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audio: replace the resampling loop in audio_pcm_sw_write()

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Replace the resampling loop in audio_pcm_sw_write() with the new
function audio_pcm_sw_resample_out(). Unlike the old resample
loop the new function will try to consume input frames even if
the output buffer is full. This is necessary when downsampling
to avoid reading less audio frames than calculated in advance.
The loop was unrolled to avoid complicated loop control conditions
in this case.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-4-vr_qemu@t-online.de>
This commit is contained in:
Volker Rümelin 2023-02-24 20:05:44 +01:00 committed by Marc-André Lureau
parent 8933882da9
commit 8a81abeeb2
1 changed files with 36 additions and 27 deletions
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63
audio/audio.c
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@ -673,11 +673,44 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
/*
* Soft voice (playback)
*/
static void audio_pcm_sw_resample_out(SWVoiceOut *sw,
size_t frames_in_max, size_t frames_out_max,
size_t *total_in, size_t *total_out)
{
HWVoiceOut *hw = sw->hw;
struct st_sample *src, *dst;
size_t live, wpos, frames_in, frames_out;
live = sw->total_hw_samples_mixed;
wpos = (hw->mix_buf.pos + live) % hw->mix_buf.size;
/* write to mix_buf from wpos to end of buffer */
src = sw->resample_buf.buffer;
frames_in = frames_in_max;
dst = hw->mix_buf.buffer + wpos;
frames_out = MIN(frames_out_max, hw->mix_buf.size - wpos);
st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out);
wpos += frames_out;
*total_in = frames_in;
*total_out = frames_out;
/* write to mix_buf from start of buffer if there are input frames left */
if (frames_in_max - frames_in > 0 && wpos == hw->mix_buf.size) {
src += frames_in;
frames_in = frames_in_max - frames_in;
dst = hw->mix_buf.buffer;
frames_out = frames_out_max - frames_out;
st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out);
*total_in += frames_in;
*total_out += frames_out;
}
}
static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
{
size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, blck;
size_t hwsamples, samples, live, dead;
size_t hw_free;
size_t ret = 0, pos = 0, total = 0;
size_t ret, total;
if (!sw) {
return size;
@ -698,8 +731,6 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
return 0;
}
wpos = (sw->hw->mix_buf.pos + live) % hwsamples;
dead = hwsamples - live;
hw_free = audio_pcm_hw_get_free(sw->hw);
hw_free = hw_free > live ? hw_free - live : 0;
@ -713,29 +744,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
}
}
while (samples) {
dead = hwsamples - live;
left = hwsamples - wpos;
blck = MIN (dead, left);
if (!blck) {
break;
}
isamp = samples;
osamp = blck;
st_rate_flow_mix (
sw->rate,
sw->resample_buf.buffer + pos,
sw->hw->mix_buf.buffer + wpos,
&isamp,
&osamp
);
ret += isamp;
samples -= isamp;
pos += isamp;
live += osamp;
wpos = (wpos + osamp) % hwsamples;
total += osamp;
}
audio_pcm_sw_resample_out(sw, samples, MIN(dead, hw_free), &ret, &total);
sw->total_hw_samples_mixed += total;
sw->empty = sw->total_hw_samples_mixed == 0;