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https://github.com/ossrs/srs.git
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Perf: Refine player cycle, use fast coroutine
This commit is contained in:
winlin
parent
29b33e6303
commit
ecef3e7f0a
6 changed files with 236 additions and 141 deletions
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@ -382,7 +382,7 @@ SrsRtcPlayStreamStatistic::~SrsRtcPlayStreamStatistic()
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SrsRtcPlayStream::SrsRtcPlayStream(SrsRtcConnection* s, const SrsContextId& cid)
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{
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cid_ = cid;
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trd = new SrsDummyCoroutine();
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trd_ = NULL;
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req_ = NULL;
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source_ = NULL;
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@ -412,7 +412,7 @@ SrsRtcPlayStream::~SrsRtcPlayStream()
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srs_freep(nack_epp);
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srs_freep(pli_worker_);
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srs_freep(trd);
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srs_freep(trd_);
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srs_freep(timer_);
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srs_freep(req_);
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@ -499,10 +499,10 @@ srs_error_t SrsRtcPlayStream::start()
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return err;
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}
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srs_freep(trd);
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trd = new SrsSTCoroutine("rtc_sender", this, cid_);
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srs_freep(trd_);
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trd_ = new SrsFastCoroutine("rtc_sender", this, cid_);
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if ((err = trd->start()) != srs_success) {
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if ((err = trd_->start()) != srs_success) {
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return srs_error_wrap(err, "rtc_sender");
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}
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@ -527,7 +527,9 @@ srs_error_t SrsRtcPlayStream::start()
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void SrsRtcPlayStream::stop()
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{
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trd->stop();
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if (trd_) {
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trd_->stop();
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}
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}
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srs_error_t SrsRtcPlayStream::cycle()
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@ -550,24 +552,14 @@ srs_error_t SrsRtcPlayStream::cycle()
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realtime = _srs_config->get_realtime_enabled(req_->vhost, true);
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mw_msgs = _srs_config->get_mw_msgs(req_->vhost, realtime, true);
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bool stat_enabled = _srs_config->get_rtc_server_perf_stat();
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SrsStatistic* stat = SrsStatistic::instance();
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// TODO: FIXME: Add cost in ms.
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SrsContextId cid = source->source_id();
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srs_trace("RTC: start play url=%s, source_id=%s/%s, realtime=%d, mw_msgs=%d, stat=%d", req_->get_stream_url().c_str(),
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cid.c_str(), source->pre_source_id().c_str(), realtime, mw_msgs, stat_enabled);
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srs_trace("RTC: start play url=%s, source_id=%s/%s, realtime=%d, mw_msgs=%d", req_->get_stream_url().c_str(),
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cid.c_str(), source->pre_source_id().c_str(), realtime, mw_msgs);
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SrsErrorPithyPrint* epp = new SrsErrorPithyPrint();
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SrsAutoFree(SrsErrorPithyPrint, epp);
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SrsPithyPrint* pprint = SrsPithyPrint::create_rtc_play();
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SrsAutoFree(SrsPithyPrint, pprint);
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// TODO: FIXME: Use cache for performance?
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vector<SrsRtpPacket2*> pkts;
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uint64_t total_pkts = 0;
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if (_srs_rtc_hijacker) {
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if ((err = _srs_rtc_hijacker->on_start_consume(session_, this, req_, consumer)) != srs_success) {
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return srs_error_wrap(err, "on start consuming");
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@ -575,64 +567,30 @@ srs_error_t SrsRtcPlayStream::cycle()
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}
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while (true) {
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if ((err = trd->pull()) != srs_success) {
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if ((err = trd_->pull()) != srs_success) {
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return srs_error_wrap(err, "rtc sender thread");
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}
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// Wait for amount of packets.
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consumer->wait(mw_msgs);
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// TODO: FIXME: Handle error.
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consumer->dump_packets(pkts);
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int msg_count = (int)pkts.size();
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if (!msg_count) {
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continue;
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try_dump_again:
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SrsRtpPacket2* pkt = NULL;
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consumer->dump_packet(&pkt);
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if (!pkt) {
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// TODO: FIXME: We should check the quit event.
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consumer->wait(mw_msgs);
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goto try_dump_again;
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}
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// Update stats for session.
