diff options
author | Volker Rümelin <vr_qemu@t-online.de> | 2023-02-24 20:05:49 +0100 |
---|---|---|
committer | Marc-André Lureau <marcandre.lureau@redhat.com> | 2023-03-06 10:30:23 +0400 |
commit | 1a01df3db89010d40eb43889c3272d864b3b9430 (patch) | |
tree | fba00f749f3ac1013c6ae9ae726dbd7d5eb25153 /audio/audio.c | |
parent | 1fe3cae39f059c9fc2010e3c51c0bbd696cbf880 (diff) | |
download | qemu-1a01df3db89010d40eb43889c3272d864b3b9430.zip qemu-1a01df3db89010d40eb43889c3272d864b3b9430.tar.gz qemu-1a01df3db89010d40eb43889c3272d864b3b9430.tar.bz2 |
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.c | 43 |
1 files changed, 18 insertions, 25 deletions
diff --git a/audio/audio.c b/audio/audio.c index 556696b..e18b5e9 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); } |