[SCM] libav/experimental: Remove usage of deprecated libavcodec/audioconvert.h functions.
siretart at users.alioth.debian.org
siretart at users.alioth.debian.org
Sun Jun 30 17:18:36 UTC 2013
The following commit has been merged in the experimental branch:
commit ba7d6e798eb11b463852479a5e09a68926e82af6
Author: Stefano Sabatini <stefano.sabatini-lala at poste.it>
Date: Wed Nov 3 20:19:34 2010 +0000
Remove usage of deprecated libavcodec/audioconvert.h functions.
Originally committed as revision 25668 to svn://svn.ffmpeg.org/ffmpeg/trunk
diff --git a/ffmpeg.c b/ffmpeg.c
index 4d1b754..ad3f670 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -37,6 +37,7 @@
#include "libavcodec/opt.h"
#include "libavcodec/audioconvert.h"
#include "libavcore/parseutils.h"
+#include "libavcore/samplefmt.h"
#include "libavutil/colorspace.h"
#include "libavutil/fifo.h"
#include "libavutil/intreadwrite.h"
@@ -769,8 +770,8 @@ static void do_audio_out(AVFormatContext *s,
int size_out, frame_bytes, ret;
AVCodecContext *enc= ost->st->codec;
AVCodecContext *dec= ist->st->codec;
- int osize= av_get_bits_per_sample_format(enc->sample_fmt)/8;
- int isize= av_get_bits_per_sample_format(dec->sample_fmt)/8;
+ int osize= av_get_bits_per_sample_fmt(enc->sample_fmt)/8;
+ int isize= av_get_bits_per_sample_fmt(dec->sample_fmt)/8;
const int coded_bps = av_get_bits_per_sample(enc->codec->id);
need_realloc:
@@ -824,8 +825,8 @@ need_realloc:
dec->sample_fmt, 1, NULL, 0);
if (!ost->reformat_ctx) {
fprintf(stderr, "Cannot convert %s sample format to %s sample format\n",
- avcodec_get_sample_fmt_name(dec->sample_fmt),
- avcodec_get_sample_fmt_name(enc->sample_fmt));
+ av_get_sample_fmt_name(dec->sample_fmt),
+ av_get_sample_fmt_name(enc->sample_fmt));
ffmpeg_exit(1);
}
ost->reformat_pair=MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt);
@@ -1443,7 +1444,7 @@ static int output_packet(AVInputStream *ist, int ist_index,
#endif
AVPacket avpkt;
- int bps = av_get_bits_per_sample_format(ist->st->codec->sample_fmt)>>3;
+ int bps = av_get_bits_per_sample_fmt(ist->st->codec->sample_fmt)>>3;
if(ist->next_pts == AV_NOPTS_VALUE)
ist->next_pts= ist->pts;
@@ -1760,7 +1761,7 @@ static int output_packet(AVInputStream *ist, int ist_index,
ret = 0;
/* encode any samples remaining in fifo */
if (fifo_bytes > 0) {
- int osize = av_get_bits_per_sample_format(enc->sample_fmt) >> 3;
+ int osize = av_get_bits_per_sample_fmt(enc->sample_fmt) >> 3;
int fs_tmp = enc->frame_size;
av_fifo_generic_read(ost->fifo, audio_buf, fifo_bytes, NULL);
@@ -2817,9 +2818,9 @@ static int opt_thread_count(const char *opt, const char *arg)
static void opt_audio_sample_fmt(const char *arg)
{
if (strcmp(arg, "list"))
- audio_sample_fmt = avcodec_get_sample_fmt(arg);
+ audio_sample_fmt = av_get_sample_fmt(arg);
else {
- list_fmts(avcodec_sample_fmt_string, SAMPLE_FMT_NB);
+ list_fmts(av_get_sample_fmt_string, SAMPLE_FMT_NB);
ffmpeg_exit(0);
}
}
diff --git a/ffplay.c b/ffplay.c
index eecf16a..0563e96 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -30,6 +30,7 @@
#include "libavutil/pixdesc.h"
#include "libavcore/imgutils.h"
#include "libavcore/parseutils.h"
+#include "libavcore/samplefmt.h"
#include "libavformat/avformat.h"
#include "libavdevice/avdevice.h"
#include "libswscale/swscale.h"
@@ -2099,8 +2100,8 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
dec->sample_fmt, 1, NULL, 0);
if (!is->reformat_ctx) {
fprintf(stderr, "Cannot convert %s sample format to %s sample format\n",
- avcodec_get_sample_fmt_name(dec->sample_fmt),
- avcodec_get_sample_fmt_name(SAMPLE_FMT_S16));
+ av_get_sample_fmt_name(dec->sample_fmt),
+ av_get_sample_fmt_name(SAMPLE_FMT_S16));
break;
}
is->audio_src_fmt= dec->sample_fmt;
@@ -2109,7 +2110,7 @@ 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_format(dec->sample_fmt)/8};
+ int istride[6]= {av_get_bits_per_sample_fmt(dec->sample_fmt)/8};
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/alsdec.c b/libavcodec/alsdec.c
index 352506b..6993bb9 100644
--- a/libavcodec/alsdec.c
+++ b/libavcodec/alsdec.c
@@ -36,6 +36,7 @@
#include "bytestream.h"
#include "bgmc.h"
#include "dsputil.h"
