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完善发送rtx rtp处理

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This commit is contained in:
xiongziliang 2021-05-16 18:06:34 +08:00
parent 7cdd5ed9df
commit bda378c340
2 changed files with 5 additions and 1 deletions
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5
webrtc/WebRtcTransport.cpp
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@ -820,11 +820,14 @@ void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t &len, void *
if (pr->second->answer_ssrc_rtx) {
//有rtx单独的ssrc,有些情况下，浏览器支持rtx，但是未指定rtx单独的ssrc
header->ssrc = htonl(pr->second->answer_ssrc_rtx);
} else {
//未单独指定rtx的ssrc，那么使用rtp的ssrc
header->ssrc = htonl(pr->second->answer_ssrc_rtp);
}
auto origin_seq = ntohs(header->seq);
//seq跟原来的不一样
header->seq = htons(origin_seq + 100);
header->seq = htons(_rtx_seq[pr->second->media->type]++);
auto payload = header->getPayloadData();
auto payload_size = header->getPayloadSize(len);
if (payload_size) {

1
webrtc/WebRtcTransport.h
View File

@ -360,6 +360,7 @@ private:
void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack);
private:
uint16_t _rtx_seq[2] = {0, 0};
//用掉的总流量
uint64_t _bytes_usage = 0;
//媒体相关元数据