#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