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[PULL 22/27] audio: replace the resampling loop in audio_pcm_sw_read()
From: |
marcandre . lureau |
Subject: |
[PULL 22/27] audio: replace the resampling loop in audio_pcm_sw_read() |
Date: |
Mon, 6 Mar 2023 10:51:57 +0400 |
From: Volker Rümelin <vr_qemu@t-online.de>
Replace the resampling loop in audio_pcm_sw_read() with the new
function audio_pcm_sw_resample_in(). 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: <20230224190555.7409-10-vr_qemu@t-online.de>
---
audio/audio.c | 59 ++++++++++++++++++++++++++++++---------------------
1 file changed, 35 insertions(+), 24 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index e18b5e98c5..9e9c03a42e 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -543,11 +543,43 @@ static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void
*pcm_buf, size_t samples)
/*
* Soft voice (capture)
*/
+static void audio_pcm_sw_resample_in(SWVoiceIn *sw,
+ size_t frames_in_max, size_t frames_out_max,
+ size_t *total_in, size_t *total_out)
+{
+ HWVoiceIn *hw = sw->hw;
+ struct st_sample *src, *dst;
+ size_t live, rpos, frames_in, frames_out;
+
+ live = hw->total_samples_captured - sw->total_hw_samples_acquired;
+ rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size);
+
+ /* resample conv_buf from rpos to end of buffer */
+ src = hw->conv_buf.buffer + rpos;
+ frames_in = MIN(frames_in_max, hw->conv_buf.size - rpos);
+ dst = sw->resample_buf.buffer;
+ frames_out = frames_out_max;
+ st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out);
+ rpos += frames_in;
+ *total_in = frames_in;
+ *total_out = frames_out;
+
+ /* resample conv_buf from start of buffer if there are input frames left */
+ if (frames_in_max - frames_in && rpos == hw->conv_buf.size) {
+ src = hw->conv_buf.buffer;
+ frames_in = frames_in_max - frames_in;
+ dst += frames_out;
+ frames_out = frames_out_max - frames_out;
+ st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out);
+ *total_in += frames_in;
+ *total_out += frames_out;
+ }
+}
+
static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
{
HWVoiceIn *hw = sw->hw;
- size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
- struct st_sample *src, *dst = sw->resample_buf.buffer;
+ size_t samples, live, ret, swlim, total;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
if (!live) {
@@ -558,33 +590,12 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf,
size_t size)
return 0;
}
- rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size);
-
samples = size / sw->info.bytes_per_frame;
swlim = (live * sw->ratio) >> 32;
swlim = MIN (swlim, samples);
- while (swlim) {
- src = hw->conv_buf.buffer + rpos;
- if (hw->conv_buf.pos > rpos) {
- isamp = hw->conv_buf.pos - rpos;
- } else {
- isamp = hw->conv_buf.size - rpos;
- }
-
- if (!isamp) {
- break;
- }
- osamp = swlim;
-
- st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
- swlim -= osamp;
- rpos = (rpos + isamp) % hw->conv_buf.size;
- dst += osamp;
- ret += osamp;
- total += isamp;
- }
+ audio_pcm_sw_resample_in(sw, live, swlim, &total, &ret);
if (!hw->pcm_ops->volume_in) {
mixeng_volume(sw->resample_buf.buffer, ret, &sw->vol);
--
2.39.2
- [PULL 11/27] alsaaudio: change default playback settings, (continued)
- [PULL 11/27] alsaaudio: change default playback settings, marcandre . lureau, 2023/03/06
- [PULL 12/27] alsaaudio: reintroduce default recording settings, marcandre . lureau, 2023/03/06
- [PULL 13/27] audio: change type of mix_buf and conv_buf, marcandre . lureau, 2023/03/06
- [PULL 14/27] audio: change type and name of the resample buffer, marcandre . lureau, 2023/03/06
- [PULL 15/27] audio: make the resampling code greedy, marcandre . lureau, 2023/03/06
- [PULL 16/27] audio: replace the resampling loop in audio_pcm_sw_write(), marcandre . lureau, 2023/03/06
- [PULL 17/27] audio: remove sw == NULL check, marcandre . lureau, 2023/03/06
- [PULL 18/27] audio: rename variables in audio_pcm_sw_write(), marcandre . lureau, 2023/03/06
- [PULL 19/27] audio: don't misuse audio_pcm_sw_write(), marcandre . lureau, 2023/03/06
- [PULL 20/27] audio: remove unused noop_conv() function, marcandre . lureau, 2023/03/06
- [PULL 22/27] audio: replace the resampling loop in audio_pcm_sw_read(),
marcandre . lureau <=
- [PULL 21/27] audio: make playback packet length calculation exact, marcandre . lureau, 2023/03/06
- [PULL 23/27] audio: rename variables in audio_pcm_sw_read(), marcandre . lureau, 2023/03/06
- [PULL 24/27] audio: make recording packet length calculation exact, marcandre . lureau, 2023/03/06
- [PULL 25/27] audio: handle leftover audio frame from upsampling, marcandre . lureau, 2023/03/06
- [PULL 26/27] audio/audio_template: substitute sw->hw with hw, marcandre . lureau, 2023/03/06
- [PULL 27/27] audio: remove sw->ratio, marcandre . lureau, 2023/03/06
- Re: [PULL 00/27] Audio patches, Peter Maydell, 2023/03/06