[PATCH 11/12] audio: fix sw->buf size for audio recording

["Volker Rümelin" <vr_qemu@t-online.de>]

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>
---
 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