Merge from Chromium at DEPS revision 03655fd3f6d7
This commit was generated by merge_to_master.py.
Change-Id: Ie97de41dee6631b70dd07c00db5bf3ad4dfe8e14
diff --git a/app/webrtc/objc/RTCEAGLVideoView+Internal.h b/app/webrtc/objc/RTCEAGLVideoView+Internal.h
deleted file mode 100644
index 10df2e3..0000000
--- a/app/webrtc/objc/RTCEAGLVideoView+Internal.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * 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.
- */
-
-#import <Foundation/Foundation.h>
-
-#import "RTCEAGLVideoView.h"
-#import "RTCVideoRenderer.h"
-
-// TODO(tkchin): Move declaration to implementation file. Exposed here in order
-// to support deprecated methods in RTCVideoRenderer.
-@interface RTCEAGLVideoView (Internal) <RTCVideoRendererDelegate>
-@end
diff --git a/app/webrtc/objc/RTCEAGLVideoView.m b/app/webrtc/objc/RTCEAGLVideoView.m
index faacef6..c55c87e 100644
--- a/app/webrtc/objc/RTCEAGLVideoView.m
+++ b/app/webrtc/objc/RTCEAGLVideoView.m
@@ -29,13 +29,12 @@
#error "This file requires ARC support."
#endif
-#import "RTCEAGLVideoView+Internal.h"
+#import "RTCEAGLVideoView.h"
#import <GLKit/GLKit.h>
+#import "RTCI420Frame.h"
#import "RTCOpenGLVideoRenderer.h"
-#import "RTCVideoRenderer.h"
-#import "RTCVideoTrack.h"
// RTCDisplayLinkTimer wraps a CADisplayLink and is set to fire every two screen
// refreshes, which should be 30fps. We wrap the display link in order to avoid
@@ -105,7 +104,6 @@
RTCDisplayLinkTimer* _timer;
GLKView* _glkView;
RTCOpenGLVideoRenderer* _glRenderer;
- RTCVideoRenderer* _videoRenderer;
}
- (instancetype)initWithFrame:(CGRect)frame {
@@ -152,7 +150,6 @@
// GLKViewDelegate method implemented below.
[strongSelf.glkView setNeedsDisplay];
}];
- _videoRenderer = [[RTCVideoRenderer alloc] initWithDelegate:self];
[self setupGL];
}
return self;
@@ -168,18 +165,6 @@
[_timer invalidate];
}
-- (void)setVideoTrack:(RTCVideoTrack*)videoTrack {
- if (_videoTrack == videoTrack) {
- return;
- }
- [_videoTrack removeRenderer:_videoRenderer];
- self.i420Frame = nil;
- _videoTrack = videoTrack;
- [_videoTrack addRenderer:_videoRenderer];
- // TODO(tkchin): potentially handle changes in track state - e.g. render
- // black if track fails.
-}
-
#pragma mark - UIView
- (void)layoutSubviews {
@@ -197,14 +182,31 @@
[_glRenderer drawFrame:self.i420Frame];
}
+#pragma mark - RTCVideoRenderer
+
+// These methods may be called on non-main thread.
+- (void)setSize:(CGSize)size {
+ __weak RTCEAGLVideoView* weakSelf = self;
+ dispatch_async(dispatch_get_main_queue(), ^{
+ RTCEAGLVideoView* strongSelf = weakSelf;
+ [strongSelf.delegate videoView:strongSelf didChangeVideoSize:size];
+ });
+}
+
+- (void)renderFrame:(RTCI420Frame*)frame {
+ self.i420Frame = frame;
+}
+
#pragma mark - Private
- (void)setupGL {
+ self.i420Frame = nil;
[_glRenderer setupGL];
_timer.isPaused = NO;
}
- (void)teardownGL {
+ self.i420Frame = nil;
_timer.isPaused = YES;
[_glkView deleteDrawable];
[_glRenderer teardownGL];
@@ -219,25 +221,3 @@
}
@end
-
-@implementation RTCEAGLVideoView (Internal)
-
-#pragma mark - RTCVideoRendererDelegate
-
-// These methods are called when the video track has frame information to
-// provide. This occurs on non-main thread.
-- (void)renderer:(RTCVideoRenderer*)renderer
- didSetSize:(CGSize)size {
- __weak RTCEAGLVideoView* weakSelf = self;
- dispatch_async(dispatch_get_main_queue(), ^{
- RTCEAGLVideoView* strongSelf = weakSelf;
- [strongSelf.delegate videoView:strongSelf didChangeVideoSize:size];
- });
-}
-
-- (void)renderer:(RTCVideoRenderer*)renderer
- didReceiveFrame:(RTCI420Frame*)frame {
- self.i420Frame = frame;
-}
-
-@end
diff --git a/app/webrtc/objc/RTCI420Frame.mm b/app/webrtc/objc/RTCI420Frame.mm
index 0b50691..9c394e5 100644
--- a/app/webrtc/objc/RTCI420Frame.mm
+++ b/app/webrtc/objc/RTCI420Frame.mm
@@ -78,6 +78,10 @@
return _videoFrame->GetVPitch();
}
+- (BOOL)makeExclusive {
+ return _videoFrame->MakeExclusive();
+}
+
@end
@implementation RTCI420Frame (Internal)
diff --git a/app/webrtc/objc/RTCMediaStream.mm b/app/webrtc/objc/RTCMediaStream.mm
index 27d20b8..a72508a 100644
--- a/app/webrtc/objc/RTCMediaStream.mm
+++ b/app/webrtc/objc/RTCMediaStream.mm
@@ -71,7 +71,7 @@
}
- (BOOL)addVideoTrack:(RTCVideoTrack*)track {
- if (self.mediaStream->AddTrack(track.videoTrack)) {
+ if (self.mediaStream->AddTrack(track.nativeVideoTrack)) {
[_videoTracks addObject:track];
return YES;
}
@@ -93,7 +93,8 @@
NSUInteger index = [_videoTracks indexOfObjectIdenticalTo:track];
NSAssert(index != NSNotFound,
@"|removeAudioTrack| called on unexpected RTCVideoTrack");
- if (index != NSNotFound && self.mediaStream->RemoveTrack(track.videoTrack)) {
+ if (index != NSNotFound &&
+ self.mediaStream->RemoveTrack(track.nativeVideoTrack)) {
[_videoTracks removeObjectAtIndex:index];
return YES;
}
diff --git a/app/webrtc/objc/RTCNSGLVideoView.m b/app/webrtc/objc/RTCNSGLVideoView.m
index 292e792..7aa4a11 100644
--- a/app/webrtc/objc/RTCNSGLVideoView.m
+++ b/app/webrtc/objc/RTCNSGLVideoView.m
@@ -33,10 +33,10 @@
#import <CoreVideo/CVDisplayLink.h>
#import <OpenGL/gl3.h>
+#import "RTCI420Frame.h"
#import "RTCOpenGLVideoRenderer.h"
-#import "RTCVideoRenderer.h"
-@interface RTCNSGLVideoView () <RTCVideoRendererDelegate>
+@interface RTCNSGLVideoView ()
// |i420Frame| is set when we receive a frame from a worker thread and is read
// from the display link callback so atomicity is required.
@property(atomic, strong) RTCI420Frame* i420Frame;
@@ -57,15 +57,6 @@
@implementation RTCNSGLVideoView {
CVDisplayLinkRef _displayLink;
- RTCVideoRenderer* _videoRenderer;
-}
-
-- (instancetype)initWithFrame:(NSRect)frame
- pixelFormat:(NSOpenGLPixelFormat*)format {
- if (self = [super initWithFrame:frame pixelFormat:format]) {
- _videoRenderer = [[RTCVideoRenderer alloc] initWithDelegate:self];
- }
- return self;
}
- (void)dealloc {
@@ -109,37 +100,16 @@
[super clearGLContext];
}
-- (void)setVideoTrack:(RTCVideoTrack*)videoTrack {
- if (_videoTrack == videoTrack) {
- return;
- }
- if (_videoTrack) {
- [_videoTrack removeRenderer:_videoRenderer];
- CVDisplayLinkStop(_displayLink);
- // Clear contents.
- self.i420Frame = nil;
- [self drawFrame];
- }
- _videoTrack = videoTrack;
- if (_videoTrack) {
- [_videoTrack addRenderer:_videoRenderer];
- CVDisplayLinkStart(_displayLink);
- }
-}
+#pragma mark - RTCVideoRenderer
-#pragma mark - RTCVideoRendererDelegate
-
-// These methods are called when the video track has frame information to
-// provide. This occurs on non-main thread.
-- (void)renderer:(RTCVideoRenderer*)renderer
- didSetSize:(CGSize)size {
+// These methods may be called on non-main thread.
+- (void)setSize:(CGSize)size {
dispatch_async(dispatch_get_main_queue(), ^{
[self.delegate videoView:self didChangeVideoSize:size];
});
}
-- (void)renderer:(RTCVideoRenderer*)renderer
- didReceiveFrame:(RTCI420Frame*)frame {
+- (void)renderFrame:(RTCI420Frame*)frame {
self.i420Frame = frame;
}
@@ -174,9 +144,7 @@
CGLPixelFormatObj cglPixelFormat = [[self pixelFormat] CGLPixelFormatObj];
CVDisplayLinkSetCurrentCGDisplayFromOpenGLContext(
_displayLink, cglContext, cglPixelFormat);
- if (_videoTrack) {
- CVDisplayLinkStart(_displayLink);
- }
+ CVDisplayLinkStart(_displayLink);
}
- (void)teardownDisplayLink {
diff --git a/app/webrtc/objc/RTCPeerConnection.mm b/app/webrtc/objc/RTCPeerConnection.mm
index 925de73..7767f76 100644
--- a/app/webrtc/objc/RTCPeerConnection.mm
+++ b/app/webrtc/objc/RTCPeerConnection.mm
@@ -68,6 +68,7 @@
[_delegate peerConnection:_peerConnection
didCreateSessionDescription:session
error:nil];
+ delete desc;
}
virtual void OnFailure(const std::string& error) OVERRIDE {
diff --git a/app/webrtc/objc/RTCVideoRenderer+Internal.h b/app/webrtc/objc/RTCVideoRendererAdapter.h
similarity index 84%
rename from app/webrtc/objc/RTCVideoRenderer+Internal.h
rename to app/webrtc/objc/RTCVideoRendererAdapter.h
index 22e445c..faf0906 100644
--- a/app/webrtc/objc/RTCVideoRenderer+Internal.h
+++ b/app/webrtc/objc/RTCVideoRendererAdapter.h
@@ -29,8 +29,12 @@
#include "talk/app/webrtc/mediastreaminterface.h"
-@interface RTCVideoRenderer (Internal)
+@interface RTCVideoRendererAdapter : NSObject
-@property(nonatomic, readonly) webrtc::VideoRendererInterface* videoRenderer;
+@property(nonatomic, readonly) id<RTCVideoRenderer> videoRenderer;
+@property(nonatomic, readonly)
+ webrtc::VideoRendererInterface* nativeVideoRenderer;
+
+- (instancetype)initWithVideoRenderer:(id<RTCVideoRenderer>)videoRenderer;
@end
diff --git a/app/webrtc/objc/RTCVideoRenderer.mm b/app/webrtc/objc/RTCVideoRendererAdapter.mm
similarity index 70%
rename from app/webrtc/objc/RTCVideoRenderer.mm
rename to app/webrtc/objc/RTCVideoRendererAdapter.mm
index 4cfe43a..e29faad 100644
--- a/app/webrtc/objc/RTCVideoRenderer.mm
+++ b/app/webrtc/objc/RTCVideoRendererAdapter.mm
@@ -29,50 +29,46 @@
#error "This file requires ARC support."
