Re: [PATCH v2 11/17] audio: replace the resampling loop in audio_pcm_sw_read()

[Marc-André Lureau]

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