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Squash: Fix bugs

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pull/2895/head
winlin 3 years ago
parent 8576fa7052
commit f05e67e1a6
10 changed files with 47 additions and 12 deletions
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6
README.md
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@ -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

2
trunk/conf/rtc.conf
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@ -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;
}

2
trunk/conf/rtc2rtmp.conf
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@ -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;
}

2
trunk/conf/rtmp2rtc.conf
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@ -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;
}

2
trunk/conf/srs.conf
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@ -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;
}

10
trunk/doc/CHANGELOG.md
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@ -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

2
trunk/src/app/srs_app_rtc_source.cpp
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@ -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();

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 203
#define VERSION_REVISION 205
#endif

5
trunk/src/kernel/srs_kernel_rtc_rtcp.cpp
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@ -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);
}

26
trunk/src/utest/srs_utest_rtc.cpp
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@ -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);

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