From: Volker Rümelin <vr_q...@t-online.de> 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. Signed-off-by: Volker Rümelin <vr_q...@t-online.de> --- audio/audio.c | 1 - audio/audio_int.h | 2 -- audio/audio_template.h | 9 +-------- 3 files changed, 1 insertion(+), 11 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index b846b89a27..b68ed4eb68 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -476,7 +476,6 @@ 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 f4ec5dcf11..3cd3539bd4 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 0cdf57760e..c053792da3 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -114,11 +114,7 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) return 0; } -#ifdef DAC - samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio; -#else - samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32; -#endif + samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq); if (samples == 0) { size_t f_fe_min; @@ -159,11 +155,8 @@ 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