#include <Channel.h>
◆ frameSizeSample
uint16_t webrtc::ACMTestFrameSizeStats::frameSizeSample |
◆ maxPayloadLen
size_t webrtc::ACMTestFrameSizeStats::maxPayloadLen |
◆ numPackets
uint32_t webrtc::ACMTestFrameSizeStats::numPackets |
◆ rateBitPerSec
double webrtc::ACMTestFrameSizeStats::rateBitPerSec |
◆ totalEncodedSamples
uint64_t webrtc::ACMTestFrameSizeStats::totalEncodedSamples |
◆ totalPayloadLenByte
uint64_t webrtc::ACMTestFrameSizeStats::totalPayloadLenByte |
◆ usageLenSec
double webrtc::ACMTestFrameSizeStats::usageLenSec |
The documentation for this struct was generated from the following file:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_coding/test/Channel.h