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fix #133, support push rtsp to srs. 2.0.120.
This commit is contained in:
winlin
parent
a954040d29
commit
9d233db27e
11 changed files with 670 additions and 80 deletions
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@ -1,10 +1,6 @@
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#Simple-RTMP-Server
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<<<<<<< HEAD
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SRS/2.0，开发代号：[ZhouGuowen](https://github.com/winlinvip/simple-rtmp-server/wiki/v1_CN_Product#release20)
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=======
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SRS/1.0，开发代号：[HuKaiqun](https://github.com/winlinvip/simple-rtmp-server/wiki/v1_CN_Product#release10)
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>>>>>>> 1.0release
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SRS定位是运营级的互联网直播服务器集群，追求更好的概念完整性和最简单实现的代码。
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@ -493,6 +489,8 @@ Supported operating systems and hardware:
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[#250](https://github.com/winlinvip/simple-rtmp-server/issues/250).
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1. Rewrite HLS(h.264+aac/mp3) streaming, read
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[#304](https://github.com/winlinvip/simple-rtmp-server/issues/304).
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1. Support push RTSP to SRS, read
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[#133](https://github.com/winlinvip/simple-rtmp-server/issues/133).
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1. [no-plan] Support <500ms latency, FRSC(Fast RTMP-compatible Stream Channel tech).
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1. [no-plan] Support RTMP 302 redirect [#92](https://github.com/winlinvip/simple-rtmp-server/issues/92).
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1. [no-plan] Support multiple processes, for both origin and edge
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@ -531,6 +529,7 @@ Supported operating systems and hardware:
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### SRS 2.0 history
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* v2.0, 2015-02-18, fix [#133](https://github.com/winlinvip/simple-rtmp-server/issues/133), support push rtsp to srs. 2.0.120.
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* v2.0, 2015-02-17, the join maybe failed, should use a variable to ensure thread terminated. 2.0.119.
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* v2.0, 2015-02-15, for [#304](https://github.com/winlinvip/simple-rtmp-server/issues/304), support config default acodec/vcodec. 2.0.118.
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* v2.0, 2015-02-15, for [#304](https://github.com/winlinvip/simple-rtmp-server/issues/304), rewrite hls/ts code, support h.264+mp3 for hls. 2.0.117.
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@ -629,7 +629,6 @@ int SrsMpegtsOverUdp::connect()
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return ret;
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}
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return ret;
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}
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@ -35,6 +35,12 @@ using namespace std;
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#include <srs_core_autofree.hpp>
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#include <srs_kernel_stream.hpp>
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#include <srs_kernel_buffer.hpp>
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#include <srs_rtmp_sdk.hpp>
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#include <srs_rtmp_amf0.hpp>
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#include <srs_rtmp_utility.hpp>
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#include <srs_kernel_utility.hpp>
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#include <srs_raw_avc.hpp>
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#include <srs_kernel_codec.hpp>
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#ifdef SRS_AUTO_STREAM_CASTER
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@ -100,15 +106,15 @@ int SrsRtpConn::on_udp_packet(sockaddr_in* from, char* buf, int nb_buf)
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}
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}
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srs_trace("rtsp: rtp %dB, vt=%d/%u, sts=%u/%#x/%#x, paylod=%dB, chunked=%d",
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nb_buf, cache->version, cache->payload_type, cache->sequence_number, cache->timestamp, cache->ssrc,
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srs_trace("rtsp: rtp #%d %dB, vt=%d/%u, sts=%u/%u/%#x, paylod=%dB, chunked=%d",
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stream_id, nb_buf, cache->version, cache->payload_type, cache->sequence_number, cache->timestamp, cache->ssrc,
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cache->payload->length(), cache->chunked
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);
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// always free it.
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SrsAutoFree(SrsRtpPacket, cache);
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if ((ret = rtsp->on_rtp_packet(cache)) != ERROR_SUCCESS) {
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if ((ret = rtsp->on_rtp_packet(cache, stream_id)) != ERROR_SUCCESS) {
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srs_error("rtsp: process rtp packet failed. ret=%d", ret);
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return ret;
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}
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@ -116,9 +122,59 @@ int SrsRtpConn::on_udp_packet(sockaddr_in* from, char* buf, int nb_buf)
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return ret;
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}
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SrsRtspAudioCache::SrsRtspAudioCache()
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{
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dts = NULL;
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audio_samples = NULL;
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payload = NULL;
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}
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SrsRtspAudioCache::~SrsRtspAudioCache()
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{
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srs_freep(audio_samples);
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srs_freep(payload);
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}
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SrsRtspJitter::SrsRtspJitter()
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{
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delta = 0;
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previous_timestamp = 0;
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pts = 0;
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}
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SrsRtspJitter::~SrsRtspJitter()
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{
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}
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int64_t SrsRtspJitter::timestamp()
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{
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return pts;
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}
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int SrsRtspJitter::correct(int64_t& ts)
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{
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int ret = ERROR_SUCCESS;
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if (previous_timestamp == 0) {
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previous_timestamp = ts;
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}
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delta = srs_max(0, ts - previous_timestamp);
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if (delta > 90000) {
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delta = 0;
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}
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previous_timestamp = ts;
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ts = pts + delta;
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pts = ts;
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return ret;
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}
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SrsRtspConn::SrsRtspConn(SrsRtspCaster* c, st_netfd_t fd, std::string o)
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{
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output = o;
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output_template = o;
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session = "";
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video_rtp = NULL;
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@ -129,6 +185,18 @@ SrsRtspConn::SrsRtspConn(SrsRtspCaster* c, st_netfd_t fd, std::string o)
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skt = new SrsStSocket(fd);
