* qatar/master: (27 commits) libxvid: Give more suitable names to libxvid-related files. libxvid: Separate libxvid encoder from libxvid rate control code. jpeglsdec: Remove write-only variable in ff_jpegls_decode_lse(). fate: cosmetics: lowercase some comments fate: Give more consistent names to some RealVideo/RealAudio tests. lavfi: add avfilter_get_audio_buffer_ref_from_arrays(). lavfi: add extended_data to AVFilterBuffer. lavc: check that extended_data is properly set in avcodec_encode_audio2(). lavc: pad last audio frame with silence when needed. samplefmt: add a function for filling a buffer with silence. samplefmt: add a function for copying audio samples. lavr: do not try to copy to uninitialized output audio data. lavr: make avresample_read() with NULL output discard samples. fate: split idroq audio and video into separate tests fate: improve dependencies fate: add convenient shorthands for ea-vp6, libavcodec, libavutil tests fate: split some combined tests into separate audio and video tests fate: fix dependencies for probe tests mips: intreadwrite: fix inline asm for gcc 4.8 mips: intreadwrite: remove unnecessary inline asm ... Conflicts: cmdutils.h configure doc/APIchanges doc/filters.texi ffmpeg.c ffplay.c libavcodec/internal.h libavcodec/jpeglsdec.c libavcodec/libschroedingerdec.c libavcodec/libxvid.c libavcodec/libxvid_rc.c libavcodec/utils.c libavcodec/version.h libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/buffersink.h tests/Makefile tests/fate/aac.mak tests/fate/audio.mak tests/fate/demux.mak tests/fate/ea.mak tests/fate/image.mak tests/fate/libavutil.mak tests/fate/lossless-audio.mak tests/fate/lossless-video.mak tests/fate/microsoft.mak tests/fate/qt.mak tests/fate/real.mak tests/fate/screen.mak tests/fate/video.mak tests/fate/voice.mak tests/fate/vqf.mak tests/ref/fate/ea-mad tests/ref/fate/ea-tqi Merged-by: Michael Niedermayer <michaelni@gmx.at>
		
			
				
	
	
		
			235 lines
		
	
	
		
			8.0 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			235 lines
		
	
	
