Commit 20f1275c by xiongziliang

rtsp拉流、rtp单端口推流新增支持获取丢包率: #1877

parent cd269672
......@@ -219,7 +219,7 @@ bool MediaSource::close(bool force) {
return listener->close(*this,force);
}
int MediaSource::getLossRate(mediakit::TrackType type) {
float MediaSource::getLossRate(mediakit::TrackType type) {
auto listener = _listener.lock();
if (!listener) {
return -1;
......@@ -720,7 +720,7 @@ void MediaSourceEventInterceptor::onRegist(MediaSource &sender, bool regist) {
}
}
int MediaSourceEventInterceptor::getLossRate(MediaSource &sender, TrackType type){
float MediaSourceEventInterceptor::getLossRate(MediaSource &sender, TrackType type){
auto listener = _listener.lock();
if (listener) {
return listener->getLossRate(sender, type);
......
......@@ -87,7 +87,7 @@ public:
//流注册或注销事件
virtual void onRegist(MediaSource &sender, bool regist) {};
// 获取丢包率
virtual int getLossRate(MediaSource &sender, TrackType type) { return -1; }
virtual float getLossRate(MediaSource &sender, TrackType type) { return -1; }
// 获取所在线程, 此函数一般强制重载
virtual toolkit::EventPoller::Ptr getOwnerPoller(MediaSource &sender) { throw NotImplemented(toolkit::demangle(typeid(*this).name()) + "::getOwnerPoller not implemented"); }
......@@ -164,7 +164,7 @@ public:
std::vector<Track::Ptr> getMediaTracks(MediaSource &sender, bool trackReady = true) const override;
void startSendRtp(MediaSource &sender, const SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb) override;
bool stopSendRtp(MediaSource &sender, const std::string &ssrc) override;
int getLossRate(MediaSource &sender, TrackType type) override;
float getLossRate(MediaSource &sender, TrackType type) override;
toolkit::EventPoller::Ptr getOwnerPoller(MediaSource &sender) override;
private:
......@@ -280,7 +280,7 @@ public:
// 停止发送ps-rtp
bool stopSendRtp(const std::string &ssrc);
// 获取丢包率
int getLossRate(mediakit::TrackType type);
float getLossRate(mediakit::TrackType type);
// 获取所在线程
toolkit::EventPoller::Ptr getOwnerPoller();
......
......@@ -89,7 +89,7 @@ public:
* 获取丢包率,只支持rtsp
* @param type 音频或视频,TrackInvalid时为总丢包率
*/
virtual float getPacketLossRate(TrackType type) const { return 0; };
virtual float getPacketLossRate(TrackType type) const { return -1; };
/**
* 获取所有track
......
......@@ -185,6 +185,10 @@ std::shared_ptr<SockInfo> PlayerProxy::getOriginSock(MediaSource &sender) const
return getSockInfo();
}
float PlayerProxy::getLossRate(MediaSource &sender, TrackType type) {
return getPacketLossRate(type);
}
void PlayerProxy::onPlaySuccess() {
GET_CONFIG(bool, reset_when_replay, General::kResetWhenRePlay);
if (dynamic_pointer_cast<RtspMediaSource>(_media_src)) {
......
......@@ -59,6 +59,7 @@ private:
MediaOriginType getOriginType(MediaSource &sender) const override;
std::string getOriginUrl(MediaSource &sender) const override;
std::shared_ptr<toolkit::SockInfo> getOriginSock(MediaSource &sender) const override;
float getLossRate(MediaSource &sender, TrackType type) override;
void rePlay(const std::string &strUrl,int iFailedCnt);
void onPlaySuccess();
......
......@@ -95,6 +95,9 @@ bool RtpProcess::inputRtp(bool is_udp, const Socket::Ptr &sock, const char *data
_process = std::make_shared<GB28181Process>(_media_info, this);
}
auto header = (RtpHeader *) data;
onRtp(ntohs(header->seq), ntohl(header->stamp), 0/*不发送sr,所以可以设置为0*/ , 90000/*ps/ts流时间戳按照90K采样率*/, len);
GET_CONFIG(string, dump_dir, RtpProxy::kDumpDir);
if (_muxer && !_muxer->isEnabled() && !dts_out && dump_dir.empty()) {
//无人访问、且不取时间戳、不导出调试文件时,我们可以直接丢弃数据
......@@ -280,20 +283,12 @@ toolkit::EventPoller::Ptr RtpProcess::getOwnerPoller(MediaSource &sender) {
return _sock ? _sock->getPoller() : EventPollerPool::Instance().getPoller();
}
void RtpProcess::setHelper(std::weak_ptr<RtcpContext> help) {
_help = std::move(help);
}
int RtpProcess::getLossRate(MediaSource &sender, TrackType type) {
auto help = _help.lock();
if (!help) {
return -1;
}
auto expected = help->getExpectedPacketsInterval();
float RtpProcess::getLossRate(MediaSource &sender, TrackType type) {
auto expected = getExpectedPacketsInterval();
if (!expected) {
return 0;
return -1;
}
return help->geLostInterval() * 100 / expected;
return geLostInterval() * 100 / expected;
}
}//namespace mediakit
......
