#include <RTPFile.h>
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virtual | ~RTPStream () |
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virtual void | Write (const uint8_t payloadType, const uint32_t timeStamp, const int16_t seqNo, const uint8_t *payloadData, const size_t payloadSize, uint32_t frequency)=0 |
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virtual size_t | Read (WebRtcRTPHeader *rtpInfo, uint8_t *payloadData, size_t payloadSize, uint32_t *offset)=0 |
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virtual bool | EndOfFile () const =0 |
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virtual | ~RTPStream () |
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virtual void | Write (const uint8_t payloadType, const uint32_t timeStamp, const int16_t seqNo, const uint8_t *payloadData, const size_t payloadSize, uint32_t frequency)=0 |
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virtual size_t | Read (WebRtcRTPHeader *rtpInfo, uint8_t *payloadData, size_t payloadSize, uint32_t *offset)=0 |
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virtual bool | EndOfFile () const =0 |
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void | MakeRTPheader (uint8_t *rtpHeader, uint8_t payloadType, int16_t seqNo, uint32_t timeStamp, uint32_t ssrc) |
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void | ParseRTPHeader (WebRtcRTPHeader *rtpInfo, const uint8_t *rtpHeader) |
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void | MakeRTPheader (uint8_t *rtpHeader, uint8_t payloadType, int16_t seqNo, uint32_t timeStamp, uint32_t ssrc) |
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void | ParseRTPHeader (WebRtcRTPHeader *rtpInfo, const uint8_t *rtpHeader) |
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◆ ~RTPStream() [1/2]
virtual webrtc::RTPStream::~RTPStream |
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inlinevirtual |
◆ ~RTPStream() [2/2]
virtual webrtc::RTPStream::~RTPStream |
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inlinevirtual |
◆ EndOfFile() [1/2]
virtual bool webrtc::RTPStream::EndOfFile |
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const |
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pure virtual |
◆ EndOfFile() [2/2]
virtual bool webrtc::RTPStream::EndOfFile |
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const |
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pure virtual |
◆ MakeRTPheader() [1/2]
◆ MakeRTPheader() [2/2]
◆ ParseRTPHeader() [1/2]
◆ ParseRTPHeader() [2/2]
◆ Read() [1/2]
◆ Read() [2/2]
◆ Write() [1/2]
◆ Write() [2/2]
The documentation for this class was generated from the following files:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_coding/test/RTPFile.h
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/RTPFile.cc