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.

audio/audio.c

@@ -31,7 +31,7 @@ static void update_redundant_info(struct mp_audio *mpa)
     assert(mp_chmap_is_empty(&mpa->channels) ||
            mp_chmap_is_valid(&mpa->channels));
     mpa->nch = mpa->channels.num;
-    mpa->bps = af_fmt2bits(mpa->format) / 8;
+    mpa->bps = af_fmt2bps(mpa->format);
     if (af_fmt_is_planar(mpa->format)) {
         mpa->spf = 1;
         mpa->num_planes = mpa->nch;

audio/audio.h

@@ -35,7 +35,7 @@ struct mp_audio {
     int nch;            // number of channels (redundant with chmap)
     int spf;            // sub-samples per sample on each plane
     int num_planes;     // number of planes
-    int bps;            // size of sub-samples (af_fmt2bits(format) / 8)
+    int bps;            // size of sub-samples (af_fmt2bps(format))
 
     // private
     int allocated[MP_NUM_CHANNELS]; // use mp_audio_get_allocated_size()

audio/decode/ad_mpg123.c

@@ -160,7 +160,7 @@ static int set_format(struct dec_audio *da)
             return MPG123_ERR;
         }
         mp_audio_set_format(&da->decoded, af);
-        con->sample_size = channels * (af_fmt2bits(af) / 8);
+        con->sample_size = channels * af_fmt2bps(af);
         con->new_format = 0;
     }
     return ret;

audio/format.c

@@ -28,21 +28,26 @@
 #include "common/common.h"
 #include "audio/filter/af.h"
 
-int af_fmt2bits(int format)
+int af_fmt2bps(int format)
 {
-    if (AF_FORMAT_IS_AC3(format)) return 16;
+    if (AF_FORMAT_IS_AC3(format)) return 2;
     if (format == AF_FORMAT_UNKNOWN)
         return 0;
     switch (format & AF_FORMAT_BITS_MASK) {
-    case AF_FORMAT_8BIT:  return 8;
-    case AF_FORMAT_16BIT: return 16;
-    case AF_FORMAT_24BIT: return 24;
-    case AF_FORMAT_32BIT: return 32;
-    case AF_FORMAT_64BIT: return 64;
+    case AF_FORMAT_8BIT:  return 1;
+    case AF_FORMAT_16BIT: return 2;
+    case AF_FORMAT_24BIT: return 3;
+    case AF_FORMAT_32BIT: return 4;
+    case AF_FORMAT_64BIT: return 8;
     }
     return 0;
 }
 
+int af_fmt2bits(int format)
+{
+    return af_fmt2bps(format) * 8;
+}
+
 static int bits_to_mask(int bits)
 {
     switch (bits) {
@@ -159,7 +164,7 @@ const char *af_fmt_to_str(int format)
 int af_fmt_seconds_to_bytes(int format, float seconds, int channels, int samplerate)
 {
     assert(!af_fmt_is_planar(format));
-    int bps      = (af_fmt2bits(format) / 8);
+    int bps      = af_fmt2bps(format);
     int framelen = channels * bps;
     int bytes    = seconds  * bps * samplerate;
     if (bytes % framelen)

audio/format.h

@@ -151,6 +151,7 @@ extern const struct af_fmt_entry af_fmtstr_table[];
 int af_str2fmt_short(bstr str);
 const char *af_fmt_to_str(int format);
 
+int af_fmt2bps(int format);
 int af_fmt2bits(int format);
 int af_fmt_change_bits(int format, int bits);
 

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;

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

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;
 

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;
 

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,

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)))

demux/demux_raw.c

@@ -94,7 +94,7 @@ static int demux_rawaudio_open(demuxer_t *demuxer, enum demux_check check)
     sh_audio->channels = channels;
     w->nChannels = sh_audio->channels.num;
     w->nSamplesPerSec = sh_audio->samplerate = samplerate;
-    int samplesize = (af_fmt2bits(aformat) + 7) / 8;
+    int samplesize = af_fmt2bps(aformat);
     w->nAvgBytesPerSec = samplerate * samplesize * w->nChannels;
     w->nBlockAlign = w->nChannels * samplesize;
     w->wBitsPerSample = 8 * samplesize;

stream/tv.c

@@ -802,7 +802,7 @@ static int demux_open_tv(demuxer_t *demuxer, enum demux_check check)
         sh_a->codec = "mp-pcm";
         sh_a->format = audio_format;
 
-        int samplesize = af_fmt2bits(audio_format) / 8;
+        int samplesize = af_fmt2bps(audio_format);
         int block_align = samplesize * sh_audio->channels.num;
         int bytes_per_second = sh_audio->samplerate * block_align;