webkit  2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
webrtc::RTPSenderAudio Member List

This is the complete list of members for webrtc::RTPSenderAudio, including all inherited members.

MarkerBit(FrameType frame_type, int8_t payload_type)webrtc::RTPSenderAudioprotected
MarkerBit(FrameType frame_type, int8_t payload_type)webrtc::RTPSenderAudioprotected
RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE], int8_t payload_type, uint32_t frequency, size_t channels, uint32_t rate, RtpUtility::Payload **payload)webrtc::RTPSenderAudio
RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE], int8_t payload_type, uint32_t frequency, size_t channels, uint32_t rate, RtpUtility::Payload **payload)webrtc::RTPSenderAudio
RTPSenderAudio(Clock *clock, RTPSender *rtp_sender)webrtc::RTPSenderAudio
RTPSenderAudio(Clock *clock, RTPSender *rtp_sender)webrtc::RTPSenderAudio
SendAudio(FrameType frame_type, int8_t payload_type, uint32_t capture_timestamp, const uint8_t *payload_data, size_t payload_size, const RTPFragmentationHeader *fragmentation)webrtc::RTPSenderAudio
SendAudio(FrameType frame_type, int8_t payload_type, uint32_t capture_timestamp, const uint8_t *payload_data, size_t payload_size, const RTPFragmentationHeader *fragmentation)webrtc::RTPSenderAudio
SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level)webrtc::RTPSenderAudio
SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level)webrtc::RTPSenderAudio
SendTelephoneEventPacket(bool ended, uint32_t dtmf_timestamp, uint16_t duration, bool marker_bit)webrtc::RTPSenderAudioprotected
SendTelephoneEventPacket(bool ended, uint32_t dtmf_timestamp, uint16_t duration, bool marker_bit)webrtc::RTPSenderAudioprotected
SetAudioLevel(uint8_t level_dbov)webrtc::RTPSenderAudio
SetAudioLevel(uint8_t level_dbov)webrtc::RTPSenderAudio
~RTPSenderAudio()webrtc::RTPSenderAudio
~RTPSenderAudio()webrtc::RTPSenderAudio