Merge from Chromium at DEPS revision db3f05efe0f9
This commit was generated by merge_to_master.py.
Change-Id: Ied615f07e5617f89603f29f94846edd58c32a3a9
diff --git a/PRESUBMIT.py b/PRESUBMIT.py
index a7b4851..9e7b5da 100644
--- a/PRESUBMIT.py
+++ b/PRESUBMIT.py
@@ -23,9 +23,6 @@
# OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
# ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-# List of files that should not be committed to
-DO_NOT_SUBMIT_FILES = ["talk/media/webrtc/webrtcvideoengine.cc"]
-
def _LicenseHeader(input_api):
"""Returns the license header regexp."""
# Accept any year number from start of project to the current year
@@ -79,26 +76,11 @@
}
return license_header
-def _ProtectedFiles(input_api, output_api):
- results = []
- changed_files = []
- for f in input_api.AffectedFiles():
- changed_files.append(f.LocalPath())
- bad_files = list(set(DO_NOT_SUBMIT_FILES) & set(changed_files))
- if bad_files:
- error_type = output_api.PresubmitError
- results.append(error_type(
- 'The following affected files are only allowed to be updated when '
- 'importing libjingle',
- bad_files))
- return results
-
def _CommonChecks(input_api, output_api):
"""Checks common to both upload and commit."""
results = []
results.extend(input_api.canned_checks.CheckLicense(
input_api, output_api, _LicenseHeader(input_api)))
- results.extend(_ProtectedFiles(input_api, output_api))
return results
def CheckChangeOnUpload(input_api, output_api):
diff --git a/app/webrtc/fakeportallocatorfactory.h b/app/webrtc/fakeportallocatorfactory.h
index eee98b0..bfdc56b 100644
--- a/app/webrtc/fakeportallocatorfactory.h
+++ b/app/webrtc/fakeportallocatorfactory.h
@@ -32,7 +32,7 @@
#define TALK_APP_WEBRTC_FAKEPORTALLOCATORFACTORY_H_
#include "talk/app/webrtc/peerconnectioninterface.h"
-#include "talk/p2p/client/fakeportallocator.h"
+#include "webrtc/p2p/client/fakeportallocator.h"
namespace webrtc {
diff --git a/app/webrtc/java/android/org/webrtc/VideoRendererGui.java b/app/webrtc/java/android/org/webrtc/VideoRendererGui.java
index e0542ed..dd1692c 100644
--- a/app/webrtc/java/android/org/webrtc/VideoRendererGui.java
+++ b/app/webrtc/java/android/org/webrtc/VideoRendererGui.java
@@ -80,6 +80,11 @@
// aspect ratio is changed if necessary.
public static enum ScalingType
{ SCALE_ASPECT_FIT, SCALE_ASPECT_FILL, SCALE_FILL };
+ private static final int EGL14_SDK_VERSION =
+ android.os.Build.VERSION_CODES.JELLY_BEAN_MR1;
+ // Current SDK version.
+ private static final int CURRENT_SDK_VERSION =
+ android.os.Build.VERSION.SDK_INT;
private final String VERTEX_SHADER_STRING =
"varying vec2 interp_tc;\n" +
@@ -248,6 +253,8 @@
private FloatBuffer textureCoords;
// Flag if texture vertices or coordinates update is needed.
private boolean updateTextureProperties;
+ // Texture properties update lock.
+ private final Object updateTextureLock = new Object();
// Viewport dimensions.
private int screenWidth;
private int screenHeight;
@@ -314,36 +321,49 @@
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;
- Log.d(TAG, "ID: " + id + ". Display: " + displayWidth +
- " x " + displayHeight + ". Video: " + videoWidth +
- " x " + videoHeight);
- 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;
+ synchronized(updateTextureLock) {
+ // 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 texOffsetU = 0;
+ float texOffsetV = 0;
+ float displayWidth = (texRight - texLeft) * screenWidth / 2;
+ float displayHeight = (texTop - texBottom) * screenHeight / 2;
+ Log.d(TAG, "ID: " + id + ". Display: " + displayWidth +
+ " x " + displayHeight + ". Video: " + videoWidth +
+ " x " + videoHeight);
+ 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;
+ }
+ }
+ 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;
+ }
}
Log.d(TAG, " Texture vertices: (" + texLeft + "," + texBottom +
") - (" + texRight + "," + texTop + ")");
- // Re-allocate vertices buffer to adjust to video aspect ratio.
float textureVeticesFloat[] = new float[] {
texLeft, texTop,
texLeft, texBottom,
@@ -351,18 +371,8 @@
texRight, texBottom
};
textureVertices = directNativeFloatBuffer(textureVeticesFloat);
- }
- if (scalingType == ScalingType.SCALE_ASPECT_FILL) {
- float texOffsetU = 0;
- float texOffsetV = 0;
- // 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;
- }
+
Log.d(TAG, " Texture UV offsets: " + texOffsetU + ", " + texOffsetV);
- // Re-allocate coordinates buffer to adjust to display aspect ratio.
float textureCoordinatesFloat[] = new float[] {
texOffsetU, texOffsetV, // left top
texOffsetU, 1.0f - texOffsetV, // left bottom
@@ -371,8 +381,8 @@
};
textureCoords = directNativeFloatBuffer(textureCoordinatesFloat);
}
+ updateTextureProperties = false;
}
- updateTextureProperties = false;
}
private void draw() {
@@ -485,18 +495,23 @@
}
public void setScreenSize(final int screenWidth, final int screenHeight) {
- this.screenWidth = screenWidth;
- this.screenHeight = screenHeight;
- updateTextureProperties = true;
+ synchronized(updateTextureLock) {
+ this.screenWidth = screenWidth;
+ this.screenHeight = screenHeight;
+ updateTextureProperties = true;
+ }
}
public void setPosition(int x, int y, int width, int height,
ScalingType scalingType) {
- 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);
- updateTextureProperties = true;
+ synchronized(updateTextureLock) {
+ 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);
+ this.scalingType = scalingType;
+ updateTextureProperties = true;
+ }
}
@Override
@@ -666,8 +681,10 @@
public void onSurfaceCreated(GL10 unused, EGLConfig config) {
Log.d(TAG, "VideoRendererGui.onSurfaceCreated");
// Store render EGL context
- eglContext = EGL14.eglGetCurrentContext();
- Log.d(TAG, "VideoRendererGui EGL Context: " + eglContext);
+ if (CURRENT_SDK_VERSION >= EGL14_SDK_VERSION) {
+ eglContext = EGL14.eglGetCurrentContext();
+ Log.d(TAG, "VideoRendererGui EGL Context: " + eglContext);
+ }
// Create YUV and OES programs.
yuvProgram = createProgram(VERTEX_SHADER_STRING,
@@ -683,7 +700,7 @@
onSurfaceCreatedCalled = true;
}
checkNoGLES2Error();
- GLES20.glClearColor(0.0f, 0.0f, 0.1f, 1.0f);
+ GLES20.glClearColor(0.15f, 0.15f, 0.15f, 1.0f);
}
@Override
diff --git a/app/webrtc/java/jni/peerconnection_jni.cc b/app/webrtc/java/jni/peerconnection_jni.cc
index d2b1639..fb36cf8 100644
--- a/app/webrtc/java/jni/peerconnection_jni.cc
+++ b/app/webrtc/java/jni/peerconnection_jni.cc
@@ -273,11 +273,19 @@
LoadClass(jni, "org/webrtc/IceCandidate");
#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");
+ jclass j_decoder_class = GetClass("org/webrtc/MediaCodecVideoDecoder");
+ jmethodID j_is_egl14_supported_method = jni->GetStaticMethodID(
+ j_decoder_class, "isEGL14Supported", "()Z");
+ bool is_egl14_supported = jni->CallStaticBooleanMethod(
+ j_decoder_class, j_is_egl14_supported_method);
+ CHECK_EXCEPTION(jni);
+ if (is_egl14_supported) {
+ LoadClass(jni, "android/opengl/EGLContext");
+ }
#endif
LoadClass(jni, "org/webrtc/MediaSource$State");
LoadClass(jni, "org/webrtc/MediaStream");
@@ -578,13 +586,6 @@
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";
- }
-
virtual void OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) OVERRIDE {
ScopedLocalRefFrame local_ref_frame(jni());
@@ -3136,12 +3137,9 @@
}
JOW(jboolean, PeerConnection_nativeAddLocalStream)(
- JNIEnv* jni, jobject j_pc, jlong native_stream, jobject j_constraints) {
- scoped_ptr<ConstraintsWrapper> constraints(
- new ConstraintsWrapper(jni, j_constraints));
+ JNIEnv* jni, jobject j_pc, jlong native_stream) {
return ExtractNativePC(jni, j_pc)->AddStream(
- reinterpret_cast<MediaStreamInterface*>(native_stream),
- constraints.get());
+ reinterpret_cast<MediaStreamInterface*>(native_stream));
}
JOW(void, PeerConnection_nativeRemoveLocalStream)(
diff --git a/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java b/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
index 96d8b1b..f93f02a 100644
--- a/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
+++ b/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
@@ -95,6 +95,10 @@
private EGLDisplay eglDisplay = EGL14.EGL_NO_DISPLAY;
private EGLContext eglContext = EGL14.EGL_NO_CONTEXT;
private EGLSurface eglSurface = EGL14.EGL_NO_SURFACE;
+ private static final int EGL14_SDK_VERSION =
+ android.os.Build.VERSION_CODES.JELLY_BEAN_MR1;
+ private static final int CURRENT_SDK_VERSION =
+ android.os.Build.VERSION.SDK_INT;
private MediaCodecVideoDecoder() { }
@@ -166,6 +170,11 @@
return null; // No HW VP8 decoder.
}
+ private static boolean isEGL14Supported() {
+ Log.d(TAG, "SDK version: " + CURRENT_SDK_VERSION);
+ return (CURRENT_SDK_VERSION >= EGL14_SDK_VERSION);
+ }
+
private static boolean isPlatformSupported() {
return findVp8Decoder(false) != null;
}
diff --git a/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java b/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
index a2d6168..4c02891 100644
--- a/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
+++ b/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
@@ -43,7 +43,7 @@
// This class is an implementation detail of the Java PeerConnection API.
// MediaCodec is thread-hostile so this class must be operated on a single
// thread.
-class MediaCodecVideoEncoder {
+public class MediaCodecVideoEncoder {
// This class is constructed, operated, and destroyed by its C++ incarnation,
// so the class and its methods have non-public visibility. The API this
// class exposes aims to mimic the webrtc::VideoEncoder API as closely as
@@ -140,7 +140,7 @@
return null; // No HW VP8 encoder.
}
- private static boolean isPlatformSupported() {
+ public static boolean isPlatformSupported() {
return findVp8HwEncoder() != null;
}
diff --git a/app/webrtc/java/src/org/webrtc/PeerConnection.java b/app/webrtc/java/src/org/webrtc/PeerConnection.java
index c2617de..3aef6ff 100644
--- a/app/webrtc/java/src/org/webrtc/PeerConnection.java
+++ b/app/webrtc/java/src/org/webrtc/PeerConnection.java
@@ -71,9 +71,6 @@
/** Triggered when a new ICE candidate has been found. */
public void onIceCandidate(IceCandidate candidate);
- /** Triggered on any error. */
- public void onError();
-
/** Triggered when media is received on a new stream from remote peer. */
public void onAddStream(MediaStream stream);
@@ -147,9 +144,8 @@
candidate.sdpMid, candidate.sdpMLineIndex, candidate.sdp);
}
- public boolean addStream(
- MediaStream stream, MediaConstraints constraints) {
- boolean ret = nativeAddLocalStream(stream.nativeStream, constraints);
+ public boolean addStream(MediaStream stream) {
+ boolean ret = nativeAddLocalStream(stream.nativeStream);
if (!ret) {
return false;
}
@@ -194,8 +190,7 @@
private native boolean nativeAddIceCandidate(
String sdpMid, int sdpMLineIndex, String iceCandidateSdp);
- private native boolean nativeAddLocalStream(
- long nativeStream, MediaConstraints constraints);
+ private native boolean nativeAddLocalStream(long nativeStream);
private native void nativeRemoveLocalStream(long nativeStream);
diff --git a/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java b/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
index 240e996..048d92b 100644
--- a/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
+++ b/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
@@ -117,11 +117,6 @@
++expectedErrors;
}
- @Override
- public synchronized void onError() {
- assertTrue(--expectedErrors >= 0);
- }
-
public synchronized void expectSetSize() {
if (RENDER_TO_GUI) {
// When new frames are delivered to the GUI renderer we don't get
@@ -489,7 +484,7 @@
lMS.addTrack(videoTrack);
lMS.addTrack(factory.createAudioTrack(
audioTrackId, factory.createAudioSource(new MediaConstraints())));
- pc.addStream(lMS, new MediaConstraints());
+ pc.addStream(lMS);
return new WeakReference<MediaStream>(lMS);
}
diff --git a/app/webrtc/jsepicecandidate.h b/app/webrtc/jsepicecandidate.h
index 7be420c..71ed2c3 100644
--- a/app/webrtc/jsepicecandidate.h
+++ b/app/webrtc/jsepicecandidate.h
@@ -33,7 +33,7 @@
#include <string>
#include "talk/app/webrtc/jsep.h"
-#include "talk/p2p/base/candidate.h"
+#include "webrtc/p2p/base/candidate.h"
#include "webrtc/base/constructormagic.h"
namespace webrtc {
diff --git a/app/webrtc/jsepsessiondescription_unittest.cc b/app/webrtc/jsepsessiondescription_unittest.cc
index 769e34a..cf992c0 100644
--- a/app/webrtc/jsepsessiondescription_unittest.cc
+++ b/app/webrtc/jsepsessiondescription_unittest.cc
@@ -29,9 +29,9 @@
#include "talk/app/webrtc/jsepicecandidate.h"
#include "talk/app/webrtc/jsepsessiondescription.h"
-#include "talk/p2p/base/candidate.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/sessiondescription.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/helpers.h"
diff --git a/app/webrtc/mediastreamsignaling_unittest.cc b/app/webrtc/mediastreamsignaling_unittest.cc
index 038c67d..d7b9734 100644
--- a/app/webrtc/mediastreamsignaling_unittest.cc
+++ b/app/webrtc/mediastreamsignaling_unittest.cc
@@ -38,8 +38,8 @@
#include "talk/app/webrtc/videotrack.h"
#include "talk/media/base/fakemediaengine.h"
#include "talk/media/devices/fakedevicemanager.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/sessiondescription.h"
#include "talk/session/media/channelmanager.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/scoped_ptr.h"
diff --git a/app/webrtc/objc/RTCPeerConnection.mm b/app/webrtc/objc/RTCPeerConnection.mm
index 72ba373..925de73 100644
--- a/app/webrtc/objc/RTCPeerConnection.mm
+++ b/app/webrtc/objc/RTCPeerConnection.mm
@@ -151,10 +151,8 @@
return self.peerConnection->AddIceCandidate(iceCandidate.get());
}
-- (BOOL)addStream:(RTCMediaStream*)stream
- constraints:(RTCMediaConstraints*)constraints {
- BOOL ret = self.peerConnection->AddStream(stream.mediaStream,
- constraints.constraints);
+- (BOOL)addStream:(RTCMediaStream*)stream {
+ BOOL ret = self.peerConnection->AddStream(stream.mediaStream);
if (!ret) {
return NO;
}
diff --git a/app/webrtc/objc/RTCPeerConnectionObserver.h b/app/webrtc/objc/RTCPeerConnectionObserver.h
index f66b567..8378ff8 100644
--- a/app/webrtc/objc/RTCPeerConnectionObserver.h
+++ b/app/webrtc/objc/RTCPeerConnectionObserver.h
@@ -41,8 +41,6 @@
RTCPeerConnectionObserver(RTCPeerConnection* peerConnection);
virtual ~RTCPeerConnectionObserver();
- virtual void OnError() OVERRIDE;
-
// Triggered when the SignalingState changed.
virtual void OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) OVERRIDE;
diff --git a/app/webrtc/objc/RTCPeerConnectionObserver.mm b/app/webrtc/objc/RTCPeerConnectionObserver.mm
index a0206e5..f4cab7f 100644
--- a/app/webrtc/objc/RTCPeerConnectionObserver.mm
+++ b/app/webrtc/objc/RTCPeerConnectionObserver.mm
@@ -46,10 +46,6 @@
RTCPeerConnectionObserver::~RTCPeerConnectionObserver() {
}
-void RTCPeerConnectionObserver::OnError() {
- [_peerConnection.delegate peerConnectionOnError:_peerConnection];
-}
-
void RTCPeerConnectionObserver::OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) {
RTCSignalingState state =
diff --git a/app/webrtc/objc/public/RTCPeerConnection.h b/app/webrtc/objc/public/RTCPeerConnection.h
index 32a9830..6d47f77 100644
--- a/app/webrtc/objc/public/RTCPeerConnection.h
+++ b/app/webrtc/objc/public/RTCPeerConnection.h
@@ -64,8 +64,7 @@
// Add a new MediaStream to be sent on this PeerConnection.
// Note that a SessionDescription negotiation is needed before the
// remote peer can receive the stream.
-- (BOOL)addStream:(RTCMediaStream *)stream
- constraints:(RTCMediaConstraints *)constraints;
+- (BOOL)addStream:(RTCMediaStream *)stream;
// Remove a MediaStream from this PeerConnection.
// Note that a SessionDescription negotiation is need before the
diff --git a/app/webrtc/objc/public/RTCPeerConnectionDelegate.h b/app/webrtc/objc/public/RTCPeerConnectionDelegate.h
index 4b177d5..ee6ec7a 100644
--- a/app/webrtc/objc/public/RTCPeerConnectionDelegate.h
+++ b/app/webrtc/objc/public/RTCPeerConnectionDelegate.h
@@ -38,9 +38,6 @@
// implemented to get messages from PeerConnection.
@protocol RTCPeerConnectionDelegate<NSObject>
-// Triggered when there is an error.
-- (void)peerConnectionOnError:(RTCPeerConnection *)peerConnection;
-
// Triggered when the SignalingState changed.
- (void)peerConnection:(RTCPeerConnection *)peerConnection
signalingStateChanged:(RTCSignalingState)stateChanged;
diff --git a/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m b/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m
index c3f898a..fbcf217 100644
--- a/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m
+++ b/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m
@@ -151,11 +151,6 @@
#pragma mark - RTCPeerConnectionDelegate methods
-- (void)peerConnectionOnError:(RTCPeerConnection*)peerConnection {
- NSLog(@"RTCPeerConnectionDelegate::onError");
- NSAssert(--_expectedErrors >= 0, @"Unexpected error");
-}
-
- (void)peerConnection:(RTCPeerConnection*)peerConnection
signalingStateChanged:(RTCSignalingState)stateChanged {
int expectedState = [self popFirstElementAsInt:_expectedSignalingChanges];
diff --git a/app/webrtc/objctests/RTCPeerConnectionTest.mm b/app/webrtc/objctests/RTCPeerConnectionTest.mm
index 909503a..6c5950b 100644
--- a/app/webrtc/objctests/RTCPeerConnectionTest.mm
+++ b/app/webrtc/objctests/RTCPeerConnectionTest.mm
@@ -89,8 +89,7 @@
[localMediaStream addVideoTrack:videoTrack];
RTCAudioTrack* audioTrack = [factory audioTrackWithID:audioTrackID];
[localMediaStream addAudioTrack:audioTrack];
- RTCMediaConstraints* constraints = [[RTCMediaConstraints alloc] init];
- [pc addStream:localMediaStream constraints:constraints];
+ [pc addStream:localMediaStream];
return localMediaStream;
}
diff --git a/app/webrtc/peerconnection.cc b/app/webrtc/peerconnection.cc
index d939f98..b64caf7 100644
--- a/app/webrtc/peerconnection.cc
+++ b/app/webrtc/peerconnection.cc
@@ -35,7 +35,7 @@
#include "talk/app/webrtc/mediaconstraintsinterface.h"
#include "talk/app/webrtc/mediastreamhandler.h"
#include "talk/app/webrtc/streamcollection.h"
-#include "talk/p2p/client/basicportallocator.h"
+#include "webrtc/p2p/client/basicportallocator.h"
#include "talk/session/media/channelmanager.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
@@ -350,9 +350,10 @@
// To handle both internal and externally created port allocator, we will
// enable BUNDLE here.
- int portallocator_flags = cricket::PORTALLOCATOR_ENABLE_BUNDLE |
- cricket::PORTALLOCATOR_ENABLE_SHARED_UFRAG |
- cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET;
+ int portallocator_flags = port_allocator_->flags();
+ portallocator_flags |= cricket::PORTALLOCATOR_ENABLE_BUNDLE |
+ 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(
@@ -403,8 +404,7 @@
return mediastream_signaling_->remote_streams();
}
-bool PeerConnection::AddStream(MediaStreamInterface* local_stream,
- const MediaConstraintsInterface* constraints) {
+bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
if (IsClosed()) {
return false;
}
@@ -412,7 +412,6 @@
local_stream))
return false;
- // TODO(perkj): Implement support for MediaConstraints in AddStream.
if (!mediastream_signaling_->AddLocalStream(local_stream)) {
return false;
}
@@ -421,6 +420,11 @@
return true;
}
+bool PeerConnection::AddStream(MediaStreamInterface* local_stream,
+ const MediaConstraintsInterface* constraints) {
+ return AddStream(local_stream);
+}
+
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
mediastream_signaling_->RemoveLocalStream(local_stream);
if (IsClosed()) {
diff --git a/app/webrtc/peerconnection.h b/app/webrtc/peerconnection.h
index fb03802..355211c 100644
--- a/app/webrtc/peerconnection.h
+++ b/app/webrtc/peerconnection.h
@@ -65,6 +65,7 @@
PeerConnectionObserver* observer);
virtual rtc::scoped_refptr<StreamCollectionInterface> local_streams();
virtual rtc::scoped_refptr<StreamCollectionInterface> remote_streams();
+ virtual bool AddStream(MediaStreamInterface* local_stream);
virtual bool AddStream(MediaStreamInterface* local_stream,
const MediaConstraintsInterface* constraints);
virtual void RemoveStream(MediaStreamInterface* local_stream);
diff --git a/app/webrtc/peerconnection_unittest.cc b/app/webrtc/peerconnection_unittest.cc
index da75e9d..c250eea 100644
--- a/app/webrtc/peerconnection_unittest.cc
+++ b/app/webrtc/peerconnection_unittest.cc
@@ -46,8 +46,8 @@
#include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
#include "talk/app/webrtc/videosourceinterface.h"
#include "talk/media/webrtc/fakewebrtcvideoengine.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/sessiondescription.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/scoped_ptr.h"
@@ -179,7 +179,7 @@
stream->AddTrack(CreateLocalVideoTrack(stream_label));
}
- EXPECT_TRUE(peer_connection_->AddStream(stream, NULL));
+ EXPECT_TRUE(peer_connection_->AddStream(stream));
}
size_t NumberOfLocalMediaStreams() {
@@ -426,7 +426,6 @@
}
// PeerConnectionObserver callbacks.
- virtual void OnError() {}
virtual void OnMessage(const std::string&) {}
virtual void OnSignalingMessage(const std::string& /*msg*/) {}
virtual void OnSignalingChange(
diff --git a/app/webrtc/peerconnectionendtoend_unittest.cc b/app/webrtc/peerconnectionendtoend_unittest.cc
index 4db3fe3..d743182 100644
--- a/app/webrtc/peerconnectionendtoend_unittest.cc
+++ b/app/webrtc/peerconnectionendtoend_unittest.cc
@@ -320,7 +320,14 @@
// Verifies that a DataChannel created after the negotiation can transition to
// "OPEN" and transfer data.
-TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=3980.
+#define MAYBE_CreateDataChannelAfterNegotiate DISABLED_CreateDataChannelAfterNegotiate
+#else
+#define MAYBE_CreateDataChannelAfterNegotiate CreateDataChannelAfterNegotiate
+#endif
+TEST_F(PeerConnectionEndToEndTest, MAYBE_CreateDataChannelAfterNegotiate) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
CreatePcs();
diff --git a/app/webrtc/peerconnectionfactory_unittest.cc b/app/webrtc/peerconnectionfactory_unittest.cc
index 5995c46..e687b8b 100644
--- a/app/webrtc/peerconnectionfactory_unittest.cc
+++ b/app/webrtc/peerconnectionfactory_unittest.cc
@@ -40,6 +40,7 @@
#include "webrtc/base/thread.h"
using webrtc::FakeVideoTrackRenderer;
+using webrtc::DataChannelInterface;
using webrtc::MediaStreamInterface;
using webrtc::PeerConnectionFactoryInterface;
using webrtc::PeerConnectionInterface;
@@ -83,13 +84,13 @@
class NullPeerConnectionObserver : public PeerConnectionObserver {
public:
- virtual void OnError() {}
virtual void OnMessage(const std::string& msg) {}
virtual void OnSignalingMessage(const std::string& msg) {}
virtual void OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) {}
virtual void OnAddStream(MediaStreamInterface* stream) {}
virtual void OnRemoveStream(MediaStreamInterface* stream) {}
+ virtual void OnDataChannel(DataChannelInterface* data_channel) {}
virtual void OnRenegotiationNeeded() {}
virtual void OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) {}
diff --git a/app/webrtc/peerconnectioninterface.h b/app/webrtc/peerconnectioninterface.h
index 6ef4847..68b7879 100644
--- a/app/webrtc/peerconnectioninterface.h
+++ b/app/webrtc/peerconnectioninterface.h
@@ -255,8 +255,15 @@
// Add a new MediaStream to be sent on this PeerConnection.
// Note that a SessionDescription negotiation is needed before the
// remote peer can receive the stream.
+ // TODO(perkj): Make pure virtual once Chrome mocks have implemented.
+ virtual bool AddStream(MediaStreamInterface* stream) { return false;}
+
+ // Deprecated:
+ // TODO(perkj): Remove once its not used by Chrome.
virtual bool AddStream(MediaStreamInterface* stream,
- const MediaConstraintsInterface* constraints) = 0;
+ const MediaConstraintsInterface* constraints) {
+ return false;
+ }
// Remove a MediaStream from this PeerConnection.
// Note that a SessionDescription negotiation is need before the
@@ -344,7 +351,9 @@
kIceState,
};
- virtual void OnError() = 0;
+ // Deprecated.
+ // TODO(perkj): Remove once its not used by Chrome.
+ virtual void OnError() {}
// Triggered when the SignalingState changed.
virtual void OnSignalingChange(
@@ -361,8 +370,7 @@
virtual void OnRemoveStream(MediaStreamInterface* stream) = 0;
// Triggered when a remote peer open a data channel.
- // TODO(perkj): Make pure virtual.
- virtual void OnDataChannel(DataChannelInterface* data_channel) {}
+ virtual void OnDataChannel(DataChannelInterface* data_channel) = 0;
// Triggered when renegotiation is needed, for example the ICE has restarted.
virtual void OnRenegotiationNeeded() = 0;
diff --git a/app/webrtc/peerconnectioninterface_unittest.cc b/app/webrtc/peerconnectioninterface_unittest.cc
index bf60673..3be6280 100644
--- a/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/app/webrtc/peerconnectioninterface_unittest.cc
@@ -132,7 +132,6 @@
state_ = pc_->signaling_state();
}
}
- virtual void OnError() {}
virtual void OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) {
EXPECT_EQ(pc_->signaling_state(), new_state);
@@ -320,7 +319,7 @@
scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack(label + "v0", video_source));
stream->AddTrack(video_track.get());
- EXPECT_TRUE(pc_->AddStream(stream, NULL));
+ EXPECT_TRUE(pc_->AddStream(stream));
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
@@ -332,7 +331,7 @@
scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack(label + "a0", NULL));
stream->AddTrack(audio_track.get());
- EXPECT_TRUE(pc_->AddStream(stream, NULL));
+ EXPECT_TRUE(pc_->AddStream(stream));
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
@@ -350,7 +349,7 @@
scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack(video_track_label, NULL));
stream->AddTrack(video_track.get());
- EXPECT_TRUE(pc_->AddStream(stream, NULL));
+ EXPECT_TRUE(pc_->AddStream(stream));
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
@@ -574,7 +573,7 @@
pc_factory_->CreateAudioTrack(
kStreamLabel3, static_cast<AudioSourceInterface*>(NULL)));
stream->AddTrack(audio_track.get());
- EXPECT_TRUE(pc_->AddStream(stream, NULL));
+ EXPECT_TRUE(pc_->AddStream(stream));
EXPECT_EQ(3u, pc_->local_streams()->count());
// Remove the third stream.
@@ -1180,7 +1179,7 @@
pc_->Close();
pc_->RemoveStream(local_stream);
- EXPECT_FALSE(pc_->AddStream(local_stream, NULL));
+ EXPECT_FALSE(pc_->AddStream(local_stream));
ASSERT_FALSE(local_stream->GetAudioTracks().empty());
rtc::scoped_refptr<webrtc::DtmfSenderInterface> dtmf_sender(
diff --git a/app/webrtc/peerconnectionproxy.h b/app/webrtc/peerconnectionproxy.h
index ed26eb8..571c676 100644
--- a/app/webrtc/peerconnectionproxy.h
+++ b/app/webrtc/peerconnectionproxy.h
@@ -39,8 +39,9 @@
local_streams)
PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>,
remote_streams)
+ PROXY_METHOD1(bool, AddStream, MediaStreamInterface*)
PROXY_METHOD2(bool, AddStream, MediaStreamInterface*,
- const MediaConstraintsInterface*)
+ const MediaConstraintsInterface*);
PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*)
PROXY_METHOD1(rtc::scoped_refptr<DtmfSenderInterface>,
CreateDtmfSender, AudioTrackInterface*)
diff --git a/app/webrtc/portallocatorfactory.cc b/app/webrtc/portallocatorfactory.cc
index 946f8ad..da4c706 100644
--- a/app/webrtc/portallocatorfactory.cc
+++ b/app/webrtc/portallocatorfactory.cc
@@ -27,8 +27,8 @@
#include "talk/app/webrtc/portallocatorfactory.h"
-#include "talk/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/client/basicportallocator.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/client/basicportallocator.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/network.h"
#include "webrtc/base/thread.h"
diff --git a/app/webrtc/proxy.h b/app/webrtc/proxy.h
index 0c21ef9..134f580 100644
--- a/app/webrtc/proxy.h
+++ b/app/webrtc/proxy.h
@@ -55,6 +55,7 @@
#ifndef TALK_APP_WEBRTC_PROXY_H_
#define TALK_APP_WEBRTC_PROXY_H_
+#include "webrtc/base/event.h"
#include "webrtc/base/thread.h"
namespace webrtc {
@@ -92,6 +93,34 @@
void value() {}
};
+namespace internal {
+
+class SynchronousMethodCall
+ : public rtc::MessageData,
+ public rtc::MessageHandler {
+ public:
+ SynchronousMethodCall(rtc::MessageHandler* proxy)
+ : e_(), proxy_(proxy) {}
+ ~SynchronousMethodCall() {}
+
+ void Invoke(rtc::Thread* t) {
+ if (t->IsCurrent()) {
+ proxy_->OnMessage(NULL);
+ } else {
+ e_.reset(new rtc::Event(false, false));
+ t->Post(this, 0);
+ e_->Wait(rtc::kForever);
+ }
+ }
+
+ private:
+ void OnMessage(rtc::Message*) { proxy_->OnMessage(NULL); e_->Set(); }
+ rtc::scoped_ptr<rtc::Event> e_;
+ rtc::MessageHandler* proxy_;
+};
+
+} // internal
+
template <typename C, typename R>
class MethodCall0 : public rtc::Message,
public rtc::MessageHandler {
@@ -100,12 +129,12 @@
MethodCall0(C* c, Method m) : c_(c), m_(m) {}
R Marshal(rtc::Thread* t) {
- t->Send(this, 0);
+ internal::SynchronousMethodCall(this).Invoke(t);
return r_.value();
}
private:
- void OnMessage(rtc::Message*) { r_.Invoke(c_, m_);}
+ void OnMessage(rtc::Message*) { r_.Invoke(c_, m_); }
C* c_;
Method m_;
@@ -120,7 +149,7 @@
ConstMethodCall0(C* c, Method m) : c_(c), m_(m) {}
R Marshal(rtc::Thread* t) {
- t->Send(this, 0);
+ internal::SynchronousMethodCall(this).Invoke(t);
return r_.value();
}
@@ -140,7 +169,7 @@
MethodCall1(C* c, Method m, T1 a1) : c_(c), m_(m), a1_(a1) {}
R Marshal(rtc::Thread* t) {
- t->Send(this, 0);
+ internal::SynchronousMethodCall(this).Invoke(t);
return r_.value();
}
@@ -161,7 +190,7 @@
ConstMethodCall1(C* c, Method m, T1 a1) : c_(c), m_(m), a1_(a1) {}
R Marshal(rtc::Thread* t) {
- t->Send(this, 0);
+ internal::SynchronousMethodCall(this).Invoke(t);
return r_.value();
}
@@ -182,7 +211,7 @@
MethodCall2(C* c, Method m, T1 a1, T2 a2) : c_(c), m_(m), a1_(a1), a2_(a2) {}
R Marshal(rtc::Thread* t) {
- t->Send(this, 0);
+ internal::SynchronousMethodCall(this).Invoke(t);
return r_.value();
}
@@ -205,7 +234,7 @@
: c_(c), m_(m), a1_(a1), a2_(a2), a3_(a3) {}
R Marshal(rtc::Thread* t) {
- t->Send(this, 0);
+ internal::SynchronousMethodCall(this).Invoke(t);
return r_.value();
}
diff --git a/app/webrtc/statscollector.cc b/app/webrtc/statscollector.cc
index 5271827..fefb0ad 100644
--- a/app/webrtc/statscollector.cc
+++ b/app/webrtc/statscollector.cc
@@ -89,8 +89,6 @@
const char StatsReport::kStatsValueNameEchoReturnLossEnhancement[] =
"googEchoCancellationReturnLossEnhancement";
-const char StatsReport::kStatsValueNameEncodeRelStdDev[] =
- "googEncodeRelStdDev";
const char StatsReport::kStatsValueNameEncodeUsagePercent[] =
"googEncodeUsagePercent";
const char StatsReport::kStatsValueNameExpandRate[] = "googExpandRate";
@@ -459,8 +457,6 @@
info.capture_queue_delay_ms_per_s);
report->AddValue(StatsReport::kStatsValueNameEncodeUsagePercent,
info.encode_usage_percent);
- report->AddValue(StatsReport::kStatsValueNameEncodeRelStdDev,
- info.encode_rsd);
}
void ExtractStats(const cricket::BandwidthEstimationInfo& info,
diff --git a/app/webrtc/statscollector_unittest.cc b/app/webrtc/statscollector_unittest.cc
index 2e55af9..c573a88 100644
--- a/app/webrtc/statscollector_unittest.cc
+++ b/app/webrtc/statscollector_unittest.cc
@@ -35,7 +35,7 @@
#include "talk/app/webrtc/videotrack.h"
#include "talk/media/base/fakemediaengine.h"
#include "talk/media/devices/fakedevicemanager.h"
-#include "talk/p2p/base/fakesession.h"
+#include "webrtc/p2p/base/fakesession.h"
#include "talk/session/media/channelmanager.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
diff --git a/app/webrtc/statstypes.h b/app/webrtc/statstypes.h
index 656b83c..8a1530a 100644
--- a/app/webrtc/statstypes.h
+++ b/app/webrtc/statstypes.h
@@ -199,7 +199,6 @@
// Internal StatsValue names
static const char kStatsValueNameAvgEncodeMs[];
- static const char kStatsValueNameEncodeRelStdDev[];
static const char kStatsValueNameEncodeUsagePercent[];
static const char kStatsValueNameCaptureJitterMs[];
static const char kStatsValueNameCaptureQueueDelayMsPerS[];
diff --git a/app/webrtc/test/peerconnectiontestwrapper.cc b/app/webrtc/test/peerconnectiontestwrapper.cc
index 24932b8..e3b8015 100644
--- a/app/webrtc/test/peerconnectiontestwrapper.cc
+++ b/app/webrtc/test/peerconnectiontestwrapper.cc
@@ -253,7 +253,7 @@
bool video, const webrtc::FakeConstraints& video_constraints) {
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
GetUserMedia(audio, audio_constraints, video, video_constraints);
- EXPECT_TRUE(peer_connection_->AddStream(stream, NULL));
+ EXPECT_TRUE(peer_connection_->AddStream(stream));
}
rtc::scoped_refptr<webrtc::MediaStreamInterface>
diff --git a/app/webrtc/test/peerconnectiontestwrapper.h b/app/webrtc/test/peerconnectiontestwrapper.h
index d4a0e4e..d8299ec 100644
--- a/app/webrtc/test/peerconnectiontestwrapper.h
+++ b/app/webrtc/test/peerconnectiontestwrapper.h
@@ -57,7 +57,6 @@
const webrtc::DataChannelInit& init);
// Implements PeerConnectionObserver.
- virtual void OnError() {}
virtual void OnSignalingChange(
webrtc::PeerConnectionInterface::SignalingState new_state) {}
virtual void OnStateChange(
diff --git a/app/webrtc/webrtcsdp.cc b/app/webrtc/webrtcsdp.cc
index 5887409..23b8f3d 100644
--- a/app/webrtc/webrtcsdp.cc
+++ b/app/webrtc/webrtcsdp.cc
@@ -39,9 +39,9 @@
#include "talk/media/base/constants.h"
#include "talk/media/base/cryptoparams.h"
#include "talk/media/sctp/sctpdataengine.h"
-#include "talk/p2p/base/candidate.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/port.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/port.h"
#include "talk/session/media/mediasession.h"
#include "talk/session/media/mediasessionclient.h"
#include "webrtc/base/common.h"
@@ -197,13 +197,17 @@
static const char kTimeDescription[] = "t=0 0";
static const char kAttrGroup[] = "a=group:BUNDLE";
static const char kConnectionNettype[] = "IN";
-static const char kConnectionAddrtype[] = "IP4";
+static const char kConnectionIpv4Addrtype[] = "IP4";
+static const char kConnectionIpv6Addrtype[] = "IP6";
static const char kMediaTypeVideo[] = "video";
static const char kMediaTypeAudio[] = "audio";
static const char kMediaTypeData[] = "application";
static const char kMediaPortRejected[] = "0";
-static const char kDefaultAddress[] = "0.0.0.0";
-static const char kDefaultPort[] = "1";
+// draft-ietf-mmusic-trickle-ice-01
+// When no candidates have been gathered, set the connection
+// address to IP6 ::.
+static const char kDummyAddress[] = "::";
+static const char kDummyPort[] = "9";
// RFC 3556
static const char kApplicationSpecificMaximum[] = "AS";
@@ -667,10 +671,13 @@
// The value of |component_id| currently supported are 1 (RTP) and 2 (RTCP).
// TODO: Decide the default destination in webrtcsession and
// pass it down via SessionDescription.
-static bool GetDefaultDestination(const std::vector<Candidate>& candidates,
- int component_id, std::string* port, std::string* ip) {
- *port = kDefaultPort;
- *ip = kDefaultAddress;
+static void GetDefaultDestination(
+ const std::vector<Candidate>& candidates,
+ int component_id, std::string* port,
+ std::string* ip, std::string* addr_type) {
+ *addr_type = kConnectionIpv6Addrtype;
+ *port = kDummyPort;
+ *ip = kDummyAddress;
int current_preference = kPreferenceUnknown;
for (std::vector<Candidate>::const_iterator it = candidates.begin();
it != candidates.end(); ++it) {
@@ -685,8 +692,13 @@
current_preference = preference;
*port = it->address().PortAsString();
*ip = it->address().ipaddr().ToString();
+ int family = it->address().ipaddr().family();
+ if (family == AF_INET) {
+ addr_type->assign(kConnectionIpv4Addrtype);
+ } else if (family == AF_INET6) {
+ addr_type->assign(kConnectionIpv6Addrtype);
+ }
}
- return true;
}
// Update |mline|'s default destination and append a c line after it.
@@ -705,55 +717,52 @@
}
std::ostringstream os;
- std::string rtp_port, rtp_ip;
- if (GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTP,
- &rtp_port, &rtp_ip)) {
- // Found default RTP candidate.
- // RFC 5245
- // The default candidates are added to the SDP as the default
- // destination for media. For streams based on RTP, this is done by
- // placing the IP address and port of the RTP candidate into the c and m
- // lines, respectively.
-
- // Update the port in the m line.
- // If this is a m-line with port equal to 0, we don't change it.
- if (fields[1] != kMediaPortRejected) {
- new_lines.replace(fields[0].size() + 1,
- fields[1].size(),
- rtp_port);
- }
- // Add the c line.
- // RFC 4566
- // c=<nettype> <addrtype> <connection-address>
- InitLine(kLineTypeConnection, kConnectionNettype, &os);
- os << " " << kConnectionAddrtype << " " << rtp_ip;
- AddLine(os.str(), &new_lines);
+ std::string rtp_port, rtp_ip, addr_type;
+ GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTP,
+ &rtp_port, &rtp_ip, &addr_type);
+ // Found default RTP candidate.
+ // RFC 5245
+ // The default candidates are added to the SDP as the default
+ // destination for media. For streams based on RTP, this is done by
+ // placing the IP address and port of the RTP candidate into the c and m
+ // lines, respectively.
+ // Update the port in the m line.
+ // If this is a m-line with port equal to 0, we don't change it.
+ if (fields[1] != kMediaPortRejected) {
+ new_lines.replace(fields[0].size() + 1,
+ fields[1].size(),
+ rtp_port);
}
+ // Add the c line.
+ // RFC 4566
+ // c=<nettype> <addrtype> <connection-address>
+ InitLine(kLineTypeConnection, kConnectionNettype, &os);
+ os << " " << addr_type << " " << rtp_ip;
+ AddLine(os.str(), &new_lines);
message->append(new_lines);
}
// Gets "a=rtcp" line if found default RTCP candidate from |candidates|.
static std::string GetRtcpLine(const std::vector<Candidate>& candidates) {
- std::string rtcp_line, rtcp_port, rtcp_ip;
- if (GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTCP,
- &rtcp_port, &rtcp_ip)) {
- // Found default RTCP candidate.
- // RFC 5245
- // If the agent is utilizing RTCP, it MUST encode the RTCP candidate
- // using the a=rtcp attribute as defined in RFC 3605.
+ std::string rtcp_line, rtcp_port, rtcp_ip, addr_type;
+ GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTCP,
+ &rtcp_port, &rtcp_ip, &addr_type);
+ // Found default RTCP candidate.
+ // RFC 5245
+ // If the agent is utilizing RTCP, it MUST encode the RTCP candidate
+ // using the a=rtcp attribute as defined in RFC 3605.
- // RFC 3605
- // rtcp-attribute = "a=rtcp:" port [nettype space addrtype space
- // connection-address] CRLF
- std::ostringstream os;
- InitAttrLine(kAttributeRtcp, &os);
- os << kSdpDelimiterColon
- << rtcp_port << " "
- << kConnectionNettype << " "
- << kConnectionAddrtype << " "
- << rtcp_ip;
- rtcp_line = os.str();
- }
+ // RFC 3605
+ // rtcp-attribute = "a=rtcp:" port [nettype space addrtype space
+ // connection-address] CRLF
+ std::ostringstream os;
+ InitAttrLine(kAttributeRtcp, &os);
+ os << kSdpDelimiterColon
+ << rtcp_port << " "
+ << kConnectionNettype << " "
+ << addr_type << " "
+ << rtcp_ip;
+ rtcp_line = os.str();
return rtcp_line;
}
@@ -1243,7 +1252,7 @@
// To reject an offered stream, the port number in the corresponding stream in
// the answer MUST be set to zero.
const std::string port = content_info->rejected ?
- kMediaPortRejected : kDefaultPort;
+ kMediaPortRejected : kDummyPort;
rtc::SSLFingerprint* fp = (transport_info) ?
transport_info->description.identity_fingerprint.get() : NULL;
diff --git a/app/webrtc/webrtcsdp_unittest.cc b/app/webrtc/webrtcsdp_unittest.cc
index 560d5da..ea590db 100644
--- a/app/webrtc/webrtcsdp_unittest.cc
+++ b/app/webrtc/webrtcsdp_unittest.cc
@@ -32,7 +32,7 @@
#include "talk/app/webrtc/jsepsessiondescription.h"
#include "talk/app/webrtc/webrtcsdp.h"
#include "talk/media/base/constants.h"
-#include "talk/p2p/base/constants.h"
+#include "webrtc/p2p/base/constants.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
@@ -212,9 +212,9 @@
"s=-\r\n"
"t=0 0\r\n"
"a=msid-semantic: WMS local_stream_1 local_stream_2\r\n"
- "m=audio 1 RTP/SAVPF 111 103 104\r\n"
- "c=IN IP4 0.0.0.0\r\n"
- "a=rtcp:1 IN IP4 0.0.0.0\r\n"
+ "m=audio 9 RTP/SAVPF 111 103 104\r\n"
+ "c=IN IP6 ::\r\n"
+ "a=rtcp:9 IN IP6 ::\r\n"
"a=ice-ufrag:ufrag_voice\r\na=ice-pwd:pwd_voice\r\n"
"a=mid:audio_content_name\r\n"
"a=sendrecv\r\n"
@@ -233,9 +233,9 @@
"a=ssrc:4 msid:local_stream_2 audio_track_id_2\r\n"
"a=ssrc:4 mslabel:local_stream_2\r\n"
"a=ssrc:4 label:audio_track_id_2\r\n"
- "m=video 1 RTP/SAVPF 120\r\n"
- "c=IN IP4 0.0.0.0\r\n"
- "a=rtcp:1 IN IP4 0.0.0.0\r\n"
+ "m=video 9 RTP/SAVPF 120\r\n"
+ "c=IN IP6 ::\r\n"
+ "a=rtcp:9 IN IP6 ::\r\n"
"a=ice-ufrag:ufrag_video\r\na=ice-pwd:pwd_video\r\n"
"a=mid:video_content_name\r\n"
"a=sendrecv\r\n"
@@ -261,9 +261,9 @@
"a=ssrc:6 label:video_track_id_3\r\n";
static const char kSdpRtpDataChannelString[] =
- "m=application 1 RTP/SAVPF 101\r\n"
- "c=IN IP4 0.0.0.0\r\n"
- "a=rtcp:1 IN IP4 0.0.0.0\r\n"
+ "m=application 9 RTP/SAVPF 101\r\n"
+ "c=IN IP6 ::\r\n"
+ "a=rtcp:9 IN IP6 ::\r\n"
"a=ice-ufrag:ufrag_data\r\n"
"a=ice-pwd:pwd_data\r\n"
"a=mid:data_content_name\r\n"
@@ -277,8 +277,8 @@
"a=ssrc:10 label:data_channeld0\r\n";
static const char kSdpSctpDataChannelString[] =
- "m=application 1 DTLS/SCTP 5000\r\n"
- "c=IN IP4 0.0.0.0\r\n"
+ "m=application 9 DTLS/SCTP 5000\r\n"
+ "c=IN IP6 ::\r\n"
"a=ice-ufrag:ufrag_data\r\n"
"a=ice-pwd:pwd_data\r\n"
"a=mid:data_content_name\r\n"
@@ -286,10 +286,10 @@
// draft-ietf-mmusic-sctp-sdp-07
static const char kSdpSctpDataChannelStringWithSctpPort[] =
- "m=application 1 DTLS/SCTP webrtc-datachannel\r\n"
+ "m=application 9 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"
+ "c=IN IP6 ::\r\n"
"a=ice-ufrag:ufrag_data\r\n"
"a=ice-pwd:pwd_data\r\n"
"a=mid:data_content_name\r\n";
@@ -315,11 +315,11 @@
"s=-\r\n"
"t=0 0\r\n"
"a=msid-semantic: WMS\r\n"
- "m=audio 1 RTP/SAVPF 111 103 104\r\n"
- "c=IN IP4 0.0.0.0\r\n"
+ "m=audio 9 RTP/SAVPF 111 103 104\r\n"
+ "c=IN IP6 ::\r\n"
"a=x-google-flag:conference\r\n"
- "m=video 1 RTP/SAVPF 120\r\n"
- "c=IN IP4 0.0.0.0\r\n"
+ "m=video 9 RTP/SAVPF 120\r\n"
+ "c=IN IP6 ::\r\n"
"a=x-google-flag:conference\r\n";
static const char kSdpSessionString[] =
@@ -330,9 +330,9 @@
"a=msid-semantic: WMS local_stream\r\n";
static const char kSdpAudioString[] =
- "m=audio 1 RTP/SAVPF 111\r\n"
- "c=IN IP4 0.0.0.0\r\n"
- "a=rtcp:1 IN IP4 0.0.0.0\r\n"
+ "m=audio 9 RTP/SAVPF 111\r\n"
+ "c=IN IP6 ::\r\n"
+ "a=rtcp:9 IN IP6 ::\r\n"
"a=ice-ufrag:ufrag_voice\r\na=ice-pwd:pwd_voice\r\n"
"a=mid:audio_content_name\r\n"
"a=sendrecv\r\n"
@@ -343,9 +343,9 @@
"a=ssrc:1 label:audio_track_id_1\r\n";
static const char kSdpVideoString[] =
- "m=video 1 RTP/SAVPF 120\r\n"
- "c=IN IP4 0.0.0.0\r\n"
- "a=rtcp:1 IN IP4 0.0.0.0\r\n"
+ "m=video 9 RTP/SAVPF 120\r\n"
+ "c=IN IP6 ::\r\n"
+ "a=rtcp:9 IN IP6 ::\r\n"
"a=ice-ufrag:ufrag_video\r\na=ice-pwd:pwd_video\r\n"
"a=mid:video_content_name\r\n"
"a=sendrecv\r\n"
@@ -1195,7 +1195,7 @@
// description.
"a=msid-semantic: WMS\r\n"
// Pl type 111 preferred.
- "m=audio 1 RTP/SAVPF 111 104 103 102\r\n"
+ "m=audio 9 RTP/SAVPF 111 104 103 102\r\n"
// Pltype 111 listed before 103 and 104 in the map.
"a=rtpmap:111 opus/48000/2\r\n"
// Pltype 103 listed before 104.
@@ -1217,7 +1217,7 @@
os.clear();
os.str("");
// Pl type 100 preferred.
- os << "m=video 1 RTP/SAVPF 99 95\r\n"
+ os << "m=video 9 RTP/SAVPF 99 95\r\n"
<< "a=rtpmap:99 VP8/90000\r\n"
<< "a=rtpmap:95 RTX/90000\r\n"
<< "a=fmtp:95 apt=99;rtx-time=1000\r\n";
@@ -1279,7 +1279,7 @@
// this parser, and will be added to the SDP when serializing a session
// description.
"a=msid-semantic: WMS\r\n"
- "m=audio 1 RTP/SAVPF 111\r\n"
+ "m=audio 9 RTP/SAVPF 111\r\n"
"a=rtpmap:111 opus/48000/2\r\n"
"a=rtcp-fb:111 nack\r\n"
"m=video 3457 RTP/SAVPF 101\r\n"
@@ -1596,7 +1596,7 @@
// TODO(pthatcher): We need to temporarily allow the SDP to control
// this for backwards-compatibility. Once we don't need that any
// more, remove this.
- InjectAfter("m=application 1 RTP/SAVPF 101\r\nc=IN IP4 0.0.0.0\r\n",
+ InjectAfter("m=application 9 RTP/SAVPF 101\r\nc=IN IP6 ::\r\n",
"b=AS:100\r\n",
&expected_sdp);
EXPECT_EQ(expected_sdp, message);
@@ -2308,7 +2308,7 @@
"o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
- "m=audio 1 RTP/SAVPF 104 103\r\n" // Pl type 104 preferred.
+ "m=audio 9 RTP/SAVPF 104 103\r\n" // Pl type 104 preferred.
"a=rtpmap:111 opus/48000/2\r\n" // Pltype 111 listed before 103 and 104
// in the map.
"a=rtpmap:103 ISAC/16000\r\n" // Pltype 103 listed before 104 in the map.
diff --git a/app/webrtc/webrtcsession.h b/app/webrtc/webrtcsession.h
index 86ae435..25e9646 100644
--- a/app/webrtc/webrtcsession.h
+++ b/app/webrtc/webrtcsession.h
@@ -36,7 +36,7 @@
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/statstypes.h"
#include "talk/media/base/mediachannel.h"
-#include "talk/p2p/base/session.h"
+#include "webrtc/p2p/base/session.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/thread.h"
diff --git a/app/webrtc/webrtcsession_unittest.cc b/app/webrtc/webrtcsession_unittest.cc
index d3480a0..e79001a 100644
--- a/app/webrtc/webrtcsession_unittest.cc
+++ b/app/webrtc/webrtcsession_unittest.cc
@@ -40,10 +40,10 @@
#include "talk/media/base/fakevideorenderer.h"
#include "talk/media/base/mediachannel.h"
#include "talk/media/devices/fakedevicemanager.h"
-#include "talk/p2p/base/stunserver.h"
-#include "talk/p2p/base/teststunserver.h"
-#include "talk/p2p/base/testturnserver.h"
-#include "talk/p2p/client/basicportallocator.h"
+#include "webrtc/p2p/base/stunserver.h"
+#include "webrtc/p2p/base/teststunserver.h"
+#include "webrtc/p2p/base/testturnserver.h"
+#include "webrtc/p2p/client/basicportallocator.h"
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/base/fakenetwork.h"
diff --git a/app/webrtc/webrtcsessiondescriptionfactory.h b/app/webrtc/webrtcsessiondescriptionfactory.h
index b870856..31f72fe 100644
--- a/app/webrtc/webrtcsessiondescriptionfactory.h
+++ b/app/webrtc/webrtcsessiondescriptionfactory.h
@@ -29,7 +29,7 @@
#define TALK_APP_WEBRTC_WEBRTCSESSIONDESCRIPTIONFACTORY_H_
#include "talk/app/webrtc/peerconnectioninterface.h"
-#include "talk/p2p/base/transportdescriptionfactory.h"
+#include "webrtc/p2p/base/transportdescriptionfactory.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/base/messagehandler.h"
diff --git a/build/common.gypi b/build/common.gypi
index 7ee4224..d682381 100644
--- a/build/common.gypi
+++ b/build/common.gypi
@@ -39,9 +39,7 @@
'libpeer_target_type%': 'static_library',
'conditions': [
['OS=="android" or OS=="linux"', {
- # TODO(henrike): make sure waterfall bots have $JAVA_HOME configured
- # properly and remove the default value below. See issue 2113.
