| Message ID | 20230206185237.8358-11-vr_qemu@t-online.de (mailing list archive) |
|---|---|
| State | New, archived |
| Headers | show |
| Series | audio: improve callback interface for audio frontends | expand |
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
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);