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rtsp = new SrsRtspStack(skt);
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trd = new SrsThread("rtsp", this, 0, false);
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req = NULL;
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io = NULL;
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client = NULL;
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stream_id = 0;
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vjitter = new SrsRtspJitter();
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ajitter = new SrsRtspJitter();
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avc = new SrsRawH264Stream();
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aac = new SrsRawAacStream();
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acodec = new SrsRawAacStreamCodec();
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acache = new SrsRtspAudioCache();
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}
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SrsRtspConn::~SrsRtspConn()
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@ -142,6 +210,13 @@ SrsRtspConn::~SrsRtspConn()
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srs_freep(trd);
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srs_freep(skt);
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srs_freep(rtsp);
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close();
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srs_freep(vjitter);
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srs_freep(ajitter);
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srs_freep(acodec);
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srs_freep(acache);
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}
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int SrsRtspConn::serve()
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@ -179,6 +254,18 @@ int SrsRtspConn::do_cycle()
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return ret;
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}
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} else if (req->is_announce()) {
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if (rtsp_tcUrl.empty()) {
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rtsp_tcUrl = req->uri;
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}
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size_t pos = string::npos;
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if ((pos = rtsp_tcUrl.rfind(".sdp")) != string::npos) {
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rtsp_tcUrl = rtsp_tcUrl.substr(0, pos);
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}
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if ((pos = rtsp_tcUrl.rfind("/")) != string::npos) {
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rtsp_stream = rtsp_tcUrl.substr(pos + 1);
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rtsp_tcUrl = rtsp_tcUrl.substr(0, pos);
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}
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srs_assert(req->sdp);
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video_id = ::atoi(req->sdp->video_stream_id.c_str());
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audio_id = ::atoi(req->sdp->audio_stream_id.c_str());
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@ -186,13 +273,13 @@ int SrsRtspConn::do_cycle()
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audio_codec = req->sdp->audio_codec;
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audio_sample_rate = ::atoi(req->sdp->audio_sample_rate.c_str());
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audio_channel = ::atoi(req->sdp->audio_channel.c_str());
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sps = req->sdp->video_sps;
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pps = req->sdp->video_pps;
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asc = req->sdp->audio_sh;
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srs_trace("rtsp: video(#%d, %s, %s/%s), audio(#%d, %s, %s/%s, %dHZ %dchannels)",
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h264_sps = req->sdp->video_sps;
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h264_pps = req->sdp->video_pps;
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aac_specific_config = req->sdp->audio_sh;
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srs_trace("rtsp: video(#%d, %s, %s/%s), audio(#%d, %s, %s/%s, %dHZ %dchannels), %s/%s",
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video_id, video_codec.c_str(), req->sdp->video_protocol.c_str(), req->sdp->video_transport_format.c_str(),
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audio_id, audio_codec.c_str(), req->sdp->audio_protocol.c_str(), req->sdp->audio_transport_format.c_str(),
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audio_sample_rate, audio_channel
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audio_sample_rate, audio_channel, rtsp_tcUrl.c_str(), rtsp_stream.c_str()
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);
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SrsRtspResponse* res = new SrsRtspResponse(req->seq);
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@ -262,9 +349,38 @@ int SrsRtspConn::do_cycle()
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return ret;
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}
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int SrsRtspConn::on_rtp_packet(SrsRtpPacket* pkt)
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int SrsRtspConn::on_rtp_packet(SrsRtpPacket* pkt, int stream_id)
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{
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int ret = ERROR_SUCCESS;
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// ensure rtmp connected.
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if ((ret = connect()) != ERROR_SUCCESS) {
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return ret;
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}
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if (stream_id == video_id) {
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// rtsp tbn is ts tbn.
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int64_t pts = pkt->timestamp;
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if ((ret = vjitter->correct(pts)) != ERROR_SUCCESS) {
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srs_error("rtsp: correct by jitter failed. ret=%d", ret);
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return ret;
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}
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// TODO: FIXME: set dts to pts, please finger out the right dts.
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int64_t dts = pts;
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return on_rtp_video(pkt, dts, pts);
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} else {
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// rtsp tbn is ts tbn.
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int64_t pts = pkt->timestamp;
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if ((ret = ajitter->correct(pts)) != ERROR_SUCCESS) {
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srs_error("rtsp: correct by jitter failed. ret=%d", ret);
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return ret;
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}
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return on_rtp_audio(pkt, pts);
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}
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return ret;
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}
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@ -307,6 +423,336 @@ void SrsRtspConn::on_thread_stop()
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caster->remove(this);
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}
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int SrsRtspConn::on_rtp_video(SrsRtpPacket* pkt, int64_t dts, int64_t pts)
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{
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int ret = ERROR_SUCCESS;
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if ((ret = kickoff_audio_cache(pkt, dts)) != ERROR_SUCCESS) {
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return ret;
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}
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if ((ret = write_h264_ipb_frame(pkt->payload->bytes(), pkt->payload->length(), dts / 90, pts / 90)) != ERROR_SUCCESS) {
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return ret;
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}
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return ret;
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}
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int SrsRtspConn::on_rtp_audio(SrsRtpPacket* pkt, int64_t dts)
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{
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int ret = ERROR_SUCCESS;
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if ((ret = kickoff_audio_cache(pkt, dts)) != ERROR_SUCCESS) {
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return ret;
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}
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// cache current audio to kickoff.
