Mercurial > libavcodec.hg
view audioconvert.c @ 8319:3e7dd8cfa97f libavcodec
undoing revision 16118 for truemotion1.c; commit was premature
author | melanson |
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date | Sun, 14 Dec 2008 03:41:11 +0000 |
parents | ea2c20535551 |
children | ffb82d54ecdc |
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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 audioconvert.c * audio conversion * @author Michael Niedermayer <michaelni@gmx.at> */ #include "avcodec.h" #include "audioconvert.h" #include <libavutil/avstring.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) { snprintf(buf, buf_size, channel_layout_map[i].name); 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]; 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; }