Merge branch 'develop' into min

AUTHORS.txt

@@ -53,4 +53,6 @@ CONTRIBUTORS ordered by first contribution.
 * alphonsetai<tyh_123@163.com>
 * Michael.Ma<wnpllr@gmail.com>
 * lam2003<linmin3@yy.com>
-* runner365<shi.weibd@hotmail.com>
+* runner365<shi.weibd@hotmail.com>
+* XiaofengWang<wasphin@gmail.com>
+* XiaLixin<xialx@yuntongxun.com>

README.md

@@ -153,6 +153,9 @@ For previous versions, please read:
 
 ## V4 changes
 
+* v4.0, 2020-03-07, For [#1612][bug #1612], fix crash bug for RTSP. 4.0.12
+* v4.0, 2020-03-07, For [#1631][bug #1631], support sei_filter for SRT. 4.0.11
+* v4.0, 2020-03-01, For [#1621][bug #1621], support mix_correct for aggregate aac for SRT. 4.0.10
 * v4.0, 2020-02-25, For [#1615][bug #1615], support default app(live) for vmix SRT. 4.0.9
 * v4.0, 2020-02-21, For [#1598][bug #1598], support SLB health checking by TCP. 4.0.8
 * v4.0, 2020-02-19, For [#1579][bug #1579], support rolling update of k8s. 4.0.7
@@ -1687,6 +1690,11 @@ Winlin
 [bug #1579]: https://github.com/ossrs/srs/issues/1579
 [bug #1598]: https://github.com/ossrs/srs/issues/1598
 [bug #1615]: https://github.com/ossrs/srs/issues/1615
+[bug #1621]: https://github.com/ossrs/srs/issues/1621
+[bug #xxxxxxxxxxxxx]: https://github.com/ossrs/srs/issues/xxxxxxxxxxxxx
+
+[bug #1631]: https://github.com/ossrs/srs/issues/1631
+[bug #1612]: https://github.com/ossrs/srs/issues/1612
 [bug #xxxxxxxxxxxxx]: https://github.com/ossrs/srs/issues/xxxxxxxxxxxxx
 
 [exo #828]: https://github.com/google/ExoPlayer/pull/828

trunk/configure

@@ -730,7 +730,6 @@ fi
 # next step
 #####################################################################################
 if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
-    ip=`ifconfig|grep "inet addr"| grep -v "127.0.0.1"|awk '{print $2}'|awk -F ':' 'NR==1 {print $2}'`
     echo ""
     echo "You can run 3rdparty applications:"
     if [ $SRS_HTTP_CALLBACK = YES ]; then

trunk/src/app/srs_app_config.cpp

@@ -3530,7 +3530,7 @@ srs_error_t SrsConfig::check_normal_config()
                 && n != "mss" && n != "latency" && n != "recvlatency"
                 && n != "peerlatency" && n != "tlpkdrop" && n != "connect_timeout"
                 && n != "sendbuf" && n != "recvbuf" && n != "payloadsize"
-                && n != "default_app") {
+                && n != "default_app" && n != "mix_correct" && n != "sei_filter") {
                 return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal srt_stream.%s", n.c_str());
             }
         }
@@ -6754,6 +6754,20 @@ unsigned short SrsConfig::get_srt_listen_port()
     return (unsigned short)atoi(conf->arg0().c_str());
 }
 
+bool SrsConfig::get_srt_mix_correct() {
+    static bool DEFAULT = true;
+    SrsConfDirective* conf = root->get("srt_server");
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("mix_correct");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    return SRS_CONF_PERFER_TRUE(conf->arg0());
+}
+
 int SrsConfig::get_srto_maxbw() {
     static int64_t DEFAULT = -1;
     SrsConfDirective* conf = root->get("srt_server");
@@ -6824,6 +6838,20 @@ int SrsConfig::get_srto_peer_latency() {
     return atoi(conf->arg0().c_str());
 }
 