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acache->dts = dts;
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acache->audio_samples = pkt->audio_samples;
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acache->payload = pkt->payload;
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pkt->audio_samples = NULL;
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pkt->payload = NULL;
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return ret;
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}
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int SrsRtspConn::kickoff_audio_cache(SrsRtpPacket* pkt, int64_t dts)
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{
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int ret = ERROR_SUCCESS;
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// nothing to kick off.
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if (!acache->payload) {
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return ret;
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}
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if (dts - acache->dts > 0 && acache->audio_samples->nb_sample_units > 0) {
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int64_t delta = (dts - acache->dts) / acache->audio_samples->nb_sample_units;
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for (int i = 0; i < acache->audio_samples->nb_sample_units; i++) {
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char* frame = acache->audio_samples->sample_units[i].bytes;
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int nb_frame = acache->audio_samples->sample_units[i].size;
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int64_t timestamp = (acache->dts + delta * i) / 90;
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acodec->aac_packet_type = 1;
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if ((ret = write_audio_raw_frame(frame, nb_frame, acodec, timestamp)) != ERROR_SUCCESS) {
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return ret;
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}
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}
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}
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acache->dts = 0;
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srs_freep(acache->audio_samples);
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srs_freep(acache->payload);
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return ret;
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}
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int SrsRtspConn::write_sequence_header()
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{
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int ret = ERROR_SUCCESS;
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// use the current dts.
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int64_t dts = vjitter->timestamp() / 90;
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// send video sps/pps
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if ((ret = write_h264_sps_pps(dts, dts)) != ERROR_SUCCESS) {
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return ret;
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}
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// generate audio sh by audio specific config.
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if (true) {
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std::string sh = aac_specific_config;
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SrsAvcAacCodec dec;
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if ((ret = dec.audio_aac_sequence_header_demux((char*)sh.c_str(), (int)sh.length())) != ERROR_SUCCESS) {
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return ret;
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}
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acodec->sound_format = SrsCodecAudioAAC;
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acodec->sound_type = (dec.aac_channels == 2)? SrsCodecAudioSoundTypeStereo : SrsCodecAudioSoundTypeMono;
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acodec->sound_size = SrsCodecAudioSampleSize16bit;
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acodec->aac_packet_type = 0;
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static int aac_sample_rates[] = {
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96000, 88200, 64000, 48000,
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44100, 32000, 24000, 22050,
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16000, 12000, 11025, 8000,
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7350, 0, 0, 0
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};
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switch (aac_sample_rates[dec.aac_sample_rate]) {
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case 11025:
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acodec->sound_rate = SrsCodecAudioSampleRate11025;
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break;
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case 22050:
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acodec->sound_rate = SrsCodecAudioSampleRate22050;
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break;
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case 44100:
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acodec->sound_rate = SrsCodecAudioSampleRate44100;
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break;
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default:
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break;
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};
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if ((ret = write_audio_raw_frame((char*)sh.data(), (int)sh.length(), acodec, dts)) != ERROR_SUCCESS) {
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return ret;
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}
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}
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return ret;
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}
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int SrsRtspConn::write_h264_sps_pps(u_int32_t dts, u_int32_t pts)
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{
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int ret = ERROR_SUCCESS;
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// h264 raw to h264 packet.
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std::string sh;
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if ((ret = avc->mux_sequence_header(h264_sps, h264_pps, dts, pts, sh)) != ERROR_SUCCESS) {
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return ret;
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}
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// h264 packet to flv packet.
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int8_t frame_type = SrsCodecVideoAVCFrameKeyFrame;
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int8_t avc_packet_type = SrsCodecVideoAVCTypeSequenceHeader;
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char* flv = NULL;
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int nb_flv = 0;
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if ((ret = avc->mux_avc2flv(sh, frame_type, avc_packet_type, dts, pts, &flv, &nb_flv)) != ERROR_SUCCESS) {
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return ret;
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}
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// the timestamp in rtmp message header is dts.
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u_int32_t timestamp = dts;
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if ((ret = rtmp_write_packet(SrsCodecFlvTagVideo, timestamp, flv, nb_flv)) != ERROR_SUCCESS) {
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return ret;
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}
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return ret;
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}
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int SrsRtspConn::write_h264_ipb_frame(char* frame, int frame_size, u_int32_t dts, u_int32_t pts)
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{
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int ret = ERROR_SUCCESS;
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std::string ibp;
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int8_t frame_type;
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if ((ret = avc->mux_ipb_frame(frame, frame_size, dts, pts, ibp, frame_type)) != ERROR_SUCCESS) {
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return ret;
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}
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int8_t avc_packet_type = SrsCodecVideoAVCTypeNALU;
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char* flv = NULL;
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int nb_flv = 0;
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if ((ret = avc->mux_avc2flv(ibp, frame_type, avc_packet_type, dts, pts, &flv, &nb_flv)) != ERROR_SUCCESS) {
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return ret;
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}
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// the timestamp in rtmp message header is dts.
