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.
Acked-by: Mark Cave-Ayland <mark.cave-ayl...@ilande.co.uk>
Reviewed-by: Marc-André Lureau <marcandre.lur...@redhat.com>
Signed-off-by: Volker Rümelin <vr_q...@t-online.de>
Message-Id: <20230224190555.7409-15-vr_q...@t-online.de>
---
audio/audio_int.h | 2 --
audio/audio_template.h | 30 +++++++++---------------------
audio/audio.c | 1 -
3 files changed, 9 insertions(+), 24 deletions(-)
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) {
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);
--
2.39.2
