diff --git a/README.md b/README.md index 0efa3ce9d..a244ba966 100755 --- a/README.md +++ b/README.md @@ -13,8 +13,9 @@ [![](https://badgen.net/twitter/follow/winlinvip)](https://twitter.com/winlinvip) [![](https://img.shields.io/youtube/channel/views/UCP6ZblCL_fIJoEyUzZxC1ng?style=social)](https://www.youtube.com/channel/UCP6ZblCL_fIJoEyUzZxC1ng) [![](https://img.shields.io/twitch/status/winlinvip?style=social)](https://www.twitch.tv/winlinvip) +[![](https://badgen.net/discord/members/yZ4BnPmHAd)](https://discord.gg/yZ4BnPmHAd) -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/5.0，[Bee](https://github.com/ossrs/srs/wiki/Product#release50) 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 is a simple, high efficiency and realtime video server, supports RTMP/WebRTC/HLS/HTTP-FLV/SRT. @@ -82,6 +83,8 @@ Other important wiki: ## AUTHORS +![](https://opencollective.com/srs-server/contributors.svg?width=800&button=false) + The [TOC(Technical Oversight Committee)](trunk/AUTHORS.md#toc) and [contributors](trunk/AUTHORS.md#contributors): * [Winlin](https://github.com/winlinvip): Focus on [ST](https://github.com/ossrs/state-threads) and [Issues/PR](https://github.com/ossrs/srs/issues). @@ -371,4 +374,3 @@ Winlin [r0.3]: https://github.com/ossrs/srs/releases/tag/v0.3.0 [r0.2]: https://github.com/ossrs/srs/releases/tag/v0.2.0 [r0.1]: https://github.com/ossrs/srs/releases/tag/v0.1.0 - diff --git a/trunk/conf/rtc.conf b/trunk/conf/rtc.conf index b7269cbdd..2a5153220 100644 --- a/trunk/conf/rtc.conf +++ b/trunk/conf/rtc.conf @@ -21,7 +21,7 @@ stats { } rtc_server { enabled on; - listen 8000; + listen 8000; # UDP port # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate candidate $CANDIDATE; } diff --git a/trunk/conf/rtc2rtmp.conf b/trunk/conf/rtc2rtmp.conf index 66f3c8cfa..87a71345e 100644 --- a/trunk/conf/rtc2rtmp.conf +++ b/trunk/conf/rtc2rtmp.conf @@ -19,7 +19,7 @@ stats { } rtc_server { enabled on; - listen 8000; + listen 8000; # UDP port # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate candidate $CANDIDATE; } diff --git a/trunk/conf/rtmp2rtc.conf b/trunk/conf/rtmp2rtc.conf index 66f3c8cfa..87a71345e 100644 --- a/trunk/conf/rtmp2rtc.conf +++ b/trunk/conf/rtmp2rtc.conf @@ -19,7 +19,7 @@ stats { } rtc_server { enabled on; - listen 8000; + listen 8000; # UDP port # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate candidate $CANDIDATE; } diff --git a/trunk/conf/srs.conf b/trunk/conf/srs.conf index d7af0b9f4..9d7d9047d 100644 --- a/trunk/conf/srs.conf +++ b/trunk/conf/srs.conf @@ -17,7 +17,7 @@ http_server { } rtc_server { enabled on; - listen 8000; + listen 8000; # UDP port # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate candidate $CANDIDATE; } diff --git a/trunk/doc/CHANGELOG.md b/trunk/doc/CHANGELOG.md index 63c3d36ca..37b99004a 100644 --- a/trunk/doc/CHANGELOG.md +++ b/trunk/doc/CHANGELOG.md @@ -30,10 +30,12 @@ The changelog for SRS. ## SRS 4.0 Changelog -* v4.0, 2021-12-04, Merge [#2768](https://github.com/ossrs/srs/pull/2768): RTC: Fix bugs for RTC2RTMP. (#2768). v4.0.203 -* v4.0, 2021-12-04, Merge [#2757](https://github.com/ossrs/srs/pull/2757): RTC: Ignore empty audio packet when transcoding (#2757). v4.0.202 -* v4.0, 2021-12-01, Fix [#2762](https://github.com/ossrs/srs/pull/2762): RTC: Refine publish security error message (#2762). v4.0.200 -* v4.0, 2021-11-25, Merge [#2751](https://github.com/ossrs/srs/pull/2751): RTC: Fix crash when pkt->payload() if pkt is nullptr (#2751). v4.0.199 +* v4.0, 2021-12-07, Merge [#2771](https://github.com/ossrs/srs/pull/2771): RTC: Fix memory leak when replace rtp packet in cache. (#2771). v4.0.205 +* v4.0, 2021-12-06, Merge [#2766](https://github.com/ossrs/srs/pull/2766): RTC: Fix nack encode seqnum. (#2766). v4.0.204 +* v4.0, 2021-12-04, Merge [#2768](https://github.com/ossrs/srs/pull/2768): RTC: Fix bugs for RTC2RTMP. (#2768). v4.0.203 +* v4.0, 2021-12-04, Merge [#2757](https://github.com/ossrs/srs/pull/2757): RTC: Ignore empty audio packet when transcoding (#2757). v4.0.202 +* v4.0, 2021-12-01, Fix [#2762](https://github.com/ossrs/srs/pull/2762): RTC: Refine publish security error message (#2762). v4.0.200 +* v4.0, 2021-11-25, Merge [#2751](https://github.com/ossrs/srs/pull/2751): RTC: Fix crash when pkt->payload() if pkt is nullptr (#2751). v4.0.199 * v4.0, 2021-11-15, For [#1708](https://github.com/ossrs/srs/pull/1708): ST: Print log when multiple thread stop one coroutine. (#1708). v4.0.198 * v4.0, 2021-11-14, Merge [#2732](https://github.com/ossrs/srs/pull/2732): WebRTC: Fail to publish RTC automatically for HTML5. (#2732). v4.0.197 * 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 diff --git a/trunk/src/app/srs_app_rtc_source.cpp b/trunk/src/app/srs_app_rtc_source.cpp index e13ae93a1..3798ebbf4 100644 --- a/trunk/src/app/srs_app_rtc_source.cpp +++ b/trunk/src/app/srs_app_rtc_source.cpp @@ -1437,6 +1437,7 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video(SrsRtpPacket* src) // store in cache int index = cache_index(pkt->header.get_sequence()); cache_video_pkts_[index].in_use = true; + srs_freep(cache_video_pkts_[index].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(); @@ -1526,6 +1527,7 @@ srs_error_t SrsRtmpFromRtcBridger::packet_video_key_frame(SrsRtpPacket* pkt) uint16_t index = cache_index(pkt->header.get_sequence()); cache_video_pkts_[index].in_use = true; + srs_freep(cache_video_pkts_[index].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(); diff --git a/trunk/src/core/srs_core_version4.hpp b/trunk/src/core/srs_core_version4.hpp index 0af98b62c..868584091 100644 --- a/trunk/src/core/srs_core_version4.hpp +++ b/trunk/src/core/srs_core_version4.hpp @@ -9,6 +9,6 @@ #define VERSION_MAJOR 4 #define VERSION_MINOR 0 -#define VERSION_REVISION 203 +#define VERSION_REVISION 205 #endif diff --git a/trunk/src/kernel/srs_kernel_rtc_rtcp.cpp b/trunk/src/kernel/srs_kernel_rtc_rtcp.cpp index 7f3120423..dd4fdcfaf 100644 --- a/trunk/src/kernel/srs_kernel_rtc_rtcp.cpp +++ b/trunk/src/kernel/srs_kernel_rtc_rtcp.cpp @@ -1348,7 +1348,10 @@ srs_error_t SrsRtcpNack::encode(SrsBuffer *buffer) } else if( (sn - pid) > 16) { // add new chunk chunks.push_back(chunk); - chunk.in_use = false; + chunk.pid = sn; + chunk.blp = 0; + chunk.in_use = true; + pid = sn; } else { chunk.blp |= 1 << (sn-pid-1); } diff --git a/trunk/src/utest/srs_utest_rtc.cpp b/trunk/src/utest/srs_utest_rtc.cpp index 83b8681bb..09f11c327 100644 --- a/trunk/src/utest/srs_utest_rtc.cpp +++ b/trunk/src/utest/srs_utest_rtc.cpp @@ -744,6 +744,32 @@ VOID TEST(KernelRTCTest, NACKFetchRTPPacket) } } +VOID TEST(KernelRTCTest, NACKEncode) +{ + uint32_t ssrc = 123; + char buf_before[kRtcpPacketSize]; + SrsBuffer stream_before(buf_before, sizeof(buf_before)); + + SrsRtcpNack rtcp_nack_encode(ssrc); + for(uint16_t i = 16; i < 50; ++i) { + rtcp_nack_encode.add_lost_sn(i); + } + srs_error_t err_before = rtcp_nack_encode.encode(&stream_before); + EXPECT_TRUE(err_before == 0); + char buf_after[kRtcpPacketSize]; + memcpy(buf_after, buf_before, kRtcpPacketSize); + SrsBuffer stream_after(buf_after, sizeof(buf_after)); + SrsRtcpNack rtcp_nack_decode(ssrc); + srs_error_t err_after = rtcp_nack_decode.decode(&stream_after); + EXPECT_TRUE(err_after == 0); + vector before = rtcp_nack_encode.get_lost_sns(); + vector after = rtcp_nack_decode.get_lost_sns(); + EXPECT_TRUE(before.size() == after.size()); + for(int i = 0; i < before.size() && i < after.size(); ++i) { + EXPECT_TRUE(before.at(i) == after.at(i)); + } +} + extern bool srs_is_stun(const uint8_t* data, size_t size); extern bool srs_is_dtls(const uint8_t* data, size_t len); extern bool srs_is_rtp_or_rtcp(const uint8_t* data, size_t len);