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https://github.com/FFmpeg/FFmpeg.git
synced 2025-01-08 13:22:53 +02:00
Remove usage of deprecated libavcodec/audioconvert.h functions.
Originally committed as revision 25668 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
parent
caa7ad5dc6
commit
ba7d6e798e
17
ffmpeg.c
17
ffmpeg.c
@ -37,6 +37,7 @@
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#include "libavcodec/opt.h"
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#include "libavcodec/audioconvert.h"
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#include "libavcore/parseutils.h"
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#include "libavcore/samplefmt.h"
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#include "libavutil/colorspace.h"
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#include "libavutil/fifo.h"
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#include "libavutil/intreadwrite.h"
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@ -769,8 +770,8 @@ static void do_audio_out(AVFormatContext *s,
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int size_out, frame_bytes, ret;
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AVCodecContext *enc= ost->st->codec;
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AVCodecContext *dec= ist->st->codec;
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int osize= av_get_bits_per_sample_format(enc->sample_fmt)/8;
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int isize= av_get_bits_per_sample_format(dec->sample_fmt)/8;
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int osize= av_get_bits_per_sample_fmt(enc->sample_fmt)/8;
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int isize= av_get_bits_per_sample_fmt(dec->sample_fmt)/8;
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const int coded_bps = av_get_bits_per_sample(enc->codec->id);
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need_realloc:
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@ -824,8 +825,8 @@ need_realloc:
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dec->sample_fmt, 1, NULL, 0);
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if (!ost->reformat_ctx) {
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fprintf(stderr, "Cannot convert %s sample format to %s sample format\n",
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avcodec_get_sample_fmt_name(dec->sample_fmt),
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avcodec_get_sample_fmt_name(enc->sample_fmt));
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av_get_sample_fmt_name(dec->sample_fmt),
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av_get_sample_fmt_name(enc->sample_fmt));
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ffmpeg_exit(1);
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}
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ost->reformat_pair=MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt);
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@ -1443,7 +1444,7 @@ static int output_packet(AVInputStream *ist, int ist_index,
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#endif
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AVPacket avpkt;
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int bps = av_get_bits_per_sample_format(ist->st->codec->sample_fmt)>>3;
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int bps = av_get_bits_per_sample_fmt(ist->st->codec->sample_fmt)>>3;
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if(ist->next_pts == AV_NOPTS_VALUE)
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ist->next_pts= ist->pts;
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@ -1760,7 +1761,7 @@ static int output_packet(AVInputStream *ist, int ist_index,
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ret = 0;
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/* encode any samples remaining in fifo */
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if (fifo_bytes > 0) {
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int osize = av_get_bits_per_sample_format(enc->sample_fmt) >> 3;
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int osize = av_get_bits_per_sample_fmt(enc->sample_fmt) >> 3;
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int fs_tmp = enc->frame_size;
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av_fifo_generic_read(ost->fifo, audio_buf, fifo_bytes, NULL);
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@ -2817,9 +2818,9 @@ static int opt_thread_count(const char *opt, const char *arg)
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static void opt_audio_sample_fmt(const char *arg)
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{
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if (strcmp(arg, "list"))
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audio_sample_fmt = avcodec_get_sample_fmt(arg);
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audio_sample_fmt = av_get_sample_fmt(arg);
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else {
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list_fmts(avcodec_sample_fmt_string, SAMPLE_FMT_NB);
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list_fmts(av_get_sample_fmt_string, SAMPLE_FMT_NB);
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ffmpeg_exit(0);
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}
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}
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7
ffplay.c
7
ffplay.c
@ -30,6 +30,7 @@
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#include "libavutil/pixdesc.h"
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#include "libavcore/imgutils.h"
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#include "libavcore/parseutils.h"
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#include "libavcore/samplefmt.h"
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#include "libavformat/avformat.h"
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#include "libavdevice/avdevice.h"
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#include "libswscale/swscale.h"
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@ -2099,8 +2100,8 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
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dec->sample_fmt, 1, NULL, 0);
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if (!is->reformat_ctx) {
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fprintf(stderr, "Cannot convert %s sample format to %s sample format\n",
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avcodec_get_sample_fmt_name(dec->sample_fmt),
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avcodec_get_sample_fmt_name(SAMPLE_FMT_S16));
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av_get_sample_fmt_name(dec->sample_fmt),
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av_get_sample_fmt_name(SAMPLE_FMT_S16));
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break;
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}
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is->audio_src_fmt= dec->sample_fmt;
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@ -2109,7 +2110,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
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if (is->reformat_ctx) {
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const void *ibuf[6]= {is->audio_buf1};
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void *obuf[6]= {is->audio_buf2};
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int istride[6]= {av_get_bits_per_sample_format(dec->sample_fmt)/8};
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int istride[6]= {av_get_bits_per_sample_fmt(dec->sample_fmt)/8};
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int ostride[6]= {2};
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int len= data_size/istride[0];
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if (av_audio_convert(is->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) {
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@ -36,6 +36,7 @@
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#include "bytestream.h"
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#include "bgmc.h"
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#include "dsputil.h"
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#include "libavcore/samplefmt.h"
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#include "libavutil/crc.h"
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#include <stdint.h>