+#include "libavcore/samplefmt.h"
#include "libavutil/crc.h"
#include <stdint.h>
@@ -1426,7 +1427,7 @@ static int decode_frame(AVCodecContext *avctx,
// check for size of decoded data
size = ctx->cur_frame_length * avctx->channels *
- (av_get_bits_per_sample_format(avctx->sample_fmt) >> 3);
+ (av_get_bits_per_sample_fmt(avctx->sample_fmt) >> 3);
if (size > *data_size) {
av_log(avctx, AV_LOG_ERROR, "Decoded data exceeds buffer size.\n");
@@ -1679,7 +1680,7 @@ 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_format(avctx->sample_fmt) >> 3));
+ (av_get_bits_per_sample_fmt(avctx->sample_fmt) >> 3));
if (!ctx->crc_buffer) {
av_log(avctx, AV_LOG_ERROR, "Allocating buffer memory failed.\n");
decode_end(avctx);
diff --git a/libavcodec/resample.c b/libavcodec/resample.c
index 222b28c..89e2d71 100644
--- a/libavcodec/resample.c
+++ b/libavcodec/resample.c
@@ -27,6 +27,7 @@
#include "avcodec.h"
#include "audioconvert.h"
#include "libavutil/opt.h"
+#include "libavcore/samplefmt.h"
struct AVResampleContext;
@@ -174,15 +175,15 @@ 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_format(s->sample_fmt[0])>>3;
- s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3;
+ 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;
if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
s->sample_fmt[0], 1, NULL, 0))) {
av_log(s, AV_LOG_ERROR,
"Cannot convert %s sample format to s16 sample format\n",
- avcodec_get_sample_fmt_name(s->sample_fmt[0]));
+ av_get_sample_fmt_name(s->sample_fmt[0]));
av_free(s);
return NULL;
}
@@ -193,7 +194,7 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
SAMPLE_FMT_S16, 1, NULL, 0))) {
av_log(s, AV_LOG_ERROR,
"Cannot convert s16 sample format to %s sample format\n",
- avcodec_get_sample_fmt_name(s->sample_fmt[1]));
+ av_get_sample_fmt_name(s->sample_fmt[1]));
av_audio_convert_free(s->convert_ctx[0]);
av_free(s);
return NULL;
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index 8169b37..4d13e0a 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -30,6 +30,7 @@
#include "libavutil/crc.h"
#include "libavutil/pixdesc.h"
#include "libavcore/imgutils.h"
+#include "libavcore/samplefmt.h"
#include "avcodec.h"
#include "dsputil.h"
#include "libavutil/opt.h"
@@ -923,7 +924,7 @@ void avcodec_string(char *buf, int buf_size, AVCodecContext *enc, int encode)
avcodec_get_channel_layout_string(buf + strlen(buf), buf_size - strlen(buf), enc->channels, enc->channel_layout);
if (enc->sample_fmt != SAMPLE_FMT_NONE) {
snprintf(buf + strlen(buf), buf_size - strlen(buf),
- ", %s", avcodec_get_sample_fmt_name(enc->sample_fmt));
+ ", %s", av_get_sample_fmt_name(enc->sample_fmt));
}
break;
case AVMEDIA_TYPE_DATA:
diff --git a/libavfilter/defaults.c b/libavfilter/defaults.c
index 1d748c2..5462b1a 100644
--- a/libavfilter/defaults.c
+++ b/libavfilter/defaults.c
@@ -20,6 +20,7 @@
*/
#include "libavcore/imgutils.h"
+#include "libavcore/samplefmt.h"
#include "libavcodec/audioconvert.h"
#include "avfilter.h"
@@ -109,7 +110,7 @@ AVFilterBufferRef *avfilter_default_get_audio_buffer(AVFilterLink *link, int per
samples->refcount = 1;
samples->free = avfilter_default_free_buffer;
- sample_size = av_get_bits_per_sample_format(sample_fmt) >>3;
+ sample_size = av_get_bits_per_sample_fmt(sample_fmt) >>3;
chans_nb = avcodec_channel_layout_num_channels(channel_layout);
per_channel_size = size/chans_nb;
diff --git a/libavformat/matroskaenc.c b/libavformat/matroskaenc.c
index 80bb08e..1430228 100644
--- a/libavformat/matroskaenc.c
+++ b/libavformat/matroskaenc.c
@@ -26,6 +26,7 @@
#include "avc.h"
#include "flacenc.h"
#include "avlanguage.h"
+#include "libavcore/samplefmt.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/random_seed.h"
#include "libavutil/lfg.h"
@@ -540,7 +541,7 @@ static int mkv_write_tracks(AVFormatContext *s)
AVMetadataTag *tag;
if (!bit_depth)
- bit_depth = av_get_bits_per_sample_format(codec->sample_fmt);
+ bit_depth = av_get_bits_per_sample_fmt(codec->sample_fmt);
if (codec->codec_id == CODEC_ID_AAC)
get_aac_sample_rates(s, codec, &sample_rate, &output_sample_rate);
--
Libav/FFmpeg packaging
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