| /external/openscreen/cast/streaming/ |
| D | sender_session_unittest.cc | 229 const Json::Value& audio_stream = streams[0]; in ConstructAnswerFromOffer() local
230 const int audio_index = audio_stream["index"].asInt(); in ConstructAnswerFromOffer()
231 const int audio_ssrc = audio_stream["ssrc"].asUInt(); in ConstructAnswerFromOffer()
367 const Json::Value& audio_stream = streams[0]; in TEST_F() local
368 EXPECT_EQ("aac", audio_stream["codecName"].asString()); in TEST_F()
369 EXPECT_EQ(0, audio_stream["index"].asInt()); in TEST_F()
370 EXPECT_EQ(32u, audio_stream["aesKey"].asString().length()); in TEST_F()
371 EXPECT_EQ(32u, audio_stream["aesIvMask"].asString().length()); in TEST_F()
372 EXPECT_EQ(5, audio_stream["channels"].asInt()); in TEST_F()
373 EXPECT_LT(0u, audio_stream["ssrc"].asUInt()); in TEST_F()
[all …]
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| D | sender_session.cc | 117 AudioStream audio_stream = in CreateRemotingOffer() local
119 audio_stream.codec = AudioCodec::kNotSpecified; in CreateRemotingOffer()
120 audio_stream.stream.rtp_payload_type = in CreateRemotingOffer()
122 offer.audio_streams.push_back(std::move(audio_stream)); in CreateRemotingOffer()
|
| /external/webrtc/modules/audio_device/ |
| D | audio_device_unittest.cc | 342 AudioStream* audio_stream, in HandleCallbacks() argument
345 audio_stream_ = audio_stream; in HandleCallbacks()
363 void HandleCallbacks(AudioStream* audio_stream) { in HandleCallbacks() argument
364 HandleCallbacks(nullptr, audio_stream, 0); in HandleCallbacks()
1141 FifoAudioStream audio_stream; in TEST_P() local
1142 mock.HandleCallbacks(event(), &audio_stream, in TEST_P()
1170 FifoAudioStream audio_stream; in TEST_P() local
1171 mock.HandleCallbacks(&audio_stream); in TEST_P()
1204 LatencyAudioStream audio_stream; in TEST_P() local
1205 mock.HandleCallbacks(event(), &audio_stream, in TEST_P()
[all …]
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| /external/crosvm/android_audio/src/ |
| D | lib.rs | 372 let audio_stream = AudioStream::new( in new_async_playback_stream() localVariable
379 Ok((Box::new(NoopStreamControl::new()), Box::new(audio_stream))) in new_async_playback_stream()
406 let audio_stream = AudioStream::new( in new_async_capture_stream() localVariable
413 Ok((Box::new(NoopStreamControl::new()), Box::new(audio_stream))) in new_async_capture_stream()
|
| /external/webrtc/test/scenario/ |
| D | BUILD.gn | 49 "audio_stream.cc",
50 "audio_stream.h",
|
| /external/webrtc/modules/audio_device/android/ |
| D | audio_device_unittest.cc | 372 AudioStreamInterface* audio_stream, in HandleCallbacks() argument
375 audio_stream_ = audio_stream; in HandleCallbacks()
|
| /external/webrtc/sdk/android/native_unittests/audio_device/ |
| D | audio_device_unittest.cc | 369 AudioStreamInterface* audio_stream, in HandleCallbacks() argument
372 audio_stream_ = audio_stream; in HandleCallbacks()
|
| /external/cronet/tot/build/ |
| D | check_gn_headers_whitelist.txt | 165 remoting/protocol/audio_stream.h
|
| /external/cronet/stable/build/ |
| D | check_gn_headers_whitelist.txt | 165 remoting/protocol/audio_stream.h
|
| /external/angle/build/ |
| D | check_gn_headers_allowlist.txt | 164 remoting/protocol/audio_stream.h
|
| /external/webrtc/call/ |
| D | call.cc | 1372 AudioReceiveStreamImpl* audio_stream = in ConfigureSync() local
1386 video_stream->SetSync(audio_stream); in ConfigureSync()
|
| /external/webrtc/pc/ |
| D | webrtc_sdp_unittest.cc | 966 StreamParams audio_stream; in WebRtcSdpTest() local
967 audio_stream.id = kAudioTrackId1; in WebRtcSdpTest()
968 audio_stream.cname = kStream1Cname; in WebRtcSdpTest()
969 audio_stream.set_stream_ids({kStreamId1}); in WebRtcSdpTest()
970 audio_stream.ssrcs.push_back(kAudioTrack1Ssrc); in WebRtcSdpTest()
971 audio_desc_->AddStream(audio_stream); in WebRtcSdpTest()
|
| /external/googleapis/google/cloud/video/livestream/v1/ |
| D | outputs.proto | 44 AudioStream audio_stream = 2; field
|
| /external/google-cloud-java/java-video-live-stream/proto-google-cloud-live-stream-v1/src/main/proto/google/cloud/video/livestream/v1/ |
| D | outputs.proto | 44 AudioStream audio_stream = 2; field
|
| /external/webrtc/logging/rtc_event_log/ |
| D | rtc_event_log_parser.cc | 1235 for (const auto& audio_stream : audio_playout_events()) { in ParseStream() local
1237 StoreFirstAndLastTimestamp(audio_stream.second); in ParseStream()
|
| /external/google-cloud-java/java-video-transcoder/proto-google-cloud-video-transcoder-v1/src/main/proto/google/cloud/video/transcoder/v1/ |
| D | resources.proto | 260 AudioStream audio_stream = 2; field
|
| /external/googleapis/google/cloud/video/transcoder/v1/ |
| D | resources.proto | 289 AudioStream audio_stream = 2; field
|