AssociateSendStream(AudioSendStream *send_stream) | webrtc::internal::AudioReceiveStream | |
AssociateSendStream(AudioSendStream *send_stream) | webrtc::internal::AudioReceiveStream | |
AudioFrameInfo enum name | webrtc::AudioMixer::Source | |
AudioFrameInfo enum name | webrtc::AudioMixer::Source | |
AudioReceiveStream(PacketRouter *packet_router, const webrtc::AudioReceiveStream::Config &config, const rtc::scoped_refptr< webrtc::AudioState > &audio_state, webrtc::RtcEventLog *event_log) | webrtc::internal::AudioReceiveStream | |
AudioReceiveStream(PacketRouter *packet_router, const webrtc::AudioReceiveStream::Config &config, const rtc::scoped_refptr< webrtc::AudioState > &audio_state, webrtc::RtcEventLog *event_log) | webrtc::internal::AudioReceiveStream | |
config() const | webrtc::internal::AudioReceiveStream | |
config() const | webrtc::internal::AudioReceiveStream | |
DeliverRtcp(const uint8_t *packet, size_t length) | webrtc::internal::AudioReceiveStream | |
DeliverRtcp(const uint8_t *packet, size_t length) | webrtc::internal::AudioReceiveStream | |
GetAudioFrameWithInfo(int sample_rate_hz, AudioFrame *audio_frame) override | webrtc::internal::AudioReceiveStream | virtual |
GetAudioFrameWithInfo(int sample_rate_hz, AudioFrame *audio_frame) override | webrtc::internal::AudioReceiveStream | virtual |
GetInfo() const override | webrtc::internal::AudioReceiveStream | virtual |
GetInfo() const override | webrtc::internal::AudioReceiveStream | virtual |
GetOutputLevel() const override | webrtc::internal::AudioReceiveStream | virtual |
GetOutputLevel() const override | webrtc::internal::AudioReceiveStream | virtual |
GetPlayoutTimestamp() const override | webrtc::internal::AudioReceiveStream | virtual |
GetPlayoutTimestamp() const override | webrtc::internal::AudioReceiveStream | virtual |
GetStats() const override | webrtc::internal::AudioReceiveStream | virtual |
GetStats() const override | webrtc::internal::AudioReceiveStream | virtual |
id() const override | webrtc::internal::AudioReceiveStream | virtual |
id() const override | webrtc::internal::AudioReceiveStream | virtual |
OnRtpPacket(const RtpPacketReceived &packet) | webrtc::internal::AudioReceiveStream | |
OnRtpPacket(const RtpPacketReceived &packet) | webrtc::internal::AudioReceiveStream | |
PreferredSampleRate() const override | webrtc::internal::AudioReceiveStream | virtual |
PreferredSampleRate() const override | webrtc::internal::AudioReceiveStream | virtual |
SetGain(float gain) override | webrtc::internal::AudioReceiveStream | virtual |
SetGain(float gain) override | webrtc::internal::AudioReceiveStream | virtual |
SetMinimumPlayoutDelay(int delay_ms) override | webrtc::internal::AudioReceiveStream | virtual |
SetMinimumPlayoutDelay(int delay_ms) override | webrtc::internal::AudioReceiveStream | virtual |
SetSink(std::unique_ptr< AudioSinkInterface > sink) override | webrtc::internal::AudioReceiveStream | virtual |
SetSink(std::unique_ptr< AudioSinkInterface > sink) override | webrtc::internal::AudioReceiveStream | virtual |
SignalNetworkState(NetworkState state) | webrtc::internal::AudioReceiveStream | |
SignalNetworkState(NetworkState state) | webrtc::internal::AudioReceiveStream | |
Ssrc() const override | webrtc::internal::AudioReceiveStream | virtual |
Ssrc() const override | webrtc::internal::AudioReceiveStream | virtual |
Start() override | webrtc::internal::AudioReceiveStream | virtual |
Start() override | webrtc::internal::AudioReceiveStream | virtual |
Stop() override | webrtc::internal::AudioReceiveStream | virtual |
Stop() override | webrtc::internal::AudioReceiveStream | virtual |
~AudioReceiveStream() override | webrtc::internal::AudioReceiveStream | virtual |
~AudioReceiveStream() override | webrtc::internal::AudioReceiveStream | virtual |
~Source() | webrtc::AudioMixer::Source | inlinevirtual |
~Source() | webrtc::AudioMixer::Source | inlinevirtual |
~Syncable() | webrtc::Syncable | virtual |
~Syncable() | webrtc::Syncable | virtual |