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winlin 2021-11-13 19:31:42 +08:00
commit ae13e784b4
5 changed files with 17 additions and 8 deletions
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8
README.md
View file

@ -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/bbs2.png)](http://bbs.ossrs.net)
[![](https://gitee.com/winlinvip/srs-wiki/raw/master/images/wechat-badge4.svg)](../../wikis/Contact#wechat)
[![](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).

1
trunk/doc/CHANGELOG.md
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@ -30,6 +30,7 @@ The changelog for SRS.
## SRS 4.0 Changelog
* 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

11
trunk/src/app/srs_app_rtc_source.cpp
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@ -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;
}
}

3
trunk/src/app/srs_app_rtc_source.hpp
View file

@ -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;
@ -325,7 +326,7 @@ private:
int32_t find_next_lost_sn(uint16_t current_sn, uint16_t& end_sn);
void clear_cached_video();
inline uint16_t cache_index(uint16_t current_sn) {
return current_sn%s_cache_size;
return current_sn % s_cache_size;
}
bool check_frame_complete(const uint16_t start, const uint16_t end);
};

2
trunk/src/core/srs_core_version4.hpp
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@ -9,6 +9,6 @@
#define VERSION_MAJOR 4
#define VERSION_MINOR 0
#define VERSION_REVISION 194
#define VERSION_REVISION 195
#endif