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RTC: Refine NACK, remove dead code
This commit is contained in:
winlin
parent
f81d35d20f
commit
ab6e3cae52
4 changed files with 157 additions and 883 deletions
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@ -507,8 +507,8 @@ SrsRtcPlayer::SrsRtcPlayer(SrsRtcSession* s, int parent_cid)
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realtime = true;
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// TODO: FIXME: Config the capacity?
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audio_queue_ = new SrsRtpRingBuffer<SrsRtpPacket2*>(100);
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video_queue_ = new SrsRtpRingBuffer<SrsRtpPacket2*>(1000);
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audio_queue_ = new SrsRtpRingBuffer(100);
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video_queue_ = new SrsRtpRingBuffer(1000);
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nn_simulate_nack_drop = 0;
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nack_enabled_ = false;
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@ -1378,9 +1378,10 @@ SrsRtcPublisher::SrsRtcPublisher(SrsRtcSession* session)
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report_timer = new SrsHourGlass(this, 200 * SRS_UTIME_MILLISECONDS);
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session_ = session;
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video_queue_ = new SrsRtpVideoQueue(1000);
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request_keyframe_ = false;
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video_queue_ = new SrsRtpRingBuffer(1000);
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video_nack_ = new SrsRtpNackForReceiver(video_queue_, 1000 * 2 / 3);
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audio_queue_ = new SrsRtpAudioQueue(100);
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audio_queue_ = new SrsRtpRingBuffer(100);
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audio_nack_ = new SrsRtpNackForReceiver(audio_queue_, 100 * 2 / 3);
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source = NULL;
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@ -1490,7 +1491,7 @@ void SrsRtcPublisher::check_send_nacks(SrsRtpNackForReceiver* nack, uint32_t ssr
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}
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}
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srs_error_t SrsRtcPublisher::send_rtcp_rr(uint32_t ssrc, SrsRtpQueue* rtp_queue)
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srs_error_t SrsRtcPublisher::send_rtcp_rr(uint32_t ssrc, SrsRtpRingBuffer* rtp_queue)
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{
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srs_error_t err = srs_success;
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@ -1507,10 +1508,12 @@ srs_error_t SrsRtcPublisher::send_rtcp_rr(uint32_t ssrc, SrsRtpQueue* rtp_queue)
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stream.write_2bytes(7);
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stream.write_4bytes(ssrc); // TODO: FIXME: Should be 1?
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uint8_t fraction_lost = rtp_queue->get_fraction_lost();
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uint32_t cumulative_number_of_packets_lost = rtp_queue->get_cumulative_number_of_packets_lost() & 0x7FFFFF;
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// TODO: FIXME: Implements it.
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// TODO: FIXME: See https://github.com/ossrs/srs/blob/f81d35d20f04ebec01915cb78a882e45b7ee8800/trunk/src/app/srs_app_rtc_queue.cpp
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uint8_t fraction_lost = 0;
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uint32_t cumulative_number_of_packets_lost = 0 & 0x7FFFFF;
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uint32_t extended_highest_sequence = rtp_queue->get_extended_highest_sequence();
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uint32_t interarrival_jitter = rtp_queue->get_interarrival_jitter();
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uint32_t interarrival_jitter = 0;
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uint32_t rr_lsr = 0;
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uint32_t rr_dlsr = 0;
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@ -1648,15 +1651,17 @@ srs_error_t SrsRtcPublisher::on_rtp(char* buf, int nb_buf)
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{
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srs_error_t err = srs_success;
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// Decode the RTP packet from buffer.
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SrsRtpPacket2* pkt = new SrsRtpPacket2();
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if (true) {
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pkt->set_decode_handler(this);
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pkt->shared_msg = new SrsSharedPtrMessage();
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pkt->shared_msg->wrap(buf, nb_buf);
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pkt->set_decode_handler(this);
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pkt->shared_msg = new SrsSharedPtrMessage();
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pkt->shared_msg->wrap(buf, nb_buf);
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SrsBuffer b(buf, nb_buf);
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if ((err = pkt->decode(&b)) != srs_success) {
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return srs_error_wrap(err, "decode rtp packet");
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SrsBuffer b(buf, nb_buf);
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if ((err = pkt->decode(&b)) != srs_success) {
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return srs_error_wrap(err, "decode rtp packet");
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}
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}
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// For NACK simulator, drop packet.
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@ -1666,15 +1671,27 @@ srs_error_t SrsRtcPublisher::on_rtp(char* buf, int nb_buf)
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return err;
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}
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// For source to consume packet.
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uint32_t ssrc = pkt->rtp_header.get_ssrc();
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if (ssrc == audio_ssrc) {
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return on_audio(pkt);
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if ((err = on_audio(pkt)) != srs_success) {
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return srs_error_wrap(err, "on audio");
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}
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} else if (ssrc == video_ssrc) {
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return on_video(pkt);
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if ((err = on_video(pkt)) != srs_success) {
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return srs_error_wrap(err, "on video");
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}
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} else {
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srs_freep(pkt);
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return srs_error_new(ERROR_RTC_RTP, "unknown ssrc=%u", ssrc);
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}
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// For NACK to handle packet.
