Squash: Fix bugs

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
-

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;
 }

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;
 }

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;
 }

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;
 }

trunk/doc/CHANGELOG.md

@@ -30,6 +30,8 @@ The changelog for SRS.
 
 ## SRS 4.0 Changelog
 
+* 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

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();

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

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);
             }

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<uint16_t> before = rtcp_nack_encode.get_lost_sns();
+    vector<uint16_t> 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);