......@@ -18,7 +18,7 @@
namespace mediakit {
class RtpProcess : public toolkit::SockInfo, public MediaSinkInterface, public MediaSourceEventInterceptor, public std::enable_shared_from_this<RtpProcess>{
class RtpProcess : public RtcpContextForRecv, public toolkit::SockInfo, public MediaSinkInterface, public MediaSourceEventInterceptor, public std::enable_shared_from_this<RtpProcess>{
public:
typedef std::shared_ptr<RtpProcess> Ptr;
friend class RtpProcessHelper;
......@@ -64,8 +64,6 @@ public:
uint16_t get_peer_port() override;
std::string getIdentifier() const override;
void setHelper(const std::weak_ptr<RtcpContext> help);
protected:
bool inputFrame(const Frame::Ptr &frame) override;
bool addTrack(const Track::Ptr & track) override;
......@@ -77,7 +75,7 @@ protected:
std::string getOriginUrl(MediaSource &sender) const override;
std::shared_ptr<SockInfo> getOriginSock(MediaSource &sender) const override;
toolkit::EventPoller::Ptr getOwnerPoller(MediaSource &sender) override;
int getLossRate(MediaSource &sender, TrackType type) override;
float getLossRate(MediaSource &sender, TrackType type) override;
private:
void emitOnPublish();
......@@ -100,7 +98,6 @@ private:
toolkit::Ticker _last_check_alive;
std::recursive_mutex _func_mtx;
std::deque<std::function<void()> > _cached_func;
std::weak_ptr<RtcpContext> _help;
};
}//namespace mediakit
......
......@@ -27,19 +27,18 @@ RtpServer::~RtpServer() {
}
}
class RtcpHelper : public RtcpContextForRecv, public std::enable_shared_from_this<RtcpHelper> {
class RtcpHelper: public std::enable_shared_from_this<RtcpHelper> {
public:
using Ptr = std::shared_ptr<RtcpHelper>;
RtcpHelper(Socket::Ptr rtcp_sock, uint32_t sample_rate) {
RtcpHelper(Socket::Ptr rtcp_sock, RtpProcess::Ptr process) {
_rtcp_sock = std::move(rtcp_sock);
_sample_rate = sample_rate;
_process = std::move(process);
}
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), 0/*不发送sr,所以可以设置为0*/ , _sample_rate, buf->size());
sendRtcp(ntohl(header->ssrc), addr, addr_len);
}
......@@ -58,7 +57,7 @@ public:
}
auto rtcps = RtcpHeader::loadFromBytes(buf->data(), buf->size());
for (auto &rtcp : rtcps) {
strong_self->onRtcp(rtcp);
strong_self->_process->onRtcp(rtcp);
}
});
}
......@@ -81,13 +80,13 @@ private:
//未收到rtcp打洞包时,采用默认的rtcp端口
rtcp_addr = addr;
}
_rtcp_sock->send(createRtcpRR(rtp_ssrc + 1, rtp_ssrc), rtcp_addr, addr_len);
_rtcp_sock->send(_process->createRtcpRR(rtp_ssrc + 1, rtp_ssrc), rtcp_addr, addr_len);
}
private:
Ticker _ticker;
Socket::Ptr _rtcp_sock;
uint32_t _sample_rate;
RtpProcess::Ptr _process;
std::shared_ptr<struct sockaddr_storage> _rtcp_addr;
};
......@@ -126,8 +125,7 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, bool enable_
if (!stream_id.empty()) {
//指定了流id,那么一个端口一个流(不管是否包含多个ssrc的多个流,绑定rtp源后,会筛选掉ip端口不匹配的流)
process = RtpSelector::Instance().getProcess(stream_id, true);
RtcpHelper::Ptr helper = std::make_shared<RtcpHelper>(std::move(rtcp_socket), 90000);
process->setHelper(helper);
RtcpHelper::Ptr helper = std::make_shared<RtcpHelper>(std::move(rtcp_socket), process);
helper->startRtcp();
rtp_socket->setOnRead([rtp_socket, process, helper, ssrc](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) {
RtpHeader *header = (RtpHeader *)buf->data();
......
......@@ -148,6 +148,6 @@ void WebRtcPusher::onRtcConfigure(RtcConfigure &configure) const {
configure.audio.direction = configure.video.direction = RtpDirection::recvonly;
}
int WebRtcPusher::getLossRate(MediaSource &sender,mediakit::TrackType type){
float WebRtcPusher::getLossRate(MediaSource &sender,mediakit::TrackType type){
return WebRtcTransportImp::getLossRate(type);
}
\ No newline at end of file
......@@ -40,7 +40,7 @@ protected:
// 获取媒体源客户端相关信息
std::shared_ptr<SockInfo> getOriginSock(mediakit::MediaSource &sender) const override;
// 获取丢包率
int getLossRate(mediakit::MediaSource &sender,mediakit::TrackType type) override;
float getLossRate(mediakit::MediaSource &sender,mediakit::TrackType type) override;
private:
WebRtcPusher(const EventPoller::Ptr &poller, const mediakit::RtspMediaSourceImp::Ptr &src,
......
......@@ -650,10 +650,10 @@ public:
return _rtcp_context.createRtcpRR(ssrc, getSSRC());
}
int getLossRate() {
float getLossRate() {
auto expected = _rtcp_context.getExpectedPacketsInterval();
if (!expected) {
return 0;
return -1;
}
return _rtcp_context.geLostInterval() * 100 / expected;
}
......@@ -698,7 +698,7 @@ std::shared_ptr<RtpChannel> MediaTrack::getRtpChannel(uint32_t ssrc) const {
return it_chn->second;
}
int WebRtcTransportImp::getLossRate(mediakit::TrackType type) {
float WebRtcTransportImp::getLossRate(mediakit::TrackType type) {
for (auto &pr : _ssrc_to_track) {
auto ssrc = pr.first;
auto &track = pr.second;
......
......@@ -263,7 +263,7 @@ protected:
void onShutdown(const SockException &ex) override;
virtual void onRecvRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp) = 0;
void updateTicker();
int getLossRate(mediakit::TrackType type);
float getLossRate(mediakit::TrackType type);
private:
void onSortedRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp);
......
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