#include <rtp_receiver.h>
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static RtpReceiver * | CreateVideoReceiver (Clock *clock, RtpData *incoming_payload_callback, RtpFeedback *incoming_messages_callback, RTPPayloadRegistry *rtp_payload_registry) |
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static RtpReceiver * | CreateAudioReceiver (Clock *clock, RtpData *incoming_payload_callback, RtpFeedback *incoming_messages_callback, RTPPayloadRegistry *rtp_payload_registry) |
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static RtpReceiver * | CreateVideoReceiver (Clock *clock, RtpData *incoming_payload_callback, RtpFeedback *incoming_messages_callback, RTPPayloadRegistry *rtp_payload_registry) |
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static RtpReceiver * | CreateAudioReceiver (Clock *clock, RtpData *incoming_payload_callback, RtpFeedback *incoming_messages_callback, RTPPayloadRegistry *rtp_payload_registry) |
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◆ ~RtpReceiver() [1/2]
virtual webrtc::RtpReceiver::~RtpReceiver |
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inlinevirtual |
◆ ~RtpReceiver() [2/2]
virtual webrtc::RtpReceiver::~RtpReceiver |
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inlinevirtual |
◆ CreateAudioReceiver() [1/2]
◆ CreateAudioReceiver() [2/2]
◆ CreateVideoReceiver() [1/2]
◆ CreateVideoReceiver() [2/2]
◆ CSRCs() [1/2]
virtual int32_t webrtc::RtpReceiver::CSRCs |
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uint32_t |
array_of_csrc[kRtpCsrcSize] | ) |
const |
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pure virtual |
◆ CSRCs() [2/2]
virtual int32_t webrtc::RtpReceiver::CSRCs |
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uint32_t |
array_of_csrc[kRtpCsrcSize] | ) |
const |
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pure virtual |
◆ DeRegisterReceivePayload() [1/2]
virtual int32_t webrtc::RtpReceiver::DeRegisterReceivePayload |
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const int8_t |
payload_type | ) |
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pure virtual |
◆ DeRegisterReceivePayload() [2/2]
virtual int32_t webrtc::RtpReceiver::DeRegisterReceivePayload |
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const int8_t |
payload_type | ) |
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pure virtual |
◆ Energy() [1/2]
virtual int32_t webrtc::RtpReceiver::Energy |
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uint8_t |
array_of_energy[kRtpCsrcSize] | ) |
const |
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pure virtual |
◆ Energy() [2/2]
virtual int32_t webrtc::RtpReceiver::Energy |
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uint8_t |
array_of_energy[kRtpCsrcSize] | ) |
const |
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pure virtual |
◆ GetTelephoneEventHandler() [1/2]
◆ GetTelephoneEventHandler() [2/2]
◆ IncomingRtpPacket() [1/2]
virtual bool webrtc::RtpReceiver::IncomingRtpPacket |
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const RTPHeader & |
rtp_header, |
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const uint8_t * |
payload, |
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size_t |
payload_length, |
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PayloadUnion |
payload_specific, |
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bool |
in_order |
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) |
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pure virtual |
◆ IncomingRtpPacket() [2/2]
virtual bool webrtc::RtpReceiver::IncomingRtpPacket |
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const RTPHeader & |
rtp_header, |
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const uint8_t * |
payload, |
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size_t |
payload_length, |
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PayloadUnion |
payload_specific, |
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bool |
in_order |
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) |
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pure virtual |
◆ LastReceivedTimeMs() [1/2]
virtual bool webrtc::RtpReceiver::LastReceivedTimeMs |
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int64_t * |
receive_time_ms | ) |
const |
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pure virtual |
◆ LastReceivedTimeMs() [2/2]
virtual bool webrtc::RtpReceiver::LastReceivedTimeMs |
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int64_t * |
receive_time_ms | ) |
const |
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pure virtual |
◆ RegisterReceivePayload() [1/4]
virtual int32_t webrtc::RtpReceiver::RegisterReceivePayload |
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const CodecInst & |
audio_codec | ) |
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pure virtual |
◆ RegisterReceivePayload() [2/4]
virtual int32_t webrtc::RtpReceiver::RegisterReceivePayload |
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const CodecInst & |
audio_codec | ) |
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pure virtual |
◆ RegisterReceivePayload() [3/4]
virtual int32_t webrtc::RtpReceiver::RegisterReceivePayload |
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const VideoCodec & |
video_codec | ) |
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pure virtual |
◆ RegisterReceivePayload() [4/4]
virtual int32_t webrtc::RtpReceiver::RegisterReceivePayload |
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const VideoCodec & |
video_codec | ) |
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pure virtual |
◆ SSRC() [1/2]
virtual uint32_t webrtc::RtpReceiver::SSRC |
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const |
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pure virtual |
◆ SSRC() [2/2]
virtual uint32_t webrtc::RtpReceiver::SSRC |
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const |
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pure virtual |
◆ Timestamp() [1/2]
virtual bool webrtc::RtpReceiver::Timestamp |
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uint32_t * |
timestamp | ) |
const |
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pure virtual |
◆ Timestamp() [2/2]
virtual bool webrtc::RtpReceiver::Timestamp |
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uint32_t * |
timestamp | ) |
const |
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pure virtual |
The documentation for this class was generated from the following files:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc