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webkit
2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
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This is the complete list of members for webrtc::AudioCodingModule, including all inherited members.
| Add10MsData(const AudioFrame &audio_frame)=0 | webrtc::AudioCodingModule | pure virtual |
| Add10MsData(const AudioFrame &audio_frame)=0 | webrtc::AudioCodingModule | pure virtual |
| AudioCodingModule() | webrtc::AudioCodingModule | inlineprotected |
| AudioCodingModule() | webrtc::AudioCodingModule | inlineprotected |
| Codec(int list_id, CodecInst *codec) | webrtc::AudioCodingModule | static |
| Codec(const char *payload_name, CodecInst *codec, int sampling_freq_hz, size_t channels) | webrtc::AudioCodingModule | static |
| Codec(const char *payload_name, int sampling_freq_hz, size_t channels) | webrtc::AudioCodingModule | static |
| Codec(int list_id, CodecInst *codec) | webrtc::AudioCodingModule | static |
| Codec(const char *payload_name, CodecInst *codec, int sampling_freq_hz, size_t channels) | webrtc::AudioCodingModule | static |
| Codec(const char *payload_name, int sampling_freq_hz, size_t channels) | webrtc::AudioCodingModule | static |
| CodecFEC() const =0 | webrtc::AudioCodingModule | pure virtual |
| CodecFEC() const =0 | webrtc::AudioCodingModule | pure virtual |
| Create(int id) | webrtc::AudioCodingModule | static |
| Create(int id, Clock *clock) | webrtc::AudioCodingModule | static |
| Create(const Config &config) | webrtc::AudioCodingModule | static |
| Create(int id) | webrtc::AudioCodingModule | static |
| Create(int id, Clock *clock) | webrtc::AudioCodingModule | static |
| Create(const Config &config) | webrtc::AudioCodingModule | static |
| DisableNack()=0 | webrtc::AudioCodingModule | pure virtual |
| DisableNack()=0 | webrtc::AudioCodingModule | pure virtual |
| DisableOpusDtx()=0 | webrtc::AudioCodingModule | pure virtual |
| DisableOpusDtx()=0 | webrtc::AudioCodingModule | pure virtual |
| EnableNack(size_t max_nack_list_size)=0 | webrtc::AudioCodingModule | pure virtual |
| EnableNack(size_t max_nack_list_size)=0 | webrtc::AudioCodingModule | pure virtual |
| EnableOpusDtx()=0 | webrtc::AudioCodingModule | pure virtual |
| EnableOpusDtx()=0 | webrtc::AudioCodingModule | pure virtual |
| FilteredCurrentDelayMs() const =0 | webrtc::AudioCodingModule | pure virtual |
| FilteredCurrentDelayMs() const =0 | webrtc::AudioCodingModule | pure virtual |
| GetDecodingCallStatistics(AudioDecodingCallStats *call_stats) const =0 | webrtc::AudioCodingModule | pure virtual |
| GetDecodingCallStatistics(AudioDecodingCallStats *call_stats) const =0 | webrtc::AudioCodingModule | pure virtual |
| GetNackList(int64_t round_trip_time_ms) const =0 | webrtc::AudioCodingModule | pure virtual |
| GetNackList(int64_t round_trip_time_ms) const =0 | webrtc::AudioCodingModule | pure virtual |
| GetNetworkStatistics(NetworkStatistics *network_statistics)=0 | webrtc::AudioCodingModule | pure virtual |
| GetNetworkStatistics(NetworkStatistics *network_statistics)=0 | webrtc::AudioCodingModule | pure virtual |
| IncomingPacket(const uint8_t *incoming_payload, const size_t payload_len_bytes, const WebRtcRTPHeader &rtp_info)=0 | webrtc::AudioCodingModule | pure virtual |
| IncomingPacket(const uint8_t *incoming_payload, const size_t payload_len_bytes, const WebRtcRTPHeader &rtp_info)=0 | webrtc::AudioCodingModule | pure virtual |
| IncomingPayload(const uint8_t *incoming_payload, const size_t payload_len_byte, const uint8_t payload_type, const uint32_t timestamp=0)=0 | webrtc::AudioCodingModule | pure virtual |
| IncomingPayload(const uint8_t *incoming_payload, const size_t payload_len_byte, const uint8_t payload_type, const uint32_t timestamp=0)=0 | webrtc::AudioCodingModule | pure virtual |
