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authorVolker Rümelin <vr_qemu@t-online.de>2023-02-24 20:05:44 +0100
committerMarc-André Lureau <marcandre.lureau@redhat.com>2023-03-06 10:30:23 +0400
commit8a81abeeb2e17630c366fccbd99ce20eac328819 (patch)
tree74a721edbdd5368b740f1cc193b330a2c981b3e8
parent8933882da9c63a0b60706828e11f3c09e2a903c7 (diff)
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audio: replace the resampling loop in audio_pcm_sw_write()
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>
-rw-r--r--audio/audio.c63
1 files changed, 36 insertions, 27 deletions
diff --git a/audio/audio.c b/audio/audio.c
index a399147..4412b5f 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -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;