| Index: content/renderer/media/webrtc/peer_connection_dependency_factory.cc
|
| diff --git a/content/renderer/media/webrtc/peer_connection_dependency_factory.cc b/content/renderer/media/webrtc/peer_connection_dependency_factory.cc
|
| index dbbe3b6bfa11a19ec5b7bd0924ea5dd00043bc89..65c438792e6960c15a076452081922e65d26dbab 100644
|
| --- a/content/renderer/media/webrtc/peer_connection_dependency_factory.cc
|
| +++ b/content/renderer/media/webrtc/peer_connection_dependency_factory.cc
|
| @@ -9,6 +9,8 @@
|
| #include <utility>
|
| #include <vector>
|
|
|
| +#include "base/bind.h"
|
| +#include "base/bind_helpers.h"
|
| #include "base/command_line.h"
|
| #include "base/location.h"
|
| #include "base/logging.h"
|
| @@ -188,8 +190,8 @@ bool PeerConnectionDependencyFactory::InitializeMediaStreamAudioSource(
|
| HarmonizeConstraintsAndEffects(&constraints,
|
| &device_info.device.input.effects);
|
|
|
| - scoped_refptr<WebRtcAudioCapturer> capturer(CreateAudioCapturer(
|
| - render_frame_id, device_info, audio_constraints, source_data));
|
| + scoped_ptr<WebRtcAudioCapturer> capturer = CreateAudioCapturer(
|
| + render_frame_id, device_info, audio_constraints, source_data);
|
| if (!capturer.get()) {
|
| const std::string log_string =
|
| "PCDF::InitializeMediaStreamAudioSource: fails to create capturer";
|
| @@ -201,7 +203,7 @@ bool PeerConnectionDependencyFactory::InitializeMediaStreamAudioSource(
|
| // be called multiple times which is likely also a bug.
|
| return false;
|
| }
|
| - source_data->SetAudioCapturer(capturer.get());
|
| + source_data->SetAudioCapturer(std::move(capturer));
|
|
|
| // Creates a LocalAudioSource object which holds audio options.
|
| // TODO(xians): The option should apply to the track instead of the source.
|
| @@ -534,20 +536,17 @@ void PeerConnectionDependencyFactory::CreateLocalAudioTrack(
|
| const blink::WebMediaStreamTrack& track) {
|
| blink::WebMediaStreamSource source = track.source();
|
| DCHECK_EQ(source.type(), blink::WebMediaStreamSource::TypeAudio);
|
| - DCHECK(!source.remote());
|
| - MediaStreamAudioSource* source_data =
|
| - static_cast<MediaStreamAudioSource*>(source.extraData());
|
| + MediaStreamAudioSource* source_data = MediaStreamAudioSource::From(source);
|
|
|
| - scoped_refptr<WebAudioCapturerSource> webaudio_source;
|
| + scoped_ptr<WebAudioCapturerSource> webaudio_source;
|
| if (!source_data) {
|
| if (source.requiresAudioConsumer()) {
|
| // We're adding a WebAudio MediaStream.
|
| // Create a specific capturer for each WebAudio consumer.
|
| webaudio_source = CreateWebAudioSource(&source);
|
| - source_data =
|
| - static_cast<MediaStreamAudioSource*>(source.extraData());
|
| + source_data = MediaStreamAudioSource::From(source);
|
| } else {
|
| - NOTREACHED() << "Local track missing source extra data.";
|
| + NOTREACHED() << "Local track missing MediaStreamAudioSource instance.";
|
| return;
|
| }
|
| }
|
| @@ -562,10 +561,25 @@ void PeerConnectionDependencyFactory::CreateLocalAudioTrack(
|
| // TODO(xians): Merge |source| to the capturer(). We can't do this today
|
| // because only one capturer() is supported while one |source| is created
|
| // for each audio track.
|
| - scoped_ptr<WebRtcLocalAudioTrack> audio_track(new WebRtcLocalAudioTrack(
|
| - adapter.get(), source_data->GetAudioCapturer(), webaudio_source.get()));
|
| -
|
| - StartLocalAudioTrack(audio_track.get());
|
| + scoped_ptr<WebRtcLocalAudioTrack> audio_track(
|
| + new WebRtcLocalAudioTrack(adapter.get()));
|
| +
|
| + // Start the source and connect the audio data flow to the track.
