On Mon, Feb 6, 2023 at 10:53 PM Volker Rümelin <[email protected]> wrote: > > Simplify the resample buffer size calculation. > > For audio playback we have > sw->ratio = ((int64_t)sw->hw->info.freq << 32) / sw->info.freq; > samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio; > > This can be simplified to > samples = muldiv64(sw->HWBUF.size, sw->info.freq, sw->hw->info.freq); > > For audio recording we have > sw->ratio = ((int64_t)sw->info.freq << 32) / sw->hw->info.freq; > samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32; > > This can be simplified to > samples = muldiv64(sw->HWBUF.size, sw->info.freq, sw->hw->info.freq); > > With hw = sw->hw this becomes in both cases > samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq); > > Now that sw->ratio is no longer needed, remove sw->ratio. > > Acked-by: Mark Cave-Ayland <[email protected]> > Signed-off-by: Volker Rümelin <[email protected]>
Reviewed-by: Marc-André Lureau <[email protected]> > --- > audio/audio.c | 1 - > audio/audio_int.h | 2 -- > audio/audio_template.h | 30 +++++++++--------------------- > 3 files changed, 9 insertions(+), 24 deletions(-) > > diff --git a/audio/audio.c b/audio/audio.c > index 4836ab8ca8..70b096713c 100644 > --- a/audio/audio.c > +++ b/audio/audio.c > @@ -478,7 +478,6 @@ static int audio_attach_capture (HWVoiceOut *hw) > sw->info = hw->info; > sw->empty = 1; > sw->active = hw->enabled; > - sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq; > sw->vol = nominal_volume; > sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq); > QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries); > diff --git a/audio/audio_int.h b/audio/audio_int.h > index 8b163e1759..d51d63f08d 100644 > --- a/audio/audio_int.h > +++ b/audio/audio_int.h > @@ -108,7 +108,6 @@ struct SWVoiceOut { > AudioState *s; > struct audio_pcm_info info; > t_sample *conv; > - int64_t ratio; > STSampleBuffer resample_buf; > void *rate; > size_t total_hw_samples_mixed; > @@ -126,7 +125,6 @@ struct SWVoiceIn { > AudioState *s; > int active; > struct audio_pcm_info info; > - int64_t ratio; > void *rate; > size_t total_hw_samples_acquired; > STSampleBuffer resample_buf; > diff --git a/audio/audio_template.h b/audio/audio_template.h > index 7e116426c7..e42326c20d 100644 > --- a/audio/audio_template.h > +++ b/audio/audio_template.h > @@ -108,32 +108,23 @@ static void glue (audio_pcm_sw_free_resources_, TYPE) > (SW *sw) > static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) > { > HW *hw = sw->hw; > - int samples; > + uint64_t samples; > > if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) { > return 0; > } > > -#ifdef DAC > - samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio; > -#else > - samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32; > -#endif > - if (audio_bug(__func__, samples < 0)) { > - dolog("Can not allocate buffer for `%s' (%d samples)\n", > - SW_NAME(sw), samples); > - return -1; > - } > - > + samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq); > if (samples == 0) { > - size_t f_fe_min; > + uint64_t f_fe_min; > + uint64_t f_be = (uint32_t)hw->info.freq; > > /* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */ > - f_fe_min = (hw->info.freq + HWBUF.size - 1) / HWBUF.size; > + f_fe_min = (f_be + HWBUF.size - 1) / HWBUF.size; > qemu_log_mask(LOG_UNIMP, > AUDIO_CAP ": The guest selected a " NAME " sample rate" > - " of %d Hz for %s. Only sample rates >= %zu Hz are" > - " supported.\n", > + " of %d Hz for %s. Only sample rates >= %" PRIu64 " Hz" > + " are supported.\n", > sw->info.freq, sw->name, f_fe_min); > return -1; > } > @@ -141,9 +132,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW > *sw) > /* > * Allocate one additional audio frame that is needed for upsampling > * if the resample buffer size is small. For large buffer sizes take > - * care of overflows. > + * care of overflows and truncation. > */ > - samples = samples < INT_MAX ? samples + 1 : INT_MAX; > + samples = samples < SIZE_MAX ? samples + 1 : SIZE_MAX; > sw->resample_buf.buffer = g_new0(st_sample, samples); > sw->resample_buf.size = samples; > sw->resample_buf.pos = 0; > @@ -170,11 +161,8 @@ static int glue (audio_pcm_sw_init_, TYPE) ( > sw->hw = hw; > sw->active = 0; > #ifdef DAC > - sw->ratio = ((int64_t) sw->hw->info.freq << 32) / sw->info.freq; > sw->total_hw_samples_mixed = 0; > sw->empty = 1; > -#else > - sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq; > #endif > > if (sw->info.is_float) { > -- > 2.35.3 > -- Marc-André Lureau