		
			8.0 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
/*
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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#include "samplefmt.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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typedef struct SampleFmtInfo {
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    char name[8];
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    int bits;
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    int planar;
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    enum AVSampleFormat altform; ///< planar<->packed alternative form
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} SampleFmtInfo;
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/** this table gives more information about formats */
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static const SampleFmtInfo sample_fmt_info[AV_SAMPLE_FMT_NB] = {
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    [AV_SAMPLE_FMT_U8]   = { .name =   "u8", .bits =  8, .planar = 0, .altform = AV_SAMPLE_FMT_U8P  },
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    [AV_SAMPLE_FMT_S16]  = { .name =  "s16", .bits = 16, .planar = 0, .altform = AV_SAMPLE_FMT_S16P },
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    [AV_SAMPLE_FMT_S32]  = { .name =  "s32", .bits = 32, .planar = 0, .altform = AV_SAMPLE_FMT_S32P },
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    [AV_SAMPLE_FMT_FLT]  = { .name =  "flt", .bits = 32, .planar = 0, .altform = AV_SAMPLE_FMT_FLTP },
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    [AV_SAMPLE_FMT_DBL]  = { .name =  "dbl", .bits = 64, .planar = 0, .altform = AV_SAMPLE_FMT_DBLP },
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    [AV_SAMPLE_FMT_U8P]  = { .name =  "u8p", .bits =  8, .planar = 1, .altform = AV_SAMPLE_FMT_U8   },
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    [AV_SAMPLE_FMT_S16P] = { .name = "s16p", .bits = 16, .planar = 1, .altform = AV_SAMPLE_FMT_S16  },
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    [AV_SAMPLE_FMT_S32P] = { .name = "s32p", .bits = 32, .planar = 1, .altform = AV_SAMPLE_FMT_S32  },
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    [AV_SAMPLE_FMT_FLTP] = { .name = "fltp", .bits = 32, .planar = 1, .altform = AV_SAMPLE_FMT_FLT  },
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    [AV_SAMPLE_FMT_DBLP] = { .name = "dblp", .bits = 64, .planar = 1, .altform = AV_SAMPLE_FMT_DBL  },
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};
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const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
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{
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    if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
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        return NULL;
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    return sample_fmt_info[sample_fmt].name;
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}
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enum AVSampleFormat av_get_sample_fmt(const char *name)
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{
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    int i;
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    for (i = 0; i < AV_SAMPLE_FMT_NB; i++)
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        if (!strcmp(sample_fmt_info[i].name, name))
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            return i;
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    return AV_SAMPLE_FMT_NONE;
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}
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enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar)
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{
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    if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
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        return AV_SAMPLE_FMT_NONE;
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    if (sample_fmt_info[sample_fmt].planar == planar)
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        return sample_fmt;
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    return sample_fmt_info[sample_fmt].altform;
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}
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enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)
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{
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    if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
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        return AV_SAMPLE_FMT_NONE;
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    if (sample_fmt_info[sample_fmt].planar)
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        return sample_fmt_info[sample_fmt].altform;
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    return sample_fmt;
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}
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enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
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{
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    if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
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        return AV_SAMPLE_FMT_NONE;
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    if (sample_fmt_info[sample_fmt].planar)
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        return sample_fmt;
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    return sample_fmt_info[sample_fmt].altform;
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}
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char *av_get_sample_fmt_string (char *buf, int buf_size, enum AVSampleFormat sample_fmt)
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{
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    /* print header */
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    if (sample_fmt < 0)
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        snprintf(buf, buf_size, "name  " " depth");
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    else if (sample_fmt < AV_SAMPLE_FMT_NB) {
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        SampleFmtInfo info = sample_fmt_info[sample_fmt];
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        snprintf (buf, buf_size, "%-6s" "   %2d ", info.name, info.bits);
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    }
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    return buf;
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}
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int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
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{
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     return sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB ?
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        0 : sample_fmt_info[sample_fmt].bits >> 3;
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}
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#if FF_API_GET_BITS_PER_SAMPLE_FMT
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int av_get_bits_per_sample_fmt(enum AVSampleFormat sample_fmt)
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{
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    return sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB ?
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        0 : sample_fmt_info[sample_fmt].bits;
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}
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#endif
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int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
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{
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     if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
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         return 0;
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     return sample_fmt_info[sample_fmt].planar;
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}
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int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
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                               enum AVSampleFormat sample_fmt, int align)
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{
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    int line_size;
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    int sample_size = av_get_bytes_per_sample(sample_fmt);
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    int planar      = av_sample_fmt_is_planar(sample_fmt);
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    /* validate parameter ranges */
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    if (!sample_size || nb_samples <= 0 || nb_channels <= 0)
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        return AVERROR(EINVAL);
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    /* auto-select alignment if not specified */
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    if (!align) {
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        align = 1;
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        nb_samples = FFALIGN(nb_samples, 32);
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    }
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    /* check for integer overflow */
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    if (nb_channels > INT_MAX / align ||
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        (int64_t)nb_channels * nb_samples > (INT_MAX - (align * nb_channels)) / sample_size)
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        return AVERROR(EINVAL);
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    line_size = planar ? FFALIGN(nb_samples * sample_size,               align) :
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                         FFALIGN(nb_samples * sample_size * nb_channels, align);
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    if (linesize)
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        *linesize = line_size;
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    return planar ? line_size * nb_channels : line_size;
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}
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int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
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                           const uint8_t *buf, int nb_channels, int nb_samples,
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                           enum AVSampleFormat sample_fmt, int align)
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{
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    int ch, planar, buf_size, line_size;
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    planar   = av_sample_fmt_is_planar(sample_fmt);
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    buf_size = av_samples_get_buffer_size(&line_size, nb_channels, nb_samples,
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                                          sample_fmt, align);
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    if (buf_size < 0)
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        return buf_size;
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    audio_data[0] = buf;
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    for (ch = 1; planar && ch < nb_channels; ch++)
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        audio_data[ch] = audio_data[ch-1] + line_size;
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    if (linesize)
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        *linesize = line_size;
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    return 0;
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}
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int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
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                     int nb_samples, enum AVSampleFormat sample_fmt, int align)
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{
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    uint8_t *buf;
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    int size = av_samples_get_buffer_size(NULL, nb_channels, nb_samples,
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                                          sample_fmt, align);
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    if (size < 0)
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        return size;
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    buf = av_mallocz(size);
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    if (!buf)
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        return AVERROR(ENOMEM);
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    size = av_samples_fill_arrays(audio_data, linesize, buf, nb_channels,
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                                  nb_samples, sample_fmt, align);
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    if (size < 0) {
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        av_free(buf);
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        return size;
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    }
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    return 0;
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}
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int av_samples_copy(uint8_t **dst, uint8_t * const *src, int dst_offset,
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                    int src_offset, int nb_samples, int nb_channels,
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                    enum AVSampleFormat sample_fmt)
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{
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    int planar      = av_sample_fmt_is_planar(sample_fmt);
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    int planes      = planar ? nb_channels : 1;
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    int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels);
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    int data_size   = nb_samples * block_align;
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    int i;
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    dst_offset *= block_align;
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    src_offset *= block_align;
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    for (i = 0; i < planes; i++)
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        memcpy(dst[i] + dst_offset, src[i] + src_offset, data_size);
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    return 0;
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}
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int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples,
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                           int nb_channels, enum AVSampleFormat sample_fmt)
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{
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    int planar      = av_sample_fmt_is_planar(sample_fmt);
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    int planes      = planar ? nb_channels : 1;
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    int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels);
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    int data_size   = nb_samples * block_align;
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    int fill_char   = (sample_fmt == AV_SAMPLE_FMT_U8 ||
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                     sample_fmt == AV_SAMPLE_FMT_U8P) ? 0x80 : 0x00;
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    int i;
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    offset *= block_align;
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    for (i = 0; i < planes; i++)
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        memset(audio_data[i] + offset, fill_char, data_size);
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    return 0;
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}
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