1
0
Fork 0
mirror of https://github.com/ossrs/srs.git synced 2025-03-09 15:49:59 +00:00

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

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>
This commit is contained in:
john 2023-10-25 11:55:32 +08:00 committed by GitHub
parent 9b07d840ed
commit 9238f09b0b
No known key found for this signature in database
GPG key ID: 4AEE18F83AFDEB23
10 changed files with 119 additions and 23 deletions

View file

@ -0,0 +1,46 @@
# WebRTC streaming config for SRS.
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
}
http_api {
enabled on;
listen 1985;
}
stats {
network 0;
}
rtc_server {
enabled on;
tcp {
enabled on;
listen 8000;
}
protocol tcp;
# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
rtc {
enabled on;
# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
rtmp_to_rtc off;
# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
rtc_to_rtmp off;
}
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
}
}