| InitializeReceiver()=0 | webrtc::AudioCodingModule | pure virtual |
| InitializeReceiver()=0 | webrtc::AudioCodingModule | pure virtual |
| IsCodecValid(const CodecInst &codec) | webrtc::AudioCodingModule | static |
| IsCodecValid(const CodecInst &codec) | webrtc::AudioCodingModule | static |
| LeastRequiredDelayMs() const =0 | webrtc::AudioCodingModule | pure virtual |
| LeastRequiredDelayMs() const =0 | webrtc::AudioCodingModule | pure virtual |
| ModifyEncoder(rtc::FunctionView< void(std::unique_ptr< AudioEncoder > *)> modifier)=0 | webrtc::AudioCodingModule | pure virtual |
| ModifyEncoder(rtc::FunctionView< void(std::unique_ptr< AudioEncoder > *)> modifier)=0 | webrtc::AudioCodingModule | pure virtual |
| NumberOfCodecs() | webrtc::AudioCodingModule | static |
| NumberOfCodecs() | webrtc::AudioCodingModule | static |
| PlayoutData10Ms(int32_t desired_freq_hz, AudioFrame *audio_frame, bool *muted)=0 | webrtc::AudioCodingModule | pure virtual |
| PlayoutData10Ms(int32_t desired_freq_hz, AudioFrame *audio_frame)=0 | webrtc::AudioCodingModule | pure virtual |
| PlayoutData10Ms(int32_t desired_freq_hz, AudioFrame *audio_frame, bool *muted)=0 | webrtc::AudioCodingModule | pure virtual |
| PlayoutData10Ms(int32_t desired_freq_hz, AudioFrame *audio_frame)=0 | webrtc::AudioCodingModule | pure virtual |
| PlayoutFrequency() const =0 | webrtc::AudioCodingModule | pure virtual |
| PlayoutFrequency() const =0 | webrtc::AudioCodingModule | pure virtual |
| PlayoutTimestamp(uint32_t *timestamp)=0 | webrtc::AudioCodingModule | pure virtual |
| PlayoutTimestamp()=0 | webrtc::AudioCodingModule | pure virtual |
| PlayoutTimestamp(uint32_t *timestamp)=0 | webrtc::AudioCodingModule | pure virtual |
| PlayoutTimestamp()=0 | webrtc::AudioCodingModule | pure virtual |
| QueryEncoder(rtc::FunctionView< void(AudioEncoder const *)> query)=0 | webrtc::AudioCodingModule | pure virtual |
| QueryEncoder(rtc::FunctionView< void(AudioEncoder const *)> query)=0 | webrtc::AudioCodingModule | pure virtual |
| ReceiveCodec(CodecInst *curr_receive_codec) const =0 | webrtc::AudioCodingModule | pure virtual |
| ReceiveCodec(CodecInst *curr_receive_codec) const =0 | webrtc::AudioCodingModule | pure virtual |
| ReceiveFormat() const =0 | webrtc::AudioCodingModule | pure virtual |
| ReceiveFormat() const =0 | webrtc::AudioCodingModule | pure virtual |
| ReceiveFrequency() const =0 | webrtc::AudioCodingModule | pure virtual |
| ReceiveFrequency() const =0 | webrtc::AudioCodingModule | pure virtual |
| REDStatus() const =0 | webrtc::AudioCodingModule | pure virtual |
| REDStatus() const =0 | webrtc::AudioCodingModule | pure virtual |
| RegisterExternalReceiveCodec(int rtp_payload_type, AudioDecoder *external_decoder, int sample_rate_hz, int num_channels, const std::string &name)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterExternalReceiveCodec(int rtp_payload_type, AudioDecoder *external_decoder, int sample_rate_hz, int num_channels, const std::string &name)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterExternalSendCodec(AudioEncoder *external_speech_encoder)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterExternalSendCodec(AudioEncoder *external_speech_encoder)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterReceiveCodec(int rtp_payload_type, const SdpAudioFormat &audio_format)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterReceiveCodec(const CodecInst &receive_codec)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterReceiveCodec(const CodecInst &receive_codec, rtc::FunctionView< std::unique_ptr< AudioDecoder >()> isac_factory)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterReceiveCodec(int