
- private:
- DISALLOW_ASSIGN(SoundDeviceLocator);
-};
-
-} // namespace cricket
-
-#endif // TALK_SOUND_SOUNDDEVICELOCATOR_H_
diff --git a/sound/soundinputstreaminterface.h b/sound/soundinputstreaminterface.h
deleted file mode 100644
index e557392..0000000
--- a/sound/soundinputstreaminterface.h
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_SOUNDINPUTSTREAMINTERFACE_H_
-#define TALK_SOUND_SOUNDINPUTSTREAMINTERFACE_H_
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/sigslot.h"
-
-namespace cricket {
-
-// Interface for consuming an input stream from a recording device.
-// Semantics and thread-safety of StartReading()/StopReading() are the same as
-// for rtc::Worker.
-class SoundInputStreamInterface {
- public:
- virtual ~SoundInputStreamInterface() {}
-
- // Starts the reading of samples on the current thread.
- virtual bool StartReading() = 0;
- // Stops the reading of samples.
- virtual bool StopReading() = 0;
-
- // Retrieves the current input volume for this stream. Nominal range is
- // defined by SoundSystemInterface::k(Max|Min)Volume, but values exceeding the
- // max may be possible in some implementations. This call retrieves the actual
- // volume currently in use by the OS, not a cached value from a previous
- // (Get|Set)Volume() call.
- virtual bool GetVolume(int *volume) = 0;
-
- // Changes the input volume for this stream. Nominal range is defined by
- // SoundSystemInterface::k(Max|Min)Volume. The effect of exceeding kMaxVolume
- // is implementation-defined.
- virtual bool SetVolume(int volume) = 0;
-
- // Closes this stream object. If currently reading then this may only be
- // called from the reading thread.
- virtual bool Close() = 0;
-
- // Get the latency of the stream.
- virtual int LatencyUsecs() = 0;
-
- // Notifies the consumer of new data read from the device.
- // The first parameter is a pointer to the data read, and is only valid for
- // the duration of the call.
- // The second parameter is the amount of data read in bytes (i.e., the valid
- // length of the memory pointed to).
- // The 3rd parameter is the stream that is issuing the callback.
- sigslot::signal3<const void *, size_t,
- SoundInputStreamInterface *> SignalSamplesRead;
-
- protected:
- SoundInputStreamInterface() {}
-
- private:
- DISALLOW_COPY_AND_ASSIGN(SoundInputStreamInterface);
-};
-
-} // namespace cricket
-
-#endif // TALK_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_
diff --git a/sound/soundoutputstreaminterface.h b/sound/soundoutputstreaminterface.h
deleted file mode 100644
index 294906d..0000000
--- a/sound/soundoutputstreaminterface.h
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_
-#define TALK_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/sigslot.h"
-
-namespace cricket {
-
-// Interface for outputting a stream to a playback device.
-// Semantics and thread-safety of EnableBufferMonitoring()/
-// DisableBufferMonitoring() are the same as for rtc::Worker.
-class SoundOutputStreamInterface {
- public:
- virtual ~SoundOutputStreamInterface() {}
-
- // Enables monitoring the available buffer space on the current thread.
- virtual bool EnableBufferMonitoring() = 0;
- // Disables the monitoring.
- virtual bool DisableBufferMonitoring() = 0;
-
- // Write the given samples to the devices. If currently monitoring then this
- // may only be called from the monitoring thread.
- virtual bool WriteSamples(const void *sample_data,
- size_t size) = 0;
-
- // Retrieves the current output volume for this stream. Nominal range is
- // defined by SoundSystemInterface::k(Max|Min)Volume, but values exceeding the
- // max may be possible in some implementations. This call retrieves the actual
- // volume currently in use by the OS, not a cached value from a previous
- // (Get|Set)Volume() call.
- virtual bool GetVolume(int *volume) = 0;
-
- // Changes the output volume for this stream. Nominal range is defined by
- // SoundSystemInterface::k(Max|Min)Volume. The effect of exceeding kMaxVolume
- // is implementation-defined.
- virtual bool SetVolume(int volume) = 0;
-
- // Closes this stream object. If currently monitoring then this may only be
- // called from the monitoring thread.
