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修改rtp pt时，确保不影响其他播放器

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This commit is contained in:
xiongziliang 2021-04-04 21:42:11 +08:00
parent 2abb5078f9
commit c70721a520
4 changed files with 10 additions and 13 deletions
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3
webrtc/SrtpSession.cpp
View File

@ -213,7 +213,7 @@ namespace RTC
}
}
bool SrtpSession::EncryptRtp(const uint8_t** data, size_t* len)
bool SrtpSession::EncryptRtp(const uint8_t** data, size_t* len, uint8_t pt)
{
MS_TRACE();
@ -226,6 +226,7 @@ namespace RTC
}
std::memcpy(EncryptBuffer, *data, *len);
EncryptBuffer[1] |= (pt & 0x7F);
srtp_err_status_t err =
srtp_protect(this->session, static_cast<void*>(EncryptBuffer), reinterpret_cast<int*>(len));

2
webrtc/SrtpSession.hpp
View File

@ -46,7 +46,7 @@ namespace RTC
~SrtpSession();
public:
bool EncryptRtp(const uint8_t** data, size_t* len);
bool EncryptRtp(const uint8_t** data, size_t* len, uint8_t pt);
bool DecryptSrtp(uint8_t* data, size_t* len);
bool EncryptRtcp(const uint8_t** data, size_t* len);
bool DecryptSrtcp(uint8_t* data, size_t* len);

16
webrtc/WebRtcTransport.cpp
View File

@ -179,11 +179,11 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *
}
}
void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush) {
void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt) {
const uint8_t *p = (uint8_t *) buf;
bool ret = false;
if (_srtp_session_send) {
ret = _srtp_session_send->EncryptRtp(&p, &len);
ret = _srtp_session_send->EncryptRtp(&p, &len, pt);
}
if (ret) {
onSendSockData((char *) p, len, flush);
@ -467,16 +467,12 @@ void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const Rtp
}
void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){
if (!_send_rtp_pt[rtp->type]) {
auto &pt = _send_rtp_pt[rtp->type];
if (!pt) {
//忽略，对方不支持该编码类型
return;
}
auto tmp = rtp->getHeader()->pt;
//设置pt
rtp->getHeader()->pt = _send_rtp_pt[rtp->type];
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush);
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, pt);
//统计rtp发送情况，好做sr汇报
_rtp_receiver[_send_rtp_pt[rtp->type]].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
//还原pt
rtp->getHeader()->pt = tmp;
_rtp_receiver[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
}

2
webrtc/WebRtcTransport.h
View File

@ -45,7 +45,7 @@ public:
* @param buf rtcp内容
* @param len rtcp长度
*/
void sendRtpPacket(char *buf, size_t len, bool flush);
void sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt);
void sendRtcpPacket(char *buf, size_t len, bool flush);
protected: