| AcmReceiver(const AudioCodingModule::Config &config) | webrtc::acm2::AcmReceiver | explicit |
| AcmReceiver(const AudioCodingModule::Config &config) | webrtc::acm2::AcmReceiver | explicit |
| AddCodec(int acm_codec_id, uint8_t payload_type, size_t channels, int sample_rate_hz, AudioDecoder *audio_decoder, const std::string &name) | webrtc::acm2::AcmReceiver | |
| AddCodec(int rtp_payload_type, const SdpAudioFormat &audio_format) | webrtc::acm2::AcmReceiver | |
| AddCodec(int acm_codec_id, uint8_t payload_type, size_t channels, int sample_rate_hz, AudioDecoder *audio_decoder, const std::string &name) | webrtc::acm2::AcmReceiver | |
| AddCodec(int rtp_payload_type, const SdpAudioFormat &audio_format) | webrtc::acm2::AcmReceiver | |
| DecoderByPayloadType(uint8_t payload_type, CodecInst *codec) const | webrtc::acm2::AcmReceiver | |
| DecoderByPayloadType(uint8_t payload_type, CodecInst *codec) const | webrtc::acm2::AcmReceiver | |
| DisableNack() | webrtc::acm2::AcmReceiver | |
| DisableNack() | webrtc::acm2::AcmReceiver | |
| EnableNack(size_t max_nack_list_size) | webrtc::acm2::AcmReceiver | |
| EnableNack(size_t max_nack_list_size) | webrtc::acm2::AcmReceiver | |
| FilteredCurrentDelayMs() const | webrtc::acm2::AcmReceiver | |
| FilteredCurrentDelayMs() const | webrtc::acm2::AcmReceiver | |
| FlushBuffers() | webrtc::acm2::AcmReceiver | |
| FlushBuffers() | webrtc::acm2::AcmReceiver | |
| GetAudio(int desired_freq_hz, AudioFrame *audio_frame, bool *muted) | webrtc::acm2::AcmReceiver | |
| GetAudio(int desired_freq_hz, AudioFrame *audio_frame, bool *muted) | webrtc::acm2::AcmReceiver | |
| GetDecodingCallStatistics(AudioDecodingCallStats *stats) const | webrtc::acm2::AcmReceiver | |
| GetDecodingCallStatistics(AudioDecodingCallStats *stats) const | webrtc::acm2::AcmReceiver | |
| GetNackList(int64_t round_trip_time_ms) const | webrtc::acm2::AcmReceiver | |
| GetNackList(int64_t round_trip_time_ms) const | webrtc::acm2::AcmReceiver | |
| GetNetworkStatistics(NetworkStatistics *statistics) | webrtc::acm2::AcmReceiver | |
| GetNetworkStatistics(NetworkStatistics *statistics) | webrtc::acm2::AcmReceiver | |
| GetPlayoutTimestamp() | webrtc::acm2::AcmReceiver | |
| GetPlayoutTimestamp() | webrtc::acm2::AcmReceiver | |
| InsertPacket(const WebRtcRTPHeader &rtp_header, rtc::ArrayView< const uint8_t > incoming_payload) | webrtc::acm2::AcmReceiver | |
| InsertPacket(const WebRtcRTPHeader &rtp_header, rtc::ArrayView< const uint8_t > incoming_payload) | webrtc::acm2::AcmReceiver | |
| last_output_sample_rate_hz() const | webrtc::acm2::AcmReceiver | |
| last_output_sample_rate_hz() const | webrtc::acm2::AcmReceiver | |
| last_packet_sample_rate_hz() const | webrtc::acm2::AcmReceiver | |
| last_packet_sample_rate_hz() const | webrtc::acm2::AcmReceiver | |
| LastAudioCodec(CodecInst *codec) const | webrtc::acm2::AcmReceiver | |
| LastAudioCodec(CodecInst *codec) const | webrtc::acm2::AcmReceiver | |
| LastAudioFormat() const | webrtc::acm2::AcmReceiver | |
| LastAudioFormat() const | webrtc::acm2::AcmReceiver | |
| LeastRequiredDelayMs() const | webrtc::acm2::AcmReceiver | |
| LeastRequiredDelayMs() const | webrtc::acm2::AcmReceiver | |
| RemoveAllCodecs() | webrtc::acm2::AcmReceiver | |
| RemoveAllCodecs() | webrtc::acm2::AcmReceiver | |
| RemoveCodec(uint8_t payload_type) | webrtc::acm2::AcmReceiver | |
| RemoveCodec(uint8_t payload_type) | webrtc::acm2::AcmReceiver | |
| ResetInitialDelay() | webrtc::acm2::AcmReceiver | |
| ResetInitialDelay() | webrtc::acm2::AcmReceiver | |
| SetMaximumDelay(int delay_ms) | webrtc::acm2::AcmReceiver | |
| SetMaximumDelay(int delay_ms) | webrtc::acm2::AcmReceiver | |
| SetMinimumDelay(int delay_ms) | webrtc::acm2::AcmReceiver | |
| SetMinimumDelay(int delay_ms) | webrtc::acm2::AcmReceiver | |
| ~AcmReceiver() | webrtc::acm2::AcmReceiver | |
| ~AcmReceiver() | webrtc::acm2::AcmReceiver | |