#endif
-#import "RTCVideoRenderer+Internal.h"
+#import "RTCVideoRendererAdapter.h"
#import "RTCI420Frame+Internal.h"
namespace webrtc {
-class RTCVideoRendererAdapter : public VideoRendererInterface {
+class RTCVideoRendererNativeAdapter : public VideoRendererInterface {
public:
- RTCVideoRendererAdapter(RTCVideoRenderer* renderer) { _renderer = renderer; }
+ RTCVideoRendererNativeAdapter(RTCVideoRendererAdapter* adapter) {
+ _adapter = adapter;
+ }
virtual void SetSize(int width, int height) OVERRIDE {
- [_renderer.delegate renderer:_renderer
- didSetSize:CGSizeMake(width, height)];
+ [_adapter.videoRenderer setSize:CGSizeMake(width, height)];
}
virtual void RenderFrame(const cricket::VideoFrame* frame) OVERRIDE {
- if (!_renderer.delegate) {
- return;
- }
RTCI420Frame* i420Frame = [[RTCI420Frame alloc] initWithVideoFrame:frame];
- [_renderer.delegate renderer:_renderer didReceiveFrame:i420Frame];
+ [_adapter.videoRenderer renderFrame:i420Frame];
}
private:
- __weak RTCVideoRenderer* _renderer;
+ __weak RTCVideoRendererAdapter* _adapter;
};
}
-@implementation RTCVideoRenderer {
- rtc::scoped_ptr<webrtc::RTCVideoRendererAdapter> _adapter;
+@implementation RTCVideoRendererAdapter {
+ id<RTCVideoRenderer> _videoRenderer;
+ rtc::scoped_ptr<webrtc::RTCVideoRendererNativeAdapter> _adapter;
}
-- (instancetype)initWithDelegate:(id<RTCVideoRendererDelegate>)delegate {
+- (instancetype)initWithVideoRenderer:(id<RTCVideoRenderer>)videoRenderer {
+ NSParameterAssert(videoRenderer);
if (self = [super init]) {
- _delegate = delegate;
- _adapter.reset(new webrtc::RTCVideoRendererAdapter(self));
+ _videoRenderer = videoRenderer;
+ _adapter.reset(new webrtc::RTCVideoRendererNativeAdapter(self));
}
return self;
}
-@end
-
-@implementation RTCVideoRenderer (Internal)
-
-- (webrtc::VideoRendererInterface*)videoRenderer {
+- (webrtc::VideoRendererInterface*)nativeVideoRenderer {
return _adapter.get();
}
diff --git a/app/webrtc/objc/RTCVideoTrack+Internal.h b/app/webrtc/objc/RTCVideoTrack+Internal.h
index 03c8f95..5f267ac 100644
--- a/app/webrtc/objc/RTCVideoTrack+Internal.h
+++ b/app/webrtc/objc/RTCVideoTrack+Internal.h
@@ -34,7 +34,7 @@
@interface RTCVideoTrack (Internal)
-@property(nonatomic, assign, readonly)
- rtc::scoped_refptr<webrtc::VideoTrackInterface> videoTrack;
+@property(nonatomic, readonly)
+ rtc::scoped_refptr<webrtc::VideoTrackInterface> nativeVideoTrack;
@end
diff --git a/app/webrtc/objc/RTCVideoTrack.mm b/app/webrtc/objc/RTCVideoTrack.mm
index beebde0..959bc6d 100644
--- a/app/webrtc/objc/RTCVideoTrack.mm
+++ b/app/webrtc/objc/RTCVideoTrack.mm
@@ -32,46 +32,55 @@
#import "RTCVideoTrack+Internal.h"
#import "RTCMediaStreamTrack+Internal.h"
-#import "RTCVideoRenderer+Internal.h"
+#import "RTCVideoRendererAdapter.h"
@implementation RTCVideoTrack {
- NSMutableArray* _rendererArray;
+ NSMutableArray* _adapters;
}
- (id)initWithMediaTrack:
(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)
mediaTrack {
if (self = [super initWithMediaTrack:mediaTrack]) {
- _rendererArray = [NSMutableArray array];
+ _adapters = [NSMutableArray array];
}
return self;
}
-- (void)addRenderer:(RTCVideoRenderer*)renderer {
- NSAssert1(![self.renderers containsObject:renderer],
- @"renderers already contains object [%@]",
- [renderer description]);
- [_rendererArray addObject:renderer];
- self.videoTrack->AddRenderer(renderer.videoRenderer);
-}
-
-- (void)removeRenderer:(RTCVideoRenderer*)renderer {
- NSUInteger index = [self.renderers indexOfObjectIdenticalTo:renderer];
- if (index != NSNotFound) {
- [_rendererArray removeObjectAtIndex:index];
- self.videoTrack->RemoveRenderer(renderer.videoRenderer);
+- (void)addRenderer:(id<RTCVideoRenderer>)renderer {
+ // Make sure we don't have this renderer yet.
+ for (RTCVideoRendererAdapter* adapter in _adapters) {
+ NSParameterAssert(adapter.videoRenderer != renderer);
}
+ // Create a wrapper that provides a native pointer for us.
+ RTCVideoRendererAdapter* adapter =
+ [[RTCVideoRendererAdapter alloc] initWithVideoRenderer:renderer];
+ [_adapters addObject:adapter];
+ self.nativeVideoTrack->AddRenderer(adapter.nativeVideoRenderer);
}
-- (NSArray*)renderers {
- return [_rendererArray copy];
+- (void)removeRenderer:(id<RTCVideoRenderer>)renderer {
+ RTCVideoRendererAdapter* adapter = nil;
+ NSUInteger indexToRemove = NSNotFound;
+ for (NSUInteger i = 0; i < _adapters.count; i++) {
+ adapter = _adapters[i];
+ if (adapter.videoRenderer == renderer) {
+ indexToRemove = i;
+ break;
+ }
+ }
+ if (indexToRemove == NSNotFound) {
+ return;
+ }
+ self.nativeVideoTrack->RemoveRenderer(adapter.nativeVideoRenderer);
+ [_adapters removeObjectAtIndex:indexToRemove];
}
@end
@implementation RTCVideoTrack (Internal)
-- (rtc::scoped_refptr<webrtc::VideoTrackInterface>)videoTrack {
+- (rtc::scoped_refptr<webrtc::VideoTrackInterface>)nativeVideoTrack {
return static_cast<webrtc::VideoTrackInterface*>(self.mediaTrack.get());
}
diff --git a/app/webrtc/objc/public/RTCEAGLVideoView.h b/app/webrtc/objc/public/RTCEAGLVideoView.h
index c38799e..526175f 100644
--- a/app/webrtc/objc/public/RTCEAGLVideoView.h
+++ b/app/webrtc/objc/public/RTCEAGLVideoView.h
@@ -37,11 +37,10 @@
@end
-@class RTCVideoTrack;
-// RTCEAGLVideoView renders |videoTrack| onto itself using OpenGLES.
-@interface RTCEAGLVideoView : UIView
+// RTCEAGLVideoView is an RTCVideoRenderer which renders i420 frames in its
+// bounds using OpenGLES 2.0.
+@interface RTCEAGLVideoView : UIView <RTCVideoRenderer>
-@property(nonatomic, strong) RTCVideoTrack* videoTrack;
@property(nonatomic, weak) id<RTCEAGLVideoViewDelegate> delegate;
@end
diff --git a/app/webrtc/objc/public/RTCI420Frame.h b/app/webrtc/objc/public/RTCI420Frame.h
index 737968c..7a8c4d4 100644
--- a/app/webrtc/objc/public/RTCI420Frame.h
+++ b/app/webrtc/objc/public/RTCI420Frame.h
@@ -43,6 +43,8 @@
@property(nonatomic, readonly) NSInteger uPitch;
@property(nonatomic, readonly) NSInteger vPitch;
+- (BOOL)makeExclusive;
+
#ifndef DOXYGEN_SHOULD_SKIP_THIS
// Disallow init and don't add to documentation
- (id)init __attribute__((
diff --git a/app/webrtc/objc/public/RTCNSGLVideoView.h b/app/webrtc/objc/public/RTCNSGLVideoView.h
index fd757cb..0af2dc5 100644
--- a/app/webrtc/objc/public/RTCNSGLVideoView.h
+++ b/app/webrtc/objc/public/RTCNSGLVideoView.h
@@ -31,7 +31,7 @@
#import <AppKit/NSOpenGLView.h>
-#import "RTCVideoTrack.h"
+#import "RTCVideoRenderer.h"
@class RTCNSGLVideoView;
@protocol RTCNSGLVideoViewDelegate
@@ -40,9 +40,8 @@
@end
-@interface RTCNSGLVideoView : NSOpenGLView
+@interface RTCNSGLVideoView : NSOpenGLView <RTCVideoRenderer>
-@property(nonatomic, strong) RTCVideoTrack* videoTrack;
@property(nonatomic, weak) id<RTCNSGLVideoViewDelegate> delegate;
@end
diff --git a/app/webrtc/objc/public/RTCVideoRenderer.h b/app/webrtc/objc/public/RTCVideoRenderer.h
index 37977ce..3c2baba 100644
--- a/app/webrtc/objc/public/RTCVideoRenderer.h
+++ b/app/webrtc/objc/public/RTCVideoRenderer.h
@@ -31,34 +31,13 @@
#endif
@class RTCI420Frame;
-@class RTCVideoRenderer;
-// RTCVideoRendererDelegate is a protocol for an object that must be
-// implemented to get messages when rendering.
-@protocol RTCVideoRendererDelegate<NSObject>
+@protocol RTCVideoRenderer<NSObject>
// The size of the frame.
-- (void)renderer:(RTCVideoRenderer*)renderer didSetSize:(CGSize)size;
+- (void)setSize:(CGSize)size;
// The frame to be displayed.
-- (void)renderer:(RTCVideoRenderer*)renderer
- didReceiveFrame:(RTCI420Frame*)frame;
-
-@end
-
-// Interface for rendering VideoFrames from a VideoTrack
-@interface RTCVideoRenderer : NSObject
-
-@property(nonatomic, weak) id<RTCVideoRendererDelegate> delegate;
-
-// Initialize the renderer. Requires a delegate which does the actual drawing
-// of frames.
-- (instancetype)initWithDelegate:(id<RTCVideoRendererDelegate>)delegate;
-
-#ifndef DOXYGEN_SHOULD_SKIP_THIS
-// Disallow init and don't add to documentation
-- (id)init __attribute__((
- unavailable("init is not a supported initializer for this class.")));
-#endif /* DOXYGEN_SHOULD_SKIP_THIS */
+- (void)renderFrame:(RTCI420Frame*)frame;
@end
diff --git a/app/webrtc/objc/public/RTCVideoTrack.h b/app/webrtc/objc/public/RTCVideoTrack.h
index 291c923..8385b71 100644
--- a/app/webrtc/objc/public/RTCVideoTrack.h
+++ b/app/webrtc/objc/public/RTCVideoTrack.h
@@ -27,19 +27,16 @@
#import "RTCMediaStreamTrack.h"
-@class RTCVideoRenderer;
+@protocol RTCVideoRenderer;
// RTCVideoTrack is an ObjectiveC wrapper for VideoTrackInterface.
@interface RTCVideoTrack : RTCMediaStreamTrack
-// The currently registered renderers.
-@property(nonatomic, strong, readonly) NSArray *renderers;
-
// Register a renderer that will render all frames received on this track.
-- (void)addRenderer:(RTCVideoRenderer *)renderer;
+- (void)addRenderer:(id<RTCVideoRenderer>)renderer;
// Deregister a renderer.
-- (void)removeRenderer:(RTCVideoRenderer *)renderer;
+- (void)removeRenderer:(id<RTCVideoRenderer>)renderer;
#ifndef DOXYGEN_SHOULD_SKIP_THIS
// Disallow init and don't add to documentation
diff --git a/app/webrtc/objctests/RTCPeerConnectionTest.mm b/app/webrtc/objctests/RTCPeerConnectionTest.mm
index 6c5950b..92d3c49 100644
--- a/app/webrtc/objctests/RTCPeerConnectionTest.mm
+++ b/app/webrtc/objctests/RTCPeerConnectionTest.mm
@@ -46,6 +46,16 @@
#error "This file requires ARC support."
#endif
+@interface RTCFakeRenderer : NSObject <RTCVideoRenderer>
+@end
+
+@implementation RTCFakeRenderer
+
+- (void)setSize:(CGSize)size {}
+- (void)renderFrame:(RTCI420Frame*)frame {}
+
+@end
+
@interface RTCPeerConnectionTest : NSObject
// Returns whether the two sessions are of the same type.
@@ -80,8 +90,7 @@
RTCMediaStream* localMediaStream = [factory mediaStreamWithLabel:streamLabel];
RTCVideoTrack* videoTrack =
[factory videoTrackWithID:videoTrackID source:videoSource];
- RTCVideoRenderer* videoRenderer =
- [[RTCVideoRenderer alloc] initWithDelegate:nil];
+ RTCFakeRenderer* videoRenderer = [[RTCFakeRenderer alloc] init];
[videoTrack addRenderer:videoRenderer];
[localMediaStream addVideoTrack:videoTrack];
// Test that removal/re-add works.
diff --git a/app/webrtc/peerconnection.cc b/app/webrtc/peerconnection.cc
index b64caf7..64ddcad 100644
--- a/app/webrtc/peerconnection.cc
+++ b/app/webrtc/peerconnection.cc
@@ -420,11 +420,6 @@
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 355211c..68aa154 100644
--- a/app/webrtc/peerconnection.h
+++ b/app/webrtc/peerconnection.h
@@ -66,8 +66,6 @@
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);
virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
diff --git a/app/webrtc/peerconnectioninterface.h b/app/webrtc/peerconnectioninterface.h
index 68b7879..73a4812 100644
--- a/app/webrtc/peerconnectioninterface.h
+++ b/app/webrtc/peerconnectioninterface.h
@@ -255,15 +255,7 @@
// 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.
