audio: remove sw->ratio
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-ayland@ilande.co.uk> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-15-vr_qemu@t-online.de>
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@ -478,7 +478,6 @@ static int audio_attach_capture (HWVoiceOut *hw)
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sw->info = hw->info;
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sw->info = hw->info;
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sw->empty = 1;
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sw->empty = 1;
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sw->active = hw->enabled;
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sw->active = hw->enabled;
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sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
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sw->vol = nominal_volume;
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sw->vol = nominal_volume;
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sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
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sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
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QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
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QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
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@ -108,7 +108,6 @@ struct SWVoiceOut {
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AudioState *s;
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AudioState *s;
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struct audio_pcm_info info;
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struct audio_pcm_info info;
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t_sample *conv;
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t_sample *conv;
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int64_t ratio;
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STSampleBuffer resample_buf;
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STSampleBuffer resample_buf;
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void *rate;
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void *rate;
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size_t total_hw_samples_mixed;
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size_t total_hw_samples_mixed;
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@ -126,7 +125,6 @@ struct SWVoiceIn {
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AudioState *s;
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AudioState *s;
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int active;
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int active;
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struct audio_pcm_info info;
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struct audio_pcm_info info;
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int64_t ratio;
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void *rate;
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void *rate;
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size_t total_hw_samples_acquired;
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size_t total_hw_samples_acquired;
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STSampleBuffer resample_buf;
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STSampleBuffer resample_buf;
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@ -108,32 +108,23 @@ static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
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static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
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static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
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{
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{
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HW *hw = sw->hw;
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HW *hw = sw->hw;
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int samples;
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uint64_t samples;
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if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) {
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if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) {
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return 0;
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return 0;
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}
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}
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#ifdef DAC
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samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq);
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samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio;
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#else
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samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32;
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#endif
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if (audio_bug(__func__, samples < 0)) {
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dolog("Can not allocate buffer for `%s' (%d samples)\n",
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SW_NAME(sw), samples);
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return -1;
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}
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if (samples == 0) {
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if (samples == 0) {
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size_t f_fe_min;
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uint64_t f_fe_min;
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uint64_t f_be = (uint32_t)hw->info.freq;
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/* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */
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/* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */
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f_fe_min = (hw->info.freq + HWBUF.size - 1) / HWBUF.size;
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f_fe_min = (f_be + HWBUF.size - 1) / HWBUF.size;
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qemu_log_mask(LOG_UNIMP,
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qemu_log_mask(LOG_UNIMP,
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AUDIO_CAP ": The guest selected a " NAME " sample rate"
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AUDIO_CAP ": The guest selected a " NAME " sample rate"
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" of %d Hz for %s. Only sample rates >= %zu Hz are"
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" of %d Hz for %s. Only sample rates >= %" PRIu64 " Hz"
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" supported.\n",
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" are supported.\n",
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sw->info.freq, sw->name, f_fe_min);
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sw->info.freq, sw->name, f_fe_min);
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return -1;
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return -1;
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}
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}
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@ -141,9 +132,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
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/*
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/*
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* Allocate one additional audio frame that is needed for upsampling
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* Allocate one additional audio frame that is needed for upsampling
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* if the resample buffer size is small. For large buffer sizes take
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* if the resample buffer size is small. For large buffer sizes take
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* care of overflows.
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* care of overflows and truncation.
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*/
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*/
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samples = samples < INT_MAX ? samples + 1 : INT_MAX;
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samples = samples < SIZE_MAX ? samples + 1 : SIZE_MAX;
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sw->resample_buf.buffer = g_new0(st_sample, samples);
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sw->resample_buf.buffer = g_new0(st_sample, samples);
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sw->resample_buf.size = samples;
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sw->resample_buf.size = samples;
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sw->resample_buf.pos = 0;
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sw->resample_buf.pos = 0;
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@ -170,11 +161,8 @@ static int glue (audio_pcm_sw_init_, TYPE) (
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sw->hw = hw;
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sw->hw = hw;
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sw->active = 0;
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sw->active = 0;
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#ifdef DAC
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#ifdef DAC
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sw->ratio = ((int64_t) sw->hw->info.freq << 32) / sw->info.freq;
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sw->total_hw_samples_mixed = 0;
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sw->total_hw_samples_mixed = 0;
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sw->empty = 1;
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sw->empty = 1;
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#else
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sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq;
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#endif
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#endif
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if (sw->info.is_float) {
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if (sw->info.is_float) {
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