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FFmpeg/libavcodec/audioconvert.c
Patrik 862c1d2f05 Replace snprintf() with av_strlcpy(). 
Fixes the following warning on some systems:
audioconvert.c:131: warning: format not a string literal and no format
arguments

Patch by Patrik patrik A yes D nu

Originally committed as revision 17110 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-02-09 23:11:26 +00:00

235 lines
8.4 KiB
C
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/*
* audio conversion
* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file libavcodec/audioconvert.c
* audio conversion
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "libavutil/avstring.h"
#include "avcodec.h"
#include "audioconvert.h"
typedef struct SampleFmtInfo {
const char *name;
int bits;
} SampleFmtInfo;
/** this table gives more information about formats */
static const SampleFmtInfo sample_fmt_info[SAMPLE_FMT_NB] = {
[SAMPLE_FMT_U8] = { .name = "u8", .bits = 8 },
[SAMPLE_FMT_S16] = { .name = "s16", .bits = 16 },
[SAMPLE_FMT_S32] = { .name = "s32", .bits = 32 },
[SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32 },
[SAMPLE_FMT_DBL] = { .name = "dbl", .bits = 64 },
};
const char *avcodec_get_sample_fmt_name(int sample_fmt)
{
if (sample_fmt < 0 || sample_fmt >= SAMPLE_FMT_NB)
return NULL;
return sample_fmt_info[sample_fmt].name;
}
enum SampleFormat avcodec_get_sample_fmt(const char* name)
{
int i;
for (i=0; i < SAMPLE_FMT_NB; i++)
if (!strcmp(sample_fmt_info[i].name, name))
return i;
return SAMPLE_FMT_NONE;
}
void avcodec_sample_fmt_string (char *buf, int buf_size, int sample_fmt)
{
/* print header */
if (sample_fmt < 0)
snprintf (buf, buf_size, "name " " depth");
else if (sample_fmt < SAMPLE_FMT_NB) {
SampleFmtInfo info= sample_fmt_info[sample_fmt];
snprintf (buf, buf_size, "%-6s" " %2d ", info.name, info.bits);
}
}
static const char* const channel_names[]={
"FL", "FR", "FC", "LFE", "BL", "BR", "FLC", "FRC",
"BC", "SL", "SR", "TC", "TFL", "TFC", "TFR", "TBL",
"TBC", "TBR",
[29] = "DL",
[30] = "DR",
};
const char *get_channel_name(int channel_id)
{
if (channel_id<0 || channel_id>=FF_ARRAY_ELEMS(channel_names))
return NULL;
return channel_names[channel_id];
}
int64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name)
{
switch(nb_channels) {
case 1: return CH_LAYOUT_MONO;
case 2: return CH_LAYOUT_STEREO;
case 3: return CH_LAYOUT_SURROUND;
case 4: return CH_LAYOUT_QUAD;
case 5: return CH_LAYOUT_5POINT0;
case 6: return CH_LAYOUT_5POINT1;
case 8: return CH_LAYOUT_7POINT1;
default: return 0;
}
}
static const struct {
const char *name;
int nb_channels;
int64_t layout;
} channel_layout_map[] = {
{ "mono", 1, CH_LAYOUT_MONO },
{ "stereo", 2, CH_LAYOUT_STEREO },
{ "surround", 3, CH_LAYOUT_SURROUND },
{ "quad", 4, CH_LAYOUT_QUAD },
{ "5.0", 5, CH_LAYOUT_5POINT0 },
{ "5.1", 6, CH_LAYOUT_5POINT1 },
{ "5.1+downmix", 8, CH_LAYOUT_5POINT1|CH_LAYOUT_STEREO_DOWNMIX, },
{ "7.1", 8, CH_LAYOUT_7POINT1 },
{ "7.1(wide)", 8, CH_LAYOUT_7POINT1_WIDE },
{ "7.1+downmix", 10, CH_LAYOUT_7POINT1|CH_LAYOUT_STEREO_DOWNMIX, },
{ 0 }
};
void avcodec_get_channel_layout_string(char *buf, int buf_size, int nb_channels, int64_t channel_layout)
{
int i;
if (channel_layout==0)
channel_layout = avcodec_guess_channel_layout(nb_channels, CODEC_ID_NONE, NULL);
for (i=0; channel_layout_map[i].name; i++)
if (nb_channels == channel_layout_map[i].nb_channels &&
channel_layout == channel_layout_map[i].layout) {
av_strlcpy(buf, channel_layout_map[i].name, buf_size);
return;
}
snprintf(buf, buf_size, "%d channels", nb_channels);
if (channel_layout) {
int i,ch;
av_strlcat(buf, " (", buf_size);
for(i=0,ch=0; i<64; i++) {
if ((channel_layout & (1L<<i))) {
const char *name = get_channel_name(i);
if (name) {
if (ch>0) av_strlcat(buf, "|", buf_size);
av_strlcat(buf, name, buf_size);
}
ch++;
}
}
av_strlcat(buf, ")", buf_size);
}
}
struct AVAudioConvert {
int in_channels, out_channels;
int fmt_pair;
};
AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
enum SampleFormat in_fmt, int in_channels,
const float *matrix, int flags)
{
AVAudioConvert *ctx;
if (in_channels!=out_channels)
return NULL; /* FIXME: not supported */
ctx = av_malloc(sizeof(AVAudioConvert));
if (!ctx)
return NULL;
ctx->in_channels = in_channels;
ctx->out_channels = out_channels;
ctx->fmt_pair = out_fmt + SAMPLE_FMT_NB*in_fmt;
return ctx;
}
void av_audio_convert_free(AVAudioConvert *ctx)
{
av_free(ctx);
}
int av_audio_convert(AVAudioConvert *ctx,
void * const out[6], const int out_stride[6],
const void * const in[6], const int in_stride[6], int len)
{
int ch;
//FIXME optimize common cases
for(ch=0; ch<ctx->out_channels; ch++){
const int is= in_stride[ch];
const int os= out_stride[ch];
const uint8_t *pi= in[ch];
uint8_t *po= out[ch];
uint8_t *end= po + os*len;
if(!out[ch])
continue;
#define CONV(ofmt, otype, ifmt, expr)\
if(ctx->fmt_pair == ofmt + SAMPLE_FMT_NB*ifmt){\
do{\
*(otype*)po = expr; pi += is; po += os;\
}while(po < end);\
}
//FIXME put things below under ifdefs so we do not waste space for cases no codec will need
//FIXME rounding and clipping ?
CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_U8 , *(const uint8_t*)pi)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8)
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24)
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S16, *(const int16_t*)pi)
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S16, *(const int16_t*)pi<<16)
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S32, *(const int32_t*)pi>>16)
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S32, *(const int32_t*)pi)
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31)))
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_FLT, lrintf(*(const float*)pi * (1<<7)) + 0x80)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_FLT, lrintf(*(const float*)pi * (1<<15)))
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_FLT, lrintf(*(const float*)pi * (1<<31)))
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_DBL, lrint(*(const double*)pi * (1<<7)) + 0x80)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_DBL, lrint(*(const double*)pi * (1<<15)))
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_DBL, lrint(*(const double*)pi * (1<<31)))
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_DBL, *(const double*)pi)
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_DBL, *(const double*)pi)
else return -1;
}
return 0;
}
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