Add sample format support functions: avcodec_get_sample_fmt_name(), avcodec_get_sample_fmt(), avcodec_sample_fmt_string()
Originally committed as revision 14484 to svn://svn.ffmpeg.org/ffmpeg/trunk
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@ -25,7 +25,49 @@
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* @author Michael Niedermayer <michaelni@gmx.at>
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* @author Michael Niedermayer <michaelni@gmx.at>
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*/
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*/
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#include "avcodec.h"
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#include "audioconvert.h"
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typedef struct SampleFmtInfo {
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const char *name;
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int bits;
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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[SAMPLE_FMT_NB] = {
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[SAMPLE_FMT_U8] = { .name = "u8", .bits = 8 },
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[SAMPLE_FMT_S16] = { .name = "s16", .bits = 16 },
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[SAMPLE_FMT_S24] = { .name = "s24", .bits = 24 },
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[SAMPLE_FMT_S32] = { .name = "s32", .bits = 32 },
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[SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32 }
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};
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const char *avcodec_get_sample_fmt_name(int sample_fmt)
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{
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if (sample_fmt < 0 || sample_fmt >= 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 SampleFormat avcodec_get_sample_fmt(const char* name)
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{
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int i;
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for (i=0; i < 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 SAMPLE_FMT_NONE;
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}
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void avcodec_sample_fmt_string (char *buf, int buf_size, int 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 < 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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}
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int av_audio_convert(void *maybe_dspcontext_or_something_av_convert_specific,
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int av_audio_convert(void *maybe_dspcontext_or_something_av_convert_specific,
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void *out[6], int out_stride[6], enum SampleFormat out_fmt,
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void *out[6], int out_stride[6], enum SampleFormat out_fmt,
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57
libavcodec/audioconvert.h
Normal file
57
libavcodec/audioconvert.h
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@ -0,0 +1,57 @@
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/*
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* audio conversion
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* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
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* Copyright (c) 2008 Peter Ross
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*
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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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#ifndef FFMPEG_AUDIOCONVERT_H
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#define FFMPEG_AUDIOCONVERT_H
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/**
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* @file audioconvert.h
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* Audio format conversion routines
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*/
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#include "avcodec.h"
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/**
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* Generate string corresponding to the sample format with
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* number sample_fmt, or a header if sample_fmt is negative.
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*
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* @param[in] buf the buffer where to write the string
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* @param[in] buf_size the size of buf
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* @param[in] sample_fmt the number of the sample format to print the corresponding info string, or
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* a negative value to print the corresponding header.
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* Meaningful values for obtaining a sample format info vary from 0 to SAMPLE_FMT_NB -1.
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*/
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void avcodec_sample_fmt_string(char *buf, int buf_size, int sample_fmt);
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/**
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* @return NULL on error
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*/
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const char *avcodec_get_sample_fmt_name(int sample_fmt);
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/**
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* @return SAMPLE_FMT_NONE on error
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*/
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enum SampleFormat avcodec_get_sample_fmt(const char* name);
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#endif /* FFMPEG_AUDIOCONVERT_H */
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