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RTC: Refine RTMP bridge to RTC, use RTP packets in consumer
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winlin
parent
54d8c36905
commit
2b1c4a188a
7 changed files with 340 additions and 286 deletions
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@ -65,12 +65,6 @@ using namespace std;
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#include <srs_app_rtc_server.hpp>
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#include <srs_app_rtc_source.hpp>
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// The RTP payload max size, reserved some paddings for SRTP as such:
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// kRtpPacketSize = kRtpMaxPayloadSize + paddings
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// For example, if kRtpPacketSize is 1500, recommend to set kRtpMaxPayloadSize to 1400,
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// which reserves 100 bytes for SRTP or paddings.
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const int kRtpMaxPayloadSize = kRtpPacketSize - 200;
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string gen_random_str(int len)
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{
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static string random_table = "0123456789abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ";
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@ -565,7 +559,6 @@ SrsRtcPlayer::SrsRtcPlayer(SrsRtcSession* s, int parent_cid)
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session_ = s;
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gso = false;
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merge_nalus = false;
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max_padding = 0;
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audio_timestamp = 0;
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@ -606,12 +599,11 @@ srs_error_t SrsRtcPlayer::initialize(const uint32_t& vssrc, const uint32_t& assr
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audio_payload_type = a_pt;
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gso = _srs_config->get_rtc_server_gso();
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merge_nalus = _srs_config->get_rtc_server_merge_nalus();
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max_padding = _srs_config->get_rtc_server_padding();
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// TODO: FIXME: Support reload.
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nack_enabled_ = _srs_config->get_rtc_nack_enabled(session_->req->vhost);
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srs_trace("RTC publisher video(ssrc=%d, pt=%d), audio(ssrc=%d, pt=%d), package(gso=%d, merge_nalus=%d), padding=%d, nack=%d",
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video_ssrc, video_payload_type, audio_ssrc, audio_payload_type, gso, merge_nalus, max_padding, nack_enabled_);
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srs_trace("RTC publisher video(ssrc=%d, pt=%d), audio(ssrc=%d, pt=%d), gso=%d, padding=%d, nack=%d",
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video_ssrc, video_payload_type, audio_ssrc, audio_payload_type, gso, max_padding, nack_enabled_);
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return err;
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}
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@ -619,10 +611,9 @@ srs_error_t SrsRtcPlayer::initialize(const uint32_t& vssrc, const uint32_t& assr
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srs_error_t SrsRtcPlayer::on_reload_rtc_server()
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{
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gso = _srs_config->get_rtc_server_gso();
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merge_nalus = _srs_config->get_rtc_server_merge_nalus();
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max_padding = _srs_config->get_rtc_server_padding();
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srs_trace("Reload rtc_server gso=%d, merge_nalus=%d, max_padding=%d", gso, merge_nalus, max_padding);
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srs_trace("Reload rtc_server gso=%d, max_padding=%d", gso, max_padding);
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return srs_success;
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}
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@ -836,7 +827,7 @@ srs_error_t SrsRtcPlayer::messages_to_packets(SrsRtcSource* source, vector<SrsRt
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info.nn_audios++;
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if ((err = package_opus(pkt)) != srs_success) {
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return srs_error_wrap(err, "opus package");
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return srs_error_wrap(err, "package opus");
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}
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continue;
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}
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@ -844,45 +835,10 @@ srs_error_t SrsRtcPlayer::messages_to_packets(SrsRtcSource* source, vector<SrsRt
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// For video, we should process all NALUs in samples.
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info.nn_videos++;
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// Well, for each IDR, we append a SPS/PPS before it, which is packaged in STAP-A.
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/*if (msg->has_idr()) {
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if ((err = package_stap_a(source, msg, info)) != srs_success) {
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return srs_error_wrap(err, "packet stap-a");
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}
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// For video, we should set the RTP packet informations about this consumer.
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if ((err = package_video(pkt)) != srs_success) {
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return srs_error_wrap(err, "package video");
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}
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// If merge Nalus, we pcakges all NALUs(samples) as one NALU, in a RTP or FUA packet.
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if (info.should_merge_nalus && nn_samples > 1) {
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if ((err = package_nalus(msg, info)) != srs_success) {
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return srs_error_wrap(err, "packet stap-a");
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}
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continue;
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}
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// By default, we package each NALU(sample) to a RTP or FUA packet.
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for (int i = 0; i < nn_samples; i++) {
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SrsSample* sample = msg->samples() + i;
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// We always ignore bframe here, if config to discard bframe,
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// the bframe flag will not be set.
