Replace usages of av_get_bits_per_sample_fmt() with av_get_bytes_per_sample().

av_get_bits_per_sample_fmt() is deprecated.

Justin Ruggles committed 15 years ago

Commit e6c52ce1 parent c5ee74012 files · +16 −16Browse files

Unified · Split

12 files

ffmpeg.c +3 −3
ffplay.c +1 −1
libavcodec/aacdec.c +1 −1
libavcodec/ac3dec.c +1 −1
libavcodec/alsdec.c +2 −2
libavcodec/dca.c +1 −1
libavcodec/resample.c +2 −2
libavcodec/utils.c +1 −1
libavcodec/vmdav.c +1 −1
libavcodec/vorbisdec.c +1 −1
libavfilter/defaults.c +1 −1
libavformat/matroskaenc.c +1 −1

@@ -778,8 +778,8 @@ static void do_audio_out(AVFormatContext *s, · 777 unmodified lines
778	int size_out, frame_bytes, ret, resample_changed;	778	int size_out, frame_bytes, ret, resample_changed;
779	AVCodecContext *enc= ost->st->codec;	779	AVCodecContext *enc= ost->st->codec;
780	AVCodecContext *dec= ist->st->codec;	780	AVCodecContext *dec= ist->st->codec;
781	int osize= av_get_bits_per_sample_fmt(enc->sample_fmt)/8;	781	int osize = av_get_bytes_per_sample(enc->sample_fmt);
782	int isize= av_get_bits_per_sample_fmt(dec->sample_fmt)/8;	782	int isize = av_get_bytes_per_sample(dec->sample_fmt);
783	const int coded_bps = av_get_bits_per_sample(enc->codec->id);	783	const int coded_bps = av_get_bits_per_sample(enc->codec->id);
784		784
785	need_realloc:	785	need_realloc:
@@ -1481,7 +1481,7 @@ static int output_packet(AVInputStream *ist, int ist_index, · 695 unmodified lines
1481	#endif	1481	#endif
1482		1482
1483	AVPacket avpkt;	1483	AVPacket avpkt;
1484	int bps = av_get_bits_per_sample_fmt(ist->st->codec->sample_fmt)>>3;	1484	int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt);
1485		1485
1486	if(ist->next_pts == AV_NOPTS_VALUE)	1486	if(ist->next_pts == AV_NOPTS_VALUE)
1487	ist->next_pts= ist->pts;	1487	ist->next_pts= ist->pts;

@@ -2032,7 +2032,7 @@ static int audio_decode_frame(VideoState is, double pts_ptr) · 2031 unmodified lines
2032	if (is->reformat_ctx) {	2032	if (is->reformat_ctx) {
2033	const void *ibuf[6]= {is->audio_buf1};	2033	const void *ibuf[6]= {is->audio_buf1};
2034	void *obuf[6]= {is->audio_buf2};	2034	void *obuf[6]= {is->audio_buf2};
2035	int istride[6]= {av_get_bits_per_sample_fmt(dec->sample_fmt)/8};	2035	int istride[6]= {av_get_bytes_per_sample(dec->sample_fmt)};
2036	int ostride[6]= {2};	2036	int ostride[6]= {2};
2037	int len= data_size/istride[0];	2037	int len= data_size/istride[0];
2038	if (av_audio_convert(is->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) {	2038	if (av_audio_convert(is->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) {

libavcodec/aacdec.c+1−1

@@ -2177,7 +2177,7 @@ static int aac_decode_frame_int(AVCodecContext avctx, void data, · 2176 unmodified lines
2177	}	2177	}
2178		2178
2179	data_size_tmp = samples * avctx->channels *	2179	data_size_tmp = samples * avctx->channels *
2180	(av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);	2180	av_get_bytes_per_sample(avctx->sample_fmt);
2181	if (*data_size < data_size_tmp) {	2181	if (*data_size < data_size_tmp) {
2182	av_log(avctx, AV_LOG_ERROR,	2182	av_log(avctx, AV_LOG_ERROR,
2183	"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",	2183	"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",

libavcodec/ac3dec.c+1−1

@@ -1422,7 +1422,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void data, int data_size, · 1421 unmodified lines
1422	}	1422	}
1423	}	1423	}
1424	data_size = s->num_blocks 256 * avctx->channels *	1424	data_size = s->num_blocks 256 * avctx->channels *
1425	(av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);	1425	av_get_bytes_per_sample(avctx->sample_fmt);
1426	return FFMIN(buf_size, s->frame_size);	1426	return FFMIN(buf_size, s->frame_size);
1427	}	1427	}
1428		1428

libavcodec/alsdec.c+2−2

@@ -1450,7 +1450,7 @@ static int decode_frame(AVCodecContext *avctx, · 1449 unmodified lines
1450		1450
1451	// check for size of decoded data	1451	// check for size of decoded data
1452	size = ctx->cur_frame_length * avctx->channels *	1452	size = ctx->cur_frame_length * avctx->channels *
1453	(av_get_bits_per_sample_fmt(avctx->sample_fmt) >> 3);	1453	av_get_bytes_per_sample(avctx->sample_fmt);
1454		1454
1455	if (size > *data_size) {	1455	if (size > *data_size) {
1456	av_log(avctx, AV_LOG_ERROR, "Decoded data exceeds buffer size.\n");	1456	av_log(avctx, AV_LOG_ERROR, "Decoded data exceeds buffer size.\n");
@@ -1714,7 +1714,7 @@ static av_cold int decode_init(AVCodecContext *avctx) · 257 unmodified lines
1714	ctx->crc_buffer = av_malloc(sizeof(ctx->crc_buffer)	1714	ctx->crc_buffer = av_malloc(sizeof(ctx->crc_buffer)
1715	ctx->cur_frame_length *	1715	ctx->cur_frame_length *
1716	avctx->channels *	1716	avctx->channels *
1717	(av_get_bits_per_sample_fmt(avctx->sample_fmt) >> 3));	1717	av_get_bytes_per_sample(avctx->sample_fmt));
1718	if (!ctx->crc_buffer) {	1718	if (!ctx->crc_buffer) {
1719	av_log(avctx, AV_LOG_ERROR, "Allocating buffer memory failed.\n");	1719	av_log(avctx, AV_LOG_ERROR, "Allocating buffer memory failed.\n");
1720	decode_end(avctx);	1720	decode_end(avctx);