- 'java_home%': '<!(python -c "import os; dir=os.getenv(\'JAVA_HOME\', \'/usr/lib/jvm/java-6-sun\'); assert os.path.exists(os.path.join(dir, \'include/jni.h\')), \'Point \\$JAVA_HOME or the java_home gyp variable to a directory containing include/jni.h!\'; print dir")',
+ 'java_home%': '<!(python -c "import os; dir=os.getenv(\'JAVA_HOME\', \'/usr/lib/jvm/java-7-openjdk-amd64\'); assert os.path.exists(os.path.join(dir, \'include/jni.h\')), \'Point \\$JAVA_HOME or the java_home gyp variable to a directory containing include/jni.h!\'; print dir")',
}],
],
},
diff --git a/build/isolate.gypi b/build/isolate.gypi
index 6f884cf..df28525 100644
--- a/build/isolate.gypi
+++ b/build/isolate.gypi
@@ -81,18 +81,6 @@
# Files that are known to be involved in this step.
'<(DEPTH)/tools/swarming_client/isolate.py',
'<(DEPTH)/tools/swarming_client/run_isolated.py',
-
- # Disable file tracking by the build driver for now. This means the
- # project must have the proper build-time dependency for their runtime
- # dependency. This improves the runtime of the build driver since it
- # doesn't have to stat() all these files.
- #
- # More importantly, it means that even if a isolate_dependency_tracked
- # file is missing, for example if a file was deleted and the .isolate
- # file was not updated, that won't break the build, especially in the
- # case where foo_tests_run is not built! This should be reenabled once
- # the switch-over to running tests on Swarm is completed.
- #'<@(isolate_dependency_tracked)',
],
'outputs': [
'<(PRODUCT_DIR)/<(RULE_INPUT_ROOT).isolated',
diff --git a/examples/android/AndroidManifest.xml b/examples/android/AndroidManifest.xml
index f898641..30fd46c 100644
--- a/examples/android/AndroidManifest.xml
+++ b/examples/android/AndroidManifest.xml
@@ -4,38 +4,46 @@
android:versionCode="1"
android:versionName="1.0">
- <uses-feature android:name="android.hardware.camera" />
- <uses-feature android:name="android.hardware.camera.autofocus" />
- <uses-feature android:glEsVersion="0x00020000" android:required="true"></uses-feature>
- <uses-sdk android:minSdkVersion="13" android:targetSdkVersion="17" />
+ <uses-feature android:name="android.hardware.camera" />
+ <uses-feature android:name="android.hardware.camera.autofocus" />
+ <uses-feature android:glEsVersion="0x00020000" android:required="true" />
+ <uses-sdk android:minSdkVersion="13" android:targetSdkVersion="19" />
- <uses-permission android:name="android.permission.CAMERA"></uses-permission>
- <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
- <uses-permission android:name="android.permission.RECORD_AUDIO" />
- <uses-permission android:name="android.permission.INTERNET" />
- <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
+ <uses-permission android:name="android.permission.CAMERA" />
+ <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
+ <uses-permission android:name="android.permission.RECORD_AUDIO" />
+ <uses-permission android:name="android.permission.INTERNET" />
+ <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
- <application android:label="@string/app_name"
- android:icon="@drawable/ic_launcher"
- android:debuggable="true"
- android:allowBackup="false">
- <activity android:name="AppRTCDemoActivity"
- android:label="@string/app_name"
- android:screenOrientation="fullUser"
- android:configChanges="orientation|screenSize"
- android:theme="@android:style/Theme.Black.NoTitleBar.Fullscreen">
- <intent-filter>
- <action android:name="android.intent.action.MAIN" />
- <category android:name="android.intent.category.LAUNCHER" />
- </intent-filter>
+ <application android:label="@string/app_name"
+ android:icon="@drawable/ic_launcher"
+ android:allowBackup="false">
- <intent-filter>
- <action android:name="android.intent.action.VIEW" />
- <category android:name="android.intent.category.DEFAULT" />
- <category android:name="android.intent.category.BROWSABLE" />
- <data android:scheme="https" android:host="apprtc.appspot.com" />
- <data android:scheme="http" android:host="apprtc.appspot.com" />
- </intent-filter>
- </activity>
- </application>
+ <activity android:name="ConnectActivity"
+ android:label="@string/app_name">
+ <intent-filter>
+ <action android:name="android.intent.action.MAIN"/>
+ <category android:name="android.intent.category.LAUNCHER"/>
+ </intent-filter>
+
+ <intent-filter>
+ <action android:name="android.intent.action.VIEW"/>
+ <category android:name="android.intent.category.DEFAULT"/>
+ <category android:name="android.intent.category.BROWSABLE"/>
+ <data android:scheme="https" android:host="apprtc.appspot.com"/>
+ <data android:scheme="http" android:host="apprtc.appspot.com"/>
+ </intent-filter>
+ </activity>
+
+ <activity android:name="SettingsActivity"
+ android:label="@string/settings_name">
+ </activity>
+
+ <activity android:name="AppRTCDemoActivity"
+ android:label="@string/app_name"
+ android:screenOrientation="fullUser"
+ android:configChanges="orientation|screenSize"
+ android:theme="@android:style/Theme.Black.NoTitleBar.Fullscreen">
+ </activity>
+ </application>
</manifest>
diff --git a/examples/android/project.properties b/examples/android/project.properties
index 8459f9b..47b7078 100644
--- a/examples/android/project.properties
+++ b/examples/android/project.properties
@@ -11,6 +11,6 @@
#proguard.config=${sdk.dir}/tools/proguard/proguard-android.txt:proguard-project.txt
# Project target.
-target=android-19
+target=android-21
java.compilerargs=-Xlint:all -Werror
diff --git a/examples/android/res/drawable-hdpi/disconnect.png b/examples/android/res/drawable-hdpi/disconnect.png
new file mode 100644
index 0000000..be36174
--- /dev/null
+++ b/examples/android/res/drawable-hdpi/disconnect.png
Binary files differ
diff --git a/examples/android/res/drawable-hdpi/ic_action_full_screen.png b/examples/android/res/drawable-hdpi/ic_action_full_screen.png
new file mode 100644
index 0000000..22f30d3
--- /dev/null
+++ b/examples/android/res/drawable-hdpi/ic_action_full_screen.png
Binary files differ
diff --git a/examples/android/res/drawable-hdpi/ic_action_return_from_full_screen.png b/examples/android/res/drawable-hdpi/ic_action_return_from_full_screen.png
new file mode 100644
index 0000000..d9436e5
--- /dev/null
+++ b/examples/android/res/drawable-hdpi/ic_action_return_from_full_screen.png
Binary files differ
diff --git a/examples/android/res/drawable-hdpi/ic_launcher.png b/examples/android/res/drawable-hdpi/ic_launcher.png
index f3e9d12..f01a31a 100644
--- a/examples/android/res/drawable-hdpi/ic_launcher.png
+++ b/examples/android/res/drawable-hdpi/ic_launcher.png
Binary files differ
diff --git a/examples/android/res/drawable-hdpi/ic_loopback_call.png b/examples/android/res/drawable-hdpi/ic_loopback_call.png
new file mode 100644
index 0000000..3931185
--- /dev/null
+++ b/examples/android/res/drawable-hdpi/ic_loopback_call.png
Binary files differ
diff --git a/examples/android/res/drawable-ldpi/disconnect.png b/examples/android/res/drawable-ldpi/disconnect.png
new file mode 100644
index 0000000..be36174
--- /dev/null
+++ b/examples/android/res/drawable-ldpi/disconnect.png
Binary files differ
diff --git a/examples/android/res/drawable-ldpi/ic_action_full_screen.png b/examples/android/res/drawable-ldpi/ic_action_full_screen.png
new file mode 100644
index 0000000..e4a9ff0
--- /dev/null
+++ b/examples/android/res/drawable-ldpi/ic_action_full_screen.png
Binary files differ
diff --git a/examples/android/res/drawable-ldpi/ic_action_return_from_full_screen.png b/examples/android/res/drawable-ldpi/ic_action_return_from_full_screen.png
new file mode 100644
index 0000000..f5c80f0
--- /dev/null
+++ b/examples/android/res/drawable-ldpi/ic_action_return_from_full_screen.png
Binary files differ
diff --git a/examples/android/res/drawable-ldpi/ic_loopback_call.png b/examples/android/res/drawable-ldpi/ic_loopback_call.png
new file mode 100644
index 0000000..3931185
--- /dev/null
+++ b/examples/android/res/drawable-ldpi/ic_loopback_call.png
Binary files differ
diff --git a/examples/android/res/drawable-mdpi/disconnect.png b/examples/android/res/drawable-mdpi/disconnect.png
new file mode 100644
index 0000000..be36174
--- /dev/null
+++ b/examples/android/res/drawable-mdpi/disconnect.png
Binary files differ
diff --git a/examples/android/res/drawable-mdpi/ic_action_full_screen.png b/examples/android/res/drawable-mdpi/ic_action_full_screen.png
new file mode 100644
index 0000000..e4a9ff0
--- /dev/null
+++ b/examples/android/res/drawable-mdpi/ic_action_full_screen.png
Binary files differ
diff --git a/examples/android/res/drawable-mdpi/ic_action_return_from_full_screen.png b/examples/android/res/drawable-mdpi/ic_action_return_from_full_screen.png
new file mode 100644
index 0000000..f5c80f0
--- /dev/null
+++ b/examples/android/res/drawable-mdpi/ic_action_return_from_full_screen.png
Binary files differ
diff --git a/examples/android/res/drawable-mdpi/ic_launcher.png b/examples/android/res/drawable-mdpi/ic_launcher.png
index 9709a1e..b8b4b0e 100644
--- a/examples/android/res/drawable-mdpi/ic_launcher.png
+++ b/examples/android/res/drawable-mdpi/ic_launcher.png
Binary files differ
diff --git a/examples/android/res/drawable-mdpi/ic_loopback_call.png b/examples/android/res/drawable-mdpi/ic_loopback_call.png
new file mode 100644
index 0000000..3931185
--- /dev/null
+++ b/examples/android/res/drawable-mdpi/ic_loopback_call.png
Binary files differ
diff --git a/examples/android/res/drawable-xhdpi/disconnect.png b/examples/android/res/drawable-xhdpi/disconnect.png
new file mode 100644
index 0000000..be36174
--- /dev/null
+++ b/examples/android/res/drawable-xhdpi/disconnect.png
Binary files differ
diff --git a/examples/android/res/drawable-xhdpi/ic_action_full_screen.png b/examples/android/res/drawable-xhdpi/ic_action_full_screen.png
new file mode 100644
index 0000000..6d90c07
--- /dev/null
+++ b/examples/android/res/drawable-xhdpi/ic_action_full_screen.png
Binary files differ
diff --git a/examples/android/res/drawable-xhdpi/ic_action_return_from_full_screen.png b/examples/android/res/drawable-xhdpi/ic_action_return_from_full_screen.png
new file mode 100644
index 0000000..a773b34
--- /dev/null
+++ b/examples/android/res/drawable-xhdpi/ic_action_return_from_full_screen.png
Binary files differ
diff --git a/examples/android/res/drawable-xhdpi/ic_launcher.png b/examples/android/res/drawable-xhdpi/ic_launcher.png
index db2c4f6..a3cd458 100644
--- a/examples/android/res/drawable-xhdpi/ic_launcher.png
+++ b/examples/android/res/drawable-xhdpi/ic_launcher.png
Binary files differ
diff --git a/examples/android/res/drawable-xhdpi/ic_loopback_call.png b/examples/android/res/drawable-xhdpi/ic_loopback_call.png
new file mode 100644
index 0000000..3931185
--- /dev/null
+++ b/examples/android/res/drawable-xhdpi/ic_loopback_call.png
Binary files differ
diff --git a/examples/android/res/layout/activity_connect.xml b/examples/android/res/layout/activity_connect.xml
new file mode 100644
index 0000000..5ca0f19
--- /dev/null
+++ b/examples/android/res/layout/activity_connect.xml
@@ -0,0 +1,72 @@
+<?xml version="1.0" encoding="utf-8"?>
+<LinearLayout
+ xmlns:android="http://schemas.android.com/apk/res/android"
+ android:layout_width="match_parent"
+ android:layout_height="match_parent"
+ android:orientation="vertical"
+ android:weightSum="1"
+ android:layout_margin="8dp"
+ android:layout_centerHorizontal="true">
+
+ <LinearLayout
+ android:layout_width="match_parent"
+ android:layout_height="wrap_content"
+ android:orientation="horizontal" >
+ <ImageButton
+ android:id="@+id/add_room_button"
+ android:background="@android:drawable/ic_menu_add"
+ android:contentDescription="@string/add_room_description"
+ android:layout_marginRight="20dp"
+ android:layout_width="48dp"
+ android:layout_height="48dp"/>
+ <ImageButton
+ android:id="@+id/remove_room_button"
+ android:background="@android:drawable/ic_delete"
+ android:contentDescription="@string/remove_room_description"
+ android:layout_marginRight="20dp"
+ android:layout_width="48dp"
+ android:layout_height="48dp"/>
+ <ImageButton
+ android:id="@+id/connect_button"
+ android:background="@android:drawable/sym_action_call"
+ android:contentDescription="@string/connect_description"
+ android:layout_marginRight="20dp"
+ android:layout_width="48dp"
+ android:layout_height="48dp"/>
+ <ImageButton
+ android:id="@+id/connect_loopback_button"
+ android:background="@drawable/ic_loopback_call"
+ android:contentDescription="@string/connect_loopback_description"
+ android:layout_width="48dp"
+ android:layout_height="48dp"/>
+ </LinearLayout>
+ <TextView
+ android:id="@+id/room_edittext_description"
+ android:layout_width="fill_parent"
+ android:layout_height="wrap_content"
+ android:layout_margin="5dp"
+ android:text="@string/room_description"/>
+ <EditText
+ android:id="@+id/room_edittext"
+ android:layout_width="match_parent"
+ android:layout_height="wrap_content"
+ android:singleLine="true"
+ android:imeOptions="actionDone"/>
+ <TextView
+ android:id="@+id/room_listview_description"
+ android:layout_width="fill_parent"
+ android:layout_height="wrap_content"
+ android:layout_marginTop="5dp"
+ android:lines="1"
+ android:maxLines="1"
+ android:textAppearance="?android:attr/textAppearanceMedium"
+ android:text="@string/room_names"/>
+ <ListView
+ android:id="@+id/room_listview"
+ android:layout_width="fill_parent"
+ android:layout_height="wrap_content"
+ android:choiceMode="singleChoice"
+ android:listSelector="@android:color/darker_gray"
+ android:drawSelectorOnTop="false" />
+
+</LinearLayout>
\ No newline at end of file
diff --git a/examples/android/res/layout/activity_fullscreen.xml b/examples/android/res/layout/activity_fullscreen.xml
new file mode 100644
index 0000000..fc9ee9e
--- /dev/null
+++ b/examples/android/res/layout/activity_fullscreen.xml
@@ -0,0 +1,33 @@
+<?xml version="1.0" encoding="utf-8"?>
+
+<RelativeLayout
+ xmlns:android="http://schemas.android.com/apk/res/android"
+ xmlns:tools="http://schemas.android.com/tools"
+ android:layout_width="match_parent"
+ android:layout_height="match_parent"
+ android:orientation="vertical">
+
+ <android.opengl.GLSurfaceView
+ android:id="@+id/glview"
+ android:layout_width="match_parent"
+ android:layout_height="match_parent" />
+
+ <TextView
+ android:id="@+id/room_name"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:layout_centerHorizontal="true"
+ android:layout_above="@+id/menubar_fragment"
+ android:textSize="24sp"
+ android:layout_margin="8dp"/>
+ <fragment
+ android:name="org.appspot.apprtc.AppRTCDemoActivity$MenuBarFragment"
+ android:id="@+id/menubar_fragment"
+ android:layout_centerHorizontal="true"
+ android:layout_width="wrap_content"
+ android:layout_height="wrap_content"
+ android:layout_alignParentBottom="true"
+ android:layout_marginBottom="32dp"
+ tools:layout="@layout/fragment_menubar"/>
+
+</RelativeLayout>
diff --git a/examples/android/res/layout/fragment_menubar.xml b/examples/android/res/layout/fragment_menubar.xml
new file mode 100644
index 0000000..77dc819
--- /dev/null
+++ b/examples/android/res/layout/fragment_menubar.xml
@@ -0,0 +1,45 @@
+<?xml version="1.0" encoding="utf-8"?>
+
+<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
+ xmlns:tools="http://schemas.android.com/tools"
+ tools:context="org.appspot.apprtc.AppRTCDemoActivity$MenuBarFragment"
+ android:id="@+id/menubar"
+ android:orientation="horizontal"
+ android:layout_width="wrap_content"
+ android:layout_height="match_parent"
+ android:layout_gravity="center_vertical|center_horizontal">
+
+ <ImageButton
+ android:id="@+id/button_disconnect"
+ android:background="@drawable/disconnect"
+ android:contentDescription="@string/disconnect_call"
+ android:layout_marginRight="16dp"
+ android:layout_width="48dp"
+ android:layout_height="48dp"/>
+
+ <!-- TODO(kjellander): Add audio and video mute buttons. -->
+
+ <ImageButton
+ android:id="@+id/button_switch_camera"
+ android:background="@android:drawable/ic_menu_camera"
+ android:contentDescription="@string/switch_camera"
+ android:layout_marginRight="8dp"
+ android:layout_width="48dp"
+ android:layout_height="48dp"/>
+
+ <ImageButton
+ android:id="@+id/button_toggle_debug"
+ android:background="@android:drawable/ic_menu_info_details"
+ android:contentDescription="@string/disconnect_call"
+ android:layout_marginRight="8dp"
+ android:layout_width="48dp"
+ android:layout_height="48dp"/>
+
+ <ImageButton
+ android:id="@+id/button_scaling_mode"
+ android:background="@drawable/ic_action_return_from_full_screen"
+ android:contentDescription="@string/disconnect_call"
+ android:layout_width="48dp"
+ android:layout_height="48dp"/>
+
+</LinearLayout>
diff --git a/examples/android/res/menu/connect_menu.xml b/examples/android/res/menu/connect_menu.xml
new file mode 100644
index 0000000..d9f9486
--- /dev/null
+++ b/examples/android/res/menu/connect_menu.xml
@@ -0,0 +1,8 @@
+<menu xmlns:android="http://schemas.android.com/apk/res/android" >
+ <item
+ android:id="@+id/action_settings"
+ android:orderInCategory="100"
+ android:icon="@android:drawable/ic_menu_preferences"
+ android:showAsAction="ifRoom"
+ android:title="@string/action_settings"/>
+</menu>
diff --git a/examples/android/res/values/arrays.xml b/examples/android/res/values/arrays.xml
new file mode 100644
index 0000000..3127a85
--- /dev/null
+++ b/examples/android/res/values/arrays.xml
@@ -0,0 +1,22 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+ <string-array name="videoResolutions">
+ <item>Default</item>
+ <item>HD (1280 x 720)</item>
+ <item>VGA (640 x 480)</item>
+ <item>QVGA (320 x 240)</item>
+ </string-array>
+ <string-array name="videoResolutionsValues">
+ <item>Default</item>
+ <item>1280 x 720</item>
+ <item>640 x 480</item>
+ <item>320 x 240</item>
+ </string-array>
+
+ <string-array name="cameraFps">
+ <item>Default</item>
+ <item>30 fps</item>
+ <item>15 fps</item>
+ </string-array>
+
+</resources>
diff --git a/examples/android/res/values/strings.xml b/examples/android/res/values/strings.xml
index bac765a..2a1d64a 100644
--- a/examples/android/res/values/strings.xml
+++ b/examples/android/res/values/strings.xml
@@ -1,4 +1,47 @@
<?xml version="1.0" encoding="utf-8"?>
<resources>
- <string name="app_name">AppRTC</string>
+ <string name="app_name" translatable="no">AppRTC</string>
+ <string name="settings_name" translatable="no">AppRTC Settings</string>
+ <string name="disconnect_call">Disconnect Call</string>
+ <string name="room_names">Room names:</string>
+ <string name="room_description">
+ Please enter a room name. Room names are shared with everyone, so think
+ of something unique and send it to a friend.
+ </string>
+ <string name="connect_text">Connect</string>
+ <string name="invalid_url_title">Invalid URL</string>
+ <string name="invalid_url_text">The URL or room name you entered resulted in an invalid URL: %1$s
+ </string>
+ <string name="channel_error_title">Connection error</string>
+ <string name="connecting_to">Connecting to: %1$s</string>
+ <string name="missing_url">FATAL ERROR: Missing URL to connect to.</string>
+ <string name="ok">OK</string>
+ <string name="switch_camera">Switch front/back camera</string>
+ <string name="action_settings">Settings</string>
+ <string name="add_room_description">Add new room to the list</string>
+ <string name="remove_room_description">Remove room from the list</string>
+ <string name="connect_description">Connect to the room</string>
+ <string name="connect_loopback_description">Loopback connection</string>
+
+ <!-- Settings strings. -->
+ <string name="pref_room_key">room_preference</string>
+ <string name="pref_room_list_key">room_list_preference</string>
+
+ <string name="pref_url_key">url_preference</string>
+ <string name="pref_url_title">Connection URL:</string>
+ <string name="pref_url_summary">AppRTC connection server URL.</string>
+ <string name="pref_url_dlg">Enter AppRTC connection server URL.</string>
+ <string name="pref_url_default">https://apprtc.appspot.com</string>
+
+ <string name="pref_resolution_key">resolution_preference</string>
+ <string name="pref_resolution_title">Video resolution.</string>
+ <string name="pref_resolution_summary">Video resolution.</string>
+ <string name="pref_resolution_dlg">Enter AppRTC local video resolution.</string>
+ <string name="pref_resolution_default">Default</string>
+
+ <string name="pref_fps_key">fps_preference</string>
+ <string name="pref_fps_title">Camera fps.</string>
+ <string name="pref_fps_summary">Camera fps.</string>
+ <string name="pref_fps_dlg">Enter local camera fps.</string>
+ <string name="pref_fps_default">Default</string>
</resources>
diff --git a/examples/android/res/xml/preferences.xml b/examples/android/res/xml/preferences.xml
new file mode 100644
index 0000000..f3f91d8
--- /dev/null
+++ b/examples/android/res/xml/preferences.xml
@@ -0,0 +1,26 @@
+<?xml version="1.0" encoding="utf-8"?>
+<PreferenceScreen xmlns:android="http://schemas.android.com/apk/res/android">
+ <EditTextPreference
+ android:key="@string/pref_url_key"
+ android:title="@string/pref_url_title"
+ android:summary="@string/pref_url_summary"
+ android:defaultValue="@string/pref_url_default"
+ android:inputType="textWebEmailAddress"
+ android:dialogTitle="@string/pref_url_dlg" />
+ <ListPreference
+ android:key="@string/pref_resolution_key"
+ android:title="@string/pref_resolution_title"
+ android:summary="@string/pref_resolution_summary"
+ android:defaultValue="@string/pref_resolution_default"
+ android:dialogTitle="@string/pref_resolution_dlg"
+ android:entries="@array/videoResolutions"
+ android:entryValues="@array/videoResolutionsValues" />
+ <ListPreference
+ android:key="@string/pref_fps_key"
+ android:title="@string/pref_fps_title"
+ android:summary="@string/pref_fps_summary"
+ android:defaultValue="@string/pref_fps_default"
+ android:dialogTitle="@string/pref_fps_dlg"
+ android:entries="@array/cameraFps"
+ android:entryValues="@array/cameraFps" />
+</PreferenceScreen>
\ No newline at end of file
diff --git a/examples/android/src/org/appspot/apprtc/AppRTCClient.java b/examples/android/src/org/appspot/apprtc/AppRTCClient.java
index 4728e51..5c34fca 100644
--- a/examples/android/src/org/appspot/apprtc/AppRTCClient.java
+++ b/examples/android/src/org/appspot/apprtc/AppRTCClient.java
@@ -80,7 +80,9 @@
}
/**
- * Signaling callbacks.
+ * Callback interface for messages delivered on signalling channel.
+ *
+ * Methods are guaranteed to be invoked on the UI thread of |activity|.
*/
public static interface AppRTCSignalingEvents {
/**
diff --git a/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java b/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
index 4718a01..ad0e2d5 100644
--- a/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
+++ b/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
@@ -29,19 +29,24 @@
import android.app.Activity;
import android.app.AlertDialog;
+import android.app.Fragment;
import android.content.DialogInterface;
import android.content.Intent;
-import android.content.res.Configuration;
+import android.content.pm.ActivityInfo;
import android.graphics.Color;
-import android.graphics.Point;
import android.media.AudioManager;
+import android.net.Uri;
+import android.opengl.GLSurfaceView;
import android.os.Bundle;
import android.util.Log;
import android.util.TypedValue;
+import android.view.LayoutInflater;
import android.view.View;
+import android.view.ViewGroup;
import android.view.ViewGroup.LayoutParams;
+import android.view.Window;
import android.view.WindowManager;
-import android.widget.EditText;
+import android.widget.ImageButton;
import android.widget.TextView;
import android.widget.Toast;
@@ -53,6 +58,7 @@
import org.webrtc.StatsReport;
import org.webrtc.VideoRenderer;
import org.webrtc.VideoRendererGui;
+import org.webrtc.VideoRendererGui.ScalingType;
/**
* Main Activity of the AppRTCDemo Android app demonstrating interoperability
@@ -66,43 +72,112 @@
private PeerConnectionClient pc;
private AppRTCClient appRtcClient = new GAERTCClient(this, this);
private AppRTCSignalingParameters appRtcParameters;
- private AppRTCGLView vsv;
+ private View rootView;
+ private View menuBar;
+ private GLSurfaceView videoView;
private VideoRenderer.Callbacks localRender;
private VideoRenderer.Callbacks remoteRender;
+ private ScalingType scalingType;
private Toast logToast;
private final LayoutParams hudLayout =
new LayoutParams(LayoutParams.WRAP_CONTENT, LayoutParams.WRAP_CONTENT);
private TextView hudView;
- // Synchronize on quit[0] to avoid teardown-related crashes.
- private final Boolean[] quit = new Boolean[] { false };
+ private TextView roomName;
+ private ImageButton videoScalingButton;
+ private boolean iceConnected;
@Override
public void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
- Thread.setDefaultUncaughtExceptionHandler(
- new UnhandledExceptionHandler(this));
-
+ // Set window styles for fullscreen-window size. Needs to be done before
+ // adding content.
+ requestWindowFeature(Window.FEATURE_NO_TITLE);
getWindow().addFlags(WindowManager.LayoutParams.FLAG_FULLSCREEN);
getWindow().addFlags(WindowManager.LayoutParams.FLAG_KEEP_SCREEN_ON);
+ getWindow().getDecorView().setSystemUiVisibility(
+ View.SYSTEM_UI_FLAG_HIDE_NAVIGATION |
+ View.SYSTEM_UI_FLAG_FULLSCREEN |
+ View.SYSTEM_UI_FLAG_IMMERSIVE_STICKY);
- Point displaySize = new Point();
- getWindowManager().getDefaultDisplay().getRealSize(displaySize);
+ setContentView(R.layout.activity_fullscreen);
- vsv = new AppRTCGLView(this, displaySize);
- VideoRendererGui.setView(vsv);
- remoteRender = VideoRendererGui.create(0, 0, 100, 100,
- VideoRendererGui.ScalingType.SCALE_ASPECT_FIT);
- localRender = VideoRendererGui.create(0, 0, 100, 100,
- VideoRendererGui.ScalingType.SCALE_ASPECT_FIT);
+ Thread.setDefaultUncaughtExceptionHandler(
+ new UnhandledExceptionHandler(this));
+ iceConnected = false;
- vsv.setOnClickListener(new View.OnClickListener() {
- @Override public void onClick(View v) {
- toggleHUD();
- }
- });
- setContentView(vsv);
- logAndToast("Tap the screen to toggle stats visibility");
+ rootView = findViewById(android.R.id.content);
+ menuBar = findViewById(R.id.menubar_fragment);
+ roomName = (TextView) findViewById(R.id.room_name);
+ videoView = (GLSurfaceView) findViewById(R.id.glview);
+
+ VideoRendererGui.setView(videoView);
+ scalingType = ScalingType.SCALE_ASPECT_FILL;
+ remoteRender = VideoRendererGui.create(0, 0, 100, 100, scalingType);
+ localRender = VideoRendererGui.create(0, 0, 100, 100, scalingType);
+
+ videoView.setOnClickListener(
+ new View.OnClickListener() {
+ @Override
+ public void onClick(View view) {
+ int visibility = menuBar.getVisibility() == View.VISIBLE
+ ? View.INVISIBLE : View.VISIBLE;
+ menuBar.setVisibility(visibility);
+ roomName.setVisibility(visibility);
+ if (visibility == View.VISIBLE) {
+ menuBar.bringToFront();
+ roomName.bringToFront();
+ rootView.invalidate();
+ }
+ }
+ });
+
+ ((ImageButton) findViewById(R.id.button_disconnect)).setOnClickListener(
+ new View.OnClickListener() {
+ @Override
+ public void onClick(View view) {
+ logAndToast("Disconnecting call.");
+ disconnect();
+ }
+ });
+
+ ((ImageButton) findViewById(R.id.button_switch_camera)).setOnClickListener(
+ new View.OnClickListener() {
+ @Override
+ public void onClick(View view) {
+ if (pc != null) {
+ pc.switchCamera();
+ }
+ }
+ });
+
+ ((ImageButton) findViewById(R.id.button_toggle_debug)).setOnClickListener(
+ new View.OnClickListener() {
+ @Override
+ public void onClick(View view) {
+ int visibility = hudView.getVisibility() == View.VISIBLE
+ ? View.INVISIBLE : View.VISIBLE;
+ hudView.setVisibility(visibility);
+ }
+ });
+
+ videoScalingButton = (ImageButton) findViewById(R.id.button_scaling_mode);
+ videoScalingButton.setOnClickListener(
+ new View.OnClickListener() {
+ @Override
+ public void onClick(View view) {
+ if (scalingType == ScalingType.SCALE_ASPECT_FILL) {
+ videoScalingButton.setBackgroundResource(
+ R.drawable.ic_action_full_screen);
+ scalingType = ScalingType.SCALE_ASPECT_FIT;
+ } else {
+ videoScalingButton.setBackgroundResource(
+ R.drawable.ic_action_return_from_full_screen);
+ scalingType = ScalingType.SCALE_ASPECT_FILL;
+ }
+ updateVideoView();
+ }
+ });
hudView = new TextView(this);
hudView.setTextColor(Color.BLACK);
@@ -123,17 +198,44 @@
audioManager.setSpeakerphoneOn(!isWiredHeadsetOn);
final Intent intent = getIntent();
- if ("android.intent.action.VIEW".equals(intent.getAction())) {
- connectToRoom(intent.getData().toString());
- return;
+ Uri url = intent.getData();
+ if (url != null) {
+ String room = url.getQueryParameter("r");
+ String loopback = url.getQueryParameter("debug");
+ if ((room != null && !room.equals("")) ||
+ (loopback != null && loopback.equals("loopback"))) {
+ logAndToast(getString(R.string.connecting_to, url));
+ appRtcClient.connectToRoom(url.toString());
+ if (room != null && !room.equals("")) {
+ roomName.setText(room);
+ } else {
+ roomName.setText("loopback");
+ }
+ } else {
+ logAndToast("Empty or missing room name!");
+ finish();
+ }
+ } else {
+ logAndToast(getString(R.string.missing_url));
+ Log.wtf(TAG, "Didn't get any URL in intent!");
+ finish();
}
- showGetRoomUI();
+ }
+
+ public static class MenuBarFragment extends Fragment {
+ @Override
+ public View onCreateView(
+ LayoutInflater inflater,
+ ViewGroup container,
+ Bundle savedInstanceState) {
+ return inflater.inflate(R.layout.fragment_menubar, container, false);
+ }
}
@Override
public void onPause() {
super.onPause();
- vsv.onPause();
+ videoView.onPause();
if (pc != null) {
pc.stopVideoSource();
}
@@ -142,56 +244,25 @@
@Override
public void onResume() {
super.onResume();
- vsv.onResume();
+ videoView.onResume();
if (pc != null) {
pc.startVideoSource();
}
}
@Override
- public void onConfigurationChanged (Configuration newConfig) {
- Point displaySize = new Point();
- getWindowManager().getDefaultDisplay().getSize(displaySize);
- vsv.updateDisplaySize(displaySize);
- super.onConfigurationChanged(newConfig);
- }
-
- @Override
protected void onDestroy() {
- disconnectAndExit();
+ disconnect();
super.onDestroy();
}
- private void showGetRoomUI() {
- final EditText roomInput = new EditText(this);
- roomInput.setText("https://apprtc.appspot.com/?r=");
- roomInput.setSelection(roomInput.getText().length());
- DialogInterface.OnClickListener listener =
- new DialogInterface.OnClickListener() {
- @Override
- public void onClick(DialogInterface dialog, int which) {
- abortUnless(which == DialogInterface.BUTTON_POSITIVE, "lolwat?");
- dialog.dismiss();
- connectToRoom(roomInput.getText().toString());
- }
- };
- AlertDialog.Builder builder = new AlertDialog.Builder(this);
- builder
- .setMessage("Enter room URL").setView(roomInput)
- .setPositiveButton("Go!", listener).show();
- }
-
- private void connectToRoom(String roomUrl) {
- logAndToast("Connecting to room...");
- appRtcClient.connectToRoom(roomUrl);
- }
-
- // Toggle visibility of the heads-up display.
- private void toggleHUD() {
- if (hudView.getVisibility() == View.VISIBLE) {
- hudView.setVisibility(View.INVISIBLE);
+ private void updateVideoView() {
+ VideoRendererGui.update(remoteRender, 0, 0, 100, 100, scalingType);
+ if (iceConnected) {
+ VideoRendererGui.update(localRender, 70, 70, 28, 28,
+ ScalingType.SCALE_ASPECT_FIT);
} else {
- hudView.setVisibility(View.VISIBLE);
+ VideoRendererGui.update(localRender, 0, 0, 100, 100, scalingType);
}
}
@@ -243,22 +314,29 @@
}
// Disconnect from remote resources, dispose of local resources, and exit.
- private void disconnectAndExit() {
- synchronized (quit[0]) {
- if (quit[0]) {
- return;
- }
- quit[0] = true;
- if (pc != null) {
- pc.close();
- pc = null;
- }
- if (appRtcClient != null) {
- appRtcClient.disconnect();
- appRtcClient = null;
- }
- finish();
+ private void disconnect() {
+ if (appRtcClient != null) {
+ appRtcClient.disconnect();
+ appRtcClient = null;
}
+ if (pc != null) {
+ pc.close();
+ pc = null;
+ }
+ finish();
+ }
+
+ private void disconnectWithMessage(String errorMessage) {
+ new AlertDialog.Builder(this)
+ .setTitle(getText(R.string.channel_error_title))
+ .setMessage(errorMessage)
+ .setCancelable(false)
+ .setNeutralButton(R.string.ok, new DialogInterface.OnClickListener() {
+ public void onClick(DialogInterface dialog, int id) {
+ dialog.cancel();
+ disconnect();
+ }
+ }).create().show();
}
// Poor-man's assert(): die with |msg| unless |condition| is true.
@@ -289,40 +367,42 @@
logAndToast("Creating peer connection...");
pc = new PeerConnectionClient(
this, localRender, remoteRender, appRtcParameters, this);
+ if (pc.isHDVideo()) {
+ setRequestedOrientation(ActivityInfo.SCREEN_ORIENTATION_LANDSCAPE);
+ } else {
+ setRequestedOrientation(ActivityInfo.SCREEN_ORIENTATION_UNSPECIFIED);
+ }
{
- final PeerConnectionClient finalPC = pc;
final Runnable repeatedStatsLogger = new Runnable() {
- public void run() {
- synchronized (quit[0]) {
- if (quit[0]) {
- return;
- }
- final Runnable runnableThis = this;
- if (hudView.getVisibility() == View.INVISIBLE) {
- vsv.postDelayed(runnableThis, 1000);
- return;
- }
- boolean success = finalPC.getStats(new StatsObserver() {
- public void onComplete(final StatsReport[] reports) {
- runOnUiThread(new Runnable() {
- public void run() {
- updateHUD(reports);
- }
- });
- for (StatsReport report : reports) {
- Log.d(TAG, "Stats: " + report.toString());
- }
- vsv.postDelayed(runnableThis, 1000);
+ public void run() {
+ if (pc == null) {
+ return;
+ }
+ final Runnable runnableThis = this;
+ if (hudView.getVisibility() == View.INVISIBLE) {
+ videoView.postDelayed(runnableThis, 1000);
+ return;
+ }
+ boolean success = pc.getStats(new StatsObserver() {
+ public void onComplete(final StatsReport[] reports) {
+ runOnUiThread(new Runnable() {
+ public void run() {
+ updateHUD(reports);
+ }
+ });
+ for (StatsReport report : reports) {
+ Log.d(TAG, "Stats: " + report.toString());
}
- }, null);
- if (!success) {
- throw new RuntimeException("getStats() return false!");
- }
+ videoView.postDelayed(runnableThis, 1000);
+ }
+ }, null);
+ if (!success) {
+ throw new RuntimeException("getStats() return false!");
}
}
- };
- vsv.postDelayed(repeatedStatsLogger, 1000);
+ };
+ videoView.postDelayed(repeatedStatsLogger, 1000);
}
logAndToast("Waiting for remote connection...");
@@ -330,6 +410,9 @@
@Override
public void onChannelOpen() {
+ if (pc == null) {
+ return;
+ }
if (appRtcParameters.initiator) {
logAndToast("Creating OFFER...");
// Create offer. Offer SDP will be sent to answering client in
@@ -340,6 +423,9 @@
@Override
public void onRemoteDescription(final SessionDescription sdp) {
+ if (pc == null) {
+ return;
+ }
logAndToast("Received remote " + sdp.type + " ...");
pc.setRemoteDescription(sdp);
if (!appRtcParameters.initiator) {
@@ -352,19 +438,20 @@
@Override
public void onRemoteIceCandidate(final IceCandidate candidate) {
- pc.addRemoteIceCandidate(candidate);
+ if (pc != null) {
+ pc.addRemoteIceCandidate(candidate);
+ }
}
@Override
public void onChannelClose() {
logAndToast("Remote end hung up; dropping PeerConnection");
- disconnectAndExit();
+ disconnect();
}
@Override
public void onChannelError(int code, String description) {
- logAndToast("Channel error: " + code + ". " + description);
- disconnectAndExit();
+ disconnectWithMessage(description);
}
// -----Implementation of PeerConnectionClient.PeerConnectionEvents.---------
@@ -372,19 +459,35 @@
// All callbacks are invoked from UI thread.
@Override
public void onLocalDescription(final SessionDescription sdp) {
- logAndToast("Sending " + sdp.type + " ...");
- appRtcClient.sendLocalDescription(sdp);
+ if (appRtcClient != null) {
+ logAndToast("Sending " + sdp.type + " ...");
+ appRtcClient.sendLocalDescription(sdp);
+ }
}
@Override
public void onIceCandidate(final IceCandidate candidate) {
- appRtcClient.sendLocalIceCandidate(candidate);
+ if (appRtcClient != null) {
+ appRtcClient.sendLocalIceCandidate(candidate);
+ }
}
@Override
public void onIceConnected() {
logAndToast("ICE connected");
- VideoRendererGui.update(localRender, 70, 70, 28, 28,
- VideoRendererGui.ScalingType.SCALE_ASPECT_FIT);
+ iceConnected = true;
+ updateVideoView();
}
+
+ @Override
+ public void onIceDisconnected() {
+ logAndToast("ICE disconnected");
+ disconnect();
+ }
+
+ @Override
+ public void onPeerConnectionError(String description) {
+ disconnectWithMessage(description);
+ }
+
}
diff --git a/examples/android/src/org/appspot/apprtc/ConnectActivity.java b/examples/android/src/org/appspot/apprtc/ConnectActivity.java
new file mode 100644
index 0000000..8f00c1c
--- /dev/null
+++ b/examples/android/src/org/appspot/apprtc/ConnectActivity.java
@@ -0,0 +1,322 @@
+/*
+ * 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.
+ */
+
+package org.appspot.apprtc;
+
+import android.app.Activity;
+import android.app.AlertDialog;
+import android.content.DialogInterface;
+import android.content.Intent;
+import android.content.SharedPreferences;
+import android.net.Uri;
+import android.os.Bundle;
+import android.preference.PreferenceManager;
+import android.util.Log;
+import android.view.KeyEvent;
+import android.view.Menu;
+import android.view.MenuItem;
+import android.view.View;
+import android.view.View.OnClickListener;
+import android.view.inputmethod.EditorInfo;
+import android.webkit.URLUtil;
+import android.widget.AdapterView;
+import android.widget.ArrayAdapter;
+import android.widget.EditText;
+import android.widget.ImageButton;
+import android.widget.ListView;
+import android.widget.TextView;
+
+import java.util.ArrayList;
+
+import org.json.JSONArray;
+import org.json.JSONException;
+import org.webrtc.MediaCodecVideoEncoder;
+
+
+/**
+ * Handles the initial setup where the user selects which room to join.
+ */
+public class ConnectActivity extends Activity {
+
+ private static final String TAG = "ConnectActivity";
+ private ImageButton addRoomButton;
+ private ImageButton removeRoomButton;
+ private ImageButton connectButton;
+ private ImageButton connectLoopbackButton;
+ private EditText roomEditText;
+ private ListView roomListView;
+ private SharedPreferences sharedPref;
+ private String keyprefUrl;
+ private String keyprefResolution;
+ private String keyprefFps;
+ private String keyprefRoom;
+ private String keyprefRoomList;
+ private ArrayList<String> roomList;
+ private ArrayAdapter<String> adapter;
+
+ @Override
+ public void onCreate(Bundle savedInstanceState) {
+ super.onCreate(savedInstanceState);
+
+ // Get setting keys.
+ PreferenceManager.setDefaultValues(this, R.xml.preferences, false);
+ sharedPref = PreferenceManager.getDefaultSharedPreferences(this);
+ keyprefUrl = getString(R.string.pref_url_key);
+ keyprefResolution = getString(R.string.pref_resolution_key);
+ keyprefFps = getString(R.string.pref_fps_key);
+ keyprefRoom = getString(R.string.pref_room_key);
+ keyprefRoomList = getString(R.string.pref_room_list_key);
+
+ // If an implicit VIEW intent is launching the app, go directly to that URL.
+ final Intent intent = getIntent();
+ if ("android.intent.action.VIEW".equals(intent.getAction())) {
+ connectToRoom(intent.getData().toString());
+ return;
+ }
+
+ setContentView(R.layout.activity_connect);
+
+ roomEditText = (EditText) findViewById(R.id.room_edittext);
+ roomEditText.setOnEditorActionListener(
+ new TextView.OnEditorActionListener() {
+ @Override
+ public boolean onEditorAction(
+ TextView textView, int i, KeyEvent keyEvent) {
+ if (i == EditorInfo.IME_ACTION_DONE) {
+ addRoomButton.performClick();
+ return true;
+ }
+ return false;
+ }
+ });
+ roomEditText.requestFocus();
+
+ roomListView = (ListView) findViewById(R.id.room_listview);
+ roomListView.setChoiceMode(ListView.CHOICE_MODE_SINGLE);
+
+ addRoomButton = (ImageButton) findViewById(R.id.add_room_button);
+ addRoomButton.setOnClickListener(addRoomListener);
+ removeRoomButton = (ImageButton) findViewById(R.id.remove_room_button);
+ removeRoomButton.setOnClickListener(removeRoomListener);
+ connectButton = (ImageButton) findViewById(R.id.connect_button);
+ connectButton.setOnClickListener(connectListener);
+ connectLoopbackButton =
+ (ImageButton) findViewById(R.id.connect_loopback_button);
+ connectLoopbackButton.setOnClickListener(connectListener);
+ }
+
+ @Override
+ public boolean onCreateOptionsMenu(Menu menu) {
+ getMenuInflater().inflate(R.menu.connect_menu, menu);
+ return true;
+ }
+
+ @Override
+ public boolean onOptionsItemSelected(MenuItem item) {
+ // Handle presses on the action bar items.
+ if (item.getItemId() == R.id.action_settings) {
+ Intent intent = new Intent(this, SettingsActivity.class);
+ startActivity(intent);
+ return true;
+ } else {
+ return super.onOptionsItemSelected(item);
+ }
+ }
+
+ @Override
+ public void onPause() {
+ super.onPause();
+ String room = roomEditText.getText().toString();
+ String roomListJson = new JSONArray(roomList).toString();
+ SharedPreferences.Editor editor = sharedPref.edit();
+ editor.putString(keyprefRoom, room);
+ editor.putString(keyprefRoomList, roomListJson);
+ editor.commit();
+ }
+
+ @Override
+ public void onResume() {
+ super.onResume();
+ String room = sharedPref.getString(keyprefRoom, "");
+ roomEditText.setText(room);
+ roomList = new ArrayList<String>();
+ String roomListJson = sharedPref.getString(keyprefRoomList, null);
+ if (roomListJson != null) {
+ try {
+ JSONArray jsonArray = new JSONArray(roomListJson);
+ for (int i = 0; i < jsonArray.length(); i++) {
+ roomList.add(jsonArray.get(i).toString());
+ }
+ } catch (JSONException e) {
+ Log.e(TAG, "Failed to load room list: " + e.toString());
+ }
+ }
+ adapter = new ArrayAdapter<String>(
+ this, android.R.layout.simple_list_item_1, roomList);
+ roomListView.setAdapter(adapter);
+ if (adapter.getCount() > 0) {
+ roomListView.requestFocus();
+ roomListView.setItemChecked(0, true);
+ }
+ }
+
+ private final OnClickListener connectListener = new OnClickListener() {
+ @Override
+ public void onClick(View view) {
+ boolean loopback = false;
+ if (view.getId() == R.id.connect_loopback_button) {
+ loopback = true;
+ }
+ String url = sharedPref.getString(keyprefUrl,
+ getString(R.string.pref_url_default));
+ if (loopback) {
+ url += "/?debug=loopback";
+ } else {
+ String roomName = getSelectedItem();
+ if (roomName == null) {
+ roomName = roomEditText.getText().toString();
+ }
+ url += "/?r=" + roomName;
+ }
+ String parametersResolution = null;
+ String parametersFps = null;
+ // Add video resolution constraints.
+ String resolution = sharedPref.getString(keyprefResolution,
+ getString(R.string.pref_resolution_default));
+ String[] dimensions = resolution.split("[ x]+");
+ if (dimensions.length == 2) {
+ try {
+ int maxWidth = Integer.parseInt(dimensions[0]);
+ int maxHeight = Integer.parseInt(dimensions[1]);
+ if (maxWidth > 0 && maxHeight > 0) {
+ parametersResolution = "minHeight=" + maxHeight + ",maxHeight=" +
+ maxHeight + ",minWidth=" + maxWidth + ",maxWidth=" + maxWidth;
+ }
+ } catch (NumberFormatException e) {
+ Log.e(TAG, "Wrong video resolution setting: " + resolution);
+ }
+ }
+ // Add camera fps constraints.
+ String fps = sharedPref.getString(keyprefFps,
+ getString(R.string.pref_fps_default));
+ String[] fpsValues = fps.split("[ x]+");
+ if (fpsValues.length == 2) {
+ try {
+ int cameraFps = Integer.parseInt(fpsValues[0]);
+ if (cameraFps > 0) {
+ parametersFps = "minFrameRate=" + cameraFps +
+ ",maxFrameRate=" + cameraFps;
+ }
+ } catch (NumberFormatException e) {
+ Log.e(TAG, "Wrong camera fps setting: " + fps);
+ }
+ }
+ // Modify connection URL.
+ if (parametersResolution != null || parametersFps != null) {
+ url += "&video=";
+ if (parametersResolution != null) {
+ url += parametersResolution;
+ if (parametersFps != null) {
+ url += ",";
+ }
+ }
+ if (parametersFps != null) {
+ url += parametersFps;
+ }
+ } else {
+ if (MediaCodecVideoEncoder.isPlatformSupported()) {
+ url += "&hd=true";
+ }
+ }
+ // TODO(kjellander): Add support for custom parameters to the URL.
+ connectToRoom(url);
+ }
+ };
+
+ private void connectToRoom(String roomUrl) {
+ if (validateUrl(roomUrl)) {
+ Uri url = Uri.parse(roomUrl);
+ Intent intent = new Intent(this, AppRTCDemoActivity.class);
+ intent.setData(url);
+ startActivity(intent);
+ }
+ }
+
+ private boolean validateUrl(String url) {
+ if (URLUtil.isHttpsUrl(url) || URLUtil.isHttpUrl(url))
+ return true;
+
+ new AlertDialog.Builder(this)
+ .setTitle(getText(R.string.invalid_url_title))
+ .setMessage(getString(R.string.invalid_url_text, url))
+ .setCancelable(false)
+ .setNeutralButton(R.string.ok, new DialogInterface.OnClickListener() {
+ public void onClick(DialogInterface dialog, int id) {
+ dialog.cancel();
+ }
+ }).create().show();
+ return false;
+ }
+
+ private final OnClickListener addRoomListener = new OnClickListener() {
+ @Override
+ public void onClick(View view) {
+ String newRoom = roomEditText.getText().toString();
+ if (newRoom.length() > 0 && !roomList.contains(newRoom)) {
+ adapter.add(newRoom);
+ adapter.notifyDataSetChanged();
+ }
+ }
+ };
+
+ private final OnClickListener removeRoomListener = new OnClickListener() {
+ @Override
+ public void onClick(View view) {
+ String selectedRoom = getSelectedItem();
+ if (selectedRoom != null) {
+ adapter.remove(selectedRoom);
+ adapter.notifyDataSetChanged();
+ }
+ }
+ };
+
+ private String getSelectedItem() {
+ int position = AdapterView.INVALID_POSITION;
+ if (roomListView.getCheckedItemCount() > 0 && adapter.getCount() > 0) {
+ position = roomListView.getCheckedItemPosition();
+ if (position >= adapter.getCount()) {
+ position = AdapterView.INVALID_POSITION;
+ }
+ }
+ if (position != AdapterView.INVALID_POSITION) {
+ return adapter.getItem(position);
+ } else {
+ return null;
+ }
+ }
+
+}
diff --git a/examples/android/src/org/appspot/apprtc/GAEChannelClient.java b/examples/android/src/org/appspot/apprtc/GAEChannelClient.java
index bcc06ab..5fd0a54 100644
--- a/examples/android/src/org/appspot/apprtc/GAEChannelClient.java
+++ b/examples/android/src/org/appspot/apprtc/GAEChannelClient.java
@@ -51,15 +51,12 @@
/**
* Callback interface for messages delivered on the Google AppEngine channel.
- *
- * Methods are guaranteed to be invoked on the UI thread of |activity| passed
- * to GAEChannelClient's constructor.
*/
public interface GAEMessageHandler {
public void onOpen();
- public void onMessage(String data);
+ public void onMessage(final String data);
public void onClose();
- public void onError(int code, String description);
+ public void onError(final int code, final String description);
}
/** Asynchronously open an AppEngine channel. */
@@ -83,8 +80,7 @@
", desc: " + description);
}
});
- proxyingMessageHandler =
- new ProxyingMessageHandler(activity, handler, token);
+ proxyingMessageHandler = new ProxyingMessageHandler(handler, token);
webView.addJavascriptInterface(
proxyingMessageHandler, "androidMessageHandler");
webView.loadUrl("file:///android_asset/channel.html");
@@ -102,72 +98,52 @@
}
// Helper class for proxying callbacks from the Java<->JS interaction
- // (private, background) thread to the Activity's UI thread.
+ // (private, background) thread.
private static class ProxyingMessageHandler {
- private final Activity activity;
private final GAEMessageHandler handler;
- private final boolean[] disconnected = { false };
+ private boolean disconnected = false;
private final String token;
- public
- ProxyingMessageHandler(Activity activity, GAEMessageHandler handler,
- String token) {
- this.activity = activity;
+ public ProxyingMessageHandler(GAEMessageHandler handler, String token) {
this.handler = handler;
this.token = token;
}
public void disconnect() {
- disconnected[0] = true;
+ disconnected = true;
}
- private boolean disconnected() {
- return disconnected[0];
- }
-
- @JavascriptInterface public String getToken() {
+ @JavascriptInterface
+ public String getToken() {
return token;
}
- @JavascriptInterface public void onOpen() {
- activity.runOnUiThread(new Runnable() {
- public void run() {
- if (!disconnected()) {
- handler.onOpen();
- }
- }
- });
+ @JavascriptInterface
+ public void onOpen() {
+ if (!disconnected) {
+ handler.onOpen();
+ }
}
- @JavascriptInterface public void onMessage(final String data) {
- activity.runOnUiThread(new Runnable() {
- public void run() {
- if (!disconnected()) {
- handler.onMessage(data);
- }
- }
- });
+ @JavascriptInterface
+ public void onMessage(final String data) {
+ if (!disconnected) {
+ handler.onMessage(data);
+ }
}
- @JavascriptInterface public void onClose() {
- activity.runOnUiThread(new Runnable() {
- public void run() {
- if (!disconnected()) {
- handler.onClose();
- }
- }
- });
+ @JavascriptInterface
+ public void onClose() {
+ if (!disconnected) {
+ handler.onClose();
+ }
}
- @JavascriptInterface public void onError(
- final int code, final String description) {
- activity.runOnUiThread(new Runnable() {
- public void run() {
- if (!disconnected()) {
- handler.onError(code, description);
- }
- }
- });
+ @JavascriptInterface
+ public void onError(final int code, final String description) {
+ if (!disconnected) {
+ handler.onError(code, description);
+ }
}
}
}
diff --git a/examples/android/src/org/appspot/apprtc/GAERTCClient.java b/examples/android/src/org/appspot/apprtc/GAERTCClient.java
index 1d1f817..c3d9564 100644
--- a/examples/android/src/org/appspot/apprtc/GAERTCClient.java
+++ b/examples/android/src/org/appspot/apprtc/GAERTCClient.java
@@ -29,7 +29,6 @@
import android.app.Activity;
import android.os.AsyncTask;
import android.util.Log;
-import android.webkit.JavascriptInterface;
import org.json.JSONArray;
import org.json.JSONException;
@@ -41,7 +40,6 @@
import java.io.IOException;
import java.io.InputStream;
-import java.net.HttpURLConnection;
import java.net.URL;
import java.net.URLConnection;
import java.util.LinkedList;
@@ -83,11 +81,6 @@
*/
@Override
public void connectToRoom(String url) {
- while (url.indexOf('?') < 0) {
- // Keep redirecting until we get a room number.
