On Mon, Feb 6, 2023 at 10:53 PM Volker Rümelin <vr_qemu@t-online.de> wrote:
>
> 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>
> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
> ---
> 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);
> --
> 2.35.3
>
--
Marc-André Lureau