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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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chundonglinlin 2023-11-04 16:21:44 +08:00 committed by GitHub
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10 changed files with 27 additions and 24 deletions

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@ -7,6 +7,7 @@ The changelog for SRS.
<a name="v6-changes"></a>
## SRS 6.0 Changelog
* 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)
* 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)
* 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)
@ -108,6 +109,7 @@ The changelog for SRS.
<a name="v5-changes"></a>
## SRS 5.0 Changelog
* 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)
* 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)
* 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)