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authorVolker Rümelin <vr_qemu@t-online.de>2023-02-24 20:05:49 +0100
committerMarc-André Lureau <marcandre.lureau@redhat.com>2023-03-06 10:30:23 +0400
commit1a01df3db89010d40eb43889c3272d864b3b9430 (patch)
treefba00f749f3ac1013c6ae9ae726dbd7d5eb25153 /audio/audio.c
parent1fe3cae39f059c9fc2010e3c51c0bbd696cbf880 (diff)
audio: make playback packet length calculation exact
Introduce the new function st_rate_frames_in() to calculate the exact number of audio input frames needed to get a given number of audio output frames. The exact number of frames depends only on the difference of opos - ipos and the number of output frames. When downsampling, this function returns the maximum number of input frames needed. This new function replaces the audio_frontend_frames_out() function, which calculated the average number of input frames rounded down to the nearest integer. Because audio_frontend_frames_out() also limited the number of input frames to the size of the resample buffer, st_rate_frames_in() is not a direct replacement and two additional MIN() functions are needed. One to prevent resample buffer overflows and one to limit the available bytes for the audio frontends. After this patch the audio packet length calculation for playback is exact. When upsampling, it's still possible that the audio frontends can't write the last audio frame. This will be fixed later. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-9-vr_qemu@t-online.de>
Diffstat (limited to 'audio/audio.c')
-rw-r--r--audio/audio.c43
1 files changed, 18 insertions, 25 deletions
diff --git a/audio/audio.c b/audio/audio.c
index 556696b095..e18b5e98c5 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -701,8 +701,8 @@ static void audio_pcm_sw_resample_out(SWVoiceOut *sw,
static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len)
{
HWVoiceOut *hw = sw->hw;
- size_t live, dead, hw_free;
- size_t frames_in_max, total_in, total_out;
+ size_t live, dead, hw_free, sw_max, fe_max;
+ size_t frames_in_max, frames_out_max, total_in, total_out;
live = sw->total_hw_samples_mixed;
if (audio_bug(__func__, live > hw->mix_buf.size)) {
@@ -720,17 +720,21 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len)
dead = hw->mix_buf.size - live;
hw_free = audio_pcm_hw_get_free(hw);
hw_free = hw_free > live ? hw_free - live : 0;
- frames_in_max = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio;
- frames_in_max = MIN(frames_in_max, buf_len / sw->info.bytes_per_frame);
- if (frames_in_max) {
- sw->conv(sw->resample_buf.buffer, buf, frames_in_max);
+ frames_out_max = MIN(dead, hw_free);
+ sw_max = st_rate_frames_in(sw->rate, frames_out_max);
+ fe_max = MIN(buf_len / sw->info.bytes_per_frame, sw->resample_buf.size);
+ frames_in_max = MIN(sw_max, fe_max);
- if (!sw->hw->pcm_ops->volume_out) {
- mixeng_volume(sw->resample_buf.buffer, frames_in_max, &sw->vol);
- }
+ if (!frames_in_max) {
+ return 0;
}
- audio_pcm_sw_resample_out(sw, frames_in_max, MIN(dead, hw_free),
+ sw->conv(sw->resample_buf.buffer, buf, frames_in_max);
+ if (!sw->hw->pcm_ops->volume_out) {
+ mixeng_volume(sw->resample_buf.buffer, frames_in_max, &sw->vol);
+ }
+
+ audio_pcm_sw_resample_out(sw, frames_in_max, frames_out_max,
&total_in, &total_out);
sw->total_hw_samples_mixed += total_out;
@@ -1000,18 +1004,6 @@ static size_t audio_get_avail (SWVoiceIn *sw)
return live;
}
-/**
- * audio_frontend_frames_out() - returns the number of frames needed to
- * get frames_out frames after resampling
- *
- * @sw: audio playback frontend
- * @frames_out: number of frames
- */
-static size_t audio_frontend_frames_out(SWVoiceOut *sw, size_t frames_out)
-{
- return ((int64_t)frames_out << 32) / sw->ratio;
-}
-
static size_t audio_get_free(SWVoiceOut *sw)
{
size_t live, dead;
@@ -1031,8 +1023,8 @@ static size_t audio_get_free(SWVoiceOut *sw)
dead = sw->hw->mix_buf.size - live;
#ifdef DEBUG_OUT
- dolog("%s: get_free live %zu dead %zu frontend frames %zu\n",
- SW_NAME(sw), live, dead, audio_frontend_frames_out(sw, dead));
+ dolog("%s: get_free live %zu dead %zu frontend frames %u\n",
+ SW_NAME(sw), live, dead, st_rate_frames_in(sw->rate, dead));
#endif
return dead;
@@ -1161,12 +1153,13 @@ static void audio_run_out (AudioState *s)
size_t free;
if (hw_free > sw->total_hw_samples_mixed) {
- free = audio_frontend_frames_out(sw,
+ free = st_rate_frames_in(sw->rate,
MIN(sw_free, hw_free - sw->total_hw_samples_mixed));
} else {
free = 0;
}
if (free > 0) {
+ free = MIN(free, sw->resample_buf.size);
sw->callback.fn(sw->callback.opaque,
free * sw->info.bytes_per_frame);
}