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if (nack_enabled_ && (err = on_nack(pkt)) != srs_success) {
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return srs_error_wrap(err, "on nack");
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}
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return err;
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}
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void SrsRtcPublisher::on_before_decode_payload(SrsRtpPacket2* pkt, SrsBuffer* buf, ISrsRtpPayloader** ppayload)
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@ -1709,40 +1726,6 @@ srs_error_t SrsRtcPublisher::on_audio(SrsRtpPacket2* pkt)
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source->on_rtp(pkt);
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return err;
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// TODO: FIXME: Directly dispatch to consumer for performance?
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std::vector<SrsRtpPacket2*> frames;
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if (nack_enabled_) {
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// TODO: FIXME: Error check.
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audio_queue_->consume(audio_nack_, pkt);
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check_send_nacks(audio_nack_, audio_ssrc);
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// Collect all audio frames.
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audio_queue_->collect_frames(audio_nack_, frames);
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} else {
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// TODO: FIXME: Error check.
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audio_queue_->consume(NULL, pkt);
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// Collect all audio frames.
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audio_queue_->collect_frames(NULL, frames);
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}
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for (size_t i = 0; i < frames.size(); ++i) {
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SrsRtpPacket2* frame = frames[i];
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// TODO: FIXME: Check error.
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source->on_rtp(frame);
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if (nn_audio_frames++ == 0) {
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SrsRtpHeader* h = &frame->rtp_header;
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SrsRtpRawPayload* payload = dynamic_cast<SrsRtpRawPayload*>(frame->payload);
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srs_trace("RTC got Opus seq=%u, ssrc=%u, ts=%u, %d bytes", h->get_sequence(), h->get_ssrc(), h->get_timestamp(), payload->nn_payload);
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}
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}
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return err;
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}
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srs_error_t SrsRtcPublisher::on_video(SrsRtpPacket2* pkt)
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@ -1754,152 +1737,38 @@ srs_error_t SrsRtcPublisher::on_video(SrsRtpPacket2* pkt)
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// TODO: FIXME: Error check.
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source->on_rtp(pkt);
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if (video_queue_->should_request_key_frame()) {
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if (request_keyframe_) {
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request_keyframe_ = false;
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// TODO: FIXME: Check error.
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send_rtcp_fb_pli(video_ssrc);
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}
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return err;
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std::vector<SrsRtpPacket2*> frames;
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if (nack_enabled_) {
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// TODO: FIXME: Error check.
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video_queue_->consume(video_nack_, pkt);
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check_send_nacks(video_nack_, video_ssrc);
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// Collect video frames.
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video_queue_->collect_frames(video_nack_, frames);
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} else {
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// TODO: FIXME: Error check.
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video_queue_->consume(NULL, pkt);
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// Collect video frames.
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video_queue_->collect_frames(NULL, frames);
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}
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for (size_t i = 0; i < frames.size(); ++i) {
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SrsRtpPacket2* frame = frames[i];
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// TODO: FIXME: Check error.
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on_video_frame(frame);
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srs_freep(frame);
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}
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if (video_queue_->should_request_key_frame()) {
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// TODO: FIXME: Check error.
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send_rtcp_fb_pli(video_ssrc);
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}
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return srs_success;
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}
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srs_error_t SrsRtcPublisher::on_video_frame(SrsRtpPacket2* frame)
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srs_error_t SrsRtcPublisher::on_nack(SrsRtpPacket2* pkt)
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{
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srs_error_t err = srs_success;
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int64_t timestamp = frame->rtp_header.get_timestamp();
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// No FU-A, because we convert it to RAW RTP packet.
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if (frame->nalu_type == (SrsAvcNaluType)kFuA) {
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return srs_error_new(ERROR_RTC_RTP_MUXER, "invalid FU-A");
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}
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// For STAP-A, it must be SPS/PPS, and only one packet.
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if (frame->nalu_type == (SrsAvcNaluType)kStapA) {
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SrsRtpSTAPPayload* payload = dynamic_cast<SrsRtpSTAPPayload*>(frame->payload);
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if (!payload) {
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return srs_error_new(ERROR_RTC_RTP_MUXER, "STAP-A payload");
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}
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SrsSample* sps = payload->get_sps();
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SrsSample* pps = payload->get_pps();
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if (!sps || !sps->size) {
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return srs_error_new(ERROR_RTC_RTP_MUXER, "STAP-A payload no sps");
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}
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if (!pps || !pps->size) {
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return srs_error_new(ERROR_RTC_RTP_MUXER, "STAP-A payload no pps");
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}
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// TODO: FIXME: Directly covert to sample for performance.
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// 5 bytes flv tag header.
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// 5 bytes sps/pps sequence header.
