#include <EncodeDecodeTest.h>
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| TestPacketization (RTPStream *rtpStream, uint16_t frequency) |
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| ~TestPacketization () |
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int32_t | SendData (const FrameType frameType, const uint8_t payloadType, const uint32_t timeStamp, const uint8_t *payloadData, const size_t payloadSize, const RTPFragmentationHeader *fragmentation) override |
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| TestPacketization (RTPStream *rtpStream, uint16_t frequency) |
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| ~TestPacketization () |
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int32_t | SendData (const FrameType frameType, const uint8_t payloadType, const uint32_t timeStamp, const uint8_t *payloadData, const size_t payloadSize, const RTPFragmentationHeader *fragmentation) override |
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virtual | ~AudioPacketizationCallback () |
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virtual | ~AudioPacketizationCallback () |
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◆ TestPacketization() [1/2]
webrtc::TestPacketization::TestPacketization |
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RTPStream * |
rtpStream, |
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uint16_t |
frequency |
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) |
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◆ ~TestPacketization() [1/2]
webrtc::TestPacketization::~TestPacketization |
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◆ TestPacketization() [2/2]
webrtc::TestPacketization::TestPacketization |
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RTPStream * |
rtpStream, |
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uint16_t |
frequency |
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) |
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◆ ~TestPacketization() [2/2]
webrtc::TestPacketization::~TestPacketization |
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◆ SendData() [1/2]
◆ SendData() [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/EncodeDecodeTest.h
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc