| /external/chromium_org/content/renderer/media/ |
| media_stream_track_extra_data.cc | 13 : track_(track),
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| media_stream_track_extra_data.h | 28 return track_;
33 scoped_refptr<webrtc::MediaStreamTrackInterface> track_; member in class:content::MediaStreamTrackExtraData
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| webaudio_capturer_source.cc | 24 : track_(NULL),
67 track_ = track;
74 track_ = NULL;
82 if (!track_)
87 track_->OnSetFormat(params_);
117 track_->Capture(capture_bus_.get(), delay.InMilliseconds(),
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| webrtc_audio_capturer_unittest.cc | 112 track_ = WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
115 track_->audio_source_provider())->SetSinkParamsForTesting(params_);
116 track_->Start();
117 EXPECT_TRUE(track_->enabled());
123 scoped_refptr<WebRtcLocalAudioTrack> track_; member in class:content::WebRtcAudioCapturerTest
132 track_->AddSink(sink.get());
169 track_->RemoveSink(sink.get());
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| webaudio_capturer_source.h | 63 WebRtcLocalAudioTrack* track_; member in class:content::WebAudioCapturerSource
72 // Flag to help notify the |track_| when the audio format has changed.
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| video_destination_handler.cc | 143 : track_(track),
156 scoped_refptr<VideoTrackInterface> track_; member in class:content::PpFrameWriterProxy
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| mock_peer_connection_impl.cc | 163 : track_(track),
185 return track_.get();
197 talk_base::scoped_refptr<AudioTrackInterface> track_; member in class:content::MockDtmfSender
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| webrtc_audio_capturer.cc | 84 TrackWrapper(WebRtcLocalAudioTrack* track) : track_(track) {}
87 return owner->IsEqual(track_);
89 WebRtcLocalAudioTrack* track_; member in struct:content::WebRtcAudioCapturer::TrackOwner::TrackWrapper
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| /external/chromium_org/chrome/renderer/media/ |
| cast_rtp_stream.cc | 114 : track_(track), sink_added_(false) {
119 RemoveFromVideoTrack(this, track_);
138 AddToVideoTrack(this, track_);
143 blink::WebMediaStreamTrack track_; member in class:CastVideoSink
158 : track_(track), sink_added_(false) {
163 RemoveFromAudioTrack(this, track_);
195 AddToAudioTrack(this, track_);
200 blink::WebMediaStreamTrack track_; member in class:CastAudioSink
236 : track_(track),
260 audio_sink_.reset(new CastAudioSink(track_));
[all...] |
| cast_rtp_stream.h | 109 blink::WebMediaStreamTrack track_; member in class:CastRtpStream
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| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| dtmfsender.cc | 94 : track_(track),
100 ASSERT(track_ != NULL);
130 return provider_->CanInsertDtmf(track_->id());
164 return track_;
227 if (!provider_->InsertDtmf(track_->id(), code, duration_)) {
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| dtmfsender_unittest.cc | 141 : track_(AudioTrack::Create(kTestAudioLabel, NULL)),
145 dtmf_ = DtmfSender::Create(track_, talk_base::Thread::Current(),
232 talk_base::scoped_refptr<AudioTrackInterface> track_; member in class:DtmfSenderTest
263 VerifyExpectedState(track_, tones1, duration, inter_tone_gap);
266 VerifyExpectedState(track_, "2", duration, inter_tone_gap);
269 VerifyExpectedState(track_, tones2, duration, inter_tone_gap);
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| dtmfsender.h | 110 talk_base::scoped_refptr<AudioTrackInterface> track_; member in class:webrtc::DtmfSender
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| mediastreamhandler.cc | 37 : track_(track),
41 track_->RegisterObserver(this);
45 track_->UnregisterObserver(this);
49 if (state_ != track_->state()) {
50 state_ = track_->state();
53 if (enabled_ != track_->enabled()) {
54 enabled_ = track_->enabled();
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| mediastreamhandler.h | 53 // Stop using |track_| on this PeerConnection.
56 MediaStreamTrackInterface* track() { return track_; }
64 talk_base::scoped_refptr<MediaStreamTrackInterface> track_; member in class:webrtc::TrackHandler
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| statscollector_unittest.cc | 238 track_= webrtc::VideoTrack::Create(kTrackId, NULL);
239 stream_->AddTrack(track_);
322 talk_base::scoped_refptr<webrtc::VideoTrack> track_; member in class:__anon11833::StatsCollectorTest
501 stats.GetStats(track_, &reports);
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| /external/valgrind/main/coregrind/ |
| pub_core_tooliface.h | 52 if (VG_(tdict).track_##fn) \
53 VG_(tdict).track_##fn(args); \
|