+bool SrsConfig::get_srt_sei_filter() {
+    static bool DEFAULT = true;
+    SrsConfDirective* conf = root->get("srt_server");
+    if (!conf) {
+        return DEFAULT;
+    }
+    
+    conf = conf->get("sei_filter");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+    return SRS_CONF_PERFER_TRUE(conf->arg0());
+}
+
 bool SrsConfig::get_srto_tlpkdrop() {
     static bool DEFAULT = true;
     SrsConfDirective* conf = root->get("srt_server");

trunk/src/app/srs_app_config.hpp

@@ -613,6 +613,8 @@ public:
     virtual int get_srto_recv_latency();
     // Get the srt SRTO_PEERLATENCY, peer latency, default is 0..
     virtual int get_srto_peer_latency();
+    // Get the srt h264 sei filter, default is on, it will drop h264 sei packet.
+    virtual bool get_srt_sei_filter();
     // Get the srt SRTO_TLPKDROP, Too-late Packet Drop, default is true.
     virtual bool get_srto_tlpkdrop();
     // Get the srt SRTO_CONNTIMEO, connection timeout, default is 3000ms.
@@ -625,6 +627,8 @@ public:
     virtual int get_srto_payloadsize();
     // Get the default app.
     virtual std::string get_default_app_name();
+    // Get the mix_correct
+    virtual bool get_srt_mix_correct();
 
 // http_hooks section
 private:

trunk/src/app/srs_app_rtsp.cpp

@@ -240,6 +240,12 @@ srs_error_t SrsRtspConn::serve()
     return err;
 }
 
+std::string SrsRtspConn::remote_ip()
+{
+    // TODO: FIXME: Implement it.
+    return "";
+}
+
 srs_error_t SrsRtspConn::do_cycle()
 {
     srs_error_t err = srs_success;
@@ -684,6 +690,7 @@ SrsRtspCaster::SrsRtspCaster(SrsConfDirective* c)
     output = _srs_config->get_stream_caster_output(c);
     local_port_min = _srs_config->get_stream_caster_rtp_port_min(c);
     local_port_max = _srs_config->get_stream_caster_rtp_port_max(c);
+    manager = new SrsCoroutineManager();
 }
 
 SrsRtspCaster::~SrsRtspCaster()
@@ -691,10 +698,21 @@ SrsRtspCaster::~SrsRtspCaster()
     std::vector<SrsRtspConn*>::iterator it;
     for (it = clients.begin(); it != clients.end(); ++it) {
         SrsRtspConn* conn = *it;
-        srs_freep(conn);
+        manager->remove(conn);
     }
     clients.clear();
     used_ports.clear();
+
+    srs_freep(manager);
+}
+
+srs_error_t SrsRtspCaster::initialize()
+{
+    srs_error_t err = srs_success;
+    if ((err = manager->start()) != srs_success) {
+        return srs_error_wrap(err, "start manager");
+    }
+    return err;
 }
 
 srs_error_t SrsRtspCaster::alloc_port(int* pport)
@@ -747,6 +765,6 @@ void SrsRtspCaster::remove(SrsRtspConn* conn)
     }
     srs_info("rtsp: remove connection from caster.");
     
-    srs_freep(conn);
+    manager->remove(conn);
 }
 

trunk/src/app/srs_app_rtsp.hpp

@@ -100,7 +100,7 @@ public:
 };
 
 // The rtsp connection serve the fd.
-class SrsRtspConn : public ISrsCoroutineHandler
+class SrsRtspConn : public ISrsCoroutineHandler, public ISrsConnection
 {
 private:
     std::string output_template;
@@ -143,6 +143,7 @@ public:
     virtual ~SrsRtspConn();
 public:
     virtual srs_error_t serve();
+    virtual std::string remote_ip();
 private:
     virtual srs_error_t do_cycle();
 // internal methods
@@ -179,6 +180,7 @@ private:
     std::map<int, bool> used_ports;
 private:
     std::vector<SrsRtspConn*> clients;
+    SrsCoroutineManager* manager;
 public:
     SrsRtspCaster(SrsConfDirective* c);
     virtual ~SrsRtspCaster();
@@ -188,6 +190,7 @@ public:
     virtual srs_error_t alloc_port(int* pport);
     // Free the alloced rtp port.
     virtual void free_port(int lpmin, int lpmax);
+    virtual srs_error_t initialize();
 // Interface ISrsTcpHandler
 public:
     virtual srs_error_t on_tcp_client(srs_netfd_t stfd);