- virtual bool Close() = 0;
-
- // Get the latency of the stream.
- virtual int LatencyUsecs() = 0;
-
- // Notifies the producer of the available buffer space for writes.
- // It fires continuously as long as the space is greater than zero.
- // The first parameter is the amount of buffer space available for data to
- // be written (i.e., the maximum amount of data that can be written right now
- // with WriteSamples() without blocking).
- // The 2nd parameter is the stream that is issuing the callback.
- sigslot::signal2<size_t, SoundOutputStreamInterface *> SignalBufferSpace;
-
- protected:
- SoundOutputStreamInterface() {}
-
- private:
- DISALLOW_COPY_AND_ASSIGN(SoundOutputStreamInterface);
-};
-
-} // namespace cricket
-
-#endif // TALK_SOUND_SOUNDOUTPUTSTREAMINTERFACE_H_
diff --git a/sound/soundsystemfactory.h b/sound/soundsystemfactory.h
deleted file mode 100644
index 06a1c3f..0000000
--- a/sound/soundsystemfactory.h
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_SOUNDSYSTEMFACTORY_H_
-#define TALK_SOUND_SOUNDSYSTEMFACTORY_H_
-
-#include "webrtc/base/referencecountedsingletonfactory.h"
-
-namespace cricket {
-
-class SoundSystemInterface;
-
-typedef rtc::ReferenceCountedSingletonFactory<SoundSystemInterface>
- SoundSystemFactory;
-
-typedef rtc::rcsf_ptr<SoundSystemInterface> SoundSystemHandle;
-
-} // namespace cricket
-
-#endif // TALK_SOUND_SOUNDSYSTEMFACTORY_H_
diff --git a/sound/soundsysteminterface.cc b/sound/soundsysteminterface.cc
deleted file mode 100644
index b432262..0000000
--- a/sound/soundsysteminterface.cc
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/sound/soundsysteminterface.h"
-
-#include "talk/sound/sounddevicelocator.h"
-
-namespace cricket {
-
-void SoundSystemInterface::ClearSoundDeviceLocatorList(
- SoundSystemInterface::SoundDeviceLocatorList *devices) {
- for (SoundDeviceLocatorList::iterator i = devices->begin();
- i != devices->end();
- ++i) {
- if (*i) {
- delete *i;
- }
- }
- devices->clear();
-}
-
-} // namespace cricket
diff --git a/sound/soundsysteminterface.h b/sound/soundsysteminterface.h
deleted file mode 100644
index 5d3e84b..0000000
--- a/sound/soundsysteminterface.h
+++ /dev/null
@@ -1,129 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_SOUNDSYSTEMINTERFACE_H_
-#define TALK_SOUND_SOUNDSYSTEMINTERFACE_H_
-
-#include <vector>
-
-#include "webrtc/base/constructormagic.h"
-
-namespace cricket {
-
-class SoundDeviceLocator;
-class SoundInputStreamInterface;
-class SoundOutputStreamInterface;
-
-// Interface for a platform's sound system.
-// Implementations must guarantee thread-safety for at least the following use
-// cases:
-// 1) Concurrent enumeration and opening of devices from different threads.
-// 2) Concurrent use of different Sound(Input|Output)StreamInterface
-// instances from different threads (but concurrent use of the _same_ one from
-// different threads need not be supported).
-class SoundSystemInterface {
- public:
- typedef std::vector<SoundDeviceLocator *> SoundDeviceLocatorList;
-
- enum SampleFormat {
- // Only one supported sample format at this time.
- // The values here may be used in lookup tables, so they shouldn't change.
- FORMAT_S16LE = 0,
- };
-
- enum Flags {
- // Enable reporting the current stream latency in
- // Sound(Input|Output)StreamInterface. See those classes for more details.
- FLAG_REPORT_LATENCY = (1 << 0),
- };
-
- struct OpenParams {
- // Format for the sound stream.
- SampleFormat format;
- // Sampling frequency in hertz.
- unsigned int freq;
- // Number of channels in the PCM stream.
- unsigned int channels;
- // Misc flags. Should be taken from the Flags enum above.