- // 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) {
- return false;
- }
+ virtual bool AddStream(MediaStreamInterface* stream) = 0;
// Remove a MediaStream from this PeerConnection.
// Note that a SessionDescription negotiation is need before the
@@ -351,10 +343,6 @@
kIceState,
};
- // Deprecated.
- // TODO(perkj): Remove once its not used by Chrome.
- virtual void OnError() {}
-
// Triggered when the SignalingState changed.
virtual void OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) {}
diff --git a/app/webrtc/peerconnectionproxy.h b/app/webrtc/peerconnectionproxy.h
index 571c676..852d852 100644
--- a/app/webrtc/peerconnectionproxy.h
+++ b/app/webrtc/peerconnectionproxy.h
@@ -40,8 +40,6 @@
PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>,
remote_streams)
PROXY_METHOD1(bool, AddStream, MediaStreamInterface*)
- PROXY_METHOD2(bool, AddStream, MediaStreamInterface*,
- const MediaConstraintsInterface*);
PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*)
PROXY_METHOD1(rtc::scoped_refptr<DtmfSenderInterface>,
CreateDtmfSender, AudioTrackInterface*)
diff --git a/app/webrtc/webrtcsdp.cc b/app/webrtc/webrtcsdp.cc
index 23b8f3d..c7b34c4 100644
--- a/app/webrtc/webrtcsdp.cc
+++ b/app/webrtc/webrtcsdp.cc
@@ -206,7 +206,9 @@
// draft-ietf-mmusic-trickle-ice-01
// When no candidates have been gathered, set the connection
// address to IP6 ::.
-static const char kDummyAddress[] = "::";
+// TODO(perkj): FF can not parse IP6 ::. See http://crbug/430333
+// Use IPV4 per default.
+static const char kDummyAddress[] = "0.0.0.0";
static const char kDummyPort[] = "9";
// RFC 3556
static const char kApplicationSpecificMaximum[] = "AS";
@@ -675,7 +677,7 @@
const std::vector<Candidate>& candidates,
int component_id, std::string* port,
std::string* ip, std::string* addr_type) {
- *addr_type = kConnectionIpv6Addrtype;
+ *addr_type = kConnectionIpv4Addrtype;
*port = kDummyPort;
*ip = kDummyAddress;
int current_preference = kPreferenceUnknown;
diff --git a/app/webrtc/webrtcsdp_unittest.cc b/app/webrtc/webrtcsdp_unittest.cc
index ea590db..8655487 100644
--- a/app/webrtc/webrtcsdp_unittest.cc
+++ b/app/webrtc/webrtcsdp_unittest.cc
@@ -213,8 +213,8 @@
"t=0 0\r\n"
"a=msid-semantic: WMS local_stream_1 local_stream_2\r\n"
"m=audio 9 RTP/SAVPF 111 103 104\r\n"
- "c=IN IP6 ::\r\n"
- "a=rtcp:9 IN IP6 ::\r\n"
+ "c=IN IP4 0.0.0.0\r\n"
+ "a=rtcp:9 IN IP4 0.0.0.0\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"
@@ -234,8 +234,8 @@
"a=ssrc:4 mslabel:local_stream_2\r\n"
"a=ssrc:4 label:audio_track_id_2\r\n"
"m=video 9 RTP/SAVPF 120\r\n"
- "c=IN IP6 ::\r\n"
- "a=rtcp:9 IN IP6 ::\r\n"
+ "c=IN IP4 0.0.0.0\r\n"
+ "a=rtcp:9 IN IP4 0.0.0.0\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"
@@ -262,8 +262,8 @@
static const char kSdpRtpDataChannelString[] =
"m=application 9 RTP/SAVPF 101\r\n"
- "c=IN IP6 ::\r\n"
- "a=rtcp:9 IN IP6 ::\r\n"
+ "c=IN IP4 0.0.0.0\r\n"
+ "a=rtcp:9 IN IP4 0.0.0.0\r\n"
"a=ice-ufrag:ufrag_data\r\n"
"a=ice-pwd:pwd_data\r\n"
"a=mid:data_content_name\r\n"
@@ -278,7 +278,7 @@
static const char kSdpSctpDataChannelString[] =
"m=application 9 DTLS/SCTP 5000\r\n"
- "c=IN IP6 ::\r\n"
+ "c=IN IP4 0.0.0.0\r\n"
"a=ice-ufrag:ufrag_data\r\n"
"a=ice-pwd:pwd_data\r\n"
"a=mid:data_content_name\r\n"
@@ -289,7 +289,7 @@
"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 IP6 ::\r\n"
+ "c=IN IP4 0.0.0.0\r\n"
"a=ice-ufrag:ufrag_data\r\n"
"a=ice-pwd:pwd_data\r\n"
"a=mid:data_content_name\r\n";
@@ -316,10 +316,10 @@
"t=0 0\r\n"
"a=msid-semantic: WMS\r\n"
"m=audio 9 RTP/SAVPF 111 103 104\r\n"
- "c=IN IP6 ::\r\n"
+ "c=IN IP4 0.0.0.0\r\n"
"a=x-google-flag:conference\r\n"
"m=video 9 RTP/SAVPF 120\r\n"
- "c=IN IP6 ::\r\n"
+ "c=IN IP4 0.0.0.0\r\n"
"a=x-google-flag:conference\r\n";
static const char kSdpSessionString[] =
@@ -331,8 +331,8 @@
static const char kSdpAudioString[] =
"m=audio 9 RTP/SAVPF 111\r\n"
- "c=IN IP6 ::\r\n"
- "a=rtcp:9 IN IP6 ::\r\n"
+ "c=IN IP4 0.0.0.0\r\n"
+ "a=rtcp:9 IN IP4 0.0.0.0\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"
@@ -344,8 +344,8 @@
static const char kSdpVideoString[] =
"m=video 9 RTP/SAVPF 120\r\n"
- "c=IN IP6 ::\r\n"
- "a=rtcp:9 IN IP6 ::\r\n"
+ "c=IN IP4 0.0.0.0\r\n"
+ "a=rtcp:9 IN IP4 0.0.0.0\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"
@@ -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 9 RTP/SAVPF 101\r\nc=IN IP6 ::\r\n",
+ InjectAfter("m=application 9 RTP/SAVPF 101\r\nc=IN IP4 0.0.0.0\r\n",
"b=AS:100\r\n",
&expected_sdp);
EXPECT_EQ(expected_sdp, message);
diff --git a/examples/android/AndroidManifest.xml b/examples/android/AndroidManifest.xml
index 30fd46c..e24942a 100644
--- a/examples/android/AndroidManifest.xml
+++ b/examples/android/AndroidManifest.xml
@@ -7,7 +7,7 @@
<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-sdk android:minSdkVersion="13" android:targetSdkVersion="21" />
<uses-permission android:name="android.permission.CAMERA" />
<uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
@@ -43,7 +43,7 @@
android:label="@string/app_name"
android:screenOrientation="fullUser"
android:configChanges="orientation|screenSize"
- android:theme="@android:style/Theme.Black.NoTitleBar.Fullscreen">
+ android:theme="@style/AppRTCDemoActivityTheme">
</activity>
</application>
</manifest>
diff --git a/examples/android/res/values-v21/styles.xml b/examples/android/res/values-v21/styles.xml
new file mode 100644
index 0000000..95f1ac6
--- /dev/null
+++ b/examples/android/res/values-v21/styles.xml
@@ -0,0 +1,8 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+ <style name="AppRTCDemoActivityTheme" parent="android:Theme.Material">
+ <item name="android:windowActionBar">false</item>
+ <item name="android:windowFullscreen">true</item>
+ <item name="android:windowNoTitle">true</item>
+ </style>
+</resources>
diff --git a/examples/android/res/values/styles.xml b/examples/android/res/values/styles.xml
new file mode 100644
index 0000000..7f809a6
--- /dev/null
+++ b/examples/android/res/values/styles.xml
@@ -0,0 +1,8 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+ <style name="AppRTCDemoActivityTheme" parent="android:Theme.Black">
+ <item name="android:windowActionBar">false</item>
+ <item name="android:windowFullscreen">true</item>
+ <item name="android:windowNoTitle">true</item>
+ </style>
+</resources>
diff --git a/examples/android/src/org/appspot/apprtc/AppRTCAudioManager.java b/examples/android/src/org/appspot/apprtc/AppRTCAudioManager.java
new file mode 100644
index 0000000..b2a1a44
--- /dev/null
+++ b/examples/android/src/org/appspot/apprtc/AppRTCAudioManager.java
@@ -0,0 +1,108 @@
+/*
+ * libjingle
+ * Copyright 2013, 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.content.Context;
+import android.media.AudioManager;
+import android.util.Log;
+
+/**
+ * AppRTCAudioManager manages all audio related parts of the AppRTC demo.
+ * TODO(henrika): add support for device enumeration, device selection etc.
+ */
+public class AppRTCAudioManager {
+ private static final String TAG = "AppRTCAudioManager";
+
+ private boolean initialized = false;
+ private AudioManager audioManager;
+ private int savedAudioMode = AudioManager.MODE_INVALID;
+ private boolean savedIsSpeakerPhoneOn = false;
+ private boolean savedIsMicrophoneMute = false;
+
+ /** Construction */
+ static AppRTCAudioManager create(Context context) {
+ return new AppRTCAudioManager(context);
+ }
+
+ private AppRTCAudioManager(Context context) {
+ Log.d(TAG, "AppRTCAudioManager");
+ audioManager = ((AudioManager) context.getSystemService(
+ Context.AUDIO_SERVICE));
+ }
+
+ public void init() {
+ Log.d(TAG, "init");
+ if (initialized) {
+ return;
+ }
+
+ // Store current audio state so we can restore it when close() is called.
+ savedAudioMode = audioManager.getMode();
+ savedIsSpeakerPhoneOn = audioManager.isSpeakerphoneOn();
+ savedIsMicrophoneMute = audioManager.isMicrophoneMute();
+
+ // The AppRTC demo shall always run in COMMUNICATION mode since it will
+ // result in best possible "VoIP settings", like audio routing, volume
+ // control etc.
+ audioManager.setMode(AudioManager.MODE_IN_COMMUNICATION);
+
+ initialized = true;
+ }
+
+ public void close() {
+ Log.d(TAG, "close");
+ if (!initialized) {
+ return;
+ }
+
+ // Restore previously stored audio states.
+ setSpeakerphoneOn(savedIsSpeakerPhoneOn);
+ setMicrophoneMute(savedIsMicrophoneMute);
+ audioManager.setMode(savedAudioMode);
+
+ initialized = false;
+ }
+
+ /** Sets the speaker phone mode. */
+ private void setSpeakerphoneOn(boolean on) {
+ boolean wasOn = audioManager.isSpeakerphoneOn();
+ if (wasOn == on) {
+ return;
+ }
+ audioManager.setSpeakerphoneOn(on);
+ }
+
+ /** Sets the microphone mute state. */
+ private void setMicrophoneMute(boolean on) {
+ boolean wasMuted = audioManager.isMicrophoneMute();
+ if (wasMuted == on) {
+ return;
+ }
+ audioManager.setMicrophoneMute(on);
+ }
+}
diff --git a/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java b/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
index ad0e2d5..3ad26af 100644
--- a/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
+++ b/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
@@ -34,7 +34,6 @@
import android.content.Intent;
import android.content.pm.ActivityInfo;
import android.graphics.Color;
-import android.media.AudioManager;
import android.net.Uri;
import android.opengl.GLSurfaceView;
import android.os.Bundle;
@@ -72,6 +71,7 @@
private PeerConnectionClient pc;
private AppRTCClient appRtcClient = new GAERTCClient(this, this);
private AppRTCSignalingParameters appRtcParameters;
+ private AppRTCAudioManager audioManager = null;
private View rootView;
private View menuBar;
private GLSurfaceView videoView;
@@ -187,15 +187,9 @@
hudView.setVisibility(View.INVISIBLE);
addContentView(hudView, hudLayout);
- AudioManager audioManager =
- ((AudioManager) getSystemService(AUDIO_SERVICE));
- // TODO(fischman): figure out how to do this Right(tm) and remove the
- // suppression.
