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 038565d69832429d822a2c051842b02598166a70..68f5b0ab2ce11c87a2ed3130e06278525960bde2 100644 |
--- a/content/renderer/media/webrtc/peer_connection_dependency_factory.cc |
+++ b/content/renderer/media/webrtc/peer_connection_dependency_factory.cc |
@@ -30,15 +30,23 @@ |
#include "content/public/common/webrtc_ip_handling_policy.h" |
#include "content/public/renderer/content_renderer_client.h" |
#include "content/renderer/media/media_stream.h" |
+#include "content/renderer/media/media_stream_audio_processor.h" |
+#include "content/renderer/media/media_stream_audio_processor_options.h" |
+#include "content/renderer/media/media_stream_audio_source.h" |
+#include "content/renderer/media/media_stream_constraints_util.h" |
#include "content/renderer/media/media_stream_video_source.h" |
#include "content/renderer/media/media_stream_video_track.h" |
#include "content/renderer/media/peer_connection_identity_store.h" |
#include "content/renderer/media/rtc_peer_connection_handler.h" |
#include "content/renderer/media/rtc_video_decoder_factory.h" |
#include "content/renderer/media/rtc_video_encoder_factory.h" |
+#include "content/renderer/media/webaudio_capturer_source.h" |
+#include "content/renderer/media/webrtc/media_stream_remote_audio_track.h" |
#include "content/renderer/media/webrtc/stun_field_trial.h" |
+#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
#include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h" |
#include "content/renderer/media/webrtc_audio_device_impl.h" |
+#include "content/renderer/media/webrtc_local_audio_track.h" |
#include "content/renderer/media/webrtc_logging.h" |
#include "content/renderer/media/webrtc_uma_histograms.h" |
#include "content/renderer/p2p/empty_network_manager.h" |
@@ -64,6 +72,7 @@ |
#include "third_party/webrtc/api/dtlsidentitystore.h" |
#include "third_party/webrtc/api/mediaconstraintsinterface.h" |
#include "third_party/webrtc/base/ssladapter.h" |
+#include "third_party/webrtc/media/base/mediachannel.h" |
#include "third_party/webrtc/modules/video_coding/codecs/h264/include/h264.h" |
#if defined(OS_ANDROID) |
@@ -119,6 +128,91 @@ |
UpdateWebRTCMethodCount(WEBKIT_RTC_PEER_CONNECTION); |
return new RTCPeerConnectionHandler(client, this); |
+} |
+ |
+bool PeerConnectionDependencyFactory::InitializeMediaStreamAudioSource( |
+ int render_frame_id, |
+ const blink::WebMediaConstraints& audio_constraints, |
+ MediaStreamAudioSource* source_data) { |
+ DVLOG(1) << "InitializeMediaStreamAudioSources()"; |
+ |
+ // Do additional source initialization if the audio source is a valid |
+ // microphone or tab audio. |
+ |
+ StreamDeviceInfo device_info = source_data->device_info(); |
+ |
+ cricket::AudioOptions options; |
+ // Apply relevant constraints. |
+ options.echo_cancellation = ConstraintToOptional( |
+ audio_constraints, &blink::WebMediaTrackConstraintSet::echoCancellation); |
+ options.delay_agnostic_aec = ConstraintToOptional( |
+ audio_constraints, |
+ &blink::WebMediaTrackConstraintSet::googDAEchoCancellation); |
+ options.auto_gain_control = ConstraintToOptional( |
+ audio_constraints, |
+ &blink::WebMediaTrackConstraintSet::googAutoGainControl); |
+ options.experimental_agc = ConstraintToOptional( |
+ audio_constraints, |
+ &blink::WebMediaTrackConstraintSet::googExperimentalAutoGainControl); |
+ options.noise_suppression = ConstraintToOptional( |
+ audio_constraints, |
+ &blink::WebMediaTrackConstraintSet::googNoiseSuppression); |
+ options.experimental_ns = ConstraintToOptional( |
+ audio_constraints, |
+ &blink::WebMediaTrackConstraintSet::googExperimentalNoiseSuppression); |
+ options.highpass_filter = ConstraintToOptional( |
+ audio_constraints, |
+ &blink::WebMediaTrackConstraintSet::googHighpassFilter); |
+ options.typing_detection = ConstraintToOptional( |
+ audio_constraints, |
+ &blink::WebMediaTrackConstraintSet::googTypingNoiseDetection); |
+ options.stereo_swapping = ConstraintToOptional( |
+ audio_constraints, |
+ &blink::WebMediaTrackConstraintSet::googAudioMirroring); |
+ |
+ MediaAudioConstraints::ApplyFixedAudioConstraints(&options); |
+ |
+ if (device_info.device.input.effects & |
+ media::AudioParameters::ECHO_CANCELLER) { |
+ // TODO(hta): Figure out if we should be looking at echoCancellation. |
+ // Previous code had googEchoCancellation only. |
+ const blink::BooleanConstraint& echoCancellation = |
+ audio_constraints.basic().googEchoCancellation; |
+ if (echoCancellation.hasExact() && !echoCancellation.exact()) { |
+ device_info.device.input.effects &= |
+ ~media::AudioParameters::ECHO_CANCELLER; |
+ } |
+ options.echo_cancellation = rtc::Optional<bool>(false); |
+ } |
+ |
+ std::unique_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"; |
+ WebRtcLogMessage(log_string); |
+ DVLOG(1) << log_string; |
+ // TODO(xians): Don't we need to check if source_observer is observing |
