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[client] audio: reduce resampler latency

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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.
This commit is contained in:
Chris Spencer
2022-02-26 18:34:10 +00:00
committed by Geoffrey McRae
parent 7efc274e81
commit c2523be4b4
1 changed files with 2 additions and 3 deletions
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5
client/src/audio.c
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@@ -328,8 +328,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
playbackStop();
int srcError;
audio.playback.spiceData.src =
src_new(SRC_SINC_BEST_QUALITY, channels, &srcError);
audio.playback.spiceData.src = src_new(SRC_SINC_FASTEST, channels, &srcError);
if (!audio.playback.spiceData.src)
{
DEBUG_ERROR("Failed to create resampler: %s", src_strerror(srcError));
@@ -614,7 +613,7 @@ void audio_playbackData(uint8_t * data, size_t size)
// the resampler startup latency
if (audio.playback.state == STREAM_STATE_KEEP_ALIVE)
{
int resamplerLatencyFrames = 144;
int resamplerLatencyFrames = 20;
targetPosition += resamplerLatencyFrames;
}