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FFmpeg
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Remove usage of deprecated libavcodec/audioconvert.h functions. 

Originally committed as revision 25668 to svn://svn.ffmpeg.org/ffmpeg/trunk
tags/n0.8
Stefano Sabatini 14 years ago
parent
caa7ad5dc6
commit
ba7d6e798e
7 changed files with 27 additions and 20 deletions
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  1. +9
    -8
      ffmpeg.c
  2. +4
    -3
      ffplay.c
  3. +3
    -2
      libavcodec/alsdec.c
  4. +5
    -4
      libavcodec/resample.c
  5. +2
    -1
      libavcodec/utils.c
  6. +2
    -1
      libavfilter/defaults.c
  7. +2
    -1
      libavformat/matroskaenc.c

+ 9
- 8
ffmpeg.c View File

@@ -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);
}
}


+ 4
- 3
ffplay.c View File

@@ -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) {


+ 3
- 2
libavcodec/alsdec.c View File

@@ -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);


+ 5
- 4
libavcodec/resample.c View File

@@ -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;


+ 2
- 1
libavcodec/utils.c View File

@@ -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:


+ 2
- 1
libavfilter/defaults.c View File

@@ -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;


+ 2
- 1
libavformat/matroskaenc.c View File

@@ -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);


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