libavcodec/dca.c+1−1

@@ -1813,7 +1813,7 @@ static int dca_decode_frame(AVCodecContext * avctx, · 1812 unmodified lines
1813	}	1813	}
1814		1814
1815	out_size = 256 / 8 * s->sample_blocks * channels *	1815	out_size = 256 / 8 * s->sample_blocks * channels *
1816	(av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8);	1816	av_get_bytes_per_sample(avctx->sample_fmt);
1817	if (*data_size < out_size)	1817	if (*data_size < out_size)
1818	return -1;	1818	return -1;
1819	*data_size = out_size;	1819	*data_size = out_size;

libavcodec/resample.c+2−2

@@ -187,8 +187,8 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, · 186 unmodified lines
187		187
188	s->sample_fmt[0] = sample_fmt_in;	188	s->sample_fmt[0] = sample_fmt_in;
189	s->sample_fmt[1] = sample_fmt_out;	189	s->sample_fmt[1] = sample_fmt_out;
190	s->sample_size[0] = av_get_bits_per_sample_fmt(s->sample_fmt[0]) >> 3;	190	s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
191	s->sample_size[1] = av_get_bits_per_sample_fmt(s->sample_fmt[1]) >> 3;	191	s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
192		192
193	if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {	193	if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
194	if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,	194	if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,

libavcodec/utils.c+1−1

@@ -1131,7 +1131,7 @@ int av_get_bits_per_sample(enum CodecID codec_id){ · 1130 unmodified lines
1131		1131
1132	#if FF_API_OLD_SAMPLE_FMT	1132	#if FF_API_OLD_SAMPLE_FMT
1133	int av_get_bits_per_sample_format(enum AVSampleFormat sample_fmt) {	1133	int av_get_bits_per_sample_format(enum AVSampleFormat sample_fmt) {
1134	return av_get_bits_per_sample_fmt(sample_fmt);	1134	return av_get_bytes_per_sample(sample_fmt) << 3;
1135	}	1135	}
1136	#endif	1136	#endif
1137		1137

libavcodec/vmdav.c+1−1

@@ -447,7 +447,7 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx) · 446 unmodified lines
447	avctx->sample_fmt = AV_SAMPLE_FMT_S16;	447	avctx->sample_fmt = AV_SAMPLE_FMT_S16;
448	else	448	else
449	avctx->sample_fmt = AV_SAMPLE_FMT_U8;	449	avctx->sample_fmt = AV_SAMPLE_FMT_U8;
450	s->out_bps = av_get_bits_per_sample_fmt(avctx->sample_fmt) >> 3;	450	s->out_bps = av_get_bytes_per_sample(avctx->sample_fmt);
451		451
452	av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "	452	av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
453	"block align = %d, sample rate = %d\n",	453	"block align = %d, sample rate = %d\n",

libavcodec/vorbisdec.c+1−1

@@ -1646,7 +1646,7 @@ static int vorbis_decode_frame(AVCodecContext *avccontext, · 1645 unmodified lines
1646	vc->audio_channels);	1646	vc->audio_channels);
1647		1647
1648	data_size = len vc->audio_channels *	1648	data_size = len vc->audio_channels *
1649	(av_get_bits_per_sample_fmt(avccontext->sample_fmt) / 8);	1649	av_get_bytes_per_sample(avccontext->sample_fmt);
1650		1650
1651	return buf_size ;	1651	return buf_size ;
1652	}	1652	}

libavfilter/defaults.c+1−1

@@ -84,7 +84,7 @@ AVFilterBufferRef avfilter_default_get_audio_buffer(AVFilterLink link, int per · 83 unmodified lines
84	samples->refcount = 1;	84	samples->refcount = 1;
85	samples->free = ff_avfilter_default_free_buffer;	85	samples->free = ff_avfilter_default_free_buffer;
86		86
87	sample_size = av_get_bits_per_sample_fmt(sample_fmt) >>3;	87	sample_size = av_get_bytes_per_sample(sample_fmt);
88	chans_nb = av_get_channel_layout_nb_channels(channel_layout);	88	chans_nb = av_get_channel_layout_nb_channels(channel_layout);
89		89
90	per_channel_size = size/chans_nb;	90	per_channel_size = size/chans_nb;

libavformat/matroskaenc.c+1−1

@@ -527,7 +527,7 @@ static int mkv_write_tracks(AVFormatContext *s) · 526 unmodified lines
527	AVDictionaryEntry *tag;	527	AVDictionaryEntry *tag;
528		528
529	if (!bit_depth)	529	if (!bit_depth)
530	bit_depth = av_get_bits_per_sample_fmt(codec->sample_fmt);	530	bit_depth = av_get_bytes_per_sample(codec->sample_fmt) << 3;
531		531
532	if (codec->codec_id == CODEC_ID_AAC)	532	if (codec->codec_id == CODEC_ID_AAC)
533	get_aac_sample_rates(s, codec, &sample_rate, &output_sample_rate);	533	get_aac_sample_rates(s, codec, &sample_rate, &output_sample_rate);