- @SuppressWarnings("deprecation")
- boolean isWiredHeadsetOn = audioManager.isWiredHeadsetOn();
- audioManager.setMode(isWiredHeadsetOn ?
- AudioManager.MODE_IN_CALL : AudioManager.MODE_IN_COMMUNICATION);
- audioManager.setSpeakerphoneOn(!isWiredHeadsetOn);
+ // Create and audio manager that will take care of audio routing,
+ // audio modes, audio device enumeration etc.
+ audioManager = AppRTCAudioManager.create(this);
final Intent intent = getIntent();
Uri url = intent.getData();
@@ -253,6 +247,10 @@
@Override
protected void onDestroy() {
disconnect();
+ if (audioManager != null) {
+ audioManager.close();
+ audioManager = null;
+ }
super.onDestroy();
}
@@ -360,6 +358,12 @@
// All events are called from UI thread.
@Override
public void onConnectedToRoom(final AppRTCSignalingParameters params) {
+ if (audioManager != null) {
+ // Store existing audio settings and change audio mode to
+ // MODE_IN_COMMUNICATION for best possible VoIP performance.
+ logAndToast("Initializing the audio manager...");
+ audioManager.init();
+ }
appRtcParameters = params;
abortUnless(PeerConnectionFactory.initializeAndroidGlobals(
this, true, true, VideoRendererGui.getEGLContext()),
diff --git a/examples/objc/AppRTCDemo/ios/APPRTCViewController.m b/examples/objc/AppRTCDemo/ios/APPRTCViewController.m
index 8042762..d8d9714 100644
--- a/examples/objc/AppRTCDemo/ios/APPRTCViewController.m
+++ b/examples/objc/AppRTCDemo/ios/APPRTCViewController.m
@@ -34,6 +34,7 @@
#import <AVFoundation/AVFoundation.h>
#import "APPRTCConnectionManager.h"
#import "RTCEAGLVideoView.h"
+#import "RTCVideoTrack.h"
// Padding space for local video view with its parent.
static CGFloat const kLocalViewPadding = 20;
@@ -47,6 +48,8 @@
@implementation APPRTCViewController {
APPRTCConnectionManager* _connectionManager;
+ RTCVideoTrack* _localVideoTrack;
+ RTCVideoTrack* _remoteVideoTrack;
CGSize _localVideoSize;
CGSize _remoteVideoSize;
}
@@ -101,13 +104,15 @@
- (void)connectionManager:(APPRTCConnectionManager*)manager
didReceiveLocalVideoTrack:(RTCVideoTrack*)localVideoTrack {
+ _localVideoTrack = localVideoTrack;
+ [_localVideoTrack addRenderer:self.localVideoView];
self.localVideoView.hidden = NO;
- self.localVideoView.videoTrack = localVideoTrack;
}
- (void)connectionManager:(APPRTCConnectionManager*)manager
didReceiveRemoteVideoTrack:(RTCVideoTrack*)remoteVideoTrack {
- self.remoteVideoView.videoTrack = remoteVideoTrack;
+ _remoteVideoTrack = remoteVideoTrack;
+ [_remoteVideoTrack addRenderer:self.remoteVideoView];
}
- (void)connectionManagerDidReceiveHangup:(APPRTCConnectionManager*)manager {
@@ -193,8 +198,16 @@
self.instructionsView.hidden = NO;
self.logView.hidden = YES;
self.logView.text = nil;
- self.localVideoView.videoTrack = nil;
- self.remoteVideoView.videoTrack = nil;
+ if (_localVideoTrack) {
+ [_localVideoTrack removeRenderer:self.localVideoView];
+ _localVideoTrack = nil;
+ [self.localVideoView renderFrame:nil];
+ }
+ if (_remoteVideoTrack) {
+ [_remoteVideoTrack removeRenderer:self.remoteVideoView];
+ _remoteVideoTrack = nil;
+ [self.remoteVideoView renderFrame:nil];
+ }
self.blackView.hidden = YES;
}
diff --git a/examples/objc/AppRTCDemo/mac/APPRTCViewController.m b/examples/objc/AppRTCDemo/mac/APPRTCViewController.m
index cf5b836..08acac9 100644
--- a/examples/objc/AppRTCDemo/mac/APPRTCViewController.m
+++ b/examples/objc/AppRTCDemo/mac/APPRTCViewController.m
@@ -30,6 +30,7 @@
#import <AVFoundation/AVFoundation.h>
#import "APPRTCConnectionManager.h"
#import "RTCNSGLVideoView.h"
+#import "RTCVideoTrack.h"
static NSUInteger const kContentWidth = 1280;
static NSUInteger const kContentHeight = 720;
@@ -227,6 +228,8 @@
@implementation APPRTCViewController {
APPRTCConnectionManager* _connectionManager;
+ RTCVideoTrack* _localVideoTrack;
+ RTCVideoTrack* _remoteVideoTrack;
}
- (instancetype)initWithNibName:(NSString*)nibName
@@ -258,12 +261,13 @@
- (void)connectionManager:(APPRTCConnectionManager*)manager
didReceiveLocalVideoTrack:(RTCVideoTrack*)localVideoTrack {
- self.mainView.localVideoView.videoTrack = localVideoTrack;
+ _localVideoTrack = localVideoTrack;
}
- (void)connectionManager:(APPRTCConnectionManager*)manager
didReceiveRemoteVideoTrack:(RTCVideoTrack*)remoteVideoTrack {
- self.mainView.remoteVideoView.videoTrack = remoteVideoTrack;
+ _remoteVideoTrack = remoteVideoTrack;
+ [_remoteVideoTrack addRenderer:self.mainView.remoteVideoView];
}
- (void)connectionManagerDidReceiveHangup:(APPRTCConnectionManager*)manager {
@@ -305,7 +309,9 @@
}
- (void)disconnect {
- self.mainView.remoteVideoView.videoTrack = nil;
+ [_remoteVideoTrack removeRenderer:self.mainView.remoteVideoView];
+ _remoteVideoTrack = nil;
+ [self.mainView.remoteVideoView renderFrame:nil];
[_connectionManager disconnect];
}
diff --git a/libjingle.gyp b/libjingle.gyp
index 803eaa3..90d1f41 100755
--- a/libjingle.gyp
+++ b/libjingle.gyp
@@ -204,8 +204,8 @@
'app/webrtc/objc/RTCStatsReport.mm',
'app/webrtc/objc/RTCVideoCapturer+Internal.h',
'app/webrtc/objc/RTCVideoCapturer.mm',
- 'app/webrtc/objc/RTCVideoRenderer+Internal.h',
- 'app/webrtc/objc/RTCVideoRenderer.mm',
+ 'app/webrtc/objc/RTCVideoRendererAdapter.h',
+ 'app/webrtc/objc/RTCVideoRendererAdapter.mm',
'app/webrtc/objc/RTCVideoSource+Internal.h',
'app/webrtc/objc/RTCVideoSource.mm',
'app/webrtc/objc/RTCVideoTrack+Internal.h',
@@ -259,7 +259,6 @@
'conditions': [
['OS=="ios"', {
'sources': [
- 'app/webrtc/objc/RTCEAGLVideoView+Internal.h',
'app/webrtc/objc/RTCEAGLVideoView.m',
'app/webrtc/objc/public/RTCEAGLVideoView.h',
],
diff --git a/libjingle_examples.gyp b/libjingle_examples.gyp
index d0f1747..f7ce53b 100755
--- a/libjingle_examples.gyp
+++ b/libjingle_examples.gyp
@@ -344,6 +344,7 @@
'examples/android/res/values/arrays.xml',
'examples/android/res/values/strings.xml',
'examples/android/res/xml/preferences.xml',
+ 'examples/android/src/org/appspot/apprtc/AppRTCAudioManager.java',
'examples/android/src/org/appspot/apprtc/AppRTCClient.java',
'examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java',
'examples/android/src/org/appspot/apprtc/ConnectActivity.java',
diff --git a/media/base/fakemediaengine.h b/media/base/fakemediaengine.h
index db5e2e4..ff6ae16 100644
--- a/media/base/fakemediaengine.h
+++ b/media/base/fakemediaengine.h
@@ -879,9 +879,6 @@
default_encoder_config_ = config;
return true;
}
- VideoEncoderConfig GetDefaultEncoderConfig() const {
- return default_encoder_config_;
- }
const VideoEncoderConfig& default_encoder_config() const {
return default_encoder_config_;
}
diff --git a/media/base/filemediaengine.h b/media/base/filemediaengine.h
index 9cc066d..ba29ab3 100644
--- a/media/base/filemediaengine.h
+++ b/media/base/filemediaengine.h
@@ -93,9 +93,6 @@
virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config) {
return true;
}
- virtual VideoEncoderConfig GetDefaultVideoEncoderConfig() const {
- return VideoEncoderConfig();
- }
virtual bool SetSoundDevices(const Device* in_dev, const Device* out_dev) {
return true;
}
diff --git a/media/base/mediaengine.h b/media/base/mediaengine.h
index e7222f2..b8d661c 100644
--- a/media/base/mediaengine.h
+++ b/media/base/mediaengine.h
@@ -99,10 +99,6 @@
// and encode video.
virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config)
= 0;
- // Gets the default (maximum) codec/resolution and encoder option used to
- // capture and encode video, as set by SetDefaultVideoEncoderConfig or the
- // default from the video engine if not previously set.
- virtual VideoEncoderConfig GetDefaultVideoEncoderConfig() const = 0;
// Device selection
// TODO(tschmelcher): Add method for selecting the soundclip device.
@@ -219,9 +215,6 @@
virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config) {
return video_.SetDefaultEncoderConfig(config);
}
- virtual VideoEncoderConfig GetDefaultVideoEncoderConfig() const {
- return video_.GetDefaultEncoderConfig();
- }
virtual bool SetSoundDevices(const Device* in_device,
const Device* out_device) {
@@ -347,9 +340,6 @@
return NULL;
}
bool SetOptions(const VideoOptions& options) { return true; }
- VideoEncoderConfig GetDefaultEncoderConfig() const {
- return VideoEncoderConfig();
- }
bool SetDefaultEncoderConfig(const VideoEncoderConfig& config) {
return true;
}
diff --git a/media/webrtc/constants.h b/media/webrtc/constants.h
index 5390c0d..821a41f 100755
--- a/media/webrtc/constants.h
+++ b/media/webrtc/constants.h
@@ -35,7 +35,9 @@
extern const char kVp8CodecName[];
extern const char kH264CodecName[];
-extern const int kDefaultFramerate;
+extern const int kDefaultVideoMaxWidth;
+extern const int kDefaultVideoMaxHeight;
+extern const int kDefaultVideoMaxFramerate;
extern const int kMinVideoBitrate;
extern const int kStartVideoBitrate;
extern const int kMaxVideoBitrate;
diff --git a/media/webrtc/webrtcmediaengine.h b/media/webrtc/webrtcmediaengine.h
index bfcfb2a..b4436cd 100644
--- a/media/webrtc/webrtcmediaengine.h
+++ b/media/webrtc/webrtcmediaengine.h
@@ -130,9 +130,6 @@
const VideoEncoderConfig& config) OVERRIDE {
return delegate_->SetDefaultVideoEncoderConfig(config);
}
- virtual VideoEncoderConfig GetDefaultVideoEncoderConfig() const OVERRIDE {
- return delegate_->GetDefaultVideoEncoderConfig();
- }
virtual bool SetSoundDevices(
const Device* in_device, const Device* out_device) OVERRIDE {
return delegate_->SetSoundDevices(in_device, out_device);
diff --git a/media/webrtc/webrtcvideoengine.cc b/media/webrtc/webrtcvideoengine.cc
index 04092f3..46d316f 100644
--- a/media/webrtc/webrtcvideoengine.cc
+++ b/media/webrtc/webrtcvideoengine.cc
@@ -55,6 +55,7 @@
#include "webrtc/base/basictypes.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/byteorder.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/cpumonitor.h"
#include "webrtc/base/logging.h"
@@ -106,7 +107,10 @@
const char kVp8CodecName[] = "VP8";
-const int kDefaultFramerate = 30;
+// TODO(ronghuawu): Change to 640x360.