- (new RedirectResolver()).execute(url);
- return; // RedirectResolver above calls us back with the next URL.
- }
(new RoomParameterGetter()).execute(url);
}
@@ -97,6 +90,7 @@
@Override
public void disconnect() {
if (channelClient != null) {
+ Log.d(TAG, "Closing GAE Channel.");
sendMessage("{\"type\": \"bye\"}");
channelClient.close();
channelClient = null;
@@ -151,68 +145,36 @@
}
}
- // Load the given URL and return the value of the Location header of the
- // resulting 302 response. If the result is not a 302, throws.
- private class RedirectResolver extends AsyncTask<String, Void, String> {
- @Override
- protected String doInBackground(String... urls) {
- if (urls.length != 1) {
- throw new RuntimeException("Must be called with a single URL");
- }
- try {
- return followRedirect(urls[0]);
- } catch (IOException e) {
- throw new RuntimeException(e);
- }
- }
-
- @Override
- protected void onPostExecute(String url) {
- connectToRoom(url);
- }
-
- private String followRedirect(String url) throws IOException {
- HttpURLConnection connection = (HttpURLConnection)
- new URL(url).openConnection();
- connection.setInstanceFollowRedirects(false);
- int code = connection.getResponseCode();
- if (code != HttpURLConnection.HTTP_MOVED_TEMP) {
- throw new IOException("Unexpected response: " + code + " for " + url +
- ", with contents: " + drainStream(connection.getInputStream()));
- }
- int n = 0;
- String name, value;
- while ((name = connection.getHeaderFieldKey(n)) != null) {
- value = connection.getHeaderField(n);
- if (name.equals("Location")) {
- return value;
- }
- ++n;
- }
- throw new IOException("Didn't find Location header!");
- }
- }
-
// AsyncTask that converts an AppRTC room URL into the set of signaling
// parameters to use with that room.
private class RoomParameterGetter
extends AsyncTask<String, Void, AppRTCSignalingParameters> {
+ private Exception exception = null;
+
@Override
protected AppRTCSignalingParameters doInBackground(String... urls) {
if (urls.length != 1) {
- throw new RuntimeException("Must be called with a single URL");
+ exception = new RuntimeException("Must be called with a single URL");
+ return null;
}
try {
+ exception = null;
return getParametersForRoomUrl(urls[0]);
} catch (JSONException e) {
- throw new RuntimeException(e);
+ exception = e;
} catch (IOException e) {
- throw new RuntimeException(e);
+ exception = e;
}
+ return null;
}
@Override
protected void onPostExecute(AppRTCSignalingParameters params) {
+ if (exception != null) {
+ Log.e(TAG, "Room connection error: " + exception.toString());
+ events.onChannelError(0, exception.getMessage());
+ return;
+ }
channelClient =
new GAEChannelClient(activity, channelToken, gaeHandler);
synchronized (sendQueue) {
@@ -445,42 +407,67 @@
// Implementation detail: handler for receiving GAE messages and dispatching
// them appropriately.
private class GAEHandler implements GAEChannelClient.GAEMessageHandler {
- @JavascriptInterface public void onOpen() {
- events.onChannelOpen();
- }
+ private boolean channelOpen = false;
- @JavascriptInterface public void onMessage(String msg) {
- Log.d(TAG, "RECEIVE: " + msg);
- try {
- JSONObject json = new JSONObject(msg);
- String type = (String) json.get("type");
- if (type.equals("candidate")) {
- IceCandidate candidate = new IceCandidate(
- (String) json.get("id"),
- json.getInt("label"),
- (String) json.get("candidate"));
- events.onRemoteIceCandidate(candidate);
- } else if (type.equals("answer") || type.equals("offer")) {
- SessionDescription sdp = new SessionDescription(
- SessionDescription.Type.fromCanonicalForm(type),
- (String)json.get("sdp"));
- events.onRemoteDescription(sdp);
- } else if (type.equals("bye")) {
- events.onChannelClose();
- } else {
- throw new RuntimeException("Unexpected message: " + msg);
+ public void onOpen() {
+ activity.runOnUiThread(new Runnable() {
+ public void run() {
+ events.onChannelOpen();
+ channelOpen = true;
}
- } catch (JSONException e) {
- throw new RuntimeException(e);
- }
+ });
}
- @JavascriptInterface public void onClose() {
- events.onChannelClose();
+ public void onMessage(final String msg) {
+ Log.d(TAG, "RECEIVE: " + msg);
+ activity.runOnUiThread(new Runnable() {
+ public void run() {
+ if (!channelOpen) {
+ return;
+ }
+ try {
+ JSONObject json = new JSONObject(msg);
+ String type = (String) json.get("type");
+ if (type.equals("candidate")) {
+ IceCandidate candidate = new IceCandidate(
+ (String) json.get("id"),
+ json.getInt("label"),
+ (String) json.get("candidate"));
+ events.onRemoteIceCandidate(candidate);
+ } else if (type.equals("answer") || type.equals("offer")) {
+ SessionDescription sdp = new SessionDescription(
+ SessionDescription.Type.fromCanonicalForm(type),
+ (String)json.get("sdp"));
+ events.onRemoteDescription(sdp);
+ } else if (type.equals("bye")) {
+ events.onChannelClose();
+ } else {
+ events.onChannelError(1, "Unexpected channel message: " + msg);
+ }
+ } catch (JSONException e) {
+ events.onChannelError(1, "Channel message JSON parsing error: " +
+ e.toString());
+ }
+ }
+ });
}
- @JavascriptInterface public void onError(int code, String description) {
- events.onChannelError(code, description);
+ public void onClose() {
+ activity.runOnUiThread(new Runnable() {
+ public void run() {
+ events.onChannelClose();
+ channelOpen = false;
+ }
+ });
+ }
+
+ public void onError(final int code, final String description) {
+ activity.runOnUiThread(new Runnable() {
+ public void run() {
+ events.onChannelError(code, description);
+ channelOpen = false;
+ }
+ });
}
}
diff --git a/examples/android/src/org/appspot/apprtc/PeerConnectionClient.java b/examples/android/src/org/appspot/apprtc/PeerConnectionClient.java
index 0376000..9c917bb 100644
--- a/examples/android/src/org/appspot/apprtc/PeerConnectionClient.java
+++ b/examples/android/src/org/appspot/apprtc/PeerConnectionClient.java
@@ -37,6 +37,7 @@
import org.webrtc.MediaStream;
import org.webrtc.MediaStreamTrack;
import org.webrtc.PeerConnection;
+import org.webrtc.MediaConstraints.KeyValuePair;
import org.webrtc.PeerConnection.IceConnectionState;
import org.webrtc.PeerConnectionFactory;
import org.webrtc.SdpObserver;
@@ -60,13 +61,17 @@
private boolean videoSourceStopped;
private final PCObserver pcObserver = new PCObserver();
private final SDPObserver sdpObserver = new SDPObserver();
+ private final VideoRenderer.Callbacks localRender;
private final VideoRenderer.Callbacks remoteRender;
private LinkedList<IceCandidate> queuedRemoteCandidates =
new LinkedList<IceCandidate>();
private MediaConstraints sdpMediaConstraints;
+ private MediaConstraints videoConstraints;
private PeerConnectionEvents events;
private boolean isInitiator;
+ private boolean useFrontFacingCamera = true;
private SessionDescription localSdp = null; // either offer or answer SDP
+ private MediaStream videoMediaStream = null;
public PeerConnectionClient(
Activity activity,
@@ -75,6 +80,7 @@
AppRTCSignalingParameters appRtcParameters,
PeerConnectionEvents events) {
this.activity = activity;
+ this.localRender = localRender;
this.remoteRender = remoteRender;
this.events = events;
isInitiator = appRtcParameters.initiator;
@@ -84,9 +90,9 @@
"OfferToReceiveAudio", "true"));
sdpMediaConstraints.mandatory.add(new MediaConstraints.KeyValuePair(
"OfferToReceiveVideo", "true"));
+ videoConstraints = appRtcParameters.videoConstraints;
factory = new PeerConnectionFactory();
-
MediaConstraints pcConstraints = appRtcParameters.pcConstraints;
pcConstraints.optional.add(
new MediaConstraints.KeyValuePair("RtpDataChannels", "true"));
@@ -101,23 +107,47 @@
// EnumSet.of(Logging.TraceLevel.TRACE_ALL),
// Logging.Severity.LS_SENSITIVE);
- Log.d(TAG, "Creating local video source");
- MediaStream lMS = factory.createLocalMediaStream("ARDAMS");
- if (appRtcParameters.videoConstraints != null) {
- VideoCapturer capturer = getVideoCapturer();
- videoSource = factory.createVideoSource(
- capturer, appRtcParameters.videoConstraints);
- VideoTrack videoTrack =
- factory.createVideoTrack("ARDAMSv0", videoSource);
- videoTrack.addRenderer(new VideoRenderer(localRender));
- lMS.addTrack(videoTrack);
+ if (videoConstraints != null) {
+ videoMediaStream = factory.createLocalMediaStream("ARDAMSVideo");
+ videoMediaStream.addTrack(createVideoTrack(useFrontFacingCamera));
+ pc.addStream(videoMediaStream);
}
+
if (appRtcParameters.audioConstraints != null) {
+ MediaStream lMS = factory.createLocalMediaStream("ARDAMSAudio");
lMS.addTrack(factory.createAudioTrack(
"ARDAMSa0",
factory.createAudioSource(appRtcParameters.audioConstraints)));
+ pc.addStream(lMS);
}
- pc.addStream(lMS, new MediaConstraints());
+ }
+
+ public boolean isHDVideo() {
+ if (videoConstraints == null) {
+ return false;
+ }
+ int minWidth = 0;
+ int minHeight = 0;
+ for (KeyValuePair keyValuePair : videoConstraints.mandatory) {
+ if (keyValuePair.getKey().equals("minWidth")) {
+ try {
+ minWidth = Integer.parseInt(keyValuePair.getValue());
+ } catch (NumberFormatException e) {
+ Log.e(TAG, "Can not parse video width from video constraints");
+ }
+ } else if (keyValuePair.getKey().equals("minHeight")) {
+ try {
+ minHeight = Integer.parseInt(keyValuePair.getValue());
+ } catch (NumberFormatException e) {
+ Log.e(TAG, "Can not parse video height from video constraints");
+ }
+ }
+ }
+ if (minWidth * minHeight >= 1280 * 720) {
+ return true;
+ } else {
+ return false;
+ }
}
public boolean getStats(StatsObserver observer, MediaStreamTrack track) {
@@ -125,33 +155,58 @@
}
public void createOffer() {
- isInitiator = true;
- pc.createOffer(sdpObserver, sdpMediaConstraints);
+ activity.runOnUiThread(new Runnable() {
+ public void run() {
+ if (pc != null) {
+ isInitiator = true;
+ pc.createOffer(sdpObserver, sdpMediaConstraints);
+ }
+ }
+ });
}
public void createAnswer() {
- isInitiator = false;
- pc.createAnswer(sdpObserver, sdpMediaConstraints);
+ activity.runOnUiThread(new Runnable() {
+ public void run() {
+ if (pc != null) {
+ isInitiator = false;
+ pc.createAnswer(sdpObserver, sdpMediaConstraints);
+ }
+ }
+ });
}
- public void addRemoteIceCandidate(IceCandidate candidate) {
- if (queuedRemoteCandidates != null) {
- queuedRemoteCandidates.add(candidate);
- } else {
- pc.addIceCandidate(candidate);
- }
+ public void addRemoteIceCandidate(final IceCandidate candidate) {
+ activity.runOnUiThread(new Runnable() {
+ public void run() {
+ if (pc != null) {
+ if (queuedRemoteCandidates != null) {
+ queuedRemoteCandidates.add(candidate);
+ } else {
+ pc.addIceCandidate(candidate);
+ }
+ }
+ }
+ });
}
- public void setRemoteDescription(SessionDescription sdp) {
- SessionDescription sdpISAC = new SessionDescription(
- sdp.type, preferISAC(sdp.description));
- Log.d(TAG, "Set remote SDP");
- pc.setRemoteDescription(sdpObserver, sdpISAC);
+ public void setRemoteDescription(final SessionDescription sdp) {
+ activity.runOnUiThread(new Runnable() {
+ public void run() {
+ if (pc != null) {
+ SessionDescription sdpISAC = new SessionDescription(
+ sdp.type, preferISAC(sdp.description));
+ Log.d(TAG, "Set remote SDP");
+ pc.setRemoteDescription(sdpObserver, sdpISAC);
+ }
+ }
+ });
}
public void stopVideoSource() {
if (videoSource != null) {
+ Log.d(TAG, "Stop video source.");
videoSource.stop();
videoSourceStopped = true;
}
@@ -159,24 +214,30 @@
public void startVideoSource() {
if (videoSource != null && videoSourceStopped) {
+ Log.d(TAG, "Restart video source.");
videoSource.restart();
videoSourceStopped = false;
}
}
public void close() {
- if (pc != null) {
- pc.dispose();
- pc = null;
- }
- if (videoSource != null) {
- videoSource.dispose();
- videoSource = null;
- }
- if (factory != null) {
- factory.dispose();
- factory = null;
- }
+ activity.runOnUiThread(new Runnable() {
+ public void run() {
+ Log.d(TAG, "Closing peer connection.");
+ if (pc != null) {
+ pc.dispose();
+ pc = null;
+ }
+ if (videoSource != null) {
+ videoSource.dispose();
+ videoSource = null;
+ }
+ if (factory != null) {
+ factory.dispose();
+ factory = null;
+ }
+ }
+ });
}
/**
@@ -198,15 +259,38 @@
* CONNECTED).
*/
public void onIceConnected();
+
+ /**
+ * Callback fired once connection is closed (IceConnectionState is
+ * DISCONNECTED).
+ */
+ public void onIceDisconnected();
+
+ /**
+ * Callback fired once peer connection error happened.
+ */
+ public void onPeerConnectionError(String description);
+ }
+
+ private void reportError(final String errorMessage) {
+ activity.runOnUiThread(new Runnable() {
+ public void run() {
+ events.onPeerConnectionError(errorMessage);
+ }
+ });
}
// Cycle through likely device names for the camera and return the first
// capturer that works, or crash if none do.
- private VideoCapturer getVideoCapturer() {
+ private VideoCapturer getVideoCapturer(boolean useFrontFacing) {
String[] cameraFacing = { "front", "back" };
- int[] cameraIndex = { 0, 1 };
- int[] cameraOrientation = { 0, 90, 180, 270 };
+ if (!useFrontFacing) {
+ cameraFacing[0] = "back";
+ cameraFacing[1] = "front";
+ }
for (String facing : cameraFacing) {
+ int[] cameraIndex = { 0, 1 };
+ int[] cameraOrientation = { 0, 90, 180, 270 };
for (int index : cameraIndex) {
for (int orientation : cameraOrientation) {
String name = "Camera " + index + ", Facing " + facing +
@@ -219,13 +303,30 @@
}
}
}
- throw new RuntimeException("Failed to open capturer");
+ reportError("Failed to open capturer");
+ return null;
+ }
+
+ private VideoTrack createVideoTrack(boolean frontFacing) {
+ VideoCapturer capturer = getVideoCapturer(frontFacing);
+ if (videoSource != null) {
+ videoSource.stop();
+ videoSource.dispose();
+ }
+
+ videoSource = factory.createVideoSource(
+ capturer, videoConstraints);
+ String trackExtension = frontFacing ? "frontFacing" : "backFacing";
+ VideoTrack videoTrack =
+ factory.createVideoTrack("ARDAMSv0" + trackExtension, videoSource);
+ videoTrack.addRenderer(new VideoRenderer(localRender));
+ return videoTrack;
}
// Poor-man's assert(): die with |msg| unless |condition| is true.
- private static void abortUnless(boolean condition, String msg) {
+ private void abortUnless(boolean condition, String msg) {
if (!condition) {
- throw new RuntimeException(msg);
+ reportError(msg);
}
}
@@ -285,24 +386,58 @@
queuedRemoteCandidates = null;
}
+ public void switchCamera() {
+ if (videoConstraints == null)
+ return; // No video is sent.
+
+ if (pc.signalingState() != PeerConnection.SignalingState.STABLE) {
+ Log.e(TAG, "Switching camera during negotiation is not handled.");
+ return;
+ }
+
+ pc.removeStream(videoMediaStream);
+ VideoTrack currentTrack = videoMediaStream.videoTracks.get(0);
+ videoMediaStream.removeTrack(currentTrack);
+
+ String trackId = currentTrack.id();
+ // On Android, there can only be one camera open at the time and we
+ // need to release our implicit references to the videoSource before the
+ // PeerConnectionFactory is released. Since createVideoTrack creates a new
+ // videoSource and frees the old one, we need to release the track here.
+ currentTrack.dispose();
+
+ useFrontFacingCamera = !useFrontFacingCamera;
+ VideoTrack newTrack = createVideoTrack(useFrontFacingCamera);
+ videoMediaStream.addTrack(newTrack);
+ pc.addStream(videoMediaStream);
+
+ SessionDescription remoteDesc = pc.getRemoteDescription();
+ if (localSdp == null || remoteDesc == null) {
+ Log.d(TAG, "Switching camera before the negotiation started.");
+ return;
+ }
+
+ localSdp = new SessionDescription(localSdp.type,
+ localSdp.description.replaceAll(trackId, newTrack.id()));
+
+ if (isInitiator) {
+ pc.setLocalDescription(new SwitchCameraSdbObserver(), localSdp);
+ pc.setRemoteDescription(new SwitchCameraSdbObserver(), remoteDesc);
+ } else {
+ pc.setRemoteDescription(new SwitchCameraSdbObserver(), remoteDesc);
+ pc.setLocalDescription(new SwitchCameraSdbObserver(), localSdp);
+ }
+ }
+
// Implementation detail: observe ICE & stream changes and react accordingly.
private class PCObserver implements PeerConnection.Observer {
@Override
public void onIceCandidate(final IceCandidate candidate){
activity.runOnUiThread(new Runnable() {
- public void run() {
- events.onIceCandidate(candidate);
- }
- });
- }
-
- @Override
- public void onError(){
- activity.runOnUiThread(new Runnable() {
- public void run() {
- throw new RuntimeException("PeerConnection error!");
- }
- });
+ public void run() {
+ events.onIceCandidate(candidate);
+ }
+ });
}
@Override
@@ -321,47 +456,50 @@
events.onIceConnected();
}
});
+ } else if (newState == IceConnectionState.DISCONNECTED) {
+ activity.runOnUiThread(new Runnable() {
+ public void run() {
+ events.onIceDisconnected();
+ }
+ });
+ } else if (newState == IceConnectionState.FAILED) {
+ reportError("ICE connection failed.");
}
}
@Override
public void onIceGatheringChange(
- PeerConnection.IceGatheringState newState) {
+ PeerConnection.IceGatheringState newState) {
}
@Override
public void onAddStream(final MediaStream stream){
activity.runOnUiThread(new Runnable() {
- public void run() {
- abortUnless(stream.audioTracks.size() <= 1 &&
- stream.videoTracks.size() <= 1,
- "Weird-looking stream: " + stream);
- if (stream.videoTracks.size() == 1) {
- stream.videoTracks.get(0).addRenderer(
- new VideoRenderer(remoteRender));
- }
+ public void run() {
+ abortUnless(stream.audioTracks.size() <= 1 &&
+ stream.videoTracks.size() <= 1,
+ "Weird-looking stream: " + stream);
+ if (stream.videoTracks.size() == 1) {
+ stream.videoTracks.get(0).addRenderer(
+ new VideoRenderer(remoteRender));
}
- });
+ }
+ });
}
@Override
public void onRemoveStream(final MediaStream stream){
activity.runOnUiThread(new Runnable() {
- public void run() {
- stream.videoTracks.get(0).dispose();
- }
- });
+ public void run() {
+ stream.videoTracks.get(0).dispose();
+ }
+ });
}
@Override
public void onDataChannel(final DataChannel dc) {
- activity.runOnUiThread(new Runnable() {
- public void run() {
- throw new RuntimeException(
- "AppRTC doesn't use data channels, but got: " + dc.label() +
- " anyway!");
- }
- });
+ reportError("AppRTC doesn't use data channels, but got: " + dc.label() +
+ " anyway!");
}
@Override
@@ -381,65 +519,82 @@
origSdp.type, preferISAC(origSdp.description));
localSdp = sdp;
activity.runOnUiThread(new Runnable() {
- public void run() {
+ public void run() {
+ if (pc != null) {
Log.d(TAG, "Set local SDP from " + sdp.type);
pc.setLocalDescription(sdpObserver, sdp);
}
- });
+ }
+ });
}
@Override
public void onSetSuccess() {
activity.runOnUiThread(new Runnable() {
- public void run() {
- if (isInitiator) {
- // For offering peer connection we first create offer and set
- // local SDP, then after receiving answer set remote SDP.
- if (pc.getRemoteDescription() == null) {
- // We've just set our local SDP so time to send it.
- Log.d(TAG, "Local SDP set succesfully");
- events.onLocalDescription(localSdp);
- } else {
- // We've just set remote description,
- // so drain remote ICE candidates.
- Log.d(TAG, "Remote SDP set succesfully");
- drainRemoteCandidates();
- }
+ public void run() {
+ if (pc == null) {
+ return;
+ }
+ if (isInitiator) {
+ // For offering peer connection we first create offer and set
+ // local SDP, then after receiving answer set remote SDP.
+ if (pc.getRemoteDescription() == null) {
+ // We've just set our local SDP so time to send it.
+ Log.d(TAG, "Local SDP set succesfully");
+ events.onLocalDescription(localSdp);
} else {
- // For answering peer connection we set remote SDP and then
- // create answer and set local SDP.
- if (pc.getLocalDescription() != null) {
- // We've just set our local SDP so time to send it and drain
- // remote ICE candidates.
- Log.d(TAG, "Local SDP set succesfully");
- events.onLocalDescription(localSdp);
- drainRemoteCandidates();
- } else {
- // We've just set remote SDP - do nothing for now -
- // answer will be created soon.
- Log.d(TAG, "Remote SDP set succesfully");
- }
+ // We've just set remote description,
+ // so drain remote ICE candidates.
+ Log.d(TAG, "Remote SDP set succesfully");
+ drainRemoteCandidates();
+ }
+ } else {
+ // For answering peer connection we set remote SDP and then
+ // create answer and set local SDP.
+ if (pc.getLocalDescription() != null) {
+ // We've just set our local SDP so time to send it and drain
+ // remote ICE candidates.
+ Log.d(TAG, "Local SDP set succesfully");
+ events.onLocalDescription(localSdp);
+ drainRemoteCandidates();
+ } else {
+ // We've just set remote SDP - do nothing for now -
+ // answer will be created soon.
+ Log.d(TAG, "Remote SDP set succesfully");
}
}
- });
+ }
+ });
}
@Override
public void onCreateFailure(final String error) {
- activity.runOnUiThread(new Runnable() {
- public void run() {
- throw new RuntimeException("createSDP error: " + error);
- }
- });
+ reportError("createSDP error: " + error);
}
@Override
public void onSetFailure(final String error) {
- activity.runOnUiThread(new Runnable() {
- public void run() {
- throw new RuntimeException("setSDP error: " + error);
- }
- });
+ reportError("setSDP error: " + error);
+ }
+ }
+
+ private class SwitchCameraSdbObserver implements SdpObserver {
+ @Override
+ public void onCreateSuccess(SessionDescription sdp) {
+ }
+
+ @Override
+ public void onSetSuccess() {
+ Log.d(TAG, "Camera switch SDP set succesfully");
+ }
+
+ @Override
+ public void onCreateFailure(final String error) {
+ }
+
+ @Override
+ public void onSetFailure(final String error) {
+ reportError("setSDP error while switching camera: " + error);
}
}
}
diff --git a/examples/android/src/org/appspot/apprtc/SettingsActivity.java b/examples/android/src/org/appspot/apprtc/SettingsActivity.java
new file mode 100644
index 0000000..2354ceb
--- /dev/null
+++ b/examples/android/src/org/appspot/apprtc/SettingsActivity.java
@@ -0,0 +1,92 @@
+/*
+ * 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.
+ */
+
+package org.appspot.apprtc;
+
+import android.app.Activity;
+import android.content.SharedPreferences;
+import android.content.SharedPreferences.OnSharedPreferenceChangeListener;
+import android.os.Bundle;
+import android.preference.Preference;
+
+public class SettingsActivity extends Activity
+ implements OnSharedPreferenceChangeListener{
+ private SettingsFragment settingsFragment;
+ private String keyprefUrl;
+ private String keyprefResolution;
+ private String keyprefFps;
+
+ @Override
+ protected void onCreate(Bundle savedInstanceState) {
+ super.onCreate(savedInstanceState);
+ keyprefUrl = getString(R.string.pref_url_key);
+ keyprefResolution = getString(R.string.pref_resolution_key);
+ keyprefFps = getString(R.string.pref_fps_key);
+
+ // Display the fragment as the main content.
+ settingsFragment = new SettingsFragment();
+ getFragmentManager().beginTransaction()
+ .replace(android.R.id.content, settingsFragment)
+ .commit();
+ }
+
+ @Override
+ protected void onResume() {
+ super.onResume();
+ // Set summary to be the user-description for the selected value
+ SharedPreferences sharedPreferences =
+ settingsFragment.getPreferenceScreen().getSharedPreferences();
+ sharedPreferences.registerOnSharedPreferenceChangeListener(this);
+ updateSummary(sharedPreferences, keyprefUrl);
+ updateSummary(sharedPreferences, keyprefResolution);
+ updateSummary(sharedPreferences, keyprefFps);
+ }
+
+ @Override
+ protected void onPause() {
+ super.onPause();
+ SharedPreferences sharedPreferences =
+ settingsFragment.getPreferenceScreen().getSharedPreferences();
+ sharedPreferences.unregisterOnSharedPreferenceChangeListener(this);
+ }
+
+ @Override
+ public void onSharedPreferenceChanged(SharedPreferences sharedPreferences,
+ String key) {
+ if (key.equals(keyprefUrl) || key.equals(keyprefResolution) ||
+ key.equals(keyprefFps)) {
+ updateSummary(sharedPreferences, key);
+ }
+ }
+
+ private void updateSummary(SharedPreferences sharedPreferences, String key) {
+ Preference updatedPref = settingsFragment.findPreference(key);
+ // Set summary to be the user-description for the selected value
+ updatedPref.setSummary(sharedPreferences.getString(key, ""));
+ }
+
+}
diff --git a/examples/android/src/org/appspot/apprtc/AppRTCGLView.java b/examples/android/src/org/appspot/apprtc/SettingsFragment.java
similarity index 65%
rename from examples/android/src/org/appspot/apprtc/AppRTCGLView.java
rename to examples/android/src/org/appspot/apprtc/SettingsFragment.java
index e622ab9..ec4b704 100644
--- a/examples/android/src/org/appspot/apprtc/AppRTCGLView.java
+++ b/examples/android/src/org/appspot/apprtc/SettingsFragment.java
@@ -27,32 +27,15 @@
package org.appspot.apprtc;
-import android.content.Context;
-import android.graphics.Point;
-import android.opengl.GLSurfaceView;
+import android.os.Bundle;
+import android.preference.PreferenceFragment;
-public class AppRTCGLView extends GLSurfaceView {
- private Point screenDimensions;
-
- public AppRTCGLView(Context c, Point screenDimensions) {
- super(c);
- this.screenDimensions = screenDimensions;
- }
-
- public void updateDisplaySize(Point screenDimensions) {
- this.screenDimensions = screenDimensions;
- }
+public class SettingsFragment extends PreferenceFragment {
@Override
- protected void onMeasure(int unusedX, int unusedY) {
- // Go big or go home!
- setMeasuredDimension(screenDimensions.x, screenDimensions.y);
- }
-
- @Override
- protected void onAttachedToWindow() {
- super.onAttachedToWindow();
- setSystemUiVisibility(SYSTEM_UI_FLAG_HIDE_NAVIGATION |
- SYSTEM_UI_FLAG_FULLSCREEN | SYSTEM_UI_FLAG_IMMERSIVE_STICKY);
+ public void onCreate(Bundle savedInstanceState) {
+ super.onCreate(savedInstanceState);
+ // Load the preferences from an XML resource
+ addPreferencesFromResource(R.xml.preferences);
}
}
diff --git a/examples/call/call_main.cc b/examples/call/call_main.cc
index cb309dc..638f6f9 100644
--- a/examples/call/call_main.cc
+++ b/examples/call/call_main.cc
@@ -41,13 +41,13 @@
#include "talk/examples/call/callclient.h"
#include "talk/examples/call/console.h"
#include "talk/examples/call/mediaenginefactory.h"
-#include "talk/p2p/base/constants.h"
+#include "webrtc/p2p/base/constants.h"
#include "talk/session/media/mediasessionclient.h"
#include "talk/session/media/srtpfilter.h"
-#include "talk/xmpp/xmppauth.h"
-#include "talk/xmpp/xmppclientsettings.h"
-#include "talk/xmpp/xmpppump.h"
-#include "talk/xmpp/xmppsocket.h"
+#include "webrtc/libjingle/xmpp/xmppauth.h"
+#include "webrtc/libjingle/xmpp/xmppclientsettings.h"
+#include "webrtc/libjingle/xmpp/xmpppump.h"
+#include "webrtc/libjingle/xmpp/xmppsocket.h"
#include "webrtc/base/pathutils.h"
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/stream.h"
diff --git a/examples/call/callclient.cc b/examples/call/callclient.cc
index b31e24a..866133e 100644
--- a/examples/call/callclient.cc
+++ b/examples/call/callclient.cc
@@ -53,17 +53,17 @@
#include "talk/media/base/videorenderer.h"
#include "talk/media/devices/devicemanager.h"
#include "talk/media/devices/videorendererfactory.h"
-#include "talk/p2p/base/sessionmanager.h"
-#include "talk/p2p/client/basicportallocator.h"
-#include "talk/p2p/client/sessionmanagertask.h"
+#include "webrtc/p2p/base/sessionmanager.h"
+#include "webrtc/p2p/client/basicportallocator.h"
+#include "webrtc/p2p/client/sessionmanagertask.h"
#include "talk/session/media/mediamessages.h"
#include "talk/session/media/mediasessionclient.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/hangoutpubsubclient.h"
-#include "talk/xmpp/mucroomconfigtask.h"
-#include "talk/xmpp/mucroomlookuptask.h"
-#include "talk/xmpp/pingtask.h"
-#include "talk/xmpp/presenceouttask.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/hangoutpubsubclient.h"
+#include "webrtc/libjingle/xmpp/mucroomconfigtask.h"
+#include "webrtc/libjingle/xmpp/mucroomlookuptask.h"
+#include "webrtc/libjingle/xmpp/pingtask.h"
+#include "webrtc/libjingle/xmpp/presenceouttask.h"
namespace {
diff --git a/examples/call/callclient.h b/examples/call/callclient.h
index 50c584b..075dc6e 100644
--- a/examples/call/callclient.h
+++ b/examples/call/callclient.h
@@ -34,12 +34,12 @@
#include "talk/examples/call/console.h"
#include "talk/media/base/mediachannel.h"
-#include "talk/p2p/base/session.h"
+#include "webrtc/p2p/base/session.h"
#include "talk/session/media/mediamessages.h"
#include "talk/session/media/mediasessionclient.h"
-#include "talk/xmpp/hangoutpubsubclient.h"
-#include "talk/xmpp/presencestatus.h"
-#include "talk/xmpp/xmppclient.h"
+#include "webrtc/libjingle/xmpp/hangoutpubsubclient.h"
+#include "webrtc/libjingle/xmpp/presencestatus.h"
+#include "webrtc/libjingle/xmpp/xmppclient.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/sslidentity.h"
diff --git a/examples/call/callclient_unittest.cc b/examples/call/callclient_unittest.cc
index d11580e..256853a 100644
--- a/examples/call/callclient_unittest.cc
+++ b/examples/call/callclient_unittest.cc
@@ -30,7 +30,7 @@
#include "talk/examples/call/callclient.h"
#include "talk/media/base/filemediaengine.h"
#include "talk/media/base/mediaengine.h"
-#include "talk/xmpp/xmppthread.h"
+#include "webrtc/libjingle/xmpp/xmppthread.h"
#include "webrtc/base/gunit.h"
TEST(CallClientTest, CreateCallClientWithDefaultMediaEngine) {
diff --git a/examples/call/friendinvitesendtask.cc b/examples/call/friendinvitesendtask.cc
index dae1006..e2b8dde 100644
--- a/examples/call/friendinvitesendtask.cc
+++ b/examples/call/friendinvitesendtask.cc
@@ -26,7 +26,7 @@
*/
#include "talk/examples/call/friendinvitesendtask.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
namespace buzz {
diff --git a/examples/call/friendinvitesendtask.h b/examples/call/friendinvitesendtask.h
index 625f077..2928452 100644
--- a/examples/call/friendinvitesendtask.h
+++ b/examples/call/friendinvitesendtask.h
@@ -28,8 +28,8 @@
#ifndef _FRIENDINVITESENDTASK_H_
#define _FRIENDINVITESENDTASK_H_
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
namespace buzz {
diff --git a/examples/call/muc.h b/examples/call/muc.h
index 0e937ca..9be192f 100644
--- a/examples/call/muc.h
+++ b/examples/call/muc.h
@@ -29,8 +29,8 @@
#define _MUC_H_
#include <map>
-#include "talk/xmpp/jid.h"
-#include "talk/xmpp/presencestatus.h"
+#include "webrtc/libjingle/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/presencestatus.h"
namespace buzz {
diff --git a/examples/call/mucinviterecvtask.cc b/examples/call/mucinviterecvtask.cc
index e32456f..32de8fe 100644
--- a/examples/call/mucinviterecvtask.cc
+++ b/examples/call/mucinviterecvtask.cc
@@ -26,7 +26,7 @@
*/
#include "talk/examples/call/mucinviterecvtask.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
namespace buzz {
diff --git a/examples/call/mucinviterecvtask.h b/examples/call/mucinviterecvtask.h
index ddfd6be..0466c94 100644
--- a/examples/call/mucinviterecvtask.h
+++ b/examples/call/mucinviterecvtask.h
@@ -30,8 +30,8 @@
#include <vector>
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
#include "webrtc/base/sigslot.h"
namespace buzz {
diff --git a/examples/call/mucinvitesendtask.cc b/examples/call/mucinvitesendtask.cc
index d648fef..2299b78 100644
--- a/examples/call/mucinvitesendtask.cc
+++ b/examples/call/mucinvitesendtask.cc
@@ -26,8 +26,8 @@
*/
#include "talk/examples/call/mucinvitesendtask.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/xmppclient.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/xmppclient.h"
namespace buzz {
diff --git a/examples/call/mucinvitesendtask.h b/examples/call/mucinvitesendtask.h
index 3ae74c1..673efb0 100644
--- a/examples/call/mucinvitesendtask.h
+++ b/examples/call/mucinvitesendtask.h
@@ -29,8 +29,8 @@
#define _MUCINVITESENDTASK_H_
#include "talk/examples/call/muc.h"
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
namespace buzz {
diff --git a/examples/call/presencepushtask.cc b/examples/call/presencepushtask.cc
index b598eb6..8d8dca8 100644
--- a/examples/call/presencepushtask.cc
+++ b/examples/call/presencepushtask.cc
@@ -28,7 +28,7 @@
#include "talk/examples/call/presencepushtask.h"
#include "talk/examples/call/muc.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/stringencode.h"
diff --git a/examples/call/presencepushtask.h b/examples/call/presencepushtask.h
index c13a4f5..e3d3268 100644
--- a/examples/call/presencepushtask.h
+++ b/examples/call/presencepushtask.h
@@ -31,9 +31,9 @@
#include <vector>
#include "talk/examples/call/callclient.h"
-#include "talk/xmpp/presencestatus.h"
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/presencestatus.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
#include "webrtc/base/sigslot.h"
namespace buzz {
diff --git a/examples/login/login_main.cc b/examples/login/login_main.cc
index bfe12af..581c986 100644
--- a/examples/login/login_main.cc
+++ b/examples/login/login_main.cc
@@ -29,10 +29,10 @@
#include <iostream>
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/xmppclientsettings.h"
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmppthread.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/xmppclientsettings.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmppthread.h"
#include "webrtc/base/thread.h"
int main(int argc, char **argv) {
diff --git a/examples/objc/AppRTCDemo/APPRTCConnectionManager.m b/examples/objc/AppRTCDemo/APPRTCConnectionManager.m
index b411a62..9a39528 100644
--- a/examples/objc/AppRTCDemo/APPRTCConnectionManager.m
+++ b/examples/objc/AppRTCDemo/APPRTCConnectionManager.m
@@ -170,7 +170,7 @@
#endif
[lms addAudioTrack:[self.peerConnectionFactory audioTrackWithID:@"ARDAMSa0"]];
- [self.peerConnection addStream:lms constraints:constraints];
+ [self.peerConnection addStream:lms];
[self.logger logMessage:@"onICEServers - added local stream."];
}
@@ -243,16 +243,6 @@
#pragma mark - RTCPeerConnectionDelegate
-- (void)peerConnectionOnError:(RTCPeerConnection*)peerConnection {
- dispatch_async(dispatch_get_main_queue(), ^{
- NSString* message = @"PeerConnection error";
- NSLog(@"%@", message);
- NSAssert(NO, @"PeerConnection failed.");
- [self.delegate connectionManager:self
- didErrorWithMessage:message];
- });
-}
-
- (void)peerConnection:(RTCPeerConnection*)peerConnection
signalingStateChanged:(RTCSignalingState)stateChanged {
dispatch_async(dispatch_get_main_queue(), ^{
diff --git a/examples/peerconnection/client/conductor.cc b/examples/peerconnection/client/conductor.cc
index f49aee6..e81f7fc 100644
--- a/examples/peerconnection/client/conductor.cc
+++ b/examples/peerconnection/client/conductor.cc
@@ -137,11 +137,6 @@
// PeerConnectionObserver implementation.
//
-void Conductor::OnError() {
- LOG(LS_ERROR) << __FUNCTION__;
- main_wnd_->QueueUIThreadCallback(PEER_CONNECTION_ERROR, NULL);
-}
-
// Called when a remote stream is added
void Conductor::OnAddStream(webrtc::MediaStreamInterface* stream) {
LOG(INFO) << __FUNCTION__ << " " << stream->label();
@@ -373,7 +368,7 @@
stream->AddTrack(audio_track);
stream->AddTrack(video_track);
- if (!peer_connection_->AddStream(stream, NULL)) {
+ if (!peer_connection_->AddStream(stream)) {
LOG(LS_ERROR) << "Adding stream to PeerConnection failed";
}
typedef std::pair<std::string,
@@ -440,10 +435,6 @@
break;
}
- case PEER_CONNECTION_ERROR:
- main_wnd_->MessageBox("Error", "an unknown error occurred", true);
- break;
-
case NEW_STREAM_ADDED: {
webrtc::MediaStreamInterface* stream =
reinterpret_cast<webrtc::MediaStreamInterface*>(
diff --git a/examples/peerconnection/client/conductor.h b/examples/peerconnection/client/conductor.h
index 0aff531..3ef5253 100644
--- a/examples/peerconnection/client/conductor.h
+++ b/examples/peerconnection/client/conductor.h
@@ -58,7 +58,6 @@
MEDIA_CHANNELS_INITIALIZED = 1,
PEER_CONNECTION_CLOSED,
SEND_MESSAGE_TO_PEER,
- PEER_CONNECTION_ERROR,
NEW_STREAM_ADDED,
STREAM_REMOVED,
};
@@ -80,11 +79,11 @@
//
// PeerConnectionObserver implementation.
//
- virtual void OnError();
virtual void OnStateChange(
webrtc::PeerConnectionObserver::StateType state_changed) {}
virtual void OnAddStream(webrtc::MediaStreamInterface* stream);
virtual void OnRemoveStream(webrtc::MediaStreamInterface* stream);
+ virtual void OnDataChannel(webrtc::DataChannelInterface* channel) {}
virtual void OnRenegotiationNeeded() {}
virtual void OnIceChange() {}
virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate);
diff --git a/examples/relayserver/relayserver_main.cc b/examples/relayserver/relayserver_main.cc
index 1a4ab78..5a8bec3 100644
--- a/examples/relayserver/relayserver_main.cc
+++ b/examples/relayserver/relayserver_main.cc
@@ -27,7 +27,7 @@
#include <iostream> // NOLINT
-#include "talk/p2p/base/relayserver.h"
+#include "webrtc/p2p/base/relayserver.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread.h"
diff --git a/examples/stunserver/stunserver_main.cc b/examples/stunserver/stunserver_main.cc
index f800f31..3cbed91 100644
--- a/examples/stunserver/stunserver_main.cc
+++ b/examples/stunserver/stunserver_main.cc
@@ -31,7 +31,7 @@
#include <iostream>
-#include "talk/p2p/base/stunserver.h"
+#include "webrtc/p2p/base/stunserver.h"
#include "webrtc/base/thread.h"
using namespace cricket;
diff --git a/examples/turnserver/turnserver_main.cc b/examples/turnserver/turnserver_main.cc
index 607b8cf..692b8a2 100644
--- a/examples/turnserver/turnserver_main.cc
+++ b/examples/turnserver/turnserver_main.cc
@@ -27,8 +27,8 @@
#include <iostream> // NOLINT
-#include "talk/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/base/turnserver.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/turnserver.h"
#include "webrtc/base/asyncudpsocket.h"
#include "webrtc/base/optionsfile.h"
#include "webrtc/base/stringencode.h"
diff --git a/libjingle.gyp b/libjingle.gyp
index 335a788..803eaa3 100755
--- a/libjingle.gyp
+++ b/libjingle.gyp
@@ -299,7 +299,7 @@
'targets': [
{
'target_name': 'libjingle',
- 'type': 'static_library',
+ 'type': 'none',
'dependencies': [
'<(DEPTH)/third_party/expat/expat.gyp:expat',
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
@@ -309,81 +309,6 @@
'<(DEPTH)/third_party/expat/expat.gyp:expat',
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
],
- 'sources': [
- 'xmpp/asyncsocket.h',
- 'xmpp/chatroommodule.h',
- 'xmpp/chatroommoduleimpl.cc',
- 'xmpp/constants.cc',
- 'xmpp/constants.h',
- 'xmpp/discoitemsquerytask.cc',
- 'xmpp/discoitemsquerytask.h',
- 'xmpp/hangoutpubsubclient.cc',
- 'xmpp/hangoutpubsubclient.h',
- 'xmpp/iqtask.cc',
- 'xmpp/iqtask.h',
- 'xmpp/jid.cc',
- 'xmpp/jid.h',
- 'xmpp/module.h',
- 'xmpp/moduleimpl.cc',
- 'xmpp/moduleimpl.h',
- 'xmpp/mucroomconfigtask.cc',
- 'xmpp/mucroomconfigtask.h',
- 'xmpp/mucroomdiscoverytask.cc',
- 'xmpp/mucroomdiscoverytask.h',
- 'xmpp/mucroomlookuptask.cc',
- 'xmpp/mucroomlookuptask.h',
- 'xmpp/mucroomuniquehangoutidtask.cc',
- 'xmpp/mucroomuniquehangoutidtask.h',
- 'xmpp/pingtask.cc',
- 'xmpp/pingtask.h',
- 'xmpp/plainsaslhandler.h',
- 'xmpp/presenceouttask.cc',
- 'xmpp/presenceouttask.h',
- 'xmpp/presencereceivetask.cc',
- 'xmpp/presencereceivetask.h',
- 'xmpp/presencestatus.cc',
- 'xmpp/presencestatus.h',
- 'xmpp/prexmppauth.h',
- 'xmpp/pubsub_task.cc',
- 'xmpp/pubsub_task.h',
- 'xmpp/pubsubclient.cc',
- 'xmpp/pubsubclient.h',
- 'xmpp/pubsubstateclient.cc',
- 'xmpp/pubsubstateclient.h',
- 'xmpp/pubsubtasks.cc',
- 'xmpp/pubsubtasks.h',
- 'xmpp/receivetask.cc',
- 'xmpp/receivetask.h',
- 'xmpp/rostermodule.h',
- 'xmpp/rostermoduleimpl.cc',
- 'xmpp/rostermoduleimpl.h',
- 'xmpp/saslcookiemechanism.h',
- 'xmpp/saslhandler.h',
- 'xmpp/saslmechanism.cc',
- 'xmpp/saslmechanism.h',
- 'xmpp/saslplainmechanism.h',
- 'xmpp/xmppauth.cc',
- 'xmpp/xmppauth.h',
- 'xmpp/xmppclient.cc',
- 'xmpp/xmppclient.h',
- 'xmpp/xmppclientsettings.h',
- 'xmpp/xmppengine.h',
- 'xmpp/xmppengineimpl.cc',
- 'xmpp/xmppengineimpl.h',
- 'xmpp/xmppengineimpl_iq.cc',
- 'xmpp/xmpplogintask.cc',
- 'xmpp/xmpplogintask.h',
- 'xmpp/xmpppump.cc',
- 'xmpp/xmpppump.h',
- 'xmpp/xmppsocket.cc',
- 'xmpp/xmppsocket.h',
- 'xmpp/xmppstanzaparser.cc',
- 'xmpp/xmppstanzaparser.h',
- 'xmpp/xmpptask.cc',
- 'xmpp/xmpptask.h',
- 'xmpp/xmppthread.cc',
- 'xmpp/xmppthread.h',
- ],
}, # target libjingle
{
'target_name': 'libjingle_media',
@@ -401,8 +326,10 @@
'<(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)/system_wrappers/source/system_wrappers.gyp:system_wrappers_default',
'<(webrtc_root)/libjingle/xmllite/xmllite.gyp:rtc_xmllite',
+ '<(webrtc_root)/libjingle/xmpp/xmpp.gyp:rtc_xmpp',
+ '<(webrtc_root)/p2p/p2p.gyp:rtc_p2p',
'libjingle',
],
'direct_dependent_settings': {
@@ -628,90 +555,6 @@
],
},
'sources': [
- 'p2p/base/asyncstuntcpsocket.cc',
- 'p2p/base/asyncstuntcpsocket.h',
- 'p2p/base/basicpacketsocketfactory.cc',
- 'p2p/base/basicpacketsocketfactory.h',
- 'p2p/base/candidate.h',
- 'p2p/base/common.h',
- 'p2p/base/constants.cc',
- 'p2p/base/constants.h',
- 'p2p/base/dtlstransportchannel.cc',
- 'p2p/base/dtlstransportchannel.h',
- 'p2p/base/p2ptransport.cc',
- 'p2p/base/p2ptransport.h',
- 'p2p/base/p2ptransportchannel.cc',
- 'p2p/base/p2ptransportchannel.h',
- 'p2p/base/packetsocketfactory.h',
- 'p2p/base/parsing.cc',
- 'p2p/base/parsing.h',
- 'p2p/base/port.cc',
- 'p2p/base/port.h',
- 'p2p/base/portallocator.cc',
- 'p2p/base/portallocator.h',
- 'p2p/base/portallocatorsessionproxy.cc',
- 'p2p/base/portallocatorsessionproxy.h',
- 'p2p/base/portinterface.h',
- 'p2p/base/portproxy.cc',
- 'p2p/base/portproxy.h',
- 'p2p/base/pseudotcp.cc',
- 'p2p/base/pseudotcp.h',
- 'p2p/base/rawtransport.cc',
- 'p2p/base/rawtransport.h',
- 'p2p/base/rawtransportchannel.cc',
- 'p2p/base/rawtransportchannel.h',
- 'p2p/base/relayport.cc',
- 'p2p/base/relayport.h',
- 'p2p/base/relayserver.cc',
- 'p2p/base/relayserver.h',
- 'p2p/base/session.cc',
- 'p2p/base/session.h',
- 'p2p/base/sessionclient.h',
- 'p2p/base/sessiondescription.cc',
- 'p2p/base/sessiondescription.h',
- 'p2p/base/sessionid.h',
- 'p2p/base/sessionmanager.cc',
- 'p2p/base/sessionmanager.h',
- 'p2p/base/sessionmessages.cc',
- 'p2p/base/sessionmessages.h',
- 'p2p/base/stun.cc',
- 'p2p/base/stun.h',
- 'p2p/base/stunport.cc',
- 'p2p/base/stunport.h',
- 'p2p/base/stunrequest.cc',
- 'p2p/base/stunrequest.h',
- 'p2p/base/stunserver.cc',
- 'p2p/base/stunserver.h',
- 'p2p/base/tcpport.cc',
- 'p2p/base/tcpport.h',
- 'p2p/base/transport.cc',
- 'p2p/base/transport.h',
- 'p2p/base/transportchannel.cc',
- 'p2p/base/transportchannel.h',
- 'p2p/base/transportchannelimpl.h',
- 'p2p/base/transportchannelproxy.cc',
- 'p2p/base/transportchannelproxy.h',
- 'p2p/base/transportdescription.cc',
- 'p2p/base/transportdescription.h',
- 'p2p/base/transportdescriptionfactory.cc',
- 'p2p/base/transportdescriptionfactory.h',
- 'p2p/base/transportinfo.h',
- 'p2p/base/turnport.cc',
- 'p2p/base/turnport.h',
- 'p2p/base/turnserver.cc',
- 'p2p/base/turnserver.h',
- 'p2p/base/udpport.h',
- 'p2p/client/autoportallocator.h',
- 'p2p/client/basicportallocator.cc',
- 'p2p/client/basicportallocator.h',
- 'p2p/client/connectivitychecker.cc',
- 'p2p/client/connectivitychecker.h',
- 'p2p/client/httpportallocator.cc',
- 'p2p/client/httpportallocator.h',
- 'p2p/client/sessionmanagertask.h',
- 'p2p/client/sessionsendtask.h',
- 'p2p/client/socketmonitor.cc',
- 'p2p/client/socketmonitor.h',
'session/tunnel/pseudotcpchannel.cc',
'session/tunnel/pseudotcpchannel.h',
'session/tunnel/tunnelsessionclient.cc',
diff --git a/libjingle_examples.gyp b/libjingle_examples.gyp
index 085da39..d0f1747 100755
--- a/libjingle_examples.gyp
+++ b/libjingle_examples.gyp
@@ -39,8 +39,8 @@
'libjingle.gyp:libjingle_p2p',
],
'sources': [
- 'xmpp/jingleinfotask.cc',
- 'xmpp/jingleinfotask.h',
+ '<(webrtc_root)/libjingle/xmpp/jingleinfotask.cc',
+ '<(webrtc_root)/libjingle/xmpp/jingleinfotask.h',
],
}, # target libjingle_xmpphelp
{
@@ -317,17 +317,41 @@
'examples/android/build.xml',
'examples/android/jni/Android.mk',
'examples/android/project.properties',
+ 'examples/android/res/drawable-hdpi/disconnect.png',
+ 'examples/android/res/drawable-hdpi/ic_action_full_screen.png',
+ 'examples/android/res/drawable-hdpi/ic_action_return_from_full_screen.png',
+ 'examples/android/res/drawable-hdpi/ic_loopback_call.png',
'examples/android/res/drawable-hdpi/ic_launcher.png',
+ 'examples/android/res/drawable-ldpi/disconnect.png',
+ 'examples/android/res/drawable-ldpi/ic_action_full_screen.png',
+ 'examples/android/res/drawable-ldpi/ic_action_return_from_full_screen.png',
+ 'examples/android/res/drawable-ldpi/ic_loopback_call.png',
'examples/android/res/drawable-ldpi/ic_launcher.png',
+ 'examples/android/res/drawable-mdpi/disconnect.png',
+ 'examples/android/res/drawable-mdpi/ic_action_full_screen.png',
+ 'examples/android/res/drawable-mdpi/ic_action_return_from_full_screen.png',
+ 'examples/android/res/drawable-mdpi/ic_loopback_call.png',
'examples/android/res/drawable-mdpi/ic_launcher.png',
+ 'examples/android/res/drawable-xhdpi/disconnect.png',
+ 'examples/android/res/drawable-xhdpi/ic_action_full_screen.png',
+ 'examples/android/res/drawable-xhdpi/ic_action_return_from_full_screen.png',
+ 'examples/android/res/drawable-xhdpi/ic_loopback_call.png',
'examples/android/res/drawable-xhdpi/ic_launcher.png',
+ 'examples/android/res/layout/activity_connect.xml',
+ 'examples/android/res/layout/activity_fullscreen.xml',
+ 'examples/android/res/layout/fragment_menubar.xml',
+ 'examples/android/res/menu/connect_menu.xml',
+ 'examples/android/res/values/arrays.xml',
'examples/android/res/values/strings.xml',
+ 'examples/android/res/xml/preferences.xml',
'examples/android/src/org/appspot/apprtc/AppRTCClient.java',
'examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java',
- 'examples/android/src/org/appspot/apprtc/AppRTCGLView.java',
+ 'examples/android/src/org/appspot/apprtc/ConnectActivity.java',
'examples/android/src/org/appspot/apprtc/GAEChannelClient.java',
'examples/android/src/org/appspot/apprtc/GAERTCClient.java',
'examples/android/src/org/appspot/apprtc/PeerConnectionClient.java',
+ 'examples/android/src/org/appspot/apprtc/SettingsActivity.java',
+ 'examples/android/src/org/appspot/apprtc/SettingsFragment.java',
'examples/android/src/org/appspot/apprtc/UnhandledExceptionHandler.java',
],
'outputs': [
diff --git a/libjingle_media_unittest.isolate b/libjingle_media_unittest.isolate
index 666478b..970a166 100644
--- a/libjingle_media_unittest.isolate
+++ b/libjingle_media_unittest.isolate
@@ -31,12 +31,9 @@
'command': [
'<(PRODUCT_DIR)/libjingle_media_unittest<(EXECUTABLE_SUFFIX)',
],
- 'isolate_dependency_tracked': [
- 'media/testdata/captured-320x240-2s-48.frames',
+ 'files': [
'<(PRODUCT_DIR)/libjingle_media_unittest<(EXECUTABLE_SUFFIX)',
- ],
- 'isolate_dependency_untracked': [
- '<(DEPTH)/tools/swarming_client/',
+ 'media/testdata/captured-320x240-2s-48.frames',
],
},
}],
diff --git a/libjingle_p2p_unittest.isolate b/libjingle_p2p_unittest.isolate
index 9ff0d77..d9f0423 100644
--- a/libjingle_p2p_unittest.isolate
+++ b/libjingle_p2p_unittest.isolate
@@ -31,12 +31,9 @@
'command': [
'<(PRODUCT_DIR)/libjingle_p2p_unittest<(EXECUTABLE_SUFFIX)',
],
- 'isolate_dependency_tracked': [
+ 'files': [
'<(PRODUCT_DIR)/libjingle_p2p_unittest<(EXECUTABLE_SUFFIX)',
],
- 'isolate_dependency_untracked': [
- '<(DEPTH)/tools/swarming_client/',
- ],
},
}],
],
diff --git a/libjingle_peerconnection_unittest.isolate b/libjingle_peerconnection_unittest.isolate
index df07d4a..4bc8c43 100644
--- a/libjingle_peerconnection_unittest.isolate
+++ b/libjingle_peerconnection_unittest.isolate
@@ -31,12 +31,9 @@
'command': [
'<(PRODUCT_DIR)/libjingle_peerconnection_unittest<(EXECUTABLE_SUFFIX)',
],
- 'isolate_dependency_tracked': [
+ 'files': [
'<(PRODUCT_DIR)/libjingle_peerconnection_unittest<(EXECUTABLE_SUFFIX)',
],
- 'isolate_dependency_untracked': [
- '<(DEPTH)/tools/swarming_client/',
- ],
},
}],
],
diff --git a/libjingle_sound_unittest.isolate b/libjingle_sound_unittest.isolate
index 728e810..b534200 100644
--- a/libjingle_sound_unittest.isolate
+++ b/libjingle_sound_unittest.isolate
@@ -31,12 +31,9 @@
'command': [
'<(PRODUCT_DIR)/libjingle_sound_unittest<(EXECUTABLE_SUFFIX)',
],
- 'isolate_dependency_tracked': [
+ 'files': [
'<(PRODUCT_DIR)/libjingle_sound_unittest<(EXECUTABLE_SUFFIX)',
],
- 'isolate_dependency_untracked': [
- '<(DEPTH)/tools/swarming_client/',
- ],
},
}],
],
diff --git a/libjingle_tests.gyp b/libjingle_tests.gyp
index 44adec3..6a25c0f 100755
--- a/libjingle_tests.gyp
+++ b/libjingle_tests.gyp
@@ -81,21 +81,7 @@
'libjingle_unittest_main',
],
'sources': [
- 'xmpp/fakexmppclient.h',
- 'xmpp/hangoutpubsubclient_unittest.cc',
- 'xmpp/jid_unittest.cc',
- 'xmpp/mucroomconfigtask_unittest.cc',
- 'xmpp/mucroomdiscoverytask_unittest.cc',
- 'xmpp/mucroomlookuptask_unittest.cc',
- 'xmpp/mucroomuniquehangoutidtask_unittest.cc',
- 'xmpp/pingtask_unittest.cc',
- 'xmpp/pubsubclient_unittest.cc',
- 'xmpp/pubsubtasks_unittest.cc',
- 'xmpp/util_unittest.cc',
- 'xmpp/util_unittest.h',
- 'xmpp/xmppengine_unittest.cc',
- 'xmpp/xmpplogintask_unittest.cc',
- 'xmpp/xmppstanzaparser_unittest.cc',
+ '<(DEPTH)/webrtc/test/testsupport/always_passing_unittest.cc',
], # sources
}, # target libjingle_unittest
{
@@ -190,28 +176,6 @@
'<(DEPTH)/third_party/libsrtp/srtp',
],
'sources': [
- 'p2p/base/dtlstransportchannel_unittest.cc',
- 'p2p/base/fakesession.h',
- 'p2p/base/p2ptransportchannel_unittest.cc',
- 'p2p/base/port_unittest.cc',
- 'p2p/base/portallocatorsessionproxy_unittest.cc',
- 'p2p/base/pseudotcp_unittest.cc',
- 'p2p/base/relayport_unittest.cc',
- 'p2p/base/relayserver_unittest.cc',
- 'p2p/base/session_unittest.cc',
- 'p2p/base/stun_unittest.cc',
- 'p2p/base/stunport_unittest.cc',
- 'p2p/base/stunrequest_unittest.cc',
- 'p2p/base/stunserver_unittest.cc',
- 'p2p/base/testrelayserver.h',
- 'p2p/base/teststunserver.h',
- 'p2p/base/testturnserver.h',
- 'p2p/base/transport_unittest.cc',
- 'p2p/base/transportdescriptionfactory_unittest.cc',
- 'p2p/base/turnport_unittest.cc',
- 'p2p/client/connectivitychecker_unittest.cc',
- 'p2p/client/fakeportallocator.h',
- 'p2p/client/portallocator_unittest.cc',
'session/media/bundlefilter_unittest.cc',
'session/media/channel_unittest.cc',
'session/media/channelmanager_unittest.cc',
@@ -438,7 +402,6 @@
],
'includes': [
'build/isolate.gypi',
- 'libjingle_media_unittest.isolate',
],
'sources': [
'libjingle_media_unittest.isolate',
@@ -452,7 +415,6 @@
],
'includes': [
'build/isolate.gypi',
- 'libjingle_p2p_unittest.isolate',
],
'sources': [
'libjingle_p2p_unittest.isolate',
@@ -466,26 +428,11 @@
],
'includes': [
'build/isolate.gypi',
- 'libjingle_peerconnection_unittest.isolate',
],
'sources': [
'libjingle_peerconnection_unittest.isolate',
],
},
- {
- 'target_name': 'libjingle_unittest_run',
- 'type': 'none',
- 'dependencies': [
- 'libjingle_unittest',
- ],
- 'includes': [
- 'build/isolate.gypi',
- 'libjingle_unittest.isolate',
- ],
- 'sources': [
- 'libjingle_unittest.isolate',
- ],
- },
],
}],
],
diff --git a/libjingle_unittest.isolate b/libjingle_unittest.isolate
index 0507f6a..9313beb 100644
--- a/libjingle_unittest.isolate
+++ b/libjingle_unittest.isolate
@@ -31,12 +31,9 @@
'command': [
'<(PRODUCT_DIR)/libjingle_unittest<(EXECUTABLE_SUFFIX)',
],
- 'isolate_dependency_tracked': [
+ 'files': [
'<(PRODUCT_DIR)/libjingle_unittest<(EXECUTABLE_SUFFIX)',
],
- 'isolate_dependency_untracked': [
- '<(DEPTH)/tools/swarming_client/',
- ],
},
}],
],
diff --git a/media/base/fakemediaengine.h b/media/base/fakemediaengine.h
index 87bc8a3..db5e2e4 100644
--- a/media/base/fakemediaengine.h
+++ b/media/base/fakemediaengine.h
@@ -38,7 +38,7 @@
#include "talk/media/base/mediaengine.h"
#include "talk/media/base/rtputils.h"
#include "talk/media/base/streamparams.h"
-#include "talk/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/base/sessiondescription.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/stringutils.h"
diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
index b1164b5..5e03ef9 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -843,7 +843,6 @@
capture_jitter_ms(0),
avg_encode_ms(0),
encode_usage_percent(0),
- encode_rsd(0),
capture_queue_delay_ms_per_s(0) {
}
@@ -865,7 +864,6 @@
int capture_jitter_ms;
int avg_encode_ms;
int encode_usage_percent;
- int encode_rsd;
int capture_queue_delay_ms_per_s;
VariableInfo<int> adapt_frame_drops;
VariableInfo<int> effects_frame_drops;
@@ -973,7 +971,7 @@
// StatsOptions::include_received_propagation_stats is true.
