| Message ID | 20230206185237.8358-4-vr_qemu@t-online.de (mailing list archive) |
|---|---|
| State | New, archived |
| Headers | show |
| Series | audio: improve callback interface for audio frontends | expand |
Hi 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_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> > Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> lgtm Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com> > --- > audio/audio.c | 63 +++++++++++++++++++++++++++++---------------------- > 1 file changed, 36 insertions(+), 27 deletions(-) > > diff --git a/audio/audio.c b/audio/audio.c > index a399147486..4412b5fad8 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; > -- > 2.35.3 > -- Marc-André Lureau
diff --git a/audio/audio.c b/audio/audio.c index a399147486..4412b5fad8 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;