+const int kDefaultVideoMaxWidth = 640;
+const int kDefaultVideoMaxHeight = 400;
+const int kDefaultVideoMaxFramerate = 30;
const int kMinVideoBitrate = 30;
const int kStartVideoBitrate = 300;
const int kMaxVideoBitrate = 2000;
@@ -176,15 +180,57 @@
static const rtc::DiffServCodePoint kVideoDscpValue =
rtc::DSCP_AF41;
-static bool IsNackEnabled(const VideoCodec& codec) {
- return codec.HasFeedbackParam(FeedbackParam(kRtcpFbParamNack,
- kParamValueEmpty));
+bool IsNackEnabled(const VideoCodec& codec) {
+ return codec.HasFeedbackParam(
+ FeedbackParam(kRtcpFbParamNack, kParamValueEmpty));
}
-// Returns true if Receiver Estimated Max Bitrate is enabled.
-static bool IsRembEnabled(const VideoCodec& codec) {
- return codec.HasFeedbackParam(FeedbackParam(kRtcpFbParamRemb,
- kParamValueEmpty));
+bool IsRembEnabled(const VideoCodec& codec) {
+ return codec.HasFeedbackParam(
+ FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty));
+}
+
+void AddDefaultFeedbackParams(VideoCodec* codec) {
+ codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamCcm, kRtcpFbCcmParamFir));
+ codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kParamValueEmpty));
+ codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli));
+ codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty));
+}
+
+bool CodecNameMatches(const std::string& name1, const std::string& name2) {
+ return _stricmp(name1.c_str(), name2.c_str()) == 0;
+}
+
+static VideoCodec MakeVideoCodecWithDefaultFeedbackParams(int payload_type,
+ const char* name) {
+ VideoCodec codec(payload_type, name, kDefaultVideoMaxWidth,
+ kDefaultVideoMaxHeight, kDefaultVideoMaxFramerate, 0);
+ AddDefaultFeedbackParams(&codec);
+ return codec;
+}
+
+static VideoCodec MakeVideoCodec(int payload_type, const char* name) {
+ return VideoCodec(payload_type, name, 0, 0, 0, 0);
+}
+
+static VideoCodec MakeRtxCodec(int payload_type, int associated_payload_type) {
+ return VideoCodec::CreateRtxCodec(payload_type, associated_payload_type);
+}
+
+bool CodecIsInternallySupported(const std::string& codec_name) {
+ if (CodecNameMatches(codec_name, kVp8CodecName)) {
+ return true;
+ }
+ return false;
+}
+
+std::vector<VideoCodec> DefaultVideoCodecList() {
+ std::vector<VideoCodec> codecs;
+ codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(100, kVp8CodecName));
+ codecs.push_back(MakeRtxCodec(96, 100));
+ codecs.push_back(MakeVideoCodec(116, kRedCodecName));
+ codecs.push_back(MakeVideoCodec(117, kUlpfecCodecName));
+ return codecs;
}
struct FlushBlackFrameData : public rtc::MessageData {
@@ -923,18 +969,6 @@
AdaptFormatType adapt_format_type_;
};
-const WebRtcVideoEngine::VideoCodecPref
- WebRtcVideoEngine::kVideoCodecPrefs[] = {
- {kVp8CodecName, 100, -1, 0},
- {kRedCodecName, 116, -1, 1},
- {kUlpfecCodecName, 117, -1, 2},
- {kRtxCodecName, 96, 100, 3},
-};
-
-const VideoFormatPod WebRtcVideoEngine::kDefaultMaxVideoFormat =
- {640, 400, FPS_TO_INTERVAL(30), FOURCC_ANY};
-// TODO(ronghuawu): Change to 640x360.
-
static bool GetCpuOveruseOptions(const VideoOptions& options,
webrtc::CpuOveruseOptions* overuse_options) {
int underuse_threshold = 0;
@@ -1018,20 +1052,14 @@
LOG_RTCERR1(SetTraceCallback, this);
}
+ default_video_codec_list_ = DefaultVideoCodecList();
+
// Set default quality levels for our supported codecs. We override them here
// if we know your cpu performance is low, and they can be updated explicitly
// by calling SetDefaultCodec. For example by a flute preference setting, or
// by the server with a jec in response to our reported system info.
- VideoCodec max_codec(kVideoCodecPrefs[0].payload_type,
- kVideoCodecPrefs[0].name,
- kDefaultMaxVideoFormat.width,
- kDefaultMaxVideoFormat.height,
- VideoFormat::IntervalToFps(
- kDefaultMaxVideoFormat.interval),
- 0);
- if (!SetDefaultCodec(max_codec)) {
- LOG(LS_ERROR) << "Failed to initialize list of supported codec types";
- }
+ CHECK(SetDefaultCodec(default_video_codec_list_.front()))
+ << "Failed to initialize list of supported codec types.";
// Consider jitter, packet loss, etc when rendering. This will
// theoretically make rendering more smooth.
@@ -1144,17 +1172,6 @@
return SetDefaultCodec(config.max_codec);
}
-VideoEncoderConfig WebRtcVideoEngine::GetDefaultEncoderConfig() const {
- ASSERT(!video_codecs_.empty());
- VideoCodec max_codec(kVideoCodecPrefs[0].payload_type,
- kVideoCodecPrefs[0].name,
- video_codecs_[0].width,
- video_codecs_[0].height,
- video_codecs_[0].framerate,
- 0);
- return VideoEncoderConfig(max_codec);
-}
-
// SetDefaultCodec may be called while the capturer is running. For example, a
// test call is started in a page with QVGA default codec, and then a real call
// is started in another page with VGA default codec. This is the corner case
@@ -1225,13 +1242,12 @@
return true;
}
}
- for (size_t j = 0; j < ARRAY_SIZE(kVideoCodecPrefs); ++j) {
- VideoCodec codec(kVideoCodecPrefs[j].payload_type,
- kVideoCodecPrefs[j].name, 0, 0, 0, 0);
- if (codec.Matches(in)) {
+ for (size_t j = 0; j != default_video_codec_list_.size(); ++j) {
+ if (default_video_codec_list_[j].Matches(in)) {
return true;
}
}
+
return false;
}
@@ -1439,17 +1455,6 @@
}
}
-static void AddDefaultFeedbackParams(VideoCodec* codec) {
- const FeedbackParam kFir(kRtcpFbParamCcm, kRtcpFbCcmParamFir);
- codec->AddFeedbackParam(kFir);
- const FeedbackParam kNack(kRtcpFbParamNack, kParamValueEmpty);
- codec->AddFeedbackParam(kNack);
- const FeedbackParam kPli(kRtcpFbParamNack, kRtcpFbNackParamPli);
- codec->AddFeedbackParam(kPli);
- const FeedbackParam kRemb(kRtcpFbParamRemb, kParamValueEmpty);
- codec->AddFeedbackParam(kRemb);
-}
-
// Rebuilds the codec list to be only those that are less intensive
// than the specified codec. Prefers internal codec over external with
// higher preference field.
@@ -1459,27 +1464,17 @@
video_codecs_.clear();
- bool found = false;
std::set<std::string> internal_codec_names;
- for (size_t i = 0; i < ARRAY_SIZE(kVideoCodecPrefs); ++i) {
- const VideoCodecPref& pref(kVideoCodecPrefs[i]);
- if (!found)
- found = (in_codec.name == pref.name);
- if (found) {
- VideoCodec codec(pref.payload_type, pref.name,
- in_codec.width, in_codec.height, in_codec.framerate,
- static_cast<int>(ARRAY_SIZE(kVideoCodecPrefs) - i));
- if (_stricmp(kVp8CodecName, codec.name.c_str()) == 0) {
- AddDefaultFeedbackParams(&codec);
- }
- if (pref.associated_payload_type != -1) {
- codec.SetParam(kCodecParamAssociatedPayloadType,
- pref.associated_payload_type);
- }
- video_codecs_.push_back(codec);
- internal_codec_names.insert(codec.name);
- }
+ for (size_t i = 0; i != default_video_codec_list_.size(); ++i) {
+ VideoCodec codec = default_video_codec_list_[i];
+ codec.width = in_codec.width;
+ codec.height = in_codec.height;
+ codec.framerate = in_codec.framerate;
+ video_codecs_.push_back(codec);
+
+ internal_codec_names.insert(codec.name);
}
+
if (encoder_factory_) {
const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs =
encoder_factory_->codecs();
@@ -1487,8 +1482,6 @@
bool is_internal_codec = internal_codec_names.find(codecs[i].name) !=
internal_codec_names.end();
if (!is_internal_codec) {
- if (!found)
- found = (in_codec.name == codecs[i].name);
VideoCodec codec(
GetExternalVideoPayloadType(static_cast<int>(i)),
codecs[i].name,
@@ -1503,7 +1496,6 @@
}
}
}
- ASSERT(found);
return true;
}
@@ -1609,12 +1601,10 @@
encoder_factory_ = encoder_factory;
// Rebuild codec list while reapplying the current default codec format.
- VideoCodec max_codec(kVideoCodecPrefs[0].payload_type,
- kVideoCodecPrefs[0].name,
- video_codecs_[0].width,
- video_codecs_[0].height,
- video_codecs_[0].framerate,
- 0);
+ VideoCodec max_codec = default_video_codec_list_[0];
+ max_codec.width = video_codecs_[0].width;
+ max_codec.height = video_codecs_[0].height;
+ max_codec.framerate = video_codecs_[0].framerate;
if (!RebuildCodecList(max_codec)) {
LOG(LS_ERROR) << "Failed to initialize list of supported codec types";
}
@@ -1661,10 +1651,12 @@
}
WebRtcVideoMediaChannel::~WebRtcVideoMediaChannel() {
- const bool send = false;
- SetSend(send);
- const bool render = false;
- SetRender(render);
+ Terminate();
+}
+
+void WebRtcVideoMediaChannel::Terminate() {
+ SetSend(false);
+ SetRender(false);
while (!send_channels_.empty()) {
if (!DeleteSendChannel(send_channels_.begin()->first)) {
@@ -1987,14 +1979,21 @@
SetReceiverReportSsrc(sp.first_ssrc());
}
- send_channel->set_stream_params(sp);
-
- // Reset send codec after stream parameters changed.
if (send_codec_) {
- if (!SetSendCodec(send_channel, *send_codec_)) {
+ send_channel->SetAdaptFormat(
+ VideoFormatFromVieCodec(*send_codec_),
+ WebRtcVideoChannelSendInfo::kAdaptFormatTypeCodec);
+
+ VideoSendParams send_params;
+ send_params.codec = *send_codec_;
+ send_params.stream = sp;
+ if (!SetSendParams(send_channel, send_params)) {
return false;
}
- LogSendCodecChange("SetSendStreamFormat()");
+ LogSendCodecChange("AddStream()");
+ } else {
+ // Save the stream params for later, when we have a codec.
+ send_channel->set_stream_params(sp);
}
if (sending_) {
@@ -2990,39 +2989,40 @@
VideoOptions original = options_;
options_.SetAll(options);
- // Set CPU options for all send channels.
+ // Set CPU options and codec options for all send channels.
for (SendChannelMap::iterator iter = send_channels_.begin();
iter != send_channels_.end(); ++iter) {
WebRtcVideoChannelSendInfo* send_channel = iter->second;
send_channel->ApplyCpuOptions(options_);
+
+ if (send_codec_) {
+ VideoSendParams send_params = send_channel->send_params();
+
+ bool conference_mode_turned_off = (
+ original.conference_mode.IsSet() &&
+ options.conference_mode.IsSet() &&
+ original.conference_mode.GetWithDefaultIfUnset(false) &&
+ !options.conference_mode.GetWithDefaultIfUnset(false));
+ if (conference_mode_turned_off) {
+ // This is a special case for turning conference mode off.
+ // Max bitrate should go back to the default maximum value instead
+ // of the current maximum.
+ send_params.codec.maxBitrate = kAutoBandwidth;
+ }
+
+ // TODO(pthatcher): Remove this. We don't need 4 ways to set bitrates.
+ int new_start_bitrate;
+ if (options.video_start_bitrate.Get(&new_start_bitrate)) {
+ send_params.codec.startBitrate = new_start_bitrate;
+ }
+
+ if (!SetSendParams(send_channel, send_params)) {
+ return false;
+ }
+ LogSendCodecChange("SetOptions()");
+ }
}
- if (send_codec_) {
- webrtc::VideoCodec new_codec = *send_codec_;
-
- bool conference_mode_turned_off = (
- original.conference_mode.IsSet() &&
- options.conference_mode.IsSet() &&
- original.conference_mode.GetWithDefaultIfUnset(false) &&
- !options.conference_mode.GetWithDefaultIfUnset(false));
- if (conference_mode_turned_off) {
- // This is a special case for turning conference mode off.
- // Max bitrate should go back to the default maximum value instead
- // of the current maximum.