int total_received_propagation_delta_ms;
std::vector<int> recent_received_propagation_delta_ms;
- std::vector<int64> recent_received_packet_group_arrival_time_ms;
+ std::vector<int64_t> recent_received_packet_group_arrival_time_ms;
};
struct VoiceMediaInfo {
diff --git a/media/base/videoengine_unittest.h b/media/base/videoengine_unittest.h
index 6ea5b8c..472d42a 100644
--- a/media/base/videoengine_unittest.h
+++ b/media/base/videoengine_unittest.h
@@ -783,10 +783,7 @@
void SetSendSetsTransportBufferSizes() {
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
- // TODO(sriniv): Remove or re-enable this.
- // As part of b/8030474, send-buffer is size now controlled through
- // portallocator flags. Its not set by channels.
- // EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
+ EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
EXPECT_EQ(64 * 1024, network_interface_.recvbuf_size());
}
// Tests that we can send frames and the right payload type is used.
@@ -1380,6 +1377,9 @@
EXPECT_EQ(0, renderer_.num_rendered_frames());
EXPECT_TRUE(SendFrame());
EXPECT_FRAME_WAIT(1, 640, 400, kTimeout);
+ // Wait for one frame so they don't get dropped because we send frames too
+ // tightly.
+ rtc::Thread::Current()->ProcessMessages(30);
// Remove the capturer.
EXPECT_TRUE(channel_->SetCapturer(kSsrc, NULL));
// Wait for one black frame for removing the capturer.
diff --git a/media/webrtc/webrtcvideoengine.cc b/media/webrtc/webrtcvideoengine.cc
index 4a4e29a..04092f3 100644
--- a/media/webrtc/webrtcvideoengine.cc
+++ b/media/webrtc/webrtcvideoengine.cc
@@ -66,6 +66,22 @@
namespace {
+cricket::VideoFormat CreateVideoFormat(int width, int height, int framerate) {
+ return cricket::VideoFormat(
+ width,
+ height,
+ cricket::VideoFormat::FpsToInterval(framerate),
+ cricket::FOURCC_ANY);
+}
+
+cricket::VideoFormat VideoFormatFromCodec(const cricket::VideoCodec& codec) {
+ return CreateVideoFormat(codec.width, codec.height, codec.framerate);
+}
+
+cricket::VideoFormat VideoFormatFromVieCodec(const webrtc::VideoCodec& codec) {
+ return CreateVideoFormat(codec.width, codec.height, codec.maxFramerate);
+}
+
template <class T>
bool Changed(cricket::Settable<T> proposed,
cricket::Settable<T> original) {
@@ -172,10 +188,12 @@
}
struct FlushBlackFrameData : public rtc::MessageData {
- FlushBlackFrameData(uint32 s, int64 t) : ssrc(s), timestamp(t) {
+ FlushBlackFrameData(uint32 s, int64 t, int i)
+ : ssrc(s), timestamp(t), interval(i) {
}
uint32 ssrc;
int64 timestamp;
+ int interval;
};
class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
@@ -592,6 +610,15 @@
class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
public:
typedef std::map<int, webrtc::VideoEncoder*> EncoderMap; // key: payload type
+
+ enum AdaptFormatType {
+ // This is how we make SetSendStreamFormat take precedence over
+ // SetSendCodecs.
+ kAdaptFormatTypeNone = 0, // Unset
+ kAdaptFormatTypeCodec = 1, // From SetSendCodec
+ kAdaptFormatTypeStream = 2, // From SetStreamFormat
+ };
+
WebRtcVideoChannelSendInfo(int channel_id, int capture_id,
webrtc::ViEExternalCapture* external_capture,
rtc::CpuMonitor* cpu_monitor)
@@ -602,15 +629,24 @@
video_capturer_(NULL),
encoder_observer_(channel_id),
external_capture_(external_capture),
- interval_(0),
cpu_monitor_(cpu_monitor),
- old_adaptation_changes_(0) {
+ old_adaptation_changes_(0),
+ adapt_format_type_(kAdaptFormatTypeNone) {
}
int channel_id() const { return channel_id_; }
int capture_id() const { return capture_id_; }
void set_sending(bool sending) { sending_ = sending; }
bool sending() const { return sending_; }
+ void set_send_params(const VideoSendParams& send_params) {
+ send_params_ = send_params;
+ }
+ const VideoSendParams& send_params() const {
+ return send_params_;
+ }
+ const Settable<CapturedFrameInfo>& last_captured_frame_info() const {
+ return last_captured_frame_info_;
+ }
void set_muted(bool on) {
// TODO(asapersson): add support.
// video_adapter_.SetBlackOutput(on);
@@ -620,25 +656,51 @@
WebRtcEncoderObserver* encoder_observer() { return &encoder_observer_; }
webrtc::ViEExternalCapture* external_capture() { return external_capture_; }
- const VideoFormat& video_format() const {
- return video_format_;
+ const VideoFormat& adapt_format() const { return adapt_format_; }
+ AdaptFormatType adapt_format_type() const { return adapt_format_type_; }
+ bool adapt_format_set() const {
+ return adapt_format_type() != kAdaptFormatTypeNone;
}
- void set_video_format(const VideoFormat& video_format) {
- video_format_ = video_format;
- if (video_format_ != cricket::VideoFormat()) {
- interval_ = video_format_.interval;
+
+ // Returns true if the last captured frame info changed.
+ void SetLastCapturedFrameInfo(
+ const VideoFrame* frame, bool screencast, bool* changed) {
+ CapturedFrameInfo last;
+ if (last_captured_frame_info_.Get(&last) &&
+ frame->GetWidth() == last.width &&
+ frame->GetHeight() == last.height &&
+ screencast == last.screencast) {
+ *changed = false;
+ return;
}
- CoordinatedVideoAdapter* adapter = video_adapter();
- if (adapter) {
- adapter->OnOutputFormatRequest(video_format_);
- }
+
+ last_captured_frame_info_.Set(CapturedFrameInfo(
+ frame->GetWidth(), frame->GetHeight(), screencast));
+ *changed = true;
}
- void set_interval(int64 interval) {
- if (video_format() == cricket::VideoFormat()) {
- interval_ = interval;
+
+ // Tells the video adapter to adapt down to a given format. The
+ // type indicates where the format came from, where different types
+ // have slightly different behavior and priority.
+ void SetAdaptFormat(const VideoFormat& format, AdaptFormatType type) {
+ if (type < adapt_format_type_) {
+ // Formats from SetSendStream format are higher priority than
+ // ones from SetSendCodecs wich is higher priority than not
+ // being set. If something lower-prioirty comes in, just ignore
+ // it.
+ return;
}
+
+ // TODO(pthatcher): Use the adapter for all max size enforcement,
+ // both codec-based and SetSendStreamFormat-based. For now, we
+ // can't do that without fixing a lot of unit tests.
+ if (video_adapter() && type == kAdaptFormatTypeStream) {
+ video_adapter()->OnOutputFormatRequest(format);
+ }
+
+ adapt_format_ = format;
+ adapt_format_type_ = type;
}
- int64 interval() { return interval_; }
int CurrentAdaptReason() const {
if (!video_adapter()) {
@@ -653,15 +715,20 @@
return old_adaptation_changes_ + video_adapter()->adaptation_changes();
}
- StreamParams* stream_params() { return stream_params_.get(); }
void set_stream_params(const StreamParams& sp) {
- stream_params_.reset(new StreamParams(sp));
+ send_params_.stream = sp;
}
- void ClearStreamParams() { stream_params_.reset(); }
- bool has_ssrc(uint32 local_ssrc) const {
- return !stream_params_ ? false :
- stream_params_->has_ssrc(local_ssrc);
+ const StreamParams& stream_params() const { return send_params_.stream; }
+ // A default send channel can be non-active if a stream hasn't been
+ // added yet, or if all streams have been removed (at which point,
+ // Deactive is called).
+ bool IsActive() {
+ return stream_params().first_ssrc() != 0;
}
+ void Deactivate() {
+ send_params_.stream = StreamParams();
+ }
+
WebRtcLocalStreamInfo* local_stream_info() {
return &local_stream_info_;
}
@@ -696,6 +763,13 @@
CoordinatedVideoAdapter* adapter = video_adapter();
ASSERT(adapter && "Video adapter should not be null here.");
+ // TODO(pthatcher): Use the adapter for all max size enforcement,
+ // both codec-based and SetSendStreamFormat-based. For now, we
+ // can't do that without fixing a lot of unit tests.
+ if (adapt_format_type_ == kAdaptFormatTypeStream) {
+ adapter->OnOutputFormatRequest(adapt_format_);
+ }
+
UpdateAdapterCpuOptions();
overuse_observer_.reset(new WebRtcOveruseObserver(adapter));
@@ -826,6 +900,9 @@
private:
int channel_id_;
int capture_id_;
+ VideoSendParams send_params_;
+ // TODO(pthatcher): Merge CapturedFrameInfo and LocalStreamInfo.
+ Settable<CapturedFrameInfo> last_captured_frame_info_;
bool sending_;
bool muted_;
VideoCapturer* video_capturer_;
@@ -833,20 +910,17 @@
webrtc::ViEExternalCapture* external_capture_;
EncoderMap registered_encoders_;
- VideoFormat video_format_;
-
- rtc::scoped_ptr<StreamParams> stream_params_;
-
WebRtcLocalStreamInfo local_stream_info_;
- int64 interval_;
-
rtc::CpuMonitor* cpu_monitor_;
rtc::scoped_ptr<WebRtcOveruseObserver> overuse_observer_;
int old_adaptation_changes_;
VideoOptions video_options_;
+
+ VideoFormat adapt_format_;
+ AdaptFormatType adapt_format_type_;
};
const WebRtcVideoEngine::VideoCodecPref
@@ -861,17 +935,6 @@
{640, 400, FPS_TO_INTERVAL(30), FOURCC_ANY};
// TODO(ronghuawu): Change to 640x360.
-static void UpdateVideoCodec(const cricket::VideoFormat& video_format,
- webrtc::VideoCodec* target_codec) {
- if ((!target_codec) || (video_format == cricket::VideoFormat())) {
- return;
- }
- target_codec->width = video_format.width;
- target_codec->height = video_format.height;
- target_codec->maxFramerate = cricket::VideoFormat::IntervalToFps(
- video_format.interval);
-}
-
static bool GetCpuOveruseOptions(const VideoOptions& options,
webrtc::CpuOveruseOptions* overuse_options) {
int underuse_threshold = 0;
@@ -1103,11 +1166,8 @@
}
ASSERT(!video_codecs_.empty());
- default_codec_format_ = VideoFormat(
- video_codecs_[0].width,
- video_codecs_[0].height,
- VideoFormat::FpsToInterval(video_codecs_[0].framerate),
- FOURCC_ANY);
+ default_codec_format_ = VideoFormatFromCodec(video_codecs_[0]);
+
return true;
}
@@ -1774,15 +1834,15 @@
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)) {
+ 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.
+ // No external encoder created, so nothing to do.
return true;
}
@@ -1813,7 +1873,9 @@
LOG(LS_ERROR) << "The specified ssrc " << ssrc << " is not in use.";
return false;
}
- send_channel->set_video_format(format);
+
+ send_channel->SetAdaptFormat(
+ format, WebRtcVideoChannelSendInfo::kAdaptFormatTypeStream);
return true;
}
@@ -1897,7 +1959,7 @@
// If the default channel is already used for sending create a new channel
// otherwise use the default channel for sending.
int channel_id = kChannelIdUnset;
- if (!DefaultSendChannelInUse()) {
+ if (!DefaultSendChannelIsActive()) {
channel_id = default_channel_id_;
} else {
if (!CreateChannel(ssrc_key, MD_SEND, &channel_id)) {
@@ -1955,9 +2017,9 @@
}
WebRtcVideoChannelSendInfo* send_channel = GetSendChannelBySsrcKey(ssrc_key);
int channel_id = send_channel->channel_id();
- if (IsDefaultChannelId(channel_id) && !send_channel->stream_params()) {
- // Default channel will still exist. However, if stream_params() is NULL
- // there is no stream to remove.
+ if (IsDefaultChannelId(channel_id) && !send_channel->IsActive()) {
+ // Default channel will still exist. However, there is no stream
+ // to remove.
return false;
}
if (sending_) {
@@ -1979,7 +2041,7 @@
// The receive channels depend on the default channel, recycle it instead.
if (IsDefaultChannelId(channel_id)) {
SetCapturer(GetDefaultSendChannelSsrc(), NULL);
- send_channel->ClearStreamParams();
+ send_channel->Deactivate();
} else {
return DeleteSendChannel(ssrc_key);
}
@@ -1995,7 +2057,7 @@
// TODO(zhurunz) Remove this once BWE works properly across different send
// and receive channels.
// Reuse default channel for recv stream in 1:1 call.
- if (!InConferenceMode() && first_receive_ssrc_ == kSsrcUnset) {
+ if (!ConferenceModeIsEnabled() && first_receive_ssrc_ == kSsrcUnset) {
LOG(LS_INFO) << "Recv stream " << sp.first_ssrc()
<< " reuse default channel #"
<< default_channel_id_;
@@ -2211,11 +2273,11 @@
bool WebRtcVideoMediaChannel::HasReadySendChannels() {
return !send_channels_.empty() &&
- ((send_channels_.size() > 1) || DefaultSendChannelInUse());
+ ((send_channels_.size() > 1) || DefaultSendChannelIsActive());
}
-bool WebRtcVideoMediaChannel::DefaultSendChannelInUse() {
- return GetDefaultSendChannel() && GetDefaultSendChannel()->stream_params();
+bool WebRtcVideoMediaChannel::DefaultSendChannelIsActive() {
+ return GetDefaultSendChannel() && GetDefaultSendChannel()->IsActive();
}
bool WebRtcVideoMediaChannel::GetSendChannelSsrcKey(uint32 local_ssrc,
@@ -2240,7 +2302,7 @@
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
- if (send_channel->has_ssrc(local_ssrc)) {
+ if (send_channel->stream_params().has_ssrc(local_ssrc)) {
*ssrc_key = iter->first;
return true;
}
@@ -2284,7 +2346,7 @@
// this point a duplicate SSRC has been detected.
return false;
}
- if (!DefaultSendChannelInUse()) {
+ if (!DefaultSendChannelIsActive()) {
// |ssrc_key| should be kDefaultChannelSsrcKey here as the default
// channel should be re-used whenever it is not used.
*ssrc_key = kDefaultChannelSsrcKey;
@@ -2310,10 +2372,7 @@
}
uint32 WebRtcVideoMediaChannel::GetDefaultSendChannelSsrc() {
- if (!DefaultSendChannelInUse()) {
- return 0;
- }
- return GetDefaultSendChannel()->stream_params()->first_ssrc();
+ return GetDefaultSendChannel()->stream_params().first_ssrc();
}
bool WebRtcVideoMediaChannel::DeleteSendChannel(uint32 ssrc_key) {
@@ -2377,7 +2436,8 @@
send_channel->set_video_capturer(NULL, engine()->vie());
const int64 timestamp = send_channel->local_stream_info()->time_stamp();
if (send_codec_) {
- QueueBlackFrame(ssrc, timestamp, send_codec_->maxFramerate);
+ QueueBlackFrame(ssrc, timestamp,
+ VideoFormat::FpsToInterval(send_codec_->maxFramerate));
}
return true;
}
@@ -2418,8 +2478,7 @@
WebRtcVideoChannelSendInfo* send_channel = iter->second;
const int channel_id = send_channel->channel_id();
VideoSenderInfo sinfo;
- const StreamParams* send_params = send_channel->stream_params();
- if (!send_params) {
+ if (!send_channel->IsActive()) {
// This should only happen if the default vie channel is not in use.
// This can happen if no streams have ever been added or the stream
// corresponding to the default channel has been removed. Note that
@@ -2439,8 +2498,9 @@
WebRtcLocalStreamInfo* channel_stream_info =
send_channel->local_stream_info();
- for (size_t i = 0; i < send_params->ssrcs.size(); ++i) {
- sinfo.add_ssrc(send_params->ssrcs[i]);
+ const StreamParams& sp = send_channel->stream_params();
+ for (size_t i = 0; i < sp.ssrcs.size(); ++i) {
+ sinfo.add_ssrc(sp.ssrcs[i]);
}
sinfo.codec_name = send_codec_->plName;
sinfo.bytes_sent = bytes_sent;
@@ -2493,7 +2553,6 @@
sinfo.capture_jitter_ms = metrics.capture_jitter_ms;
sinfo.avg_encode_ms = metrics.avg_encode_time_ms;
sinfo.encode_usage_percent = metrics.encode_usage_percent;
- sinfo.encode_rsd = metrics.encode_rsd;
sinfo.capture_queue_delay_ms_per_s = metrics.capture_queue_delay_ms_per_s;
#else
sinfo.capture_jitter_ms = -1;
@@ -2723,7 +2782,8 @@
}
const int64 timestamp = send_channel->local_stream_info()->time_stamp();
if (send_codec_) {
- QueueBlackFrame(ssrc, timestamp, send_codec_->maxFramerate);
+ QueueBlackFrame(ssrc, timestamp,
+ VideoFormat::FpsToInterval(send_codec_->maxFramerate));
}
return true;
}
@@ -2745,7 +2805,7 @@
int processing_channel_id = GetRecvChannelId(ssrc);
if (processing_channel_id == kChannelIdUnset) {
// Allocate an unsignalled recv channel for processing in conference mode.
- if (!InConferenceMode()) {
+ if (!ConferenceModeIsEnabled()) {
// If we can't find or allocate one, use the default.
processing_channel_id = default_channel_id_;
} else if (!CreateUnsignalledRecvChannel(ssrc, &processing_channel_id)) {
@@ -3048,12 +3108,13 @@
rtc::Socket::OPT_RCVBUF,
kVideoRtpBufferSize);
- // TODO(sriniv): Remove or re-enable this.
- // As part of b/8030474, send-buffer is size now controlled through
- // portallocator flags.
- // network_interface_->SetOption(NetworkInterface::ST_RTP,
- // rtc::Socket::OPT_SNDBUF,
- // kVideoRtpBufferSize);
+ // Speculative change to increase the outbound socket buffer size.
+ // In b/15152257, we are seeing a significant number of packets discarded
+ // due to lack of socket buffer space, although it's not yet clear what the
+ // ideal value should be.
+ MediaChannel::SetOption(NetworkInterface::ST_RTP,
+ rtc::Socket::OPT_SNDBUF,
+ kVideoRtpBufferSize);
}
void WebRtcVideoMediaChannel::UpdateAspectRatio(int ratio_w, int ratio_h) {
@@ -3140,23 +3201,32 @@
// Send codec has not been set. No reason to process the frame any further.
return false;
}
- const VideoFormat& video_format = send_channel->video_format();
+
+ // TODO(pthatcher): Move drop logic to adapter.
// If the frame should be dropped.
- const bool video_format_set = video_format != cricket::VideoFormat();
- if (video_format_set &&
- (video_format.width == 0 && video_format.height == 0)) {
+ if (send_channel->adapt_format_set() &&
+ send_channel->adapt_format().width == 0 &&
+ send_channel->adapt_format().height == 0) {
return true;
}
- // Checks if we need to reset vie send codec.
- if (!MaybeResetVieSendCodec(send_channel,
- static_cast<int>(frame->GetWidth()),
- static_cast<int>(frame->GetHeight()),
- is_screencast, NULL)) {
- LOG(LS_ERROR) << "MaybeResetVieSendCodec failed with "
- << frame->GetWidth() << "x" << frame->GetHeight();
- return false;
+ bool changed;
+ send_channel->SetLastCapturedFrameInfo(frame, is_screencast, &changed);
+ if (changed) {
+ // If the last captured frame info changed, then calling
+ // SetSendParams will update to the latest resolution.
+ VideoSendParams send_params = send_channel->send_params();
+ // Note: We must copy the send_params because otherwise the memory
+ // checker will complain.
+ if (!SetSendParams(send_channel, send_params)) {
+ LOG(LS_ERROR) << "SetSendParams from SendFrame failed with "
+ << frame->GetWidth() << "x" << frame->GetHeight()
+ << " screencast? " << is_screencast;
+ return false;
+ }
+ LogSendCodecChange("Captured frame size changed");
}
+
const VideoFrame* frame_out = frame;
rtc::scoped_ptr<VideoFrame> processed_frame;
// TODO(hellner): Remove the need for disabling mute when screencasting.
@@ -3458,6 +3528,8 @@
LOG_RTCERR2(ConnectCaptureDevice, vie_capture, channel_id);
return false;
}
+
+ // Set up a new send channel.
rtc::scoped_ptr<WebRtcVideoChannelSendInfo> send_channel(
new WebRtcVideoChannelSendInfo(channel_id, vie_capture,
external_capture,
@@ -3549,7 +3621,7 @@
int fec_payload_type,
bool nack_enabled) {
bool enable = (red_payload_type != -1 && fec_payload_type != -1 &&
- !InConferenceMode());
+ !ConferenceModeIsEnabled());
if (enable) {
if (engine_->vie()->rtp()->SetHybridNACKFECStatus(
channel_id, nack_enabled, red_payload_type, fec_payload_type) != 0) {
@@ -3600,67 +3672,29 @@
return false;
}
+ send_channel->SetAdaptFormat(
+ VideoFormatFromVieCodec(codec),
+ WebRtcVideoChannelSendInfo::kAdaptFormatTypeCodec);
+
+ MaybeRegisterExternalEncoder(send_channel, codec);
+
+ VideoSendParams send_params = send_channel->send_params();
+ send_params.codec = codec;
+ if (!SetSendParams(send_channel, send_params)) {
+ 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.
const int channel_id = send_channel->channel_id();
- // Make a copy of the codec
- webrtc::VideoCodec target_codec = codec;
-
- // Set the default number of temporal layers for VP8.
- if (webrtc::kVideoCodecVP8 == codec.codecType) {
- target_codec.codecSpecific.VP8.numberOfTemporalLayers =
- kDefaultNumberOfTemporalLayers;
-
- // Turn off the VP8 error resilience
- target_codec.codecSpecific.VP8.resilience = webrtc::kResilienceOff;
-
- bool enable_denoising =
- options_.video_noise_reduction.GetWithDefaultIfUnset(true);
- target_codec.codecSpecific.VP8.denoisingOn = enable_denoising;
+ if (send_rtx_type_ != -1 &&
+ engine()->vie()->rtp()->SetRtxSendPayloadType(channel_id,
+ send_rtx_type_) != 0) {
+ LOG_RTCERR2(SetRtxSendPayloadType, channel_id, send_rtx_type_);
+ return false;
}
- MaybeRegisterExternalEncoder(send_channel, target_codec);
-
- // Resolution and framerate may vary for different send channels.
- const VideoFormat& video_format = send_channel->video_format();
- UpdateVideoCodec(video_format, &target_codec);
-
- if (target_codec.width == 0 && target_codec.height == 0) {
- const uint32 ssrc = send_channel->stream_params()->first_ssrc();
- LOG(LS_INFO) << "0x0 resolution selected. Captured frames will be dropped "
- << "for ssrc: " << ssrc << ".";
- } else {
- 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.
- if (current_codec == target_codec) {
- // Codec is already configured on channel. no need to apply.
- return true;
- }
- }
-
- if (0 != engine()->vie()->codec()->SetSendCodec(channel_id, target_codec)) {
- LOG_RTCERR2(SetSendCodec, channel_id, target_codec.plName);
- 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.
- if (send_rtx_type_ != -1 &&
- engine()->vie()->rtp()->SetRtxSendPayloadType(channel_id,
- send_rtx_type_) != 0) {
- LOG_RTCERR2(SetRtxSendPayloadType, channel_id, send_rtx_type_);
- return false;
- }
- }
- send_channel->set_interval(
- cricket::VideoFormat::FpsToInterval(target_codec.maxFramerate));
return true;
}
@@ -3833,117 +3867,107 @@
return recv_channel_id;
}
-// If the new frame size is different from the send codec size we set on vie,
-// we need to reset the send codec on vie.
-// The new send codec size should not exceed send_codec_ which is controlled
-// only by the 'jec' logic.
-// TODO(pthatcher): Get rid of this function, so we only ever set up
-// codecs in a single place.
-bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec(
+bool WebRtcVideoMediaChannel::SetSendParams(
WebRtcVideoChannelSendInfo* send_channel,
- int new_width,
- int new_height,
- bool is_screencast,
- bool* reset) {
- if (reset) {
- *reset = false;
- }
- ASSERT(send_codec_.get() != NULL);
+ const VideoSendParams& send_params) {
+ const int channel_id = send_channel->channel_id();
- webrtc::VideoCodec target_codec = *send_codec_;
- const VideoFormat& video_format = send_channel->video_format();
- UpdateVideoCodec(video_format, &target_codec);
+ CapturedFrameInfo frame;
+ send_channel->last_captured_frame_info().Get(&frame);
- // Vie send codec size should not exceed target_codec.
- int target_width = new_width;
- int target_height = new_height;
- if (!is_screencast &&
- (new_width > target_codec.width || new_height > target_codec.height)) {
- target_width = target_codec.width;
- target_height = target_codec.height;
+ // TODO(pthatcher): This checking of the max height and width is
+ // only needed because some unit tests bypass the VideoAdapter, and
+ // others expect behavior from the adapter different than what it
+ // actually does. We should fix the tests and remove this block.
+ VideoFormat max = send_channel->adapt_format();
+ size_t max_width = static_cast<size_t>(max.width);
+ size_t max_height = static_cast<size_t>(max.height);
+ if (!send_channel->last_captured_frame_info().IsSet() ||
+ (!frame.screencast &&
+ (frame.width > max_width || frame.height > max_height))) {
+ frame.width = max_width;
+ frame.height = max_height;
}
+ webrtc::VideoCodec codec;
+ ConfigureVieCodecFromSendParams(channel_id, send_params, frame, &codec);
+ // TODO(pthatcher): Figure out a clean way to configure the max
+ // framerate and sanitize the bitrates inside of
+ // ConfigureVieCodecFromSendParams.
+ codec.maxFramerate = max.framerate();
+ SanitizeBitrates(channel_id, &codec);
+
// Get current vie codec.
- webrtc::VideoCodec vie_codec;
- const int channel_id = send_channel->channel_id();
- if (engine()->vie()->codec()->GetSendCodec(channel_id, vie_codec) != 0) {
+ webrtc::VideoCodec current;
+ if (engine()->vie()->codec()->GetSendCodec(channel_id, current) != 0) {
LOG_RTCERR1(GetSendCodec, channel_id);
return false;
}
- const int cur_width = vie_codec.width;
- const int cur_height = vie_codec.height;
- // Only reset send codec when there is a size change. Additionally,
- // automatic resize needs to be turned off when screencasting and on when
- // not screencasting.
- // Don't allow automatic resizing for screencasting.
- bool automatic_resize = !is_screencast;
- // Turn off VP8 frame dropping when screensharing as the current model does
- // not work well at low fps.
- bool vp8_frame_dropping = !is_screencast;
- // TODO(pbos): Remove |video_noise_reduction| and enable it for all
- // non-screencast.
- bool enable_denoising =
- options_.video_noise_reduction.GetWithDefaultIfUnset(true);
- // Disable denoising for screencasting.
- if (is_screencast) {
- enable_denoising = false;
- }
- int screencast_min_bitrate =
- options_.screencast_min_bitrate.GetWithDefaultIfUnset(0);
- 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) {
- reset_send_codec = reset_send_codec ||
- automatic_resize != vie_codec.codecSpecific.VP8.automaticResizeOn ||
- enable_denoising != vie_codec.codecSpecific.VP8.denoisingOn ||
- vp8_frame_dropping != vie_codec.codecSpecific.VP8.frameDroppingOn;
- }
-
- if (reset_send_codec) {
- // Set the new codec on vie.
- vie_codec.width = target_width;
- vie_codec.height = target_height;
- vie_codec.maxFramerate = target_codec.maxFramerate;
- vie_codec.startBitrate = target_codec.startBitrate;
- vie_codec.minBitrate = target_codec.minBitrate;
- vie_codec.maxBitrate = target_codec.maxBitrate;
- vie_codec.targetBitrate = 0;
- if (vie_codec.codecType == webrtc::kVideoCodecVP8) {
- vie_codec.codecSpecific.VP8.automaticResizeOn = automatic_resize;
- vie_codec.codecSpecific.VP8.denoisingOn = enable_denoising;
- vie_codec.codecSpecific.VP8.frameDroppingOn = vp8_frame_dropping;
- }
- SanitizeBitrates(channel_id, &vie_codec);
-
- if (engine()->vie()->codec()->SetSendCodec(channel_id, vie_codec) != 0) {
+ if (current != codec) {
+ if (engine()->vie()->codec()->SetSendCodec(channel_id, codec) != 0) {
LOG_RTCERR1(SetSendCodec, channel_id);
return false;
}
-
- if (is_screencast) {
- engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id,
- screencast_min_bitrate);
- } else {
- // In case of switching from screencast to regular capture, set
- // min bitrate padding and pacer back to defaults.
- engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id, 0);
- }
- engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id, true);
- // 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;
- }
- LogSendCodecChange("Capture size changed");
}
+ if (frame.screencast) {
+ int screencast_min_bitrate =
+ options_.screencast_min_bitrate.GetWithDefaultIfUnset(0);
+ engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id,
+ screencast_min_bitrate);
+ } else {
+ // In case of switching from screencast to regular capture, set
+ // min bitrate padding and pacer back to defaults.
+ engine()->vie()->rtp()->SetMinTransmitBitrate(channel_id, 0);
+ }
+ engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id, true);
+
+ if (send_channel->IsActive()) {
+ if (!SetSendSsrcs(channel_id, send_params.stream, codec)) {
+ return false;
+ }
+ }
+
+ send_channel->set_send_params(send_params);
+ return true;
+}
+
+bool WebRtcVideoMediaChannel::ConfigureVieCodecFromSendParams(
+ int channel_id,
+ const VideoSendParams& send_params,
+ const CapturedFrameInfo& last_captured_frame_info,
+ webrtc::VideoCodec* codec_out) {
+ webrtc::VideoCodec codec = send_params.codec;
+
+ codec.width = static_cast<int>(last_captured_frame_info.width);
+ codec.height = static_cast<int>(last_captured_frame_info.height);
+ codec.targetBitrate = 0;
+ if (codec.codecType == webrtc::kVideoCodecVP8) {
+ codec.codecSpecific.VP8.numberOfTemporalLayers =
+ kDefaultNumberOfTemporalLayers;
+ codec.codecSpecific.VP8.resilience = webrtc::kResilienceOff;
+ }
+
+ if (last_captured_frame_info.screencast) {
+ codec.mode = webrtc::kScreensharing;
+ if (codec.codecType == webrtc::kVideoCodecVP8) {
+ codec.codecSpecific.VP8.denoisingOn = false;
+ codec.codecSpecific.VP8.automaticResizeOn = false;
+ codec.codecSpecific.VP8.frameDroppingOn = false;
+ }
+ } else {
+ codec.mode = webrtc::kRealtimeVideo;
+ if (codec.codecType == webrtc::kVideoCodecVP8) {
+ // TODO(pthatcher): Pass in options in VideoSendParams.
+ codec.codecSpecific.VP8.denoisingOn =
+ options_.video_noise_reduction.GetWithDefaultIfUnset(true);
+ codec.codecSpecific.VP8.automaticResizeOn = true;
+ codec.codecSpecific.VP8.frameDroppingOn = true;
+ }
+ }
+
+ *codec_out = codec;
return true;
}
@@ -3987,10 +4011,9 @@
}
void WebRtcVideoMediaChannel::OnMessage(rtc::Message* msg) {
- FlushBlackFrameData* black_frame_data =
- static_cast<FlushBlackFrameData*>(msg->pdata);
- FlushBlackFrame(black_frame_data->ssrc, black_frame_data->timestamp);
- delete black_frame_data;
+ FlushBlackFrameData* data = static_cast<FlushBlackFrameData*>(msg->pdata);
+ FlushBlackFrame(data->ssrc, data->timestamp, data->interval);
+ delete data;
}
int WebRtcVideoMediaChannel::SendPacket(int channel, const void* data,
@@ -4007,19 +4030,18 @@
}
void WebRtcVideoMediaChannel::QueueBlackFrame(uint32 ssrc, int64 timestamp,
- int framerate) {
+ int interval) {
if (timestamp) {
FlushBlackFrameData* black_frame_data = new FlushBlackFrameData(
- ssrc,
- timestamp);
+ ssrc, timestamp, interval);
const int delay_ms = static_cast<int>(
- 2 * cricket::VideoFormat::FpsToInterval(framerate) *
- rtc::kNumMillisecsPerSec / rtc::kNumNanosecsPerSec);
+ 2 * interval * rtc::kNumMillisecsPerSec / rtc::kNumNanosecsPerSec);
worker_thread()->PostDelayed(delay_ms, this, 0, black_frame_data);
}
}
-void WebRtcVideoMediaChannel::FlushBlackFrame(uint32 ssrc, int64 timestamp) {
+void WebRtcVideoMediaChannel::FlushBlackFrame(
+ uint32 ssrc, int64 timestamp, int interval) {
WebRtcVideoChannelSendInfo* send_channel = GetSendChannelBySsrc(ssrc);
if (!send_channel) {
return;
@@ -4041,7 +4063,7 @@
WebRtcVideoFrame black_frame;
// Black frame is not screencast.
const bool screencasting = false;
- const int64 timestamp_delta = send_channel->interval();
+ const int64 timestamp_delta = interval;
if (!black_frame.InitToBlack(send_codec_->width, send_codec_->height, 1, 1,
last_frame_elapsed_time + timestamp_delta,
last_frame_time_stamp + timestamp_delta) ||
@@ -4094,7 +4116,7 @@
bool WebRtcVideoMediaChannel::SetPrimaryAndRtxSsrcs(
int channel_id, int idx, uint32 primary_ssrc,
- const StreamParams& send_params) {
+ const StreamParams& sp) {
LOG(LS_INFO) << "Set primary ssrc " << primary_ssrc
<< " on channel " << channel_id << " idx " << idx;
if (engine()->vie()->rtp()->SetLocalSSRC(
@@ -4105,7 +4127,7 @@
}
uint32 rtx_ssrc = 0;
- if (send_params.GetFidSsrc(primary_ssrc, &rtx_ssrc)) {
+ if (sp.GetFidSsrc(primary_ssrc, &rtx_ssrc)) {
LOG(LS_INFO) << "Set rtx ssrc " << rtx_ssrc
<< " on channel " << channel_id << " idx " << idx;
if (engine()->vie()->rtp()->SetLocalSSRC(
diff --git a/media/webrtc/webrtcvideoengine.h b/media/webrtc/webrtcvideoengine.h
index 72ba9df..cc81ee9 100644
--- a/media/webrtc/webrtcvideoengine.h
+++ b/media/webrtc/webrtcvideoengine.h
@@ -234,6 +234,22 @@
rtc::scoped_ptr<rtc::CpuMonitor> cpu_monitor_;
};
+struct CapturedFrameInfo {
+ CapturedFrameInfo() : width(0), height(0), screencast(false) {}
+ CapturedFrameInfo(size_t width, size_t height, bool screencast) :
+ width(width), height(height), screencast(screencast) {}
+
+ size_t width;
+ size_t height;
+ bool screencast;
+};
+
+// TODO(pthatcher): Add VideoOptions.
+struct VideoSendParams {
+ webrtc::VideoCodec codec;
+ StreamParams stream;
+};
+
class WebRtcVideoMediaChannel : public rtc::MessageHandler,
public VideoMediaChannel,
public webrtc::Transport {
@@ -316,6 +332,22 @@
virtual int SendPacket(int channel, const void* data, int len) OVERRIDE;
virtual int SendRTCPPacket(int channel, const void* data, int len) OVERRIDE;
+ bool ConferenceModeIsEnabled() const {
+ return options_.conference_mode.GetWithDefaultIfUnset(false);
+ }
+
+ // We take lots of things as input from applications (packaged in
+ // params), but ViE wants lots of those packed instead as a
+ // webrtc::VideoCodec. This is where we convert between the inputs
+ // we get from the applications and the input to give to ViE. We
+ // also configure the codec differently depending on the latest
+ // frame that we have received (in particular, depending on the
+ // resolution and whether the it was a screencast frame or not).
+ virtual bool ConfigureVieCodecFromSendParams(
+ int channel_id,
+ const VideoSendParams& send_params,
+ const CapturedFrameInfo& last_captured_frame_info,
+ webrtc::VideoCodec* codec);
// Checks the current bitrate estimate and modifies the bitrates
// accordingly, including converting kAutoBandwidth to the correct defaults.
virtual void SanitizeBitrates(
@@ -359,6 +391,9 @@
bool SetSendCodec(const webrtc::VideoCodec& codec);
bool SetSendCodec(WebRtcVideoChannelSendInfo* send_channel,
const webrtc::VideoCodec& codec);
+ bool SetSendParams(WebRtcVideoChannelSendInfo* send_channel,
+ const VideoSendParams& params);
+
// Prepares the channel with channel id |info->channel_id()| to receive all
// codecs in |receive_codecs_| and start receive packets.
bool SetReceiveCodecs(WebRtcVideoChannelRecvInfo* info);
@@ -370,12 +405,6 @@
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);
// 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();
@@ -387,7 +416,7 @@
bool SendIntraFrame(int channel_id);
bool HasReadySendChannels();
- bool DefaultSendChannelInUse();
+ bool DefaultSendChannelIsActive();
// Returns the ssrc key corresponding to the provided local SSRC in
// |ssrc_key|. The return value is true upon success. If the local
@@ -414,14 +443,11 @@
WebRtcVideoChannelRecvInfo* GetDefaultRecvChannel();
WebRtcVideoChannelRecvInfo* GetRecvChannelBySsrc(uint32 ssrc);
- bool InConferenceMode() const {
- return options_.conference_mode.GetWithDefaultIfUnset(false);
- }
bool RemoveCapturer(uint32 ssrc);
rtc::MessageQueue* worker_thread() { return engine_->worker_thread(); }
- void QueueBlackFrame(uint32 ssrc, int64 timestamp, int framerate);
- void FlushBlackFrame(uint32 ssrc, int64 timestamp);
+ void QueueBlackFrame(uint32 ssrc, int64 timestamp, int interval);
+ void FlushBlackFrame(uint32 ssrc, int64 timestamp, int interval);
void SetNetworkTransmissionState(bool is_transmitting);
diff --git a/media/webrtc/webrtcvideoengine2.cc b/media/webrtc/webrtcvideoengine2.cc
index 96df0f7..d79f71d 100644
--- a/media/webrtc/webrtcvideoengine2.cc
+++ b/media/webrtc/webrtcvideoengine2.cc
@@ -42,6 +42,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/call.h"
+#include "webrtc/video_decoder.h"
#include "webrtc/video_encoder.h"
#define UNIMPLEMENTED \
@@ -56,10 +57,18 @@
return _stricmp(name1.c_str(), name2.c_str()) == 0;
}
+const char* kInternallySupportedCodecs[] = {
+ kVp8CodecName,
+};
+
// True if codec is supported by a software implementation that's always
// available.
static bool CodecIsInternallySupported(const std::string& codec_name) {
- return CodecNameMatches(codec_name, kVp8CodecName);
+ for (size_t i = 0; i < ARRAY_SIZE(kInternallySupportedCodecs); ++i) {
+ if (CodecNameMatches(codec_name, kInternallySupportedCodecs[i]))
+ return true;
+ }
+ return false;
}
static std::string CodecVectorToString(const std::vector<VideoCodec>& codecs) {
@@ -117,6 +126,8 @@
static const int kDefaultRtcpReceiverReportSsrc = 1;
+static const int kConferenceModeTemporalLayerBitrateBps = 100000;
+
// External video encoders are given payloads 120-127. This also means that we
// only support up to 8 external payload types.
static const int kExternalVideoPayloadTypeBase = 120;
@@ -238,8 +249,9 @@
const VideoOptions& options,
size_t num_streams) {
if (num_streams != 1) {
- LOG(LS_ERROR) << "Unsupported number of streams: " << num_streams;
- return std::vector<webrtc::VideoStream>();
+ LOG(LS_WARNING) << "Unsupported number of streams (" << num_streams
+ << "), falling back to one.";
+ num_streams = 1;
}
webrtc::VideoStream stream;
@@ -250,6 +262,12 @@
int min_bitrate = kMinVideoBitrate;
codec.GetParam(kCodecParamMinBitrate, &min_bitrate);
+ // Clamp the min video bitrate, this is set from JavaScript directly and needs
+ // to be sanitized.
+ if (min_bitrate < kMinVideoBitrate) {
+ min_bitrate = kMinVideoBitrate;
+ }
+
int max_bitrate = kMaxVideoBitrate;
codec.GetParam(kCodecParamMaxBitrate, &max_bitrate);
stream.min_bitrate_bps = min_bitrate * 1000;
@@ -419,10 +437,12 @@
assert(initialized_);
LOG(LS_INFO) << "CreateChannel: "
<< (voice_channel != NULL ? "With" : "Without")
- << " voice channel.";
+ << " voice channel. Options: " << options.ToString();
WebRtcVideoChannel2* channel =
new WebRtcVideoChannel2(call_factory_,
+ voice_engine_,
voice_channel,
+ options,
external_encoder_factory_,
external_decoder_factory_,
GetVideoEncoderFactory());
@@ -430,7 +450,6 @@
delete channel;
return NULL;
}
- channel->SetOptions(options);
channel->SetRecvCodecs(video_codecs_);
return channel;
}
@@ -744,24 +763,35 @@
WebRtcVideoChannel2::WebRtcVideoChannel2(
WebRtcCallFactory* call_factory,
+ WebRtcVoiceEngine* voice_engine,
VoiceMediaChannel* voice_channel,
+ const VideoOptions& options,
WebRtcVideoEncoderFactory* external_encoder_factory,
WebRtcVideoDecoderFactory* external_decoder_factory,
WebRtcVideoEncoderFactory2* encoder_factory)
: unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
+ voice_channel_(voice_channel),
external_encoder_factory_(external_encoder_factory),
external_decoder_factory_(external_decoder_factory),
encoder_factory_(encoder_factory) {
- // TODO(pbos): Connect the video and audio with |voice_channel|.
+ SetDefaultOptions();
+ options_.SetAll(options);
webrtc::Call::Config config(this);
config.overuse_callback = this;
+ if (voice_engine != NULL) {
+ config.voice_engine = voice_engine->voe()->engine();
+ }
+
+ // Set start bitrate for the call. A default is provided by SetDefaultOptions.
+ int start_bitrate_kbps;
+ options_.video_start_bitrate.Get(&start_bitrate_kbps);
+ config.stream_start_bitrate_bps = start_bitrate_kbps * 1000;
+
call_.reset(call_factory->CreateCall(config));
rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
sending_ = false;
default_send_ssrc_ = 0;
-
- SetDefaultOptions();
}
void WebRtcVideoChannel2::SetDefaultOptions() {
@@ -769,6 +799,9 @@
options_.suspend_below_min_bitrate.Set(false);
options_.use_payload_padding.Set(false);
options_.video_noise_reduction.Set(true);
+ options_.video_start_bitrate.Set(
+ webrtc::Call::Config::kDefaultStartBitrateBps / 1000);
+ options_.screencast_min_bitrate.Set(0);
}
WebRtcVideoChannel2::~WebRtcVideoChannel2() {
@@ -789,6 +822,37 @@
bool WebRtcVideoChannel2::Init() { return true; }
+bool WebRtcVideoChannel2::CodecIsExternallySupported(
+ const std::string& name) const {
+ if (external_encoder_factory_ == NULL) {
+ return false;
+ }
+
+ const std::vector<WebRtcVideoEncoderFactory::VideoCodec> external_codecs =
+ external_encoder_factory_->codecs();
+ for (size_t c = 0; c < external_codecs.size(); ++c) {
+ if (CodecNameMatches(name, external_codecs[c].name)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+std::vector<WebRtcVideoChannel2::VideoCodecSettings>
+WebRtcVideoChannel2::FilterSupportedCodecs(
+ const std::vector<WebRtcVideoChannel2::VideoCodecSettings>& mapped_codecs)
+ const {
+ std::vector<VideoCodecSettings> supported_codecs;
+ for (size_t i = 0; i < mapped_codecs.size(); ++i) {
+ const VideoCodecSettings& codec = mapped_codecs[i];
+ if (CodecIsInternallySupported(codec.codec.name) ||
+ CodecIsExternallySupported(codec.codec.name)) {
+ supported_codecs.push_back(codec);
+ }
+ }
+ return supported_codecs;
+}
+
bool WebRtcVideoChannel2::SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
LOG(LS_INFO) << "SetRecvCodecs: " << CodecVectorToString(codecs);
if (!ValidateCodecFormats(codecs)) {
@@ -797,21 +861,19 @@
const std::vector<VideoCodecSettings> mapped_codecs = MapCodecs(codecs);
if (mapped_codecs.empty()) {
- LOG(LS_ERROR) << "SetRecvCodecs called without video codec payloads.";
+ LOG(LS_ERROR) << "SetRecvCodecs called without any video codecs.";
return false;
}
- // TODO(pbos): Add a decoder factory which controls supported codecs.
