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Searched refs:AudioSendStream (Results 1 – 25 of 37) sorted by relevance

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/external/webrtc/call/
Daudio_send_stream.cc21 AudioSendStream::Stats::Stats() = default;
22 AudioSendStream::Stats::~Stats() = default;
24 AudioSendStream::Config::Config(Transport* send_transport) in Config()
27 AudioSendStream::Config::~Config() = default;
29 std::string AudioSendStream::Config::ToString() const { in ToString()
43 AudioSendStream::Config::Rtp::Rtp() = default;
45 AudioSendStream::Config::Rtp::~Rtp() = default;
47 std::string AudioSendStream::Config::Rtp::ToString() const { in ToString()
65 AudioSendStream::Config::SendCodecSpec::SendCodecSpec( in SendCodecSpec()
69 AudioSendStream::Config::SendCodecSpec::~SendCodecSpec() = default;
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Dcall_unittest.cc102 AudioSendStream::Config config(&send_transport); in TEST()
104 AudioSendStream* stream = call->CreateAudioSendStream(config); in TEST()
129 AudioSendStream::Config config(&send_transport); in TEST()
130 std::list<AudioSendStream*> streams; in TEST()
134 AudioSendStream* stream = call->CreateAudioSendStream(config); in TEST()
193 AudioSendStream::Config send_config(&send_transport); in TEST()
195 AudioSendStream* send_stream = call->CreateAudioSendStream(send_config); in TEST()
215 AudioSendStream::Config send_config(&send_transport); in TEST()
217 AudioSendStream* send_stream = call->CreateAudioSendStream(send_config); in TEST()
328 AudioSendStream::Config config(&send_transport); in TEST()
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Ddegraded_call.h54 AudioSendStream* CreateAudioSendStream(
55 const AudioSendStream::Config& config) override;
56 void DestroyAudioSendStream(AudioSendStream* send_stream) override;
161 std::map<AudioSendStream*, std::unique_ptr<FakeNetworkPipeTransportAdapter>>
Daudio_send_stream.h36 class AudioSendStream : public AudioSender {
168 virtual ~AudioSendStream() = default;
170 virtual const webrtc::AudioSendStream::Config& GetConfig() const = 0;
Dcall.h92 virtual AudioSendStream* CreateAudioSendStream(
93 const AudioSendStream::Config& config) = 0;
95 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
Ddegraded_call.cc156 AudioSendStream* DegradedCall::CreateAudioSendStream( in CreateAudioSendStream()
157 const AudioSendStream::Config& config) { in CreateAudioSendStream()
161 AudioSendStream::Config degrade_config = config; in CreateAudioSendStream()
163 AudioSendStream* send_stream = call_->CreateAudioSendStream(degrade_config); in CreateAudioSendStream()
173 void DegradedCall::DestroyAudioSendStream(AudioSendStream* send_stream) { in DestroyAudioSendStream()
Dcall.cc229 webrtc::AudioSendStream* CreateAudioSendStream(
230 const webrtc::AudioSendStream::Config& config) override;
231 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
374 std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_
760 webrtc::AudioSendStream* Call::CreateAudioSendStream( in CreateAudioSendStream()
761 const webrtc::AudioSendStream::Config& config) { in CreateAudioSendStream()
777 AudioSendStream* send_stream = new AudioSendStream( in CreateAudioSendStream()
796 void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { in DestroyAudioSendStream()
804 webrtc::internal::AudioSendStream* audio_send_stream = in DestroyAudioSendStream()
805 static_cast<webrtc::internal::AudioSendStream*>(send_stream); in DestroyAudioSendStream()
/external/webrtc/audio/
Daudio_send_stream.cc48 const AudioSendStream::Config& config, in UpdateEventLogStreamConfig()
49 const AudioSendStream::Config* old_config) { in UpdateEventLogStreamConfig()
50 using SendCodecSpec = AudioSendStream::Config::SendCodecSpec; in UpdateEventLogStreamConfig()
100 AudioSendStream::AudioSendStream( in AudioSendStream() function in webrtc::internal::AudioSendStream
102 const webrtc::AudioSendStream::Config& config, in AudioSendStream()
111 : AudioSendStream(clock, in AudioSendStream()
134 AudioSendStream::AudioSendStream( in AudioSendStream() function in webrtc::internal::AudioSendStream
136 const webrtc::AudioSendStream::Config& config, in AudioSendStream()
177 AudioSendStream::~AudioSendStream() { in ~AudioSendStream()
189 const webrtc::AudioSendStream::Config& AudioSendStream::GetConfig() const { in GetConfig()
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Daudio_send_stream.h55 class AudioSendStream final : public webrtc::AudioSendStream,
58 AudioSendStream(Clock* clock,
59 const webrtc::AudioSendStream::Config& config,
69 AudioSendStream(Clock* clock,
70 const webrtc::AudioSendStream::Config& config,
78 ~AudioSendStream() override;
81 const webrtc::AudioSendStream::Config& GetConfig() const override;
82 void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
91 webrtc::AudioSendStream::Stats GetStats() const override;
92 webrtc::AudioSendStream::Stats GetStats(
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Daudio_state.h27 class AudioSendStream; variable
55 void AddSendingStream(webrtc::AudioSendStream* stream,
58 void RemoveSendingStream(webrtc::AudioSendStream* stream);
84 std::map<webrtc::AudioSendStream*, StreamProperties> sending_streams_;
