#include <EncodeDecodeTest.h>
◆ Receiver() [1/2]
◆ ~Receiver() [1/2]
virtual webrtc::Receiver::~Receiver |
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inlinevirtual |
◆ Receiver() [2/2]
webrtc::Receiver::Receiver |
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◆ ~Receiver() [2/2]
virtual webrtc::Receiver::~Receiver |
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inlinevirtual |
◆ IncomingPacket() [1/2]
bool Receiver::IncomingPacket |
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virtual |
◆ IncomingPacket() [2/2]
virtual bool webrtc::Receiver::IncomingPacket |
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virtual |
◆ PlayoutData() [1/2]
bool Receiver::PlayoutData |
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◆ PlayoutData() [2/2]
bool webrtc::Receiver::PlayoutData |
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◆ Run() [1/2]
◆ Run() [2/2]
void webrtc::Receiver::Run |
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◆ Setup() [1/2]
◆ Setup() [2/2]
◆ Teardown() [1/2]
void Receiver::Teardown |
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◆ Teardown() [2/2]
void webrtc::Receiver::Teardown |
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◆ _acm
◆ _incomingPayload
◆ _nextTime
◆ _payloadSizeBytes
size_t Receiver::_payloadSizeBytes |
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◆ _realPayloadSizeBytes
size_t Receiver::_realPayloadSizeBytes |
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◆ _rtpInfo
◆ _rtpStream
◆ codeId
◆ testMode
The documentation for this class was generated from the following files:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_coding/test/EncodeDecodeTest.h
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc