Remove sh_audio->samplesize

This member was redundant. sh_audio->sample_format indicates the sample
size already.

The TV code is a bit strange: the redundant sample size was part of the
internal TV interface. Assume it's really redundant and not something
else. The PCM decoder ignores the sample size anyway.

audio/decode/ad_lavc.c

@@ -184,7 +184,6 @@ static int setup_format(sh_audio_t *sh_audio,
         sh_audio->channels = lavc_chmap;
         sh_audio->samplerate = samplerate;
         sh_audio->sample_format = sample_format;
-        sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
         return 1;
     }
     return 0;

audio/decode/ad_mpg123.c

@@ -206,22 +206,18 @@ static int set_format(sh_audio_t *sh, struct ad_mpg123_context *con)
         switch (encoding) {
         case MPG123_ENC_SIGNED_8:
             sh->sample_format = AF_FORMAT_S8;
-            sh->samplesize    = 1;
             break;
         case MPG123_ENC_SIGNED_16:
             sh->sample_format = AF_FORMAT_S16_NE;
-            sh->samplesize    = 2;
             break;
         /* To stay compatible with the oldest libmpg123 headers, do not rely
          * on float and 32 bit encoding symbols being defined.
          * Those formats came later */
         case 0x1180: /* MPG123_ENC_SIGNED_32 */
             sh->sample_format = AF_FORMAT_S32_NE;
-            sh->samplesize    = 4;
             break;
         case 0x200: /* MPG123_ENC_FLOAT_32 */
             sh->sample_format = AF_FORMAT_FLOAT_NE;
-            sh->samplesize    = 4;
             break;
         default:
             /* This means we got a funny custom build of libmpg123 that only supports an unknown format. */
@@ -233,7 +229,7 @@ static int set_format(sh_audio_t *sh, struct ad_mpg123_context *con)
         /* Going to decode directly to MPlayer's memory. It is important
          * to have MPG123_AUTO_RESAMPLE disabled for the buffer size
          * being an all-time limit. */
-        sh->audio_out_minsize = 1152 * 2 * sh->samplesize;
+        sh->audio_out_minsize = 1152 * 2 * (af_fmt2bits(sh->sample_format) / 8);
 #endif
         con->new_format = 0;
     }

audio/decode/dec_audio.c

@@ -58,7 +58,6 @@ static int init_audio_codec(sh_audio_t *sh_audio, const char *decoder)
 {
     assert(!sh_audio->initialized);
     resync_audio_stream(sh_audio);
-    sh_audio->samplesize = 4;
     sh_audio->sample_format = AF_FORMAT_FLOAT_NE;
     sh_audio->audio_out_minsize = 8192; // default, preinit() may change it
     if (!sh_audio->ad_driver->preinit(sh_audio)) {
@@ -305,7 +304,8 @@ int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen)
     // Indicates that a filter seems to be buffering large amounts of data
     int huge_filter_buffer = 0;
     // Decoded audio must be cut at boundaries of this many bytes
-    int unitsize = sh_audio->channels.num * sh_audio->samplesize * 16;
+    int bps = af_fmt2bits(sh_audio->sample_format) / 8;
+    int unitsize = sh_audio->channels.num * bps * 16;
 
     /* Filter output size will be about filter_multiplier times input size.
      * If some filter buffers audio in big blocks this might only hold

demux/demux.c

@@ -259,7 +259,6 @@ struct sh_stream *new_sh_stream(demuxer_t *demuxer, enum stream_type type)
             struct sh_audio *sht = talloc_zero(demuxer, struct sh_audio);
             sht->gsh = sh;
             sht->opts = sh->opts;
-            sht->samplesize = 2;
             sht->sample_format = AF_FORMAT_S16_NE;
             sh->audio = sht;
             break;

demux/stheader.h

@@ -90,7 +90,6 @@ typedef struct sh_audio {
     // output format:
     int sample_format;
     int samplerate;
-    int samplesize;
     struct mp_chmap channels;
     int i_bps; // == bitrate  (compressed bytes/sec)
     // decoder buffers:

mpvcore/player/audio.c

@@ -177,7 +177,7 @@ double written_audio_pts(struct MPContext *mpctx)
         return MP_NOPTS_VALUE;
 
     double bps = sh_audio->channels.num * sh_audio->samplerate *
-                 sh_audio->samplesize;
+                 (af_fmt2bits(sh_audio->sample_format) / 8);
 
