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RTC: Refine FFmpeg opus audio noisy issue. v5.0.197 v6.0.97 (#3852)

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### 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
parent 4a100616fc
commit e7b629cd39
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10 changed files with 27 additions and 24 deletions
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trunk/research/players/js/srs.page.js
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@ -13,7 +13,6 @@ function srs_get_player_height() { return srs_get_player_width() * 9 / 19; }
* update the navigator, add same query string.
*/
function update_nav() {
$("#srs_index").attr("href", "index.html" + window.location.search);
$("#nav_srs_player").attr("href", "srs_player.html" + window.location.search);
$("#nav_rtc_player").attr("href", "rtc_player.html" + window.location.search);
$("#nav_rtc_publisher").attr("href", "rtc_publisher.html" + window.location.search);