|
| + if (webaudio_source.get()) {
|
| + webaudio_source->Start(audio_track.get());
|
| + // The stop callback takes ownership of the |webaudio_source|, which will
|
| + // cause it to be auto-destroyed when the track is stopped.
|
| + //
|
| + // TODO(miu): In a future change, WebAudioCapturerSource will become a
|
| + // subclass of MediaStreamAudioSource, and this will allow it to be owned by
|
| + // the blink::WebMediaStreamSource so we don't need this hacky thing here.
|
| + audio_track->AddStopObserver(base::Bind(
|
| + &WebAudioCapturerSource::Stop, base::Owned(webaudio_source.release())));
|
| + } else if (WebRtcAudioCapturer* capturer = source_data->audio_capturer()) {
|
| + capturer->AddTrack(audio_track.get());
|
| + } else {
|
| + NOTREACHED();
|
| + }
|
|
|
| // Pass the ownership of the native local audio track to the blink track.
|
| blink::WebMediaStreamTrack writable_track = track;
|
| @@ -577,48 +591,26 @@ void PeerConnectionDependencyFactory::CreateRemoteAudioTrack(
|
| blink::WebMediaStreamSource source = track.source();
|
| DCHECK_EQ(source.type(), blink::WebMediaStreamSource::TypeAudio);
|
| DCHECK(source.remote());
|
| - DCHECK(source.extraData());
|
| + DCHECK(MediaStreamAudioSource::From(source));
|
|
|
| blink::WebMediaStreamTrack writable_track = track;
|
| writable_track.setExtraData(
|
| new MediaStreamRemoteAudioTrack(source, track.isEnabled()));
|
| }
|
|
|
| -void PeerConnectionDependencyFactory::StartLocalAudioTrack(
|
| - WebRtcLocalAudioTrack* audio_track) {
|
| - // Start the audio track. This will hook the |audio_track| to the capturer
|
| - // as the sink of the audio, and only start the source of the capturer if
|
| - // it is the first audio track connecting to the capturer.
|
| - audio_track->Start();
|
| -}
|
| -
|
| -scoped_refptr<WebAudioCapturerSource>
|
| +scoped_ptr<WebAudioCapturerSource>
|
| PeerConnectionDependencyFactory::CreateWebAudioSource(
|
| blink::WebMediaStreamSource* source) {
|
| DVLOG(1) << "PeerConnectionDependencyFactory::CreateWebAudioSource()";
|
|
|
| - scoped_refptr<WebAudioCapturerSource>
|
| - webaudio_capturer_source(new WebAudioCapturerSource(*source));
|
| MediaStreamAudioSource* source_data = new MediaStreamAudioSource();
|
|
|
| - // Use the current default capturer for the WebAudio track so that the
|
| - // WebAudio track can pass a valid delay value and |need_audio_processing|
|
| - // flag to PeerConnection.
|
| - // TODO(xians): Remove this after moving APM to Chrome.
|
| - if (GetWebRtcAudioDevice()) {
|
| - source_data->SetAudioCapturer(
|
| - GetWebRtcAudioDevice()->GetDefaultCapturer());
|
| - }
|
| -
|
| // Create a LocalAudioSource object which holds audio options.
|
| // SetLocalAudioSource() affects core audio parts in third_party/Libjingle.
|
| source_data->SetLocalAudioSource(CreateLocalAudioSource(NULL).get());
|
| source->setExtraData(source_data);
|
|
|
| - // Replace the default source with WebAudio as source instead.
|
| - source->addAudioConsumer(webaudio_capturer_source.get());
|
| -
|
| - return webaudio_capturer_source;
|
| + return make_scoped_ptr(new WebAudioCapturerSource(source));
|
| }
|
|
|
| scoped_refptr<webrtc::VideoTrackInterface>
|
| @@ -759,7 +751,7 @@ void PeerConnectionDependencyFactory::CleanupPeerConnectionFactory() {
|
| }
|
| }
|
|
|
| -scoped_refptr<WebRtcAudioCapturer>
|
| +scoped_ptr<WebRtcAudioCapturer>
|
| PeerConnectionDependencyFactory::CreateAudioCapturer(
|
| int render_frame_id,
|
| const StreamDeviceInfo& device_info,
|
|
|