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RTC: Refine FFmpeg opus audio noisy issue. v5.0.197 v6.0.97 (#3852)
### Description When converting between AAC and Opus formats (aac2opus or opus2aac), the `av_frame_get_buffer` API is frequently called. ### Objective The goal is to optimize the code logic and reduce the frequent allocation and deallocation of memory. In the case of aac2opus, av_frame_get_buffer is still frequently called. In the case of opus2aac, the goal is to avoid calling av_frame_get_buffer and reduce memory allocations. ### Additional Note Before calling the `av_audio_fifo_read` API, use `av_frame_make_writable` to check if the frame is writable. If it is not writable, create a new frame. --------- Co-authored-by: john <hondaxiao@tencent.com>
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@ -7,6 +7,7 @@ The changelog for SRS.
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<a name="v6-changes"></a>
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## SRS 6.0 Changelog
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* v6.0, 2023-11-04, Merge [#3852](https://github.com/ossrs/srs/pull/3852): RTC: Refine FFmpeg opus audio noisy issue. v6.0.97 (#3852)
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* v6.0, 2023-11-01, Merge [#3858](https://github.com/ossrs/srs/pull/3858): Support build without cache to test if actions fail. v6.0.96 (#3858)
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* 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)
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* 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)
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@ -108,6 +109,7 @@ The changelog for SRS.
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<a name="v5-changes"></a>
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## SRS 5.0 Changelog
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* v5.0, 2023-11-04, Merge [#3852](https://github.com/ossrs/srs/pull/3852): RTC: Refine FFmpeg opus audio noisy issue. v5.0.197 (#3852)
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* v5.0, 2023-11-01, Merge [#3858](https://github.com/ossrs/srs/pull/3858): Support build without cache to test if actions fail. v5.0.196 (#3858)
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* 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)
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* 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)
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