- // Blocked on webrtc:2854.
- for (size_t i = 0; i < mapped_codecs.size(); ++i) {
- if (!CodecNameMatches(mapped_codecs[i].codec.name, kVp8CodecName)) {
- LOG(LS_ERROR) << "SetRecvCodecs called with unsupported codec: '"
- << mapped_codecs[i].codec.name << "'";
- return false;
- }
+ const std::vector<VideoCodecSettings> supported_codecs =
+ FilterSupportedCodecs(mapped_codecs);
+
+ if (mapped_codecs.size() != supported_codecs.size()) {
+ LOG(LS_ERROR) << "SetRecvCodecs called with unsupported video codecs.";
+ return false;
}
- recv_codecs_ = mapped_codecs;
+ recv_codecs_ = supported_codecs;
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it =
@@ -998,8 +1060,20 @@
webrtc::VideoReceiveStream::Config config;
ConfigureReceiverRtp(&config, sp);
- receive_streams_[ssrc] =
- new WebRtcVideoReceiveStream(call_.get(), config, recv_codecs_);
+
+ // Set up A/V sync if there is a VoiceChannel.
+ // TODO(pbos): The A/V is synched by the receiving channel. So we need to know
+ // the SSRC of the remote audio channel in order to sync the correct webrtc
+ // VoiceEngine channel. For now sync the first channel in non-conference to
+ // match existing behavior in WebRtcVideoEngine.
+ if (voice_channel_ != NULL && receive_streams_.empty() &&
+ !options_.conference_mode.GetWithDefaultIfUnset(false)) {
+ config.audio_channel_id =
+ static_cast<WebRtcVoiceMediaChannel*>(voice_channel_)->voe_channel();
+ }
+
+ receive_streams_[ssrc] = new WebRtcVideoReceiveStream(
+ call_.get(), external_decoder_factory_, config, recv_codecs_);
return true;
}
@@ -1284,12 +1358,13 @@
rtc::Socket::OPT_RCVBUF,
kVideoRtpBufferSize);
- // TODO(sriniv): Remove or re-enable this.
- // As part of b/8030474, send-buffer is size now controlled through
- // portallocator flags.
- // network_interface_->SetOption(NetworkInterface::ST_RTP,
- // rtc::Socket::OPT_SNDBUF,
- // kVideoRtpBufferSize);
+ // Speculative change to increase the outbound socket buffer size.
+ // In b/15152257, we are seeing a significant number of packets discarded
+ // due to lack of socket buffer space, although it's not yet clear what the
+ // ideal value should be.
+ MediaChannel::SetOption(NetworkInterface::ST_RTP,
+ rtc::Socket::OPT_SNDBUF,
+ kVideoRtpBufferSize);
}
void WebRtcVideoChannel2::UpdateAspectRatio(int ratio_w, int ratio_h) {
@@ -1663,30 +1738,47 @@
void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions(
int width,
int height,
- bool override_max) {
+ bool is_screencast) {
+ if (last_dimensions_.width == width && last_dimensions_.height == height &&
+ last_dimensions_.is_screencast == is_screencast) {
+ // Configured using the same parameters, do not reconfigure.
+ return;
+ }
+
+ last_dimensions_.width = width;
+ last_dimensions_.height = height;
+ last_dimensions_.is_screencast = is_screencast;
+
assert(!parameters_.encoder_config.streams.empty());
LOG(LS_VERBOSE) << "SetDimensions: " << width << "x" << height;
VideoCodecSettings codec_settings;
parameters_.codec_settings.Get(&codec_settings);
// Restrict dimensions according to codec max.
- if (!override_max) {
+ if (!is_screencast) {
if (codec_settings.codec.width < width)
width = codec_settings.codec.width;
if (codec_settings.codec.height < height)
height = codec_settings.codec.height;
}
- if (parameters_.encoder_config.streams.back().width == width &&
- parameters_.encoder_config.streams.back().height == height) {
- return;
- }
-
webrtc::VideoEncoderConfig encoder_config = parameters_.encoder_config;
encoder_config.encoder_specific_settings =
encoder_factory_->CreateVideoEncoderSettings(codec_settings.codec,
parameters_.options);
+ if (is_screencast) {
+ int screencast_min_bitrate_kbps;
+ parameters_.options.screencast_min_bitrate.Get(
+ &screencast_min_bitrate_kbps);
+ encoder_config.min_transmit_bitrate_bps =
+ screencast_min_bitrate_kbps * 1000;
+ encoder_config.content_type = webrtc::VideoEncoderConfig::kScreenshare;
+ } else {
+ encoder_config.min_transmit_bitrate_bps = 0;
+ encoder_config.content_type = webrtc::VideoEncoderConfig::kRealtimeVideo;
+ }
+
VideoCodec codec = codec_settings.codec;
codec.width = width;
codec.height = height;
@@ -1694,6 +1786,14 @@
encoder_config.streams = encoder_factory_->CreateVideoStreams(
codec, parameters_.options, parameters_.config.rtp.ssrcs.size());
+ // Conference mode screencast uses 2 temporal layers split at 100kbit.
+ if (parameters_.options.conference_mode.GetWithDefaultIfUnset(false) &&
+ is_screencast && encoder_config.streams.size() == 1) {
+ encoder_config.streams[0].temporal_layer_thresholds_bps.clear();
+ encoder_config.streams[0].temporal_layer_thresholds_bps.push_back(
+ kConferenceModeTemporalLayerBitrateBps);
+ }
+
bool stream_reconfigured = stream_->ReconfigureVideoEncoder(encoder_config);
encoder_factory_->DestroyVideoEncoderSettings(
@@ -1826,11 +1926,13 @@
WebRtcVideoChannel2::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream(
webrtc::Call* call,
+ WebRtcVideoDecoderFactory* external_decoder_factory,
const webrtc::VideoReceiveStream::Config& config,
const std::vector<VideoCodecSettings>& recv_codecs)
: call_(call),
stream_(NULL),
config_(config),
+ external_decoder_factory_(external_decoder_factory),
renderer_(NULL),
last_width_(-1),
last_height_(-1) {
@@ -1841,39 +1943,67 @@
WebRtcVideoChannel2::WebRtcVideoReceiveStream::~WebRtcVideoReceiveStream() {
call_->DestroyVideoReceiveStream(stream_);
+ ClearDecoders(&allocated_decoders_);
+}
+
+WebRtcVideoChannel2::WebRtcVideoReceiveStream::AllocatedDecoder
+WebRtcVideoChannel2::WebRtcVideoReceiveStream::CreateOrReuseVideoDecoder(
+ std::vector<AllocatedDecoder>* old_decoders,
+ const VideoCodec& codec) {
+ webrtc::VideoCodecType type = CodecTypeFromName(codec.name);
+
+ for (size_t i = 0; i < old_decoders->size(); ++i) {
+ if ((*old_decoders)[i].type == type) {
+ AllocatedDecoder decoder = (*old_decoders)[i];
+ (*old_decoders)[i] = old_decoders->back();
+ old_decoders->pop_back();
+ return decoder;
+ }
+ }
+
+ if (external_decoder_factory_ != NULL) {
+ webrtc::VideoDecoder* decoder =
+ external_decoder_factory_->CreateVideoDecoder(type);
+ if (decoder != NULL) {
+ return AllocatedDecoder(decoder, type, true);
+ }
+ }
+
+ if (type == webrtc::kVideoCodecVP8) {
+ return AllocatedDecoder(
+ webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kVp8), type, false);
+ }
+
+ // This shouldn't happen, we should not be trying to create something we don't
+ // support.
+ assert(false);
+ return AllocatedDecoder(NULL, webrtc::kVideoCodecUnknown, false);
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRecvCodecs(
const std::vector<VideoCodecSettings>& recv_codecs) {
+ std::vector<AllocatedDecoder> old_decoders = allocated_decoders_;
+ allocated_decoders_.clear();
+ config_.decoders.clear();
+ for (size_t i = 0; i < recv_codecs.size(); ++i) {
+ AllocatedDecoder allocated_decoder =
+ CreateOrReuseVideoDecoder(&old_decoders, recv_codecs[i].codec);
+ allocated_decoders_.push_back(allocated_decoder);
+
+ webrtc::VideoReceiveStream::Decoder decoder;
+ decoder.decoder = allocated_decoder.decoder;
+ decoder.payload_type = recv_codecs[i].codec.id;
+ decoder.payload_name = recv_codecs[i].codec.name;
+ config_.decoders.push_back(decoder);
+ }
+
// TODO(pbos): Reconfigure RTX based on incoming recv_codecs.
- // TODO(pbos): Base receive codecs off recv_codecs_ and set up using a
- // DecoderFactory similar to send side. Pending webrtc:2854.
- // Also set up default codecs if there's nothing in recv_codecs_.
- webrtc::VideoCodec codec;
- memset(&codec, 0, sizeof(codec));
-
- codec.plType = kDefaultVideoCodecPref.payload_type;
- strcpy(codec.plName, kDefaultVideoCodecPref.name);
- codec.codecType = webrtc::kVideoCodecVP8;
- codec.codecSpecific.VP8.resilience = webrtc::kResilientStream;
- codec.codecSpecific.VP8.numberOfTemporalLayers = 1;
- codec.codecSpecific.VP8.denoisingOn = true;
- codec.codecSpecific.VP8.errorConcealmentOn = false;
- codec.codecSpecific.VP8.automaticResizeOn = false;
- codec.codecSpecific.VP8.frameDroppingOn = true;
- codec.codecSpecific.VP8.keyFrameInterval = 3000;
- // Bitrates don't matter and are ignored for the receiver. This is put in to
- // have the current underlying implementation accept the VideoCodec.
- codec.minBitrate = codec.startBitrate = codec.maxBitrate = 300;
- config_.codecs.clear();
- config_.codecs.push_back(codec);
-
config_.rtp.fec = recv_codecs.front().fec;
-
config_.rtp.nack.rtp_history_ms =
IsNackEnabled(recv_codecs.begin()->codec) ? kNackHistoryMs : 0;
config_.rtp.remb = IsRembEnabled(recv_codecs.begin()->codec);
+ ClearDecoders(&old_decoders);
RecreateWebRtcStream();
}
@@ -1891,6 +2021,19 @@
stream_->Start();
}
+void WebRtcVideoChannel2::WebRtcVideoReceiveStream::ClearDecoders(
+ std::vector<AllocatedDecoder>* allocated_decoders) {
+ for (size_t i = 0; i < allocated_decoders->size(); ++i) {
+ if ((*allocated_decoders)[i].external) {
+ external_decoder_factory_->DestroyVideoDecoder(
+ (*allocated_decoders)[i].decoder);
+ } else {
+ delete (*allocated_decoders)[i].decoder;
+ }
+ }
+ allocated_decoders->clear();
+}
+
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RenderFrame(
const webrtc::I420VideoFrame& frame,
int time_to_render_ms) {
@@ -2050,31 +2193,6 @@
return video_codecs;
}
-std::vector<WebRtcVideoChannel2::VideoCodecSettings>
-WebRtcVideoChannel2::FilterSupportedCodecs(
- const std::vector<WebRtcVideoChannel2::VideoCodecSettings>& mapped_codecs) {
- std::vector<VideoCodecSettings> supported_codecs;
- for (size_t i = 0; i < mapped_codecs.size(); ++i) {
- const VideoCodecSettings& codec = mapped_codecs[i];
- if (CodecIsInternallySupported(codec.codec.name)) {
- supported_codecs.push_back(codec);
- }
-
- if (external_encoder_factory_ == NULL) {
- continue;
- }
- const std::vector<WebRtcVideoEncoderFactory::VideoCodec> external_codecs =
- external_encoder_factory_->codecs();
- for (size_t c = 0; c < external_codecs.size(); ++c) {
- if (CodecNameMatches(codec.codec.name, external_codecs[c].name)) {
- supported_codecs.push_back(codec);
- break;
- }
- }
- }
- return supported_codecs;
-}
-
} // namespace cricket
#endif // HAVE_WEBRTC_VIDEO
diff --git a/media/webrtc/webrtcvideoengine2.h b/media/webrtc/webrtcvideoengine2.h
index 8831a1b..0b812ef 100644
--- a/media/webrtc/webrtcvideoengine2.h
+++ b/media/webrtc/webrtcvideoengine2.h
@@ -211,7 +211,9 @@
public webrtc::LoadObserver {
public:
WebRtcVideoChannel2(WebRtcCallFactory* call_factory,
+ WebRtcVoiceEngine* voice_engine,
VoiceMediaChannel* voice_channel,
+ const VideoOptions& options,
WebRtcVideoEncoderFactory* external_encoder_factory,
WebRtcVideoDecoderFactory* external_decoder_factory,
WebRtcVideoEncoderFactory2* encoder_factory);
@@ -274,6 +276,7 @@
private:
void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config,
const StreamParams& sp) const;
+ bool CodecIsExternallySupported(const std::string& name) const;
struct VideoCodecSettings {
VideoCodecSettings();
@@ -345,6 +348,13 @@
bool external;
};
+ struct LastDimensions {
+ LastDimensions() : width(-1), height(-1), is_screencast(false) {}
+ int width;
+ int height;
+ bool is_screencast;
+ };
+
AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec)
EXCLUSIVE_LOCKS_REQUIRED(lock_);
void DestroyVideoEncoder(AllocatedEncoder* encoder);
@@ -352,8 +362,7 @@
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)
+ void SetDimensions(int width, int height, bool is_screencast)
EXCLUSIVE_LOCKS_REQUIRED(lock_);
webrtc::Call* const call_;
@@ -364,6 +373,7 @@
webrtc::VideoSendStream* stream_ GUARDED_BY(lock_);
VideoSendStreamParameters parameters_ GUARDED_BY(lock_);
AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_);
+ LastDimensions last_dimensions_ GUARDED_BY(lock_);
VideoCapturer* capturer_ GUARDED_BY(lock_);
bool sending_ GUARDED_BY(lock_);
@@ -381,6 +391,7 @@
public:
WebRtcVideoReceiveStream(
webrtc::Call*,
+ WebRtcVideoDecoderFactory* external_decoder_factory,
const webrtc::VideoReceiveStream::Config& config,
const std::vector<VideoCodecSettings>& recv_codecs);
~WebRtcVideoReceiveStream();
@@ -397,14 +408,32 @@
VideoReceiverInfo GetVideoReceiverInfo();
private:
+ struct AllocatedDecoder {
+ AllocatedDecoder(webrtc::VideoDecoder* decoder,
+ webrtc::VideoCodecType type,
+ bool external)
+ : decoder(decoder), type(type), external(external) {}
+ webrtc::VideoDecoder* decoder;
+ webrtc::VideoCodecType type;
+ bool external;
+ };
+
void SetSize(int width, int height);
void RecreateWebRtcStream();
+ AllocatedDecoder CreateOrReuseVideoDecoder(
+ std::vector<AllocatedDecoder>* old_decoder,
+ const VideoCodec& codec);
+ void ClearDecoders(std::vector<AllocatedDecoder>* allocated_decoders);
+
webrtc::Call* const call_;
webrtc::VideoReceiveStream* stream_;
webrtc::VideoReceiveStream::Config config_;
+ WebRtcVideoDecoderFactory* const external_decoder_factory_;
+ std::vector<AllocatedDecoder> allocated_decoders_;
+
rtc::CriticalSection renderer_lock_;
cricket::VideoRenderer* renderer_ GUARDED_BY(renderer_lock_);
int last_width_ GUARDED_BY(renderer_lock_);
@@ -423,7 +452,7 @@
static std::vector<VideoCodecSettings> MapCodecs(
const std::vector<VideoCodec>& codecs);
std::vector<VideoCodecSettings> FilterSupportedCodecs(
- const std::vector<VideoCodecSettings>& mapped_codecs);
+ const std::vector<VideoCodecSettings>& mapped_codecs) const;
void FillSenderStats(VideoMediaInfo* info);
void FillReceiverStats(VideoMediaInfo* info);
@@ -449,6 +478,7 @@
Settable<VideoCodecSettings> send_codec_;
std::vector<webrtc::RtpExtension> send_rtp_extensions_;
+ VoiceMediaChannel* const voice_channel_;
WebRtcVideoEncoderFactory* const external_encoder_factory_;
WebRtcVideoDecoderFactory* const external_decoder_factory_;
WebRtcVideoEncoderFactory2* const encoder_factory_;
diff --git a/media/webrtc/webrtcvideoengine2_unittest.cc b/media/webrtc/webrtcvideoengine2_unittest.cc
index 85b7b9e..0b85723 100644
--- a/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -34,6 +34,7 @@
#include "talk/media/webrtc/webrtcvideochannelfactory.h"
#include "talk/media/webrtc/webrtcvideoengine2.h"
#include "talk/media/webrtc/webrtcvideoengine2_unittest.h"
+#include "talk/media/webrtc/webrtcvoiceengine.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/video_encoder.h"
@@ -79,10 +80,14 @@
ReconfigureVideoEncoder(encoder_config);
}
-webrtc::VideoSendStream::Config FakeVideoSendStream::GetConfig() {
+webrtc::VideoSendStream::Config FakeVideoSendStream::GetConfig() const {
return config_;
}
+webrtc::VideoEncoderConfig FakeVideoSendStream::GetEncoderConfig() const {
+ return encoder_config_;
+}
+
std::vector<webrtc::VideoStream> FakeVideoSendStream::GetVideoStreams() {
return encoder_config_.streams;
}
@@ -170,9 +175,6 @@
receiving_ = false;
}
-void FakeVideoReceiveStream::GetCurrentReceiveCodec(webrtc::VideoCodec* codec) {
-}
-
FakeCall::FakeCall(const webrtc::Call::Config& config)
: config_(config), network_state_(kNetworkUp) {
SetVideoCodecs(GetDefaultVideoCodecs());
@@ -327,9 +329,32 @@
}
protected:
+ class FakeCallFactory : public WebRtcCallFactory {
+ public:
+ FakeCallFactory() : fake_call_(NULL) {}
+ FakeCall* GetCall() { return fake_call_; }
+
+ private:
+ virtual webrtc::Call* CreateCall(
+ const webrtc::Call::Config& config) OVERRIDE {
+ assert(fake_call_ == NULL);
+ fake_call_ = new FakeCall(config);
+ return fake_call_;
+ }
+
+ FakeCall* fake_call_;
+ };
+
VideoMediaChannel* SetUpForExternalEncoderFactory(
cricket::WebRtcVideoEncoderFactory* encoder_factory,
const std::vector<VideoCodec>& codecs);
+
+ VideoMediaChannel* SetUpForExternalDecoderFactory(
+ cricket::WebRtcVideoDecoderFactory* decoder_factory,
+ const std::vector<VideoCodec>& codecs);
+
+ void TestStartBitrate(bool override_start_bitrate, int start_bitrate_bps);
+
WebRtcVideoEngine2 engine_;
VideoCodec default_codec_;
VideoCodec default_red_codec_;
@@ -337,9 +362,42 @@
VideoCodec default_rtx_codec_;
};
-// TODO(pbos): Add test that verifies that sync is configured properly.
-TEST_F(WebRtcVideoEngine2Test, DISABLED_CreateChannelWithVoiceEngine) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
+TEST_F(WebRtcVideoEngine2Test, ConfiguresAvSyncForFirstReceiveChannel) {
+ FakeCallFactory call_factory;
+ engine_.SetCallFactory(&call_factory);
+
+ WebRtcVoiceEngine voice_engine;
+ engine_.SetVoiceEngine(&voice_engine);
+ voice_engine.Init(rtc::Thread::Current());
+ engine_.Init(rtc::Thread::Current());
+
+ rtc::scoped_ptr<VoiceMediaChannel> voice_channel(
+ voice_engine.CreateChannel());
+ ASSERT_TRUE(voice_channel.get() != NULL);
+ WebRtcVoiceMediaChannel* webrtc_voice_channel =
+ static_cast<WebRtcVoiceMediaChannel*>(voice_channel.get());
+ ASSERT_NE(webrtc_voice_channel->voe_channel(), -1);
+ rtc::scoped_ptr<VideoMediaChannel> channel(
+ engine_.CreateChannel(cricket::VideoOptions(), voice_channel.get()));
+
+ FakeCall* fake_call = call_factory.GetCall();
+ ASSERT_TRUE(fake_call != NULL);
+
+ webrtc::Call::Config call_config = fake_call->GetConfig();
+
+ ASSERT_TRUE(voice_engine.voe()->engine() != NULL);
+ ASSERT_EQ(voice_engine.voe()->engine(), call_config.voice_engine);
+
+ EXPECT_TRUE(channel->AddRecvStream(StreamParams::CreateLegacy(kSsrc)));
+ EXPECT_TRUE(channel->AddRecvStream(StreamParams::CreateLegacy(kSsrc + 1)));
+ std::vector<FakeVideoReceiveStream*> receive_streams =
+ fake_call->GetVideoReceiveStreams();
+
+ ASSERT_EQ(2u, receive_streams.size());
+ EXPECT_EQ(webrtc_voice_channel->voe_channel(),
+ receive_streams[0]->GetConfig().audio_channel_id);
+ EXPECT_EQ(-1, receive_streams[1]->GetConfig().audio_channel_id)
+ << "AV sync should only be set up for the first receive channel.";
}
TEST_F(WebRtcVideoEngine2Test, FindCodec) {
@@ -425,6 +483,39 @@
FAIL() << "Absolute Sender Time extension not in header-extension list.";
}
+void WebRtcVideoEngine2Test::TestStartBitrate(bool override_start_bitrate,
+ int start_bitrate_bps) {
+ FakeCallFactory call_factory;
+ engine_.SetCallFactory(&call_factory);
+
+ engine_.Init(rtc::Thread::Current());
+
+ cricket::VideoOptions options;
+ if (override_start_bitrate) {
+ options.video_start_bitrate.Set(start_bitrate_bps / 1000);
+ }
+
+ rtc::scoped_ptr<VideoMediaChannel> channel(
+ engine_.CreateChannel(options, NULL));
+
+ EXPECT_EQ(override_start_bitrate
+ ? start_bitrate_bps
+ : webrtc::Call::Config::kDefaultStartBitrateBps,
+ call_factory.GetCall()->GetConfig().stream_start_bitrate_bps);
+}
+
+TEST_F(WebRtcVideoEngine2Test, UsesCorrectDefaultStartBitrate) {
+ TestStartBitrate(false, -1);
+}
+
+TEST_F(WebRtcVideoEngine2Test, CreateChannelCanUseIncreasedStartBitrate) {
+ TestStartBitrate(true, 2 * webrtc::Call::Config::kDefaultStartBitrateBps);
+}
+
+TEST_F(WebRtcVideoEngine2Test, CreateChannelCanUseDecreasedStartBitrate) {
+ TestStartBitrate(true, webrtc::Call::Config::kDefaultStartBitrateBps / 2);
+}
+
TEST_F(WebRtcVideoEngine2Test, SetSendFailsBeforeSettingCodecs) {
engine_.Init(rtc::Thread::Current());
rtc::scoped_ptr<VideoMediaChannel> channel(
@@ -492,6 +583,19 @@
return channel;
}
+VideoMediaChannel* WebRtcVideoEngine2Test::SetUpForExternalDecoderFactory(
+ cricket::WebRtcVideoDecoderFactory* decoder_factory,
+ const std::vector<VideoCodec>& codecs) {
+ engine_.SetExternalDecoderFactory(decoder_factory);
+ engine_.Init(rtc::Thread::Current());
+
+ VideoMediaChannel* channel =
+ engine_.CreateChannel(cricket::VideoOptions(), NULL);
+ EXPECT_TRUE(channel->SetRecvCodecs(codecs));
+
+ return channel;
+}
+
TEST_F(WebRtcVideoEngine2Test, ChannelWithExternalH264CanChangeToInternalVp8) {
cricket::FakeWebRtcVideoEncoderFactory encoder_factory;
encoder_factory.AddSupportedVideoCodecType(webrtc::kVideoCodecH264, "H264");
@@ -545,6 +649,50 @@
EXPECT_EQ("H264", external_codec.name);
}
+TEST_F(WebRtcVideoEngine2Test, RegisterExternalDecodersIfSupported) {
+ cricket::FakeWebRtcVideoDecoderFactory decoder_factory;
+ decoder_factory.AddSupportedVideoCodecType(webrtc::kVideoCodecVP8);
+ std::vector<cricket::VideoCodec> codecs;
+ codecs.push_back(kVp8Codec);
+
+ rtc::scoped_ptr<VideoMediaChannel> channel(
+ SetUpForExternalDecoderFactory(&decoder_factory, codecs));
+
+ EXPECT_TRUE(
+ channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
+ ASSERT_EQ(1u, decoder_factory.decoders().size());
+
+ // Setting codecs of the same type should not reallocate the decoder.
+ EXPECT_TRUE(channel->SetRecvCodecs(codecs));
+ EXPECT_EQ(1, decoder_factory.GetNumCreatedDecoders());
+
+ // Remove stream previously added to free the external decoder instance.
+ EXPECT_TRUE(channel->RemoveRecvStream(kSsrc));
+ EXPECT_EQ(0u, decoder_factory.decoders().size());
+}
+
+// Verifies that we can set up decoders that are not internally supported.
+TEST_F(WebRtcVideoEngine2Test, RegisterExternalH264DecoderIfSupported) {
+ // TODO(pbos): Do not assume that encoder/decoder support is symmetric. We
+ // can't even query the WebRtcVideoDecoderFactory for supported codecs.
+ // For now we add a FakeWebRtcVideoEncoderFactory to add H264 to supported
+ // codecs.
+ cricket::FakeWebRtcVideoEncoderFactory encoder_factory;
+ encoder_factory.AddSupportedVideoCodecType(webrtc::kVideoCodecH264, "H264");
+ engine_.SetExternalEncoderFactory(&encoder_factory);
+ cricket::FakeWebRtcVideoDecoderFactory decoder_factory;
+ decoder_factory.AddSupportedVideoCodecType(webrtc::kVideoCodecH264);
+ std::vector<cricket::VideoCodec> codecs;
+ codecs.push_back(kH264Codec);
+
+ rtc::scoped_ptr<VideoMediaChannel> channel(
+ SetUpForExternalDecoderFactory(&decoder_factory, codecs));
+
+ EXPECT_TRUE(
+ channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
+ ASSERT_EQ(1u, decoder_factory.decoders().size());
+}
+
class WebRtcVideoEngine2BaseTest
: public VideoEngineTest<cricket::WebRtcVideoEngine2> {
protected:
@@ -670,7 +818,7 @@
EXPECT_TRUE(channel_->SetCapturer(kSsrc, video_capturer_.get()));
// Capture format HD -> adapt (OnOutputFormatRequest VGA) -> VGA.
- cricket::VideoCodec codec(100, "VP8", 1280, 720, 30, 0);
+ cricket::VideoCodec codec = kVp8Codec720p;
EXPECT_TRUE(SetOneCodec(codec));
codec.width /= 2;
codec.height /= 2;
@@ -841,60 +989,6 @@
uint32 last_ssrc_;
};
-TEST_F(WebRtcVideoChannel2Test, DISABLED_MaxBitrateResetsWithConferenceMode) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
-}
-
-TEST_F(WebRtcVideoChannel2Test, DISABLED_StartSendBitrate) {
- // TODO(pbos): Is this test testing vie_ ? this is confusing. No API to set
- // start send bitrate from outside? Add defaults here that should be kept?
- std::vector<cricket::VideoCodec> codec_list;
- codec_list.push_back(kVp8Codec);
- EXPECT_TRUE(channel_->SetSendCodecs(codec_list));
- const unsigned int kVideoMinSendBitrateKbps = 50;
- const unsigned int kVideoTargetSendBitrateKbps = 300;
- const unsigned int kVideoMaxSendBitrateKbps = 2000;
- FakeVideoSendStream* stream = AddSendStream();
- std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
- ASSERT_EQ(1u, video_streams.size());
- EXPECT_EQ(kVideoMinSendBitrateKbps,
- video_streams.back().min_bitrate_bps / 1000);
- EXPECT_EQ(kVideoTargetSendBitrateKbps,
- video_streams.back().target_bitrate_bps / 1000);
- EXPECT_EQ(kVideoMaxSendBitrateKbps,
- video_streams.back().max_bitrate_bps / 1000);
-#if 0
- // TODO(pbos): un-#if
- VerifyVP8SendCodec(send_channel, kVP8Codec.width, kVP8Codec.height, 0,
- kVideoMaxSendBitrateKbps, kVideoMinSendBitrateKbps,
- kVideoDefaultStartSendBitrateKbps);
- EXPECT_EQ(0, vie_.StartSend(send_channel));
-
- // Increase the send bitrate and verify it is used as start bitrate.
- const unsigned int kVideoSendBitrateBps = 768000;
- vie_.SetSendBitrates(send_channel, kVideoSendBitrateBps, 0, 0);
- EXPECT_TRUE(channel_->SetSendCodecs(codec_list));
- VerifyVP8SendCodec(send_channel, kVP8Codec.width, kVP8Codec.height, 0,
- kVideoMaxSendBitrateKbps, kVideoMinSendBitrateKbps,
- kVideoSendBitrateBps / 1000);
-
- // Never set a start bitrate higher than the max bitrate.
- vie_.SetSendBitrates(send_channel, kVideoMaxSendBitrateKbps + 500, 0, 0);
- EXPECT_TRUE(channel_->SetSendCodecs(codec_list));
- VerifyVP8SendCodec(send_channel, kVP8Codec.width, kVP8Codec.height, 0,
- kVideoMaxSendBitrateKbps, kVideoMinSendBitrateKbps,
- kVideoDefaultStartSendBitrateKbps);
-
- // Use the default start bitrate if the send bitrate is lower.
- vie_.SetSendBitrates(send_channel, kVideoDefaultStartSendBitrateKbps - 50, 0,
- 0);
- EXPECT_TRUE(channel_->SetSendCodecs(codec_list));
- VerifyVP8SendCodec(send_channel, kVP8Codec.width, kVP8Codec.height, 0,
- kVideoMaxSendBitrateKbps, kVideoMinSendBitrateKbps,
- kVideoDefaultStartSendBitrateKbps);
-#endif
-}
-
TEST_F(WebRtcVideoChannel2Test, RecvStreamWithSimAndRtx) {
EXPECT_TRUE(channel_->SetSendCodecs(engine_.codecs()));
EXPECT_TRUE(channel_->SetSend(true));
@@ -1188,8 +1282,97 @@
FAIL() << "Not implemented."; // TODO(pbos): Implement.
}
-TEST_F(WebRtcVideoChannel2Test, DISABLED_SetBandwidthScreencast) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
+TEST_F(WebRtcVideoChannel2Test, UsesCorrectSettingsForScreencast) {
+ static const int kScreenshareMinBitrateKbps = 800;
+ cricket::VideoCodec codec = kVp8Codec360p;
+ std::vector<cricket::VideoCodec> codecs;
+ codecs.push_back(codec);
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ VideoOptions options;
+ options.screencast_min_bitrate.Set(kScreenshareMinBitrateKbps);
+ channel_->SetOptions(options);
+
+ AddSendStream();
+
+ cricket::FakeVideoCapturer capturer;
+ capturer.SetScreencast(false);
+ EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+ cricket::VideoFormat capture_format_hd =
+ capturer.GetSupportedFormats()->front();
+ EXPECT_EQ(1280, capture_format_hd.width);
+ EXPECT_EQ(720, capture_format_hd.height);
+ EXPECT_EQ(cricket::CS_RUNNING, capturer.Start(capture_format_hd));
+
+ EXPECT_TRUE(channel_->SetSend(true));
+
+ EXPECT_TRUE(capturer.CaptureFrame());
+ ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size());
+ FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
+
+ EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames());
+
+ // Verify non-screencast settings.
+ webrtc::VideoEncoderConfig encoder_config = send_stream->GetEncoderConfig();
+ EXPECT_EQ(webrtc::VideoEncoderConfig::kRealtimeVideo,
+ encoder_config.content_type);
+ EXPECT_EQ(codec.width, encoder_config.streams.front().width);
+ EXPECT_EQ(codec.height, encoder_config.streams.front().height);
+ EXPECT_EQ(0, encoder_config.min_transmit_bitrate_bps)
+ << "Non-screenshare shouldn't use min-transmit bitrate.";
+
+ capturer.SetScreencast(true);
+ EXPECT_TRUE(capturer.CaptureFrame());
+
+ EXPECT_EQ(2, send_stream->GetNumberOfSwappedFrames());
+
+ // Verify screencast settings.
+ encoder_config = send_stream->GetEncoderConfig();
+ EXPECT_EQ(webrtc::VideoEncoderConfig::kScreenshare,
+ encoder_config.content_type);
+ EXPECT_EQ(kScreenshareMinBitrateKbps * 1000,
+ encoder_config.min_transmit_bitrate_bps);
+
+ EXPECT_EQ(capture_format_hd.width, encoder_config.streams.front().width);
+ EXPECT_EQ(capture_format_hd.height, encoder_config.streams.front().height);
+ EXPECT_TRUE(encoder_config.streams[0].temporal_layer_thresholds_bps.empty());
+
+ EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
+}
+
+TEST_F(WebRtcVideoChannel2Test,
+ ConferenceModeScreencastConfiguresTemporalLayer) {
+ static const int kConferenceScreencastTemporalBitrateBps = 100000;
+ VideoOptions options;
+ options.conference_mode.Set(true);
+ channel_->SetOptions(options);
+
+ AddSendStream();
+
+ cricket::FakeVideoCapturer capturer;
+ capturer.SetScreencast(true);
+ EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+ cricket::VideoFormat capture_format_hd =
+ capturer.GetSupportedFormats()->front();
+ EXPECT_EQ(cricket::CS_RUNNING, capturer.Start(capture_format_hd));
+
+ EXPECT_TRUE(channel_->SetSend(true));
+
+ EXPECT_TRUE(capturer.CaptureFrame());
+ ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size());
+ FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
+
+ webrtc::VideoEncoderConfig encoder_config = send_stream->GetEncoderConfig();
+
+ // Verify screencast settings.
+ encoder_config = send_stream->GetEncoderConfig();
+ EXPECT_EQ(webrtc::VideoEncoderConfig::kScreenshare,
+ encoder_config.content_type);
+ ASSERT_EQ(1u, encoder_config.streams.size());
+ ASSERT_EQ(1u, encoder_config.streams[0].temporal_layer_thresholds_bps.size());
+ EXPECT_EQ(kConferenceScreencastTemporalBitrateBps,
+ encoder_config.streams[0].temporal_layer_thresholds_bps[0]);
+
+ EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
}
TEST_F(WebRtcVideoChannel2Test, DISABLED_SetSendSsrcAndCname) {
@@ -1289,7 +1472,7 @@
}
void WebRtcVideoChannel2Test::TestCpuAdaptation(bool enable_overuse) {
- cricket::VideoCodec codec(100, "VP8", 1280, 720, 30, 0);
+ cricket::VideoCodec codec = kVp8Codec720p;
std::vector<cricket::VideoCodec> codecs;
codecs.push_back(codec);
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
@@ -1451,13 +1634,13 @@
}
TEST_F(WebRtcVideoChannel2Test, SetSendCodecsWithMinMaxBitrate) {
- SetSendCodecsShouldWorkForBitrates("10", "20");
+ SetSendCodecsShouldWorkForBitrates("100", "200");
}
TEST_F(WebRtcVideoChannel2Test, SetSendCodecsRejectsMaxLessThanMinBitrate) {
std::vector<VideoCodec> video_codecs = engine_.codecs();
- video_codecs[0].params[kCodecParamMinBitrate] = "30";
- video_codecs[0].params[kCodecParamMaxBitrate] = "20";
+ video_codecs[0].params[kCodecParamMinBitrate] = "300";
+ video_codecs[0].params[kCodecParamMaxBitrate] = "200";
EXPECT_FALSE(channel_->SetSendCodecs(video_codecs));
}
@@ -1565,8 +1748,8 @@
FakeVideoReceiveStream* stream = AddRecvStream();
webrtc::VideoReceiveStream::Config config = stream->GetConfig();
- EXPECT_STREQ(engine_.codecs()[0].name.c_str(), config.codecs[0].plName);
- EXPECT_EQ(engine_.codecs()[0].id, config.codecs[0].plType);
+ EXPECT_EQ(engine_.codecs()[0].name, config.decoders[0].payload_name);
+ EXPECT_EQ(engine_.codecs()[0].id, config.decoders[0].payload_type);
}
TEST_F(WebRtcVideoChannel2Test, SetRecvCodecsRejectUnsupportedCodec) {
@@ -1678,55 +1861,6 @@
FAIL() << "Not implemented."; // TODO(pbos): Implement.
}
-TEST_F(WebRtcVideoChannel2Test, DISABLED_SetOptionsWithLoweredBitrate) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
-}
-
-TEST_F(WebRtcVideoChannel2Test, DISABLED_SetOptionsSucceedsWhenSending) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
-}
-
-TEST_F(WebRtcVideoChannel2Test, DISABLED_ResetCodecOnScreencast) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
-}
-
-TEST_F(WebRtcVideoChannel2Test, DISABLED_DontResetCodecOnSendFrame) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
-}
-
-TEST_F(WebRtcVideoChannel2Test,
- DISABLED_DontRegisterDecoderIfFactoryIsNotGiven) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
-}
-
-TEST_F(WebRtcVideoChannel2Test, DISABLED_RegisterDecoderIfFactoryIsGiven) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
-}
-
-TEST_F(WebRtcVideoChannel2Test, DISABLED_DontRegisterDecoderMultipleTimes) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
-}
-
-TEST_F(WebRtcVideoChannel2Test, DISABLED_DontRegisterDecoderForNonVP8) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
-}
-
-TEST_F(WebRtcVideoChannel2Test, DISABLED_FeedbackParamsForNonVP8) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
-}
-
-TEST_F(WebRtcVideoChannel2Test, DISABLED_ExternalCodecAddedToTheEnd) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
-}
-
-TEST_F(WebRtcVideoChannel2Test, DISABLED_ExternalCodecIgnored) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
-}
-
-TEST_F(WebRtcVideoChannel2Test, DISABLED_UpdateEncoderCodecsAfterSetFactory) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
-}
-
TEST_F(WebRtcVideoChannel2Test, OnReadyToSendSignalsNetworkState) {
EXPECT_EQ(webrtc::Call::kNetworkUp, fake_call_->GetNetworkState());
@@ -1737,7 +1871,4 @@
EXPECT_EQ(webrtc::Call::kNetworkUp, fake_call_->GetNetworkState());
}
-TEST_F(WebRtcVideoChannel2Test, DISABLED_CaptureFrameTimestampToNtpTimestamp) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
-}
} // namespace cricket
diff --git a/media/webrtc/webrtcvideoengine2_unittest.h b/media/webrtc/webrtcvideoengine2_unittest.h
index b3014ad..3b62289 100644
--- a/media/webrtc/webrtcvideoengine2_unittest.h
+++ b/media/webrtc/webrtcvideoengine2_unittest.h
@@ -41,7 +41,8 @@
public:
FakeVideoSendStream(const webrtc::VideoSendStream::Config& config,
const webrtc::VideoEncoderConfig& encoder_config);
- webrtc::VideoSendStream::Config GetConfig();
+ webrtc::VideoSendStream::Config GetConfig() const;
+ webrtc::VideoEncoderConfig GetEncoderConfig() const;
std::vector<webrtc::VideoStream> GetVideoStreams();
bool IsSending() const;
@@ -86,7 +87,6 @@
virtual void Start() OVERRIDE;
virtual void Stop() OVERRIDE;
- virtual void GetCurrentReceiveCodec(webrtc::VideoCodec* codec);
webrtc::VideoReceiveStream::Config config_;
bool receiving_;
diff --git a/media/webrtc/webrtcvideoengine_unittest.cc b/media/webrtc/webrtcvideoengine_unittest.cc
index 4060493..00d6e4b 100644
--- a/media/webrtc/webrtcvideoengine_unittest.cc
+++ b/media/webrtc/webrtcvideoengine_unittest.cc
@@ -1475,7 +1475,7 @@
kMaxBandwidthKbps, kMinBandwidthKbps, kStartBandwidthKbps);
// Set the start bitrate option.
- int kBoostedStartBandwidthKbps = 1000;
+ unsigned int kBoostedStartBandwidthKbps = 1000;
ASSERT_NE(kStartBandwidthKbps, kBoostedStartBandwidthKbps);
cricket::VideoOptions options;
options.video_start_bitrate.Set(kBoostedStartBandwidthKbps);
diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc
index f9e7878..95e16e4 100644
--- a/media/webrtc/webrtcvoiceengine.cc
+++ b/media/webrtc/webrtcvoiceengine.cc
@@ -110,13 +110,26 @@
static const char kIsacCodecName[] = "ISAC";
static const char kL16CodecName[] = "L16";
-// Codec parameters for Opus.
-static const int kOpusMonoBitrate = 32000;
+
// Parameter used for NACK.
// This value is equivalent to 5 seconds of audio data at 20 ms per packet.
static const int kNackMaxPackets = 250;
-static const int kOpusStereoBitrate = 64000;
+
+// Codec parameters for Opus.
// draft-spittka-payload-rtp-opus-03
+
+// Recommended bitrates:
+// 8-12 kb/s for NB speech,
+// 16-20 kb/s for WB speech,
+// 28-40 kb/s for FB speech,
+// 48-64 kb/s for FB mono music, and
+// 64-128 kb/s for FB stereo music.
+// The current implementation applies the following values to mono signals,
+// and multiplies them by 2 for stereo.
+static const int kOpusBitrateNb = 12000;
+static const int kOpusBitrateWb = 20000;
+static const int kOpusBitrateFb = 32000;
+
// Opus bitrate should be in the range between 6000 and 510000.
static const int kOpusMinBitrate = 6000;
static const int kOpusMaxBitrate = 510000;
@@ -405,22 +418,37 @@
return codec.GetParam(kCodecParamStereo, &value) && value == 1;
}
-// TODO(minyue): Clamp bitrate when invalid.
-static bool IsValidOpusBitrate(int bitrate) {
- return (bitrate >= kOpusMinBitrate && bitrate <= kOpusMaxBitrate);
-}
-
-// Returns 0 if params[kCodecParamMaxAverageBitrate] is not defined or invalid.
-// Returns the value of params[kCodecParamMaxAverageBitrate] otherwise.
-static int GetOpusBitrateFromParams(const AudioCodec& codec) {
+// Use params[kCodecParamMaxAverageBitrate] if it is defined, use codec.bitrate
+// otherwise. If the value (either from params or codec.bitrate) <=0, use the
+// default configuration. If the value is beyond feasible bit rate of Opus,
+// clamp it. Returns the Opus bit rate for operation.
+static int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) {
int bitrate = 0;
+ bool use_param = true;
if (!codec.GetParam(kCodecParamMaxAverageBitrate, &bitrate)) {
- return 0;
+ bitrate = codec.bitrate;
+ use_param = false;
}
- if (!IsValidOpusBitrate(bitrate)) {
- LOG(LS_WARNING) << "Codec parameter \"maxaveragebitrate\" has an "
- << "invalid value: " << bitrate;
- return 0;
+ if (bitrate <= 0) {
+ if (max_playback_rate <= 8000) {
+ bitrate = kOpusBitrateNb;
+ } else if (max_playback_rate <= 16000) {
+ bitrate = kOpusBitrateWb;
+ } else {
+ bitrate = kOpusBitrateFb;
+ }
+
+ if (IsOpusStereoEnabled(codec)) {
+ bitrate *= 2;
+ }
+ } else if (bitrate < kOpusMinBitrate || bitrate > kOpusMaxBitrate) {
+ bitrate = (bitrate < kOpusMinBitrate) ? kOpusMinBitrate : kOpusMaxBitrate;
+ std::string rate_source =
+ use_param ? "Codec parameter \"maxaveragebitrate\"" :
+ "Supplied Opus bitrate";
+ LOG(LS_WARNING) << rate_source
+ << " is invalid and is replaced by: "
+ << bitrate;
}
return bitrate;
}
@@ -450,39 +478,11 @@
// 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
+ // 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;
- }
+ voe_codec->channels = IsOpusStereoEnabled(codec) ? 2 : 1;
+ voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate);
}
void WebRtcVoiceEngine::ConstructCodecs() {
@@ -514,7 +514,7 @@
ARRAY_SIZE(kCodecPrefs) - (pref - kCodecPrefs));
LOG(LS_INFO) << ToString(codec);
if (IsIsac(codec)) {
- // Indicate auto-bandwidth in signaling.
+ // Indicate auto-bitrate in signaling.
codec.bitrate = 0;
}
if (IsOpus(codec)) {
@@ -1246,7 +1246,7 @@
// Apply codec-specific settings.
if (IsIsac(codec)) {
// If ISAC and an explicit bitrate is not specified,
- // enable auto bandwidth adjustment.
+ // enable auto bitrate adjustment.
voe_codec.rate = (in.bitrate > 0) ? in.bitrate : -1;
}
*out = voe_codec;
@@ -1811,8 +1811,8 @@
: WebRtcMediaChannel<VoiceMediaChannel, WebRtcVoiceEngine>(
engine,
engine->CreateMediaVoiceChannel()),
- send_bw_setting_(false),
- send_bw_bps_(0),
+ send_bitrate_setting_(false),
+ send_bitrate_bps_(0),
options_(),
dtmf_allowed_(false),
desired_playout_(false),
@@ -2047,9 +2047,7 @@
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;
+ int opus_max_playback_rate = 0;
// Set send codec (the first non-telephone-event/CN codec)
for (std::vector<AudioCodec>::const_iterator it = codecs.begin();
@@ -2067,7 +2065,6 @@
continue;
}
-
// 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.
// "red", for RED audio, is a special case where the actual codec to be
@@ -2099,7 +2096,8 @@
// 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);
+ GetOpusConfig(*it, &send_codec, &enable_codec_fec,
+ &opus_max_playback_rate);
}
}
found_send_codec = true;
@@ -2135,15 +2133,16 @@
}
// maxplaybackrate should be set after SetSendCodec.
- if (max_playback_rate > 0) {
+ // If opus_max_playback_rate <= 0, the default maximum playback rate of 48 kHz
+ // will be used.
+ if (opus_max_playback_rate > 0) {
LOG(LS_INFO) << "Attempt to set maximum playback rate to "
- << max_playback_rate
+ << opus_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) {
+ channel, opus_max_playback_rate) == -1) {
LOG(LS_WARNING) << "Could not set maximum playback rate.";
}
#endif
@@ -2152,8 +2151,8 @@
// Always update the |send_codec_| to the currently set send codec.
send_codec_.reset(new webrtc::CodecInst(send_codec));
- if (send_bw_setting_) {
- SetSendBandwidthInternal(send_bw_bps_);
+ if (send_bitrate_setting_) {
+ SetSendBitrateInternal(send_bitrate_bps_);
}
// Loop through the codecs list again to config the telephone-event/CN codec.
@@ -3206,25 +3205,27 @@
return true;
}
+// TODO(minyue): SetMaxSendBandwidth() is subject to be renamed to
+// SetMaxSendBitrate() in future.
bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) {
- LOG(LS_INFO) << "WebRtcVoiceMediaChanne::SetSendBandwidth.";
+ LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBandwidth.";
- return SetSendBandwidthInternal(bps);
+ return SetSendBitrateInternal(bps);
}
-bool WebRtcVoiceMediaChannel::SetSendBandwidthInternal(int bps) {
- LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBandwidthInternal.";
+bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
+ LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBitrateInternal.";
- send_bw_setting_ = true;
- send_bw_bps_ = bps;
+ send_bitrate_setting_ = true;
+ send_bitrate_bps_ = bps;
if (!send_codec_) {
LOG(LS_INFO) << "The send codec has not been set up yet. "
- << "The send bandwidth setting will be applied later.";
+ << "The send bitrate setting will be applied later.";
return true;
}
- // Bandwidth is auto by default.
+ // Bitrate is auto by default.
// TODO(bemasc): Fix this so that if SetMaxSendBandwidth(50) is followed by
// SetMaxSendBandwith(0), the second call removes the previous limit.
if (bps <= 0)
diff --git a/media/webrtc/webrtcvoiceengine.h b/media/webrtc/webrtcvoiceengine.h
index 8d762d4..f19059b 100644
--- a/media/webrtc/webrtcvoiceengine.h
+++ b/media/webrtc/webrtcvoiceengine.h
@@ -435,7 +435,7 @@
return channel_id == voe_channel();
}
bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
- bool SetSendBandwidthInternal(int bps);
+ bool SetSendBitrateInternal(int bps);
bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
const RtpHeaderExtension* extension);
@@ -453,8 +453,8 @@
std::vector<AudioCodec> recv_codecs_;
std::vector<AudioCodec> send_codecs_;
rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
- bool send_bw_setting_;
- int send_bw_bps_;
+ bool send_bitrate_setting_;
+ int send_bitrate_bps_;
AudioOptions options_;
bool dtmf_allowed_;
bool desired_playout_;
diff --git a/media/webrtc/webrtcvoiceengine_unittest.cc b/media/webrtc/webrtcvoiceengine_unittest.cc
index b044e92..5deabd2 100644
--- a/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -40,7 +40,7 @@
#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 "webrtc/p2p/base/fakesession.h"
#include "talk/session/media/channel.h"
// Tests for the WebRtcVoiceEngine/VoiceChannel code.
@@ -876,21 +876,20 @@
codecs[0].params["stereo"] = "0";
webrtc::CodecInst gcodec;
- // bitrate that's out of the range between 6000 and 510000 will be considered
- // as invalid and ignored.
+ // bitrate that's out of the range between 6000 and 510000 will be clamped.
codecs[0].bitrate = 5999;
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
EXPECT_STREQ("opus", gcodec.plname);
EXPECT_EQ(1, gcodec.channels);
- EXPECT_EQ(32000, gcodec.rate);
+ EXPECT_EQ(6000, gcodec.rate);
codecs[0].bitrate = 510001;
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
EXPECT_STREQ("opus", gcodec.plname);
EXPECT_EQ(1, gcodec.channels);
- EXPECT_EQ(32000, gcodec.rate);
+ EXPECT_EQ(510000, gcodec.rate);
}
// Test that with bitrate=0 and stereo=1,
@@ -920,21 +919,20 @@
codecs[0].params["stereo"] = "1";
webrtc::CodecInst gcodec;
- // bitrate that's out of the range between 6000 and 510000 will be considered
- // as invalid and ignored.
+ // bitrate that's out of the range between 6000 and 510000 will be clamped.
codecs[0].bitrate = 5999;
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
EXPECT_STREQ("opus", gcodec.plname);
EXPECT_EQ(2, gcodec.channels);
- EXPECT_EQ(64000, gcodec.rate);
+ EXPECT_EQ(6000, gcodec.rate);
codecs[0].bitrate = 510001;
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
EXPECT_STREQ("opus", gcodec.plname);
EXPECT_EQ(2, gcodec.channels);
- EXPECT_EQ(64000, gcodec.rate);
+ EXPECT_EQ(510000, gcodec.rate);
}
// Test that with bitrate=N and stereo unset,
@@ -1020,13 +1018,13 @@
codecs[0].params["maxaveragebitrate"] = "5999";
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
- EXPECT_EQ(30000, gcodec.rate);
+ EXPECT_EQ(6000, gcodec.rate);
// Ignore if larger than 510000.
codecs[0].params["maxaveragebitrate"] = "510001";
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
- EXPECT_EQ(30000, gcodec.rate);
+ EXPECT_EQ(510000, gcodec.rate);
codecs[0].params["maxaveragebitrate"] = "200000";
EXPECT_TRUE(channel_->SetSendCodecs(codecs));
@@ -1256,9 +1254,12 @@
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);
+
+ EXPECT_EQ(12000, gcodec.rate);
+ codecs[0].SetParam(cricket::kCodecParamStereo, "1");
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+ EXPECT_EQ(24000, gcodec.rate);
}
// Test 8000 < maxplaybackrate <= 12000 triggers Opus medium band mode.
@@ -1275,9 +1276,12 @@
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);
+
+ EXPECT_EQ(20000, gcodec.rate);
+ codecs[0].SetParam(cricket::kCodecParamStereo, "1");
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+ EXPECT_EQ(40000, gcodec.rate);
}
// Test 12000 < maxplaybackrate <= 16000 triggers Opus wide band mode.
@@ -1294,9 +1298,12 @@
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);
+
+ EXPECT_EQ(20000, gcodec.rate);
+ codecs[0].SetParam(cricket::kCodecParamStereo, "1");
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+ EXPECT_EQ(40000, gcodec.rate);
}
// Test 16000 < maxplaybackrate <= 24000 triggers Opus super wide band mode.
@@ -1313,9 +1320,12 @@
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);
+ codecs[0].SetParam(cricket::kCodecParamStereo, "1");
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+ EXPECT_EQ(64000, gcodec.rate);
}
// Test 24000 < maxplaybackrate triggers Opus full band mode.