+ // something? If not, then it looks like we have a leak here. |
+ // OTOH, if it _is_ observing something, then the callback might |
+ // be called multiple times which is likely also a bug. |
+ return false; |
+ } |
+ 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. |
+ // TODO(perkj): Move audio constraints parsing to Chrome. |
+ // Currently there are a few constraints that are parsed by libjingle and |
+ // the state is set to ended if parsing fails. |
+ scoped_refptr<webrtc::AudioSourceInterface> rtc_source( |
+ CreateLocalAudioSource(options).get()); |
+ if (rtc_source->state() != webrtc::MediaSourceInterface::kLive) { |
+ DLOG(WARNING) << "Failed to create rtc LocalAudioSource."; |
+ return false; |
+ } |
+ source_data->SetLocalAudioSource(rtc_source.get()); |
+ return true; |
} |
WebRtcVideoCapturerAdapter* |
@@ -431,6 +525,84 @@ |
return source; |
} |
+void PeerConnectionDependencyFactory::CreateLocalAudioTrack( |
+ const blink::WebMediaStreamTrack& track) { |
+ blink::WebMediaStreamSource source = track.source(); |
+ DCHECK_EQ(source.getType(), blink::WebMediaStreamSource::TypeAudio); |
+ MediaStreamAudioSource* source_data = MediaStreamAudioSource::From(source); |
+ |
+ if (!source_data) { |
+ if (source.requiresAudioConsumer()) { |
+ // We're adding a WebAudio MediaStream. |
+ // Create a specific capturer for each WebAudio consumer. |
+ CreateWebAudioSource(&source); |
+ source_data = MediaStreamAudioSource::From(source); |
+ DCHECK(source_data->webaudio_capturer()); |
+ } else { |
+ NOTREACHED() << "Local track missing MediaStreamAudioSource instance."; |
+ return; |
+ } |
+ } |
+ |
+ // Creates an adapter to hold all the libjingle objects. |
+ scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
+ WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(), |
+ source_data->local_audio_source())); |
+ static_cast<webrtc::AudioTrackInterface*>(adapter.get())->set_enabled( |
+ track.isEnabled()); |
+ |
+ // 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. |
+ std::unique_ptr<WebRtcLocalAudioTrack> audio_track( |
+ new WebRtcLocalAudioTrack(adapter.get())); |
+ |
+ // Start the source and connect the audio data flow to the track. |
+ // |
+ // TODO(miu): This logic will me moved to MediaStreamAudioSource (or a |
+ // subclass of it) in soon-upcoming changes. |
+ audio_track->Start(base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, |
+ source_data->GetWeakPtr(), |
+ audio_track.get())); |
+ if (source_data->webaudio_capturer()) |
+ source_data->webaudio_capturer()->Start(audio_track.get()); |
+ else if (source_data->audio_capturer()) |
+ source_data->audio_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; |
+ writable_track.setExtraData(audio_track.release()); |
+} |
+ |
+void PeerConnectionDependencyFactory::CreateRemoteAudioTrack( |
+ const blink::WebMediaStreamTrack& track) { |
+ blink::WebMediaStreamSource source = track.source(); |
+ DCHECK_EQ(source.getType(), blink::WebMediaStreamSource::TypeAudio); |
+ DCHECK(source.remote()); |
+ DCHECK(MediaStreamAudioSource::From(source)); |
+ |
+ blink::WebMediaStreamTrack writable_track = track; |
+ writable_track.setExtraData( |
+ new MediaStreamRemoteAudioTrack(source, track.isEnabled())); |
+} |
+ |
+void PeerConnectionDependencyFactory::CreateWebAudioSource( |
+ blink::WebMediaStreamSource* source) { |
+ DVLOG(1) << "PeerConnectionDependencyFactory::CreateWebAudioSource()"; |
+ |
+ MediaStreamAudioSource* source_data = new MediaStreamAudioSource(); |
+ source_data->SetWebAudioCapturer( |
+ base::WrapUnique(new WebAudioCapturerSource(source))); |
+ |
+ // Create a LocalAudioSource object which holds audio options. |
+ // SetLocalAudioSource() affects core audio parts in third_party/Libjingle. |
+ cricket::AudioOptions options; |
+ source_data->SetLocalAudioSource(CreateLocalAudioSource(options).get()); |
+ source->setExtraData(source_data); |
+} |
+ |
scoped_refptr<webrtc::VideoTrackInterface> |
PeerConnectionDependencyFactory::CreateLocalVideoTrack( |
const std::string& id, |
@@ -569,6 +741,23 @@ |
} |
} |
+std::unique_ptr<WebRtcAudioCapturer> |
+PeerConnectionDependencyFactory::CreateAudioCapturer( |
+ int render_frame_id, |
+ const StreamDeviceInfo& device_info, |
+ const blink::WebMediaConstraints& constraints, |
+ MediaStreamAudioSource* audio_source) { |
+ // TODO(xians): Handle the cases when gUM is called without a proper render |
+ // view, for example, by an extension. |
+ DCHECK_GE(render_frame_id, 0); |
+ |
+ EnsureWebRtcAudioDeviceImpl(); |
+ DCHECK(GetWebRtcAudioDevice()); |
+ return WebRtcAudioCapturer::CreateCapturer( |
+ render_frame_id, device_info, constraints, GetWebRtcAudioDevice(), |
+ audio_source); |
+} |
+ |
void PeerConnectionDependencyFactory::EnsureInitialized() { |
DCHECK(CalledOnValidThread()); |
GetPcFactory(); |