| /external/webrtc/webrtc/audio/ |
| D | audio_state_unittest.cc | 40 rtc::scoped_refptr<AudioState> audio_state = in TEST() local
42 EXPECT_TRUE(audio_state.get()); in TEST()
47 rtc::scoped_ptr<internal::AudioState> audio_state( in TEST() local
53 rtc::scoped_ptr<internal::AudioState> audio_state( in TEST() local
55 EXPECT_EQ(audio_state->voice_engine(), &helper.voice_engine()); in TEST()
60 rtc::scoped_ptr<internal::AudioState> audio_state( in TEST() local
63 static_cast<VoiceEngineObserver*>(audio_state.get()); in TEST()
64 EXPECT_FALSE(audio_state->typing_noise_detected()); in TEST()
67 EXPECT_FALSE(audio_state->typing_noise_detected()); in TEST()
70 EXPECT_TRUE(audio_state->typing_noise_detected()); in TEST()
[all …]
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| D | audio_send_stream.cc | 61 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, in AudioSendStream() argument
63 : config_(config), audio_state_(audio_state) { in AudioSendStream()
201 internal::AudioState* audio_state = in GetStats() local
203 stats.typing_noise_detected = audio_state->typing_noise_detected(); in GetStats()
214 internal::AudioState* audio_state = in voice_engine() local
216 VoiceEngine* voice_engine = audio_state->voice_engine(); in voice_engine()
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| D | audio_send_stream_unittest.cc | 108 rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } in audio_state() function
187 internal::AudioSendStream send_stream(helper.config(), helper.audio_state(), in TEST()
193 internal::AudioSendStream send_stream(helper.config(), helper.audio_state(), in TEST()
202 internal::AudioSendStream send_stream(helper.config(), helper.audio_state(), in TEST()
230 internal::AudioSendStream send_stream(helper.config(), helper.audio_state(), in TEST()
236 static_cast<internal::AudioState*>(helper.audio_state().get()); in TEST()
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| D | audio_receive_stream.cc | 86 const rtc::scoped_refptr<webrtc::AudioState>& audio_state) in AudioReceiveStream() argument
88 audio_state_(audio_state), in AudioReceiveStream()
249 internal::AudioState* audio_state = in voice_engine() local
251 VoiceEngine* voice_engine = audio_state->voice_engine(); in voice_engine()
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| D | audio_receive_stream_unittest.cc | 119 rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } in audio_state() function
223 helper.congestion_controller(), helper.config(), helper.audio_state()); in TEST()
242 helper.congestion_controller(), helper.config(), helper.audio_state()); in TEST()
268 helper.congestion_controller(), helper.config(), helper.audio_state()); in TEST()
289 helper.congestion_controller(), helper.config(), helper.audio_state()); in TEST()
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| D | webrtc_audio.gypi | 22 'audio/audio_state.cc',
23 'audio/audio_state.h',
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| D | BUILD.gn | 18 "audio_state.cc",
19 "audio_state.h",
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| D | audio_send_stream.h | 31 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
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| D | audio_receive_stream.h | 33 const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
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| /external/webrtc/webrtc/call/ |
| D | call.cc | 109 internal::AudioState* audio_state = in voice_engine() local
110 static_cast<internal::AudioState*>(config_.audio_state.get()); in voice_engine()
111 if (audio_state) in voice_engine()
112 return audio_state->voice_engine(); in voice_engine()
212 if (config.audio_state.get()) { in Call()
304 config, config_.audio_state, congestion_controller_.get()); in CreateAudioSendStream()
339 congestion_controller_.get(), config, config_.audio_state); in CreateAudioReceiveStream()
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| D | call_unittest.cc | 26 config.audio_state = webrtc::AudioState::Create(audio_state_config); in CallHelper()
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| D | call_perf_tests.cc | 245 sender_config.audio_state = AudioState::Create(send_audio_state_config); in TestAudioVideoSync()
247 receiver_config.audio_state = sender_config.audio_state; in TestAudioVideoSync()
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| D | bitrate_estimator_tests.cc | 113 config.audio_state = AudioState::Create(audio_state_config); in SetUp()
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| /external/webrtc/talk/app/webrtc/ |
| D | mediacontroller.cc | 70 config.audio_state = media_engine->GetAudioState(); in Construct_w()
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| /external/webrtc/webrtc/ |
| D | call.h | 85 rtc::scoped_refptr<AudioState> audio_state; member
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| D | webrtc.gyp | 100 'audio_state.h',
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| /external/webrtc/webrtc/test/ |
| D | call_test.cc | 51 send_config.audio_state = AudioState::Create(audio_state_config); in RunBaseTest()
59 recv_config.audio_state = AudioState::Create(audio_state_config); in RunBaseTest()
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