audio: fix sw->buf size for audio recording

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>

diff --git a/audio/audio.c b/audio/audio.c
index ed2b9d5f7e1567e8f77186e585565bc055ca08dd..886725747bdaec8e74742d0a9928d943bbc28b7d 100644 (file)
--- 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 98ab557684d83dd6638e995a958d021507101df9..720a32e57e7dd98e29d6f3d0cb529464f6a27a8f 100644 (file)
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -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) {