/external/webrtc/pc/ |
D | media_stream_unittest.cc | 108 scoped_refptr<webrtc::MediaStreamTrackInterface> audio_track( in TEST_F() local 110 EXPECT_EQ(0, audio_track->id().compare(kAudioTrackId)); in TEST_F() 111 EXPECT_TRUE(audio_track->enabled()); in TEST_F() 113 EXPECT_TRUE(stream_->GetAudioTracks()[0].get() == audio_track.get()); in TEST_F() 114 EXPECT_TRUE(stream_->FindAudioTrack(audio_track->id()).get() == in TEST_F() 115 audio_track.get()); in TEST_F() 116 audio_track = stream_->GetAudioTracks()[0]; in TEST_F() 117 EXPECT_EQ(0, audio_track->id().compare(kAudioTrackId)); in TEST_F() 118 EXPECT_TRUE(audio_track->enabled()); in TEST_F() 148 scoped_refptr<webrtc::AudioTrackInterface> audio_track( in TEST_F() local [all …]
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D | legacy_stats_collector_interface.h | 30 virtual void AddLocalAudioTrack(AudioTrackInterface* audio_track, 35 virtual void RemoveLocalAudioTrack(AudioTrackInterface* audio_track,
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D | peer_connection_rtp_unittest.cc | 1032 auto audio_track = caller->CreateAudioTrack("audio track"); in TEST_F() local 1033 auto transceiver = caller->AddTransceiver(audio_track); in TEST_F() 1037 EXPECT_EQ(audio_track, sender->track()); in TEST_F() 1052 auto audio_track = caller->CreateAudioTrack("audio track"); in TEST_F() local 1054 auto transceiver1 = caller->AddTransceiver(audio_track); in TEST_F() 1055 auto transceiver2 = caller->AddTransceiver(audio_track); in TEST_F() 1062 EXPECT_EQ(audio_track, sender1->track()); in TEST_F() 1063 EXPECT_EQ(audio_track, sender2->track()); in TEST_F() 1095 auto audio_track = caller->CreateAudioTrack("a"); in TEST_F() local 1096 auto sender = caller->AddTrack(audio_track); in TEST_F() [all …]
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D | rtp_sender.cc | 669 audio_track()->RemoveSink(sink_adapter_.get()); in DetachTrack() 675 audio_track()->AddSink(sink_adapter_.get()); in AttachTrack() 680 legacy_stats_->AddLocalAudioTrack(audio_track().get(), ssrc_); in AddTrackToStats() 686 legacy_stats_->RemoveLocalAudioTrack(audio_track().get(), ssrc_); in RemoveTrackFromStats() 716 if (track_->enabled() && audio_track()->GetSource() && in SetSend() 717 !audio_track()->GetSource()->remote()) { in SetSend() 718 options = audio_track()->GetSource()->options(); in SetSend()
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D | legacy_stats_collector.h | 71 void AddLocalAudioTrack(AudioTrackInterface* audio_track, 76 void RemoveLocalAudioTrack(AudioTrackInterface* audio_track,
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D | legacy_stats_collector.cc | 577 void LegacyStatsCollector::AddLocalAudioTrack(AudioTrackInterface* audio_track, in AddLocalAudioTrack() argument 580 RTC_DCHECK(audio_track != NULL); in AddLocalAudioTrack() 583 RTC_DCHECK(track.first != audio_track || track.second != ssrc); in AddLocalAudioTrack() 586 local_audio_tracks_.push_back(std::make_pair(audio_track, ssrc)); in AddLocalAudioTrack() 591 audio_track->id())); in AddLocalAudioTrack() 595 report->AddString(StatsReport::kStatsValueNameTrackId, audio_track->id()); in AddLocalAudioTrack() 600 AudioTrackInterface* audio_track, in RemoveLocalAudioTrack() argument 602 RTC_DCHECK(audio_track != NULL); in RemoveLocalAudioTrack() 606 [audio_track, ssrc](const LocalAudioTrackVector::value_type& track) { in RemoveLocalAudioTrack() 607 return track.first == audio_track && track.second == ssrc; in RemoveLocalAudioTrack()
