Squash: Merge #2721, #2729

README.md

@@ -5,12 +5,12 @@
 [![](https://github.com/ossrs/srs/actions/workflows/release.yml/badge.svg)](https://github.com/ossrs/srs/actions/workflows/release.yml?query=workflow%3ARelease)
 [![](https://github.com/ossrs/srs/actions/workflows/test.yml/badge.svg?branch=develop)](https://github.com/ossrs/srs/actions?query=workflow%3ATest+branch%3Adevelop)
 [![](https://codecov.io/gh/ossrs/srs/branch/develop/graph/badge.svg)](https://codecov.io/gh/ossrs/srs/branch/develop)
-[![](https://gitee.com/winlinvip/srs-wiki/raw/master/images/wechat-badge2.png)](../../wikis/Contact#wechat)
+[![](https://gitee.com/winlinvip/srs-wiki/raw/master/images/wechat-badge4.svg)](../../wikis/Contact#wechat)
-[![](https://gitee.com/winlinvip/srs-wiki/raw/master/images/bbs2.png)](http://bbs.ossrs.net)
+[![](https://gitee.com/winlinvip/srs-wiki/raw/master/images/srs-faq.svg)](https://github.com/ossrs/srs/issues/2716)
 
-SRS/4.0 [Leo](https://github.com/ossrs/srs/wiki/v4_CN_Product#release40) 是一个简单高效的实时视频服务器，支持RTMP/WebRTC/HLS/HTTP-FLV/SRT。
+SRS/5.0 是一个简单高效的实时视频服务器，支持RTMP/WebRTC/HLS/HTTP-FLV/SRT。
 
-SRS/4.0 [Leo](https://github.com/ossrs/srs/wiki/v4_CN_Product#release40) is a simple, high efficiency and realtime video server, supports RTMP/WebRTC/HLS/HTTP-FLV/SRT.
+SRS/5.0 is a simple, high efficiency and realtime video server, supports RTMP/WebRTC/HLS/HTTP-FLV/SRT.
 
 SRS is licenced under [MIT](https://github.com/ossrs/srs/blob/develop/LICENSE), but some depended libraries are distributed using their [own licenses](https://github.com/ossrs/srs/wiki/LicenseMixing).
 

trunk/doc/CHANGELOG.md

@@ -30,6 +30,8 @@ The changelog for SRS.
 
 ## SRS 4.0 Changelog
 
+* v4.0, 2021-11-13, Merge [#2729](https://github.com/ossrs/srs/pull/2729): RTC: check audio track exist when negotiate (#2729). v4.0.196
+* v4.0, 2021-11-09, Merge [#2721](https://github.com/ossrs/srs/pull/2721): Rtc2Rtmp: Use RTP timestamp to identify video frames. v4.0.195
 * v4.0, 2021-11-07, Merge [#2711](https://github.com/ossrs/srs/pull/2711): Config: Guess config files by [FHS](https://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard). v4.0.194
 * v4.0, 2021-11-07, Merge [#2714](https://github.com/ossrs/srs/pull/2714): DVR: copy req from publish. v4.0.193
 * v4.0, 2021-11-04, Merge [#2707](https://github.com/ossrs/srs/pull/2707): Refuse edge request when state is stopping. v4.0.192

trunk/src/app/srs_app_rtc_source.cpp

@@ -659,6 +659,9 @@ std::vector<SrsRtcTrackDescription*> SrsRtcSource::get_track_desc(std::string ty
     }
 
     if (type == "audio") {
+        if (! stream_desc_->audio_track_desc_) {
+            return track_descs;
+        }
         if (stream_desc_->audio_track_desc_->media_->name_ == media_name) {
             track_descs.push_back(stream_desc_->audio_track_desc_);
         }
@@ -1437,6 +1440,7 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video(SrsRtpPacket* src)
     cache_video_pkts_[index].pkt = pkt;
     cache_video_pkts_[index].sn = pkt->header.get_sequence();
     cache_video_pkts_[index].ts = pkt->get_avsync_time();
+    cache_video_pkts_[index].rtp_ts = pkt->header.get_timestamp();
 
     // check whether to recovery lost packet and can construct a video frame
     if (lost_sn_ == pkt->header.get_sequence()) {
@@ -1525,6 +1529,7 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video_key_frame(SrsRtpPacket* pkt)
     cache_video_pkts_[index].pkt = pkt;
     cache_video_pkts_[index].sn = pkt->header.get_sequence();
     cache_video_pkts_[index].ts = pkt->get_avsync_time();
+    cache_video_pkts_[index].rtp_ts = pkt->header.get_timestamp();
 
     int32_t sn = lost_sn_;
     uint16_t tail_sn = 0;
@@ -1622,6 +1627,7 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video_rtmp(const uint16_t start, const
         cache_video_pkts_[index].in_use = false;
         cache_video_pkts_[index].pkt = NULL;
         cache_video_pkts_[index].ts = 0;
+        cache_video_pkts_[index].rtp_ts = 0;
         cache_video_pkts_[index].sn = 0;
 
         SrsRtpFUAPayload2* fua_payload = dynamic_cast<SrsRtpFUAPayload2*>(pkt->payload());
@@ -1692,7 +1698,7 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video_rtmp(const uint16_t start, const
 
 int32_t SrsRtmpFromRtcBridger::find_next_lost_sn(uint16_t current_sn, uint16_t& end_sn)
 {
-    uint32_t last_ts = cache_video_pkts_[cache_index(header_sn_)].ts;
+    uint32_t last_rtp_ts = cache_video_pkts_[cache_index(header_sn_)].rtp_ts;
     for (int i = 0; i < s_cache_size; ++i) {
         uint16_t lost_sn = current_sn + i;
         int index = cache_index(lost_sn);
@@ -1701,7 +1707,7 @@ int32_t SrsRtmpFromRtcBridger::find_next_lost_sn(uint16_t current_sn, uint16_t&
             return lost_sn;
         }
         //check time first, avoid two small frame mixed case decode fail
-        if (last_ts != cache_video_pkts_[index].ts) {
+        if (last_rtp_ts != cache_video_pkts_[index].rtp_ts) {
             end_sn = lost_sn - 1;
             return -1;
         }
@@ -1724,6 +1730,7 @@ void SrsRtmpFromRtcBridger::clear_cached_video()
             srs_freep(cache_video_pkts_[i].pkt);
             cache_video_pkts_[i].sn = 0;
             cache_video_pkts_[i].ts = 0;
+            cache_video_pkts_[i].rtp_ts = 0;
             cache_video_pkts_[i].in_use = false;
         }
     }

trunk/src/app/srs_app_rtc_source.hpp

@@ -300,6 +300,7 @@ private:
         bool in_use;
         uint16_t sn;
         uint32_t ts;
+        uint32_t rtp_ts;
         SrsRtpPacket* pkt;
     };
     const static uint16_t s_cache_size = 512;

trunk/src/core/srs_core_version4.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       4
 #define VERSION_MINOR       0
-#define VERSION_REVISION    194
+#define VERSION_REVISION    196
 
 #endif