- new_codec.maxBitrate = kAutoBandwidth;
- }
-
- // TODO(pthatcher): Remove this. We don't need 4 ways to set bitrates.
- int new_start_bitrate;
- if (options.video_start_bitrate.Get(&new_start_bitrate)) {
- new_codec.startBitrate = new_start_bitrate;
- }
-
- if (!SetSendCodec(new_codec)) {
- return false;
- }
- LogSendCodecChange("SetOptions()");
- }
int buffer_latency;
if (Changed(options.buffered_mode_latency,
@@ -3676,26 +3676,9 @@
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();
- 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;
- }
-
- return true;
+ return SetSendParams(send_channel, send_params);
}
static std::string ToString(webrtc::VideoCodecComplexity complexity) {
@@ -3872,6 +3855,8 @@
const VideoSendParams& send_params) {
const int channel_id = send_channel->channel_id();
+ MaybeRegisterExternalEncoder(send_channel, send_params.codec);
+
CapturedFrameInfo frame;
send_channel->last_captured_frame_info().Get(&frame);
@@ -3923,10 +3908,18 @@
}
engine()->vie()->rtp()->SetTransmissionSmoothingStatus(channel_id, true);
- if (send_channel->IsActive()) {
- if (!SetSendSsrcs(channel_id, send_params.stream, codec)) {
- return false;
- }
+ if (!SetSendSsrcs(channel_id, send_params.stream, codec)) {
+ return false;
+ }
+
+ // NOTE: SetRtxSendPayloadType must be called after all 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_send_params(send_params);
diff --git a/media/webrtc/webrtcvideoengine.h b/media/webrtc/webrtcvideoengine.h
index cc81ee9..db091af 100644
--- a/media/webrtc/webrtcvideoengine.h
+++ b/media/webrtc/webrtcvideoengine.h
@@ -85,6 +85,15 @@
struct CapturedFrame;
struct Device;
+// This set of methods is declared here for the sole purpose of sharing code
+// between webrtc video engine v1 and v2.
+std::vector<VideoCodec> DefaultVideoCodecList();
+bool CodecNameMatches(const std::string& name1, const std::string& name2);
+bool CodecIsInternallySupported(const std::string& codec_name);
+bool IsNackEnabled(const VideoCodec& codec);
+bool IsRembEnabled(const VideoCodec& codec);
+void AddDefaultFeedbackParams(VideoCodec* codec);
+
class WebRtcVideoEngine : public sigslot::has_slots<>,
public webrtc::TraceCallback {
public:
@@ -108,7 +117,6 @@
int GetCapabilities();
bool SetDefaultEncoderConfig(const VideoEncoderConfig& config);
- VideoEncoderConfig GetDefaultEncoderConfig() const;
// TODO(pbos): Remove when all call sites use VideoOptions.
virtual WebRtcVideoMediaChannel* CreateChannel(
@@ -177,6 +185,10 @@
rtc::CpuMonitor* cpu_monitor() { return cpu_monitor_.get(); }
protected:
+ bool initialized() const {
+ return initialized_;
+ }
+
// When a video processor registers with the engine.
// SignalMediaFrame will be invoked for every video frame.
// See videoprocessor.h for param reference.
@@ -184,18 +196,6 @@
private:
typedef std::vector<WebRtcVideoMediaChannel*> VideoChannels;
- struct VideoCodecPref {
- const char* name;
- int payload_type;
- // For RTX, this field is the payload-type that RTX applies to.
- // For other codecs, it should be set to -1.
- int associated_payload_type;
- int pref;
- };
-
- static const VideoCodecPref kVideoCodecPrefs[];
- static const VideoFormatPod kVideoFormats[];
- static const VideoFormatPod kDefaultMaxVideoFormat;
void Construct(ViEWrapper* vie_wrapper,
ViETraceWrapper* tracing,
@@ -222,6 +222,7 @@
WebRtcVideoEncoderFactory* encoder_factory_;
WebRtcVideoDecoderFactory* decoder_factory_;
std::vector<VideoCodec> video_codecs_;
+ std::vector<VideoCodec> default_video_codec_list_;
std::vector<RtpHeaderExtension> rtp_header_extensions_;
VideoFormat default_codec_format_;
@@ -326,6 +327,7 @@
virtual void OnMessage(rtc::Message* msg) OVERRIDE;
protected:
+ void Terminate();
int GetLastEngineError() { return engine()->GetLastEngineError(); }
// webrtc::Transport:
diff --git a/media/webrtc/webrtcvideoengine2.cc b/media/webrtc/webrtcvideoengine2.cc
index d79f71d..5b5f12e 100644
--- a/media/webrtc/webrtcvideoengine2.cc
+++ b/media/webrtc/webrtcvideoengine2.cc
@@ -36,6 +36,7 @@
#include "talk/media/base/videorenderer.h"
#include "talk/media/webrtc/constants.h"
#include "talk/media/webrtc/webrtcvideocapturer.h"
+#include "talk/media/webrtc/webrtcvideoengine.h"
#include "talk/media/webrtc/webrtcvideoframe.h"
#include "talk/media/webrtc/webrtcvoiceengine.h"
#include "webrtc/base/buffer.h"
@@ -51,26 +52,6 @@
namespace cricket {
namespace {
-
-static bool CodecNameMatches(const std::string& name1,
- const std::string& name2) {
- 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) {
- 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) {
std::stringstream out;
out << '{';
@@ -116,6 +97,29 @@
return out.str();
}
+// Merges two fec configs and logs an error if a conflict arises
+// such that merging in diferent order would trigger a diferent output.
+static void MergeFecConfig(const webrtc::FecConfig& other,
+ webrtc::FecConfig* output) {
+ if (other.ulpfec_payload_type != -1) {
+ if (output->ulpfec_payload_type != -1 &&
+ output->ulpfec_payload_type != other.ulpfec_payload_type) {
+ LOG(LS_WARNING) << "Conflict merging ulpfec_payload_type configs: "
+ << output->ulpfec_payload_type << " and "
+ << other.ulpfec_payload_type;
+ }
+ output->ulpfec_payload_type = other.ulpfec_payload_type;
+ }
+ if (other.red_payload_type != -1) {
+ if (output->red_payload_type != -1 &&
+ output->red_payload_type != other.red_payload_type) {
+ LOG(LS_WARNING) << "Conflict merging red_payload_type configs: "
+ << output->red_payload_type << " and "
+ << other.red_payload_type;
+ }
+ output->red_payload_type = other.red_payload_type;
+ }
+}
} // namespace
// This constant is really an on/off, lower-level configurable NACK history
@@ -135,19 +139,8 @@
static const size_t kMaxExternalVideoCodecs = 8;
#endif
-struct VideoCodecPref {
- int payload_type;
- int width;
- int height;
- const char* name;
- int rtx_payload_type;
-} kDefaultVideoCodecPref = {100, 640, 400, kVp8CodecName, 96};
-
const char kH264CodecName[] = "H264";
-VideoCodecPref kRedPref = {116, -1, -1, kRedCodecName, -1};
-VideoCodecPref kUlpfecPref = {117, -1, -1, kUlpfecCodecName, -1};
-
static bool FindFirstMatchingCodec(const std::vector<VideoCodec>& codecs,
const VideoCodec& requested_codec,
VideoCodec* matching_codec) {
@@ -160,59 +153,6 @@
return false;
}
-static void AddDefaultFeedbackParams(VideoCodec* codec) {
- const FeedbackParam kFir(kRtcpFbParamCcm, kRtcpFbCcmParamFir);
- codec->AddFeedbackParam(kFir);
- const FeedbackParam kNack(kRtcpFbParamNack, kParamValueEmpty);
- codec->AddFeedbackParam(kNack);
- const FeedbackParam kPli(kRtcpFbParamNack, kRtcpFbNackParamPli);
- codec->AddFeedbackParam(kPli);
- const FeedbackParam kRemb(kRtcpFbParamRemb, kParamValueEmpty);
- codec->AddFeedbackParam(kRemb);
-}
-
-static bool IsNackEnabled(const VideoCodec& codec) {
- return codec.HasFeedbackParam(
- FeedbackParam(kRtcpFbParamNack, kParamValueEmpty));
-}
-
-static bool IsRembEnabled(const VideoCodec& codec) {
- return codec.HasFeedbackParam(
- FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty));
-}
-
-static VideoCodec DefaultVideoCodec() {
- VideoCodec default_codec(kDefaultVideoCodecPref.payload_type,
- kDefaultVideoCodecPref.name,
- kDefaultVideoCodecPref.width,
- kDefaultVideoCodecPref.height,
- kDefaultFramerate,
- 0);
- AddDefaultFeedbackParams(&default_codec);
- return default_codec;
-}
-
-static VideoCodec DefaultRedCodec() {
- return VideoCodec(kRedPref.payload_type, kRedPref.name, 0, 0, 0, 0);
-}
-
-static VideoCodec DefaultUlpfecCodec() {
- return VideoCodec(kUlpfecPref.payload_type, kUlpfecPref.name, 0, 0, 0, 0);
-}
-
-static std::vector<VideoCodec> DefaultVideoCodecs() {
- std::vector<VideoCodec> codecs;
- codecs.push_back(DefaultVideoCodec());
- codecs.push_back(DefaultRedCodec());
- codecs.push_back(DefaultUlpfecCodec());
- if (kDefaultVideoCodecPref.rtx_payload_type != -1) {
- codecs.push_back(
- VideoCodec::CreateRtxCodec(kDefaultVideoCodecPref.rtx_payload_type,
- kDefaultVideoCodecPref.payload_type));
- }
- return codecs;
-}
-
static bool ValidateRtpHeaderExtensionIds(
const std::vector<RtpHeaderExtension>& extensions) {
std::set<int> extensions_used;
@@ -258,7 +198,7 @@
stream.width = codec.width;
stream.height = codec.height;
stream.max_framerate =
- codec.framerate != 0 ? codec.framerate : kDefaultFramerate;
+ codec.framerate != 0 ? codec.framerate : kDefaultVideoMaxFramerate;
int min_bitrate = kMinVideoBitrate;
codec.GetParam(kCodecParamMinBitrate, &min_bitrate);
@@ -350,9 +290,9 @@
WebRtcVideoEngine2::WebRtcVideoEngine2()
: worker_thread_(NULL),
voice_engine_(NULL),
- default_codec_format_(kDefaultVideoCodecPref.width,
- kDefaultVideoCodecPref.height,
- FPS_TO_INTERVAL(kDefaultFramerate),
+ default_codec_format_(kDefaultVideoMaxWidth,
+ kDefaultVideoMaxHeight,
+ FPS_TO_INTERVAL(kDefaultVideoMaxFramerate),
FOURCC_ANY),
initialized_(false),
cpu_monitor_(new rtc::CpuMonitor(NULL)),
@@ -427,10 +367,6 @@
return true;
}
-VideoEncoderConfig WebRtcVideoEngine2::GetDefaultEncoderConfig() const {
- return VideoEncoderConfig(DefaultVideoCodec());
-}
-
WebRtcVideoChannel2* WebRtcVideoEngine2::CreateChannel(
const VideoOptions& options,
VoiceMediaChannel* voice_channel) {
@@ -575,7 +511,7 @@
}
std::vector<VideoCodec> WebRtcVideoEngine2::GetSupportedCodecs() const {
- std::vector<VideoCodec> supported_codecs = DefaultVideoCodecs();
+ std::vector<VideoCodec> supported_codecs = DefaultVideoCodecList();
if (external_encoder_factory_ == NULL) {
return supported_codecs;
@@ -900,9 +836,16 @@
return false;
}
- send_codec_.Set(supported_codecs.front());
LOG(LS_INFO) << "Using codec: " << supported_codecs.front().codec.ToString();
+ VideoCodecSettings old_codec;
+ if (send_codec_.Get(&old_codec) && supported_codecs.front() == old_codec) {
+ // Using same codec, avoid reconfiguring.