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if (sample->bframe) {
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continue;
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}
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if (sample->size <= kRtpMaxPayloadSize) {
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if ((err = package_single_nalu(msg, sample, info)) != srs_success) {
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return srs_error_wrap(err, "packet single nalu");
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}
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} else {
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if ((err = package_fu_a(msg, sample, kRtpMaxPayloadSize, info)) != srs_success) {
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return srs_error_wrap(err, "packet fu-a");
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}
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}
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if (i == nn_samples - 1) {
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info.back()->rtp_header.set_marker(true);
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}
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}*/
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}
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return err;
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@ -1203,101 +1159,10 @@ srs_error_t SrsRtcPlayer::send_packets_gso(SrsRtcOutgoingPackets& packets)
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return err;
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}
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srs_error_t SrsRtcPlayer::package_nalus(SrsSharedPtrMessage* msg, SrsRtcOutgoingPackets& packets)
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{
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srs_error_t err = srs_success;
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SrsRtpRawNALUs* raw = new SrsRtpRawNALUs();
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for (int i = 0; i < msg->nn_samples(); i++) {
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SrsSample* sample = msg->samples() + i;
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// We always ignore bframe here, if config to discard bframe,
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// the bframe flag will not be set.
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if (sample->bframe) {
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continue;
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}
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raw->push_back(sample->copy());
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}
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// Ignore empty.
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int nn_bytes = raw->nb_bytes();
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if (nn_bytes <= 0) {
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srs_freep(raw);
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return err;
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}
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if (nn_bytes < kRtpMaxPayloadSize) {
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// Package NALUs in a single RTP packet.
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SrsRtpPacket2* packet = packets.fetch();
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if (!packet) {
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srs_freep(raw);
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return srs_error_new(ERROR_RTC_RTP_MUXER, "cache empty");
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}
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packet->rtp_header.set_timestamp(msg->timestamp * 90);
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packet->rtp_header.set_sequence(video_sequence++);
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packet->rtp_header.set_ssrc(video_ssrc);
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packet->rtp_header.set_payload_type(video_payload_type);
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packet->payload = raw;
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} else {
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// We must free it, should never use RTP packets to free it,
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// because more than one RTP packet will refer to it.
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SrsAutoFree(SrsRtpRawNALUs, raw);
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// Package NALUs in FU-A RTP packets.
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int fu_payload_size = kRtpMaxPayloadSize;
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// The first byte is store in FU-A header.
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uint8_t header = raw->skip_first_byte();
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uint8_t nal_type = header & kNalTypeMask;
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int nb_left = nn_bytes - 1;
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int num_of_packet = 1 + (nn_bytes - 1) / fu_payload_size;
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for (int i = 0; i < num_of_packet; ++i) {
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int packet_size = srs_min(nb_left, fu_payload_size);
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SrsRtpPacket2* packet = packets.fetch();
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if (!packet) {
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srs_freep(raw);
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return srs_error_new(ERROR_RTC_RTP_MUXER, "cache empty");
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}
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packet->rtp_header.set_timestamp(msg->timestamp * 90);
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packet->rtp_header.set_sequence(video_sequence++);
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packet->rtp_header.set_ssrc(video_ssrc);
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packet->rtp_header.set_payload_type(video_payload_type);
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SrsRtpFUAPayload* fua = new SrsRtpFUAPayload();
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packet->payload = fua;
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fua->nri = (SrsAvcNaluType)header;
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fua->nalu_type = (SrsAvcNaluType)nal_type;
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fua->start = bool(i == 0);
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fua->end = bool(i == num_of_packet - 1);
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if ((err = raw->read_samples(fua->nalus, packet_size)) != srs_success) {
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return srs_error_wrap(err, "read samples %d bytes, left %d, total %d", packet_size, nb_left, nn_bytes);
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}
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nb_left -= packet_size;
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}
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}
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if (packets.size() > 0) {
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packets.back()->rtp_header.set_marker(true);
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}
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return err;
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}
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srs_error_t SrsRtcPlayer::package_opus(SrsRtpPacket2* pkt)
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{
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srs_error_t err = srs_success;
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pkt->rtp_header.set_marker(true);
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pkt->rtp_header.set_timestamp(audio_timestamp);
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pkt->rtp_header.set_sequence(audio_sequence++);
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pkt->rtp_header.set_ssrc(audio_ssrc);
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@ -1313,118 +1178,13 @@ srs_error_t SrsRtcPlayer::package_opus(SrsRtpPacket2* pkt)
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return err;
