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Unified Diff: content/renderer/media/webrtc/peer_connection_dependency_factory.cc

Issue 1834323002: MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Reworked unit tests around structural changes, and added exhaustive media_stream_audio_unittest.cc. Created 4 years, 8 months ago
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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 03bc115f2dbce4287614c10cf94c9a99f91fd165..cf30f3333eef5fbf5e6cb2256260e7c1f0d97985 100644
--- a/content/renderer/media/webrtc/peer_connection_dependency_factory.cc
+++ b/content/renderer/media/webrtc/peer_connection_dependency_factory.cc
@@ -30,23 +30,15 @@
#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"
@@ -72,7 +64,6 @@
#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)
@@ -130,91 +121,6 @@ PeerConnectionDependencyFactory::CreateRTCPeerConnectionHandler(
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*
PeerConnectionDependencyFactory::CreateVideoCapturer(
bool is_screeencast) {
@@ -525,84 +431,6 @@ PeerConnectionDependencyFactory::CreateLocalAudioSource(
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,
@@ -741,23 +569,6 @@ void PeerConnectionDependencyFactory::CleanupPeerConnectionFactory() {
}
}
-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();

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