af_fmt2bits: change to af_fmt2bps (bytes/sample) where appropriate

In most places where af_fmt2bits is called to get the bits/sample, the
result is immediately converted to bytes/sample. Avoid this by getting
bytes/sample directly by introducing af_fmt2bps.

6 files changed, 9 insertions, 9 deletions

diff --git a/audio/out/ao.c b/audio/out/ao.c
index 313f4f7554..358762b73c 100644
--- a/audio/out/ao.c
+++ b/audio/out/ao.c
@@ -171,7 +171,7 @@ static struct ao *ao_create(bool probing, struct mpv_global *global,
     if (ao->driver->init(ao) < 0)
         goto error;
 
-    ao->sstride = af_fmt2bits(ao->format) / 8;
+    ao->sstride = af_fmt2bps(ao->format);
     ao->num_planes = 1;
     if (af_fmt_is_planar(ao->format)) {
         ao->num_planes = ao->channels.num;
diff --git a/audio/out/ao_dsound.c b/audio/out/ao_dsound.c
index da0c14cdc2..e8d37e48f7 100644
--- a/audio/out/ao_dsound.c
+++ b/audio/out/ao_dsound.c
@@ -402,7 +402,7 @@ static int init(struct ao *ao)
     //set our audio parameters
     ao->samplerate = rate;
     ao->format = format;
-    ao->bps = ao->channels.num * rate * (af_fmt2bits(format) >> 3);
+    ao->bps = ao->channels.num * rate * af_fmt2bps(format);
     int buffersize = ao->bps; // space for 1 sec
     MP_VERBOSE(ao, "Samplerate:%iHz Channels:%i Format:%s\n", rate,
                ao->channels.num, af_fmt_to_str(format));
@@ -422,9 +422,9 @@ static int init(struct ao *ao)
     } else {
         wformat.Format.wFormatTag = (ao->channels.num > 2)
                                     ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
-        wformat.Format.wBitsPerSample = af_fmt2bits(format);
-        wformat.Format.nBlockAlign = wformat.Format.nChannels *
-                                     (wformat.Format.wBitsPerSample >> 3);
+        int bps = af_fmt2bps(format);
+        wformat.Format.wBitsPerSample = bps * 8;
+        wformat.Format.nBlockAlign = wformat.Format.nChannels * bps;
     }
 
     // fill in primary sound buffer descriptor
diff --git a/audio/out/ao_lavc.c b/audio/out/ao_lavc.c
index 0bfe2eb9cb..7eb81f1d4e 100644
--- a/audio/out/ao_lavc.c
+++ b/audio/out/ao_lavc.c
@@ -137,7 +137,7 @@ static int init(struct ao *ao)
 
     select_format(ao, codec);
 
-    ac->sample_size = af_fmt2bits(ao->format) / 8;
+    ac->sample_size = af_fmt2bps(ao->format);
     ac->stream->codec->sample_fmt = af_to_avformat(ao->format);
     ac->stream->codec->bits_per_raw_sample = ac->sample_size * 8;
 
diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c
index ca7539b590..545ddca3f8 100644
--- a/audio/out/ao_oss.c
+++ b/audio/out/ao_oss.c
@@ -419,7 +419,7 @@ ac3_retry:
 #endif
     }
 
-    ao->bps = ao->channels.num * (af_fmt2bits(ao->format) / 8);
+    ao->bps = ao->channels.num * af_fmt2bps(ao->format);
     p->outburst -= p->outburst % ao->bps; // round down
     ao->bps *= ao->samplerate;
 
diff --git a/audio/out/ao_pcm.c b/audio/out/ao_pcm.c
index ab5faf6173..1e8a0adf75 100644
--- a/audio/out/ao_pcm.c
+++ b/audio/out/ao_pcm.c
@@ -143,7 +143,7 @@ static int init(struct ao *ao)
     if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
         return -1;
 
-    ao->bps = ao->channels.num * ao->samplerate * (af_fmt2bits(ao->format) / 8);
+    ao->bps = ao->channels.num * ao->samplerate * af_fmt2bps(ao->format);
 
     MP_INFO(ao, "File: %s (%s)\nPCM: Samplerate: %d Hz Channels: %d Format: %s\n",
             priv->outputfilename,
diff --git a/audio/out/ao_portaudio.c b/audio/out/ao_portaudio.c
index ae8b76e830..52c67d2a3c 100644
--- a/audio/out/ao_portaudio.c
+++ b/audio/out/ao_portaudio.c
@@ -204,7 +204,7 @@ static int init(struct ao *ao)
 
     ao->format = fmt->mp_format;
     sp.sampleFormat = fmt->pa_format;
-    int framelen = ao->channels.num * (af_fmt2bits(ao->format) / 8);
+    int framelen = ao->channels.num * af_fmt2bps(ao->format);
     ao->bps = ao->samplerate * framelen;
 
     if (!CHECK_PA_RET(Pa_IsFormatSupported(NULL, &sp, ao->samplerate)))