+ return true;
+ }
+
+ send_codec_.Set(supported_codecs.front());
+
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
send_streams_.begin();
@@ -1101,19 +1044,19 @@
}
for (size_t i = 0; i < recv_codecs_.size(); ++i) {
- if (recv_codecs_[i].codec.id == kDefaultVideoCodecPref.payload_type) {
- config->rtp.fec = recv_codecs_[i].fec;
- uint32 rtx_ssrc;
- if (recv_codecs_[i].rtx_payload_type != -1 &&
- sp.GetFidSsrc(ssrc, &rtx_ssrc)) {
- config->rtp.rtx[kDefaultVideoCodecPref.payload_type].ssrc = rtx_ssrc;
- config->rtp.rtx[kDefaultVideoCodecPref.payload_type].payload_type =
- recv_codecs_[i].rtx_payload_type;
- }
- break;
- }
+ MergeFecConfig(recv_codecs_[i].fec, &config->rtp.fec);
}
+ for (size_t i = 0; i < recv_codecs_.size(); ++i) {
+ uint32 rtx_ssrc;
+ if (recv_codecs_[i].rtx_payload_type != -1 &&
+ sp.GetFidSsrc(ssrc, &rtx_ssrc)) {
+ webrtc::VideoReceiveStream::Config::Rtp::Rtx& rtx =
+ config->rtp.rtx[recv_codecs_[i].codec.id];
+ rtx.ssrc = rtx_ssrc;
+ rtx.payload_type = recv_codecs_[i].rtx_payload_type;
+ }
+ }
}
bool WebRtcVideoChannel2::RemoveRecvStream(uint32 ssrc) {
@@ -1202,7 +1145,21 @@
void WebRtcVideoChannel2::FillBandwidthEstimationStats(
VideoMediaInfo* video_media_info) {
- // TODO(pbos): Implement.
+ BandwidthEstimationInfo bwe_info;
+ webrtc::Call::Stats stats = call_->GetStats();
+ bwe_info.available_send_bandwidth = stats.send_bandwidth_bps;
+ bwe_info.available_recv_bandwidth = stats.recv_bandwidth_bps;
+ bwe_info.bucket_delay = stats.pacer_delay_ms;
+
+ // Get send stream bitrate stats.
+ rtc::CritScope stream_lock(&stream_crit_);
+ for (std::map<uint32, WebRtcVideoSendStream*>::iterator stream =
+ send_streams_.begin();
+ stream != send_streams_.end();
+ ++stream) {
+ stream->second->FillBandwidthEstimationInfo(&bwe_info);
+ }
+ video_media_info->bw_estimations.push_back(bwe_info);
}
bool WebRtcVideoChannel2::SetCapturer(uint32 ssrc, VideoCapturer* capturer) {
@@ -1322,6 +1279,7 @@
return false;
send_rtp_extensions_ = FilterRtpExtensions(extensions);
+
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
send_streams_.begin();
@@ -1339,8 +1297,13 @@
}
bool WebRtcVideoChannel2::SetOptions(const VideoOptions& options) {
- LOG(LS_VERBOSE) << "SetOptions: " << options.ToString();
+ LOG(LS_INFO) << "SetOptions: " << options.ToString();
+ VideoOptions old_options = options_;
options_.SetAll(options);
+ if (options_ == old_options) {
+ // No new options to set.
+ return true;
+ }
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32, WebRtcVideoSendStream*>::iterator it =
send_streams_.begin();
@@ -1552,13 +1515,16 @@
LOG(LS_VERBOSE) << "Disabling capturer, sending black frame.";
webrtc::I420VideoFrame black_frame;
+ // TODO(pbos): Base width/height on last_dimensions_. This will however
+ // fail the test AddRemoveCapturer which needs to be fixed to permit
+ // sending black frames in the same size that was previously sent.
int width = format_.width;
int height = format_.height;
int half_width = (width + 1) / 2;
black_frame.CreateEmptyFrame(
width, height, width, half_width, half_width);
SetWebRtcFrameToBlack(&black_frame);
- SetDimensions(width, height, false);
+ SetDimensions(width, height, last_dimensions_.is_screencast);
stream_->Input()->SwapFrame(&black_frame);
}
@@ -1685,13 +1651,17 @@
void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodecAndOptions(
const VideoCodecSettings& codec_settings,
const VideoOptions& options) {
- std::vector<webrtc::VideoStream> video_streams =
- encoder_factory_->CreateVideoStreams(
- codec_settings.codec, options, parameters_.config.rtp.ssrcs.size());
- if (video_streams.empty()) {
+ if (last_dimensions_.width == -1) {
+ last_dimensions_.width = codec_settings.codec.width;
+ last_dimensions_.height = codec_settings.codec.height;
+ last_dimensions_.is_screencast = false;
+ }
+ parameters_.encoder_config =
+ CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
+ if (parameters_.encoder_config.streams.empty()) {
return;
}
- parameters_.encoder_config.streams = video_streams;
+
format_ = VideoFormat(codec_settings.codec.width,
codec_settings.codec.height,
VideoFormat::FpsToInterval(30),
@@ -1735,39 +1705,12 @@
RecreateWebRtcStream();
}
-void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions(
- int width,
- int height,
- 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 (!is_screencast) {
- if (codec_settings.codec.width < width)
- width = codec_settings.codec.width;
- if (codec_settings.codec.height < height)
- height = codec_settings.codec.height;
- }
-
- webrtc::VideoEncoderConfig encoder_config = parameters_.encoder_config;
- encoder_config.encoder_specific_settings =
- encoder_factory_->CreateVideoEncoderSettings(codec_settings.codec,
- parameters_.options);
-
- if (is_screencast) {
+webrtc::VideoEncoderConfig
+WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
+ const Dimensions& dimensions,
+ const VideoCodec& codec) const {
+ webrtc::VideoEncoderConfig encoder_config;
+ if (dimensions.is_screencast) {
int screencast_min_bitrate_kbps;
parameters_.options.screencast_min_bitrate.Get(
&screencast_min_bitrate_kbps);
@@ -1779,20 +1722,60 @@
encoder_config.content_type = webrtc::VideoEncoderConfig::kRealtimeVideo;
}
- VideoCodec codec = codec_settings.codec;
- codec.width = width;
- codec.height = height;
+ // Restrict dimensions according to codec max.
+ int width = dimensions.width;
+ int height = dimensions.height;
+ if (!dimensions.is_screencast) {
+ if (codec.width < width)
+ width = codec.width;
+ if (codec.height < height)
+ height = codec.height;
+ }
+
+ VideoCodec clamped_codec = codec;
+ clamped_codec.width = width;
+ clamped_codec.height = height;
encoder_config.streams = encoder_factory_->CreateVideoStreams(
- codec, parameters_.options, parameters_.config.rtp.ssrcs.size());
+ clamped_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) {
+ dimensions.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);
}
+ return encoder_config;
+}
+
+void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions(
+ int width,
+ int height,
+ 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;
+ }
+ LOG(LS_INFO) << "SetDimensions: " << width << "x" << height
+ << (is_screencast ? " (screencast)" : " (not screencast)");
+
+ last_dimensions_.width = width;
+ last_dimensions_.height = height;
+ last_dimensions_.is_screencast = is_screencast;
+
+ assert(!parameters_.encoder_config.streams.empty());
+
+ VideoCodecSettings codec_settings;
+ parameters_.codec_settings.Get(&codec_settings);
+
+ webrtc::VideoEncoderConfig encoder_config =
+ CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
+
+ encoder_config.encoder_specific_settings =
+ encoder_factory_->CreateVideoEncoderSettings(codec_settings.codec,
+ parameters_.options);
bool stream_reconfigured = stream_->ReconfigureVideoEncoder(encoder_config);
@@ -1842,12 +1825,12 @@
info.framerate_input = stats.input_frame_rate;
info.framerate_sent = stats.encode_frame_rate;
- for (std::map<uint32_t, webrtc::StreamStats>::iterator it =
+ for (std::map<uint32_t, webrtc::SsrcStats>::iterator it =
stats.substreams.begin();
it != stats.substreams.end();
++it) {
// TODO(pbos): Wire up additional stats, such as padding bytes.
- webrtc::StreamStats stream_stats = it->second;
+ webrtc::SsrcStats stream_stats = it->second;
info.bytes_sent += stream_stats.rtp_stats.bytes +
stream_stats.rtp_stats.header_bytes +
stream_stats.rtp_stats.padding_bytes;
@@ -1857,7 +1840,7 @@
if (!stats.substreams.empty()) {
// TODO(pbos): Report fraction lost per SSRC.
- webrtc::StreamStats first_stream_stats = stats.substreams.begin()->second;
+ webrtc::SsrcStats first_stream_stats = stats.substreams.begin()->second;
info.fraction_lost =
static_cast<float>(first_stream_stats.rtcp_stats.fraction_lost) /
(1 << 8);
@@ -1884,6 +1867,23 @@
return info;
}
+void WebRtcVideoChannel2::WebRtcVideoSendStream::FillBandwidthEstimationInfo(
+ BandwidthEstimationInfo* bwe_info) {
+ rtc::CritScope cs(&lock_);
+ if (stream_ == NULL) {
+ return;
+ }
+ webrtc::VideoSendStream::Stats stats = stream_->GetStats();
+ for (std::map<uint32_t, webrtc::SsrcStats>::iterator it =
+ stats.substreams.begin();
+ it != stats.substreams.end();
+ ++it) {
+ bwe_info->transmit_bitrate += it->second.total_bitrate_bps;
+ bwe_info->retransmit_bitrate += it->second.retransmit_bitrate_bps;
+ }
+ bwe_info->actual_enc_bitrate = stats.media_bitrate_bps;
+}
+
void WebRtcVideoChannel2::WebRtcVideoSendStream::OnCpuResolutionRequest(
CoordinatedVideoAdapter::AdaptRequest adapt_request) {
rtc::CritScope cs(&lock_);
@@ -2106,6 +2106,14 @@
WebRtcVideoChannel2::VideoCodecSettings::VideoCodecSettings()
: rtx_payload_type(-1) {}
+bool WebRtcVideoChannel2::VideoCodecSettings::operator==(
+ const WebRtcVideoChannel2::VideoCodecSettings& other) const {
+ return codec == other.codec &&
+ fec.ulpfec_payload_type == other.fec.ulpfec_payload_type &&
+ fec.red_payload_type == other.fec.red_payload_type &&
+ rtx_payload_type == other.rtx_payload_type;
+}
+
std::vector<WebRtcVideoChannel2::VideoCodecSettings>
WebRtcVideoChannel2::MapCodecs(const std::vector<VideoCodec>& codecs) {
assert(!codecs.empty());
diff --git a/media/webrtc/webrtcvideoengine2.h b/media/webrtc/webrtcvideoengine2.h
index 0b812ef..9a5fe65 100644
--- a/media/webrtc/webrtcvideoengine2.h
+++ b/media/webrtc/webrtcvideoengine2.h
@@ -145,7 +145,6 @@
int GetCapabilities();
bool SetDefaultEncoderConfig(const VideoEncoderConfig& config);
- VideoEncoderConfig GetDefaultEncoderConfig() const;
WebRtcVideoChannel2* CreateChannel(const VideoOptions& options,
VoiceMediaChannel* voice_channel);
@@ -281,6 +280,8 @@
struct VideoCodecSettings {
VideoCodecSettings();
+ bool operator ==(const VideoCodecSettings& other) const;
+
VideoCodec codec;
webrtc::FecConfig fec;
int rtx_payload_type;
@@ -315,6 +316,7 @@
void Stop();
VideoSenderInfo GetVideoSenderInfo();
+ void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info);
void OnCpuResolutionRequest(
CoordinatedVideoAdapter::AdaptRequest adapt_request);
@@ -348,8 +350,8 @@
bool external;
};
- struct LastDimensions {
- LastDimensions() : width(-1), height(-1), is_screencast(false) {}
+ struct Dimensions {
+ Dimensions() : width(-1), height(-1), is_screencast(false) {}
int width;
int height;
bool is_screencast;
@@ -357,23 +359,28 @@
AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec)
EXCLUSIVE_LOCKS_REQUIRED(lock_);
- void DestroyVideoEncoder(AllocatedEncoder* encoder);
+ void DestroyVideoEncoder(AllocatedEncoder* encoder)
+ EXCLUSIVE_LOCKS_REQUIRED(lock_);
void SetCodecAndOptions(const VideoCodecSettings& codec,
const VideoOptions& options)
EXCLUSIVE_LOCKS_REQUIRED(lock_);
void RecreateWebRtcStream() EXCLUSIVE_LOCKS_REQUIRED(lock_);
+ webrtc::VideoEncoderConfig CreateVideoEncoderConfig(
+ const Dimensions& dimensions,
+ const VideoCodec& codec) const EXCLUSIVE_LOCKS_REQUIRED(lock_);
void SetDimensions(int width, int height, bool is_screencast)
EXCLUSIVE_LOCKS_REQUIRED(lock_);
webrtc::Call* const call_;
- WebRtcVideoEncoderFactory* const external_encoder_factory_;
- WebRtcVideoEncoderFactory2* const encoder_factory_;
+ WebRtcVideoEncoderFactory* const external_encoder_factory_
+ GUARDED_BY(lock_);
+ WebRtcVideoEncoderFactory2* const encoder_factory_ GUARDED_BY(lock_);
rtc::CriticalSection lock_;
webrtc::VideoSendStream* stream_ GUARDED_BY(lock_);
VideoSendStreamParameters parameters_ GUARDED_BY(lock_);
AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_);
- LastDimensions last_dimensions_ GUARDED_BY(lock_);
+ Dimensions last_dimensions_ GUARDED_BY(lock_);
VideoCapturer* capturer_ GUARDED_BY(lock_);
bool sending_ GUARDED_BY(lock_);
diff --git a/media/webrtc/webrtcvideoengine2_unittest.cc b/media/webrtc/webrtcvideoengine2_unittest.cc
index 0b85723..afea370 100644
--- a/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -294,12 +294,9 @@
return NULL;
}
-uint32_t FakeCall::SendBitrateEstimate() {
- return 0;
-}
-
-uint32_t FakeCall::ReceiveBitrateEstimate() {
- return 0;
+webrtc::Call::Stats FakeCall::GetStats() const {
+ webrtc::Call::Stats stats;
+ return stats;
}
void FakeCall::SignalNetworkState(webrtc::Call::NetworkState state) {
diff --git a/media/webrtc/webrtcvideoengine2_unittest.h b/media/webrtc/webrtcvideoengine2_unittest.h
index 3b62289..48c4f64 100644
--- a/media/webrtc/webrtcvideoengine2_unittest.h
+++ b/media/webrtc/webrtcvideoengine2_unittest.h
@@ -127,8 +127,7 @@
webrtc::VideoReceiveStream* receive_stream) OVERRIDE;
virtual webrtc::PacketReceiver* Receiver() OVERRIDE;
- virtual uint32_t SendBitrateEstimate() OVERRIDE;
- virtual uint32_t ReceiveBitrateEstimate() OVERRIDE;
+ virtual webrtc::Call::Stats GetStats() const OVERRIDE;
virtual void SignalNetworkState(webrtc::Call::NetworkState state) OVERRIDE;
diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc
index 95e16e4..74b3163 100644
--- a/media/webrtc/webrtcvoiceengine.cc
+++ b/media/webrtc/webrtcvoiceengine.cc
@@ -74,7 +74,7 @@
{ "ISAC", 32000, 1, 104, true },
{ "CELT", 32000, 1, 109, true },
{ "CELT", 32000, 2, 110, true },
- { "G722", 16000, 1, 9, false },
+ { "G722", 8000, 1, 9, false },
{ "ILBC", 8000, 1, 102, false },
{ "PCMU", 8000, 1, 0, false },
{ "PCMA", 8000, 1, 8, false },
@@ -110,6 +110,7 @@
static const char kIsacCodecName[] = "ISAC";
static const char kL16CodecName[] = "L16";
+static const char kG722CodecName[] = "G722";
// Parameter used for NACK.
