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audio: fix sw->buf size for audio recording

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The calculation of the buffer size needed to store audio samples
after resampling is wrong for audio recording. For audio recording
sw->ratio is calculated as

sw->ratio = frontend sample rate / backend sample rate.

From this follows

frontend samples = frontend sample rate / backend sample rate
 * backend samples
frontend samples = sw->ratio * backend samples

In 2 of 3 places in the audio recording code where sw->ratio
is used in a calculation to get the number of frontend frames,
the calculation is wrong. Fix this. The 3rd formula in
audio_pcm_sw_read() is correct.

Resolves: https://gitlab.com/qemu-project/qemu/-/issues/71
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Message-Id: <20220923183640.8314-11-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This commit is contained in:
Volker Rümelin 2022-09-23 20:36:39 +02:00 committed by Gerd Hoffmann
parent 0724c57988
commit b73ef11ff6
2 changed files with 5 additions and 1 deletions
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2
audio/audio.c
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@ -995,7 +995,7 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
*/
static size_t audio_frontend_frames_in(SWVoiceIn *sw, size_t frames_in)
{
return ((int64_t)frames_in << 32) / sw->ratio;
return (int64_t)frames_in * sw->ratio >> 32;
}
static size_t audio_get_avail (SWVoiceIn *sw)

4
audio/audio_template.h
View File

@ -110,7 +110,11 @@ 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
sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample));
if (!sw->buf) {