Replace usages of av_get_bits_per_sample_fmt() with av_get_bytes_per_sample().

av_get_bits_per_sample_fmt() is deprecated.

ffmpeg.c

@@ -778,8 +778,8 @@ static void do_audio_out(AVFormatContext *s,
     int size_out, frame_bytes, ret, resample_changed;
     AVCodecContext *enc= ost->st->codec;
     AVCodecContext *dec= ist->st->codec;
-    int osize= av_get_bits_per_sample_fmt(enc->sample_fmt)/8;
-    int isize= av_get_bits_per_sample_fmt(dec->sample_fmt)/8;
+    int osize = av_get_bytes_per_sample(enc->sample_fmt);
+    int isize = av_get_bytes_per_sample(dec->sample_fmt);
     const int coded_bps = av_get_bits_per_sample(enc->codec->id);
 
 need_realloc:
@@ -1481,7 +1481,7 @@ static int output_packet(AVInputStream *ist, int ist_index,
 #endif
 
     AVPacket avpkt;
-    int bps = av_get_bits_per_sample_fmt(ist->st->codec->sample_fmt)>>3;
+    int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt);
 
     if(ist->next_pts == AV_NOPTS_VALUE)
         ist->next_pts= ist->pts;

ffplay.c

@@ -2032,7 +2032,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
             if (is->reformat_ctx) {
                 const void *ibuf[6]= {is->audio_buf1};
                 void *obuf[6]= {is->audio_buf2};
-                int istride[6]= {av_get_bits_per_sample_fmt(dec->sample_fmt)/8};
+                int istride[6]= {av_get_bytes_per_sample(dec->sample_fmt)};
                 int ostride[6]= {2};
                 int len= data_size/istride[0];
                 if (av_audio_convert(is->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) {

libavcodec/aacdec.c

@@ -2177,7 +2177,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
     }
 
     data_size_tmp = samples * avctx->channels *
-                    (av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
+                    av_get_bytes_per_sample(avctx->sample_fmt);
     if (*data_size < data_size_tmp) {
         av_log(avctx, AV_LOG_ERROR,
                "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",

libavcodec/ac3dec.c

@@ -1422,7 +1422,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
         }
     }
     *data_size = s->num_blocks * 256 * avctx->channels *
-                 (av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
+                 av_get_bytes_per_sample(avctx->sample_fmt);
     return FFMIN(buf_size, s->frame_size);
 }
 

libavcodec/alsdec.c

@@ -1450,7 +1450,7 @@ static int decode_frame(AVCodecContext *avctx,
 
     // check for size of decoded data
     size = ctx->cur_frame_length * avctx->channels *
-           (av_get_bits_per_sample_fmt(avctx->sample_fmt) >> 3);
+           av_get_bytes_per_sample(avctx->sample_fmt);
 
     if (size > *data_size) {
         av_log(avctx, AV_LOG_ERROR, "Decoded data exceeds buffer size.\n");
@@ -1714,7 +1714,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
         ctx->crc_buffer = av_malloc(sizeof(*ctx->crc_buffer) *
                                     ctx->cur_frame_length *
                                     avctx->channels *
-                                    (av_get_bits_per_sample_fmt(avctx->sample_fmt) >> 3));
+                                    av_get_bytes_per_sample(avctx->sample_fmt));
         if (!ctx->crc_buffer) {
             av_log(avctx, AV_LOG_ERROR, "Allocating buffer memory failed.\n");
             decode_end(avctx);

libavcodec/dca.c

@@ -1813,7 +1813,7 @@ static int dca_decode_frame(AVCodecContext * avctx,
     }
 
     out_size = 256 / 8 * s->sample_blocks * channels *
-               (av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);
+               av_get_bytes_per_sample(avctx->sample_fmt);
     if (*data_size < out_size)
         return -1;
     *data_size = out_size;

libavcodec/resample.c

@@ -187,8 +187,8 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
 
     s->sample_fmt[0]  = sample_fmt_in;
     s->sample_fmt[1]  = sample_fmt_out;
-    s->sample_size[0] = av_get_bits_per_sample_fmt(s->sample_fmt[0]) >> 3;
-    s->sample_size[1] = av_get_bits_per_sample_fmt(s->sample_fmt[1]) >> 3;
+    s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
+    s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
 
     if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
         if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,

libavcodec/utils.c

@@ -1131,7 +1131,7 @@ int av_get_bits_per_sample(enum CodecID codec_id){
 
 #if FF_API_OLD_SAMPLE_FMT
 int av_get_bits_per_sample_format(enum AVSampleFormat sample_fmt) {
-    return av_get_bits_per_sample_fmt(sample_fmt);
+    return av_get_bytes_per_sample(sample_fmt) << 3;
 }
 #endif
 

libavcodec/vmdav.c

@@ -447,7 +447,7 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
         avctx->sample_fmt = AV_SAMPLE_FMT_S16;
     else
         avctx->sample_fmt = AV_SAMPLE_FMT_U8;
-    s->out_bps = av_get_bits_per_sample_fmt(avctx->sample_fmt) >> 3;
+    s->out_bps = av_get_bytes_per_sample(avctx->sample_fmt);
 
     av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
            "block align = %d, sample rate = %d\n",

libavcodec/vorbisdec.c

@@ -1646,7 +1646,7 @@ static int vorbis_decode_frame(AVCodecContext *avccontext,
                                                vc->audio_channels);
 
     *data_size = len * vc->audio_channels *
-                 (av_get_bits_per_sample_fmt(avccontext->sample_fmt) / 8);
+                 av_get_bytes_per_sample(avccontext->sample_fmt);
 
     return buf_size ;
 }

libavfilter/defaults.c

@@ -84,7 +84,7 @@ AVFilterBufferRef *avfilter_default_get_audio_buffer(AVFilterLink *link, int per
     samples->refcount   = 1;
     samples->free       = ff_avfilter_default_free_buffer;
 
-    sample_size = av_get_bits_per_sample_fmt(sample_fmt) >>3;
+    sample_size = av_get_bytes_per_sample(sample_fmt);
     chans_nb = av_get_channel_layout_nb_channels(channel_layout);
 
     per_channel_size = size/chans_nb;

libavformat/matroskaenc.c

@@ -527,7 +527,7 @@ static int mkv_write_tracks(AVFormatContext *s)
         AVDictionaryEntry *tag;
 
         if (!bit_depth)
-            bit_depth = av_get_bits_per_sample_fmt(codec->sample_fmt);
+            bit_depth = av_get_bytes_per_sample(codec->sample_fmt) << 3;
 
         if (codec->codec_id == CODEC_ID_AAC)
             get_aac_sample_rates(s, codec, &sample_rate, &output_sample_rate);