// This value is equivalent to 5 seconds of audio data at 20 ms per packet.
@@ -485,12 +486,24 @@
voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate);
}
+// Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
+// which says that G722 should be advertised as 8 kHz although it is a 16 kHz
+// codec.
+static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
+ if (_stricmp(voe_codec->plname, kG722CodecName) == 0) {
+ // If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
+ // has changed, and this special case is no longer needed.
+ ASSERT(voe_codec->plfreq != new_plfreq);
+ voe_codec->plfreq = new_plfreq;
+ }
+}
+
void WebRtcVoiceEngine::ConstructCodecs() {
LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
for (int i = 0; i < ncodecs; ++i) {
webrtc::CodecInst voe_codec;
- if (voe_wrapper_->codec()->GetCodec(i, voe_codec) != -1) {
+ if (GetVoeCodec(i, voe_codec)) {
// Skip uncompressed formats.
if (_stricmp(voe_codec.plname, kL16CodecName) == 0) {
continue;
@@ -540,6 +553,15 @@
std::sort(codecs_.begin(), codecs_.end(), &AudioCodec::Preferable);
}
+bool WebRtcVoiceEngine::GetVoeCodec(int index, webrtc::CodecInst& codec) {
+ if (voe_wrapper_->codec()->GetCodec(index, codec) != -1) {
+ // Change the sample rate of G722 to 8000 to match SDP.
+ MaybeFixupG722(&codec, 8000);
+ return true;
+ }
+ return false;
+}
+
WebRtcVoiceEngine::~WebRtcVoiceEngine() {
LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
if (voe_wrapper_->base()->DeRegisterVoiceEngineObserver() == -1) {
@@ -1224,7 +1246,7 @@
int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
for (int i = 0; i < ncodecs; ++i) {
webrtc::CodecInst voe_codec;
- if (voe_wrapper_->codec()->GetCodec(i, voe_codec) != -1) {
+ if (GetVoeCodec(i, voe_codec)) {
AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq,
voe_codec.rate, voe_codec.channels, 0);
bool multi_rate = IsCodecMultiRate(voe_codec);
@@ -1243,6 +1265,9 @@
voe_codec.rate = in.bitrate;
}
+ // Reset G722 sample rate to 16000 to match WebRTC.
+ MaybeFixupG722(&voe_codec, 16000);
+
// Apply codec-specific settings.
if (IsIsac(codec)) {
// If ISAC and an explicit bitrate is not specified,
diff --git a/media/webrtc/webrtcvoiceengine.h b/media/webrtc/webrtcvoiceengine.h
index f19059b..34b9f3c 100644
--- a/media/webrtc/webrtcvoiceengine.h
+++ b/media/webrtc/webrtcvoiceengine.h
@@ -199,6 +199,7 @@
void Construct();
void ConstructCodecs();
+ bool GetVoeCodec(int index, webrtc::CodecInst& codec);
bool InitInternal();
bool EnsureSoundclipEngineInit();
void SetTraceFilter(int filter);
diff --git a/media/webrtc/webrtcvoiceengine_unittest.cc b/media/webrtc/webrtcvoiceengine_unittest.cc
index 5deabd2..5eb6e24 100644
--- a/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -52,14 +52,16 @@
static const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1, 0);
static const cricket::AudioCodec kCeltCodec(110, "CELT", 32000, 64000, 2, 0);
static const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 64000, 2, 0);
+static const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1, 0);
+static const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1, 0);
static const cricket::AudioCodec kRedCodec(117, "red", 8000, 0, 1, 0);
static const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1, 0);
static const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1, 0);
static const cricket::AudioCodec
kTelephoneEventCodec(106, "telephone-event", 8000, 0, 1, 0);
static const cricket::AudioCodec* const kAudioCodecs[] = {
- &kPcmuCodec, &kIsacCodec, &kCeltCodec, &kOpusCodec, &kRedCodec,
- &kCn8000Codec, &kCn16000Codec, &kTelephoneEventCodec,
+ &kPcmuCodec, &kIsacCodec, &kCeltCodec, &kOpusCodec, &kG722CodecVoE,
+ &kRedCodec, &kCn8000Codec, &kCn16000Codec, &kTelephoneEventCodec,
};
const char kRingbackTone[] = "RIFF____WAVE____ABCD1234";
static uint32 kSsrc1 = 0x99;
@@ -770,6 +772,20 @@
EXPECT_EQ(1, voe_.GetNumSetSendCodecs());
}
+// Verify that G722 is set with 16000 samples per second to WebRTC.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecG722) {
+ EXPECT_TRUE(SetupEngine());
+ int channel_num = voe_.GetLastChannel();
+ std::vector<cricket::AudioCodec> codecs;
+ codecs.push_back(kG722CodecSdp);
+ EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+ webrtc::CodecInst gcodec;
+ EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+ EXPECT_STREQ("G722", gcodec.plname);
+ EXPECT_EQ(1, gcodec.channels);
+ EXPECT_EQ(16000, gcodec.plfreq);
+}
+
// Test that if clockrate is not 48000 for opus, we fail.
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) {
EXPECT_TRUE(SetupEngine());
@@ -3208,7 +3224,7 @@
EXPECT_TRUE(engine.FindCodec(
cricket::AudioCodec(96, "PCMA", 8000, 0, 1, 0)));
EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "G722", 16000, 0, 1, 0)));
+ cricket::AudioCodec(96, "G722", 8000, 0, 1, 0)));
EXPECT_TRUE(engine.FindCodec(
cricket::AudioCodec(96, "red", 8000, 0, 1, 0)));
EXPECT_TRUE(engine.FindCodec(
@@ -3225,7 +3241,7 @@
EXPECT_TRUE(engine.FindCodec(
cricket::AudioCodec(8, "", 8000, 0, 1, 0))); // PCMA
EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(9, "", 16000, 0, 1, 0))); // G722
+ cricket::AudioCodec(9, "", 8000, 0, 1, 0))); // G722
EXPECT_TRUE(engine.FindCodec(
cricket::AudioCodec(13, "", 8000, 0, 1, 0))); // CN
// Check sample/bitrate matching.
@@ -3248,7 +3264,7 @@
EXPECT_EQ(103, it->id);
} else if (it->name == "ISAC" && it->clockrate == 32000) {
EXPECT_EQ(104, it->id);
- } else if (it->name == "G722" && it->clockrate == 16000) {
+ } else if (it->name == "G722" && it->clockrate == 8000) {
EXPECT_EQ(9, it->id);
} else if (it->name == "telephone-event") {
EXPECT_EQ(126, it->id);
diff --git a/session/media/mediasessionclient_unittest.cc b/session/media/mediasessionclient_unittest.cc
index 3e8a90f..eb052ef 100644
--- a/session/media/mediasessionclient_unittest.cc
+++ b/session/media/mediasessionclient_unittest.cc
@@ -61,7 +61,7 @@
cricket::AudioCodec(119, "ISACLC", 16000, 40000, 1, 16),
cricket::AudioCodec(99, "speex", 16000, 22000, 1, 15),
cricket::AudioCodec(97, "IPCMWB", 16000, 80000, 1, 14),
- cricket::AudioCodec(9, "G722", 16000, 64000, 1, 13),
+ cricket::AudioCodec(9, "G722", 8000, 64000, 1, 13),
cricket::AudioCodec(102, "iLBC", 8000, 13300, 1, 12),
cricket::AudioCodec(98, "speex", 8000, 11000, 1, 11),
cricket::AudioCodec(3, "GSM", 8000, 13000, 1, 10),
@@ -81,7 +81,7 @@
static const cricket::AudioCodec kAudioCodecsDifferentPreference[] = {
cricket::AudioCodec(104, "ISAC", 32000, -1, 1, 17),
cricket::AudioCodec(97, "IPCMWB", 16000, 80000, 1, 14),
- cricket::AudioCodec(9, "G722", 16000, 64000, 1, 13),
+ cricket::AudioCodec(9, "G722", 8000, 64000, 1, 13),
cricket::AudioCodec(119, "ISACLC", 16000, 40000, 1, 16),
cricket::AudioCodec(103, "ISAC", 16000, -1, 1, 18),
cricket::AudioCodec(99, "speex", 16000, 22000, 1, 15),
@@ -197,7 +197,7 @@
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='97' name='IPCMWB' clockrate='16000' bitrate='80000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
- " id='9' name='G722' clockrate='16000' bitrate='64000' /> " \
+ " id='9' name='G722' clockrate='8000' bitrate='64000' /> " \
" <payload-type xmlns='http://www.google.com/session/phone' " \
" id='102' name='iLBC' clockrate='8000' bitrate='13300' />" \
" <payload-type xmlns='http://www.google.com/session/phone' " \
@@ -248,7 +248,7 @@
" <parameter name='bitrate' value='80000'/> " \
" </payload-type> " \
" <payload-type " \
- " id='9' name='G722' clockrate='16000'> " \
+ " id='9' name='G722' clockrate='8000'> " \
" <parameter name='bitrate' value='64000'/> " \
" </payload-type> " \
" <payload-type " \
@@ -1918,7 +1918,7 @@
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
codec = AudioCodecFromPayloadType(e);
- VerifyAudioCodec(codec, 9, "G722", 16000, 64000, 1);
+ VerifyAudioCodec(codec, 9, "G722", 8000, 64000, 1);
e = NextFromPayloadType(e);
ASSERT_TRUE(e != NULL);
@@ -2112,7 +2112,7 @@
codec = AudioCodecFromPayloadType(e);
ASSERT_EQ(9, codec.id);
ASSERT_EQ("G722", codec.name);
- ASSERT_EQ(16000, codec.clockrate);
+ ASSERT_EQ(8000, codec.clockrate);
ASSERT_EQ(64000, codec.bitrate);
ASSERT_EQ(1, codec.channels);
