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RTC: Fix FFmpeg opus audio noisy issue. v5.0.195 v6.0.95 (#3845)

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Follow the example in FFmpeg's doc, before calling the API
`avcodec_send_frame`, always use `av_frame_alloc` to create a new frame.

---------

Co-authored-by: Haibo Chen <495810242@qq.com>
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john 2023-10-25 11:55:32 +08:00 • committed by GitHub
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trunk/doc/CHANGELOG.md
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@ -7,6 +7,7 @@ The changelog for SRS.
<a name="v6-changes"></a>
## SRS 6.0 Changelog
* v6.0, 2023-10-25, Merge [#3845](https://github.com/ossrs/srs/pull/3845): RTC: Fix FFmpeg opus audio noisy issue. v6.0.95 (#3845)
* v6.0, 2023-10-21, Merge [#3847](https://github.com/ossrs/srs/pull/3847): WebRTC: TCP transport should use read_fully instead of read. v6.0.94 (#3847)
* v6.0, 2023-10-20, Merge [#3846](https://github.com/ossrs/srs/pull/3846): Added system library option for ffmpeg, srtp, srt libraries. v6.0.93 (#3846)
* v6.0, 2023-10-17, Merge [#3840](https://github.com/ossrs/srs/pull/3840): Disable asan by default. v6.0.92 (#3840)
@ -106,6 +107,7 @@ The changelog for SRS.
<a name="v5-changes"></a>
## SRS 5.0 Changelog
* v5.0, 2023-10-25, Merge [#3845](https://github.com/ossrs/srs/pull/3845): RTC: Fix FFmpeg opus audio noisy issue. v5.0.195 (#3845)
* v5.0, 2023-10-21, Merge [#3847](https://github.com/ossrs/srs/pull/3847): WebRTC: TCP transport should use read_fully instead of read. v5.0.194 (#3847)
* v5.0, 2023-10-20, Merge [#3846](https://github.com/ossrs/srs/pull/3846): Added system library option for ffmpeg, srtp, srt libraries. v5.0.193 (#3846)
* v5.0, 2023-10-17, Merge [#3840](https://github.com/ossrs/srs/pull/3840): Disable asan by default. v5.0.192 (#3840)