@@ -1332,9 +1342,12 @@
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);
+ codecs[0].SetParam(cricket::kCodecParamStereo, "1");
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+ EXPECT_EQ(64000, gcodec.rate);
}
// Test Opus that without maxplaybackrate, default playback rate is used.
diff --git a/p2p/base/asyncstuntcpsocket.cc b/p2p/base/asyncstuntcpsocket.cc
index c1aeead..73bb152 100644
--- a/p2p/base/asyncstuntcpsocket.cc
+++ b/p2p/base/asyncstuntcpsocket.cc
@@ -25,11 +25,11 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/asyncstuntcpsocket.h"
+#include "webrtc/p2p/base/asyncstuntcpsocket.h"
#include <string.h>
-#include "talk/p2p/base/stun.h"
+#include "webrtc/p2p/base/stun.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
diff --git a/p2p/base/asyncstuntcpsocket.h b/p2p/base/asyncstuntcpsocket.h
index 136b4df..3675c9e 100644
--- a/p2p/base/asyncstuntcpsocket.h
+++ b/p2p/base/asyncstuntcpsocket.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_ASYNCSTUNTCPSOCKET_H_
-#define TALK_P2P_BASE_ASYNCSTUNTCPSOCKET_H_
+#ifndef WEBRTC_P2P_BASE_ASYNCSTUNTCPSOCKET_H_
+#define WEBRTC_P2P_BASE_ASYNCSTUNTCPSOCKET_H_
#include "webrtc/base/asynctcpsocket.h"
#include "webrtc/base/scoped_ptr.h"
@@ -64,4 +64,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_ASYNCSTUNTCPSOCKET_H_
+#endif // WEBRTC_P2P_BASE_ASYNCSTUNTCPSOCKET_H_
diff --git a/p2p/base/asyncstuntcpsocket_unittest.cc b/p2p/base/asyncstuntcpsocket_unittest.cc
index 46e5d63..ba407e7 100644
--- a/p2p/base/asyncstuntcpsocket_unittest.cc
+++ b/p2p/base/asyncstuntcpsocket_unittest.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/asyncstuntcpsocket.h"
+#include "webrtc/p2p/base/asyncstuntcpsocket.h"
#include "webrtc/base/asyncsocket.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/physicalsocketserver.h"
diff --git a/p2p/base/basicpacketsocketfactory.cc b/p2p/base/basicpacketsocketfactory.cc
index 2b3b06f..1e0ddf8 100644
--- a/p2p/base/basicpacketsocketfactory.cc
+++ b/p2p/base/basicpacketsocketfactory.cc
@@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/base/asyncstuntcpsocket.h"
-#include "talk/p2p/base/stun.h"
+#include "webrtc/p2p/base/asyncstuntcpsocket.h"
+#include "webrtc/p2p/base/stun.h"
#include "webrtc/base/asynctcpsocket.h"
#include "webrtc/base/asyncudpsocket.h"
#include "webrtc/base/logging.h"
diff --git a/p2p/base/basicpacketsocketfactory.h b/p2p/base/basicpacketsocketfactory.h
index 77b1652..f895efa 100644
--- a/p2p/base/basicpacketsocketfactory.h
+++ b/p2p/base/basicpacketsocketfactory.h
@@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_BASICPACKETSOCKETFACTORY_H_
-#define TALK_P2P_BASE_BASICPACKETSOCKETFACTORY_H_
+#ifndef WEBRTC_P2P_BASE_BASICPACKETSOCKETFACTORY_H_
+#define WEBRTC_P2P_BASE_BASICPACKETSOCKETFACTORY_H_
-#include "talk/p2p/base/packetsocketfactory.h"
+#include "webrtc/p2p/base/packetsocketfactory.h"
namespace rtc {
@@ -65,4 +65,4 @@
} // namespace rtc
-#endif // TALK_P2P_BASE_BASICPACKETSOCKETFACTORY_H_
+#endif // WEBRTC_P2P_BASE_BASICPACKETSOCKETFACTORY_H_
diff --git a/p2p/base/candidate.h b/p2p/base/candidate.h
index cb0b1bc..64872f3 100644
--- a/p2p/base/candidate.h
+++ b/p2p/base/candidate.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_CANDIDATE_H_
-#define TALK_P2P_BASE_CANDIDATE_H_
+#ifndef WEBRTC_P2P_BASE_CANDIDATE_H_
+#define WEBRTC_P2P_BASE_CANDIDATE_H_
#include <limits.h>
#include <math.h>
@@ -35,7 +35,7 @@
#include <sstream>
#include <string>
-#include "talk/p2p/base/constants.h"
+#include "webrtc/p2p/base/constants.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/socketaddress.h"
@@ -226,4 +226,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_CANDIDATE_H_
+#endif // WEBRTC_P2P_BASE_CANDIDATE_H_
diff --git a/p2p/base/common.h b/p2p/base/common.h
index a33e9e0..0aeb77e 100644
--- a/p2p/base/common.h
+++ b/p2p/base/common.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_COMMON_H_
-#define TALK_P2P_BASE_COMMON_H_
+#ifndef WEBRTC_P2P_BASE_COMMON_H_
+#define WEBRTC_P2P_BASE_COMMON_H_
#include "webrtc/base/logging.h"
@@ -34,4 +34,4 @@
#define LOG_J(sev, obj) LOG(sev) << "Jingle:" << obj->ToString() << ": "
#define LOG_JV(sev, obj) LOG_V(sev) << "Jingle:" << obj->ToString() << ": "
-#endif // TALK_P2P_BASE_COMMON_H_
+#endif // WEBRTC_P2P_BASE_COMMON_H_
diff --git a/p2p/base/constants.cc b/p2p/base/constants.cc
index 278a615..54cfb75 100644
--- a/p2p/base/constants.cc
+++ b/p2p/base/constants.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/constants.h"
+#include "webrtc/p2p/base/constants.h"
#include <string>
diff --git a/p2p/base/constants.h b/p2p/base/constants.h
index 4cd1166..9c53b7d 100644
--- a/p2p/base/constants.h
+++ b/p2p/base/constants.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_CONSTANTS_H_
-#define TALK_P2P_BASE_CONSTANTS_H_
+#ifndef WEBRTC_P2P_BASE_CONSTANTS_H_
+#define WEBRTC_P2P_BASE_CONSTANTS_H_
#include <string>
#include "webrtc/libjingle/xmllite/qname.h"
@@ -273,4 +273,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_CONSTANTS_H_
+#endif // WEBRTC_P2P_BASE_CONSTANTS_H_
diff --git a/p2p/base/dtlstransport.h b/p2p/base/dtlstransport.h
index 318c14a..66b08b4 100644
--- a/p2p/base/dtlstransport.h
+++ b/p2p/base/dtlstransport.h
@@ -25,11 +25,11 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_DTLSTRANSPORT_H_
-#define TALK_P2P_BASE_DTLSTRANSPORT_H_
+#ifndef WEBRTC_P2P_BASE_DTLSTRANSPORT_H_
+#define WEBRTC_P2P_BASE_DTLSTRANSPORT_H_
-#include "talk/p2p/base/dtlstransportchannel.h"
-#include "talk/p2p/base/transport.h"
+#include "webrtc/p2p/base/dtlstransportchannel.h"
+#include "webrtc/p2p/base/transport.h"
namespace rtc {
class SSLIdentity;
@@ -254,4 +254,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_DTLSTRANSPORT_H_
+#endif // WEBRTC_P2P_BASE_DTLSTRANSPORT_H_
diff --git a/p2p/base/dtlstransportchannel.cc b/p2p/base/dtlstransportchannel.cc
index e0f4141..78d4622 100644
--- a/p2p/base/dtlstransportchannel.cc
+++ b/p2p/base/dtlstransportchannel.cc
@@ -26,9 +26,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/dtlstransportchannel.h"
+#include "webrtc/p2p/base/dtlstransportchannel.h"
-#include "talk/p2p/base/common.h"
+#include "webrtc/p2p/base/common.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/dscp.h"
#include "webrtc/base/messagequeue.h"
diff --git a/p2p/base/dtlstransportchannel.h b/p2p/base/dtlstransportchannel.h
index 44769c9..6706013 100644
--- a/p2p/base/dtlstransportchannel.h
+++ b/p2p/base/dtlstransportchannel.h
@@ -26,13 +26,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_DTLSTRANSPORTCHANNEL_H_
-#define TALK_P2P_BASE_DTLSTRANSPORTCHANNEL_H_
+#ifndef WEBRTC_P2P_BASE_DTLSTRANSPORTCHANNEL_H_
+#define WEBRTC_P2P_BASE_DTLSTRANSPORTCHANNEL_H_
#include <string>
#include <vector>
-#include "talk/p2p/base/transportchannelimpl.h"
+#include "webrtc/p2p/base/transportchannelimpl.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/sslstreamadapter.h"
@@ -261,4 +261,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_DTLSTRANSPORTCHANNEL_H_
+#endif // WEBRTC_P2P_BASE_DTLSTRANSPORTCHANNEL_H_
diff --git a/p2p/base/dtlstransportchannel_unittest.cc b/p2p/base/dtlstransportchannel_unittest.cc
index 64825bc..44854c7 100644
--- a/p2p/base/dtlstransportchannel_unittest.cc
+++ b/p2p/base/dtlstransportchannel_unittest.cc
@@ -28,8 +28,8 @@
#include <set>
-#include "talk/p2p/base/dtlstransport.h"
-#include "talk/p2p/base/fakesession.h"
+#include "webrtc/p2p/base/dtlstransport.h"
+#include "webrtc/p2p/base/fakesession.h"
#include "webrtc/base/common.h"
#include "webrtc/base/dscp.h"
#include "webrtc/base/gunit.h"
diff --git a/p2p/base/fakesession.h b/p2p/base/fakesession.h
index 7c99719..cb9111f 100644
--- a/p2p/base/fakesession.h
+++ b/p2p/base/fakesession.h
@@ -25,17 +25,17 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_FAKESESSION_H_
-#define TALK_P2P_BASE_FAKESESSION_H_
+#ifndef WEBRTC_P2P_BASE_FAKESESSION_H_
+#define WEBRTC_P2P_BASE_FAKESESSION_H_
#include <map>
#include <string>
#include <vector>
-#include "talk/p2p/base/session.h"
-#include "talk/p2p/base/transport.h"
-#include "talk/p2p/base/transportchannel.h"
-#include "talk/p2p/base/transportchannelimpl.h"
+#include "webrtc/p2p/base/session.h"
+#include "webrtc/p2p/base/transport.h"
+#include "webrtc/p2p/base/transportchannel.h"
+#include "webrtc/p2p/base/transportchannelimpl.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/fakesslidentity.h"
#include "webrtc/base/messagequeue.h"
@@ -506,4 +506,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_FAKESESSION_H_
+#endif // WEBRTC_P2P_BASE_FAKESESSION_H_
diff --git a/p2p/base/p2ptransport.cc b/p2p/base/p2ptransport.cc
index 06941ac..bc316cf 100644
--- a/p2p/base/p2ptransport.cc
+++ b/p2p/base/p2ptransport.cc
@@ -25,19 +25,19 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/p2ptransport.h"
+#include "webrtc/p2p/base/p2ptransport.h"
#include <string>
#include <vector>
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/p2ptransportchannel.h"
-#include "talk/p2p/base/parsing.h"
-#include "talk/p2p/base/sessionmanager.h"
-#include "talk/p2p/base/sessionmessages.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/p2ptransportchannel.h"
+#include "webrtc/p2p/base/parsing.h"
+#include "webrtc/p2p/base/sessionmanager.h"
+#include "webrtc/p2p/base/sessionmessages.h"
#include "webrtc/libjingle/xmllite/qname.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/common.h"
#include "webrtc/base/stringencode.h"
diff --git a/p2p/base/p2ptransport.h b/p2p/base/p2ptransport.h
index 500bb9b..fc65bab 100644
--- a/p2p/base/p2ptransport.h
+++ b/p2p/base/p2ptransport.h
@@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_P2PTRANSPORT_H_
-#define TALK_P2P_BASE_P2PTRANSPORT_H_
+#ifndef WEBRTC_P2P_BASE_P2PTRANSPORT_H_
+#define WEBRTC_P2P_BASE_P2PTRANSPORT_H_
#include <string>
#include <vector>
-#include "talk/p2p/base/transport.h"
+#include "webrtc/p2p/base/transport.h"
namespace cricket {
@@ -100,4 +100,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_P2PTRANSPORT_H_
+#endif // WEBRTC_P2P_BASE_P2PTRANSPORT_H_
diff --git a/p2p/base/p2ptransportchannel.cc b/p2p/base/p2ptransportchannel.cc
index b1dc223..daedbeb 100644
--- a/p2p/base/p2ptransportchannel.cc
+++ b/p2p/base/p2ptransportchannel.cc
@@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/p2ptransportchannel.h"
+#include "webrtc/p2p/base/p2ptransportchannel.h"
#include <set>
-#include "talk/p2p/base/common.h"
-#include "talk/p2p/base/relayport.h" // For RELAY_PORT_TYPE.
-#include "talk/p2p/base/stunport.h" // For STUN_PORT_TYPE.
+#include "webrtc/p2p/base/common.h"
+#include "webrtc/p2p/base/relayport.h" // For RELAY_PORT_TYPE.
+#include "webrtc/p2p/base/stunport.h" // For STUN_PORT_TYPE.
#include "webrtc/base/common.h"
#include "webrtc/base/crc32.h"
#include "webrtc/base/logging.h"
diff --git a/p2p/base/p2ptransportchannel.h b/p2p/base/p2ptransportchannel.h
index 229e512..6d45999 100644
--- a/p2p/base/p2ptransportchannel.h
+++ b/p2p/base/p2ptransportchannel.h
@@ -34,18 +34,18 @@
// When all of the available connections become invalid (non-writable), we
// kick off a process of determining more candidates and more connections.
//
-#ifndef TALK_P2P_BASE_P2PTRANSPORTCHANNEL_H_
-#define TALK_P2P_BASE_P2PTRANSPORTCHANNEL_H_
+#ifndef WEBRTC_P2P_BASE_P2PTRANSPORTCHANNEL_H_
+#define WEBRTC_P2P_BASE_P2PTRANSPORTCHANNEL_H_
#include <map>
#include <string>
#include <vector>
-#include "talk/p2p/base/candidate.h"
-#include "talk/p2p/base/p2ptransport.h"
-#include "talk/p2p/base/portallocator.h"
-#include "talk/p2p/base/portinterface.h"
-#include "talk/p2p/base/transport.h"
-#include "talk/p2p/base/transportchannelimpl.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/p2ptransport.h"
+#include "webrtc/p2p/base/portallocator.h"
+#include "webrtc/p2p/base/portinterface.h"
+#include "webrtc/p2p/base/transport.h"
+#include "webrtc/p2p/base/transportchannelimpl.h"
#include "webrtc/base/asyncpacketsocket.h"
#include "webrtc/base/sigslot.h"
@@ -256,4 +256,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_P2PTRANSPORTCHANNEL_H_
+#endif // WEBRTC_P2P_BASE_P2PTRANSPORTCHANNEL_H_
diff --git a/p2p/base/p2ptransportchannel_unittest.cc b/p2p/base/p2ptransportchannel_unittest.cc
index 44c76ee..d7c0b9f 100644
--- a/p2p/base/p2ptransportchannel_unittest.cc
+++ b/p2p/base/p2ptransportchannel_unittest.cc
@@ -25,11 +25,11 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/p2ptransportchannel.h"
-#include "talk/p2p/base/testrelayserver.h"
-#include "talk/p2p/base/teststunserver.h"
-#include "talk/p2p/base/testturnserver.h"
-#include "talk/p2p/client/basicportallocator.h"
+#include "webrtc/p2p/base/p2ptransportchannel.h"
+#include "webrtc/p2p/base/testrelayserver.h"
+#include "webrtc/p2p/base/teststunserver.h"
+#include "webrtc/p2p/base/testturnserver.h"
+#include "webrtc/p2p/client/basicportallocator.h"
#include "webrtc/base/dscp.h"
#include "webrtc/base/fakenetwork.h"
#include "webrtc/base/firewallsocketserver.h"
diff --git a/p2p/base/packetsocketfactory.h b/p2p/base/packetsocketfactory.h
index 46767c2..0160cac 100644
--- a/p2p/base/packetsocketfactory.h
+++ b/p2p/base/packetsocketfactory.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_PACKETSOCKETFACTORY_H_
-#define TALK_P2P_BASE_PACKETSOCKETFACTORY_H_
+#ifndef WEBRTC_P2P_BASE_PACKETSOCKETFACTORY_H_
+#define WEBRTC_P2P_BASE_PACKETSOCKETFACTORY_H_
#include "webrtc/base/proxyinfo.h"
@@ -66,4 +66,4 @@
} // namespace rtc
-#endif // TALK_P2P_BASE_PACKETSOCKETFACTORY_H_
+#endif // WEBRTC_P2P_BASE_PACKETSOCKETFACTORY_H_
diff --git a/p2p/base/parsing.cc b/p2p/base/parsing.cc
index 1465096..eace88b 100644
--- a/p2p/base/parsing.cc
+++ b/p2p/base/parsing.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/parsing.h"
+#include "webrtc/p2p/base/parsing.h"
#include <stdlib.h>
#include <algorithm>
diff --git a/p2p/base/parsing.h b/p2p/base/parsing.h
index 2aab2f8..f150747 100644
--- a/p2p/base/parsing.h
+++ b/p2p/base/parsing.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_PARSING_H_
-#define TALK_P2P_BASE_PARSING_H_
+#ifndef WEBRTC_P2P_BASE_PARSING_H_
+#define WEBRTC_P2P_BASE_PARSING_H_
#include <string>
#include <vector>
@@ -154,4 +154,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_PARSING_H_
+#endif // WEBRTC_P2P_BASE_PARSING_H_
diff --git a/p2p/base/port.cc b/p2p/base/port.cc
index 0cf46eb..aa88872 100644
--- a/p2p/base/port.cc
+++ b/p2p/base/port.cc
@@ -25,12 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/port.h"
+#include "webrtc/p2p/base/port.h"
#include <algorithm>
#include <vector>
-#include "talk/p2p/base/common.h"
+#include "webrtc/p2p/base/common.h"
+#include "webrtc/p2p/base/portallocator.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/crc32.h"
#include "webrtc/base/helpers.h"
@@ -187,7 +188,8 @@
ice_protocol_(ICEPROTO_HYBRID),
ice_role_(ICEROLE_UNKNOWN),
tiebreaker_(0),
- shared_socket_(true) {
+ shared_socket_(true),
+ candidate_filter_(CF_ALL) {
Construct();
}
@@ -213,7 +215,8 @@
ice_protocol_(ICEPROTO_HYBRID),
ice_role_(ICEROLE_UNKNOWN),
tiebreaker_(0),
- shared_socket_(false) {
+ shared_socket_(false),
+ candidate_filter_(CF_ALL) {
ASSERT(factory_ != NULL);
Construct();
}
@@ -1334,7 +1337,7 @@
void Connection::OnMessage(rtc::Message *pmsg) {
ASSERT(pmsg->message_id == MSG_DELETE);
- LOG_J(LS_INFO, this) << "Connection deleted";
+ LOG_J(LS_INFO, this) << "Connection deleted due to read or write timeout";
SignalDestroyed(this);
delete this;
}
diff --git a/p2p/base/port.h b/p2p/base/port.h
index 4893586..490aee5 100644
--- a/p2p/base/port.h
+++ b/p2p/base/port.h
@@ -25,20 +25,20 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_PORT_H_
-#define TALK_P2P_BASE_PORT_H_
+#ifndef WEBRTC_P2P_BASE_PORT_H_
+#define WEBRTC_P2P_BASE_PORT_H_
#include <map>
#include <set>
#include <string>
#include <vector>
-#include "talk/p2p/base/candidate.h"
-#include "talk/p2p/base/packetsocketfactory.h"
-#include "talk/p2p/base/portinterface.h"
-#include "talk/p2p/base/stun.h"
-#include "talk/p2p/base/stunrequest.h"
-#include "talk/p2p/base/transport.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/packetsocketfactory.h"
+#include "webrtc/p2p/base/portinterface.h"
+#include "webrtc/p2p/base/stun.h"
+#include "webrtc/p2p/base/stunrequest.h"
+#include "webrtc/p2p/base/transport.h"
#include "webrtc/base/asyncpacketsocket.h"
#include "webrtc/base/network.h"
#include "webrtc/base/proxyinfo.h"
@@ -308,6 +308,10 @@
// Returns if Hybrid ICE protocol is used.
bool IsHybridIce() const;
+ void set_candidate_filter(uint32 candidate_filter) {
+ candidate_filter_ = candidate_filter;
+ }
+
protected:
enum {
MSG_CHECKTIMEOUT = 0,
@@ -351,6 +355,8 @@
return rtc::DSCP_NO_CHANGE;
}
+ uint32 candidate_filter() { return candidate_filter_; }
+
private:
void Construct();
// Called when one of our connections deletes itself.
@@ -392,6 +398,12 @@
std::string user_agent_;
rtc::ProxyInfo proxy_;
+ // Candidate filter is pushed down to Port such that each Port could
+ // make its own decision on how to create candidates. For example,
+ // when IceTransportsType is set to relay, both RelayPort and
+ // TurnPort will hide raddr to avoid local address leakage.
+ uint32 candidate_filter_;
+
friend class Connection;
};
@@ -604,4 +616,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_PORT_H_
+#endif // WEBRTC_P2P_BASE_PORT_H_
diff --git a/p2p/base/port_unittest.cc b/p2p/base/port_unittest.cc
index efa3f65..bd4520f 100644
--- a/p2p/base/port_unittest.cc
+++ b/p2p/base/port_unittest.cc
@@ -25,16 +25,16 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/base/portproxy.h"
-#include "talk/p2p/base/relayport.h"
-#include "talk/p2p/base/stunport.h"
-#include "talk/p2p/base/tcpport.h"
-#include "talk/p2p/base/testrelayserver.h"
-#include "talk/p2p/base/teststunserver.h"
-#include "talk/p2p/base/testturnserver.h"
-#include "talk/p2p/base/transport.h"
-#include "talk/p2p/base/turnport.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/portproxy.h"
+#include "webrtc/p2p/base/relayport.h"
+#include "webrtc/p2p/base/stunport.h"
+#include "webrtc/p2p/base/tcpport.h"
+#include "webrtc/p2p/base/testrelayserver.h"
+#include "webrtc/p2p/base/teststunserver.h"
+#include "webrtc/p2p/base/testturnserver.h"
+#include "webrtc/p2p/base/transport.h"
+#include "webrtc/p2p/base/turnport.h"
#include "webrtc/base/crc32.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/helpers.h"
diff --git a/p2p/base/portallocator.cc b/p2p/base/portallocator.cc
index 369af86..84fb344 100644
--- a/p2p/base/portallocator.cc
+++ b/p2p/base/portallocator.cc
@@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/portallocator.h"
+#include "webrtc/p2p/base/portallocator.h"
-#include "talk/p2p/base/portallocatorsessionproxy.h"
+#include "webrtc/p2p/base/portallocatorsessionproxy.h"
namespace cricket {
diff --git a/p2p/base/portallocator.h b/p2p/base/portallocator.h
index 5bc389e..29a1b0e 100644
--- a/p2p/base/portallocator.h
+++ b/p2p/base/portallocator.h
@@ -25,13 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_PORTALLOCATOR_H_
-#define TALK_P2P_BASE_PORTALLOCATOR_H_
+#ifndef WEBRTC_P2P_BASE_PORTALLOCATOR_H_
+#define WEBRTC_P2P_BASE_PORTALLOCATOR_H_
#include <string>
#include <vector>
-#include "talk/p2p/base/portinterface.h"
+#include "webrtc/p2p/base/portinterface.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/proxyinfo.h"
#include "webrtc/base/sigslot.h"
@@ -206,4 +206,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_PORTALLOCATOR_H_
+#endif // WEBRTC_P2P_BASE_PORTALLOCATOR_H_
diff --git a/p2p/base/portallocatorsessionproxy.cc b/p2p/base/portallocatorsessionproxy.cc
index a6c8065..2801f26 100644
--- a/p2p/base/portallocatorsessionproxy.cc
+++ b/p2p/base/portallocatorsessionproxy.cc
@@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/portallocatorsessionproxy.h"
+#include "webrtc/p2p/base/portallocatorsessionproxy.h"
-#include "talk/p2p/base/portallocator.h"
-#include "talk/p2p/base/portproxy.h"
+#include "webrtc/p2p/base/portallocator.h"
+#include "webrtc/p2p/base/portproxy.h"
#include "webrtc/base/thread.h"
namespace cricket {
diff --git a/p2p/base/portallocatorsessionproxy.h b/p2p/base/portallocatorsessionproxy.h
index 659c730..fdd5d02 100644
--- a/p2p/base/portallocatorsessionproxy.h
+++ b/p2p/base/portallocatorsessionproxy.h
@@ -25,13 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_
-#define TALK_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_
+#ifndef WEBRTC_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_
+#define WEBRTC_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_
#include <string>
-#include "talk/p2p/base/candidate.h"
-#include "talk/p2p/base/portallocator.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/portallocator.h"
namespace cricket {
class PortAllocator;
@@ -120,4 +120,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_
+#endif // WEBRTC_P2P_BASE_PORTALLOCATORSESSIONPROXY_H_
diff --git a/p2p/base/portallocatorsessionproxy_unittest.cc b/p2p/base/portallocatorsessionproxy_unittest.cc
index 3a69b46..30353ee 100644
--- a/p2p/base/portallocatorsessionproxy_unittest.cc
+++ b/p2p/base/portallocatorsessionproxy_unittest.cc
@@ -27,10 +27,10 @@
#include <vector>
-#include "talk/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/base/portallocatorsessionproxy.h"
-#include "talk/p2p/client/basicportallocator.h"
-#include "talk/p2p/client/fakeportallocator.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/portallocatorsessionproxy.h"
+#include "webrtc/p2p/client/basicportallocator.h"
+#include "webrtc/p2p/client/fakeportallocator.h"
#include "webrtc/base/fakenetwork.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/thread.h"
diff --git a/p2p/base/portinterface.h b/p2p/base/portinterface.h
index 8cab5b4..d53dc2e 100644
--- a/p2p/base/portinterface.h
+++ b/p2p/base/portinterface.h
@@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_PORTINTERFACE_H_
-#define TALK_P2P_BASE_PORTINTERFACE_H_
+#ifndef WEBRTC_P2P_BASE_PORTINTERFACE_H_
+#define WEBRTC_P2P_BASE_PORTINTERFACE_H_
#include <string>
-#include "talk/p2p/base/transport.h"
+#include "webrtc/p2p/base/transport.h"
#include "webrtc/base/socketaddress.h"
namespace rtc {
@@ -140,4 +140,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_PORTINTERFACE_H_
+#endif // WEBRTC_P2P_BASE_PORTINTERFACE_H_
diff --git a/p2p/base/portproxy.cc b/p2p/base/portproxy.cc
index 841cd85..5a5d1ff 100644
--- a/p2p/base/portproxy.cc
+++ b/p2p/base/portproxy.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/portproxy.h"
+#include "webrtc/p2p/base/portproxy.h"
namespace cricket {
diff --git a/p2p/base/portproxy.h b/p2p/base/portproxy.h
index 8c28223..6f79318 100644
--- a/p2p/base/portproxy.h
+++ b/p2p/base/portproxy.h
@@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_PORTPROXY_H_
-#define TALK_P2P_BASE_PORTPROXY_H_
+#ifndef WEBRTC_P2P_BASE_PORTPROXY_H_
+#define WEBRTC_P2P_BASE_PORTPROXY_H_
-#include "talk/p2p/base/portinterface.h"
+#include "webrtc/p2p/base/portinterface.h"
#include "webrtc/base/sigslot.h"
namespace rtc {
@@ -101,4 +101,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_PORTPROXY_H_
+#endif // WEBRTC_P2P_BASE_PORTPROXY_H_
diff --git a/p2p/base/pseudotcp.cc b/p2p/base/pseudotcp.cc
index 9a944f0..a82cd67 100644
--- a/p2p/base/pseudotcp.cc
+++ b/p2p/base/pseudotcp.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/pseudotcp.h"
+#include "webrtc/p2p/base/pseudotcp.h"
#include <stdio.h>
#include <stdlib.h>
diff --git a/p2p/base/pseudotcp.h b/p2p/base/pseudotcp.h
index 46e9d3b..e3455de 100644
--- a/p2p/base/pseudotcp.h
+++ b/p2p/base/pseudotcp.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_PSEUDOTCP_H_
-#define TALK_P2P_BASE_PSEUDOTCP_H_
+#ifndef WEBRTC_P2P_BASE_PSEUDOTCP_H_
+#define WEBRTC_P2P_BASE_PSEUDOTCP_H_
#include <list>
@@ -255,4 +255,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_PSEUDOTCP_H_
+#endif // WEBRTC_P2P_BASE_PSEUDOTCP_H_
diff --git a/p2p/base/pseudotcp_unittest.cc b/p2p/base/pseudotcp_unittest.cc
index d9435cf..522eb18 100644
--- a/p2p/base/pseudotcp_unittest.cc
+++ b/p2p/base/pseudotcp_unittest.cc
@@ -27,7 +27,7 @@
#include <vector>
-#include "talk/p2p/base/pseudotcp.h"
+#include "webrtc/p2p/base/pseudotcp.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/messagehandler.h"
diff --git a/p2p/base/rawtransport.cc b/p2p/base/rawtransport.cc
index 2af1864..f363dd1 100644
--- a/p2p/base/rawtransport.cc
+++ b/p2p/base/rawtransport.cc
@@ -27,14 +27,14 @@
#include <string>
#include <vector>
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/parsing.h"
-#include "talk/p2p/base/rawtransport.h"
-#include "talk/p2p/base/rawtransportchannel.h"
-#include "talk/p2p/base/sessionmanager.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/parsing.h"
+#include "webrtc/p2p/base/rawtransport.h"
+#include "webrtc/p2p/base/rawtransportchannel.h"
+#include "webrtc/p2p/base/sessionmanager.h"
#include "webrtc/libjingle/xmllite/qname.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/common.h"
#if defined(FEATURE_ENABLE_PSTN)
diff --git a/p2p/base/rawtransport.h b/p2p/base/rawtransport.h
index 3a20ef5..7458191 100644
--- a/p2p/base/rawtransport.h
+++ b/p2p/base/rawtransport.h
@@ -25,11 +25,11 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_RAWTRANSPORT_H_
-#define TALK_P2P_BASE_RAWTRANSPORT_H_
+#ifndef WEBRTC_P2P_BASE_RAWTRANSPORT_H_
+#define WEBRTC_P2P_BASE_RAWTRANSPORT_H_
#include <string>
-#include "talk/p2p/base/transport.h"
+#include "webrtc/p2p/base/transport.h"
#if defined(FEATURE_ENABLE_PSTN)
namespace cricket {
@@ -78,4 +78,4 @@
#endif // defined(FEATURE_ENABLE_PSTN)
-#endif // TALK_P2P_BASE_RAWTRANSPORT_H_
+#endif // WEBRTC_P2P_BASE_RAWTRANSPORT_H_
diff --git a/p2p/base/rawtransportchannel.cc b/p2p/base/rawtransportchannel.cc
index ae268e3..ef2286b 100644
--- a/p2p/base/rawtransportchannel.cc
+++ b/p2p/base/rawtransportchannel.cc
@@ -25,20 +25,20 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/rawtransportchannel.h"
+#include "webrtc/p2p/base/rawtransportchannel.h"
#include <string>
#include <vector>
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/portallocator.h"
-#include "talk/p2p/base/portinterface.h"
-#include "talk/p2p/base/rawtransport.h"
-#include "talk/p2p/base/relayport.h"
-#include "talk/p2p/base/sessionmanager.h"
-#include "talk/p2p/base/stunport.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/portallocator.h"
+#include "webrtc/p2p/base/portinterface.h"
+#include "webrtc/p2p/base/rawtransport.h"
+#include "webrtc/p2p/base/relayport.h"
+#include "webrtc/p2p/base/sessionmanager.h"
+#include "webrtc/p2p/base/stunport.h"
#include "webrtc/libjingle/xmllite/qname.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/common.h"
#if defined(FEATURE_ENABLE_PSTN)
diff --git a/p2p/base/rawtransportchannel.h b/p2p/base/rawtransportchannel.h
index 8fbf073..25f6e4f 100644
--- a/p2p/base/rawtransportchannel.h
+++ b/p2p/base/rawtransportchannel.h
@@ -25,14 +25,14 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_RAWTRANSPORTCHANNEL_H_
-#define TALK_P2P_BASE_RAWTRANSPORTCHANNEL_H_
+#ifndef WEBRTC_P2P_BASE_RAWTRANSPORTCHANNEL_H_
+#define WEBRTC_P2P_BASE_RAWTRANSPORTCHANNEL_H_
#include <string>
#include <vector>
-#include "talk/p2p/base/candidate.h"
-#include "talk/p2p/base/rawtransport.h"
-#include "talk/p2p/base/transportchannelimpl.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/rawtransport.h"
+#include "webrtc/p2p/base/transportchannelimpl.h"
#include "webrtc/base/messagequeue.h"
#if defined(FEATURE_ENABLE_PSTN)
@@ -203,4 +203,4 @@
} // namespace cricket
#endif // defined(FEATURE_ENABLE_PSTN)
-#endif // TALK_P2P_BASE_RAWTRANSPORTCHANNEL_H_
+#endif // WEBRTC_P2P_BASE_RAWTRANSPORTCHANNEL_H_
diff --git a/p2p/base/relayport.cc b/p2p/base/relayport.cc
index 55052a5..2db42c9 100644
--- a/p2p/base/relayport.cc
+++ b/p2p/base/relayport.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/relayport.h"
+#include "webrtc/p2p/base/relayport.h"
#include "webrtc/base/asyncpacketsocket.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
diff --git a/p2p/base/relayport.h b/p2p/base/relayport.h
index f22d045..bc0c5d9 100644
--- a/p2p/base/relayport.h
+++ b/p2p/base/relayport.h
@@ -25,16 +25,16 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_RELAYPORT_H_
-#define TALK_P2P_BASE_RELAYPORT_H_
+#ifndef WEBRTC_P2P_BASE_RELAYPORT_H_
+#define WEBRTC_P2P_BASE_RELAYPORT_H_
#include <deque>
#include <string>
#include <utility>
#include <vector>
-#include "talk/p2p/base/port.h"
-#include "talk/p2p/base/stunrequest.h"
+#include "webrtc/p2p/base/port.h"
+#include "webrtc/p2p/base/stunrequest.h"
namespace cricket {
@@ -115,4 +115,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_RELAYPORT_H_
+#endif // WEBRTC_P2P_BASE_RELAYPORT_H_
diff --git a/p2p/base/relayport_unittest.cc b/p2p/base/relayport_unittest.cc
index ebb16c5..2f4515e 100644
--- a/p2p/base/relayport_unittest.cc
+++ b/p2p/base/relayport_unittest.cc
@@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/base/relayport.h"
-#include "talk/p2p/base/relayserver.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/relayport.h"
+#include "webrtc/p2p/base/relayserver.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
diff --git a/p2p/base/relayserver.cc b/p2p/base/relayserver.cc
index b5d1ac6..ebf165b 100644
--- a/p2p/base/relayserver.cc
+++ b/p2p/base/relayserver.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/relayserver.h"
+#include "webrtc/p2p/base/relayserver.h"
#ifdef POSIX
#include <errno.h>
diff --git a/p2p/base/relayserver.h b/p2p/base/relayserver.h
index 248d0e7..8065174 100644
--- a/p2p/base/relayserver.h
+++ b/p2p/base/relayserver.h
@@ -25,15 +25,15 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_RELAYSERVER_H_
-#define TALK_P2P_BASE_RELAYSERVER_H_
+#ifndef WEBRTC_P2P_BASE_RELAYSERVER_H_
+#define WEBRTC_P2P_BASE_RELAYSERVER_H_
#include <map>
#include <string>
#include <vector>
-#include "talk/p2p/base/port.h"
-#include "talk/p2p/base/stun.h"
+#include "webrtc/p2p/base/port.h"
+#include "webrtc/p2p/base/stun.h"
#include "webrtc/base/asyncudpsocket.h"
#include "webrtc/base/socketaddresspair.h"
#include "webrtc/base/thread.h"
@@ -249,4 +249,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_RELAYSERVER_H_
+#endif // WEBRTC_P2P_BASE_RELAYSERVER_H_
diff --git a/p2p/base/relayserver_unittest.cc b/p2p/base/relayserver_unittest.cc
index 5d77ca6..5ea5e10 100644
--- a/p2p/base/relayserver_unittest.cc
+++ b/p2p/base/relayserver_unittest.cc
@@ -27,7 +27,7 @@
#include <string>
-#include "talk/p2p/base/relayserver.h"
+#include "webrtc/p2p/base/relayserver.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
diff --git a/p2p/base/session.cc b/p2p/base/session.cc
index a9bf815..c4db1cf 100644
--- a/p2p/base/session.cc
+++ b/p2p/base/session.cc
@@ -25,16 +25,16 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/session.h"
+#include "webrtc/p2p/base/session.h"
-#include "talk/p2p/base/dtlstransport.h"
-#include "talk/p2p/base/p2ptransport.h"
-#include "talk/p2p/base/sessionclient.h"
-#include "talk/p2p/base/transport.h"
-#include "talk/p2p/base/transportchannelproxy.h"
-#include "talk/p2p/base/transportinfo.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/jid.h"
+#include "webrtc/p2p/base/dtlstransport.h"
+#include "webrtc/p2p/base/p2ptransport.h"
+#include "webrtc/p2p/base/sessionclient.h"
+#include "webrtc/p2p/base/transport.h"
+#include "webrtc/p2p/base/transportchannelproxy.h"
+#include "webrtc/p2p/base/transportinfo.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/jid.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/common.h"
#include "webrtc/base/helpers.h"
@@ -42,7 +42,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/sslstreamadapter.h"
-#include "talk/p2p/base/constants.h"
+#include "webrtc/p2p/base/constants.h"
namespace cricket {
diff --git a/p2p/base/session.h b/p2p/base/session.h
index e06cf00..cbe350f 100644
--- a/p2p/base/session.h
+++ b/p2p/base/session.h
@@ -25,22 +25,22 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_SESSION_H_
-#define TALK_P2P_BASE_SESSION_H_
+#ifndef WEBRTC_P2P_BASE_SESSION_H_
+#define WEBRTC_P2P_BASE_SESSION_H_
#include <list>
#include <map>
#include <string>
#include <vector>
-#include "talk/p2p/base/parsing.h"
-#include "talk/p2p/base/port.h"
-#include "talk/p2p/base/sessionclient.h"
-#include "talk/p2p/base/sessionmanager.h"
-#include "talk/p2p/base/sessionmessages.h"
-#include "talk/p2p/base/transport.h"
+#include "webrtc/p2p/base/parsing.h"
+#include "webrtc/p2p/base/port.h"
+#include "webrtc/p2p/base/sessionclient.h"
+#include "webrtc/p2p/base/sessionmanager.h"
+#include "webrtc/p2p/base/sessionmessages.h"
+#include "webrtc/p2p/base/transport.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/scoped_ref_ptr.h"
@@ -744,4 +744,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_SESSION_H_
+#endif // WEBRTC_P2P_BASE_SESSION_H_
diff --git a/p2p/base/session_unittest.cc b/p2p/base/session_unittest.cc
index 4674d2c..c7a3db9 100644
--- a/p2p/base/session_unittest.cc
+++ b/p2p/base/session_unittest.cc
@@ -31,22 +31,22 @@
#include <map>
#include <sstream>
-#include "talk/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/p2ptransport.h"
-#include "talk/p2p/base/parsing.h"
-#include "talk/p2p/base/portallocator.h"
-#include "talk/p2p/base/relayport.h"
-#include "talk/p2p/base/relayserver.h"
-#include "talk/p2p/base/session.h"
-#include "talk/p2p/base/sessionclient.h"
-#include "talk/p2p/base/sessionmanager.h"
-#include "talk/p2p/base/stunport.h"
-#include "talk/p2p/base/stunserver.h"
-#include "talk/p2p/base/transportchannel.h"
-#include "talk/p2p/base/transportchannelproxy.h"
-#include "talk/p2p/base/udpport.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/p2ptransport.h"
+#include "webrtc/p2p/base/parsing.h"
+#include "webrtc/p2p/base/portallocator.h"
+#include "webrtc/p2p/base/relayport.h"
+#include "webrtc/p2p/base/relayserver.h"
+#include "webrtc/p2p/base/session.h"
+#include "webrtc/p2p/base/sessionclient.h"
+#include "webrtc/p2p/base/sessionmanager.h"
+#include "webrtc/p2p/base/stunport.h"
+#include "webrtc/p2p/base/stunserver.h"
+#include "webrtc/p2p/base/transportchannel.h"
+#include "webrtc/p2p/base/transportchannelproxy.h"
+#include "webrtc/p2p/base/udpport.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/common.h"
#include "webrtc/base/gunit.h"
diff --git a/p2p/base/sessionclient.h b/p2p/base/sessionclient.h
index 10b0c92..f7952f0 100644
--- a/p2p/base/sessionclient.h
+++ b/p2p/base/sessionclient.h
@@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_SESSIONCLIENT_H_
-#define TALK_P2P_BASE_SESSIONCLIENT_H_
+#ifndef WEBRTC_P2P_BASE_SESSIONCLIENT_H_
+#define WEBRTC_P2P_BASE_SESSIONCLIENT_H_
-#include "talk/p2p/base/constants.h"
+#include "webrtc/p2p/base/constants.h"
namespace buzz {
class XmlElement;
@@ -92,4 +92,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_SESSIONCLIENT_H_
+#endif // WEBRTC_P2P_BASE_SESSIONCLIENT_H_
diff --git a/p2p/base/sessiondescription.cc b/p2p/base/sessiondescription.cc
index 7ad3d48..57f08c9 100644
--- a/p2p/base/sessiondescription.cc
+++ b/p2p/base/sessiondescription.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/base/sessiondescription.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
diff --git a/p2p/base/sessiondescription.h b/p2p/base/sessiondescription.h
index 56dd412..bc76761 100644
--- a/p2p/base/sessiondescription.h
+++ b/p2p/base/sessiondescription.h
@@ -25,13 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_SESSIONDESCRIPTION_H_
-#define TALK_P2P_BASE_SESSIONDESCRIPTION_H_
+#ifndef WEBRTC_P2P_BASE_SESSIONDESCRIPTION_H_
+#define WEBRTC_P2P_BASE_SESSIONDESCRIPTION_H_
#include <string>
#include <vector>
-#include "talk/p2p/base/transportinfo.h"
+#include "webrtc/p2p/base/transportinfo.h"
#include "webrtc/base/constructormagic.h"
namespace cricket {
@@ -199,4 +199,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_SESSIONDESCRIPTION_H_
+#endif // WEBRTC_P2P_BASE_SESSIONDESCRIPTION_H_
diff --git a/p2p/base/sessionid.h b/p2p/base/sessionid.h
index 6942942..52ee990 100644
--- a/p2p/base/sessionid.h
+++ b/p2p/base/sessionid.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_SESSIONID_H_
-#define TALK_P2P_BASE_SESSIONID_H_
+#ifndef WEBRTC_P2P_BASE_SESSIONID_H_
+#define WEBRTC_P2P_BASE_SESSIONID_H_
// TODO: Remove this file.
@@ -34,4 +34,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_SESSIONID_H_
+#endif // WEBRTC_P2P_BASE_SESSIONID_H_
diff --git a/p2p/base/sessionmanager.cc b/p2p/base/sessionmanager.cc
index d173517..7f27f6c 100644
--- a/p2p/base/sessionmanager.cc
+++ b/p2p/base/sessionmanager.cc
@@ -25,13 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/sessionmanager.h"
+#include "webrtc/p2p/base/sessionmanager.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/session.h"
-#include "talk/p2p/base/sessionmessages.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/jid.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/session.h"
+#include "webrtc/p2p/base/sessionmessages.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/jid.h"
#include "webrtc/base/common.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
diff --git a/p2p/base/sessionmanager.h b/p2p/base/sessionmanager.h
index 92aeb3d..0e0e9a4 100644
--- a/p2p/base/sessionmanager.h
+++ b/p2p/base/sessionmanager.h
@@ -25,16 +25,16 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_SESSIONMANAGER_H_
-#define TALK_P2P_BASE_SESSIONMANAGER_H_
+#ifndef WEBRTC_P2P_BASE_SESSIONMANAGER_H_
+#define WEBRTC_P2P_BASE_SESSIONMANAGER_H_
#include <map>
#include <string>
#include <utility>
#include <vector>
-#include "talk/p2p/base/portallocator.h"
-#include "talk/p2p/base/transportdescriptionfactory.h"
+#include "webrtc/p2p/base/portallocator.h"
+#include "webrtc/p2p/base/transportdescriptionfactory.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/thread.h"
@@ -208,4 +208,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_SESSIONMANAGER_H_
+#endif // WEBRTC_P2P_BASE_SESSIONMANAGER_H_
diff --git a/p2p/base/sessionmessages.cc b/p2p/base/sessionmessages.cc
index b2fd9d6..0534271 100644
--- a/p2p/base/sessionmessages.cc
+++ b/p2p/base/sessionmessages.cc
@@ -25,19 +25,19 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/sessionmessages.h"
+#include "webrtc/p2p/base/sessionmessages.h"
#include <stdio.h>
#include <string>
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/p2ptransport.h"
-#include "talk/p2p/base/parsing.h"
-#include "talk/p2p/base/sessionclient.h"
-#include "talk/p2p/base/sessiondescription.h"
-#include "talk/p2p/base/transport.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/p2ptransport.h"
+#include "webrtc/p2p/base/parsing.h"
+#include "webrtc/p2p/base/sessionclient.h"
+#include "webrtc/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/base/transport.h"
#include "webrtc/libjingle/xmllite/xmlconstants.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/stringutils.h"
diff --git a/p2p/base/sessionmessages.h b/p2p/base/sessionmessages.h
index 7e1f8ac..0ab56ce 100644
--- a/p2p/base/sessionmessages.h
+++ b/p2p/base/sessionmessages.h
@@ -25,17 +25,17 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_SESSIONMESSAGES_H_
-#define TALK_P2P_BASE_SESSIONMESSAGES_H_
+#ifndef WEBRTC_P2P_BASE_SESSIONMESSAGES_H_
+#define WEBRTC_P2P_BASE_SESSIONMESSAGES_H_
#include <map>
#include <string>
#include <vector>
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/parsing.h"
-#include "talk/p2p/base/sessiondescription.h" // Needed to delete contents.
-#include "talk/p2p/base/transportinfo.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/parsing.h"
+#include "webrtc/p2p/base/sessiondescription.h" // Needed to delete contents.
+#include "webrtc/p2p/base/transportinfo.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "webrtc/base/basictypes.h"
@@ -240,4 +240,4 @@
SessionRedirect* redirect);
} // namespace cricket
-#endif // TALK_P2P_BASE_SESSIONMESSAGES_H_
+#endif // WEBRTC_P2P_BASE_SESSIONMESSAGES_H_
diff --git a/p2p/base/stun.cc b/p2p/base/stun.cc
index 061fd9a..6bdf77e 100644
--- a/p2p/base/stun.cc
+++ b/p2p/base/stun.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/stun.h"
+#include "webrtc/p2p/base/stun.h"
#include <string.h>
diff --git a/p2p/base/stun.h b/p2p/base/stun.h
index c4f522b..5a43a3c 100644
--- a/p2p/base/stun.h
+++ b/p2p/base/stun.h
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_STUN_H_
-#define TALK_P2P_BASE_STUN_H_
+#ifndef WEBRTC_P2P_BASE_STUN_H_
+#define WEBRTC_P2P_BASE_STUN_H_
// This file contains classes for dealing with the STUN protocol, as specified
// in RFC 5389, and its descendants.
@@ -646,4 +646,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_STUN_H_
+#endif // WEBRTC_P2P_BASE_STUN_H_
diff --git a/p2p/base/stun_unittest.cc b/p2p/base/stun_unittest.cc
index 00dffed..34a47cb 100644
--- a/p2p/base/stun_unittest.cc
+++ b/p2p/base/stun_unittest.cc
@@ -27,7 +27,7 @@
#include <string>
-#include "talk/p2p/base/stun.h"
+#include "webrtc/p2p/base/stun.h"
#include "webrtc/base/bytebuffer.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
diff --git a/p2p/base/stunport.cc b/p2p/base/stunport.cc
index a185a35..0602c5a 100644
--- a/p2p/base/stunport.cc
+++ b/p2p/base/stunport.cc
@@ -25,10 +25,11 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/stunport.h"
+#include "webrtc/p2p/base/stunport.h"
-#include "talk/p2p/base/common.h"
-#include "talk/p2p/base/stun.h"
+#include "webrtc/p2p/base/common.h"
+#include "webrtc/p2p/base/portallocator.h"
+#include "webrtc/p2p/base/stun.h"
#include "webrtc/base/common.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
@@ -395,8 +396,17 @@
// For STUN, related address is the local socket address.
if ((!SharedSocket() || stun_reflected_addr != socket_->GetLocalAddress()) &&
!HasCandidateWithAddress(stun_reflected_addr)) {
+
+ rtc::SocketAddress related_address = socket_->GetLocalAddress();
+ if (!(candidate_filter() & CF_HOST)) {
+ // If candidate filter doesn't have CF_HOST specified, empty raddr to
+ // avoid local address leakage.
+ related_address = rtc::EmptySocketAddressWithFamily(
+ related_address.family());
+ }
+
AddAddress(stun_reflected_addr, socket_->GetLocalAddress(),
- socket_->GetLocalAddress(), UDP_PROTOCOL_NAME, "",
+ related_address, UDP_PROTOCOL_NAME, "",
STUN_PORT_TYPE, ICE_TYPE_PREFERENCE_SRFLX, 0, false);
}
MaybeSetPortCompleteOrError();
diff --git a/p2p/base/stunport.h b/p2p/base/stunport.h
index 1fc1f64..9767546 100644
--- a/p2p/base/stunport.h
+++ b/p2p/base/stunport.h
@@ -25,13 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_STUNPORT_H_
-#define TALK_P2P_BASE_STUNPORT_H_
+#ifndef WEBRTC_P2P_BASE_STUNPORT_H_
+#define WEBRTC_P2P_BASE_STUNPORT_H_
#include <string>
-#include "talk/p2p/base/port.h"
-#include "talk/p2p/base/stunrequest.h"
+#include "webrtc/p2p/base/port.h"
+#include "webrtc/p2p/base/stunrequest.h"
#include "webrtc/base/asyncpacketsocket.h"
// TODO(mallinath) - Rename stunport.cc|h to udpport.cc|h.
@@ -252,4 +252,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_STUNPORT_H_
+#endif // WEBRTC_P2P_BASE_STUNPORT_H_
diff --git a/p2p/base/stunport_unittest.cc b/p2p/base/stunport_unittest.cc
index 5c97ecc..7b289ea 100644
--- a/p2p/base/stunport_unittest.cc
+++ b/p2p/base/stunport_unittest.cc
@@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/base/stunport.h"
-#include "talk/p2p/base/teststunserver.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/stunport.h"
+#include "webrtc/p2p/base/teststunserver.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/physicalsocketserver.h"
diff --git a/p2p/base/stunrequest.cc b/p2p/base/stunrequest.cc
index 148718f..55d270e 100644
--- a/p2p/base/stunrequest.cc
+++ b/p2p/base/stunrequest.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/stunrequest.h"
+#include "webrtc/p2p/base/stunrequest.h"
#include "webrtc/base/common.h"
#include "webrtc/base/helpers.h"
diff --git a/p2p/base/stunrequest.h b/p2p/base/stunrequest.h
index 374f7e7..be73445 100644
--- a/p2p/base/stunrequest.h
+++ b/p2p/base/stunrequest.h
@@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_STUNREQUEST_H_
-#define TALK_P2P_BASE_STUNREQUEST_H_
+#ifndef WEBRTC_P2P_BASE_STUNREQUEST_H_
+#define WEBRTC_P2P_BASE_STUNREQUEST_H_
#include <map>
#include <string>
-#include "talk/p2p/base/stun.h"
+#include "webrtc/p2p/base/stun.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/thread.h"
@@ -130,4 +130,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_STUNREQUEST_H_
+#endif // WEBRTC_P2P_BASE_STUNREQUEST_H_
diff --git a/p2p/base/stunrequest_unittest.cc b/p2p/base/stunrequest_unittest.cc
index 2783aa3..5c7b97b 100644
--- a/p2p/base/stunrequest_unittest.cc
+++ b/p2p/base/stunrequest_unittest.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/stunrequest.h"
+#include "webrtc/p2p/base/stunrequest.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
diff --git a/p2p/base/stunserver.cc b/p2p/base/stunserver.cc
index df428f9..a4beca4 100644
--- a/p2p/base/stunserver.cc
+++ b/p2p/base/stunserver.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/stunserver.h"
+#include "webrtc/p2p/base/stunserver.h"
#include "webrtc/base/bytebuffer.h"
#include "webrtc/base/logging.h"
diff --git a/p2p/base/stunserver.h b/p2p/base/stunserver.h
index a141e5b..5bbe258 100644
--- a/p2p/base/stunserver.h
+++ b/p2p/base/stunserver.h
@@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_STUNSERVER_H_
-#define TALK_P2P_BASE_STUNSERVER_H_
+#ifndef WEBRTC_P2P_BASE_STUNSERVER_H_
+#define WEBRTC_P2P_BASE_STUNSERVER_H_
-#include "talk/p2p/base/stun.h"
+#include "webrtc/p2p/base/stun.h"
#include "webrtc/base/asyncudpsocket.h"
#include "webrtc/base/scoped_ptr.h"
@@ -80,4 +80,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_STUNSERVER_H_
+#endif // WEBRTC_P2P_BASE_STUNSERVER_H_
diff --git a/p2p/base/stunserver_unittest.cc b/p2p/base/stunserver_unittest.cc
index 405d8ae..4cf3c33 100644
--- a/p2p/base/stunserver_unittest.cc
+++ b/p2p/base/stunserver_unittest.cc
@@ -27,7 +27,7 @@
#include <string>
-#include "talk/p2p/base/stunserver.h"
+#include "webrtc/p2p/base/stunserver.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/physicalsocketserver.h"
diff --git a/p2p/base/tcpport.cc b/p2p/base/tcpport.cc
index cb5b726..aa08a92 100644
--- a/p2p/base/tcpport.cc
+++ b/p2p/base/tcpport.cc
@@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/tcpport.h"
+#include "webrtc/p2p/base/tcpport.h"
-#include "talk/p2p/base/common.h"
+#include "webrtc/p2p/base/common.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
@@ -304,20 +304,23 @@
// the one we asked for. This is seen in Chrome, where TCP sockets cannot be
// given a binding address, and the platform is expected to pick the
// correct local address.
