7aba442a38
## Summary Removes the deprecated `hls_acodec` and `hls_vcodec` configuration options and implements automatic codec detection for HLS streams, fixing issues with video-only streams incorrectly showing audio information. ## Problem - When streaming video-only content via RTMP, HLS output incorrectly contained audio track information due to hardcoded default codec settings - The static `hls_acodec` and `hls_vcodec` configurations were inflexible and caused compatibility issues with some players - Users had to manually configure `hls_acodec an` to fix video-only streams ## Solution - **Remove deprecated configs**: Eliminates `hls_acodec` and `hls_vcodec` configuration options entirely - **Dynamic codec detection**: HLS muxer now automatically detects and uses actual stream codecs in real-time - **Improved defaults**: Changes from hardcoded AAC/H.264 defaults to disabled state, letting actual stream content determine codec information - **Real-time codec switching**: Supports codec changes during streaming with proper logging ## Changes - Remove `get_hls_acodec()` and `get_hls_vcodec()` from SrsConfig - Update HLS muxer to use `latest_acodec_`/`latest_vcodec_` for codec detection - Add codec detection logic in `write_audio()` and `write_video()` methods - Remove deprecated config options from all configuration files - Add comprehensive unit tests for codec detection functionality Fixes #4223 --------- Co-authored-by: Haibo Chen <495810242@qq.com> Co-authored-by: OSSRS-AI <winlinam@gmail.com>
32 lines
686 B
Plaintext
32 lines
686 B
Plaintext
listen 1935;
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max_connections 1000;
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daemon off;
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srs_log_tank console;
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http_server {
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enabled on;
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listen 8080;
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dir ./objs/nginx/html;
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}
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rtc_server {
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enabled on;
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listen 8000; # UDP port
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# @see https://ossrs.io/lts/en-us/docs/v7/doc/webrtc#config-candidate
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candidate $CANDIDATE;
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}
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vhost __defaultVhost__ {
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http_remux {
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enabled on;
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mount [vhost]/[app]/[stream].flv;
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}
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hls {
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enabled on;
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}
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rtc {
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enabled on;
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# @see https://ossrs.io/lts/en-us/docs/v7/doc/webrtc#rtmp-to-rtc
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rtmp_to_rtc on;
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# @see https://ossrs.io/lts/en-us/docs/v7/doc/webrtc#rtc-to-rtmp
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rtc_to_rtmp on;
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}
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}
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