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session_->stat_->nn_out_rtp += msg_count;
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total_pkts += msg_count;
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// Send-out all RTP packets and do cleanup
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if (true) {
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if ((err = send_packets(source, pkts, info)) != srs_success) {
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uint32_t nn = 0;
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if (epp->can_print(err, &nn)) {
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srs_warn("play send packets=%u, nn=%u/%u, err: %s", pkts.size(), epp->nn_count, nn, srs_error_desc(err).c_str());
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}
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srs_freep(err);
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// Send-out the RTP packet and do cleanup
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if ((err = send_packet(pkt)) != srs_success) {
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uint32_t nn = 0;
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if (epp->can_print(err, &nn)) {
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srs_warn("play send packets=%u, nn=%u/%u, err: %s", 1, epp->nn_count, nn, srs_error_desc(err).c_str());
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}
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for (int i = 0; i < msg_count; i++) {
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SrsRtpPacket2* pkt = pkts[i];
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_srs_rtp_cache->recycle(pkt);
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}
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pkts.clear();
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srs_freep(err);
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}
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// Stat for performance analysis.
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if (!stat_enabled) {
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continue;
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}
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// Stat the original RAW AV frame, maybe h264+aac.
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stat->perf_on_msgs(msg_count);
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// Stat the RTC packets, RAW AV frame, maybe h.264+opus.
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int nn_rtc_packets = srs_max(info.nn_audios, info.nn_extras) + info.nn_videos;
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stat->perf_on_rtc_packets(nn_rtc_packets);
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// Stat the RAW RTP packets, which maybe group by GSO.
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stat->perf_on_rtp_packets(msg_count);
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// Stat the bytes and paddings.
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stat->perf_on_rtc_bytes(info.nn_bytes, info.nn_rtp_bytes, info.nn_padding_bytes);
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pprint->elapse();
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if (pprint->can_print()) {
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// TODO: FIXME: Print stat like frame/s, packet/s, loss_packets.
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srs_trace("-> RTC PLAY %d msgs, %d/%d packets, %d audios, %d extras, %d videos, %d samples, %d/%d/%d bytes, %d pad, %d/%d cache",
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total_pkts, msg_count, info.nn_rtp_pkts, info.nn_audios, info.nn_extras, info.nn_videos, info.nn_samples, info.nn_bytes,
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info.nn_rtp_bytes, info.nn_padding_bytes, info.nn_paddings, msg_count, msg_count);
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}
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_srs_rtp_cache->recycle(pkt);
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}
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}
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@ -640,7 +598,6 @@ srs_error_t SrsRtcPlayStream::send_packets(SrsRtcStream* source, const vector<Sr
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{
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srs_error_t err = srs_success;
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vector<SrsRtpPacket2*> send_pkts;
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// Covert kernel messages to RTP packets.
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for (int i = 0; i < (int)pkts.size(); i++) {
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SrsRtpPacket2* pkt = pkts[i];
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@ -656,7 +613,7 @@ srs_error_t SrsRtcPlayStream::send_packets(SrsRtcStream* source, const vector<Sr
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// TODO: FIXME: Any simple solution?
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SrsRtcAudioSendTrack* audio_track = audio_tracks_[pkt->header.get_ssrc()];
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if ((err = audio_track->on_rtp(pkt, info)) != srs_success) {
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if ((err = audio_track->on_rtp(pkt)) != srs_success) {
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return srs_error_wrap(err, "audio track, SSRC=%u, SEQ=%u", pkt->header.get_ssrc(), pkt->header.get_sequence());
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}
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@ -665,7 +622,7 @@ srs_error_t SrsRtcPlayStream::send_packets(SrsRtcStream* source, const vector<Sr
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// TODO: FIXME: Any simple solution?
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SrsRtcVideoSendTrack* video_track = video_tracks_[pkt->header.get_ssrc()];
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if ((err = video_track->on_rtp(pkt, info)) != srs_success) {
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if ((err = video_track->on_rtp(pkt)) != srs_success) {
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return srs_error_wrap(err, "video track, SSRC=%u, SEQ=%u", pkt->header.get_ssrc(), pkt->header.get_sequence());
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}
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}
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@ -678,6 +635,42 @@ srs_error_t SrsRtcPlayStream::send_packets(SrsRtcStream* source, const vector<Sr
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return err;
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}
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srs_error_t SrsRtcPlayStream::send_packet(SrsRtpPacket2* pkt)
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{
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srs_error_t err = srs_success;
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// TODO: FIXME: Maybe refine for performance issue.