rtp_payload_type, const SdpAudioFormat &audio_format)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterReceiveCodec(const CodecInst &receive_codec)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterReceiveCodec(const CodecInst &receive_codec, rtc::FunctionView< std::unique_ptr< AudioDecoder >()> isac_factory)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterSendCodec(const CodecInst &send_codec)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterSendCodec(const CodecInst &send_codec)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterTransportCallback(AudioPacketizationCallback *transport)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterTransportCallback(AudioPacketizationCallback *transport)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterVADCallback(ACMVADCallback *vad_callback)=0 | webrtc::AudioCodingModule | pure virtual |
| RegisterVADCallback(ACMVADCallback *vad_callback)=0 | webrtc::AudioCodingModule | pure virtual |
| SendCodec() const =0 | webrtc::AudioCodingModule | pure virtual |
| SendCodec() const =0 | webrtc::AudioCodingModule | pure virtual |
| SendFrequency() const =0 | webrtc::AudioCodingModule | pure virtual |
| SendFrequency() const =0 | webrtc::AudioCodingModule | pure virtual |
| SetBitRate(int bitrate_bps)=0 | webrtc::AudioCodingModule | pure virtual |
| SetBitRate(int bitrate_bps)=0 | webrtc::AudioCodingModule | pure virtual |
| SetCodecFEC(bool enable_codec_fec)=0 | webrtc::AudioCodingModule | pure virtual |
| SetCodecFEC(bool enable_codec_fec)=0 | webrtc::AudioCodingModule | pure virtual |
| SetEncoder(std::unique_ptr< AudioEncoder > new_encoder) | webrtc::AudioCodingModule | inline |
| SetEncoder(std::unique_ptr< AudioEncoder > new_encoder) | webrtc::AudioCodingModule | inline |
| SetMaximumPlayoutDelay(int time_ms)=0 | webrtc::AudioCodingModule | pure virtual |
| SetMaximumPlayoutDelay(int time_ms)=0 | webrtc::AudioCodingModule | pure virtual |
| SetMinimumPlayoutDelay(int time_ms)=0 | webrtc::AudioCodingModule | pure virtual |
| SetMinimumPlayoutDelay(int time_ms)=0 | webrtc::AudioCodingModule | pure virtual |
| SetOpusApplication(OpusApplicationMode application)=0 | webrtc::AudioCodingModule | pure virtual |
| SetOpusApplication(OpusApplicationMode application)=0 | webrtc::AudioCodingModule | pure virtual |
| SetOpusMaxPlaybackRate(int frequency_hz)=0 | webrtc::AudioCodingModule | pure virtual |
| SetOpusMaxPlaybackRate(int frequency_hz)=0 | webrtc::AudioCodingModule | pure virtual |
| SetPacketLossRate(int packet_loss_rate)=0 | webrtc::AudioCodingModule | pure virtual |
| SetPacketLossRate(int packet_loss_rate)=0 | webrtc::AudioCodingModule | pure virtual |
| SetREDStatus(bool enable_red)=0 | webrtc::AudioCodingModule | pure virtual |
| SetREDStatus(bool enable_red)=0 | webrtc::AudioCodingModule | pure virtual |
| SetVAD(const bool enable_dtx=true, const bool enable_vad=false, const ACMVADMode vad_mode=VADNormal)=0 | webrtc::AudioCodingModule | pure virtual |
| SetVAD(const bool enable_dtx=true, const bool enable_vad=false, const ACMVADMode vad_mode=VADNormal)=0 | webrtc::AudioCodingModule | pure virtual |
| UnregisterReceiveCodec(uint8_t payload_type)=0 | webrtc::AudioCodingModule | pure virtual |
| UnregisterReceiveCodec(uint8_t payload_type)=0 | webrtc::AudioCodingModule | pure virtual |
| VAD(bool *dtx_enabled, bool *vad_enabled, ACMVADMode *vad_mode) const =0 | webrtc::AudioCodingModule | pure virtual |
| VAD(bool *dtx_enabled, bool *vad_enabled, ACMVADMode *vad_mode) const =0 | webrtc::AudioCodingModule | pure virtual |
| ~AudioCodingModule()=default | webrtc::AudioCodingModule | virtual |
| ~AudioCodingModule()=default | webrtc::AudioCodingModule | virtual |
1.8.13