[FFmpeg-cvslog] Replace usages of av_get_bits_per_sample_fmt() with av_get_bytes_per_sample ().
Justin Ruggles
git at videolan.org
Tue Jun 21 03:45:24 CEST 2011
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Tue Jun 7 13:40:22 2011 -0400| [e6c52cee541ba23a7aec525f72dff73c188dad06] | committer: Justin Ruggles
Replace usages of av_get_bits_per_sample_fmt() with av_get_bytes_per_sample().
av_get_bits_per_sample_fmt() is deprecated.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e6c52cee541ba23a7aec525f72dff73c188dad06
---
ffmpeg.c | 6 +++---
ffplay.c | 2 +-
libavcodec/aacdec.c | 2 +-
libavcodec/ac3dec.c | 2 +-
libavcodec/alsdec.c | 4 ++--
libavcodec/dca.c | 2 +-
libavcodec/resample.c | 4 ++--
libavcodec/utils.c | 2 +-
libavcodec/vmdav.c | 2 +-
libavcodec/vorbisdec.c | 2 +-
libavfilter/defaults.c | 2 +-
libavformat/matroskaenc.c | 2 +-
12 files changed, 16 insertions(+), 16 deletions(-)
diff --git a/ffmpeg.c b/ffmpeg.c
index b284087..9f2d5b9 100644
--- a/ffmpeg.c
+++ b/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;
diff --git a/ffplay.c b/ffplay.c
index c9891d3..ffd4c23 100644
--- a/ffplay.c
+++ b/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) {
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index 69aacb8..26ce204 100644
--- a/libavcodec/aacdec.c
+++ b/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",
diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c
index 2966c33..42b62ef 100644
--- a/libavcodec/ac3dec.c
+++ b/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);
}
diff --git a/libavcodec/alsdec.c b/libavcodec/alsdec.c
index 17c5490..055bfd0 100644
--- a/libavcodec/alsdec.c
+++ b/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);
diff --git a/libavcodec/dca.c b/libavcodec/dca.c
index a9b2c9b..68731c9 100644
--- a/libavcodec/dca.c
+++ b/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;
diff --git a/libavcodec/resample.c b/libavcodec/resample.c
index 0bebe1a..04bbbf0 100644
--- a/libavcodec/resample.c
+++ b/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,
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index 1e58864..146dd30 100644
--- a/libavcodec/utils.c
+++ b/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
diff --git a/libavcodec/vmdav.c b/libavcodec/vmdav.c
index d258252..283c213 100644
--- a/libavcodec/vmdav.c
+++ b/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",
diff --git a/libavcodec/vorbisdec.c b/libavcodec/vorbisdec.c
index 017102e..9fc6068 100644
--- a/libavcodec/vorbisdec.c
+++ b/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 ;
}
diff --git a/libavfilter/defaults.c b/libavfilter/defaults.c
index 146f1c7..b891ab1 100644
--- a/libavfilter/defaults.c
+++ b/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;
diff --git a/libavformat/matroskaenc.c b/libavformat/matroskaenc.c
index fde1470..e485539 100644
--- a/libavformat/matroskaenc.c
+++ b/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);
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