| /external/webrtc/audio/ |
| D | audio_state_unittest.cc | 99 auto audio_state = AudioState::Create(helper.config()); in TEST() local
100 EXPECT_TRUE(audio_state.get()); in TEST()
107 rtc::scoped_refptr<internal::AudioState> audio_state( in TEST() local
115 rtc::scoped_refptr<internal::AudioState> audio_state( in TEST() local
119 audio_state->AddSendingStream(&stream, 8000, 2); in TEST()
136 audio_state->audio_processing()); in TEST()
147 audio_state->audio_transport()->RecordedDataIsAvailable( in TEST()
152 audio_state->RemoveSendingStream(&stream); in TEST()
159 rtc::scoped_refptr<internal::AudioState> audio_state( in TEST() local
164 audio_state->AddSendingStream(&stream_1, 8001, 2); in TEST()
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| D | audio_receive_stream.cc | 70 webrtc::AudioState* audio_state, in CreateChannelReceive() argument
75 RTC_DCHECK(audio_state); in CreateChannelReceive()
77 static_cast<internal::AudioState*>(audio_state); in CreateChannelReceive()
97 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, in AudioReceiveStream() argument
103 audio_state, in AudioReceiveStream()
106 audio_state.get(), in AudioReceiveStream()
117 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, in AudioReceiveStream() argument
120 : audio_state_(audio_state), in AudioReceiveStream()
163 audio_state()->AddReceivingStream(this); in Start()
173 audio_state()->RemoveReceivingStream(this); in Stop()
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| D | audio_send_stream.cc | 103 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, in AudioSendStream() argument
113 audio_state, in AudioSendStream()
137 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, in AudioSendStream() argument
154 audio_state_(audio_state), in AudioSendStream()
395 audio_state()->AddSendingStream(this, encoder_sample_rate_hz_, in Start()
408 audio_state()->RemoveSendingStream(this); in Stop()
497 stats.typing_noise_detected = audio_state()->typing_noise_detected(); in GetStats()
590 internal::AudioState* AudioSendStream::audio_state() { in audio_state() function in webrtc::internal::AudioSendStream
591 internal::AudioState* audio_state = in audio_state() local
593 RTC_DCHECK(audio_state); in audio_state()
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| D | audio_send_stream.h | 60 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
71 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
117 internal::AudioState* audio_state();
118 const internal::AudioState* audio_state() const;
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| D | audio_receive_stream.h | 53 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
61 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
108 AudioState* audio_state() const;
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| D | BUILD.gn | 23 "audio_state.cc",
24 "audio_state.h",
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| /external/webrtc/test/scenario/ |
| D | call_client.cc | 47 setup.audio_state = AudioState::Create(audio_state_config); in InitAudio()
49 setup.audio_state->audio_transport()); in InitAudio()
57 rtc::scoped_refptr<AudioState> audio_state, in CreateCall() argument
68 call_config.audio_state = audio_state; in CreateCall()
224 fake_audio_setup_.audio_state, module_thread_)); in CallClient()
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| D | call_client.h | 91 rtc::scoped_refptr<AudioState> audio_state; member
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| /external/webrtc/call/ |
| D | call_config.h | 38 rtc::scoped_refptr<AudioState> audio_state; member
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| D | call_perf_tests.cc | 208 auto audio_state = AudioState::Create(send_audio_state_config); in TestAudioVideoSync() local
209 fake_audio_device->RegisterAudioCallback(audio_state->audio_transport()); in TestAudioVideoSync()
210 sender_config.audio_state = audio_state; in TestAudioVideoSync()
212 receiver_config.audio_state = audio_state; in TestAudioVideoSync()
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| D | BUILD.gn | 16 "audio_state.cc",
17 "audio_state.h",
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| D | call_unittest.cc | 59 config.audio_state = webrtc::AudioState::Create(audio_state_config); in CallHelper()
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| D | call.cc | 778 clock_, config, config_.audio_state, task_queue_factory_, in CreateAudioSendStream()
831 config_.audio_state, event_log_); in CreateAudioReceiveStream()
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| /external/igt-gpu-tools/tests/ |
| D | kms_chamelium.c | 1244 struct audio_state { struct
1272 static void audio_state_init(struct audio_state *state, data_t *data, in audio_state_init() argument
1294 static void audio_state_fini(struct audio_state *state) in audio_state_fini()
1308 static void audio_state_start(struct audio_state *state, const char *name) in audio_state_start()
1382 static void audio_state_receive(struct audio_state *state, in audio_state_receive()
1406 static void audio_state_stop(struct audio_state *state, bool success) in audio_state_stop()
1449 static void check_audio_infoframe(struct audio_state *state) in check_audio_infoframe()
1505 struct audio_state *state = data; in audio_output_frequencies_callback()
1518 static bool test_audio_frequencies(struct audio_state *state) in test_audio_frequencies()
1635 struct audio_state *state = data; in audio_output_flatline_callback()
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| /external/webrtc/test/ |
| D | call_test.cc | 109 send_config.audio_state = AudioState::Create(audio_state_config); in RunBaseTest()
111 send_config.audio_state->audio_transport()); in RunBaseTest()
122 recv_config.audio_state = AudioState::Create(audio_state_config); in RunBaseTest()
124 recv_config.audio_state->audio_transport()); in RunBaseTest()
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| /external/webrtc/media/engine/ |
| D | webrtc_voice_engine.h | 98 webrtc::AudioState* audio_state();
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| D | webrtc_voice_engine.cc | 322 adm()->RegisterAudioCallback(audio_state()->audio_transport()); in Init()
482 audio_state()->SetStereoChannelSwapping(*options.stereo_swapping); in ApplyOptions()
666 webrtc::AudioState* WebRtcVoiceEngine::audio_state() { in audio_state() function in cricket::WebRtcVoiceEngine
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| /external/webrtc/pc/ |
| D | peer_connection_factory.cc | 364 call_config.audio_state = in CreateCall_w()
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| D | peer_connection.cc | 4178 auto audio_state = in SetAudioPlayout() local
4180 audio_state->SetPlayout(playout); in SetAudioPlayout()
4190 auto audio_state = in SetAudioRecording() local
4192 audio_state->SetRecording(recording); in SetAudioRecording()
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| /external/webrtc/video/ |
| D | video_quality_test.cc | 1378 send_call_config->audio_state = AudioState::Create(audio_state_config); in InitializeAudioDevice()
1379 recv_call_config->audio_state = AudioState::Create(audio_state_config); in InitializeAudioDevice()
1390 send_call_config->audio_state->audio_transport()) == 0); in InitializeAudioDevice()
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| /external/webrtc/ |
| D | Android.bp | 3681 "call/audio_state.cc",
4202 "audio/audio_state.cc",
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| /external/webrtc/android_tools/ |
| D | sorted_targets.txt | 60266 "//call/audio_state.cc": [
60267 "obj/call/call_interfaces/audio_state.o"
60283 "//call/audio_state.cc",
60284 "//call/audio_state.h",
64381 "//audio/audio_state.cc": [
64382 "obj/audio/audio/audio_state.o"
64413 "//audio/audio_state.cc",
64414 "//audio/audio_state.h",
|