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_q...@t-online.de> --- audio/audio.c | 2 +- audio/audio_template.h | 4 ++++ 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/audio/audio.c b/audio/audio.c index ba0c62b120..60c7472d37 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -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) diff --git a/audio/audio_template.h b/audio/audio_template.h index 7192b19e73..6a0337ac6b 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -112,7 +112,11 @@ 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) { -- 2.35.3
