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ffmpeg
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Replace usages of av_get_bits_per_sample_fmt() with av_get_bytes_per_sample(). 

av_get_bits_per_sample_fmt() is deprecated.
This commit is contained in:
Justin Ruggles 2011-06-07 13:40:22 -04:00
parent c5ee740745
commit e6c52cee54
12 changed files with 16 additions and 16 deletions
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6
ffmpeg.c
View File

@ -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;

2
ffplay.c
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@ -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) {

2
libavcodec/aacdec.c
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@ -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",

2
libavcodec/ac3dec.c
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@ -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);
}

4
libavcodec/alsdec.c
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@ -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);

2
libavcodec/dca.c
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@ -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;

4
libavcodec/resample.c
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@ -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,

2
libavcodec/utils.c
View File

@ -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

2
libavcodec/vmdav.c
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@ -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",

2
libavcodec/vorbisdec.c
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@ -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 ;
}

2
libavfilter/defaults.c
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@ -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;

2
libavformat/matroskaenc.c
View File

@ -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);