/external/webrtc/pc/ |
D | media_stream_unittest.cc | 109 scoped_refptr<webrtc::MediaStreamTrackInterface> audio_track( in TEST_F() local 111 EXPECT_EQ(0, audio_track->id().compare(kAudioTrackId)); in TEST_F() 112 EXPECT_TRUE(audio_track->enabled()); in TEST_F() 114 EXPECT_TRUE(stream_->GetAudioTracks()[0].get() == audio_track.get()); in TEST_F() 115 EXPECT_TRUE(stream_->FindAudioTrack(audio_track->id()).get() == in TEST_F() 116 audio_track.get()); in TEST_F() 117 audio_track = stream_->GetAudioTracks()[0]; in TEST_F() 118 EXPECT_EQ(0, audio_track->id().compare(kAudioTrackId)); in TEST_F() 119 EXPECT_TRUE(audio_track->enabled()); in TEST_F() 149 scoped_refptr<webrtc::AudioTrackInterface> audio_track( in TEST_F() local [all …]
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D | rtp_sender.cc | 486 audio_track()->RemoveSink(sink_adapter_.get()); in DetachTrack() 492 audio_track()->AddSink(sink_adapter_.get()); in AttachTrack() 497 stats_->AddLocalAudioTrack(audio_track().get(), ssrc_); in AddTrackToStats() 503 stats_->RemoveLocalAudioTrack(audio_track().get(), ssrc_); in RemoveTrackFromStats() 523 if (track_->enabled() && audio_track()->GetSource() && in SetSend() 524 !audio_track()->GetSource()->remote()) { in SetSend() 525 options = audio_track()->GetSource()->options(); in SetSend()
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D | peer_connection_rtp_unittest.cc | 948 auto audio_track = caller->CreateAudioTrack("audio track"); in TEST_F() local 949 auto transceiver = caller->AddTransceiver(audio_track); in TEST_F() 953 EXPECT_EQ(audio_track, sender->track()); in TEST_F() 968 auto audio_track = caller->CreateAudioTrack("audio track"); in TEST_F() local 970 auto transceiver1 = caller->AddTransceiver(audio_track); in TEST_F() 971 auto transceiver2 = caller->AddTransceiver(audio_track); in TEST_F() 978 EXPECT_EQ(audio_track, sender1->track()); in TEST_F() 979 EXPECT_EQ(audio_track, sender2->track()); in TEST_F() 1011 auto audio_track = caller->CreateAudioTrack("a"); in TEST_F() local 1012 auto sender = caller->AddTrack(audio_track); in TEST_F() [all …]
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D | stats_collector.h | 60 void AddLocalAudioTrack(AudioTrackInterface* audio_track, uint32_t ssrc); 64 void RemoveLocalAudioTrack(AudioTrackInterface* audio_track, uint32_t ssrc);
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D | stats_collector.cc | 544 void StatsCollector::AddLocalAudioTrack(AudioTrackInterface* audio_track, in AddLocalAudioTrack() argument 547 RTC_DCHECK(audio_track != NULL); in AddLocalAudioTrack() 550 RTC_DCHECK(track.first != audio_track || track.second != ssrc); in AddLocalAudioTrack() 553 local_audio_tracks_.push_back(std::make_pair(audio_track, ssrc)); in AddLocalAudioTrack() 558 audio_track->id())); in AddLocalAudioTrack() 562 report->AddString(StatsReport::kStatsValueNameTrackId, audio_track->id()); in AddLocalAudioTrack() 566 void StatsCollector::RemoveLocalAudioTrack(AudioTrackInterface* audio_track, in RemoveLocalAudioTrack() argument 568 RTC_DCHECK(audio_track != NULL); in RemoveLocalAudioTrack() 572 [audio_track, ssrc](const LocalAudioTrackVector::value_type& track) { in RemoveLocalAudioTrack() 573 return track.first == audio_track && track.second == ssrc; in RemoveLocalAudioTrack()
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D | peer_connection_interface_unittest.cc | 559 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( in CreateStreamCollection() local 562 stream->AddTrack(audio_track); in CreateStreamCollection() 1135 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( in AddAudioTrack() local 1137 ASSERT_TRUE(stream->AddTrack(audio_track)); in AddAudioTrack() 1449 rtc::scoped_refptr<AudioTrackInterface> audio_track( in TEST_F() local 1452 stream->AddTrack(audio_track.get()); in TEST_F() 1516 rtc::scoped_refptr<AudioTrackInterface> audio_track( in TEST_F() local 1520 auto audio_sender = pc_->AddTrack(audio_track, {kStreamId1}).MoveValue(); in TEST_F() 1525 EXPECT_EQ(audio_track, audio_sender->track()); in TEST_F() 1574 rtc::scoped_refptr<AudioTrackInterface> audio_track( in TEST_P() local [all …]