Daudio_receive_stream.h41 class AudioSendStream; variable
98 void AssociateSendStream(AudioSendStream* send_stream);
101 const AudioSendStream* GetAssociatedSendStreamForTesting() const;
116 AudioSendStream* associated_send_stream_ = nullptr;
Daudio_send_stream_tests.cc65 AudioSendStream::Config* send_config, in TEST_F()
94 AudioSendStream::Config* send_config, in TEST_F()
133 AudioSendStream::Config* send_config, in TEST_F()
175 AudioSendStream::Config* send_config, in ModifyAudioConfigs()
234 AudioSendStream* send_stream, in TEST_F()
Daudio_send_stream_unittest.cc177 AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat); in ConfigHelper()
190 std::unique_ptr<internal::AudioSendStream> CreateAudioSendStream() { in CreateAudioSendStream()
193 return std::unique_ptr<internal::AudioSendStream>( in CreateAudioSendStream()
194 new internal::AudioSendStream( in CreateAudioSendStream()
201 AudioSendStream::Config& config() { return stream_config_; } in config()
210 static void AddBweToConfig(AudioSendStream::Config* config) { in AddBweToConfig()
328 AudioSendStream::Config stream_config_;
364 AudioSendStream::Config config(/*send_transport=*/nullptr); in TEST()
370 AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat); in TEST()
440 AudioSendStream::Stats stats = send_stream->GetStats(true); in TEST()
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Daudio_state.cc91 void AudioState::AddSendingStream(webrtc::AudioSendStream* stream, in AddSendingStream()
113 void AudioState::RemoveSendingStream(webrtc::AudioSendStream* stream) { in RemoveSendingStream()
/external/webrtc/media/engine/
Dfake_webrtc_call.h39 class FakeAudioSendStream final : public webrtc::AudioSendStream {
49 const webrtc::AudioSendStream::Config& config);
52 const webrtc::AudioSendStream::Config& GetConfig() const override;
53 void SetStats(const webrtc::AudioSendStream::Stats& stats);
60 void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
70 webrtc::AudioSendStream::Stats GetStats() const override;
71 webrtc::AudioSendStream::Stats GetStats(
76 webrtc::AudioSendStream::Config config_;
77 webrtc::AudioSendStream::Stats stats_;
324 webrtc::AudioSendStream* CreateAudioSendStream(
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Dfake_webrtc_call.cc24 const webrtc::AudioSendStream::Config& config) in FakeAudioSendStream()
28 const webrtc::AudioSendStream::Config& config) { in Reconfigure()
32 const webrtc::AudioSendStream::Config& FakeAudioSendStream::GetConfig() const { in GetConfig()
37 const webrtc::AudioSendStream::Stats& stats) { in SetStats()
61 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const { in GetStats()
65 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats( in GetStats()
478 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream( in CreateAudioSendStream()
479 const webrtc::AudioSendStream::Config& config) { in CreateAudioSendStream()
487 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { in DestroyAudioSendStream()
/external/webrtc/audio/test/
Daudio_end_to_end_test.h30 const AudioSendStream* send_stream() const { return send_stream_; } in send_stream()
53 AudioSendStream::Config* send_config,
56 AudioSendStream* send_stream,
63 AudioSendStream* send_stream_ = nullptr;
Daudio_end_to_end_test.cc88 AudioSendStream::Config* send_config, in ModifyAudioConfigs()
93 send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec( in ModifyAudioConfigs()
98 AudioSendStream* send_stream, in OnAudioStreamsCreated()
Dlow_bandwidth_audio_test.cc77 AudioSendStream::Config* send_config, in ModifyAudioConfigs()
79 send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec( in ModifyAudioConfigs()
Daudio_bwe_integration_test.cc109 AudioSendStream::Config* send_config, in ModifyAudioConfigs()
111 send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec( in ModifyAudioConfigs()
/external/webrtc/test/
Dcall_test.h98 void SetAudioConfig(const AudioSendStream::Config& config);
132 const AudioSendStream::Config& send_config,
193 AudioSendStream::Config audio_send_config_;
194 AudioSendStream* audio_send_stream_;
290 AudioSendStream::Config* send_config,
293 AudioSendStream* send_stream,
Dcall_test.cc281 AudioSendStream::Config audio_send_config(send_transport); in CreateAudioAndFecSendConfigs()
283 audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec( in CreateAudioAndFecSendConfigs()
295 void CallTest::SetAudioConfig(const AudioSendStream::Config& config) { in SetAudioConfig()
423 const AudioSendStream::Config& send_config, in CreateMatchingAudioConfig()
814 AudioSendStream::Config* send_config, in ModifyAudioConfigs()
818 AudioSendStream* send_stream, in OnAudioStreamsCreated()
/external/webrtc/call/test/
Dmock_audio_send_stream.h22 class MockAudioSendStream : public AudioSendStream {
24 MOCK_METHOD(const webrtc::AudioSendStream::Config&,
/external/webrtc/test/scenario/
Daudio_stream.cc76 AudioSendStream::Config send_config(send_transport); in SendAudioStream()
91 send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec( in SendAudioStream()
164 AudioSendStream::Stats stats = send_stream_->GetStats(); in StatsPrinter()
Daudio_stream.h44 AudioSendStream* send_stream_ = nullptr;

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