- int flags;
- // Desired latency, measured as number of bytes of sample data
- int latency;
- };
-
- // Special values for the "latency" field of OpenParams.
- // Use this one to say you don't care what the latency is. The sound system
- // will optimize for other things instead.
- static const int kNoLatencyRequirements = -1;
- // Use this one to say that you want the sound system to pick an appropriate
- // small latency value. The sound system may pick the minimum allowed one, or
- // a slightly higher one in the event that the true minimum requires an
- // undesirable trade-off.
- static const int kLowLatency = 0;
-
- // Max value for the volume parameters for Sound(Input|Output)StreamInterface.
- static const int kMaxVolume = 255;
- // Min value for the volume parameters for Sound(Input|Output)StreamInterface.
- static const int kMinVolume = 0;
-
- // Helper for clearing a locator list and deleting the entries.
- static void ClearSoundDeviceLocatorList(SoundDeviceLocatorList *devices);
-
- virtual ~SoundSystemInterface() {}
-
- virtual bool Init() = 0;
- virtual void Terminate() = 0;
-
- // Enumerates the available devices. (Any pre-existing locators in the lists
- // are deleted.)
- virtual bool EnumeratePlaybackDevices(SoundDeviceLocatorList *devices) = 0;
- virtual bool EnumerateCaptureDevices(SoundDeviceLocatorList *devices) = 0;
-
- // Gets a special locator for the default device.
- virtual bool GetDefaultPlaybackDevice(SoundDeviceLocator **device) = 0;
- virtual bool GetDefaultCaptureDevice(SoundDeviceLocator **device) = 0;
-
- // Opens the given device, or returns NULL on error.
- virtual SoundOutputStreamInterface *OpenPlaybackDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms) = 0;
- virtual SoundInputStreamInterface *OpenCaptureDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms) = 0;
-
- // A human-readable name for this sound system.
- virtual const char *GetName() const = 0;
-
- protected:
- SoundSystemInterface() {}
-
- private:
- DISALLOW_COPY_AND_ASSIGN(SoundSystemInterface);
-};
-
-} // namespace cricket
-
-#endif // TALK_SOUND_SOUNDSYSTEMINTERFACE_H_
diff --git a/sound/soundsystemproxy.cc b/sound/soundsystemproxy.cc
deleted file mode 100644
index 737a6bb..0000000
--- a/sound/soundsystemproxy.cc
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/sound/soundsystemproxy.h"
-
-namespace cricket {
-
-bool SoundSystemProxy::EnumeratePlaybackDevices(
- SoundDeviceLocatorList *devices) {
- return wrapped_ ? wrapped_->EnumeratePlaybackDevices(devices) : false;
-}
-
-bool SoundSystemProxy::EnumerateCaptureDevices(
- SoundDeviceLocatorList *devices) {
- return wrapped_ ? wrapped_->EnumerateCaptureDevices(devices) : false;
-}
-
-bool SoundSystemProxy::GetDefaultPlaybackDevice(
- SoundDeviceLocator **device) {
- return wrapped_ ? wrapped_->GetDefaultPlaybackDevice(device) : false;
-}
-
-bool SoundSystemProxy::GetDefaultCaptureDevice(
- SoundDeviceLocator **device) {
- return wrapped_ ? wrapped_->GetDefaultCaptureDevice(device) : false;
-}
-
-SoundOutputStreamInterface *SoundSystemProxy::OpenPlaybackDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms) {
- return wrapped_ ? wrapped_->OpenPlaybackDevice(device, params) : NULL;
-}
-
-SoundInputStreamInterface *SoundSystemProxy::OpenCaptureDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms) {
- return wrapped_ ? wrapped_->OpenCaptureDevice(device, params) : NULL;
-}
-
-} // namespace cricket
diff --git a/sound/soundsystemproxy.h b/sound/soundsystemproxy.h
deleted file mode 100644
index 0b8a3da..0000000
--- a/sound/soundsystemproxy.h
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_SOUND_SOUNDSYSTEMPROXY_H_
-#define TALK_SOUND_SOUNDSYSTEMPROXY_H_
-
-#include "talk/sound/soundsysteminterface.h"
-#include "webrtc/base/basictypes.h" // for NULL
-
-namespace cricket {
-
-// A SoundSystemProxy is a sound system that defers to another one.
