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.

3 files changed, 3 insertions, 8 deletions

diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
index abd47f2fa3..1e63f0c3f2 100644
--- a/audio/decode/ad_lavc.c
+++ b/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;
diff --git a/audio/decode/ad_mpg123.c b/audio/decode/ad_mpg123.c
index df9e82306d..47cb5d2039 100644
--- a/audio/decode/ad_mpg123.c
+++ b/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;
     }
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
index 127139ff60..e381a12a3c 100644
--- a/audio/decode/dec_audio.c
+++ b/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