- if (socket->GetLocalAddress().ipaddr() == port()->ip()) {
+ const rtc::IPAddress& socket_ip = socket->GetLocalAddress().ipaddr();
+ if (socket_ip == port()->ip()) {
LOG_J(LS_VERBOSE, this) << "Connection established to "
<< socket->GetRemoteAddress().ToSensitiveString();
set_connected(true);
} else {
- LOG_J(LS_WARNING, this) << "Dropping connection as TCP socket bound to a "
- << "different address from the local candidate.";
+ LOG_J(LS_WARNING, this) << "Dropping connection as TCP socket bound to IP "
+ << socket_ip.ToSensitiveString()
+ << ", different from the local candidate IP "
+ << port()->ip().ToSensitiveString();
socket_->Close();
}
}
void TCPConnection::OnClose(rtc::AsyncPacketSocket* socket, int error) {
ASSERT(socket == socket_);
- LOG_J(LS_VERBOSE, this) << "Connection closed with error " << error;
+ LOG_J(LS_INFO, this) << "Connection closed with error " << error;
set_connected(false);
set_write_state(STATE_WRITE_TIMEOUT);
}
diff --git a/p2p/base/tcpport.h b/p2p/base/tcpport.h
index 97bd977..c331f6b 100644
--- a/p2p/base/tcpport.h
+++ b/p2p/base/tcpport.h
@@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_TCPPORT_H_
-#define TALK_P2P_BASE_TCPPORT_H_
+#ifndef WEBRTC_P2P_BASE_TCPPORT_H_
+#define WEBRTC_P2P_BASE_TCPPORT_H_
#include <list>
#include <string>
-#include "talk/p2p/base/port.h"
+#include "webrtc/p2p/base/port.h"
#include "webrtc/base/asyncpacketsocket.h"
namespace cricket {
@@ -150,4 +150,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_TCPPORT_H_
+#endif // WEBRTC_P2P_BASE_TCPPORT_H_
diff --git a/p2p/base/testrelayserver.h b/p2p/base/testrelayserver.h
index eaeefc8..003b42c 100644
--- a/p2p/base/testrelayserver.h
+++ b/p2p/base/testrelayserver.h
@@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_TESTRELAYSERVER_H_
-#define TALK_P2P_BASE_TESTRELAYSERVER_H_
+#ifndef WEBRTC_P2P_BASE_TESTRELAYSERVER_H_
+#define WEBRTC_P2P_BASE_TESTRELAYSERVER_H_
-#include "talk/p2p/base/relayserver.h"
+#include "webrtc/p2p/base/relayserver.h"
#include "webrtc/base/asynctcpsocket.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/sigslot.h"
@@ -115,4 +115,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_TESTRELAYSERVER_H_
+#endif // WEBRTC_P2P_BASE_TESTRELAYSERVER_H_
diff --git a/p2p/base/teststunserver.h b/p2p/base/teststunserver.h
index 840150b..eef15d7 100644
--- a/p2p/base/teststunserver.h
+++ b/p2p/base/teststunserver.h
@@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_TESTSTUNSERVER_H_
-#define TALK_P2P_BASE_TESTSTUNSERVER_H_
+#ifndef WEBRTC_P2P_BASE_TESTSTUNSERVER_H_
+#define WEBRTC_P2P_BASE_TESTSTUNSERVER_H_
-#include "talk/p2p/base/stunserver.h"
+#include "webrtc/p2p/base/stunserver.h"
#include "webrtc/base/socketaddress.h"
#include "webrtc/base/thread.h"
@@ -72,4 +72,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_TESTSTUNSERVER_H_
+#endif // WEBRTC_P2P_BASE_TESTSTUNSERVER_H_
diff --git a/p2p/base/testturnserver.h b/p2p/base/testturnserver.h
index 6c30afe..64da877 100644
--- a/p2p/base/testturnserver.h
+++ b/p2p/base/testturnserver.h
@@ -25,15 +25,15 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_TESTTURNSERVER_H_
-#define TALK_P2P_BASE_TESTTURNSERVER_H_
+#ifndef WEBRTC_P2P_BASE_TESTTURNSERVER_H_
+#define WEBRTC_P2P_BASE_TESTTURNSERVER_H_
#include <string>
#include <vector>
-#include "talk/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/base/stun.h"
-#include "talk/p2p/base/turnserver.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/stun.h"
+#include "webrtc/p2p/base/turnserver.h"
#include "webrtc/base/asyncudpsocket.h"
#include "webrtc/base/thread.h"
@@ -117,4 +117,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_TESTTURNSERVER_H_
+#endif // WEBRTC_P2P_BASE_TESTTURNSERVER_H_
diff --git a/p2p/base/transport.cc b/p2p/base/transport.cc
index d88f5e7..f33f942 100644
--- a/p2p/base/transport.cc
+++ b/p2p/base/transport.cc
@@ -25,16 +25,16 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/transport.h"
+#include "webrtc/p2p/base/transport.h"
-#include "talk/p2p/base/candidate.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/parsing.h"
-#include "talk/p2p/base/port.h"
-#include "talk/p2p/base/sessionmanager.h"
-#include "talk/p2p/base/transportchannelimpl.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/parsing.h"
+#include "webrtc/p2p/base/port.h"
+#include "webrtc/p2p/base/sessionmanager.h"
+#include "webrtc/p2p/base/transportchannelimpl.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
diff --git a/p2p/base/transport.h b/p2p/base/transport.h
index 21661f6..fbc7f6f 100644
--- a/p2p/base/transport.h
+++ b/p2p/base/transport.h
@@ -43,16 +43,16 @@
// It is not possible to do so here because the subclass constructor will
// already have run.
-#ifndef TALK_P2P_BASE_TRANSPORT_H_
-#define TALK_P2P_BASE_TRANSPORT_H_
+#ifndef WEBRTC_P2P_BASE_TRANSPORT_H_
+#define WEBRTC_P2P_BASE_TRANSPORT_H_
#include <map>
#include <string>
#include <vector>
-#include "talk/p2p/base/candidate.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/sessiondescription.h"
-#include "talk/p2p/base/transportinfo.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/base/transportinfo.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/messagequeue.h"
#include "webrtc/base/sigslot.h"
@@ -527,4 +527,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_TRANSPORT_H_
+#endif // WEBRTC_P2P_BASE_TRANSPORT_H_
diff --git a/p2p/base/transport_unittest.cc b/p2p/base/transport_unittest.cc
index 8f7dae2..9ef7d60 100644
--- a/p2p/base/transport_unittest.cc
+++ b/p2p/base/transport_unittest.cc
@@ -25,14 +25,14 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/fakesession.h"
-#include "talk/p2p/base/p2ptransport.h"
-#include "talk/p2p/base/parsing.h"
-#include "talk/p2p/base/rawtransport.h"
-#include "talk/p2p/base/sessionmessages.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/fakesession.h"
+#include "webrtc/p2p/base/p2ptransport.h"
+#include "webrtc/p2p/base/parsing.h"
+#include "webrtc/p2p/base/rawtransport.h"
+#include "webrtc/p2p/base/sessionmessages.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/fakesslidentity.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/thread.h"
diff --git a/p2p/base/transportchannel.cc b/p2p/base/transportchannel.cc
index 50ebfb9..13048ed 100644
--- a/p2p/base/transportchannel.cc
+++ b/p2p/base/transportchannel.cc
@@ -26,7 +26,7 @@
*/
#include <sstream>
-#include "talk/p2p/base/transportchannel.h"
+#include "webrtc/p2p/base/transportchannel.h"
namespace cricket {
diff --git a/p2p/base/transportchannel.h b/p2p/base/transportchannel.h
index b0beb04..56d9467 100644
--- a/p2p/base/transportchannel.h
+++ b/p2p/base/transportchannel.h
@@ -25,15 +25,15 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_TRANSPORTCHANNEL_H_
-#define TALK_P2P_BASE_TRANSPORTCHANNEL_H_
+#ifndef WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_
+#define WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_
#include <string>
#include <vector>
-#include "talk/p2p/base/candidate.h"
-#include "talk/p2p/base/transport.h"
-#include "talk/p2p/base/transportdescription.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/transport.h"
+#include "webrtc/p2p/base/transportdescription.h"
#include "webrtc/base/asyncpacketsocket.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/dscp.h"
@@ -157,4 +157,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_TRANSPORTCHANNEL_H_
+#endif // WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_
diff --git a/p2p/base/transportchannelimpl.h b/p2p/base/transportchannelimpl.h
index fde980b..73afdee 100644
--- a/p2p/base/transportchannelimpl.h
+++ b/p2p/base/transportchannelimpl.h
@@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_TRANSPORTCHANNELIMPL_H_
-#define TALK_P2P_BASE_TRANSPORTCHANNELIMPL_H_
+#ifndef WEBRTC_P2P_BASE_TRANSPORTCHANNELIMPL_H_
+#define WEBRTC_P2P_BASE_TRANSPORTCHANNELIMPL_H_
#include <string>
-#include "talk/p2p/base/transport.h"
-#include "talk/p2p/base/transportchannel.h"
+#include "webrtc/p2p/base/transport.h"
+#include "webrtc/p2p/base/transportchannel.h"
namespace buzz { class XmlElement; }
@@ -125,4 +125,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_TRANSPORTCHANNELIMPL_H_
+#endif // WEBRTC_P2P_BASE_TRANSPORTCHANNELIMPL_H_
diff --git a/p2p/base/transportchannelproxy.cc b/p2p/base/transportchannelproxy.cc
index 9fa0b6a..259b8dd 100644
--- a/p2p/base/transportchannelproxy.cc
+++ b/p2p/base/transportchannelproxy.cc
@@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/transport.h"
-#include "talk/p2p/base/transportchannelimpl.h"
-#include "talk/p2p/base/transportchannelproxy.h"
+#include "webrtc/p2p/base/transport.h"
+#include "webrtc/p2p/base/transportchannelimpl.h"
+#include "webrtc/p2p/base/transportchannelproxy.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/thread.h"
diff --git a/p2p/base/transportchannelproxy.h b/p2p/base/transportchannelproxy.h
index 07b9090..dee4313 100644
--- a/p2p/base/transportchannelproxy.h
+++ b/p2p/base/transportchannelproxy.h
@@ -25,14 +25,14 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_TRANSPORTCHANNELPROXY_H_
-#define TALK_P2P_BASE_TRANSPORTCHANNELPROXY_H_
+#ifndef WEBRTC_P2P_BASE_TRANSPORTCHANNELPROXY_H_
+#define WEBRTC_P2P_BASE_TRANSPORTCHANNELPROXY_H_
#include <string>
#include <utility>
#include <vector>
-#include "talk/p2p/base/transportchannel.h"
+#include "webrtc/p2p/base/transportchannel.h"
#include "webrtc/base/messagehandler.h"
namespace rtc {
@@ -109,4 +109,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_TRANSPORTCHANNELPROXY_H_
+#endif // WEBRTC_P2P_BASE_TRANSPORTCHANNELPROXY_H_
diff --git a/p2p/base/transportdescription.cc b/p2p/base/transportdescription.cc
index 08460d7..8bdcbb3 100644
--- a/p2p/base/transportdescription.cc
+++ b/p2p/base/transportdescription.cc
@@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/transportdescription.h"
+#include "webrtc/p2p/base/transportdescription.h"
-#include "talk/p2p/base/constants.h"
+#include "webrtc/p2p/base/constants.h"
#include "webrtc/base/stringutils.h"
namespace cricket {
diff --git a/p2p/base/transportdescription.h b/p2p/base/transportdescription.h
index efd04ff..5763d23 100644
--- a/p2p/base/transportdescription.h
+++ b/p2p/base/transportdescription.h
@@ -25,15 +25,15 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_TRANSPORTDESCRIPTION_H_
-#define TALK_P2P_BASE_TRANSPORTDESCRIPTION_H_
+#ifndef WEBRTC_P2P_BASE_TRANSPORTDESCRIPTION_H_
+#define WEBRTC_P2P_BASE_TRANSPORTDESCRIPTION_H_
#include <algorithm>
#include <string>
#include <vector>
-#include "talk/p2p/base/candidate.h"
-#include "talk/p2p/base/constants.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/constants.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/sslfingerprint.h"
@@ -185,4 +185,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_TRANSPORTDESCRIPTION_H_
+#endif // WEBRTC_P2P_BASE_TRANSPORTDESCRIPTION_H_
diff --git a/p2p/base/transportdescriptionfactory.cc b/p2p/base/transportdescriptionfactory.cc
index 4eb603c..7260b9c 100644
--- a/p2p/base/transportdescriptionfactory.cc
+++ b/p2p/base/transportdescriptionfactory.cc
@@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/transportdescriptionfactory.h"
+#include "webrtc/p2p/base/transportdescriptionfactory.h"
-#include "talk/p2p/base/transportdescription.h"
+#include "webrtc/p2p/base/transportdescription.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/messagedigest.h"
diff --git a/p2p/base/transportdescriptionfactory.h b/p2p/base/transportdescriptionfactory.h
index 84f25ac..bcd14c6 100644
--- a/p2p/base/transportdescriptionfactory.h
+++ b/p2p/base/transportdescriptionfactory.h
@@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_
-#define TALK_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_
+#ifndef WEBRTC_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_
+#define WEBRTC_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_
-#include "talk/p2p/base/transportdescription.h"
+#include "webrtc/p2p/base/transportdescription.h"
namespace rtc {
class SSLIdentity;
@@ -80,4 +80,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_
+#endif // WEBRTC_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_
diff --git a/p2p/base/transportdescriptionfactory_unittest.cc b/p2p/base/transportdescriptionfactory_unittest.cc
index 5c17f96..a793853 100644
--- a/p2p/base/transportdescriptionfactory_unittest.cc
+++ b/p2p/base/transportdescriptionfactory_unittest.cc
@@ -28,9 +28,9 @@
#include <string>
#include <vector>
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/transportdescription.h"
-#include "talk/p2p/base/transportdescriptionfactory.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/transportdescription.h"
+#include "webrtc/p2p/base/transportdescriptionfactory.h"
#include "webrtc/base/fakesslidentity.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/ssladapter.h"
diff --git a/p2p/base/transportinfo.h b/p2p/base/transportinfo.h
index 5ffb10a..414b6ca 100644
--- a/p2p/base/transportinfo.h
+++ b/p2p/base/transportinfo.h
@@ -25,15 +25,15 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_TRANSPORTINFO_H_
-#define TALK_P2P_BASE_TRANSPORTINFO_H_
+#ifndef WEBRTC_P2P_BASE_TRANSPORTINFO_H_
+#define WEBRTC_P2P_BASE_TRANSPORTINFO_H_
#include <string>
#include <vector>
-#include "talk/p2p/base/candidate.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/transportdescription.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/transportdescription.h"
#include "webrtc/base/helpers.h"
namespace cricket {
@@ -57,4 +57,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_TRANSPORTINFO_H_
+#endif // WEBRTC_P2P_BASE_TRANSPORTINFO_H_
diff --git a/p2p/base/turnport.cc b/p2p/base/turnport.cc
index 2908d71..b38317b 100644
--- a/p2p/base/turnport.cc
+++ b/p2p/base/turnport.cc
@@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/turnport.h"
+#include "webrtc/p2p/base/turnport.h"
#include <functional>
-#include "talk/p2p/base/common.h"
-#include "talk/p2p/base/stun.h"
+#include "webrtc/p2p/base/common.h"
+#include "webrtc/p2p/base/stun.h"
#include "webrtc/base/asyncpacketsocket.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/common.h"
@@ -588,10 +588,18 @@
void TurnPort::OnAllocateSuccess(const rtc::SocketAddress& address,
const rtc::SocketAddress& stun_address) {
connected_ = true;
+
+ rtc::SocketAddress related_address = stun_address;
+ if (!(candidate_filter() & CF_REFLEXIVE)) {
+ // If candidate filter only allows relay type of address, empty raddr to
+ // avoid local address leakage.
+ related_address = rtc::EmptySocketAddressWithFamily(stun_address.family());
+ }
+
// For relayed candidate, Base is the candidate itself.
- AddAddress(address, // Candidate address.
- address, // Base address.
- stun_address, // Related address.
+ AddAddress(address, // Candidate address.
+ address, // Base address.
+ related_address, // Related address.
UDP_PROTOCOL_NAME,
"", // TCP canddiate type, empty for turn candidates.
RELAY_PORT_TYPE,
diff --git a/p2p/base/turnport.h b/p2p/base/turnport.h
index ab7d4e7..6345aea 100644
--- a/p2p/base/turnport.h
+++ b/p2p/base/turnport.h
@@ -25,16 +25,16 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_TURNPORT_H_
-#define TALK_P2P_BASE_TURNPORT_H_
+#ifndef WEBRTC_P2P_BASE_TURNPORT_H_
+#define WEBRTC_P2P_BASE_TURNPORT_H_
#include <stdio.h>
#include <list>
#include <set>
#include <string>
-#include "talk/p2p/base/port.h"
-#include "talk/p2p/client/basicportallocator.h"
+#include "webrtc/p2p/base/port.h"
+#include "webrtc/p2p/client/basicportallocator.h"
#include "webrtc/base/asyncpacketsocket.h"
namespace rtc {
@@ -251,4 +251,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_TURNPORT_H_
+#endif // WEBRTC_P2P_BASE_TURNPORT_H_
diff --git a/p2p/base/turnport_unittest.cc b/p2p/base/turnport_unittest.cc
index c5261e4..95615ff 100644
--- a/p2p/base/turnport_unittest.cc
+++ b/p2p/base/turnport_unittest.cc
@@ -28,12 +28,12 @@
#include <dirent.h>
#endif
-#include "talk/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/tcpport.h"
-#include "talk/p2p/base/testturnserver.h"
-#include "talk/p2p/base/turnport.h"
-#include "talk/p2p/base/udpport.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/tcpport.h"
+#include "webrtc/p2p/base/testturnserver.h"
+#include "webrtc/p2p/base/turnport.h"
+#include "webrtc/p2p/base/udpport.h"
#include "webrtc/base/asynctcpsocket.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/dscp.h"
diff --git a/p2p/base/turnserver.cc b/p2p/base/turnserver.cc
index dbcbcd4..28096ae 100644
--- a/p2p/base/turnserver.cc
+++ b/p2p/base/turnserver.cc
@@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/turnserver.h"
+#include "webrtc/p2p/base/turnserver.h"
-#include "talk/p2p/base/asyncstuntcpsocket.h"
-#include "talk/p2p/base/common.h"
-#include "talk/p2p/base/packetsocketfactory.h"
-#include "talk/p2p/base/stun.h"
+#include "webrtc/p2p/base/asyncstuntcpsocket.h"
+#include "webrtc/p2p/base/common.h"
+#include "webrtc/p2p/base/packetsocketfactory.h"
+#include "webrtc/p2p/base/stun.h"
#include "webrtc/base/bytebuffer.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
diff --git a/p2p/base/turnserver.h b/p2p/base/turnserver.h
index 553d00c..44878fd 100644
--- a/p2p/base/turnserver.h
+++ b/p2p/base/turnserver.h
@@ -25,15 +25,15 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_TURNSERVER_H_
-#define TALK_P2P_BASE_TURNSERVER_H_
+#ifndef WEBRTC_P2P_BASE_TURNSERVER_H_
+#define WEBRTC_P2P_BASE_TURNSERVER_H_
#include <list>
#include <map>
#include <set>
#include <string>
-#include "talk/p2p/base/portinterface.h"
+#include "webrtc/p2p/base/portinterface.h"
#include "webrtc/base/asyncpacketsocket.h"
#include "webrtc/base/messagequeue.h"
#include "webrtc/base/sigslot.h"
@@ -204,4 +204,4 @@
} // namespace cricket
-#endif // TALK_P2P_BASE_TURNSERVER_H_
+#endif // WEBRTC_P2P_BASE_TURNSERVER_H_
diff --git a/p2p/base/udpport.h b/p2p/base/udpport.h
index fc981fd..35be1a1 100644
--- a/p2p/base/udpport.h
+++ b/p2p/base/udpport.h
@@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_BASE_UDPPORT_H_
-#define TALK_P2P_BASE_UDPPORT_H_
+#ifndef WEBRTC_P2P_BASE_UDPPORT_H_
+#define WEBRTC_P2P_BASE_UDPPORT_H_
// StunPort will be handling UDPPort functionality.
-#include "talk/p2p/base/stunport.h"
+#include "webrtc/p2p/base/stunport.h"
-#endif // TALK_P2P_BASE_UDPPORT_H_
+#endif // WEBRTC_P2P_BASE_UDPPORT_H_
diff --git a/p2p/client/autoportallocator.h b/p2p/client/autoportallocator.h
index ed87162..e7f2169 100644
--- a/p2p/client/autoportallocator.h
+++ b/p2p/client/autoportallocator.h
@@ -25,15 +25,15 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_CLIENT_AUTOPORTALLOCATOR_H_
-#define TALK_P2P_CLIENT_AUTOPORTALLOCATOR_H_
+#ifndef WEBRTC_P2P_CLIENT_AUTOPORTALLOCATOR_H_
+#define WEBRTC_P2P_CLIENT_AUTOPORTALLOCATOR_H_
#include <string>
#include <vector>
-#include "talk/p2p/client/httpportallocator.h"
-#include "talk/xmpp/jingleinfotask.h"
-#include "talk/xmpp/xmppclient.h"
+#include "webrtc/p2p/client/httpportallocator.h"
+#include "webrtc/libjingle/xmpp/jingleinfotask.h"
+#include "webrtc/libjingle/xmpp/xmppclient.h"
#include "webrtc/base/sigslot.h"
// This class sets the relay and stun servers using XmppClient.
@@ -66,4 +66,4 @@
}
};
-#endif // TALK_P2P_CLIENT_AUTOPORTALLOCATOR_H_
+#endif // WEBRTC_P2P_CLIENT_AUTOPORTALLOCATOR_H_
diff --git a/p2p/client/basicportallocator.cc b/p2p/client/basicportallocator.cc
index 0ec13e7..27d0db2 100644
--- a/p2p/client/basicportallocator.cc
+++ b/p2p/client/basicportallocator.cc
@@ -25,19 +25,19 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/client/basicportallocator.h"
+#include "webrtc/p2p/client/basicportallocator.h"
#include <string>
#include <vector>
-#include "talk/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/base/common.h"
-#include "talk/p2p/base/port.h"
-#include "talk/p2p/base/relayport.h"
-#include "talk/p2p/base/stunport.h"
-#include "talk/p2p/base/tcpport.h"
-#include "talk/p2p/base/turnport.h"
-#include "talk/p2p/base/udpport.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/common.h"
+#include "webrtc/p2p/base/port.h"
+#include "webrtc/p2p/base/relayport.h"
+#include "webrtc/p2p/base/stunport.h"
+#include "webrtc/p2p/base/tcpport.h"
+#include "webrtc/p2p/base/turnport.h"
+#include "webrtc/p2p/base/udpport.h"
#include "webrtc/base/common.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
@@ -497,6 +497,9 @@
port->set_send_retransmit_count_attribute((allocator_->flags() &
PORTALLOCATOR_ENABLE_STUN_RETRANSMIT_ATTRIBUTE) != 0);
+ // Push down the candidate_filter to individual port.
+ port->set_candidate_filter(allocator_->candidate_filter());
+
PortData data(port, seq);
ports_.push_back(data);
diff --git a/p2p/client/basicportallocator.h b/p2p/client/basicportallocator.h
index d424772..e2389d3 100644
--- a/p2p/client/basicportallocator.h
+++ b/p2p/client/basicportallocator.h
@@ -25,14 +25,14 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_CLIENT_BASICPORTALLOCATOR_H_
-#define TALK_P2P_CLIENT_BASICPORTALLOCATOR_H_
+#ifndef WEBRTC_P2P_CLIENT_BASICPORTALLOCATOR_H_
+#define WEBRTC_P2P_CLIENT_BASICPORTALLOCATOR_H_
#include <string>
#include <vector>
-#include "talk/p2p/base/port.h"
-#include "talk/p2p/base/portallocator.h"
+#include "webrtc/p2p/base/port.h"
+#include "webrtc/p2p/base/portallocator.h"
#include "webrtc/base/messagequeue.h"
#include "webrtc/base/network.h"
#include "webrtc/base/scoped_ptr.h"
@@ -255,4 +255,4 @@
} // namespace cricket
-#endif // TALK_P2P_CLIENT_BASICPORTALLOCATOR_H_
+#endif // WEBRTC_P2P_CLIENT_BASICPORTALLOCATOR_H_
diff --git a/p2p/client/connectivitychecker.cc b/p2p/client/connectivitychecker.cc
index 723c5a1..38a7dd2 100644
--- a/p2p/client/connectivitychecker.cc
+++ b/p2p/client/connectivitychecker.cc
@@ -27,14 +27,14 @@
#include <string>
-#include "talk/p2p/client/connectivitychecker.h"
+#include "webrtc/p2p/client/connectivitychecker.h"
-#include "talk/p2p/base/candidate.h"
-#include "talk/p2p/base/common.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/port.h"
-#include "talk/p2p/base/relayport.h"
-#include "talk/p2p/base/stunport.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/common.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/port.h"
+#include "webrtc/p2p/base/relayport.h"
+#include "webrtc/p2p/base/stunport.h"
#include "webrtc/base/asynchttprequest.h"
#include "webrtc/base/autodetectproxy.h"
#include "webrtc/base/helpers.h"
diff --git a/p2p/client/connectivitychecker.h b/p2p/client/connectivitychecker.h
index d4cda1e..ba3d003 100644
--- a/p2p/client/connectivitychecker.h
+++ b/p2p/client/connectivitychecker.h
@@ -1,14 +1,14 @@
// Copyright 2011 Google Inc. All Rights Reserved.
-#ifndef TALK_P2P_CLIENT_CONNECTIVITYCHECKER_H_
-#define TALK_P2P_CLIENT_CONNECTIVITYCHECKER_H_
+#ifndef WEBRTC_P2P_CLIENT_CONNECTIVITYCHECKER_H_
+#define WEBRTC_P2P_CLIENT_CONNECTIVITYCHECKER_H_
#include <map>
#include <string>
-#include "talk/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/client/httpportallocator.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/client/httpportallocator.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/messagehandler.h"
#include "webrtc/base/network.h"
@@ -271,4 +271,4 @@
} // namespace cricket
-#endif // TALK_P2P_CLIENT_CONNECTIVITYCHECKER_H_
+#endif // WEBRTC_P2P_CLIENT_CONNECTIVITYCHECKER_H_
diff --git a/p2p/client/connectivitychecker_unittest.cc b/p2p/client/connectivitychecker_unittest.cc
index 187505a..5a2a608 100644
--- a/p2p/client/connectivitychecker_unittest.cc
+++ b/p2p/client/connectivitychecker_unittest.cc
@@ -27,11 +27,11 @@
#include <string>
-#include "talk/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/base/relayport.h"
-#include "talk/p2p/base/stunport.h"
-#include "talk/p2p/client/connectivitychecker.h"
-#include "talk/p2p/client/httpportallocator.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/relayport.h"
+#include "webrtc/p2p/base/stunport.h"
+#include "webrtc/p2p/client/connectivitychecker.h"
+#include "webrtc/p2p/client/httpportallocator.h"
#include "webrtc/base/asynchttprequest.h"
#include "webrtc/base/fakenetwork.h"
#include "webrtc/base/gunit.h"
diff --git a/p2p/client/fakeportallocator.h b/p2p/client/fakeportallocator.h
index e1a04dd..04c5f47 100644
--- a/p2p/client/fakeportallocator.h
+++ b/p2p/client/fakeportallocator.h
@@ -25,13 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_CLIENT_FAKEPORTALLOCATOR_H_
-#define TALK_P2P_CLIENT_FAKEPORTALLOCATOR_H_
+#ifndef WEBRTC_P2P_CLIENT_FAKEPORTALLOCATOR_H_
+#define WEBRTC_P2P_CLIENT_FAKEPORTALLOCATOR_H_
#include <string>
-#include "talk/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/base/portallocator.h"
-#include "talk/p2p/base/udpport.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/portallocator.h"
+#include "webrtc/p2p/base/udpport.h"
#include "webrtc/base/scoped_ptr.h"
namespace rtc {
@@ -135,4 +135,4 @@
} // namespace cricket
-#endif // TALK_P2P_CLIENT_FAKEPORTALLOCATOR_H_
+#endif // WEBRTC_P2P_CLIENT_FAKEPORTALLOCATOR_H_
diff --git a/p2p/client/httpportallocator.cc b/p2p/client/httpportallocator.cc
index 31c9b51..ce52593 100644
--- a/p2p/client/httpportallocator.cc
+++ b/p2p/client/httpportallocator.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/client/httpportallocator.h"
+#include "webrtc/p2p/client/httpportallocator.h"
#include <algorithm>
#include <map>
diff --git a/p2p/client/httpportallocator.h b/p2p/client/httpportallocator.h
index 7ace943..cab8e7e 100644
--- a/p2p/client/httpportallocator.h
+++ b/p2p/client/httpportallocator.h
@@ -25,14 +25,14 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_CLIENT_HTTPPORTALLOCATOR_H_
-#define TALK_P2P_CLIENT_HTTPPORTALLOCATOR_H_
+#ifndef WEBRTC_P2P_CLIENT_HTTPPORTALLOCATOR_H_
+#define WEBRTC_P2P_CLIENT_HTTPPORTALLOCATOR_H_
#include <list>
#include <string>
#include <vector>
-#include "talk/p2p/client/basicportallocator.h"
+#include "webrtc/p2p/client/basicportallocator.h"
class HttpPortAllocatorTest_TestSessionRequestUrl_Test;
@@ -187,4 +187,4 @@
} // namespace cricket
-#endif // TALK_P2P_CLIENT_HTTPPORTALLOCATOR_H_
+#endif // WEBRTC_P2P_CLIENT_HTTPPORTALLOCATOR_H_
diff --git a/p2p/client/portallocator_unittest.cc b/p2p/client/portallocator_unittest.cc
index 84b8d03..5b1a869 100644
--- a/p2p/client/portallocator_unittest.cc
+++ b/p2p/client/portallocator_unittest.cc
@@ -25,15 +25,15 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/base/basicpacketsocketfactory.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/p2ptransportchannel.h"
-#include "talk/p2p/base/portallocatorsessionproxy.h"
-#include "talk/p2p/base/testrelayserver.h"
-#include "talk/p2p/base/teststunserver.h"
-#include "talk/p2p/base/testturnserver.h"
-#include "talk/p2p/client/basicportallocator.h"
-#include "talk/p2p/client/httpportallocator.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/p2ptransportchannel.h"
+#include "webrtc/p2p/base/portallocatorsessionproxy.h"
+#include "webrtc/p2p/base/testrelayserver.h"
+#include "webrtc/p2p/base/teststunserver.h"
+#include "webrtc/p2p/base/testturnserver.h"
+#include "webrtc/p2p/client/basicportallocator.h"
+#include "webrtc/p2p/client/httpportallocator.h"
#include "webrtc/base/fakenetwork.h"
#include "webrtc/base/firewallsocketserver.h"
#include "webrtc/base/gunit.h"
@@ -113,6 +113,8 @@
candidate_allocation_done_(false) {
cricket::ServerAddresses stun_servers;
stun_servers.insert(kStunAddr);
+ // Passing the addresses of GTURN servers will enable GTURN in
+ // Basicportallocator.
allocator_.reset(new cricket::BasicPortAllocator(
&network_manager_,
stun_servers,
@@ -137,6 +139,14 @@
allocator().set_step_delay(cricket::kMinimumStepDelay);
}
+ // Create a BasicPortAllocator without GTURN and add the TURN servers.
+ void ResetWithTurnServers(const rtc::SocketAddress& udp_turn,
+ const rtc::SocketAddress& tcp_turn) {
+ allocator_.reset(new cricket::BasicPortAllocator(&network_manager_));
+ allocator().set_step_delay(cricket::kMinimumStepDelay);
+ AddTurnServers(udp_turn, tcp_turn);
+ }
+
void AddTurnServers(const rtc::SocketAddress& udp_turn,
const rtc::SocketAddress& tcp_turn) {
cricket::RelayServerConfig relay_server(cricket::RELAY_TURN);
@@ -565,17 +575,32 @@
}
// Test ICE candidate filter mechanism with options Relay/Host/Reflexive.
+// This test also verifies that when the allocator is only allowed to use
+// relay (i.e. IceTransportsType is relay), the raddr is an empty
+// address with the correct family. This is to prevent any local
+// reflective address leakage in the sdp line.
TEST_F(PortAllocatorTest, TestCandidateFilterWithRelayOnly) {
AddInterface(kClientAddr);
+ // GTURN is not configured here.
+ ResetWithTurnServers(kTurnUdpIntAddr, rtc::SocketAddress());
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_PRED5(CheckCandidate,
+ candidates_[0],
+ cricket::ICE_CANDIDATE_COMPONENT_RTP,
+ "relay",
+ "udp",
+ rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
+
+ EXPECT_EQ(1U, 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());
+ EXPECT_EQ(
+ candidates_[0].related_address(),
+ rtc::EmptySocketAddressWithFamily(candidates_[0].address().family()));
}
}
@@ -611,6 +636,9 @@
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());
+ EXPECT_EQ(
+ candidates_[0].related_address(),
+ rtc::EmptySocketAddressWithFamily(candidates_[0].address().family()));
}
}
diff --git a/p2p/client/sessionmanagertask.h b/p2p/client/sessionmanagertask.h
index e16d9d6..753d938 100644
--- a/p2p/client/sessionmanagertask.h
+++ b/p2p/client/sessionmanagertask.h
@@ -25,13 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_CLIENT_SESSIONMANAGERTASK_H_
-#define TALK_P2P_CLIENT_SESSIONMANAGERTASK_H_
+#ifndef WEBRTC_P2P_CLIENT_SESSIONMANAGERTASK_H_
+#define WEBRTC_P2P_CLIENT_SESSIONMANAGERTASK_H_
-#include "talk/p2p/base/sessionmanager.h"
-#include "talk/p2p/client/sessionsendtask.h"
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/p2p/base/sessionmanager.h"
+#include "webrtc/p2p/client/sessionsendtask.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
namespace cricket {
@@ -90,4 +90,4 @@
} // namespace cricket
-#endif // TALK_P2P_CLIENT_SESSIONMANAGERTASK_H_
+#endif // WEBRTC_P2P_CLIENT_SESSIONMANAGERTASK_H_
diff --git a/p2p/client/sessionsendtask.h b/p2p/client/sessionsendtask.h
index c8734d2..7dafc1e 100644
--- a/p2p/client/sessionsendtask.h
+++ b/p2p/client/sessionsendtask.h
@@ -25,14 +25,14 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_CLIENT_SESSIONSENDTASK_H_
-#define TALK_P2P_CLIENT_SESSIONSENDTASK_H_
+#ifndef WEBRTC_P2P_CLIENT_SESSIONSENDTASK_H_
+#define WEBRTC_P2P_CLIENT_SESSIONSENDTASK_H_
-#include "talk/p2p/base/sessionmanager.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/xmppclient.h"
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/p2p/base/sessionmanager.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/xmppclient.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
#include "webrtc/base/common.h"
namespace cricket {
@@ -142,4 +142,4 @@
}
-#endif // TALK_P2P_CLIENT_SESSIONSENDTASK_H_
+#endif // WEBRTC_P2P_CLIENT_SESSIONSENDTASK_H_
diff --git a/p2p/client/socketmonitor.cc b/p2p/client/socketmonitor.cc
index 1924c70..a3a33e3 100644
--- a/p2p/client/socketmonitor.cc
+++ b/p2p/client/socketmonitor.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/p2p/client/socketmonitor.h"
+#include "webrtc/p2p/client/socketmonitor.h"
#include "webrtc/base/common.h"
diff --git a/p2p/client/socketmonitor.h b/p2p/client/socketmonitor.h
index 87b11cf..77241fe 100644
--- a/p2p/client/socketmonitor.h
+++ b/p2p/client/socketmonitor.h
@@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#ifndef TALK_P2P_CLIENT_SOCKETMONITOR_H_
-#define TALK_P2P_CLIENT_SOCKETMONITOR_H_
+#ifndef WEBRTC_P2P_CLIENT_SOCKETMONITOR_H_
+#define WEBRTC_P2P_CLIENT_SOCKETMONITOR_H_
#include <vector>
-#include "talk/p2p/base/transportchannel.h"
+#include "webrtc/p2p/base/transportchannel.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/thread.h"
@@ -68,4 +68,4 @@
} // namespace cricket
-#endif // TALK_P2P_CLIENT_SOCKETMONITOR_H_
+#endif // WEBRTC_P2P_CLIENT_SOCKETMONITOR_H_
diff --git a/session/media/audiomonitor.h b/session/media/audiomonitor.h
index e5585a4..454f0fe 100644
--- a/session/media/audiomonitor.h
+++ b/session/media/audiomonitor.h
@@ -29,7 +29,7 @@
#define TALK_SESSION_MEDIA_AUDIOMONITOR_H_
#include <vector>
-#include "talk/p2p/base/port.h"
+#include "webrtc/p2p/base/port.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/thread.h"
diff --git a/session/media/call.cc b/session/media/call.cc
index ad28130..7e59ac6 100644
--- a/session/media/call.cc
+++ b/session/media/call.cc
@@ -28,7 +28,7 @@
#include <string>
#include "talk/media/base/constants.h"
#include "talk/media/base/screencastid.h"
-#include "talk/p2p/base/parsing.h"
+#include "webrtc/p2p/base/parsing.h"
#include "talk/session/media/call.h"
#include "talk/session/media/currentspeakermonitor.h"
#include "talk/session/media/mediasessionclient.h"
diff --git a/session/media/call.h b/session/media/call.h
index be9397e..5a2d49b 100644
--- a/session/media/call.h
+++ b/session/media/call.h
@@ -37,13 +37,13 @@
#include "talk/media/base/screencastid.h"
#include "talk/media/base/streamparams.h"
#include "talk/media/base/videocommon.h"
-#include "talk/p2p/base/session.h"
-#include "talk/p2p/client/socketmonitor.h"
+#include "webrtc/p2p/base/session.h"
+#include "webrtc/p2p/client/socketmonitor.h"
#include "talk/session/media/audiomonitor.h"
#include "talk/session/media/currentspeakermonitor.h"
#include "talk/session/media/mediamessages.h"
#include "talk/session/media/mediasession.h"
-#include "talk/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/jid.h"
#include "webrtc/base/messagequeue.h"
namespace cricket {
diff --git a/session/media/channel.cc b/session/media/channel.cc
index 3615da8..414a131 100644
--- a/session/media/channel.cc
+++ b/session/media/channel.cc
@@ -29,7 +29,7 @@
#include "talk/media/base/constants.h"
#include "talk/media/base/rtputils.h"
-#include "talk/p2p/base/transportchannel.h"
+#include "webrtc/p2p/base/transportchannel.h"
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/mediamessages.h"
#include "talk/session/media/typingmonitor.h"
diff --git a/session/media/channel.h b/session/media/channel.h
index 6da29a4..29980bb 100644
--- a/session/media/channel.h
+++ b/session/media/channel.h
@@ -35,8 +35,8 @@
#include "talk/media/base/mediaengine.h"
#include "talk/media/base/streamparams.h"
#include "talk/media/base/videocapturer.h"
-#include "talk/p2p/base/session.h"
-#include "talk/p2p/client/socketmonitor.h"
+#include "webrtc/p2p/base/session.h"
+#include "webrtc/p2p/client/socketmonitor.h"
#include "talk/session/media/audiomonitor.h"
#include "talk/session/media/bundlefilter.h"
#include "talk/session/media/mediamonitor.h"
diff --git a/session/media/channel_unittest.cc b/session/media/channel_unittest.cc
index 698a34a..0fba14e 100644
--- a/session/media/channel_unittest.cc
+++ b/session/media/channel_unittest.cc
@@ -31,7 +31,7 @@
#include "talk/media/base/rtpdump.h"
#include "talk/media/base/screencastid.h"
#include "talk/media/base/testutils.h"
-#include "talk/p2p/base/fakesession.h"
+#include "webrtc/p2p/base/fakesession.h"
#include "talk/session/media/channel.h"
#include "talk/session/media/mediamessages.h"
#include "talk/session/media/mediarecorder.h"
diff --git a/session/media/channelmanager.h b/session/media/channelmanager.h
index 98a3c96..764451d 100644
--- a/session/media/channelmanager.h
+++ b/session/media/channelmanager.h
@@ -33,7 +33,7 @@
#include "talk/media/base/capturemanager.h"
#include "talk/media/base/mediaengine.h"
-#include "talk/p2p/base/session.h"
+#include "webrtc/p2p/base/session.h"
#include "talk/session/media/voicechannel.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/fileutils.h"
diff --git a/session/media/channelmanager_unittest.cc b/session/media/channelmanager_unittest.cc
index 1363364..4c6f4ab 100644
--- a/session/media/channelmanager_unittest.cc
+++ b/session/media/channelmanager_unittest.cc
@@ -29,7 +29,7 @@
#include "talk/media/base/nullvideorenderer.h"
#include "talk/media/base/testutils.h"
#include "talk/media/devices/fakedevicemanager.h"
-#include "talk/p2p/base/fakesession.h"
+#include "webrtc/p2p/base/fakesession.h"
#include "talk/session/media/channelmanager.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
diff --git a/session/media/mediamessages.cc b/session/media/mediamessages.cc
index 6c9f681..011fd7d 100644
--- a/session/media/mediamessages.cc
+++ b/session/media/mediamessages.cc
@@ -31,8 +31,8 @@
#include "talk/session/media/mediamessages.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/parsing.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/parsing.h"
#include "talk/session/media/mediasessionclient.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "webrtc/base/logging.h"
diff --git a/session/media/mediamessages.h b/session/media/mediamessages.h
index 4a56e4b..d918e3e 100644
--- a/session/media/mediamessages.h
+++ b/session/media/mediamessages.h
@@ -41,8 +41,8 @@
#include "talk/media/base/mediachannel.h" // For RtpHeaderExtension
#include "talk/media/base/streamparams.h"
-#include "talk/p2p/base/parsing.h"
-#include "talk/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/base/parsing.h"
+#include "webrtc/p2p/base/sessiondescription.h"
#include "webrtc/base/basictypes.h"
namespace cricket {
diff --git a/session/media/mediamessages_unittest.cc b/session/media/mediamessages_unittest.cc
index 9ebd38c..a9f00e4 100644
--- a/session/media/mediamessages_unittest.cc
+++ b/session/media/mediamessages_unittest.cc
@@ -30,7 +30,7 @@
#include <string>
#include <vector>
-#include "talk/p2p/base/constants.h"
+#include "webrtc/p2p/base/constants.h"
#include "talk/session/media/mediasessionclient.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "webrtc/base/gunit.h"
diff --git a/session/media/mediarecorder_unittest.cc b/session/media/mediarecorder_unittest.cc
index 9feb22a..b71a984 100644
--- a/session/media/mediarecorder_unittest.cc
+++ b/session/media/mediarecorder_unittest.cc
@@ -28,7 +28,7 @@
#include "talk/media/base/fakemediaengine.h"
#include "talk/media/base/rtpdump.h"
#include "talk/media/base/testutils.h"
-#include "talk/p2p/base/fakesession.h"
+#include "webrtc/p2p/base/fakesession.h"
#include "talk/session/media/channel.h"
#include "talk/session/media/mediarecorder.h"
#include "webrtc/base/bytebuffer.h"
diff --git a/session/media/mediasession.cc b/session/media/mediasession.cc
index 54eeffa..96e45a6 100644
--- a/session/media/mediasession.cc
+++ b/session/media/mediasession.cc
@@ -34,10 +34,10 @@
#include "talk/media/base/constants.h"
#include "talk/media/base/cryptoparams.h"
-#include "talk/p2p/base/constants.h"
+#include "webrtc/p2p/base/constants.h"
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/srtpfilter.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
@@ -1191,8 +1191,14 @@
}
video_added = true;
} else if (IsMediaContentOfType(&*it, MEDIA_TYPE_DATA)) {
- if (!AddDataContentForOffer(options, current_description, &data_codecs,
- ¤t_streams, offer.get())) {
+ MediaSessionOptions options_copy(options);
+ if (IsSctp(static_cast<const MediaContentDescription*>(
+ it->description))) {
+ options_copy.data_channel_type = DCT_SCTP;
+ }
+ if (!AddDataContentForOffer(options_copy, current_description,
+ &data_codecs, ¤t_streams,
+ offer.get())) {
return NULL;
}
data_added = true;
diff --git a/session/media/mediasession.h b/session/media/mediasession.h
index 992439c..462ddd2 100644
--- a/session/media/mediasession.h
+++ b/session/media/mediasession.h
@@ -40,9 +40,9 @@
#include "talk/media/base/mediachannel.h"
#include "talk/media/base/mediaengine.h" // For DataChannelType
#include "talk/media/base/streamparams.h"
-#include "talk/p2p/base/sessiondescription.h"
-#include "talk/p2p/base/transport.h"
-#include "talk/p2p/base/transportdescriptionfactory.h"
+#include "webrtc/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/base/transport.h"
+#include "webrtc/p2p/base/transportdescriptionfactory.h"
#include "webrtc/base/scoped_ptr.h"
namespace cricket {
diff --git a/session/media/mediasession_unittest.cc b/session/media/mediasession_unittest.cc
index ef155f0..9ee6065 100644
--- a/session/media/mediasession_unittest.cc
+++ b/session/media/mediasession_unittest.cc
@@ -30,9 +30,9 @@
#include "talk/media/base/codec.h"
#include "talk/media/base/testutils.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/transportdescription.h"
-#include "talk/p2p/base/transportinfo.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/transportdescription.h"
+#include "webrtc/p2p/base/transportinfo.h"
#include "talk/session/media/mediasession.h"
#include "talk/session/media/srtpfilter.h"
#include "webrtc/base/fakesslidentity.h"
@@ -628,6 +628,33 @@
EXPECT_TRUE(offer->GetContentByName("data") != NULL);
}
+// Test creating an sctp data channel from an already generated offer.
+TEST_F(MediaSessionDescriptionFactoryTest, TestCreateImplicitSctpDataOffer) {
+ MediaSessionOptions opts;
+ opts.recv_audio = false;
+ opts.bundle_enabled = true;
+ opts.data_channel_type = cricket::DCT_SCTP;
+ f1_.set_secure(SEC_ENABLED);
+ rtc::scoped_ptr<SessionDescription> offer1(f1_.CreateOffer(opts, NULL));
+ ASSERT_TRUE(offer1.get() != NULL);
+ const ContentInfo* data = offer1->GetContentByName("data");
+ ASSERT_TRUE(data != NULL);
+ const MediaContentDescription* mdesc =
+ static_cast<const MediaContentDescription*>(data->description);
+ ASSERT_EQ(cricket::kMediaProtocolSctp, mdesc->protocol());
+
+ // Now set data_channel_type to 'none' (default) and make sure that the
+ // datachannel type that gets generated from the previous offer, is of the
+ // same type.
+ opts.data_channel_type = cricket::DCT_NONE;
+ rtc::scoped_ptr<SessionDescription> offer2(
+ f1_.CreateOffer(opts, offer1.get()));
+ data = offer2->GetContentByName("data");
+ ASSERT_TRUE(data != NULL);
+ mdesc = static_cast<const MediaContentDescription*>(data->description);
+ EXPECT_EQ(cricket::kMediaProtocolSctp, mdesc->protocol());
+}
+
// Create an audio, video offer without legacy StreamParams.
TEST_F(MediaSessionDescriptionFactoryTest,
TestCreateOfferWithoutLegacyStreams) {
diff --git a/session/media/mediasessionclient.cc b/session/media/mediasessionclient.cc
index 826909a..5cb7917 100644
--- a/session/media/mediasessionclient.cc
+++ b/session/media/mediasessionclient.cc
@@ -32,13 +32,13 @@
#include "talk/media/base/capturemanager.h"
#include "talk/media/base/cryptoparams.h"
#include "talk/media/sctp/sctpdataengine.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/parsing.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/parsing.h"
#include "talk/session/media/mediamessages.h"
#include "talk/session/media/srtpfilter.h"
#include "webrtc/libjingle/xmllite/qname.h"
#include "webrtc/libjingle/xmllite/xmlconstants.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
diff --git a/session/media/mediasessionclient.h b/session/media/mediasessionclient.h
index 61d1f1d..c66c221 100644
--- a/session/media/mediasessionclient.h
+++ b/session/media/mediasessionclient.h
@@ -33,10 +33,10 @@
#include <string>
#include <vector>
#include "talk/media/base/cryptoparams.h"
-#include "talk/p2p/base/session.h"
-#include "talk/p2p/base/sessionclient.h"
-#include "talk/p2p/base/sessiondescription.h"
-#include "talk/p2p/base/sessionmanager.h"
+#include "webrtc/p2p/base/session.h"
+#include "webrtc/p2p/base/sessionclient.h"
+#include "webrtc/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/base/sessionmanager.h"
#include "talk/session/media/call.h"
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/mediasession.h"
diff --git a/session/media/mediasessionclient_unittest.cc b/session/media/mediasessionclient_unittest.cc
index bb3043d..3e8a90f 100644
--- a/session/media/mediasessionclient_unittest.cc
+++ b/session/media/mediasessionclient_unittest.cc
@@ -31,13 +31,13 @@
#include "talk/media/base/fakemediaengine.h"
#include "talk/media/base/testutils.h"
#include "talk/media/devices/fakedevicemanager.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/client/basicportallocator.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/client/basicportallocator.h"
#include "talk/session/media/mediasessionclient.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/libjingle/xmpp/constants.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
diff --git a/session/media/rtcpmuxfilter.h b/session/media/rtcpmuxfilter.h
index 40951a2..948b3c3 100644
--- a/session/media/rtcpmuxfilter.h
+++ b/session/media/rtcpmuxfilter.h
@@ -28,7 +28,7 @@
#ifndef TALK_SESSION_MEDIA_RTCPMUXFILTER_H_
#define TALK_SESSION_MEDIA_RTCPMUXFILTER_H_
-#include "talk/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/base/sessiondescription.h"
#include "webrtc/base/basictypes.h"
namespace cricket {
diff --git a/session/media/srtpfilter.cc b/session/media/srtpfilter.cc
index c79bf13..5a774db 100644
--- a/session/media/srtpfilter.cc
+++ b/session/media/srtpfilter.cc
@@ -150,7 +150,7 @@
const uint8* send_key, int send_key_len,
const std::string& recv_cs,
const uint8* recv_key, int recv_key_len) {
- if (state_ == ST_ACTIVE) {
+ if (IsActive()) {
LOG(LS_ERROR) << "Tried to set SRTP Params when filter already active";
return false;
}
@@ -212,6 +212,7 @@
LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active";
return false;
}
+ ASSERT(send_session_ != NULL);
return send_session_->ProtectRtp(p, in_len, max_len, out_len);
}
@@ -221,7 +222,7 @@
LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active";
return false;
}
-
+ ASSERT(send_session_ != NULL);
return send_session_->ProtectRtp(p, in_len, max_len, out_len, index);
}
@@ -233,6 +234,7 @@
if (send_rtcp_session_) {
return send_rtcp_session_->ProtectRtcp(p, in_len, max_len, out_len);
} else {
+ ASSERT(send_session_ != NULL);
return send_session_->ProtectRtcp(p, in_len, max_len, out_len);
}
}
@@ -242,6 +244,7 @@
LOG(LS_WARNING) << "Failed to UnprotectRtp: SRTP not active";
return false;
}
+ ASSERT(recv_session_ != NULL);
return recv_session_->UnprotectRtp(p, in_len, out_len);
}
@@ -253,6 +256,7 @@
if (recv_rtcp_session_) {
return recv_rtcp_session_->UnprotectRtcp(p, in_len, out_len);
} else {
+ ASSERT(recv_session_ != NULL);
return recv_session_->UnprotectRtcp(p, in_len, out_len);
}
}
@@ -263,13 +267,16 @@
return false;
}
+ ASSERT(send_session_ != NULL);
return send_session_->GetRtpAuthParams(key, key_len, tag_len);
}
void SrtpFilter::set_signal_silent_time(uint32 signal_silent_time_in_ms) {
signal_silent_time_in_ms_ = signal_silent_time_in_ms;
- if (state_ == ST_ACTIVE) {
+ if (IsActive()) {
+ ASSERT(send_session_ != NULL);
send_session_->set_signal_silent_time(signal_silent_time_in_ms);
+ ASSERT(recv_session_ != NULL);
recv_session_->set_signal_silent_time(signal_silent_time_in_ms);
if (send_rtcp_session_)
send_rtcp_session_->set_signal_silent_time(signal_silent_time_in_ms);
@@ -292,7 +299,7 @@
offer_params_ = params;
if (state_ == ST_INIT) {
state_ = (source == CS_LOCAL) ? ST_SENTOFFER : ST_RECEIVEDOFFER;
- } else { // state >= ST_ACTIVE
+ } else if (state_ == ST_ACTIVE) {
state_ =
(source == CS_LOCAL) ? ST_SENTUPDATEDOFFER : ST_RECEIVEDUPDATEDOFFER;
}
diff --git a/session/media/srtpfilter.h b/session/media/srtpfilter.h
index ce3368f..3134418 100644
--- a/session/media/srtpfilter.h
+++ b/session/media/srtpfilter.h
@@ -34,7 +34,7 @@
#include <vector>
#include "talk/media/base/cryptoparams.h"
-#include "talk/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/base/sessiondescription.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/sigslotrepeater.h"
diff --git a/session/media/srtpfilter_unittest.cc b/session/media/srtpfilter_unittest.cc
index 19efa2a..51d701d 100644
--- a/session/media/srtpfilter_unittest.cc
+++ b/session/media/srtpfilter_unittest.cc
@@ -27,7 +27,7 @@
#include "talk/media/base/cryptoparams.h"
#include "talk/media/base/fakertp.h"
-#include "talk/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/base/sessiondescription.h"
#include "talk/session/media/srtpfilter.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/gunit.h"
@@ -82,9 +82,12 @@
const std::vector<CryptoParams>& params2) {
EXPECT_TRUE(f1_.SetOffer(params1, CS_LOCAL));
EXPECT_TRUE(f2_.SetOffer(params1, CS_REMOTE));
+ EXPECT_FALSE(f1_.IsActive());
+ EXPECT_FALSE(f2_.IsActive());
EXPECT_TRUE(f2_.SetAnswer(params2, CS_LOCAL));
EXPECT_TRUE(f1_.SetAnswer(params2, CS_REMOTE));
EXPECT_TRUE(f1_.IsActive());
+ EXPECT_TRUE(f2_.IsActive());
}
void TestProtectUnprotect(const std::string& cs1, const std::string& cs2) {
char rtp_packet[sizeof(kPcmuFrame) + 10];
@@ -139,6 +142,7 @@
// Test that we can set up the session and keys properly.