-// Init(), Terminate(), and GetName() are left as pure virtual, so a sub-class
-// must define them.
-class SoundSystemProxy : public SoundSystemInterface {
- public:
- SoundSystemProxy() : wrapped_(NULL) {}
-
- // Each of these methods simply defers to wrapped_ if non-NULL, else fails.
-
- virtual bool EnumeratePlaybackDevices(SoundDeviceLocatorList *devices);
- virtual bool EnumerateCaptureDevices(SoundDeviceLocatorList *devices);
-
- virtual bool GetDefaultPlaybackDevice(SoundDeviceLocator **device);
- virtual bool GetDefaultCaptureDevice(SoundDeviceLocator **device);
-
- virtual SoundOutputStreamInterface *OpenPlaybackDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms);
- virtual SoundInputStreamInterface *OpenCaptureDevice(
- const SoundDeviceLocator *device,
- const OpenParams ¶ms);
-
- protected:
- SoundSystemInterface *wrapped_;
-};
-
-} // namespace cricket
-
-#endif // TALK_SOUND_SOUNDSYSTEMPROXY_H_
diff --git a/xmpp/asyncsocket.h b/xmpp/asyncsocket.h
index 5e83b63..2514742 100644
--- a/xmpp/asyncsocket.h
+++ b/xmpp/asyncsocket.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef _ASYNCSOCKET_H_
-#define _ASYNCSOCKET_H_
+#ifndef TALK_XMPP_ASYNCSOCKET_H_
+#define TALK_XMPP_ASYNCSOCKET_H_
#include <string>
@@ -86,4 +86,4 @@
}
-#endif
+#endif // TALK_XMPP_ASYNCSOCKET_H_
diff --git a/xmpp/chatroommodule.h b/xmpp/chatroommodule.h
index 47a7106..8358fc1 100644
--- a/xmpp/chatroommodule.h
+++ b/xmpp/chatroommodule.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef _multiuserchatmodule_h_
-#define _multiuserchatmodule_h_
+#ifndef TALK_XMPP_CHATROOMMODULE_H_
+#define TALK_XMPP_CHATROOMMODULE_H_
#include "talk/xmpp/module.h"
#include "talk/xmpp/rostermodule.h"
@@ -267,4 +267,4 @@
}
-#endif
+#endif // TALK_XMPP_CHATROOMMODULE_H_
diff --git a/xmpp/constants.cc b/xmpp/constants.cc
index a2179ca..297eafd 100644
--- a/xmpp/constants.cc
+++ b/xmpp/constants.cc
@@ -29,9 +29,9 @@
#include <string>
-#include "talk/xmllite/qname.h"
-#include "talk/xmllite/xmlconstants.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/xmlconstants.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/jid.h"
#include "webrtc/base/basicdefs.h"
diff --git a/xmpp/constants.h b/xmpp/constants.h
index 6d94095..6aa1a54 100644
--- a/xmpp/constants.h
+++ b/xmpp/constants.h
@@ -29,7 +29,7 @@
#define TALK_XMPP_CONSTANTS_H_
#include <string>
-#include "talk/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/qname.h"
#include "talk/xmpp/jid.h"
namespace buzz {
diff --git a/xmpp/hangoutpubsubclient.cc b/xmpp/hangoutpubsubclient.cc
index 834f844..63f5bcf 100644
--- a/xmpp/hangoutpubsubclient.cc
+++ b/xmpp/hangoutpubsubclient.cc
@@ -27,8 +27,8 @@
#include "talk/xmpp/hangoutpubsubclient.h"
-#include "talk/xmllite/qname.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/jid.h"
#include "webrtc/base/logging.h"
diff --git a/xmpp/hangoutpubsubclient_unittest.cc b/xmpp/hangoutpubsubclient_unittest.cc
index c12b791..555ee5c 100644
--- a/xmpp/hangoutpubsubclient_unittest.cc
+++ b/xmpp/hangoutpubsubclient_unittest.cc
