| Application enum name | webrtc::AudioEncoder | |
| Application enum name | webrtc::AudioEncoder | |
| CodecType enum name | webrtc::AudioEncoder | |
| CodecType enum name | webrtc::AudioEncoder | |
| DisableAudioNetworkAdaptor() | webrtc::AudioEncoder | virtual |
| DisableAudioNetworkAdaptor() | webrtc::AudioEncoder | virtual |
| EnableAudioNetworkAdaptor(const std::string &config_string, RtcEventLog *event_log, const Clock *clock) | webrtc::AudioEncoder | virtual |
| EnableAudioNetworkAdaptor(const std::string &config_string, RtcEventLog *event_log, const Clock *clock) | webrtc::AudioEncoder | virtual |
| Encode(uint32_t rtp_timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded) | webrtc::AudioEncoder | |
| Encode(uint32_t rtp_timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded) | webrtc::AudioEncoder | |
| EncodeImpl(uint32_t rtp_timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded)=0 | webrtc::AudioEncoder | protectedpure virtual |
| EncodeImpl(uint32_t rtp_timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded)=0 | webrtc::AudioEncoder | protectedpure virtual |
| GetDtx() const | webrtc::AudioEncoder | virtual |
| GetDtx() const | webrtc::AudioEncoder | virtual |
| GetTargetBitrate() const =0 | webrtc::AudioEncoder | pure virtual |
| GetTargetBitrate() const =0 | webrtc::AudioEncoder | pure virtual |
| Max10MsFramesInAPacket() const =0 | webrtc::AudioEncoder | pure virtual |
| Max10MsFramesInAPacket() const =0 | webrtc::AudioEncoder | pure virtual |
| MOCK_CONST_METHOD0(SampleRateHz, int()) | webrtc::MockAudioEncoder | |
| MOCK_CONST_METHOD0(NumChannels, size_t()) | webrtc::MockAudioEncoder | |
| MOCK_CONST_METHOD0(RtpTimestampRateHz, int()) | webrtc::MockAudioEncoder | |
| MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, size_t()) | webrtc::MockAudioEncoder | |
| MOCK_CONST_METHOD0(Max10MsFramesInAPacket, size_t()) | webrtc::MockAudioEncoder | |
| MOCK_CONST_METHOD0(GetTargetBitrate, int()) | webrtc::MockAudioEncoder | |
| MOCK_CONST_METHOD0(SampleRateHz, int()) | webrtc::MockAudioEncoder | |
| MOCK_CONST_METHOD0(NumChannels, size_t()) | webrtc::MockAudioEncoder | |
| MOCK_CONST_METHOD0(RtpTimestampRateHz, int()) | webrtc::MockAudioEncoder | |
| MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, size_t()) | webrtc::MockAudioEncoder | |
| MOCK_CONST_METHOD0(Max10MsFramesInAPacket, size_t()) | webrtc::MockAudioEncoder | |
| MOCK_CONST_METHOD0(GetTargetBitrate, int()) | webrtc::MockAudioEncoder | |
| MOCK_METHOD0(Die, void()) | webrtc::MockAudioEncoder | |
| MOCK_METHOD0(Reset, void()) | webrtc::MockAudioEncoder | |
| MOCK_METHOD0(Die, void()) | webrtc::MockAudioEncoder | |
| MOCK_METHOD0(Reset, void()) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(Mark, void(std::string desc)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(SetFec, bool(bool enable)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(SetDtx, bool(bool enable)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(SetApplication, bool(Application application)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(SetMaxPlaybackRate, void(int frequency_hz)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(SetMaxBitrate, void(int max_bps)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(OnReceivedUplinkPacketLossFraction, void(float uplink_packet_loss_fraction)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(Mark, void(std::string desc)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(SetFec, bool(bool enable)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(SetDtx, bool(bool enable)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(SetApplication, bool(Application application)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(SetMaxPlaybackRate, void(int