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D | peer_connection_interface_unittest.cc | 544 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( in CreateStreamCollection() local 547 stream->AddTrack(audio_track); in CreateStreamCollection() 1142 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( in AddAudioTrack() local 1144 ASSERT_TRUE(stream->AddTrack(audio_track)); in AddAudioTrack() 1464 rtc::scoped_refptr<AudioTrackInterface> audio_track( in TEST_F() local 1467 stream->AddTrack(audio_track); in TEST_F() 1531 rtc::scoped_refptr<AudioTrackInterface> audio_track( in TEST_F() local 1535 auto audio_sender = pc_->AddTrack(audio_track, {kStreamId1}).MoveValue(); in TEST_F() 1540 EXPECT_EQ(audio_track, audio_sender->track()); in TEST_F() 1588 rtc::scoped_refptr<AudioTrackInterface> audio_track( in TEST_F() local [all …]
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D | rtc_stats_collector.cc | 996 AudioTrackInterface& audio_track, in ProduceMediaStreamTrackStatsFromVoiceSenderInfo() argument 1005 audio_track, audio_track_stats.get()); in ProduceMediaStreamTrackStatsFromVoiceSenderInfo() 1015 auto audio_processor(audio_track.GetAudioProcessor()); in ProduceMediaStreamTrackStatsFromVoiceSenderInfo() 1029 const AudioTrackInterface& audio_track, in ProduceMediaStreamTrackStatsFromVoiceReceiverInfo() argument 1040 audio_track, audio_track_stats.get()); in ProduceMediaStreamTrackStatsFromVoiceReceiverInfo() 1851 AudioTrackInterface* audio_track = in ProduceMediaSourceStats_s() local 1879 auto audio_processor(audio_track->GetAudioProcessor()); in ProduceMediaSourceStats_s() 1983 rtc::scoped_refptr<AudioTrackInterface> audio_track = in ProduceAudioRTPStreamStats_n() local 1985 if (audio_track) { in ProduceAudioRTPStreamStats_n() 1989 .GetAttachmentIdByTrack(audio_track.get()) in ProduceAudioRTPStreamStats_n() [all …]
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D | legacy_stats_collector_unittest.cc | 557 AudioTrackInterface* audio_track, in UpdateVoiceSenderInfoFromAudioTrack() argument 560 audio_track->GetSignalLevel(&voice_sender_info->audio_level); in UpdateVoiceSenderInfoFromAudioTrack() 562 audio_track->GetAudioProcessor()->GetStats(has_remote_tracks); in UpdateVoiceSenderInfoFromAudioTrack() 608 void VerifyAudioTrackStats(FakeAudioTrack* audio_track, in VerifyAudioTrackStats() argument 623 EXPECT_EQ(audio_track->id(), track_id); in VerifyAudioTrackStats() 642 stats->GetStats(audio_track, &track_reports); in VerifyAudioTrackStats() 649 EXPECT_EQ(audio_track->id(), track_id); in VerifyAudioTrackStats() 818 stream_->AddTrack(audio_track()); in AddOutgoingAudioTrack() 833 stream_->AddTrack(audio_track()); in AddIncomingAudioTrack() 841 rtc::scoped_refptr<AudioTrackInterface> audio_track() { return audio_track_; } in audio_track() function in webrtc::StatsCollectorTrackTest [all …]
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D | audio_rtp_receiver.cc | 245 existing_stream->RemoveTrack(audio_track()); in SetStreams() 259 stream->AddTrack(audio_track()); in SetStreams()
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D | rtp_transmission_manager.cc | 560 rtc::scoped_refptr<AudioTrackInterface> audio_track = in OnRemoteSenderRemoved() local 562 if (audio_track) { in OnRemoteSenderRemoved() 563 stream->RemoveTrack(audio_track); in OnRemoteSenderRemoved()
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D | audio_rtp_receiver.h | 73 rtc::scoped_refptr<AudioTrackInterface> audio_track() const { return track_; } in audio_track() function
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D | BUILD.gn | 707 ":audio_track", 1476 "../pc:audio_track", 1674 ":audio_track", 1763 rtc_library("audio_track") { 1766 "audio_track.cc", 1767 "audio_track.h", 2280 ":audio_track", 2499 ":audio_track",
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D | rtp_sender.h | 382 rtc::scoped_refptr<AudioTrackInterface> audio_track() const { in audio_track() function
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D | webrtc_sdp_unittest.cc | 1299 StreamParams audio_track; in MakeUnifiedPlanDescriptionNoSsrcSignaling() local 1300 audio_track.id = kAudioTrackId1; in MakeUnifiedPlanDescriptionNoSsrcSignaling() 1301 audio_track.set_stream_ids({kStreamId1}); in MakeUnifiedPlanDescriptionNoSsrcSignaling() 1302 audio_desc->AddStream(audio_track); in MakeUnifiedPlanDescriptionNoSsrcSignaling()