@@ -3,8 +3,8 @@
#include <string>
-#include "talk/xmllite/qname.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/fakexmppclient.h"
#include "talk/xmpp/hangoutpubsubclient.h"
diff --git a/xmpp/jid.h b/xmpp/jid.h
index 7a6370c..2bfe0d9 100644
--- a/xmpp/jid.h
+++ b/xmpp/jid.h
@@ -29,7 +29,7 @@
#define TALK_XMPP_JID_H_
#include <string>
-#include "talk/xmllite/xmlconstants.h"
+#include "webrtc/libjingle/xmllite/xmlconstants.h"
#include "webrtc/base/basictypes.h"
namespace buzz {
diff --git a/xmpp/jingleinfotask.h b/xmpp/jingleinfotask.h
index 5865a77..18cb5a9 100644
--- a/xmpp/jingleinfotask.h
+++ b/xmpp/jingleinfotask.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_EXAMPLES_LOGIN_JINGLEINFOTASK_H_
-#define TALK_EXAMPLES_LOGIN_JINGLEINFOTASK_H_
+#ifndef TALK_XMPP_JINGLEINFOTASK_H_
+#define TALK_XMPP_JINGLEINFOTASK_H_
#include <vector>
@@ -58,4 +58,4 @@
};
}
-#endif // TALK_EXAMPLES_LOGIN_JINGLEINFOTASK_H_
+#endif // TALK_XMPP_JINGLEINFOTASK_H_
diff --git a/xmpp/module.h b/xmpp/module.h
index 37396ef..a5d0687 100644
--- a/xmpp/module.h
+++ b/xmpp/module.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef _module_h_
-#define _module_h_
+#ifndef TALK_XMPP_MODULE_H_
+#define TALK_XMPP_MODULE_H_
#include "talk/xmpp/xmppengine.h"
@@ -49,4 +49,4 @@
};
}
-#endif
+#endif // TALK_XMPP_MODULE_H_
diff --git a/xmpp/moduleimpl.h b/xmpp/moduleimpl.h
index 32a182b..897bfce 100644
--- a/xmpp/moduleimpl.h
+++ b/xmpp/moduleimpl.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef _moduleimpl_h_
-#define _moduleimpl_h_
+#ifndef TALK_XMPP_MODULEIMPL_H_
+#define TALK_XMPP_MODULEIMPL_H_
#include "talk/xmpp/module.h"
#include "talk/xmpp/xmppengine.h"
@@ -90,4 +90,4 @@
}
-#endif
+#endif // TALK_XMPP_MODULEIMPL_H_
diff --git a/xmpp/mucroomconfigtask_unittest.cc b/xmpp/mucroomconfigtask_unittest.cc
index 00a9c6e..bf5e7b4 100644
--- a/xmpp/mucroomconfigtask_unittest.cc
+++ b/xmpp/mucroomconfigtask_unittest.cc
@@ -28,7 +28,7 @@
#include <string>
#include <vector>
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/fakexmppclient.h"
#include "talk/xmpp/mucroomconfigtask.h"
diff --git a/xmpp/mucroomdiscoverytask_unittest.cc b/xmpp/mucroomdiscoverytask_unittest.cc
index ef7e477..e1a633e 100644
--- a/xmpp/mucroomdiscoverytask_unittest.cc
+++ b/xmpp/mucroomdiscoverytask_unittest.cc
@@ -28,7 +28,7 @@
#include <string>
#include <vector>
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/fakexmppclient.h"
#include "talk/xmpp/mucroomdiscoverytask.h"
diff --git a/xmpp/mucroomlookuptask_unittest.cc b/xmpp/mucroomlookuptask_unittest.cc
index 341fd3f..03be292 100644
--- a/xmpp/mucroomlookuptask_unittest.cc
+++ b/xmpp/mucroomlookuptask_unittest.cc
@@ -28,7 +28,7 @@
#include <string>
#include <vector>
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/fakexmppclient.h"
#include "talk/xmpp/mucroomlookuptask.h"
diff --git a/xmpp/mucroomuniquehangoutidtask_unittest.cc b/xmpp/mucroomuniquehangoutidtask_unittest.cc
index 5cabcbe..42bed13 100644
--- a/xmpp/mucroomuniquehangoutidtask_unittest.cc