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}
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srs_error_t SrsRtcPlayer::package_fu_a(SrsSharedPtrMessage* msg, SrsSample* sample, int fu_payload_size, SrsRtcOutgoingPackets& packets)
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srs_error_t SrsRtcPlayer::package_video(SrsRtpPacket2* pkt)
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{
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srs_error_t err = srs_success;
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char* p = sample->bytes + 1;
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int nb_left = sample->size - 1;
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uint8_t header = sample->bytes[0];
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uint8_t nal_type = header & kNalTypeMask;
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int num_of_packet = 1 + (sample->size - 1) / fu_payload_size;
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for (int i = 0; i < num_of_packet; ++i) {
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int packet_size = srs_min(nb_left, fu_payload_size);
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SrsRtpPacket2* packet = packets.fetch();
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if (!packet) {
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return srs_error_new(ERROR_RTC_RTP_MUXER, "cache empty");
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}
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packet->rtp_header.set_timestamp(msg->timestamp * 90);
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packet->rtp_header.set_sequence(video_sequence++);
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packet->rtp_header.set_ssrc(video_ssrc);
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packet->rtp_header.set_payload_type(video_payload_type);
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SrsRtpFUAPayload2* fua = packet->reuse_fua();
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fua->nri = (SrsAvcNaluType)header;
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fua->nalu_type = (SrsAvcNaluType)nal_type;
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fua->start = bool(i == 0);
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fua->end = bool(i == num_of_packet - 1);
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fua->payload = p;
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fua->size = packet_size;
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p += packet_size;
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nb_left -= packet_size;
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}
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return err;
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}
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// Single NAL Unit Packet @see https://tools.ietf.org/html/rfc6184#section-5.6
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srs_error_t SrsRtcPlayer::package_single_nalu(SrsSharedPtrMessage* msg, SrsSample* sample, SrsRtcOutgoingPackets& packets)
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{
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srs_error_t err = srs_success;
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SrsRtpPacket2* packet = packets.fetch();
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if (!packet) {
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return srs_error_new(ERROR_RTC_RTP_MUXER, "cache empty");
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}
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packet->rtp_header.set_timestamp(msg->timestamp * 90);
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packet->rtp_header.set_sequence(video_sequence++);
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packet->rtp_header.set_ssrc(video_ssrc);
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packet->rtp_header.set_payload_type(video_payload_type);
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SrsRtpRawPayload* raw = packet->reuse_raw();
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raw->payload = sample->bytes;
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raw->nn_payload = sample->size;
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return err;
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}
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srs_error_t SrsRtcPlayer::package_stap_a(SrsRtcSource* source, SrsSharedPtrMessage* msg, SrsRtcOutgoingPackets& packets)
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{
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srs_error_t err = srs_success;
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SrsMetaCache* meta = source->cached_meta();
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if (!meta) {
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return err;
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}
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SrsFormat* format = meta->vsh_format();
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if (!format || !format->vcodec) {
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return err;
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}
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const vector<char>& sps = format->vcodec->sequenceParameterSetNALUnit;
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const vector<char>& pps = format->vcodec->pictureParameterSetNALUnit;
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if (sps.empty() || pps.empty()) {
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return srs_error_new(ERROR_RTC_RTP_MUXER, "sps/pps empty");
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}
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SrsRtpPacket2* packet = packets.fetch();
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if (!packet) {
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return srs_error_new(ERROR_RTC_RTP_MUXER, "cache empty");
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}
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packet->rtp_header.set_marker(false);
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packet->rtp_header.set_timestamp(msg->timestamp * 90);
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packet->rtp_header.set_sequence(video_sequence++);
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packet->rtp_header.set_ssrc(video_ssrc);
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packet->rtp_header.set_payload_type(video_payload_type);
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SrsRtpSTAPPayload* stap = new SrsRtpSTAPPayload();
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packet->payload = stap;
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uint8_t header = sps[0];
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stap->nri = (SrsAvcNaluType)header;
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if (true) {
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SrsSample* sample = new SrsSample();
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sample->bytes = (char*)&sps[0];
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sample->size = (int)sps.size();
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stap->nalus.push_back(sample);
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}
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if (true) {
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SrsSample* sample = new SrsSample();
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sample->bytes = (char*)&pps[0];
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sample->size = (int)pps.size();
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stap->nalus.push_back(sample);
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}
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srs_trace("RTC STAP-A seq=%u, sps %d, pps %d bytes", packet->rtp_header.get_sequence(), sps.size(), pps.size());
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pkt->rtp_header.set_sequence(video_sequence++);
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pkt->rtp_header.set_ssrc(video_ssrc);
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pkt->rtp_header.set_payload_type(video_payload_type);
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return err;
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}
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