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// 6 bytes size for sps/pps, each is 3 bytes.
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int nn_payload = sps->size + pps->size + 5 + 5 + 6;
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char* data = new char[nn_payload];
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SrsBuffer buf(data, nn_payload);
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buf.write_1bytes(0x17); // Keyframe.
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buf.write_1bytes(0x00); // Sequence header.
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buf.write_3bytes(0x00); // CTS.
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// FIXME: Replace magic number for avc_demux_sps_pps.
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buf.write_1bytes(0x01); // configurationVersion
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buf.write_1bytes(0x42); // AVCProfileIndication, 0x42 = Baseline
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buf.write_1bytes(0xC0); // profile_compatibility
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buf.write_1bytes(0x1f); // AVCLevelIndication, 0x1f = Level3.1
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buf.write_1bytes(0x03); // lengthSizeMinusOne, size of length for NALU.
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buf.write_1bytes(0x01); // numOfSequenceParameterSets
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buf.write_2bytes(sps->size); // sequenceParameterSetLength
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buf.write_bytes(sps->bytes, sps->size); // sps
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buf.write_1bytes(0x01); // numOfPictureParameterSets
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buf.write_2bytes(pps->size); // pictureParameterSetLength
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buf.write_bytes(pps->bytes, pps->size); // pps
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SrsMessageHeader header;
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header.message_type = RTMP_MSG_VideoMessage;
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// TODO: FIXME: Maybe the tbn is not 90k.
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header.timestamp = (timestamp / 90) & 0x3fffffff;
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SrsCommonMessage* shared_video = new SrsCommonMessage();
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SrsAutoFree(SrsCommonMessage, shared_video);
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// TODO: FIXME: Check error.
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shared_video->create(&header, data, nn_payload);
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return source->on_video(shared_video);
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}
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// For RAW NALU, should be one RAW packet.
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SrsRtpRawPayload* payload = dynamic_cast<SrsRtpRawPayload*>(frame->payload);
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if (!payload) {
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return srs_error_new(ERROR_RTC_RTP_MUXER, "RAW-NALU payload");
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}
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if (!payload->nn_payload) {
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uint16_t seq = pkt->rtp_header.get_sequence();
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SrsRtpNackInfo* nack_info = audio_nack_->find(seq);
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if (nack_info) {
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return err;
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}
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// TODO: FIXME: Directly covert to sample for performance.
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// 5 bytes FLV tag header.
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// 4 bytes NALU IBMF header, define by sequence header.
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int nn_payload = payload->nn_payload + 5 + 4;
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char* data = new char[nn_payload];
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SrsBuffer buf(data, nn_payload);
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if (frame->nalu_type == SrsAvcNaluTypeIDR) {
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buf.write_1bytes(0x17); // Keyframe.
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SrsRtpHeader* h = &frame->rtp_header;
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srs_trace("RTC got IDR seq=%u, ssrc=%u, ts=%u, %d bytes", h->get_sequence(), h->get_ssrc(), h->get_timestamp(), nn_payload);
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} else {
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buf.write_1bytes(0x27); // Not-Keyframe.
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uint16_t nack_first = 0, nack_last = 0;
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if (!audio_queue_->update(seq, nack_first, nack_last)) {
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srs_warn("too old seq %u, range [%u, %u]", seq, audio_queue_->begin, audio_queue_->end);
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}
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buf.write_1bytes(0x01); // Not-SequenceHeader.
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buf.write_3bytes(0x00); // CTS.
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buf.write_4bytes(payload->nn_payload); // Size of NALU.
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buf.write_bytes(payload->payload, payload->nn_payload); // NALU.
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if (srs_rtp_seq_distance(nack_first, nack_last) > 0) {
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srs_trace("update seq=%u, nack range [%u, %u]", seq, nack_first, nack_last);
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audio_nack_->insert(nack_first, nack_last);
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audio_nack_->check_queue_size();
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}
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SrsMessageHeader header;
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header.message_type = RTMP_MSG_VideoMessage;
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// TODO: FIXME: Maybe the tbn is not 90k.
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header.timestamp = (timestamp / 90) & 0x3fffffff;
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SrsCommonMessage* shared_video = new SrsCommonMessage();
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SrsAutoFree(SrsCommonMessage, shared_video);
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// TODO: FIXME: Check error.
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shared_video->create(&header, data, nn_payload);
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return source->on_video(shared_video);
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return err;
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}
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srs_error_t SrsRtcPublisher::on_rtcp(char* data, int nb_data)
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@ -2250,7 +2119,7 @@ void SrsRtcPublisher::request_keyframe()
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int pcid = session_->context_id();
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srs_trace("RTC play=[%d][%d] request keyframe from publish=[%d][%d]", ::getpid(), scid, ::getpid(), pcid);
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video_queue_->request_keyframe();
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request_keyframe_ = true;
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}
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srs_error_t SrsRtcPublisher::notify(int type, srs_utime_t interval, srs_utime_t tick)
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