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u_int32_t timestamp = dts;
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return rtmp_write_packet(SrsCodecFlvTagVideo, timestamp, flv, nb_flv);
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}
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int SrsRtspConn::write_audio_raw_frame(char* frame, int frame_size, SrsRawAacStreamCodec* codec, u_int32_t dts)
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{
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int ret = ERROR_SUCCESS;
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char* data = NULL;
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int size = 0;
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if ((ret = aac->mux_aac2flv(frame, frame_size, codec, dts, &data, &size)) != ERROR_SUCCESS) {
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return ret;
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}
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return rtmp_write_packet(SrsCodecFlvTagAudio, dts, data, size);
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}
|
||||
|
||||
int SrsRtspConn::rtmp_write_packet(char type, u_int32_t timestamp, char* data, int size)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
SrsSharedPtrMessage* msg = NULL;
|
||||
|
||||
if ((ret = srs_rtmp_create_msg(type, timestamp, data, size, stream_id, &msg)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: create shared ptr msg failed. ret=%d", ret);
|
||||
return ret;
|
||||
}
|
||||
srs_assert(msg);
|
||||
|
||||
// send out encoded msg.
|
||||
if ((ret = client->send_and_free_message(msg, stream_id)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
// TODO: FIXME: merge all client code.
|
||||
int SrsRtspConn::connect()
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
// when ok, ignore.
|
||||
if (io || client) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
// parse uri
|
||||
if (!req) {
|
||||
req = new SrsRequest();
|
||||
|
||||
std::string schema, host, vhost, app, port, param;
|
||||
srs_discovery_tc_url(rtsp_tcUrl, schema, host, vhost, app, port, param);
|
||||
|
||||
// generate output by template.
|
||||
std::string output = output_template;
|
||||
output = srs_string_replace(output, "[app]", app);
|
||||
output = srs_string_replace(output, "[stream]", rtsp_stream);
|
||||
|
||||
size_t pos = string::npos;
|
||||
string uri = req->tcUrl = output;
|
||||
|
||||
// tcUrl, stream
|
||||
if ((pos = uri.rfind("/")) != string::npos) {
|
||||
req->stream = uri.substr(pos + 1);
|
||||
req->tcUrl = uri = uri.substr(0, pos);
|
||||
}
|
||||
|
||||
srs_discovery_tc_url(req->tcUrl,
|
||||
req->schema, req->host, req->vhost, req->app, req->port,
|
||||
req->param);
|
||||
}
|
||||
|
||||
// connect host.
|
||||
if ((ret = srs_socket_connect(req->host, ::atoi(req->port.c_str()), ST_UTIME_NO_TIMEOUT, &stfd)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: connect server %s:%s failed. ret=%d", req->host.c_str(), req->port.c_str(), ret);
|
||||
return ret;
|
||||
}
|
||||
io = new SrsStSocket(stfd);
|
||||
client = new SrsRtmpClient(io);
|
||||
|
||||
client->set_recv_timeout(SRS_CONSTS_RTMP_RECV_TIMEOUT_US);
|
||||
client->set_send_timeout(SRS_CONSTS_RTMP_SEND_TIMEOUT_US);
|
||||
|
||||
// connect to vhost/app
|
||||
if ((ret = client->handshake()) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: handshake with server failed. ret=%d", ret);
|
||||
return ret;
|
||||
}
|
||||
if ((ret = connect_app(req->host, req->port)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: connect with server failed. ret=%d", ret);
|
||||
return ret;
|
||||
}
|
||||
if ((ret = client->create_stream(stream_id)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: connect with server failed, stream_id=%d. ret=%d", stream_id, ret);
|
||||
return ret;
|
||||
}
|
||||
|
||||
// publish.
|
||||
if ((ret = client->publish(req->stream, stream_id)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: publish failed, stream=%s, stream_id=%d. ret=%d",
|
||||
req->stream.c_str(), stream_id, ret);
|
||||
return ret;
|
||||
}
|
||||
|
||||
return write_sequence_header();
|
||||
}
|
||||
|
||||
// TODO: FIXME: refine the connect_app.
|
||||
int SrsRtspConn::connect_app(string ep_server, string ep_port)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
// args of request takes the srs info.
|
||||
if (req->args == NULL) {
|
||||
req->args = SrsAmf0Any::object();
|
||||
}
|
||||
|
||||
// notify server the edge identity,
|
||||
// @see https://github.com/winlinvip/simple-rtmp-server/issues/147
|
||||
SrsAmf0Object* data = req->args;
|
||||
data->set("srs_sig", SrsAmf0Any::str(RTMP_SIG_SRS_KEY));
|
||||
data->set("srs_server", SrsAmf0Any::str(RTMP_SIG_SRS_KEY" "RTMP_SIG_SRS_VERSION" ("RTMP_SIG_SRS_URL_SHORT")"));
|
||||
data->set("srs_license", SrsAmf0Any::str(RTMP_SIG_SRS_LICENSE));
|
||||
data->set("srs_role", SrsAmf0Any::str(RTMP_SIG_SRS_ROLE));
|
||||
data->set("srs_url", SrsAmf0Any::str(RTMP_SIG_SRS_URL));
|
||||
data->set("srs_version", SrsAmf0Any::str(RTMP_SIG_SRS_VERSION));
|
||||
data->set("srs_site", SrsAmf0Any::str(RTMP_SIG_SRS_WEB));
|
||||
data->set("srs_email", SrsAmf0Any::str(RTMP_SIG_SRS_EMAIL));
|
||||
data->set("srs_copyright", SrsAmf0Any::str(RTMP_SIG_SRS_COPYRIGHT));
|
||||
data->set("srs_primary", SrsAmf0Any::str(RTMP_SIG_SRS_PRIMARY));
|
||||
data->set("srs_authors", SrsAmf0Any::str(RTMP_SIG_SRS_AUTHROS));
|
||||
// for edge to directly get the id of client.