frequency_hz)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(SetMaxBitrate, void(int max_bps)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD1(OnReceivedUplinkPacketLossFraction, void(float uplink_packet_loss_fraction)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD2(OnReceivedUplinkBandwidth, void(int target_audio_bitrate_bps, rtc::Optional< int64_t > probing_interval_ms)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD2(OnReceivedUplinkBandwidth, void(int target_audio_bitrate_bps, rtc::Optional< int64_t > probing_interval_ms)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD3(EncodeImpl, EncodedInfo(uint32_t timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded)) | webrtc::MockAudioEncoder | |
| MOCK_METHOD3(EncodeImpl, EncodedInfo(uint32_t timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded)) | webrtc::MockAudioEncoder | |
| Num10MsFramesInNextPacket() const =0 | webrtc::AudioEncoder | pure virtual |
| Num10MsFramesInNextPacket() const =0 | webrtc::AudioEncoder | pure virtual |
| NumChannels() const =0 | webrtc::AudioEncoder | pure virtual |
| NumChannels() const =0 | webrtc::AudioEncoder | pure virtual |
| OnReceivedOverhead(size_t overhead_bytes_per_packet) | webrtc::AudioEncoder | virtual |
| OnReceivedOverhead(size_t overhead_bytes_per_packet) | webrtc::AudioEncoder | virtual |
| OnReceivedRtt(int rtt_ms) | webrtc::AudioEncoder | virtual |
| OnReceivedRtt(int rtt_ms) | webrtc::AudioEncoder | virtual |
| OnReceivedTargetAudioBitrate(int target_bps) | webrtc::AudioEncoder | virtual |
| OnReceivedTargetAudioBitrate(int target_bps) | webrtc::AudioEncoder | virtual |
| OnReceivedUplinkBandwidth(int target_audio_bitrate_bps, rtc::Optional< int64_t > probing_interval_ms) | webrtc::AudioEncoder | virtual |
| OnReceivedUplinkBandwidth(int target_audio_bitrate_bps, rtc::Optional< int64_t > probing_interval_ms) | webrtc::AudioEncoder | virtual |
| OnReceivedUplinkPacketLossFraction(float uplink_packet_loss_fraction) | webrtc::AudioEncoder | virtual |
| OnReceivedUplinkPacketLossFraction(float uplink_packet_loss_fraction) | webrtc::AudioEncoder | virtual |
| ReclaimContainedEncoders() | webrtc::AudioEncoder | virtual |
| ReclaimContainedEncoders() | webrtc::AudioEncoder | virtual |
| Reset()=0 | webrtc::AudioEncoder | pure virtual |
| Reset()=0 | webrtc::AudioEncoder | pure virtual |
| RtpTimestampRateHz() const | webrtc::AudioEncoder | virtual |
| RtpTimestampRateHz() const | webrtc::AudioEncoder | virtual |
| SampleRateHz() const =0 | webrtc::AudioEncoder | pure virtual |
| SampleRateHz() const =0 | webrtc::AudioEncoder | pure virtual |
| SetApplication(Application application) | webrtc::AudioEncoder | virtual |
| SetApplication(Application application) | webrtc::AudioEncoder | virtual |
| SetDtx(bool enable) | webrtc::AudioEncoder | virtual |
| SetDtx(bool enable) | webrtc::AudioEncoder | virtual |
| SetFec(bool enable) | webrtc::AudioEncoder | virtual |
| SetFec(bool enable) | webrtc::AudioEncoder | virtual |
| SetMaxPlaybackRate(int frequency_hz) | webrtc::AudioEncoder | virtual |
| SetMaxPlaybackRate(int frequency_hz) | webrtc::AudioEncoder | virtual |
| SetReceiverFrameLengthRange(int min_frame_length_ms, int max_frame_length_ms) | webrtc::AudioEncoder | virtual |
| SetReceiverFrameLengthRange(int min_frame_length_ms, int max_frame_length_ms) | webrtc::AudioEncoder | virtual |
| SetTargetBitrate(int target_bps) | webrtc::AudioEncoder | virtual |
| SetTargetBitrate(int target_bps) | webrtc::AudioEncoder | virtual |
| ~AudioEncoder()=default | webrtc::AudioEncoder | virtual |
| ~AudioEncoder()=default | webrtc::AudioEncoder | virtual |
| ~MockAudioEncoder() | webrtc::MockAudioEncoder | inline |
| ~MockAudioEncoder() | webrtc::MockAudioEncoder | inline |