TEST_F(SrtpFilterTest, TestGoodSetupOneCipherSuite) {
EXPECT_TRUE(f1_.SetOffer(MakeVector(kTestCryptoParams1), CS_LOCAL));
+ EXPECT_FALSE(f1_.IsActive());
EXPECT_TRUE(f1_.SetAnswer(MakeVector(kTestCryptoParams2), CS_REMOTE));
EXPECT_TRUE(f1_.IsActive());
}
@@ -153,6 +157,7 @@
answer[0].tag = 2;
answer[0].cipher_suite = CS_AES_CM_128_HMAC_SHA1_32;
EXPECT_TRUE(f1_.SetOffer(offer, CS_LOCAL));
+ EXPECT_FALSE(f1_.IsActive());
EXPECT_TRUE(f1_.SetAnswer(answer, CS_REMOTE));
EXPECT_TRUE(f1_.IsActive());
}
@@ -188,6 +193,7 @@
TEST_F(SrtpFilterTest, TestGoodSetupMultipleOffers) {
EXPECT_TRUE(f1_.SetOffer(MakeVector(kTestCryptoParams1), CS_LOCAL));
EXPECT_TRUE(f1_.SetOffer(MakeVector(kTestCryptoParams2), CS_LOCAL));
+ EXPECT_FALSE(f1_.IsActive());
EXPECT_TRUE(f1_.SetAnswer(MakeVector(kTestCryptoParams2), CS_REMOTE));
EXPECT_TRUE(f1_.IsActive());
EXPECT_TRUE(f1_.SetOffer(MakeVector(kTestCryptoParams1), CS_LOCAL));
@@ -196,6 +202,7 @@
EXPECT_TRUE(f2_.SetOffer(MakeVector(kTestCryptoParams1), CS_REMOTE));
EXPECT_TRUE(f2_.SetOffer(MakeVector(kTestCryptoParams2), CS_REMOTE));
+ EXPECT_FALSE(f2_.IsActive());
EXPECT_TRUE(f2_.SetAnswer(MakeVector(kTestCryptoParams2), CS_LOCAL));
EXPECT_TRUE(f2_.IsActive());
EXPECT_TRUE(f2_.SetOffer(MakeVector(kTestCryptoParams1), CS_REMOTE));
@@ -206,6 +213,7 @@
TEST_F(SrtpFilterTest, TestBadSetupMultipleOffers) {
EXPECT_TRUE(f1_.SetOffer(MakeVector(kTestCryptoParams1), CS_LOCAL));
EXPECT_FALSE(f1_.SetOffer(MakeVector(kTestCryptoParams2), CS_REMOTE));
+ EXPECT_FALSE(f1_.IsActive());
EXPECT_TRUE(f1_.SetAnswer(MakeVector(kTestCryptoParams1), CS_REMOTE));
EXPECT_TRUE(f1_.IsActive());
EXPECT_TRUE(f1_.SetOffer(MakeVector(kTestCryptoParams2), CS_LOCAL));
@@ -214,6 +222,7 @@
EXPECT_TRUE(f2_.SetOffer(MakeVector(kTestCryptoParams2), CS_REMOTE));
EXPECT_FALSE(f2_.SetOffer(MakeVector(kTestCryptoParams1), CS_LOCAL));
+ EXPECT_FALSE(f2_.IsActive());
EXPECT_TRUE(f2_.SetAnswer(MakeVector(kTestCryptoParams2), CS_LOCAL));
EXPECT_TRUE(f2_.IsActive());
EXPECT_TRUE(f2_.SetOffer(MakeVector(kTestCryptoParams2), CS_REMOTE));
@@ -402,6 +411,8 @@
EXPECT_TRUE(f1_.SetOffer(offer, CS_LOCAL));
EXPECT_TRUE(f2_.SetOffer(offer, CS_REMOTE));
+ EXPECT_FALSE(f1_.IsActive());
+ EXPECT_FALSE(f2_.IsActive());
EXPECT_TRUE(f2_.SetProvisionalAnswer(answer, CS_LOCAL));
EXPECT_TRUE(f1_.SetProvisionalAnswer(answer, CS_REMOTE));
EXPECT_TRUE(f1_.IsActive());
@@ -425,6 +436,8 @@
EXPECT_TRUE(f1_.SetOffer(offer, CS_LOCAL));
EXPECT_TRUE(f2_.SetOffer(offer, CS_REMOTE));
+ EXPECT_FALSE(f1_.IsActive());
+ EXPECT_FALSE(f2_.IsActive());
EXPECT_TRUE(f2_.SetProvisionalAnswer(answer, CS_LOCAL));
EXPECT_TRUE(f1_.SetProvisionalAnswer(answer, CS_REMOTE));
EXPECT_FALSE(f1_.IsActive());
@@ -438,6 +451,33 @@
TestProtectUnprotect(CS_AES_CM_128_HMAC_SHA1_80, CS_AES_CM_128_HMAC_SHA1_80);
}
+// Test that if we get a new local offer after a provisional answer
+// with no crypto, that we are in an inactive state.
+TEST_F(SrtpFilterTest, TestLocalOfferAfterProvisionalAnswerWithoutCrypto) {
+ std::vector<CryptoParams> offer(MakeVector(kTestCryptoParams1));
+ std::vector<CryptoParams> answer;
+
+ EXPECT_TRUE(f1_.SetOffer(offer, CS_LOCAL));
+ EXPECT_TRUE(f2_.SetOffer(offer, CS_REMOTE));
+ EXPECT_TRUE(f1_.SetProvisionalAnswer(answer, CS_REMOTE));
+ EXPECT_TRUE(f2_.SetProvisionalAnswer(answer, CS_LOCAL));
+ EXPECT_FALSE(f1_.IsActive());
+ EXPECT_FALSE(f2_.IsActive());
+ // The calls to set an offer after a provisional answer fail, so the
+ // state doesn't change.
+ EXPECT_FALSE(f1_.SetOffer(offer, CS_LOCAL));
+ EXPECT_FALSE(f2_.SetOffer(offer, CS_REMOTE));
+ EXPECT_FALSE(f1_.IsActive());
+ EXPECT_FALSE(f2_.IsActive());
+
+ answer.push_back(kTestCryptoParams2);
+ EXPECT_TRUE(f2_.SetAnswer(answer, CS_LOCAL));
+ EXPECT_TRUE(f1_.SetAnswer(answer, CS_REMOTE));
+ EXPECT_TRUE(f1_.IsActive());
+ EXPECT_TRUE(f2_.IsActive());
+ TestProtectUnprotect(CS_AES_CM_128_HMAC_SHA1_80, CS_AES_CM_128_HMAC_SHA1_80);
+}
+
// Test that we can disable encryption.
TEST_F(SrtpFilterTest, TestDisableEncryption) {
std::vector<CryptoParams> offer(MakeVector(kTestCryptoParams1));
diff --git a/session/media/typingmonitor_unittest.cc b/session/media/typingmonitor_unittest.cc
index 95fa7a4..66e2401 100644
--- a/session/media/typingmonitor_unittest.cc
+++ b/session/media/typingmonitor_unittest.cc
@@ -26,7 +26,7 @@
*/
#include "talk/media/base/fakemediaengine.h"
-#include "talk/p2p/base/fakesession.h"
+#include "webrtc/p2p/base/fakesession.h"
#include "talk/session/media/channel.h"
#include "talk/session/media/currentspeakermonitor.h"
#include "talk/session/media/typingmonitor.h"
diff --git a/session/tunnel/pseudotcpchannel.cc b/session/tunnel/pseudotcpchannel.cc
index f277614..861e178 100644
--- a/session/tunnel/pseudotcpchannel.cc
+++ b/session/tunnel/pseudotcpchannel.cc
@@ -27,8 +27,8 @@
#include <string>
#include "pseudotcpchannel.h"
-#include "talk/p2p/base/candidate.h"
-#include "talk/p2p/base/transportchannel.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/transportchannel.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
diff --git a/session/tunnel/pseudotcpchannel.h b/session/tunnel/pseudotcpchannel.h
index 5ccfa4c..ad6d907 100644
--- a/session/tunnel/pseudotcpchannel.h
+++ b/session/tunnel/pseudotcpchannel.h
@@ -28,8 +28,8 @@
#ifndef TALK_SESSION_TUNNEL_PSEUDOTCPCHANNEL_H_
#define TALK_SESSION_TUNNEL_PSEUDOTCPCHANNEL_H_
-#include "talk/p2p/base/pseudotcp.h"
-#include "talk/p2p/base/session.h"
+#include "webrtc/p2p/base/pseudotcp.h"
+#include "webrtc/p2p/base/session.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/messagequeue.h"
#include "webrtc/base/stream.h"
diff --git a/session/tunnel/securetunnelsessionclient.cc b/session/tunnel/securetunnelsessionclient.cc
index cb41c3b..4ed8185 100644
--- a/session/tunnel/securetunnelsessionclient.cc
+++ b/session/tunnel/securetunnelsessionclient.cc
@@ -27,7 +27,7 @@
// SecureTunnelSessionClient and SecureTunnelSession implementation.
-#include "talk/p2p/base/transportchannel.h"
+#include "webrtc/p2p/base/transportchannel.h"
#include "talk/session/tunnel/pseudotcpchannel.h"
#include "talk/session/tunnel/securetunnelsessionclient.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
diff --git a/session/tunnel/tunnelsessionclient.cc b/session/tunnel/tunnelsessionclient.cc
index 7221db4..7d2a7d1 100644
--- a/session/tunnel/tunnelsessionclient.cc
+++ b/session/tunnel/tunnelsessionclient.cc
@@ -26,8 +26,8 @@
*/
#include "pseudotcpchannel.h"
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/transportchannel.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/transportchannel.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
#include "tunnelsessionclient.h"
#include "webrtc/base/basicdefs.h"
diff --git a/session/tunnel/tunnelsessionclient.h b/session/tunnel/tunnelsessionclient.h
index d3fa64a..87beaed 100644
--- a/session/tunnel/tunnelsessionclient.h
+++ b/session/tunnel/tunnelsessionclient.h
@@ -30,14 +30,14 @@
#include <vector>
-#include "talk/p2p/base/constants.h"
-#include "talk/p2p/base/pseudotcp.h"
-#include "talk/p2p/base/session.h"
-#include "talk/p2p/base/sessionclient.h"
-#include "talk/p2p/base/sessiondescription.h"
-#include "talk/p2p/base/sessionmanager.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/pseudotcp.h"
+#include "webrtc/p2p/base/session.h"
+#include "webrtc/p2p/base/sessionclient.h"
+#include "webrtc/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/base/sessionmanager.h"
#include "webrtc/libjingle/xmllite/qname.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/stream.h"
diff --git a/session/tunnel/tunnelsessionclient_unittest.cc b/session/tunnel/tunnelsessionclient_unittest.cc
index 6d46918..6bd746a 100644
--- a/session/tunnel/tunnelsessionclient_unittest.cc
+++ b/session/tunnel/tunnelsessionclient_unittest.cc
@@ -26,9 +26,9 @@
*/
#include <string>
-#include "talk/p2p/base/sessionmanager.h"
-#include "talk/p2p/base/transport.h"
-#include "talk/p2p/client/fakeportallocator.h"
+#include "webrtc/p2p/base/sessionmanager.h"
+#include "webrtc/p2p/base/transport.h"
+#include "webrtc/p2p/client/fakeportallocator.h"
#include "talk/session/tunnel/tunnelsessionclient.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/messagehandler.h"
diff --git a/xmpp/chatroommodule.h b/xmpp/chatroommodule.h
index 8358fc1..145ab89 100644
--- a/xmpp/chatroommodule.h
+++ b/xmpp/chatroommodule.h
@@ -28,8 +28,8 @@
#ifndef TALK_XMPP_CHATROOMMODULE_H_
#define TALK_XMPP_CHATROOMMODULE_H_
-#include "talk/xmpp/module.h"
-#include "talk/xmpp/rostermodule.h"
+#include "webrtc/libjingle/xmpp/module.h"
+#include "webrtc/libjingle/xmpp/rostermodule.h"
namespace buzz {
diff --git a/xmpp/chatroommoduleimpl.cc b/xmpp/chatroommoduleimpl.cc
index 45db014..33c752e 100644
--- a/xmpp/chatroommoduleimpl.cc
+++ b/xmpp/chatroommoduleimpl.cc
@@ -31,9 +31,9 @@
#include <sstream>
#include <string>
#include <vector>
-#include "talk/xmpp/chatroommodule.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/moduleimpl.h"
+#include "webrtc/libjingle/xmpp/chatroommodule.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/moduleimpl.h"
#include "webrtc/base/common.h"
namespace buzz {
diff --git a/xmpp/constants.cc b/xmpp/constants.cc
index 297eafd..fce9c75 100644
--- a/xmpp/constants.cc
+++ b/xmpp/constants.cc
@@ -25,14 +25,14 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include <string>
#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/libjingle/xmpp/jid.h"
#include "webrtc/base/basicdefs.h"
namespace buzz {
diff --git a/xmpp/constants.h b/xmpp/constants.h
index 6aa1a54..52088ca 100644
--- a/xmpp/constants.h
+++ b/xmpp/constants.h
@@ -30,7 +30,7 @@
#include <string>
#include "webrtc/libjingle/xmllite/qname.h"
-#include "talk/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/jid.h"
namespace buzz {
diff --git a/xmpp/discoitemsquerytask.cc b/xmpp/discoitemsquerytask.cc
index d900b85..b739ba3 100644
--- a/xmpp/discoitemsquerytask.cc
+++ b/xmpp/discoitemsquerytask.cc
@@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/discoitemsquerytask.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/discoitemsquerytask.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
#include "webrtc/base/scoped_ptr.h"
namespace buzz {
diff --git a/xmpp/discoitemsquerytask.h b/xmpp/discoitemsquerytask.h
index 409fc90..c31d261 100644
--- a/xmpp/discoitemsquerytask.h
+++ b/xmpp/discoitemsquerytask.h
@@ -54,7 +54,7 @@
#include <string>
#include <vector>
-#include "talk/xmpp/iqtask.h"
+#include "webrtc/libjingle/xmpp/iqtask.h"
namespace buzz {
diff --git a/xmpp/fakexmppclient.h b/xmpp/fakexmppclient.h
index 3522ba9..2e37dd1 100644
--- a/xmpp/fakexmppclient.h
+++ b/xmpp/fakexmppclient.h
@@ -33,7 +33,7 @@
#include <string>
#include <vector>
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
namespace buzz {
diff --git a/xmpp/hangoutpubsubclient.cc b/xmpp/hangoutpubsubclient.cc
index 63f5bcf..dacccf1 100644
--- a/xmpp/hangoutpubsubclient.cc
+++ b/xmpp/hangoutpubsubclient.cc
@@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/hangoutpubsubclient.h"
+#include "webrtc/libjingle/xmpp/hangoutpubsubclient.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/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/jid.h"
#include "webrtc/base/logging.h"
diff --git a/xmpp/hangoutpubsubclient.h b/xmpp/hangoutpubsubclient.h
index 5692fc6..03c96f2 100644
--- a/xmpp/hangoutpubsubclient.h
+++ b/xmpp/hangoutpubsubclient.h
@@ -32,9 +32,9 @@
#include <string>
#include <vector>
-#include "talk/xmpp/jid.h"
-#include "talk/xmpp/pubsubclient.h"
-#include "talk/xmpp/pubsubstateclient.h"
+#include "webrtc/libjingle/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/pubsubclient.h"
+#include "webrtc/libjingle/xmpp/pubsubstateclient.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/sigslotrepeater.h"
diff --git a/xmpp/hangoutpubsubclient_unittest.cc b/xmpp/hangoutpubsubclient_unittest.cc
index 555ee5c..574e21a 100644
--- a/xmpp/hangoutpubsubclient_unittest.cc
+++ b/xmpp/hangoutpubsubclient_unittest.cc
@@ -5,10 +5,10 @@
#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"
-#include "talk/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/fakexmppclient.h"
+#include "webrtc/libjingle/xmpp/hangoutpubsubclient.h"
+#include "webrtc/libjingle/xmpp/jid.h"
#include "webrtc/base/faketaskrunner.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/sigslot.h"
diff --git a/xmpp/iqtask.cc b/xmpp/iqtask.cc
index f6a21d4..d4cf310 100644
--- a/xmpp/iqtask.cc
+++ b/xmpp/iqtask.cc
@@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/iqtask.h"
+#include "webrtc/libjingle/xmpp/iqtask.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/xmppclient.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/xmppclient.h"
namespace buzz {
diff --git a/xmpp/iqtask.h b/xmpp/iqtask.h
index 312c4af..8b0c04b 100644
--- a/xmpp/iqtask.h
+++ b/xmpp/iqtask.h
@@ -30,8 +30,8 @@
#include <string>
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
namespace buzz {
diff --git a/xmpp/jid.cc b/xmpp/jid.cc
index 702e477..0410458 100644
--- a/xmpp/jid.cc
+++ b/xmpp/jid.cc
@@ -25,14 +25,14 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/jid.h"
#include <ctype.h>
#include <algorithm>
#include <string>
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
diff --git a/xmpp/jid_unittest.cc b/xmpp/jid_unittest.cc
index 05835cf..9cae2f8 100644
--- a/xmpp/jid_unittest.cc
+++ b/xmpp/jid_unittest.cc
@@ -1,7 +1,7 @@
// Copyright 2004 Google Inc. All Rights Reserved
-#include "talk/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/jid.h"
#include "webrtc/base/gunit.h"
using buzz::Jid;
diff --git a/xmpp/jingleinfotask.cc b/xmpp/jingleinfotask.cc
index 3ceae07..3b8fdde 100644
--- a/xmpp/jingleinfotask.cc
+++ b/xmpp/jingleinfotask.cc
@@ -25,11 +25,11 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/jingleinfotask.h"
+#include "webrtc/libjingle/xmpp/jingleinfotask.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/xmppclient.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/xmppclient.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
#include "webrtc/base/socketaddress.h"
namespace buzz {
diff --git a/xmpp/jingleinfotask.h b/xmpp/jingleinfotask.h
index 18cb5a9..ac03ecf 100644
--- a/xmpp/jingleinfotask.h
+++ b/xmpp/jingleinfotask.h
@@ -30,9 +30,9 @@
#include <vector>
-#include "talk/p2p/client/httpportallocator.h"
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/p2p/client/httpportallocator.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
#include "webrtc/base/sigslot.h"
namespace buzz {
diff --git a/xmpp/module.h b/xmpp/module.h
index a5d0687..f1cff11 100644
--- a/xmpp/module.h
+++ b/xmpp/module.h
@@ -28,7 +28,7 @@
#ifndef TALK_XMPP_MODULE_H_
#define TALK_XMPP_MODULE_H_
-#include "talk/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
namespace buzz {
diff --git a/xmpp/moduleimpl.cc b/xmpp/moduleimpl.cc
index 4635bee..8ac755d 100644
--- a/xmpp/moduleimpl.cc
+++ b/xmpp/moduleimpl.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/moduleimpl.h"
+#include "webrtc/libjingle/xmpp/moduleimpl.h"
#include "webrtc/base/common.h"
namespace buzz {
diff --git a/xmpp/moduleimpl.h b/xmpp/moduleimpl.h
index 897bfce..f1986f3 100644
--- a/xmpp/moduleimpl.h
+++ b/xmpp/moduleimpl.h
@@ -28,8 +28,8 @@
#ifndef TALK_XMPP_MODULEIMPL_H_
#define TALK_XMPP_MODULEIMPL_H_
-#include "talk/xmpp/module.h"
-#include "talk/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/module.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
namespace buzz {
diff --git a/xmpp/mucroomconfigtask.cc b/xmpp/mucroomconfigtask.cc
index 536eb9b..86ba5f6 100644
--- a/xmpp/mucroomconfigtask.cc
+++ b/xmpp/mucroomconfigtask.cc
@@ -28,9 +28,9 @@
#include <string>
#include <vector>
-#include "talk/xmpp/mucroomconfigtask.h"
+#include "webrtc/libjingle/xmpp/mucroomconfigtask.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/scoped_ptr.h"
namespace buzz {
diff --git a/xmpp/mucroomconfigtask.h b/xmpp/mucroomconfigtask.h
index ba0dbaa..5626d3e 100644
--- a/xmpp/mucroomconfigtask.h
+++ b/xmpp/mucroomconfigtask.h
@@ -29,7 +29,7 @@
#define TALK_XMPP_MUCROOMCONFIGTASK_H_
#include <string>
-#include "talk/xmpp/iqtask.h"
+#include "webrtc/libjingle/xmpp/iqtask.h"
namespace buzz {
diff --git a/xmpp/mucroomconfigtask_unittest.cc b/xmpp/mucroomconfigtask_unittest.cc
index bf5e7b4..e9e1281 100644
--- a/xmpp/mucroomconfigtask_unittest.cc
+++ b/xmpp/mucroomconfigtask_unittest.cc
@@ -29,9 +29,9 @@
#include <vector>
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/fakexmppclient.h"
-#include "talk/xmpp/mucroomconfigtask.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/fakexmppclient.h"
+#include "webrtc/libjingle/xmpp/mucroomconfigtask.h"
#include "webrtc/base/faketaskrunner.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/sigslot.h"
diff --git a/xmpp/mucroomdiscoverytask.cc b/xmpp/mucroomdiscoverytask.cc
index c7477ae..0ec4fbe 100644
--- a/xmpp/mucroomdiscoverytask.cc
+++ b/xmpp/mucroomdiscoverytask.cc
@@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/mucroomdiscoverytask.h"
+#include "webrtc/libjingle/xmpp/mucroomdiscoverytask.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
namespace buzz {
diff --git a/xmpp/mucroomdiscoverytask.h b/xmpp/mucroomdiscoverytask.h
index 4097cc6..dc5918d 100644
--- a/xmpp/mucroomdiscoverytask.h
+++ b/xmpp/mucroomdiscoverytask.h
@@ -30,7 +30,7 @@
#include <map>
#include <string>
-#include "talk/xmpp/iqtask.h"
+#include "webrtc/libjingle/xmpp/iqtask.h"
namespace buzz {
diff --git a/xmpp/mucroomdiscoverytask_unittest.cc b/xmpp/mucroomdiscoverytask_unittest.cc
index e1a633e..77fdf86 100644
--- a/xmpp/mucroomdiscoverytask_unittest.cc
+++ b/xmpp/mucroomdiscoverytask_unittest.cc
@@ -29,9 +29,9 @@
#include <vector>
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/fakexmppclient.h"
-#include "talk/xmpp/mucroomdiscoverytask.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/fakexmppclient.h"
+#include "webrtc/libjingle/xmpp/mucroomdiscoverytask.h"
#include "webrtc/base/faketaskrunner.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/sigslot.h"
diff --git a/xmpp/mucroomlookuptask.cc b/xmpp/mucroomlookuptask.cc
index 8e1895c..6832ade 100644
--- a/xmpp/mucroomlookuptask.cc
+++ b/xmpp/mucroomlookuptask.cc
@@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/mucroomlookuptask.h"
+#include "webrtc/libjingle/xmpp/mucroomlookuptask.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
diff --git a/xmpp/mucroomlookuptask.h b/xmpp/mucroomlookuptask.h
index 48f2484..9bcca2c 100644
--- a/xmpp/mucroomlookuptask.h
+++ b/xmpp/mucroomlookuptask.h
@@ -29,7 +29,7 @@
#define TALK_XMPP_MUCROOMLOOKUPTASK_H_
#include <string>
-#include "talk/xmpp/iqtask.h"
+#include "webrtc/libjingle/xmpp/iqtask.h"
namespace buzz {
diff --git a/xmpp/mucroomlookuptask_unittest.cc b/xmpp/mucroomlookuptask_unittest.cc
index 03be292..961e664 100644
--- a/xmpp/mucroomlookuptask_unittest.cc
+++ b/xmpp/mucroomlookuptask_unittest.cc
@@ -29,9 +29,9 @@
#include <vector>
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/fakexmppclient.h"
-#include "talk/xmpp/mucroomlookuptask.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/fakexmppclient.h"
+#include "webrtc/libjingle/xmpp/mucroomlookuptask.h"
#include "webrtc/base/faketaskrunner.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/sigslot.h"
diff --git a/xmpp/mucroomuniquehangoutidtask.cc b/xmpp/mucroomuniquehangoutidtask.cc
index 78a8edf..db94741 100644
--- a/xmpp/mucroomuniquehangoutidtask.cc
+++ b/xmpp/mucroomuniquehangoutidtask.cc
@@ -1,9 +1,9 @@
// Copyright 2012 Google Inc. All Rights Reserved.
-#include "talk/xmpp/mucroomuniquehangoutidtask.h"
+#include "webrtc/libjingle/xmpp/mucroomuniquehangoutidtask.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
namespace buzz {
diff --git a/xmpp/mucroomuniquehangoutidtask.h b/xmpp/mucroomuniquehangoutidtask.h
index d222bac..83de434 100644
--- a/xmpp/mucroomuniquehangoutidtask.h
+++ b/xmpp/mucroomuniquehangoutidtask.h
@@ -4,7 +4,7 @@
#ifndef TALK_XMPP_MUCROOMUNIQUEHANGOUTIDTASK_H_
#define TALK_XMPP_MUCROOMUNIQUEHANGOUTIDTASK_H_
-#include "talk/xmpp/iqtask.h"
+#include "webrtc/libjingle/xmpp/iqtask.h"
namespace buzz {
diff --git a/xmpp/mucroomuniquehangoutidtask_unittest.cc b/xmpp/mucroomuniquehangoutidtask_unittest.cc
index 42bed13..d8f7c20 100644
--- a/xmpp/mucroomuniquehangoutidtask_unittest.cc
+++ b/xmpp/mucroomuniquehangoutidtask_unittest.cc
@@ -29,9 +29,9 @@
#include <vector>
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/fakexmppclient.h"
-#include "talk/xmpp/mucroomuniquehangoutidtask.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/fakexmppclient.h"
+#include "webrtc/libjingle/xmpp/mucroomuniquehangoutidtask.h"
#include "webrtc/base/faketaskrunner.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/sigslot.h"
diff --git a/xmpp/pingtask.cc b/xmpp/pingtask.cc
index 7922ea0..619c3a8 100644
--- a/xmpp/pingtask.cc
+++ b/xmpp/pingtask.cc
@@ -1,9 +1,9 @@
// Copyright 2011 Google Inc. All Rights Reserved.
-#include "talk/xmpp/pingtask.h"
+#include "webrtc/libjingle/xmpp/pingtask.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
diff --git a/xmpp/pingtask.h b/xmpp/pingtask.h
index 2a869a3..87d74b4 100644
--- a/xmpp/pingtask.h
+++ b/xmpp/pingtask.h
@@ -28,7 +28,7 @@
#ifndef TALK_XMPP_PINGTASK_H_
#define TALK_XMPP_PINGTASK_H_
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
#include "webrtc/base/messagehandler.h"
#include "webrtc/base/messagequeue.h"
diff --git a/xmpp/pingtask_unittest.cc b/xmpp/pingtask_unittest.cc
index fe88a5c..a0b6618 100644
--- a/xmpp/pingtask_unittest.cc
+++ b/xmpp/pingtask_unittest.cc
@@ -29,9 +29,9 @@
#include <vector>
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/fakexmppclient.h"
-#include "talk/xmpp/pingtask.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/fakexmppclient.h"
+#include "webrtc/libjingle/xmpp/pingtask.h"
#include "webrtc/base/faketaskrunner.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/sigslot.h"
diff --git a/xmpp/plainsaslhandler.h b/xmpp/plainsaslhandler.h
index 31032e4..f23782d 100644
--- a/xmpp/plainsaslhandler.h
+++ b/xmpp/plainsaslhandler.h
@@ -29,8 +29,8 @@
#define TALK_XMPP_PLAINSASLHANDLER_H_
#include <algorithm>
-#include "talk/xmpp/saslhandler.h"
-#include "talk/xmpp/saslplainmechanism.h"
+#include "webrtc/libjingle/xmpp/saslhandler.h"
+#include "webrtc/libjingle/xmpp/saslplainmechanism.h"
#include "webrtc/base/cryptstring.h"
namespace buzz {
diff --git a/xmpp/presenceouttask.cc b/xmpp/presenceouttask.cc
index a77d9d6..39ef646 100644
--- a/xmpp/presenceouttask.cc
+++ b/xmpp/presenceouttask.cc
@@ -27,9 +27,9 @@
#include <time.h>
#include <sstream>
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/presenceouttask.h"
-#include "talk/xmpp/xmppclient.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/presenceouttask.h"
+#include "webrtc/libjingle/xmpp/xmppclient.h"
#include "webrtc/base/stringencode.h"
namespace buzz {
diff --git a/xmpp/presenceouttask.h b/xmpp/presenceouttask.h
index 53bbae5..f6b04af 100644
--- a/xmpp/presenceouttask.h
+++ b/xmpp/presenceouttask.h
@@ -28,9 +28,9 @@
#ifndef TALK_XMPP_PRESENCEOUTTASK_H_
#define TALK_XMPP_PRESENCEOUTTASK_H_
-#include "talk/xmpp/presencestatus.h"
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/presencestatus.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
namespace buzz {
diff --git a/xmpp/presencereceivetask.cc b/xmpp/presencereceivetask.cc
index 940c53b..0a159ac 100644
--- a/xmpp/presencereceivetask.cc
+++ b/xmpp/presencereceivetask.cc
@@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/presencereceivetask.h"
+#include "webrtc/libjingle/xmpp/presencereceivetask.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/stringencode.h"
namespace buzz {
diff --git a/xmpp/presencereceivetask.h b/xmpp/presencereceivetask.h
index 6a090f3..80e11d8 100644
--- a/xmpp/presencereceivetask.h
+++ b/xmpp/presencereceivetask.h
@@ -30,8 +30,8 @@
#include "webrtc/base/sigslot.h"
-#include "talk/xmpp/presencestatus.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/presencestatus.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
namespace buzz {
diff --git a/xmpp/presencestatus.cc b/xmpp/presencestatus.cc
index c75b705..2ab89b2 100644
--- a/xmpp/presencestatus.cc
+++ b/xmpp/presencestatus.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/presencestatus.h"
+#include "webrtc/libjingle/xmpp/presencestatus.h"
namespace buzz {
PresenceStatus::PresenceStatus()
diff --git a/xmpp/presencestatus.h b/xmpp/presencestatus.h
index 45c5471..c133e40 100644
--- a/xmpp/presencestatus.h
+++ b/xmpp/presencestatus.h
@@ -28,8 +28,8 @@
#ifndef THIRD_PARTY_LIBJINGLE_FILES_TALK_XMPP_PRESENCESTATUS_H_
#define THIRD_PARTY_LIBJINGLE_FILES_TALK_XMPP_PRESENCESTATUS_H_
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/jid.h"
namespace buzz {
diff --git a/xmpp/prexmppauth.h b/xmpp/prexmppauth.h
index 71fcae4..7ddc85b 100644
--- a/xmpp/prexmppauth.h
+++ b/xmpp/prexmppauth.h
@@ -28,7 +28,7 @@
#ifndef TALK_XMPP_PREXMPPAUTH_H_
#define TALK_XMPP_PREXMPPAUTH_H_
-#include "talk/xmpp/saslhandler.h"
+#include "webrtc/libjingle/xmpp/saslhandler.h"
#include "webrtc/base/cryptstring.h"
#include "webrtc/base/sigslot.h"
diff --git a/xmpp/pubsub_task.cc b/xmpp/pubsub_task.cc
index 3552764..98446fd 100644
--- a/xmpp/pubsub_task.cc
+++ b/xmpp/pubsub_task.cc
@@ -25,13 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/pubsub_task.h"
+#include "webrtc/libjingle/xmpp/pubsub_task.h"
#include <map>
#include <string>
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
#include "webrtc/base/common.h"
namespace buzz {
diff --git a/xmpp/pubsub_task.h b/xmpp/pubsub_task.h
index 2787cbc..a219b60 100644
--- a/xmpp/pubsub_task.h
+++ b/xmpp/pubsub_task.h
@@ -31,8 +31,8 @@
#include <map>
#include <string>
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/jid.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
namespace buzz {
diff --git a/xmpp/pubsubclient.cc b/xmpp/pubsubclient.cc
index b627587..a5793ef 100644
--- a/xmpp/pubsubclient.cc
+++ b/xmpp/pubsubclient.cc
@@ -25,14 +25,14 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/pubsubclient.h"
+#include "webrtc/libjingle/xmpp/pubsubclient.h"
#include <string>
#include <vector>
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/jid.h"
-#include "talk/xmpp/pubsubtasks.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/pubsubtasks.h"
namespace buzz {
diff --git a/xmpp/pubsubclient.h b/xmpp/pubsubclient.h
index d7b5066..6465957 100644
--- a/xmpp/pubsubclient.h
+++ b/xmpp/pubsubclient.h
@@ -31,8 +31,8 @@
#include <string>
#include <vector>
-#include "talk/xmpp/jid.h"
-#include "talk/xmpp/pubsubtasks.h"
+#include "webrtc/libjingle/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/pubsubtasks.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/sigslotrepeater.h"
#include "webrtc/base/task.h"
diff --git a/xmpp/pubsubclient_unittest.cc b/xmpp/pubsubclient_unittest.cc
index f191a18..b3049d8 100644
--- a/xmpp/pubsubclient_unittest.cc
+++ b/xmpp/pubsubclient_unittest.cc
@@ -5,10 +5,10 @@
#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"
-#include "talk/xmpp/pubsubclient.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/fakexmppclient.h"
+#include "webrtc/libjingle/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/pubsubclient.h"
#include "webrtc/base/faketaskrunner.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/sigslot.h"
diff --git a/xmpp/pubsubstateclient.cc b/xmpp/pubsubstateclient.cc
index 5cd7b1a..2ed7b20 100644
--- a/xmpp/pubsubstateclient.cc
+++ b/xmpp/pubsubstateclient.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/pubsubstateclient.h"
+#include "webrtc/libjingle/xmpp/pubsubstateclient.h"
namespace buzz {
diff --git a/xmpp/pubsubstateclient.h b/xmpp/pubsubstateclient.h
index 09ef0f4..79b1329 100644
--- a/xmpp/pubsubstateclient.h
+++ b/xmpp/pubsubstateclient.h
@@ -34,9 +34,9 @@
#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"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/pubsubclient.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/sigslotrepeater.h"
diff --git a/xmpp/pubsubtasks.cc b/xmpp/pubsubtasks.cc
index 015708e..091a7ad 100644
--- a/xmpp/pubsubtasks.cc
+++ b/xmpp/pubsubtasks.cc
@@ -25,13 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/pubsubtasks.h"
+#include "webrtc/libjingle/xmpp/pubsubtasks.h"
#include <string>
#include <vector>
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/receivetask.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/receivetask.h"
// An implementation of the tasks for XEP-0060
// (http://xmpp.org/extensions/xep-0060.html).
diff --git a/xmpp/pubsubtasks.h b/xmpp/pubsubtasks.h
index 94be276..280c613 100644
--- a/xmpp/pubsubtasks.h
+++ b/xmpp/pubsubtasks.h
@@ -30,8 +30,8 @@
#include <vector>
-#include "talk/xmpp/iqtask.h"
-#include "talk/xmpp/receivetask.h"
+#include "webrtc/libjingle/xmpp/iqtask.h"
+#include "webrtc/libjingle/xmpp/receivetask.h"
#include "webrtc/base/sigslot.h"
namespace buzz {
diff --git a/xmpp/pubsubtasks_unittest.cc b/xmpp/pubsubtasks_unittest.cc
index 48cd04a..7ea21e7 100644
--- a/xmpp/pubsubtasks_unittest.cc
+++ b/xmpp/pubsubtasks_unittest.cc
@@ -5,11 +5,11 @@
#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"
-#include "talk/xmpp/jid.h"
-#include "talk/xmpp/pubsubtasks.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/fakexmppclient.h"
+#include "webrtc/libjingle/xmpp/iqtask.h"
+#include "webrtc/libjingle/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/pubsubtasks.h"
#include "webrtc/base/faketaskrunner.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/sigslot.h"
diff --git a/xmpp/receivetask.cc b/xmpp/receivetask.cc
index c2fb244..4c4fae5 100644
--- a/xmpp/receivetask.cc
+++ b/xmpp/receivetask.cc
@@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/receivetask.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/receivetask.h"
namespace buzz {
diff --git a/xmpp/receivetask.h b/xmpp/receivetask.h
index b18e0f0..7055339 100644
--- a/xmpp/receivetask.h
+++ b/xmpp/receivetask.h
@@ -28,7 +28,7 @@
#ifndef TALK_XMPP_RECEIVETASK_H_
#define TALK_XMPP_RECEIVETASK_H_
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
namespace buzz {
diff --git a/xmpp/rostermodule.h b/xmpp/rostermodule.h
index dfb647d..9ae7f70 100644
--- a/xmpp/rostermodule.h
+++ b/xmpp/rostermodule.h
@@ -28,7 +28,7 @@
#ifndef TALK_XMPP_ROSTERMODULE_H_
#define TALK_XMPP_ROSTERMODULE_H_
-#include "talk/xmpp/module.h"
+#include "webrtc/libjingle/xmpp/module.h"
namespace buzz {
diff --git a/xmpp/rostermodule_unittest.cc b/xmpp/rostermodule_unittest.cc
index cb7f773..fb82546 100644
--- a/xmpp/rostermodule_unittest.cc
+++ b/xmpp/rostermodule_unittest.cc
@@ -30,10 +30,10 @@
#include <string>
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/rostermodule.h"
-#include "talk/xmpp/util_unittest.h"
-#include "talk/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/rostermodule.h"
+#include "webrtc/libjingle/xmpp/util_unittest.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/scoped_ptr.h"
diff --git a/xmpp/rostermoduleimpl.cc b/xmpp/rostermoduleimpl.cc
index e52e78e..8841db6 100644
--- a/xmpp/rostermoduleimpl.cc
+++ b/xmpp/rostermoduleimpl.cc
@@ -31,8 +31,8 @@
#include <sstream>
#include <string>
#include <vector>
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/rostermoduleimpl.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/rostermoduleimpl.h"
#include "webrtc/base/common.h"
#include "webrtc/base/stringencode.h"
diff --git a/xmpp/rostermoduleimpl.h b/xmpp/rostermoduleimpl.h
index 37d1117..efef5ed 100644
--- a/xmpp/rostermoduleimpl.h
+++ b/xmpp/rostermoduleimpl.h
@@ -28,8 +28,8 @@
#ifndef TALK_XMPP_XMPPTHREAD_H_
#define TALK_XMPP_XMPPTHREAD_H_
-#include "talk/xmpp/moduleimpl.h"
-#include "talk/xmpp/rostermodule.h"
+#include "webrtc/libjingle/xmpp/moduleimpl.h"
+#include "webrtc/libjingle/xmpp/rostermodule.h"
namespace buzz {
diff --git a/xmpp/saslcookiemechanism.h b/xmpp/saslcookiemechanism.h
index eda142c..0f193f2 100644
--- a/xmpp/saslcookiemechanism.h
+++ b/xmpp/saslcookiemechanism.h
@@ -30,8 +30,8 @@
#include "webrtc/libjingle/xmllite/qname.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/saslmechanism.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/saslmechanism.h"
namespace buzz {
diff --git a/xmpp/saslmechanism.cc b/xmpp/saslmechanism.cc
index 8877084..9fb01bc 100644
--- a/xmpp/saslmechanism.cc
+++ b/xmpp/saslmechanism.cc
@@ -26,8 +26,8 @@
*/
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/saslmechanism.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/saslmechanism.h"
#include "webrtc/base/base64.h"
using rtc::Base64;
diff --git a/xmpp/saslplainmechanism.h b/xmpp/saslplainmechanism.h
index 36b9bb9..c9d428e 100644
--- a/xmpp/saslplainmechanism.h
+++ b/xmpp/saslplainmechanism.h
@@ -28,7 +28,7 @@
#ifndef TALK_XMPP_SASLPLAINMECHANISM_H_
#define TALK_XMPP_SASLPLAINMECHANISM_H_
-#include "talk/xmpp/saslmechanism.h"
+#include "webrtc/libjingle/xmpp/saslmechanism.h"
#include "webrtc/base/cryptstring.h"
namespace buzz {
diff --git a/xmpp/util_unittest.cc b/xmpp/util_unittest.cc
index 3e47d3f..d245efe 100644
--- a/xmpp/util_unittest.cc
+++ b/xmpp/util_unittest.cc
@@ -5,8 +5,8 @@
#include <sstream>
#include <string>
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/util_unittest.h"
-#include "talk/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/util_unittest.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
#include "webrtc/base/gunit.h"
namespace buzz {
diff --git a/xmpp/util_unittest.h b/xmpp/util_unittest.h
index c9377ff..806b505 100644
--- a/xmpp/util_unittest.h
+++ b/xmpp/util_unittest.h
@@ -30,7 +30,7 @@
#include <sstream>
#include <string>
-#include "talk/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
namespace buzz {
diff --git a/xmpp/xmppauth.cc b/xmpp/xmppauth.cc
index d828475..255f3a7 100644
--- a/xmpp/xmppauth.cc
+++ b/xmpp/xmppauth.cc
@@ -25,13 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/xmppauth.h"
+#include "webrtc/libjingle/xmpp/xmppauth.h"
#include <algorithm>
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/saslcookiemechanism.h"
-#include "talk/xmpp/saslplainmechanism.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/saslcookiemechanism.h"
+#include "webrtc/libjingle/xmpp/saslplainmechanism.h"
XmppAuth::XmppAuth() : done_(false) {
}
diff --git a/xmpp/xmppauth.h b/xmpp/xmppauth.h
index fec73e6..9fabd5e 100644
--- a/xmpp/xmppauth.h
+++ b/xmpp/xmppauth.h
@@ -30,9 +30,9 @@
#include <vector>
-#include "talk/xmpp/jid.h"
-#include "talk/xmpp/prexmppauth.h"
-#include "talk/xmpp/saslhandler.h"
+#include "webrtc/libjingle/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/prexmppauth.h"
+#include "webrtc/libjingle/xmpp/saslhandler.h"
#include "webrtc/base/cryptstring.h"
#include "webrtc/base/sigslot.h"
diff --git a/xmpp/xmppclient.cc b/xmpp/xmppclient.cc
index 27aa5e4..66d1e97 100644
--- a/xmpp/xmppclient.cc
+++ b/xmpp/xmppclient.cc
@@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/xmppclient.h"
+#include "webrtc/libjingle/xmpp/xmppclient.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/plainsaslhandler.h"
-#include "talk/xmpp/prexmppauth.h"
-#include "talk/xmpp/saslplainmechanism.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/plainsaslhandler.h"
+#include "webrtc/libjingle/xmpp/prexmppauth.h"
+#include "webrtc/libjingle/xmpp/saslplainmechanism.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/sigslot.h"
diff --git a/xmpp/xmppclient.h b/xmpp/xmppclient.h
index 53b2d24..74e4030 100644
--- a/xmpp/xmppclient.h
+++ b/xmpp/xmppclient.h
@@ -29,10 +29,10 @@
#define TALK_XMPP_XMPPCLIENT_H_
#include <string>
-#include "talk/xmpp/asyncsocket.h"
-#include "talk/xmpp/xmppclientsettings.h"
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/asyncsocket.h"
+#include "webrtc/libjingle/xmpp/xmppclientsettings.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
#include "webrtc/base/basicdefs.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/task.h"
diff --git a/xmpp/xmppclientsettings.h b/xmpp/xmppclientsettings.h
index 50028b7..04acf36 100644
--- a/xmpp/xmppclientsettings.h
+++ b/xmpp/xmppclientsettings.h
@@ -28,8 +28,8 @@
#ifndef TALK_XMPP_XMPPCLIENTSETTINGS_H_
#define TALK_XMPP_XMPPCLIENTSETTINGS_H_
-#include "talk/p2p/base/port.h"
-#include "talk/xmpp/xmppengine.h"
+#include "webrtc/p2p/base/port.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
#include "webrtc/base/cryptstring.h"
namespace buzz {
diff --git a/xmpp/xmppengine.h b/xmpp/xmppengine.h
index 461e90f..1806ba9 100644
--- a/xmpp/xmppengine.h
+++ b/xmpp/xmppengine.h
@@ -31,7 +31,7 @@
// also part of the API
#include "webrtc/libjingle/xmllite/qname.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/jid.h"
namespace buzz {
diff --git a/xmpp/xmppengine_unittest.cc b/xmpp/xmppengine_unittest.cc
index b519a65..6b0dc96 100644
--- a/xmpp/xmppengine_unittest.cc
+++ b/xmpp/xmppengine_unittest.cc
@@ -5,11 +5,11 @@
#include <sstream>
#include <string>
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/plainsaslhandler.h"
-#include "talk/xmpp/saslplainmechanism.h"
-#include "talk/xmpp/util_unittest.h"
-#include "talk/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/plainsaslhandler.h"
+#include "webrtc/libjingle/xmpp/saslplainmechanism.h"
+#include "webrtc/libjingle/xmpp/util_unittest.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
#include "webrtc/base/common.h"
#include "webrtc/base/gunit.h"
diff --git a/xmpp/xmppengineimpl.cc b/xmpp/xmppengineimpl.cc
index 5de9de7..98d89e2 100644
--- a/xmpp/xmppengineimpl.cc
+++ b/xmpp/xmppengineimpl.cc
@@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/xmppengineimpl.h"
+#include "webrtc/libjingle/xmpp/xmppengineimpl.h"
#include <algorithm>
#include <sstream>
@@ -33,9 +33,9 @@
#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"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/saslhandler.h"
+#include "webrtc/libjingle/xmpp/xmpplogintask.h"
#include "webrtc/base/common.h"
namespace buzz {
diff --git a/xmpp/xmppengineimpl.h b/xmpp/xmppengineimpl.h
index 4eacf2f..a3da795 100644
--- a/xmpp/xmppengineimpl.h
+++ b/xmpp/xmppengineimpl.h
@@ -30,8 +30,8 @@
#include <sstream>
#include <vector>
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmppstanzaparser.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmppstanzaparser.h"
namespace buzz {
diff --git a/xmpp/xmppengineimpl_iq.cc b/xmpp/xmppengineimpl_iq.cc
index 208e164..d48f021 100644
--- a/xmpp/xmppengineimpl_iq.cc
+++ b/xmpp/xmppengineimpl_iq.cc
@@ -27,8 +27,8 @@
#include <algorithm>
#include <vector>
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/xmppengineimpl.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/xmppengineimpl.h"
#include "webrtc/base/common.h"
namespace buzz {
diff --git a/xmpp/xmpplogintask.cc b/xmpp/xmpplogintask.cc
index a48a94c..7788c0d 100644
--- a/xmpp/xmpplogintask.cc
+++ b/xmpp/xmpplogintask.cc
@@ -25,16 +25,16 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/xmpplogintask.h"
+#include "webrtc/libjingle/xmpp/xmpplogintask.h"
#include <string>
#include <vector>
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/jid.h"
-#include "talk/xmpp/saslmechanism.h"
-#include "talk/xmpp/xmppengineimpl.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/saslmechanism.h"
+#include "webrtc/libjingle/xmpp/xmppengineimpl.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/common.h"
diff --git a/xmpp/xmpplogintask.h b/xmpp/xmpplogintask.h
index 8685658..631ebdf 100644
--- a/xmpp/xmpplogintask.h
+++ b/xmpp/xmpplogintask.h
@@ -31,8 +31,8 @@
#include <string>
#include <vector>
-#include "talk/xmpp/jid.h"
-#include "talk/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/jid.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
diff --git a/xmpp/xmpplogintask_unittest.cc b/xmpp/xmpplogintask_unittest.cc
index ae9a554..82973c4 100644
--- a/xmpp/xmpplogintask_unittest.cc
+++ b/xmpp/xmpplogintask_unittest.cc
@@ -5,11 +5,11 @@
#include <sstream>
#include <string>
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/plainsaslhandler.h"
-#include "talk/xmpp/saslplainmechanism.h"
-#include "talk/xmpp/util_unittest.h"
-#include "talk/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/plainsaslhandler.h"
+#include "webrtc/libjingle/xmpp/saslplainmechanism.h"
+#include "webrtc/libjingle/xmpp/util_unittest.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
#include "webrtc/base/common.h"
#include "webrtc/base/cryptstring.h"
#include "webrtc/base/gunit.h"
diff --git a/xmpp/xmpppump.cc b/xmpp/xmpppump.cc
index cf7aa7b..e9e7823 100644
--- a/xmpp/xmpppump.cc
+++ b/xmpp/xmpppump.cc
@@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/xmpppump.h"
+#include "webrtc/libjingle/xmpp/xmpppump.h"
-#include "talk/xmpp/xmppauth.h"
+#include "webrtc/libjingle/xmpp/xmppauth.h"
namespace buzz {
diff --git a/xmpp/xmpppump.h b/xmpp/xmpppump.h
index ff72f54..082afb9 100644
--- a/xmpp/xmpppump.h
+++ b/xmpp/xmpppump.h
@@ -28,9 +28,9 @@
#ifndef TALK_XMPP_XMPPPUMP_H_
#define TALK_XMPP_XMPPPUMP_H_
-#include "talk/xmpp/xmppclient.h"
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/xmppclient.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
#include "webrtc/base/messagequeue.h"
#include "webrtc/base/taskrunner.h"
#include "webrtc/base/thread.h"
diff --git a/xmpp/xmppsocket.h b/xmpp/xmppsocket.h
index 3d6df52..240bbe9 100644
--- a/xmpp/xmppsocket.h
+++ b/xmpp/xmppsocket.h
@@ -28,8 +28,8 @@
#ifndef TALK_XMPP_XMPPSOCKET_H_
#define TALK_XMPP_XMPPSOCKET_H_
-#include "talk/xmpp/asyncsocket.h"
-#include "talk/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/asyncsocket.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
#include "webrtc/base/asyncsocket.h"
#include "webrtc/base/bytebuffer.h"
#include "webrtc/base/sigslot.h"
diff --git a/xmpp/xmppstanzaparser.cc b/xmpp/xmppstanzaparser.cc
index 4795839..a4ca2ed 100644
--- a/xmpp/xmppstanzaparser.cc
+++ b/xmpp/xmppstanzaparser.cc
@@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/xmppstanzaparser.h"
+#include "webrtc/libjingle/xmpp/xmppstanzaparser.h"
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/constants.h"
#include "webrtc/base/common.h"
#ifdef EXPAT_RELATIVE_PATH
#include "expat.h"
diff --git a/xmpp/xmppstanzaparser_unittest.cc b/xmpp/xmppstanzaparser_unittest.cc
index 0b114c0..433de5e 100644
--- a/xmpp/xmppstanzaparser_unittest.cc
+++ b/xmpp/xmppstanzaparser_unittest.cc
@@ -5,7 +5,7 @@
#include <sstream>
#include <string>
#include "webrtc/libjingle/xmllite/xmlelement.h"
-#include "talk/xmpp/xmppstanzaparser.h"
+#include "webrtc/libjingle/xmpp/xmppstanzaparser.h"
#include "webrtc/base/common.h"
#include "webrtc/base/gunit.h"
diff --git a/xmpp/xmpptask.cc b/xmpp/xmpptask.cc
index c5e86fa..31aba62 100644
--- a/xmpp/xmpptask.cc
+++ b/xmpp/xmpptask.cc
@@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/constants.h"
-#include "talk/xmpp/xmppclient.h"
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmpptask.h"
+#include "webrtc/libjingle/xmpp/constants.h"
+#include "webrtc/libjingle/xmpp/xmppclient.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmpptask.h"
namespace buzz {
diff --git a/xmpp/xmpptask.h b/xmpp/xmpptask.h
index f6dd156..a8d1124 100644
--- a/xmpp/xmpptask.h
+++ b/xmpp/xmpptask.h
@@ -30,7 +30,7 @@
#include <deque>
#include <string>
-#include "talk/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/task.h"
#include "webrtc/base/taskparent.h"
diff --git a/xmpp/xmppthread.cc b/xmpp/xmppthread.cc
index e67bffe..ad9246b 100644
--- a/xmpp/xmppthread.cc
+++ b/xmpp/xmppthread.cc
@@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/xmpp/xmppthread.h"
+#include "webrtc/libjingle/xmpp/xmppthread.h"
-#include "talk/xmpp/xmppauth.h"
-#include "talk/xmpp/xmppclientsettings.h"
+#include "webrtc/libjingle/xmpp/xmppauth.h"
+#include "webrtc/libjingle/xmpp/xmppclientsettings.h"
namespace buzz {
namespace {
diff --git a/xmpp/xmppthread.h b/xmpp/xmppthread.h
index bfb74d6..5a77f00 100644
--- a/xmpp/xmppthread.h
+++ b/xmpp/xmppthread.h
@@ -28,10 +28,10 @@
#ifndef TALK_XMPP_XMPPTHREAD_H_
#define TALK_XMPP_XMPPTHREAD_H_
-#include "talk/xmpp/xmppclientsettings.h"
-#include "talk/xmpp/xmppengine.h"
-#include "talk/xmpp/xmpppump.h"
-#include "talk/xmpp/xmppsocket.h"
+#include "webrtc/libjingle/xmpp/xmppclientsettings.h"
+#include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/libjingle/xmpp/xmpppump.h"
+#include "webrtc/libjingle/xmpp/xmppsocket.h"
#include "webrtc/base/thread.h"
namespace buzz {
