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webkit
2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
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This is the complete list of members for webrtc::Channel, including all inherited members.
| BitRate() | webrtc::Channel | |
| BitRate() | webrtc::Channel | |
| Channel(int16_t chID=-1) | webrtc::Channel | |
| Channel(int16_t chID=-1) | webrtc::Channel | |
| LastInTimestamp() | webrtc::Channel | |
| LastInTimestamp() | webrtc::Channel | |
| PrintStats(CodecInst &codecInst) | webrtc::Channel | |
| PrintStats(CodecInst &codecInst) | webrtc::Channel | |
| RegisterReceiverACM(AudioCodingModule *acm) | webrtc::Channel | |
| RegisterReceiverACM(AudioCodingModule *acm) | webrtc::Channel | |
| ResetStats() | webrtc::Channel | |
| ResetStats() | webrtc::Channel | |
| SendData(FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t *payloadData, size_t payloadSize, const RTPFragmentationHeader *fragmentation) override | webrtc::Channel | virtual |
| SendData(FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t *payloadData, size_t payloadSize, const RTPFragmentationHeader *fragmentation) override | webrtc::Channel | virtual |
| set_num_packets_to_drop(int new_num_packets_to_drop) | webrtc::Channel | inline |
| set_num_packets_to_drop(int new_num_packets_to_drop) | webrtc::Channel | inline |
| set_send_timestamp(uint32_t new_send_ts) | webrtc::Channel | inline |
| set_send_timestamp(uint32_t new_send_ts) | webrtc::Channel | inline |
| set_sequence_number(uint16_t new_sequence_number) | webrtc::Channel | inline |
| set_sequence_number(uint16_t new_sequence_number) | webrtc::Channel | inline |
| SetFECTestWithPacketLoss(bool usePacketLoss) | webrtc::Channel | inline |
| SetFECTestWithPacketLoss(bool usePacketLoss) | webrtc::Channel | inline |
| SetIsStereo(bool isStereo) | webrtc::Channel | inline |
| SetIsStereo(bool isStereo) | webrtc::Channel | inline |
| Stats(CodecInst &codecInst, ACMTestPayloadStats &payloadStats) | webrtc::Channel | |
| Stats(uint32_t *numPackets) | webrtc::Channel | |
| Stats(uint8_t *payloadType, uint32_t *payloadLenByte) | webrtc::Channel | |
| Stats(CodecInst &codecInst, ACMTestPayloadStats &payloadStats) | webrtc::Channel | |
| Stats(uint32_t *numPackets) | webrtc::Channel | |
| Stats(uint8_t *payloadType, uint32_t *payloadLenByte) | webrtc::Channel | |
| ~AudioPacketizationCallback() | webrtc::AudioPacketizationCallback | inlinevirtual |
| ~AudioPacketizationCallback() | webrtc::AudioPacketizationCallback | inlinevirtual |
| ~Channel() override | webrtc::Channel | |
| ~Channel() override | webrtc::Channel |
1.8.13