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fix err memory leak in rtc to rtmp bridge. v6.0.174 v7.0.55 (#4441)

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1. print the error messages before dismiss it;
2. free the err to avoid memory leak;

## Cause
found this issue when research #4434 .


## Impact

 1. develop
 2. 5.0release
 3. 6.0release

---------

Co-authored-by: winlin <winlinvip@gmail.com>
This commit is contained in:
Jacob Su 2025-08-16 21:56:49 +08:00 committed by GitHub
parent f40ca0ff9f
commit 8cd4616147
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GPG Key ID: B5690EEEBB952194
4 changed files with 6 additions and 3 deletions
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2
trunk/doc/CHANGELOG.md
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@ -7,6 +7,7 @@ The changelog for SRS.
<a name="v7-changes"></a> <a name="v7-changes"></a>
## SRS 7.0 Changelog ## SRS 7.0 Changelog
* v7.0, 2025-08-16, Merge [#4441](https://github.com/ossrs/srs/pull/4441): fix err memory leak in rtc to rtmp bridge. v7.0.55 (#4441)
* v7.0, 2025-08-14, Merge [#4161](https://github.com/ossrs/srs/pull/4161): fix hls & dash segments cleanup. v7.0.54 (#4161) * v7.0, 2025-08-14, Merge [#4161](https://github.com/ossrs/srs/pull/4161): fix hls & dash segments cleanup. v7.0.54 (#4161)
* v7.0, 2025-08-13, Merge [#4225](https://github.com/ossrs/srs/pull/4225): issue #4223: remove hls_acodec and hls_vcodec config. v7.0.53 (#4225) * v7.0, 2025-08-13, Merge [#4225](https://github.com/ossrs/srs/pull/4225): issue #4223: remove hls_acodec and hls_vcodec config. v7.0.53 (#4225)
* v7.0, 2025-08-12, Merge [#4230](https://github.com/ossrs/srs/pull/4230): MP4 DVR: Fix audio/video synchronization issues in WebRTC recordings. v7.0.52 (#4230) * v7.0, 2025-08-12, Merge [#4230](https://github.com/ossrs/srs/pull/4230): MP4 DVR: Fix audio/video synchronization issues in WebRTC recordings. v7.0.52 (#4230)
@ -68,6 +69,7 @@ The changelog for SRS.
<a name="v6-changes"></a> <a name="v6-changes"></a>
## SRS 6.0 Changelog ## SRS 6.0 Changelog
* v6.0, 2025-08-16, Merge [#4441](https://github.com/ossrs/srs/pull/4441): fix err memory leak in rtc to rtmp bridge. v6.0.174 (#4441)
* v6.0, 2025-08-14, Merge [#4161](https://github.com/ossrs/srs/pull/4161): fix hls & dash segments cleanup. v6.0.173 (#4161) * v6.0, 2025-08-14, Merge [#4161](https://github.com/ossrs/srs/pull/4161): fix hls & dash segments cleanup. v6.0.173 (#4161)
* v6.0, 2025-08-12, Merge [#4230](https://github.com/ossrs/srs/pull/4230): MP4 DVR: Fix audio/video synchronization issues in WebRTC recordings. v6.0.172 (#4230) * v6.0, 2025-08-12, Merge [#4230](https://github.com/ossrs/srs/pull/4230): MP4 DVR: Fix audio/video synchronization issues in WebRTC recordings. v6.0.172 (#4230)
* v6.0, 2025-08-11, Merge [#4432](https://github.com/ossrs/srs/pull/4432): AI: HTTP-FLV: Fix heap-use-after-free crash during stream unmount. v6.0.171 (#4432) * v6.0, 2025-08-11, Merge [#4432](https://github.com/ossrs/srs/pull/4432): AI: HTTP-FLV: Fix heap-use-after-free crash during stream unmount. v6.0.171 (#4432)

3
trunk/src/app/srs_app_rtc_source.cpp
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@ -2413,7 +2413,8 @@ srs_error_t SrsRtcFrameBuilder::packet_video_rtmp(const uint16_t start, const ui
} }
if ((err = bridge_->on_frame(&msg)) != srs_success) { if ((err = bridge_->on_frame(&msg)) != srs_success) {
srs_warn("fail to pack video frame"); srs_warn("fail to pack video frame: %s", srs_error_summary(err).c_str());
srs_freep(err);
} }
// Try to detect and detach next RTMP packet. // Try to detect and detach next RTMP packet.

2
trunk/src/core/srs_core_version6.hpp
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@ -9,6 +9,6 @@
#define VERSION_MAJOR 6 #define VERSION_MAJOR 6
#define VERSION_MINOR 0 #define VERSION_MINOR 0
#define VERSION_REVISION 173 #define VERSION_REVISION 174
#endif #endif

2
trunk/src/core/srs_core_version7.hpp
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@ -9,6 +9,6 @@
#define VERSION_MAJOR 7 #define VERSION_MAJOR 7
#define VERSION_MINOR 0 #define VERSION_MINOR 0
#define VERSION_REVISION 54 #define VERSION_REVISION 55
#endif #endif