Mercurial > libavcodec.hg
view audioconvert.c @ 7774:a570a1e80400 libavcodec
Prevent a division by 0 in the g726 decoder when the configured samplerate is 0.
patch by Laurent Aimar, fenrir via.ecp fr
author | diego |
---|---|
date | Tue, 02 Sep 2008 23:09:14 +0000 |
parents | 1302ec81afc0 |
children | 8d5e06d2eed8 |
line wrap: on
line source
/* * 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 audioconvert.c * audio conversion * @author Michael Niedermayer <michaelni@gmx.at> */ #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_S24] = { .name = "s24", .bits = 24 }, [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); } } 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 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]; 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 , *(uint8_t*)pi) else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_U8 , (*(uint8_t*)pi - 0x80)<<8) else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_U8 , (*(uint8_t*)pi - 0x80)<<24) else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_U8 , (*(uint8_t*)pi - 0x80)*(1.0 / (1<<7))) else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_U8 , (*(uint8_t*)pi - 0x80)*(1.0 / (1<<7))) else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S16, (*(int16_t*)pi>>8) + 0x80) else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S16, *(int16_t*)pi) else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S16, *(int16_t*)pi<<16) else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S16, *(int16_t*)pi*(1.0 / (1<<15))) else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S16, *(int16_t*)pi*(1.0 / (1<<15))) else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S32, (*(int32_t*)pi>>24) + 0x80) else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S32, *(int32_t*)pi>>16) else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S32, *(int32_t*)pi) else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S32, *(int32_t*)pi*(1.0 / (1<<31))) else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S32, *(int32_t*)pi*(1.0 / (1<<31))) else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_FLT, lrintf(*(float*)pi * (1<<7)) + 0x80) else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_FLT, lrintf(*(float*)pi * (1<<15))) else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_FLT, lrintf(*(float*)pi * (1<<31))) else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_FLT, *(float*)pi) else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_FLT, *(float*)pi) else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_DBL, lrint(*(double*)pi * (1<<7)) + 0x80) else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_DBL, lrint(*(double*)pi * (1<<15))) else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_DBL, lrint(*(double*)pi * (1<<31))) else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_DBL, *(double*)pi) else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_DBL, *(double*)pi) else return -1; } return 0; }