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/external/libwebm/testing/ |
D | mkvparser_tests.cc | 160 const AudioTrack* const audio_track = in TEST_F() local 162 EXPECT_EQ(kSampleRate, audio_track->GetSamplingRate()); in TEST_F() 163 EXPECT_EQ(kChannels, audio_track->GetChannels()); in TEST_F() 164 EXPECT_EQ(kBitDepth, audio_track->GetBitDepth()); in TEST_F() 165 EXPECT_STREQ(kVorbisCodecId, audio_track->GetCodecId()); in TEST_F() 167 EXPECT_EQ(kTrackUid, audio_track->GetUid()); in TEST_F() 423 const AudioTrack* const audio_track = in TEST_F() local 425 EXPECT_EQ(48000, audio_track->GetSamplingRate()); in TEST_F() 426 EXPECT_EQ(6, audio_track->GetChannels()); in TEST_F() 427 EXPECT_EQ(32, audio_track->GetBitDepth()); in TEST_F() [all …]
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/external/webrtc/sdk/android/src/jni/pc/ |
D | media_stream.cc | 33 [this](AudioTrackInterface* audio_track, in JavaMediaStream() 35 OnAudioTrackAddedToStream(audio_track, media_stream); in JavaMediaStream() 37 [this](AudioTrackInterface* audio_track, in JavaMediaStream() argument 39 OnAudioTrackRemovedFromStream(audio_track, media_stream); in JavaMediaStream()
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/external/webrtc/pc/test/ |
D | peer_connection_test_wrapper.cc | 308 for (const auto& audio_track : stream->GetAudioTracks()) { in GetAndAddUserMedia() local 309 EXPECT_TRUE(peer_connection_->AddTrack(audio_track, {stream->id()}).ok()); in GetAndAddUserMedia() 332 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( in GetUserMedia() local 335 stream->AddTrack(audio_track); in GetUserMedia()
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D | mock_peer_connection_observers.h | 40 for (auto audio_track : stream->GetAudioTracks()) { in AddTrackEvent() local 41 tracks.push_back(audio_track); in AddTrackEvent()
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/external/webrtc/examples/unityplugin/ |
D | simple_peer_connection.cc | 395 rtc::scoped_refptr<webrtc::AudioTrackInterface>& audio_track = tracks[0]; in SetAudioControl() local 397 audio_track->AddSink(this); in SetAudioControl() 399 audio_track->RemoveSink(this); in SetAudioControl() 427 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( in AddStreams() local 432 stream->AddTrack(audio_track); in AddStreams()
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/external/webrtc/modules/audio_device/android/ |
D | audio_track_jni.cc | 28 std::unique_ptr<GlobalRef> audio_track) in JavaAudioTrack() argument 29 : audio_track_(std::move(audio_track)), in JavaAudioTrack()
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D | audio_record_jni.h | 51 std::unique_ptr<GlobalRef> audio_track);
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D | audio_track_jni.h | 47 std::unique_ptr<GlobalRef> audio_track);
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/external/libwebm/ |
D | webm_info.cc | 643 const mkvparser::AudioTrack* const audio_track = in OutputTracks() local 645 const int64_t channels = audio_track->GetChannels(); in OutputTracks() 646 const int64_t bit_depth = audio_track->GetBitDepth(); in OutputTracks() 647 const uint64_t codec_delay = audio_track->GetCodecDelay(); in OutputTracks() 648 const uint64_t seek_preroll = audio_track->GetSeekPreRoll(); in OutputTracks() 655 audio_track->GetSamplingRate()); in OutputTracks()
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/external/webrtc/examples/peerconnection/client/ |
D | conductor.cc | 456 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( in AddTracks() local 461 auto result_or_error = peer_connection_->AddTrack(audio_track, {kStreamId}); in AddTracks()
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