|
||||
data->set("srs_pid", SrsAmf0Any::number(getpid()));
|
||||
data->set("srs_id", SrsAmf0Any::number(_srs_context->get_id()));
|
||||
|
||||
// local ip of edge
|
||||
std::vector<std::string> ips = srs_get_local_ipv4_ips();
|
||||
assert(_srs_config->get_stats_network() < (int)ips.size());
|
||||
std::string local_ip = ips[_srs_config->get_stats_network()];
|
||||
data->set("srs_server_ip", SrsAmf0Any::str(local_ip.c_str()));
|
||||
|
||||
// generate the tcUrl
|
||||
std::string param = "";
|
||||
std::string tc_url = srs_generate_tc_url(ep_server, req->vhost, req->app, ep_port, param);
|
||||
|
||||
// upnode server identity will show in the connect_app of client.
|
||||
// @see https://github.com/winlinvip/simple-rtmp-server/issues/160
|
||||
// the debug_srs_upnode is config in vhost and default to true.
|
||||
bool debug_srs_upnode = _srs_config->get_debug_srs_upnode(req->vhost);
|
||||
if ((ret = client->connect_app(req->app, tc_url, req, debug_srs_upnode)) != ERROR_SUCCESS) {
|
||||
srs_error("rtsp: connect with server failed, tcUrl=%s, dsu=%d. ret=%d",
|
||||
tc_url.c_str(), debug_srs_upnode, ret);
|
||||
return ret;
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
void SrsRtspConn::close()
|
||||
{
|
||||
srs_freep(client);
|
||||
srs_freep(io);
|
||||
srs_freep(req);
|
||||
srs_close_stfd(stfd);
|
||||
}
|
||||
|
||||
SrsRtspCaster::SrsRtspCaster(SrsConfDirective* c)
|
||||
{
|
||||
// TODO: FIXME: support reload.
|
||||
|
|
|
|||
|
|
@ -46,6 +46,15 @@ class SrsRtspStack;
|
|||
class SrsRtspCaster;
|
||||
class SrsConfDirective;
|
||||
class SrsRtpPacket;
|
||||
class SrsRequest;
|
||||
class SrsStSocket;
|
||||
class SrsRtmpClient;
|
||||
class SrsRawH264Stream;
|
||||
class SrsRawAacStream;
|
||||
class SrsRawAacStreamCodec;
|
||||
class SrsSharedPtrMessage;
|
||||
class SrsCodecSample;
|
||||
class SrsSimpleBuffer;
|
||||
|
||||
/**
|
||||
* a rtp connection which transport a stream.
|
||||
|
|
@ -69,13 +78,46 @@ public:
|
|||
virtual int on_udp_packet(sockaddr_in* from, char* buf, int nb_buf);
|
||||
};
|
||||
|
||||
/**
|
||||
* audio is group by frames.
|
||||
*/
|
||||
struct SrsRtspAudioCache
|
||||
{
|
||||
int64_t dts;
|
||||
SrsCodecSample* audio_samples;
|
||||
SrsSimpleBuffer* payload;
|
||||
|
||||
SrsRtspAudioCache();
|
||||
virtual ~SrsRtspAudioCache();
|
||||
};
|
||||
|
||||
/**
|
||||
* the time jitter correct for rtsp.
|
||||
*/
|
||||
class SrsRtspJitter
|
||||
{
|
||||
private:
|
||||
int64_t previous_timestamp;
|
||||
int64_t pts;
|
||||
int delta;
|
||||
public:
|
||||
SrsRtspJitter();
|
||||
virtual ~SrsRtspJitter();
|
||||
public:
|
||||
virtual int64_t timestamp();
|
||||
virtual int correct(int64_t& ts);
|
||||
};
|
||||
|
||||
/**
|
||||
* the rtsp connection serve the fd.
|
||||
*/
|
||||
class SrsRtspConn : public ISrsThreadHandler
|
||||
{
|
||||
private:
|
||||
std::string output;
|
||||
std::string output_template;
|
||||
std::string rtsp_tcUrl;
|
||||
std::string rtsp_stream;
|
||||
|
||||
private:
|
||||
std::string session;
|
||||
// video stream.
|
||||
|
|
@ -88,17 +130,28 @@ private:
|
|||
int audio_sample_rate;
|
||||
int audio_channel;
|
||||
SrsRtpConn* audio_rtp;
|
||||
// video sequence header.
|
||||
std::string sps;
|
||||
std::string pps;
|
||||
// audio sequence header.
