| /external/webrtc/audio/ |
| D | audio_state.cc | 30 AudioState::AudioState(const AudioState::Config& config) in AudioState() function in webrtc::internal::AudioState
40 AudioState::~AudioState() { in ~AudioState()
47 AudioProcessing* AudioState::audio_processing() { in audio_processing()
51 AudioTransport* AudioState::audio_transport() { in audio_transport()
55 void AudioState::AddReceivingStream( in AddReceivingStream()
79 void AudioState::RemoveReceivingStream( in RemoveReceivingStream()
92 void AudioState::AddSendingStream(webrtc::AudioSendStream* stream, in AddSendingStream()
114 void AudioState::RemoveSendingStream(webrtc::AudioSendStream* stream) { in RemoveSendingStream()
124 void AudioState::SetPlayout(bool enabled) { in SetPlayout()
141 void AudioState::SetRecording(bool enabled) { in SetRecording()
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| D | audio_state.h | 32 class AudioState : public webrtc::AudioState {
34 explicit AudioState(const AudioState::Config& config);
36 AudioState() = delete;
37 AudioState(const AudioState&) = delete;
38 AudioState& operator=(const AudioState&) = delete;
40 ~AudioState() override;
69 const webrtc::AudioState::Config config_;
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| D | audio_state_unittest.cc | 114 AudioState::Config& config() { return audio_state_config; } in config()
122 AudioState::Config audio_state_config;
174 auto audio_state = AudioState::Create(helper.config()); in TEST_P()
180 rtc::scoped_refptr<internal::AudioState> audio_state( in TEST_P()
181 rtc::make_ref_counted<internal::AudioState>(helper.config())); in TEST_P()
193 rtc::scoped_refptr<internal::AudioState> audio_state( in TEST_P()
194 rtc::make_ref_counted<internal::AudioState>(helper.config())); in TEST_P()
242 rtc::scoped_refptr<internal::AudioState> audio_state( in TEST_P()
243 rtc::make_ref_counted<internal::AudioState>(helper.config())); in TEST_P()
305 rtc::scoped_refptr<internal::AudioState> audio_state( in TEST_P()
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| D | audio_send_stream.h | 56 class AudioState; variable
63 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
74 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
127 internal::AudioState* audio_state();
128 const internal::AudioState* audio_state() const;
185 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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| D | audio_receive_stream.h | 54 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
61 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
152 internal::AudioState* audio_state() const;
164 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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| D | audio_receive_stream.cc | 71 webrtc::AudioState* audio_state, in CreateChannelReceive()
76 internal::AudioState* internal_audio_state = in CreateChannelReceive()
77 static_cast<internal::AudioState*>(audio_state); in CreateChannelReceive()
94 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, in AudioReceiveStreamImpl()
111 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, in AudioReceiveStreamImpl()
507 internal::AudioState* AudioReceiveStreamImpl::audio_state() const { in audio_state()
508 auto* audio_state = static_cast<internal::AudioState*>(audio_state_.get()); in audio_state()
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| D | audio_send_stream.cc | 105 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, in AudioSendStream()
140 const rtc::scoped_refptr<webrtc::AudioState>& audio_state, in AudioSendStream()
579 internal::AudioState* AudioSendStream::audio_state() { in audio_state()
580 internal::AudioState* audio_state = in audio_state()
581 static_cast<internal::AudioState*>(audio_state_.get()); in audio_state()
586 const internal::AudioState* AudioSendStream::audio_state() const { in audio_state()
587 internal::AudioState* audio_state = in audio_state()
588 static_cast<internal::AudioState*>(audio_state_.get()); in audio_state()
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| D | audio_receive_stream_unittest.cc | 112 AudioState::Config config; in ConfigHelper()
120 audio_state_ = AudioState::Create(config); in ConfigHelper()
191 rtc::scoped_refptr<AudioState> audio_state_;
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| D | audio_send_stream_unittest.cc | 164 AudioState::Config config; in ConfigHelper()
168 audio_state_ = AudioState::Create(config); in ConfigHelper()
328 rtc::scoped_refptr<AudioState> audio_state_;
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| /external/webrtc/call/ |
| D | audio_state.h | 26 class AudioState : public rtc::RefCountInterface {
62 static rtc::scoped_refptr<AudioState> Create(
63 const AudioState::Config& config);
65 ~AudioState() override {} in ~AudioState()
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| D | audio_state.cc | 15 AudioState::Config::Config() = default;
16 AudioState::Config::~Config() = default;
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| D | call_config.h | 46 rtc::scoped_refptr<AudioState> audio_state;
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| D | call_unittest.cc | 51 webrtc::AudioState::Config audio_state_config; in CallHelper()
62 config.audio_state = webrtc::AudioState::Create(audio_state_config); in CallHelper()
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| D | call_perf_tests.cc | 216 AudioState::Config send_audio_state_config; in TestAudioVideoSync()
223 auto audio_state = AudioState::Create(send_audio_state_config); in TestAudioVideoSync()
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| /external/webrtc/media/engine/ |
| D | webrtc_voice_engine.h | 69 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
103 webrtc::AudioState* audio_state();
121 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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| D | webrtc_voice_engine.cc | 366 webrtc::AudioState::Config config; in Init()
378 audio_state_ = webrtc::AudioState::Create(config); in Init()
405 rtc::scoped_refptr<webrtc::AudioState> WebRtcVoiceEngine::GetAudioState() in GetAudioState()
639 webrtc::AudioState* WebRtcVoiceEngine::audio_state() { in audio_state()
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| D | webrtc_voice_engine_unittest.cc | 3768 webrtc::AudioState::Config config; in TEST()
3772 call_config.audio_state = webrtc::AudioState::Create(config); in TEST()
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| /external/webrtc/test/scenario/ |
| D | call_client.cc | 48 AudioState::Config audio_state_config; in InitAudio()
52 setup.audio_state = AudioState::Create(audio_state_config); in InitAudio()
62 rtc::scoped_refptr<AudioState> audio_state) { in CreateCall()
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| D | call_client.h | 97 rtc::scoped_refptr<AudioState> audio_state;
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| /external/webrtc/test/ |
| D | call_test.cc | 105 AudioState::Config audio_state_config; in RunBaseTest()
109 send_config.audio_state = AudioState::Create(audio_state_config); in RunBaseTest()
118 AudioState::Config audio_state_config; in RunBaseTest()
122 recv_config.audio_state = AudioState::Create(audio_state_config); in RunBaseTest()
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| /external/webrtc/media/base/ |
| D | media_engine.h | 95 virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
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| D | fake_media_engine.cc | 441 rtc::scoped_refptr<webrtc::AudioState> FakeVoiceEngine::GetAudioState() const { in GetAudioState()
442 return rtc::scoped_refptr<webrtc::AudioState>(); in GetAudioState()
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| D | fake_media_engine.h | 495 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
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| /external/webrtc/video/ |
| D | video_quality_test.cc | 1380 AudioState::Config audio_state_config; in InitializeAudioDevice()
1384 send_call_config->audio_state = AudioState::Create(audio_state_config); in InitializeAudioDevice()
1385 recv_call_config->audio_state = AudioState::Create(audio_state_config); in InitializeAudioDevice()
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| /external/webrtc/modules/audio_device/g3doc/ |
| D | audio_device_module.md | 119 `WebRtcVoiceEngine` also owns an [`AudioState`][30] member and this class is
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