     // first calculate the end pts of audio that has been output by decoder
     double a_pts = sh_audio->pts;

stream/tv.c

@@ -794,8 +794,6 @@ static int demux_open_tv(demuxer_t *demuxer, enum demux_check check)
 
 	funcs->control(tvh->priv, TVI_CONTROL_AUD_GET_SAMPLERATE,
                    &sh_audio->samplerate);
-	funcs->control(tvh->priv, TVI_CONTROL_AUD_GET_SAMPLESIZE,
-                   &sh_audio->samplesize);
         int nchannels = sh_audio->channels.num;
 	funcs->control(tvh->priv, TVI_CONTROL_AUD_GET_CHANNELS,
                    &nchannels);
@@ -804,17 +802,18 @@ static int demux_open_tv(demuxer_t *demuxer, enum demux_check check)
         sh_audio->gsh->codec = "mp-pcm";
 	sh_audio->format = audio_format;
 
+        int samplesize = af_fmt2bits(audio_format) / 8;
+
 	sh_audio->i_bps =
-	    sh_audio->samplerate * sh_audio->samplesize *
-	    sh_audio->channels.num;
+	    sh_audio->samplerate * samplesize * sh_audio->channels.num;
 
 	// emulate WF for win32 codecs:
 	sh_audio->wf = malloc(sizeof(*sh_audio->wf));
 	sh_audio->wf->wFormatTag = sh_audio->format;
 	sh_audio->wf->nChannels = sh_audio->channels.num;
-	sh_audio->wf->wBitsPerSample = sh_audio->samplesize * 8;
+	sh_audio->wf->wBitsPerSample = samplesize * 8;
 	sh_audio->wf->nSamplesPerSec = sh_audio->samplerate;
-	sh_audio->wf->nBlockAlign = sh_audio->samplesize * sh_audio->channels.num;
+	sh_audio->wf->nBlockAlign = samplesize * sh_audio->channels.num;
 	sh_audio->wf->nAvgBytesPerSec = sh_audio->i_bps;
 
 	mp_tmsg(MSGT_DECVIDEO, MSGL_V, "  TV audio: %d channels, %d bits, %d Hz\n",

stream/tv.h

@@ -188,7 +188,6 @@ typedef struct {
 /* AUDIO controls */
 #define TVI_CONTROL_AUD_GET_FORMAT	0x301
 #define TVI_CONTROL_AUD_GET_SAMPLERATE	0x302
-#define TVI_CONTROL_AUD_GET_SAMPLESIZE	0x303
 #define TVI_CONTROL_AUD_GET_CHANNELS	0x304
 #define TVI_CONTROL_AUD_SET_SAMPLERATE	0x305
 

stream/tvi_v4l2.c

@@ -974,13 +974,6 @@ static int do_control(priv_t *priv, int cmd, void *arg)
         mp_msg(MSGT_TV, MSGL_V, "%s: get audio samplerate: %d\n",
                info.short_name, *(int *)arg);
         return TVI_CONTROL_TRUE;
-    case TVI_CONTROL_AUD_GET_SAMPLESIZE:
-        init_audio(priv);
-        if (!priv->audio_initialized) return TVI_CONTROL_FALSE;
-        *(int *)arg = priv->audio_in.bytes_per_sample;
-        mp_msg(MSGT_TV, MSGL_V, "%s: get audio samplesize: %d\n",
-               info.short_name, *(int *)arg);
-        return TVI_CONTROL_TRUE;
     case TVI_CONTROL_AUD_GET_CHANNELS:
         init_audio(priv);
         if (!priv->audio_initialized) return TVI_CONTROL_FALSE;