|
||||
std::string asc;
|
||||
private:
|
||||
st_netfd_t stfd;
|
||||
SrsStSocket* skt;
|
||||
SrsRtspStack* rtsp;
|
||||
SrsRtspCaster* caster;
|
||||
SrsThread* trd;
|
||||
private:
|
||||
SrsRequest* req;
|
||||
SrsStSocket* io;
|
||||
SrsRtmpClient* client;
|
||||
SrsRtspJitter* vjitter;
|
||||
SrsRtspJitter* ajitter;
|
||||
int stream_id;
|
||||
private:
|
||||
SrsRawH264Stream* avc;
|
||||
std::string h264_sps;
|
||||
std::string h264_pps;
|
||||
private:
|
||||
SrsRawAacStream* aac;
|
||||
SrsRawAacStreamCodec* acodec;
|
||||
std::string aac_specific_config;
|
||||
SrsRtspAudioCache* acache;
|
||||
public:
|
||||
SrsRtspConn(SrsRtspCaster* c, st_netfd_t fd, std::string o);
|
||||
virtual ~SrsRtspConn();
|
||||
|
|
@ -108,11 +161,28 @@ private:
|
|||
virtual int do_cycle();
|
||||
// internal methods
|
||||
public:
|
||||
virtual int on_rtp_packet(SrsRtpPacket* pkt);
|
||||
virtual int on_rtp_packet(SrsRtpPacket* pkt, int stream_id);
|
||||
// interface ISrsThreadHandler
|
||||
public:
|
||||
virtual int cycle();
|
||||
virtual void on_thread_stop();
|
||||
private:
|
||||
virtual int on_rtp_video(SrsRtpPacket* pkt, int64_t dts, int64_t pts);
|
||||
virtual int on_rtp_audio(SrsRtpPacket* pkt, int64_t dts);
|
||||
virtual int kickoff_audio_cache(SrsRtpPacket* pkt, int64_t dts);
|
||||
private:
|
||||
virtual int write_sequence_header();
|
||||
virtual int write_h264_sps_pps(u_int32_t dts, u_int32_t pts);
|
||||
virtual int write_h264_ipb_frame(char* frame, int frame_size, u_int32_t dts, u_int32_t pts);
|
||||
virtual int write_audio_raw_frame(char* frame, int frame_size, SrsRawAacStreamCodec* codec, u_int32_t dts);
|
||||
virtual int rtmp_write_packet(char type, u_int32_t timestamp, char* data, int size);
|
||||
private:
|
||||
// connect to rtmp output url.
|
||||
// @remark ignore when not connected, reconnect when disconnected.
|
||||
virtual int connect();
|
||||
virtual int connect_app(std::string ep_server, std::string ep_port);
|
||||
// close the connected io and rtmp to ready to be re-connect.
|
||||
virtual void close();
|
||||
};
|
||||
|
||||
/**
|
||||
|
|
|
|||
|
|
@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
|
|||
// current release version
|
||||
#define VERSION_MAJOR 2
|
||||
#define VERSION_MINOR 0
|
||||
#define VERSION_REVISION 119
|
||||
#define VERSION_REVISION 120
|
||||
|
||||
// server info.
|
||||
#define RTMP_SIG_SRS_KEY "SRS"
|
||||
|
|
|
|||
|
|
@ -279,58 +279,12 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
|
|||
srs_freep(aac_extra_data);
|
||||
aac_extra_data = new char[aac_extra_size];
|
||||
memcpy(aac_extra_data, stream->data() + stream->pos(), aac_extra_size);
|
||||
}
|
||||
|
||||
// only need to decode the first 2bytes:
|
||||
// audioObjectType, aac_profile, 5bits.
|
||||
// samplingFrequencyIndex, aac_sample_rate, 4bits.
|
||||
// channelConfiguration, aac_channels, 4bits
|
||||
if (!stream->require(2)) {
|
||||
ret = ERROR_HLS_DECODE_ERROR;
|
||||
srs_error("audio codec decode aac sequence header failed. ret=%d", ret);
|
||||
return ret;
|
||||
}
|
||||
u_int8_t profile_ObjectType = stream->read_1bytes();
|
||||
u_int8_t samplingFrequencyIndex = stream->read_1bytes();
|
||||
|
||||
aac_channels = (samplingFrequencyIndex >> 3) & 0x0f;
|
||||
samplingFrequencyIndex = ((profile_ObjectType << 1) & 0x0e) | ((samplingFrequencyIndex >> 7) & 0x01);
|
||||
profile_ObjectType = (profile_ObjectType >> 3) & 0x1f;
|
||||
|
||||
// set the aac sample rate.
|
||||
aac_sample_rate = samplingFrequencyIndex;
|
||||
|
||||
// the profile = object_id + 1
|
||||
// @see aac-mp4a-format-ISO_IEC_14496-3+2001.pdf, page 78,
|
||||
// Table 1. A.9 ¨C MPEG-2 Audio profiles and MPEG-4 Audio object types
|
||||
aac_profile = profile_ObjectType + 1;
|
||||
|
||||
// the valid aac profile:
|
||||
// MPEG-2 profile
|
||||
// Main profile (ID == 1)
|
||||
// Low Complexity profile (LC) (ID == 2)
|
||||
// Scalable Sampling Rate profile (SSR) (ID == 3)
|
||||
// (reserved) (ID == 4)
|
||||
// @see aac-mp4a-format-ISO_IEC_14496-3+2001.pdf, page 78,
|
||||
// Table 1. A.9 ¨C MPEG-2 Audio profiles and MPEG-4 Audio object types
|
||||
if (aac_profile > 4) {
|
||||
ret = ERROR_HLS_DECODE_ERROR;
|
||||
srs_error("audio codec decode aac sequence header failed, "
|
||||
"adts object=%d invalid. ret=%d", profile_ObjectType, ret);
|
||||
return ret;
|
||||
// demux the sequence header.