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D | rtc_stats_collector.cc | 630 const AudioTrackInterface& audio_track, in ProduceMediaStreamTrackStatsFromVoiceSenderInfo() argument 639 audio_track, audio_track_stats.get()); in ProduceMediaStreamTrackStatsFromVoiceSenderInfo() 659 const AudioTrackInterface& audio_track, in ProduceMediaStreamTrackStatsFromVoiceReceiverInfo() argument 670 audio_track, audio_track_stats.get()); in ProduceMediaStreamTrackStatsFromVoiceReceiverInfo() 1633 rtc::scoped_refptr<AudioTrackInterface> audio_track = in ProduceAudioRTPStreamStats_n() local 1635 if (audio_track) { in ProduceAudioRTPStreamStats_n() 1639 track_media_info_map.GetAttachmentIdByTrack(audio_track).value()); in ProduceAudioRTPStreamStats_n() 1655 rtc::scoped_refptr<AudioTrackInterface> audio_track = in ProduceAudioRTPStreamStats_n() local 1657 if (audio_track) { in ProduceAudioRTPStreamStats_n() 1659 track_media_info_map.GetAttachmentIdByTrack(audio_track).value(); in ProduceAudioRTPStreamStats_n()
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D | audio_rtp_receiver.h | 54 rtc::scoped_refptr<AudioTrackInterface> audio_track() const { in audio_track() function
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D | stats_collector_unittest.cc | 558 AudioTrackInterface* audio_track, in UpdateVoiceSenderInfoFromAudioTrack() argument 561 audio_track->GetSignalLevel(&voice_sender_info->audio_level); in UpdateVoiceSenderInfoFromAudioTrack() 563 audio_track->GetAudioProcessor()->GetStats(has_remote_tracks); in UpdateVoiceSenderInfoFromAudioTrack() 611 void VerifyAudioTrackStats(FakeAudioTrack* audio_track, in VerifyAudioTrackStats() argument 626 EXPECT_EQ(audio_track->id(), track_id); in VerifyAudioTrackStats() 645 stats->GetStats(audio_track, &track_reports); in VerifyAudioTrackStats() 652 EXPECT_EQ(audio_track->id(), track_id); in VerifyAudioTrackStats()
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D | rtp_sender.h | 295 rtc::scoped_refptr<AudioTrackInterface> audio_track() const { in audio_track() function
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D | BUILD.gn | 162 "audio_track.cc", 163 "audio_track.h",
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D | webrtc_sdp_unittest.cc | 1351 StreamParams audio_track; in MakeUnifiedPlanDescriptionNoSsrcSignaling() local 1352 audio_track.id = kAudioTrackId1; in MakeUnifiedPlanDescriptionNoSsrcSignaling() 1353 audio_track.set_stream_ids({kStreamId1}); in MakeUnifiedPlanDescriptionNoSsrcSignaling() 1354 audio_desc->AddStream(audio_track); in MakeUnifiedPlanDescriptionNoSsrcSignaling()
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D | rtp_sender_receiver_unittest.cc | 295 audio_track_ = audio_rtp_receiver_->audio_track();
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D | peer_connection_integrationtest.cc | 2932 auto audio_track = caller()->CreateLocalAudioTrack(); in TEST_P() local 2934 caller()->AddTrack(audio_track); in TEST_P() 2945 EXPECT_GT(caller()->OldGetStatsForTrack(audio_track)->BytesSent(), 0); in TEST_P()
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D | peer_connection.cc | 5423 rtc::scoped_refptr<AudioTrackInterface> audio_track = in OnRemoteSenderRemoved() local 5425 if (audio_track) { in OnRemoteSenderRemoved() 5426 stream->RemoveTrack(audio_track); in OnRemoteSenderRemoved()
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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/examples/unityplugin/ |
D | simple_peer_connection.cc | 393 webrtc::AudioTrackInterface* audio_track = tracks[0]; in SetAudioControl() local 394 std::string id = audio_track->id(); in SetAudioControl() 396 audio_track->AddSink(this); in SetAudioControl() 398 audio_track->RemoveSink(this); in SetAudioControl() 426 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( in AddStreams() local 430 std::string id = audio_track->id(); in AddStreams() 431 stream->AddTrack(audio_track); in AddStreams()
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/external/webrtc/pc/test/ |
D | peer_connection_test_wrapper.cc | 298 for (const auto& audio_track : stream->GetAudioTracks()) { in GetAndAddUserMedia() local 299 EXPECT_TRUE(peer_connection_->AddTrack(audio_track, {stream->id()}).ok()); in GetAndAddUserMedia() 322 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( in GetUserMedia() local 325 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/modules/audio_device/android/ |
D | audio_track_jni.cc | 29 std::unique_ptr<GlobalRef> audio_track) in JavaAudioTrack() argument 30 : 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 | 439 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( in AddTracks() local 443 auto result_or_error = peer_connection_->AddTrack(audio_track, {kStreamId}); in AddTracks()
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/external/webrtc/sdk/android/ |
D | BUILD.gn | 672 "src/jni/pc/audio_track.cc",
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