trunk/src/app/srs_app_server.cpp

@@ -179,6 +179,9 @@ SrsRtspListener::SrsRtspListener(SrsServer* svr, SrsListenerType t, SrsConfDirec
     srs_assert(type == SrsListenerRtsp);
     if (type == SrsListenerRtsp) {
         caster = new SrsRtspCaster(c);
+
+        // TODO: FIXME: Must check error.
+        caster->initialize();
     }
 }
 

trunk/src/app/srs_app_server.hpp

@@ -50,6 +50,7 @@ class ISrsUdpHandler;
 class SrsUdpListener;
 class SrsTcpListener;
 class SrsAppCasterFlv;
+class SrsRtspCaster;
 class SrsCoroutineManager;
 
 // The listener type for server to identify the connection,
@@ -107,7 +108,7 @@ class SrsRtspListener : virtual public SrsListener, virtual public ISrsTcpHandle
 {
 private:
     SrsTcpListener* listener;
-    ISrsTcpHandler* caster;
+    SrsRtspCaster* caster;
 public:
     SrsRtspListener(SrsServer* svr, SrsListenerType t, SrsConfDirective* c);
     virtual ~SrsRtspListener();

trunk/src/core/srs_core_version4.hpp

@@ -24,6 +24,6 @@
 #ifndef SRS_CORE_VERSION4_HPP
 #define SRS_CORE_VERSION4_HPP
 
-#define SRS_VERSION4_REVISION 9
+#define SRS_VERSION4_REVISION 12
 
 #endif

trunk/src/srt/srt_handle.cpp

@@ -145,6 +145,8 @@ void srt_handle::add_newconn(SRT_CONN_PTR conn_ptr, int events) {
     srs_trace("srto SRTO_RCVBUF=%d", val_i);
     srt_getsockopt(conn_ptr->get_conn(), 0, SRTO_MAXBW, &val_i, &opt_len);
     srs_trace("srto SRTO_MAXBW=%d", val_i);
+    srs_trace("srt mix_correct is %s.", _srs_config->get_srt_mix_correct() ? "enable" : "disable");
+    srs_trace("srt h264 sei filter is %s.", _srs_config->get_srt_sei_filter() ? "enable" : "disable");
 
     if (conn_ptr->get_mode() == PULL_SRT_MODE) {
         add_new_puller(conn_ptr, conn_ptr->get_subpath());

trunk/src/srt/srt_server.cpp

@@ -257,8 +257,9 @@ void srt_server::on_work()
         }
     }
 
-    // @see 2020-01-28 https://github.com/Haivision/srt/commit/b8c70ec801a56bea151ecce9c09c4ebb720c2f68#diff-fb66028e8746fea578788532533a296bR786
-#if SRT_VERSION_MAJOR > 1 || SRT_VERSION_MINOR > 4 || SRT_VERSION_PATCH > 1
+    // New API at 2020-01-28, >1.4.1
+    // @see https://github.com/Haivision/srt/commit/b8c70ec801a56bea151ecce9c09c4ebb720c2f68#diff-fb66028e8746fea578788532533a296bR786
+#if (SRT_VERSION_MAJOR<<24 | SRT_VERSION_MINOR<<16 | SRT_VERSION_PATCH<<8) > 0x01040100
     srt_epoll_clear_usocks(_pollid);
 #endif
 }

trunk/src/srt/srt_to_rtmp.cpp

@@ -322,12 +322,13 @@ srs_error_t rtmp_client::write_h264_sps_pps(uint32_t dts, uint32_t pts) {
         return srs_error_wrap(err, "avc to flv");
     }
     
