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Libopus update broke WebRTC #1982

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Basilisk-Devが3ヶ月前に作成 · 3件のコメント
Basilisk-Dev が 3ヶ月前 にコメント

The libopus update performed in #1890

broke WebRTC.

It appears that the function tonality_analysis_init in the file media/libopus/src/analysis.h added a new argument and calls to this function were not updated in the WebRTC code.

The file platform/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_classifier.cc calls this function without the new argument and so compilation fails.

The libopus commit adding this new function argument can be found here

The libopus update performed in #1890 broke WebRTC. It appears that the function `tonality_analysis_init` in the file `media/libopus/src/analysis.h` added a new argument and calls to this function were not updated in the WebRTC code. The file `platform/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_classifier.cc` calls this function without the new argument and so compilation fails. The libopus commit adding this new function argument can be found [here](https://gitlab.xiph.org/xiph/opus/-/commit/cf9409fe51fa425d176e74c2e3bc4b7a9b5e9086)
Moonchild が 3ヶ月前 にコメント
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Looks like the Fs parameter added is the sample rate. We should be able to just pass that down. Default is defined statically as kDefaultSampleRateHz

Looks like the `Fs` parameter added is the sample rate. We should be able to just pass that down. Default is defined statically as `kDefaultSampleRateHz`
Moonchild が 3ヶ月前 にコメント
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I merged the PR, let me know if this is resolved.

I merged the PR, let me know if this is resolved.
Basilisk-Dev が 3ヶ月前 にコメント
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@Moonchild This is resolved. I am now able to compile Basilisk with --enable-webrtc and was able to join a Jitsi Meet call successfully.

@Moonchild This is resolved. I am now able to compile Basilisk with `--enable-webrtc` and was able to join a Jitsi Meet call successfully.
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リファレンス: MoonchildProductions/UXP#1982
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