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@ -1426,7 +1427,7 @@ static int decode_frame(AVCodecContext *avctx,
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// check for size of decoded data
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size = ctx->cur_frame_length * avctx->channels *
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(av_get_bits_per_sample_format(avctx->sample_fmt) >> 3);
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(av_get_bits_per_sample_fmt(avctx->sample_fmt) >> 3);
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if (size > *data_size) {
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av_log(avctx, AV_LOG_ERROR, "Decoded data exceeds buffer size.\n");
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@ -1679,7 +1680,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
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ctx->crc_buffer = av_malloc(sizeof(*ctx->crc_buffer) *
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ctx->cur_frame_length *
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avctx->channels *
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(av_get_bits_per_sample_format(avctx->sample_fmt) >> 3));
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(av_get_bits_per_sample_fmt(avctx->sample_fmt) >> 3));
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if (!ctx->crc_buffer) {
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av_log(avctx, AV_LOG_ERROR, "Allocating buffer memory failed.\n");
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decode_end(avctx);
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@ -27,6 +27,7 @@
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#include "avcodec.h"
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#include "audioconvert.h"
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#include "libavutil/opt.h"
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#include "libavcore/samplefmt.h"
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struct AVResampleContext;
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@ -174,15 +175,15 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
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s->sample_fmt [0] = sample_fmt_in;
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s->sample_fmt [1] = sample_fmt_out;
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s->sample_size[0] = av_get_bits_per_sample_format(s->sample_fmt[0])>>3;
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s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3;
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s->sample_size[0] = av_get_bits_per_sample_fmt(s->sample_fmt[0])>>3;
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s->sample_size[1] = av_get_bits_per_sample_fmt(s->sample_fmt[1])>>3;
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if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
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if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
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s->sample_fmt[0], 1, NULL, 0))) {
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av_log(s, AV_LOG_ERROR,
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"Cannot convert %s sample format to s16 sample format\n",
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avcodec_get_sample_fmt_name(s->sample_fmt[0]));
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av_get_sample_fmt_name(s->sample_fmt[0]));
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av_free(s);
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return NULL;
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}
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@ -193,7 +194,7 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
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SAMPLE_FMT_S16, 1, NULL, 0))) {
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av_log(s, AV_LOG_ERROR,
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"Cannot convert s16 sample format to %s sample format\n",
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avcodec_get_sample_fmt_name(s->sample_fmt[1]));
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av_get_sample_fmt_name(s->sample_fmt[1]));
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av_audio_convert_free(s->convert_ctx[0]);
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av_free(s);
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return NULL;
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@ -30,6 +30,7 @@
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#include "libavutil/crc.h"
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#include "libavutil/pixdesc.h"
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#include "libavcore/imgutils.h"
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#include "libavcore/samplefmt.h"
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#include "avcodec.h"
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#include "dsputil.h"
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#include "libavutil/opt.h"
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@ -923,7 +924,7 @@ void avcodec_string(char *buf, int buf_size, AVCodecContext *enc, int encode)
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avcodec_get_channel_layout_string(buf + strlen(buf), buf_size - strlen(buf), enc->channels, enc->channel_layout);
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if (enc->sample_fmt != SAMPLE_FMT_NONE) {
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snprintf(buf + strlen(buf), buf_size - strlen(buf),
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", %s", avcodec_get_sample_fmt_name(enc->sample_fmt));
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", %s", av_get_sample_fmt_name(enc->sample_fmt));
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}
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break;
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case AVMEDIA_TYPE_DATA:
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@ -20,6 +20,7 @@
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*/
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#include "libavcore/imgutils.h"
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#include "libavcore/samplefmt.h"
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#include "libavcodec/audioconvert.h"
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#include "avfilter.h"
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@ -109,7 +110,7 @@ AVFilterBufferRef *avfilter_default_get_audio_buffer(AVFilterLink *link, int per
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samples->refcount = 1;
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samples->free = avfilter_default_free_buffer;
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sample_size = av_get_bits_per_sample_format(sample_fmt) >>3;
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sample_size = av_get_bits_per_sample_fmt(sample_fmt) >>3;
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chans_nb = avcodec_channel_layout_num_channels(channel_layout);
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per_channel_size = size/chans_nb;
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@ -26,6 +26,7 @@
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#include "avc.h"
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#include "flacenc.h"
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#include "avlanguage.h"
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#include "libavcore/samplefmt.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/random_seed.h"
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#include "libavutil/lfg.h"
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@ -540,7 +541,7 @@ static int mkv_write_tracks(AVFormatContext *s)
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AVMetadataTag *tag;
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if (!bit_depth)
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bit_depth = av_get_bits_per_sample_format(codec->sample_fmt);
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bit_depth = av_get_bits_per_sample_fmt(codec->sample_fmt);
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if (codec->codec_id == CODEC_ID_AAC)
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get_aac_sample_rates(s, codec, &sample_rate, &output_sample_rate);
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