-    // the timestamp in rtmp message header is dts.
-    uint32_t timestamp = dts;
-    if ((err = rtmp_write_packet(SrsFrameTypeVideo, timestamp, flv, nb_flv)) != srs_success) {
-        return srs_error_wrap(err, "write packet");
+    if (_srs_config->get_srt_mix_correct()) {
+        _rtmp_queue.insert_rtmp_data((unsigned char*)flv, nb_flv, (int64_t)dts, SrsFrameTypeVideo);
+        rtmp_write_work();
+    } else {
+        rtmp_write_packet(SrsFrameTypeVideo, dts, flv, nb_flv);
     }
-    
+
     // reset sps and pps.
     _h264_sps_changed = false;
     _h264_pps_changed = false;
@@ -367,10 +368,14 @@ srs_error_t rtmp_client::write_h264_ipb_frame(char* frame, int frame_size, uint3
     if ((err = _avc_ptr->mux_avc2flv(ibp, frame_type, avc_packet_type, dts, pts, &flv, &nb_flv)) != srs_success) {
         return srs_error_wrap(err, "mux avc to flv");
     }
-    
-    // the timestamp in rtmp message header is dts.
-    uint32_t timestamp = dts;
-    return rtmp_write_packet(SrsFrameTypeVideo, timestamp, flv, nb_flv);
+    if (_srs_config->get_srt_mix_correct()) {
+        _rtmp_queue.insert_rtmp_data((unsigned char*)flv, nb_flv, (int64_t)dts, SrsFrameTypeVideo);
+        rtmp_write_work();
+    } else {
+        rtmp_write_packet(SrsFrameTypeVideo, dts, flv, nb_flv);
+    }
+
+    return err;
 }
 
 srs_error_t rtmp_client::write_audio_raw_frame(char* frame, int frame_size, SrsRawAacStreamCodec* codec, uint32_t dts) {
@@ -381,14 +386,20 @@ srs_error_t rtmp_client::write_audio_raw_frame(char* frame, int frame_size, SrsR
     if ((err = _aac_ptr->mux_aac2flv(frame, frame_size, codec, dts, &data, &size)) != srs_success) {
         return srs_error_wrap(err, "mux aac to flv");
     }
-    
-    return rtmp_write_packet(SrsFrameTypeAudio, dts, data, size);
+    if (_srs_config->get_srt_mix_correct()) {
+        _rtmp_queue.insert_rtmp_data((unsigned char*)data, size, (int64_t)dts, SrsFrameTypeAudio);
+        rtmp_write_work();
+    } else {
+        rtmp_write_packet(SrsFrameTypeAudio, dts, data, size);
+    }
+
+    return err;
 }
 
 srs_error_t rtmp_client::rtmp_write_packet(char type, uint32_t timestamp, char* data, int size) {
     srs_error_t err = srs_success;
     SrsSharedPtrMessage* msg = NULL;
-    
+
     if ((err = srs_rtmp_create_msg(type, timestamp, data, size, _rtmp_conn_ptr->sid(), &msg)) != srs_success) {
         return srs_error_wrap(err, "create message");
     }
@@ -403,6 +414,19 @@ srs_error_t rtmp_client::rtmp_write_packet(char type, uint32_t timestamp, char*
     return err;
 }
 
+void rtmp_client::rtmp_write_work() {
+    rtmp_packet_info_s packet_info;
+    bool ret = false;
+    
+    do {
+        ret = _rtmp_queue.get_rtmp_data(packet_info);
+        if (ret) {
+            rtmp_write_packet(packet_info._type, packet_info._dts, (char*)packet_info._data, packet_info._len);
+        }
+    } while(ret);
+    return;
+}
+
 srs_error_t rtmp_client::on_ts_video(std::shared_ptr<SrsBuffer> avs_ptr, uint64_t dts, uint64_t pts) {
     srs_error_t err = srs_success;
 