|
||||
if ((ret = audio_aac_sequence_header_demux(aac_extra_data, aac_extra_size)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
}
|
||||
|
||||
// TODO: FIXME: to support aac he/he-v2, see: ngx_rtmp_codec_parse_aac_header
|
||||
// @see: https://github.com/winlinvip/nginx-rtmp-module/commit/3a5f9eea78fc8d11e8be922aea9ac349b9dcbfc2
|
||||
//
|
||||
// donot force to LC, @see: https://github.com/winlinvip/simple-rtmp-server/issues/81
|
||||
// the source will print the sequence header info.
|
||||
//if (aac_profile > 3) {
|
||||
// Mark all extended profiles as LC
|
||||
// to make Android as happy as possible.
|
||||
// @see: ngx_rtmp_hls_parse_aac_header
|
||||
//aac_profile = 1;
|
||||
//}
|
||||
} else if (aac_packet_type == SrsCodecAudioTypeRawData) {
|
||||
// ensure the sequence header demuxed
|
||||
if (aac_extra_size <= 0 || !aac_extra_data) {
|
||||
|
|
@ -403,6 +357,68 @@ int SrsAvcAacCodec::audio_mp3_demux(char* data, int size, SrsCodecSample* sample
|
|||
return ret;
|
||||
}
|
||||
|
||||
int SrsAvcAacCodec::audio_aac_sequence_header_demux(char* data, int size)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
if ((ret = stream->initialize(data, size)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
// only need to decode the first 2bytes:
|
||||
// audioObjectType, aac_profile, 5bits.
|
||||
// samplingFrequencyIndex, aac_sample_rate, 4bits.
|
||||
// channelConfiguration, aac_channels, 4bits
|
||||
if (!stream->require(2)) {
|
||||
ret = ERROR_HLS_DECODE_ERROR;
|
||||
srs_error("audio codec decode aac sequence header failed. ret=%d", ret);
|
||||
return ret;
|
||||
}
|
||||
u_int8_t profile_ObjectType = stream->read_1bytes();
|
||||
u_int8_t samplingFrequencyIndex = stream->read_1bytes();
|
||||
|
||||
aac_channels = (samplingFrequencyIndex >> 3) & 0x0f;
|
||||
samplingFrequencyIndex = ((profile_ObjectType << 1) & 0x0e) | ((samplingFrequencyIndex >> 7) & 0x01);
|
||||
profile_ObjectType = (profile_ObjectType >> 3) & 0x1f;
|
||||
|
||||
// set the aac sample rate.
|
||||
aac_sample_rate = samplingFrequencyIndex;
|
||||
|
||||
// the profile = object_id + 1
|
||||
// @see aac-mp4a-format-ISO_IEC_14496-3+2001.pdf, page 78,
|
||||
// Table 1. A.9 ¨C MPEG-2 Audio profiles and MPEG-4 Audio object types
|
||||
aac_profile = profile_ObjectType + 1;
|
||||
|
||||
// the valid aac profile:
|
||||
// MPEG-2 profile
|
||||
// Main profile (ID == 1)
|
||||
// Low Complexity profile (LC) (ID == 2)
|
||||
// Scalable Sampling Rate profile (SSR) (ID == 3)
|
||||
// (reserved) (ID == 4)
|
||||
// @see aac-mp4a-format-ISO_IEC_14496-3+2001.pdf, page 78,
|
||||
// Table 1. A.9 ¨C MPEG-2 Audio profiles and MPEG-4 Audio object types
|
||||
if (aac_profile > 4) {
|
||||
ret = ERROR_HLS_DECODE_ERROR;
|
||||
srs_error("audio codec decode aac sequence header failed, "
|
||||
"adts object=%d invalid. ret=%d", profile_ObjectType, ret);
|
||||
return ret;
|
||||
}
|
||||
|
||||
// TODO: FIXME: to support aac he/he-v2, see: ngx_rtmp_codec_parse_aac_header
|
||||
// @see: https://github.com/winlinvip/nginx-rtmp-module/commit/3a5f9eea78fc8d11e8be922aea9ac349b9dcbfc2
|
||||
//
|
||||
// donot force to LC, @see: https://github.com/winlinvip/simple-rtmp-server/issues/81
|
||||
// the source will print the sequence header info.
|
||||
//if (aac_profile > 3) {
|
||||
// Mark all extended profiles as LC
|
||||
// to make Android as happy as possible.
|
||||
// @see: ngx_rtmp_hls_parse_aac_header
|
||||
//aac_profile = 1;
|
||||
//}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
|
|
|||
|
|
@ -475,6 +475,11 @@ public:
|
|||
* demux the h.264 NALUs to sampe units.
|
||||
*/
|
||||
virtual int video_avc_demux(char* data, int size, SrsCodecSample* sample);
|
||||
public:
|
||||
/**
|
||||
* directly demux the sequence header, without RTMP packet header.