+++ b/xmpp/mucroomuniquehangoutidtask_unittest.cc
@@ -28,7 +28,7 @@
#include <string>
#include <vector>
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/fakexmppclient.h"
#include "talk/xmpp/mucroomuniquehangoutidtask.h"
diff --git a/xmpp/pingtask_unittest.cc b/xmpp/pingtask_unittest.cc
index ce595a8..fe88a5c 100644
--- a/xmpp/pingtask_unittest.cc
+++ b/xmpp/pingtask_unittest.cc
@@ -28,7 +28,7 @@
#include <string>
#include <vector>
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/fakexmppclient.h"
#include "talk/xmpp/pingtask.h"
diff --git a/xmpp/plainsaslhandler.h b/xmpp/plainsaslhandler.h
index 2ca364c..31032e4 100644
--- a/xmpp/plainsaslhandler.h
+++ b/xmpp/plainsaslhandler.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef _PLAINSASLHANDLER_H_
-#define _PLAINSASLHANDLER_H_
+#ifndef TALK_XMPP_PLAINSASLHANDLER_H_
+#define TALK_XMPP_PLAINSASLHANDLER_H_
#include <algorithm>
#include "talk/xmpp/saslhandler.h"
@@ -78,4 +78,4 @@
}
-#endif
+#endif // TALK_XMPP_PLAINSASLHANDLER_H_
diff --git a/xmpp/presenceouttask.h b/xmpp/presenceouttask.h
index a72457e..53bbae5 100644
--- a/xmpp/presenceouttask.h
+++ b/xmpp/presenceouttask.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef _PRESENCEOUTTASK_H_
-#define _PRESENCEOUTTASK_H_
+#ifndef TALK_XMPP_PRESENCEOUTTASK_H_
+#define TALK_XMPP_PRESENCEOUTTASK_H_
#include "talk/xmpp/presencestatus.h"
#include "talk/xmpp/xmppengine.h"
@@ -51,4 +51,4 @@
}
-#endif
+#endif // TALK_XMPP_PRESENCEOUTTASK_H_
diff --git a/xmpp/pubsub_task.h b/xmpp/pubsub_task.h
index 45a7462..2787cbc 100644
--- a/xmpp/pubsub_task.h
+++ b/xmpp/pubsub_task.h
@@ -30,7 +30,7 @@
#include <map>
#include <string>
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/jid.h"
#include "talk/xmpp/xmpptask.h"
diff --git a/xmpp/pubsubclient_unittest.cc b/xmpp/pubsubclient_unittest.cc
index 135dc37..f191a18 100644
--- a/xmpp/pubsubclient_unittest.cc
+++ b/xmpp/pubsubclient_unittest.cc
@@ -3,8 +3,8 @@
#include <string>
-#include "talk/xmllite/qname.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/fakexmppclient.h"
#include "talk/xmpp/jid.h"
diff --git a/xmpp/pubsubstateclient.h b/xmpp/pubsubstateclient.h
index 2acbdaa..09ef0f4 100644
--- a/xmpp/pubsubstateclient.h
+++ b/xmpp/pubsubstateclient.h
@@ -32,8 +32,8 @@
#include <string>
#include <vector>
-#include "talk/xmllite/qname.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/jid.h"
#include "talk/xmpp/pubsubclient.h"
diff --git a/xmpp/pubsubtasks_unittest.cc b/xmpp/pubsubtasks_unittest.cc
index 98cb7c1..48cd04a 100644
--- a/xmpp/pubsubtasks_unittest.cc
+++ b/xmpp/pubsubtasks_unittest.cc
@@ -3,8 +3,8 @@
#include <string>
-#include "talk/xmllite/qname.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/fakexmppclient.h"
#include "talk/xmpp/iqtask.h"
diff --git a/xmpp/rostermodule.h b/xmpp/rostermodule.h
index 7e14dc1..dfb647d 100644
--- a/xmpp/rostermodule.h
+++ b/xmpp/rostermodule.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef _rostermodule_h_
-#define _rostermodule_h_
+#ifndef TALK_XMPP_ROSTERMODULE_H_
+#define TALK_XMPP_ROSTERMODULE_H_
