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[client] audio: reduce resampler latency
The best quality resampler has an intrinsic latency of about 3ms, and the processing itself takes another 1-2ms per 10ms block. The faster setting has an intrinsic latency of about 0.4ms, with about 0.04ms processing time. This makes for an overall saving of about 4ms, with negligible loss in quality.
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@ -328,8 +328,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
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playbackStop();
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int srcError;
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audio.playback.spiceData.src =
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src_new(SRC_SINC_BEST_QUALITY, channels, &srcError);
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audio.playback.spiceData.src = src_new(SRC_SINC_FASTEST, channels, &srcError);
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if (!audio.playback.spiceData.src)
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{
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DEBUG_ERROR("Failed to create resampler: %s", src_strerror(srcError));
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@ -614,7 +613,7 @@ void audio_playbackData(uint8_t * data, size_t size)
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// the resampler startup latency
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if (audio.playback.state == STREAM_STATE_KEEP_ALIVE)
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{
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int resamplerLatencyFrames = 144;
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int resamplerLatencyFrames = 20;
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targetPosition += resamplerLatencyFrames;
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}
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