|
||||
*/
|
||||
virtual int audio_aac_sequence_header_demux(char* data, int size);
|
||||
private:
|
||||
/**
|
||||
* when avc packet type is SrsCodecVideoAVCTypeSequenceHeader,
|
||||
|
|
|
|||
|
|
@ -147,6 +147,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
|
|||
#define ERROR_RTP_HEADER_CORRUPT 2044
|
||||
#define ERROR_RTP_TYPE96_CORRUPT 2045
|
||||
#define ERROR_RTP_TYPE97_CORRUPT 2046
|
||||
#define ERROR_RTSP_AUDIO_CONFIG 2047
|
||||
//
|
||||
// system control message,
|
||||
// not an error, but special control logic.
|
||||
|
|
|
|||
|
|
@ -621,3 +621,41 @@ char* srs_av_base64_encode(char* out, int out_size, const u_int8_t* in, int in_s
|
|||
return ret;
|
||||
}
|
||||
|
||||
#define SPACE_CHARS " \t\r\n"
|
||||
|
||||
int av_toupper(int c)
|
||||
{
|
||||
if (c >= 'a' && c <= 'z') {
|
||||
c ^= 0x20;
|
||||
}
|
||||
return c;
|
||||
}
|
||||
|
||||
int ff_hex_to_data(u_int8_t* data, const char* p)
|
||||
{
|
||||
int c, len, v;
|
||||
|
||||
len = 0;
|
||||
v = 1;
|
||||
for (;;) {
|
||||
p += strspn(p, SPACE_CHARS);
|
||||
if (*p == '\0')
|
||||
break;
|
||||
c = av_toupper((unsigned char) *p++);
|
||||
if (c >= '0' && c <= '9')
|
||||
c = c - '0';
|
||||
else if (c >= 'A' && c <= 'F')
|
||||
c = c - 'A' + 10;
|
||||
else
|
||||
break;
|
||||
v = (v << 4) | c;
|
||||
if (v & 0x100) {
|
||||
if (data)
|
||||
data[len] = v;
|
||||
len++;
|
||||
v = 1;
|
||||
}
|
||||
}
|
||||
return len;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -115,5 +115,12 @@ extern char* srs_av_base64_encode(char* out, int out_size, const u_int8_t* in, i
|
|||
*/
|
||||
#define SRS_AV_BASE64_SIZE(x) (((x)+2) / 3 * 4 + 1)
|
||||
|
||||
/**
|
||||
* convert hex string to data.
|
||||
* for example, p=config='139056E5A0'
|
||||
* output hex to data={0x13, 0x90, 0x56, 0xe5, 0xa0}
|
||||
*/
|
||||
extern int ff_hex_to_data(u_int8_t* data, const char* p);
|
||||
|
||||
#endif
|
||||
|
||||
|
|
|
|||
|
|
@ -200,6 +200,8 @@ int SrsRtpPacket::decode(SrsStream* stream)
|
|||
timestamp = stream->read_4bytes();
|
||||
ssrc = stream->read_4bytes();
|
||||
|
||||
// TODO: FIXME: check sequence number.
|
||||
|
||||
// video codec.
|
||||
if (payload_type == 96) {
|
||||
return decode_96(stream);
|
||||
|
|
@ -232,7 +234,6 @@ int SrsRtpPacket::decode_97(SrsStream* stream)
|
|||
}
|
||||
|
||||
int nb_samples = au_size / 2;
|
||||
int guess_sample_size = (stream->size() - stream->pos() - au_size) / nb_samples;
|
||||
int required_size = 0;
|
||||
|
||||
// append left bytes to payload.
|
||||
|
|
@ -247,11 +248,9 @@ int SrsRtpPacket::decode_97(SrsStream* stream)
|
|||
lasv = stream->read_1bytes();
|
||||
|
||||
u_int16_t sample_size = ((hasv << 5) & 0xE0) | ((lasv >> 3) & 0x1f);
|
||||
if (sample_size != guess_sample_size) {
|
||||
// guess the size lost 0x100.
|
||||
if (guess_sample_size == (sample_size | 0x100)) {
|
||||
sample_size = guess_sample_size;
|
||||
}
|
||||
// TODO: FIXME: finger out how to parse the size of sample.
|
||||
if (sample_size < 0x100 && stream->require(required_size + sample_size + 0x100)) {
|
||||
sample_size = sample_size | 0x100;
|
||||
}
|
||||
|
||||
char* sample = p + required_size;
|
||||
|
|
@ -541,7 +540,17 @@ int SrsRtspSdp::parse_fmtp_attribute(string attr)
|
|||
} else if (item_key == "indexdeltalength") {
|
||||
audio_index_delta_length = item_value;
|
||||
} else if (item_key == "config") {
|
||||
audio_sh = base64_decode(item_value);
|
||||
if (item_value.length() <= 0) {
|
||||
ret = ERROR_RTSP_AUDIO_CONFIG;
|
||||
srs_error("rtsp: audio config failed. ret=%d", ret);
|
||||
return ret;
|
||||
}
|
||||
|
||||
char* tmp_sh = new char[item_value.length()];
|
||||
SrsAutoFree(char, tmp_sh);
|
||||
int nb_tmp_sh = ff_hex_to_data((u_int8_t*)tmp_sh, item_value.c_str());
|
||||
srs_assert(nb_tmp_sh > 0);
|
||||
audio_sh.append(tmp_sh, nb_tmp_sh);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue