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 966a295fdf8fdd3fa61f66ff58cc7a996eb21dc2..c50c9cdd266e2b31184cc622ac66605214bb5fad 100644 |
--- a/content/renderer/media/webrtc/peer_connection_dependency_factory.cc |
+++ b/content/renderer/media/webrtc/peer_connection_dependency_factory.cc |
@@ -26,8 +26,6 @@ |
#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_video_source.h" |
#include "content/renderer/media/media_stream_video_track.h" |
#include "content/renderer/media/peer_connection_identity_store.h" |
@@ -35,14 +33,9 @@ |
#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" |
#include "content/renderer/p2p/filtering_network_manager.h" |
@@ -97,52 +90,6 @@ WebRTCIPHandlingPolicy GetWebRTCIPHandlingPolicy( |
} // namespace |
-// Map of corresponding media constraints and platform effects. |
-struct { |
- const char* constraint; |
- const media::AudioParameters::PlatformEffectsMask effect; |
-} const kConstraintEffectMap[] = { |
- { webrtc::MediaConstraintsInterface::kGoogEchoCancellation, |
- media::AudioParameters::ECHO_CANCELLER }, |
-}; |
- |
-// If any platform effects are available, check them against the constraints. |
-// Disable effects to match false constraints, but if a constraint is true, set |
-// the constraint to false to later disable the software effect. |
-// |
-// This function may modify both |constraints| and |effects|. |
-void HarmonizeConstraintsAndEffects(RTCMediaConstraints* constraints, |
- int* effects) { |
- if (*effects != media::AudioParameters::NO_EFFECTS) { |
- for (size_t i = 0; i < arraysize(kConstraintEffectMap); ++i) { |
- bool value; |
- size_t is_mandatory = 0; |
- if (!webrtc::FindConstraint(constraints, |
- kConstraintEffectMap[i].constraint, |
- &value, |
- &is_mandatory) || !value) { |
- // If the constraint is false, or does not exist, disable the platform |
- // effect. |
- *effects &= ~kConstraintEffectMap[i].effect; |
- DVLOG(1) << "Disabling platform effect: " |
- << kConstraintEffectMap[i].effect; |
- } else if (*effects & kConstraintEffectMap[i].effect) { |
- // If the constraint is true, leave the platform effect enabled, and |
- // set the constraint to false to later disable the software effect. |
- if (is_mandatory) { |
- constraints->AddMandatory(kConstraintEffectMap[i].constraint, |
- webrtc::MediaConstraintsInterface::kValueFalse, true); |
- } else { |
- constraints->AddOptional(kConstraintEffectMap[i].constraint, |
- webrtc::MediaConstraintsInterface::kValueFalse, true); |
- } |
- DVLOG(1) << "Disabling constraint: " |
- << kConstraintEffectMap[i].constraint; |
- } |
- } |
- } |
-} |
- |
PeerConnectionDependencyFactory::PeerConnectionDependencyFactory( |
P2PSocketDispatcher* p2p_socket_dispatcher) |
: network_manager_(NULL), |
@@ -170,53 +117,6 @@ 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. |
- RTCMediaConstraints native_audio_constraints(audio_constraints); |
- MediaAudioConstraints::ApplyFixedAudioConstraints(&native_audio_constraints); |
- |
- StreamDeviceInfo device_info = source_data->device_info(); |
- RTCMediaConstraints constraints = native_audio_constraints; |
- // May modify both |constraints| and |effects|. |
- HarmonizeConstraintsAndEffects(&constraints, |
- &device_info.device.input.effects); |
- |
- scoped_refptr<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(capturer.get()); |
- |
- // 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(&constraints).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) { |
@@ -533,97 +433,6 @@ PeerConnectionDependencyFactory::CreateLocalAudioSource( |
return source; |
} |
-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()); |
- |
- scoped_refptr<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()); |
- } else { |
- NOTREACHED() << "Local track missing source extra data."; |
- 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. |
- scoped_ptr<WebRtcLocalAudioTrack> audio_track(new WebRtcLocalAudioTrack( |
- adapter.get(), source_data->GetAudioCapturer(), webaudio_source.get())); |
- |
- StartLocalAudioTrack(audio_track.get()); |
- |
- // 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.type(), blink::WebMediaStreamSource::TypeAudio); |
- DCHECK(source.remote()); |
- DCHECK(source.extraData()); |
- |
- 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> |
-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; |
-} |
- |
scoped_refptr<webrtc::VideoTrackInterface> |
PeerConnectionDependencyFactory::CreateLocalVideoTrack( |
const std::string& id, |
@@ -762,23 +571,6 @@ void PeerConnectionDependencyFactory::CleanupPeerConnectionFactory() { |
} |
} |
-scoped_refptr<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(); |