#include "talk/xmpp/module.h"
@@ -345,4 +345,4 @@
}
-#endif
+#endif // TALK_XMPP_ROSTERMODULE_H_
diff --git a/xmpp/rostermodule_unittest.cc b/xmpp/rostermodule_unittest.cc
index 0e2b268..cb7f773 100644
--- a/xmpp/rostermodule_unittest.cc
+++ b/xmpp/rostermodule_unittest.cc
@@ -29,7 +29,7 @@
#include <sstream>
#include <string>
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/rostermodule.h"
#include "talk/xmpp/util_unittest.h"
diff --git a/xmpp/rostermoduleimpl.h b/xmpp/rostermoduleimpl.h
index a6b15cf..37d1117 100644
--- a/xmpp/rostermoduleimpl.h
+++ b/xmpp/rostermoduleimpl.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef _rostermoduleimpl_h_
-#define _rostermoduleimpl_h_
+#ifndef TALK_XMPP_XMPPTHREAD_H_
+#define TALK_XMPP_XMPPTHREAD_H_
#include "talk/xmpp/moduleimpl.h"
#include "talk/xmpp/rostermodule.h"
@@ -299,4 +299,4 @@
}
-#endif
+#endif // TALK_XMPP_XMPPTHREAD_H_
diff --git a/xmpp/saslcookiemechanism.h b/xmpp/saslcookiemechanism.h
index 7a19d3e..eda142c 100644
--- a/xmpp/saslcookiemechanism.h
+++ b/xmpp/saslcookiemechanism.h
@@ -28,8 +28,8 @@
#ifndef TALK_XMPP_SASLCOOKIEMECHANISM_H_
#define TALK_XMPP_SASLCOOKIEMECHANISM_H_
-#include "talk/xmllite/qname.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/saslmechanism.h"
diff --git a/xmpp/saslhandler.h b/xmpp/saslhandler.h
index bead8aa..a4a73e4 100644
--- a/xmpp/saslhandler.h
+++ b/xmpp/saslhandler.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef _SASLHANDLER_H_
-#define _SASLHANDLER_H_
+#ifndef TALK_XMPP_SASLHANDLER_H_
+#define TALK_XMPP_SASLHANDLER_H_
#include <string>
#include <vector>
@@ -56,4 +56,4 @@
}
-#endif
+#endif // TALK_XMPP_SASLHANDLER_H_
diff --git a/xmpp/saslmechanism.cc b/xmpp/saslmechanism.cc
index 488007c..8877084 100644
--- a/xmpp/saslmechanism.cc
+++ b/xmpp/saslmechanism.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/saslmechanism.h"
#include "webrtc/base/base64.h"
diff --git a/xmpp/saslmechanism.h b/xmpp/saslmechanism.h
index f2e5adc..ee419d1 100644
--- a/xmpp/saslmechanism.h
+++ b/xmpp/saslmechanism.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef _SASLMECHANISM_H_
-#define _SASLMECHANISM_H_
+#ifndef TALK_XMPP_SASLMECHANISM_H_
+#define TALK_XMPP_SASLMECHANISM_H_
#include <string>
@@ -71,4 +71,4 @@
}
-#endif
+#endif // TALK_XMPP_SASLMECHANISM_H_
diff --git a/xmpp/util_unittest.cc b/xmpp/util_unittest.cc
index 66fb5ef..3e47d3f 100644
--- a/xmpp/util_unittest.cc
+++ b/xmpp/util_unittest.cc
@@ -4,7 +4,7 @@
#include <iostream>
#include <sstream>
#include <string>
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/util_unittest.h"
#include "talk/xmpp/xmppengine.h"
#include "webrtc/base/gunit.h"
diff --git a/xmpp/xmppengine.h b/xmpp/xmppengine.h
index 68f10fe..461e90f 100644
--- a/xmpp/xmppengine.h
+++ b/xmpp/xmppengine.h
@@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef _xmppengine_h_
-#define _xmppengine_h_
+#ifndef TALK_XMPP_XMPPENGINE_H_
+#define TALK_XMPP_XMPPENGINE_H_
// also part of the API
-#include "talk/xmllite/qname.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/qname.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/jid.h"
@@ -346,4 +346,4 @@
} while (false) \
-#endif
+#endif // TALK_XMPP_XMPPENGINE_H_
diff --git a/xmpp/xmppengine_unittest.cc b/xmpp/xmppengine_unittest.cc
index 8d6c858..b519a65 100644
--- a/xmpp/xmppengine_unittest.cc
+++ b/xmpp/xmppengine_unittest.cc
@@ -4,7 +4,7 @@
#include <iostream>
#include <sstream>
#include <string>
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/plainsaslhandler.h"
#include "talk/xmpp/saslplainmechanism.h"
diff --git a/xmpp/xmppengineimpl.cc b/xmpp/xmppengineimpl.cc
index bba96a4..5de9de7 100644
--- a/xmpp/xmppengineimpl.cc
+++ b/xmpp/xmppengineimpl.cc
@@ -31,8 +31,8 @@
#include <sstream>
#include <vector>
-#include "talk/xmllite/xmlelement.h"
-#include "talk/xmllite/xmlprinter.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlprinter.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/saslhandler.h"
#include "talk/xmpp/xmpplogintask.h"
diff --git a/xmpp/xmpplogintask.cc b/xmpp/xmpplogintask.cc
index 2183c40..a48a94c 100644
--- a/xmpp/xmpplogintask.cc
+++ b/xmpp/xmpplogintask.cc
@@ -30,7 +30,7 @@
#include <string>
#include <vector>
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/jid.h"
#include "talk/xmpp/saslmechanism.h"
diff --git a/xmpp/xmpplogintask_unittest.cc b/xmpp/xmpplogintask_unittest.cc
index 1cc16cc..ae9a554 100644
--- a/xmpp/xmpplogintask_unittest.cc
+++ b/xmpp/xmpplogintask_unittest.cc
@@ -4,7 +4,7 @@
#include <iostream>
#include <sstream>
#include <string>
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "talk/xmpp/plainsaslhandler.h"
#include "talk/xmpp/saslplainmechanism.h"
diff --git a/xmpp/xmppstanzaparser.cc b/xmpp/xmppstanzaparser.cc
index 3e6ad47..4795839 100644
--- a/xmpp/xmppstanzaparser.cc
+++ b/xmpp/xmppstanzaparser.cc
@@ -27,7 +27,7 @@
#include "talk/xmpp/xmppstanzaparser.h"
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/constants.h"
#include "webrtc/base/common.h"
#ifdef EXPAT_RELATIVE_PATH
diff --git a/xmpp/xmppstanzaparser.h b/xmpp/xmppstanzaparser.h
index d9a8933..ffb3275 100644
--- a/xmpp/xmppstanzaparser.h
+++ b/xmpp/xmppstanzaparser.h
@@ -25,11 +25,11 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef _xmppstanzaparser_h_
-#define _xmppstanzaparser_h_
+#ifndef TALK_XMPP_XMPPSTANZAPARSER_H_
+#define TALK_XMPP_XMPPSTANZAPARSER_H_
-#include "talk/xmllite/xmlbuilder.h"
-#include "talk/xmllite/xmlparser.h"
+#include "webrtc/libjingle/xmllite/xmlbuilder.h"
+#include "webrtc/libjingle/xmllite/xmlparser.h"
namespace buzz {
@@ -94,4 +94,4 @@
}
-#endif
+#endif // TALK_XMPP_XMPPSTANZAPARSER_H_
diff --git a/xmpp/xmppstanzaparser_unittest.cc b/xmpp/xmppstanzaparser_unittest.cc
index 4e60d81..0b114c0 100644
--- a/xmpp/xmppstanzaparser_unittest.cc
+++ b/xmpp/xmppstanzaparser_unittest.cc
@@ -4,7 +4,7 @@
#include <iostream>
#include <sstream>
#include <string>
-#include "talk/xmllite/xmlelement.h"
+#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "talk/xmpp/xmppstanzaparser.h"
#include "webrtc/base/common.h"
#include "webrtc/base/gunit.h"