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if (!audio_tracks_.count(pkt->header.get_ssrc()) && !video_tracks_.count(pkt->header.get_ssrc())) {
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srs_warn("ssrc %u not found", pkt->header.get_ssrc());
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return err;
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}
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// For audio, we transcoded AAC to opus in extra payloads.
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if (pkt->is_audio()) {
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// TODO: FIXME: Any simple solution?
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SrsRtcAudioSendTrack* audio_track = audio_tracks_[pkt->header.get_ssrc()];
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if ((err = audio_track->on_rtp(pkt)) != srs_success) {
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return srs_error_wrap(err, "audio track, SSRC=%u, SEQ=%u", pkt->header.get_ssrc(), pkt->header.get_sequence());
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}
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// TODO: FIXME: Padding audio to the max payload in RTP packets.
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} else {
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// TODO: FIXME: Any simple solution?
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SrsRtcVideoSendTrack* video_track = video_tracks_[pkt->header.get_ssrc()];
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if ((err = video_track->on_rtp(pkt)) != srs_success) {
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return srs_error_wrap(err, "video track, SSRC=%u, SEQ=%u", pkt->header.get_ssrc(), pkt->header.get_sequence());
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}
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}
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// Detail log, should disable it in release version.
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srs_info("RTC: Update PT=%u, SSRC=%#x, Time=%u, %u bytes", pkt->header.get_payload_type(), pkt->header.get_ssrc(),
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pkt->header.get_timestamp(), pkt->nb_bytes());
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return err;
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}
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void SrsRtcPlayStream::set_all_tracks_status(bool status)
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{
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std::ostringstream merged_log;
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@ -805,7 +798,7 @@ srs_error_t SrsRtcPlayStream::on_rtcp_nack(SrsRtcpNack* rtcp)
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}
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vector<uint16_t> seqs = rtcp->get_lost_sns();
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if((err = target->on_recv_nack(seqs, info)) != srs_success) {
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if((err = target->on_recv_nack(seqs)) != srs_success) {
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return srs_error_wrap(err, "track response nack. id:%s, ssrc=%u", target->get_track_id().c_str(), ssrc);
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}
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@ -2569,6 +2562,51 @@ srs_error_t SrsRtcConnection::do_send_packets(const std::vector<SrsRtpPacket2*>&
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return err;
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}
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srs_error_t SrsRtcConnection::do_send_packet(SrsRtpPacket2* pkt)
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{
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srs_error_t err = srs_success;
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// For this message, select the first iovec.
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iovec* iov = cache_iov_;
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iov->iov_len = kRtpPacketSize;
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cache_buffer_->skip(-1 * cache_buffer_->pos());
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// Marshal packet to bytes in iovec.
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if (true) {
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if ((err = pkt->encode(cache_buffer_)) != srs_success) {
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return srs_error_wrap(err, "encode packet");
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}
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iov->iov_len = cache_buffer_->pos();
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}
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// Cipher RTP to SRTP packet.
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if (true) {
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int nn_encrypt = (int)iov->iov_len;
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if ((err = transport_->protect_rtp(iov->iov_base, &nn_encrypt)) != srs_success) {
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return srs_error_wrap(err, "srtp protect");
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}
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iov->iov_len = (size_t)nn_encrypt;
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}
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// For NACK simulator, drop packet.
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if (nn_simulate_player_nack_drop) {
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simulate_player_drop_packet(&pkt->header, (int)iov->iov_len);
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iov->iov_len = 0;
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return err;
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}
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++_srs_pps_srtps->sugar;
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// TODO: FIXME: Handle error.
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sendonly_skt->sendto(iov->iov_base, iov->iov_len, 0);
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// Detail log, should disable it in release version.
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srs_info("RTC: SEND PT=%u, SSRC=%#x, SEQ=%u, Time=%u, %u/%u bytes", pkt->header.get_payload_type(), pkt->header.get_ssrc(),
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pkt->header.get_sequence(), pkt->header.get_timestamp(), pkt->nb_bytes(), iov->iov_len);
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return err;
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}
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void SrsRtcConnection::set_all_tracks_status(std::string stream_uri, bool is_publish, bool status)
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{
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// For publishers.
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