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RtcpContext修改时间戳单位、整理WebRTC相关代码

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This commit is contained in:
ziyue
2021-06-25 14:59:27 +08:00
parent 6c01cf336e
commit 964cf39145
8 changed files with 49 additions and 52 deletions
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6
src/Rtcp/RtcpContext.cpp
View File

@@ -18,8 +18,7 @@ void RtcpContext::clear() {
memset(this, 0, sizeof(RtcpContext));
}
RtcpContext::RtcpContext(uint32_t sample_rate, bool is_receiver) {
_sample_rate = sample_rate;
RtcpContext::RtcpContext(bool is_receiver) {
_is_receiver = is_receiver;
}
@@ -35,7 +34,6 @@ void RtcpContext::onRtp(uint16_t seq, uint32_t stamp, size_t bytes) {
diff = -diff;
}
//抖动单位为采样次数
diff *= (_sample_rate / 1000.0);
_jitter += (diff - _jitter) / 16.0;
} else {
_jitter = 0;
@@ -129,7 +127,7 @@ Buffer::Ptr RtcpContext::createRtcpSR(uint32_t rtcp_ssrc) {
rtcp->setNtpStamp(tv);
//转换成rtp时间戳
rtcp->rtpts = htonl(uint32_t(_last_rtp_stamp * (_sample_rate / 1000.0)));
rtcp->rtpts = htonl(_last_rtp_stamp);
rtcp->packet_count = htonl((uint32_t) _packets);
rtcp->octet_count = htonl((uint32_t) _bytes);
return RtcpHeader::toBuffer(std::move(rtcp));

7
src/Rtcp/RtcpContext.h
View File

@@ -22,15 +22,14 @@ public:
using Ptr = std::shared_ptr<RtcpContext>;
/**
* 创建rtcp上下文
* @param sample_rate 音频采用率，视频一般为90000
* @param is_receiver 是否为rtp接收者，接收者更消耗性能
*/
RtcpContext(uint32_t sample_rate, bool is_receiver);
RtcpContext(bool is_receiver);
/**
* 输出或输入rtp时调用
* @param seq rtp的seq
* @param stamp rtp的时间戳，单位毫秒
* @param stamp rtp的时间戳，单位采样数(非毫秒)
* @param bytes rtp数据长度
*/
void onRtp(uint16_t seq, uint32_t stamp, size_t bytes);
@@ -87,8 +86,6 @@ private:
bool _is_receiver;
//时间戳抖动值
double _jitter = 0;
//视频默认90000,音频为采样率
uint32_t _sample_rate;
//收到或发送的rtp的字节数
size_t _bytes = 0;
//收到或发送的rtp的个数

4
src/Rtp/RtpServer.cpp
View File

@@ -27,7 +27,7 @@ class RtcpHelper : public RtcpContext, public std::enable_shared_from_this<RtcpH
public:
using Ptr = std::shared_ptr<RtcpHelper>;
RtcpHelper(Socket::Ptr rtcp_sock, uint32_t sample_rate) : RtcpContext(sample_rate, true){
RtcpHelper(Socket::Ptr rtcp_sock, uint32_t sample_rate) : RtcpContext(true){
_rtcp_sock = std::move(rtcp_sock);
_sample_rate = sample_rate;
}
@@ -35,7 +35,7 @@ public:
void onRecvRtp(const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len){
//统计rtp接受情况，用于发送rr包
auto header = (RtpHeader *) buf->data();
onRtp(ntohs(header->seq), ntohl(header->stamp) * uint64_t(1000) / _sample_rate, buf->size());
onRtp(ntohs(header->seq), ntohl(header->stamp), buf->size());
sendRtcp(ntohl(header->ssrc), addr, addr_len);
}

4
src/Rtsp/RtspPlayer.cpp
View File

@@ -205,7 +205,7 @@ void RtspPlayer::handleResDESCRIBE(const Parser& parser) {
}
_rtcp_context.clear();
for (auto &track : _sdp_track) {
_rtcp_context.emplace_back(std::make_shared<RtcpContext>(track->_samplerate, true));
_rtcp_context.emplace_back(std::make_shared<RtcpContext>(true));
}
sendSetup(0);
}
@@ -591,7 +591,7 @@ void RtspPlayer::sendRtspRequest(const string &cmd, const string &url,const StrC
void RtspPlayer::onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_idx){
auto &rtcp_ctx = _rtcp_context[track_idx];
rtcp_ctx->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
rtcp_ctx->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
auto &ticker = _rtcp_send_ticker[track_idx];
if (ticker.elapsedTime() < 3 * 1000) {

4
src/Rtsp/RtspPusher.cpp
View File

@@ -179,7 +179,7 @@ void RtspPusher::sendAnnounce() {
}
_rtcp_context.clear();
for (auto &track : _track_vec) {
_rtcp_context.emplace_back(std::make_shared<RtcpContext>(track->_samplerate, false));
_rtcp_context.emplace_back(std::make_shared<RtcpContext>(false));
}
_on_res_func = std::bind(&RtspPusher::handleResAnnounce, this, placeholders::_1);
sendRtspRequest("ANNOUNCE", _url, {}, src->getSdp());
@@ -360,7 +360,7 @@ void RtspPusher::updateRtcpContext(const RtpPacket::Ptr &rtp){
int track_index = getTrackIndexByTrackType(rtp->type);
auto &ticker = _rtcp_send_ticker[track_index];
auto &rtcp_ctx = _rtcp_context[track_index];
rtcp_ctx->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
rtcp_ctx->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
//send rtcp every 5 second
if (ticker.elapsedTime() > 5 * 1000) {

6
src/Rtsp/RtspSession.cpp
View File

@@ -252,7 +252,7 @@ void RtspSession::handleReq_ANNOUNCE(const Parser &parser) {
}
_rtcp_context.clear();
for (auto &track : _sdp_track) {
_rtcp_context.emplace_back(std::make_shared<RtcpContext>(track->_samplerate, true));
_rtcp_context.emplace_back(std::make_shared<RtcpContext>(true));
}
_push_src = std::make_shared<RtspMediaSourceImp>(_media_info._vhost, _media_info._app, _media_info._streamid);
_push_src->setListener(dynamic_pointer_cast<MediaSourceEvent>(shared_from_this()));
@@ -413,7 +413,7 @@ void RtspSession::onAuthSuccess() {
}
strongSelf->_rtcp_context.clear();
for (auto &track : strongSelf->_sdp_track) {
strongSelf->_rtcp_context.emplace_back(std::make_shared<RtcpContext>(track->_samplerate, false));
strongSelf->_rtcp_context.emplace_back(std::make_shared<RtcpContext>(false));
}
strongSelf->_sessionid = makeRandStr(12);
strongSelf->_play_src = rtsp_src;
@@ -1126,7 +1126,7 @@ void RtspSession::onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index){
void RtspSession::updateRtcpContext(const RtpPacket::Ptr &rtp){
int track_index = getTrackIndexByTrackType(rtp->type);
auto &rtcp_ctx = _rtcp_context[track_index];
rtcp_ctx->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
rtcp_ctx->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
auto &ticker = _rtcp_send_tickers[track_index];
//send rtcp every 5 second