@@ -410,7 +434,13 @@ srs_error_t rtmp_client::on_ts_video(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
     if ((err = connect()) != srs_success) {
         return srs_error_wrap(err, "connect");
     }
-    
+    dts = dts / 90;
+    pts = pts / 90;
+
+    if (dts == 0) {
+        dts = pts;
+    }
+
     // send each frame.
     while (!avs_ptr->empty()) {
         char* frame = NULL;
@@ -429,6 +459,13 @@ srs_error_t rtmp_client::on_ts_video(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
         if (nal_unit_type == SrsAvcNaluTypeAccessUnitDelimiter) {
             continue;
         }
+
+        // TODO: FIXME: Should cache this config, it's better not to get it for each video frame.
+        if (_srs_config->get_srt_sei_filter()) {
+            if (nal_unit_type == SrsAvcNaluTypeSEI) {
+                continue;
+            }
+        }
         
         // for sps
         if (_avc_ptr->is_sps(frame, frame_size)) {
@@ -480,14 +517,47 @@ srs_error_t rtmp_client::on_ts_video(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
     return err;
 }
 
+int rtmp_client::get_sample_rate(char sound_rate) {
+    int sample_rate = 44100;
+
+    switch (sound_rate)
+    {
+    case SrsAudioSampleRate44100:
+        sample_rate = 44100;
+        break;
+    case SrsAudioSampleRate22050:
+        sample_rate = 22050;
+        break;
+    case SrsAudioSampleRate11025:
+        sample_rate = 11025;
+        break;
+    case SrsAudioSampleRate5512:
+        sample_rate = 5512;
+        break;
+    default:
+        break;
+    }
+    return sample_rate;
+}
+
 srs_error_t rtmp_client::on_ts_audio(std::shared_ptr<SrsBuffer> avs_ptr, uint64_t dts, uint64_t pts) {
     srs_error_t err = srs_success;
+    uint64_t base_dts;
+    uint64_t real_dts;
+    uint64_t first_dts;
+    int index = 0;
+    int sample_size = 1024;
 
     // ensure rtmp connected.
     if ((err = connect()) != srs_success) {
         return srs_error_wrap(err, "connect");
     }
     
+    base_dts = dts/90;
+    if (base_dts == 0) {
+        base_dts = pts/90;
+    }
+    
     // send each frame.
     while (!avs_ptr->empty()) {
         char* frame = NULL;
@@ -496,16 +566,24 @@ srs_error_t rtmp_client::on_ts_audio(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
         if ((err = _aac_ptr->adts_demux(avs_ptr.get(), &frame, &frame_size, codec)) != srs_success) {
             return srs_error_wrap(err, "demux adts");
         }
-        //srs_trace("audio annexb demux sampling_frequency_index:%d, aac_packet_type:%d, sound_rate:%d, sound_size:%d", 
-        //    codec.sampling_frequency_index, codec.aac_packet_type, codec.sound_rate,
-        //    codec.sound_size);
-        //srs_trace_data(frame, frame_size, "audio annexb demux:");
-        // ignore invalid frame,
-        //  * atleast 1bytes for aac to decode the data.
+
         if (frame_size <= 0) {
             continue;
         }
-        
+        int sample_rate = get_sample_rate(codec.sound_rate);
+
+        if (codec.aac_packet_type > SrsAudioOpusFrameTraitRaw) {
+            sample_size = 2048;
+        } else {
+            sample_size = 1024;
+        }
+
+        real_dts = base_dts + index * 1000.0 * sample_size / sample_rate;
+        if (index == 0) {
+            first_dts = real_dts;
+        }
+        index++;
+
         // generate sh.
         if (_aac_specific_config.empty()) {
             std::string sh;
@@ -516,19 +594,25 @@ srs_error_t rtmp_client::on_ts_audio(std::shared_ptr<SrsBuffer> avs_ptr, uint64_
             
             codec.aac_packet_type = 0;
             
-            if ((err = write_audio_raw_frame((char*)sh.data(), (int)sh.length(), &codec, dts)) != srs_success) {
+            if ((err = write_audio_raw_frame((char*)sh.data(), (int)sh.length(), &codec, real_dts)) != srs_success) {
                 return srs_error_wrap(err, "write raw audio frame");
             }
         }
         
         // audio raw data.
         codec.aac_packet_type = 1;
-        if ((err = write_audio_raw_frame(frame, frame_size, &codec, dts)) != srs_success) {
+        if ((err = write_audio_raw_frame(frame, frame_size, &codec, real_dts)) != srs_success) {
             return srs_error_wrap(err, "write audio raw frame");
         }
         _last_live_ts = now_ms();
     }
     
+    uint64_t diff_t = real_dts - first_dts;
+    diff_t += 100;
+    if ((diff_t > 200) && (diff_t < 600)) {
+        srs_info("set_queue_timeout timeout:%lu", diff_t);
+        _rtmp_queue.set_queue_timeout(diff_t);
+    }
     return err;
 }
 
@@ -541,16 +625,85 @@ void rtmp_client::on_data_callback(SRT_DATA_MSG_PTR data_ptr, unsigned int media
     }
 
     auto avs_ptr = std::make_shared<SrsBuffer>((char*)data_ptr->get_data(), data_ptr->data_len());
-    dts = dts / 90;
-    pts = pts / 90;
 
     if (media_type == STREAM_TYPE_VIDEO_H264) {
         on_ts_video(avs_ptr, dts, pts);
     } else if (media_type == STREAM_TYPE_AUDIO_AAC) {
         on_ts_audio(avs_ptr, dts, pts);
     } else {
-        srs_error("mpegts demux unkown stream type:0x%02x", media_type);
-        assert(0);
+        srs_error("mpegts demux unkown stream type:0x%02x, only support h264+aac", media_type);
+    }
+    return;
+}
+
+rtmp_packet_queue::rtmp_packet_queue():_queue_timeout(QUEUE_DEF_TIMEOUT)
+    ,_queue_maxlen(QUEUE_LEN_MAX)
+    ,_first_packet_t(-1)
+    ,_first_local_t(-1) {
+
+}
+
+rtmp_packet_queue::~rtmp_packet_queue() {
+    for (auto item : _send_map) {
+        rtmp_packet_info_s info = item.second;
+        if (info._data) {
+            delete info._data;
+        }
+    }
+    _send_map.clear();
+}
+
+void rtmp_packet_queue::set_queue_timeout(int64_t queue_timeout) {
+    _queue_timeout = queue_timeout;
+}
+
+void rtmp_packet_queue::insert_rtmp_data(unsigned char* data, int len, int64_t dts, char media_type) {
+    rtmp_packet_info_s packet_info;
+
+    packet_info._data = data;
+    packet_info._len  = len;
+    packet_info._dts  = dts;
+    packet_info._type = media_type;
+
+    if (_first_packet_t == -1) {
+        _first_packet_t = dts;
+        _first_local_t = (int64_t)now_ms();
     }
+
+    _send_map.insert(std::make_pair(dts, packet_info));
     return;
 }
+
+bool rtmp_packet_queue::is_ready() {
+    if (!_srs_config->get_srt_mix_correct() && !_send_map.empty()) {
+        return true;
+    }
+    if (_send_map.size() < 2) {
+        return false;
+    }
+
+    if (_send_map.size() >= (size_t)_queue_maxlen) {
+        return true;
+    }
+
+    auto first_item = _send_map.begin();
+    int64_t now_t = (int64_t)now_ms();
+
+    int64_t diff_t = (now_t - _first_local_t) - (first_item->first - _first_packet_t);
+
+    if (diff_t >= _queue_timeout) {
+        return true;
+    }
+    return false;
+}
+
+bool rtmp_packet_queue::get_rtmp_data(rtmp_packet_info_s& packet_info) {
+    if (!is_ready()) {
+        return false;
+    }
+    auto iter = _send_map.begin();
+    packet_info = iter->second;
+    _send_map.erase(iter);
+
+    return true;
+}

trunk/src/srt/srt_to_rtmp.hpp

@@ -4,6 +4,7 @@
 #include <string>
 #include <thread>
 #include <queue>
+#include <map>
 #include <mutex>
 #include <condition_variable>
 #include <srs_kernel_ts.hpp>
@@ -24,6 +25,37 @@ typedef std::shared_ptr<SrsRawAacStream> AAC_PTR;
 
 #define DEFAULT_VHOST "__default_host__"
 
+#define QUEUE_DEF_TIMEOUT 500
+#define QUEUE_LEN_MAX     100
+
+typedef struct {
+    unsigned char* _data;
+    int _len;
+    int64_t _dts;
+    char _type;
+    char reserve[3];
+} rtmp_packet_info_s;
+
+class rtmp_packet_queue {
+public:
+    rtmp_packet_queue();
+    ~rtmp_packet_queue();
+
+    void set_queue_timeout(int64_t queue_timeout);
+    void insert_rtmp_data(unsigned char* data, int len, int64_t dts, char media_type);
+    bool get_rtmp_data(rtmp_packet_info_s& packet_info);
+
+private:
+    bool is_ready();
+
+private:
+    int64_t _queue_timeout;
+    int64_t _queue_maxlen;
+    int64_t _first_packet_t;
+    int64_t _first_local_t;
+    std::multimap<int64_t, rtmp_packet_info_s> _send_map;//key:dts, value:rtmp_packet_info
+};
+
 class rtmp_client : public ts_media_data_callback_I, public std::enable_shared_from_this<rtmp_client> {
 public:
     rtmp_client(std::string key_path);
@@ -46,6 +78,10 @@ private:
     virtual srs_error_t write_h264_ipb_frame(char* frame, int frame_size, uint32_t dts, uint32_t pts);
     virtual srs_error_t write_audio_raw_frame(char* frame, int frame_size, SrsRawAacStreamCodec* codec, uint32_t dts);
 
+    int get_sample_rate(char sound_rate);
+
+    void rtmp_write_work();
+
 private:
     virtual srs_error_t rtmp_write_packet(char type, uint32_t timestamp, char* data, int size);
 
@@ -71,6 +107,9 @@ private:
     RTMP_CONN_PTR _rtmp_conn_ptr;
     bool _connect_flag;
     int64_t _last_live_ts;
+
+private:
+    rtmp_packet_queue _rtmp_queue;
 };
 
 typedef std::shared_ptr<rtmp_client> RTMP_CLIENT_PTR;

trunk/src/srt/ts_demux.cpp

@@ -259,11 +259,21 @@ int ts_demux::decode_unit(unsigned char* data_p, std::string key_path, TS_DATA_C
                     if(ts_header_info._payload_unit_start_indicator){
                         unsigned char* ret_data_p = nullptr;
                         size_t ret_size = 0;
+                        uint64_t dts = 0;
+                        uint64_t pts = 0;
                         
                         //callback last media data in data buffer
                         on_callback(callback, _last_pid, key_path, _last_dts, _last_pts);
 
-                        pes_parse(data_p+npos, npos, &ret_data_p, ret_size, _last_dts, _last_pts);
+                        int ret = pes_parse(data_p+npos, npos, &ret_data_p, ret_size, dts, pts);
+                        assert(ret <= 188);
+                        if (ret > 188) {
+                            return -1;
+                        }
+
+                        _last_pts = pts;
+                        _last_dts = (dts == 0) ? pts : dts;
+
                         if ((ret_data_p != nullptr) && (ret_size > 0)) {
                             insert_into_databuf(ret_data_p, ret_size, key_path, ts_header_info._PID);
                         }
@@ -291,7 +301,6 @@ int ts_demux::decode(SRT_DATA_MSG_PTR data_ptr, TS_DATA_CALLBACK_PTR callback)
         return -1;
     }
 
-
     unsigned int count = data_ptr->data_len()/188;
     path = data_ptr->get_path();
     for (unsigned int index = 0; index < count; index++)