Mercurial > libavcodec.hg
annotate audioconvert.c @ 7497:9dbaea2302fd libavcodec
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author | michael |
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date | Tue, 05 Aug 2008 11:34:46 +0000 |
parents | 283eeda62184 |
children | 1302ec81afc0 |
rev | line source |
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3594 | 1 /* |
3596 | 2 * audio conversion |
3594 | 3 * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at> |
4 * | |
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5 * This file is part of FFmpeg. |
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
3594 | 8 * modify it under the terms of the GNU Lesser General Public |
9 * License as published by the Free Software Foundation; either | |
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10 * version 2.1 of the License, or (at your option) any later version. |
3594 | 11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
3594 | 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
15 * Lesser General Public License for more details. | |
16 * | |
17 * You should have received a copy of the GNU Lesser General Public | |
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18 * License along with FFmpeg; if not, write to the Free Software |
3594 | 19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
20 */ | |
21 | |
22 /** | |
23 * @file audioconvert.c | |
3596 | 24 * audio conversion |
3594 | 25 * @author Michael Niedermayer <michaelni@gmx.at> |
26 */ | |
27 | |
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28 #include "avcodec.h" |
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29 #include "audioconvert.h" |
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30 |
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31 typedef struct SampleFmtInfo { |
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32 const char *name; |
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33 int bits; |
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34 } SampleFmtInfo; |
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35 |
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36 /** this table gives more information about formats */ |
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37 static const SampleFmtInfo sample_fmt_info[SAMPLE_FMT_NB] = { |
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38 [SAMPLE_FMT_U8] = { .name = "u8", .bits = 8 }, |
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39 [SAMPLE_FMT_S16] = { .name = "s16", .bits = 16 }, |
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40 [SAMPLE_FMT_S24] = { .name = "s24", .bits = 24 }, |
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41 [SAMPLE_FMT_S32] = { .name = "s32", .bits = 32 }, |
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42 [SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32 } |
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43 }; |
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44 |
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45 const char *avcodec_get_sample_fmt_name(int sample_fmt) |
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46 { |
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47 if (sample_fmt < 0 || sample_fmt >= SAMPLE_FMT_NB) |
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48 return NULL; |
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49 return sample_fmt_info[sample_fmt].name; |
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50 } |
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51 |
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52 enum SampleFormat avcodec_get_sample_fmt(const char* name) |
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53 { |
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54 int i; |
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55 |
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56 for (i=0; i < SAMPLE_FMT_NB; i++) |
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57 if (!strcmp(sample_fmt_info[i].name, name)) |
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58 return i; |
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59 return SAMPLE_FMT_NONE; |
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60 } |
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61 |
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62 void avcodec_sample_fmt_string (char *buf, int buf_size, int sample_fmt) |
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63 { |
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64 /* print header */ |
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65 if (sample_fmt < 0) |
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66 snprintf (buf, buf_size, "name " " depth"); |
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67 else if (sample_fmt < SAMPLE_FMT_NB) { |
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68 SampleFmtInfo info= sample_fmt_info[sample_fmt]; |
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69 snprintf (buf, buf_size, "%-6s" " %2d ", info.name, info.bits); |
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70 } |
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71 } |
3594 | 72 |
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73 struct AVAudioConvert { |
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74 int in_channels, out_channels; |
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75 int fmt_pair; |
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76 }; |
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77 |
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78 AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels, |
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79 enum SampleFormat in_fmt, int in_channels, |
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80 const const float *matrix, int flags) |
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81 { |
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82 AVAudioConvert *ctx; |
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83 if (in_channels!=out_channels) |
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84 return NULL; /* FIXME: not supported */ |
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85 ctx = av_malloc(sizeof(AVAudioConvert)); |
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86 if (!ctx) |
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87 return NULL; |
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88 ctx->in_channels = in_channels; |
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89 ctx->out_channels = out_channels; |
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90 ctx->fmt_pair = out_fmt + SAMPLE_FMT_NB*in_fmt; |
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91 return ctx; |
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92 } |
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93 |
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94 void av_audio_convert_free(AVAudioConvert *ctx) |
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95 { |
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96 av_free(ctx); |
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97 } |
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98 |
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99 int av_audio_convert(AVAudioConvert *ctx, |
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100 void * const out[6], const int out_stride[6], |
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101 const void * const in[6], const int in_stride[6], int len) |
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102 { |
3594 | 103 int ch; |
104 | |
105 //FIXME optimize common cases | |
106 | |
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107 for(ch=0; ch<ctx->out_channels; ch++){ |
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108 const int is= in_stride[ch]; |
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109 const int os= out_stride[ch]; |
3594 | 110 uint8_t *pi= in[ch]; |
111 uint8_t *po= out[ch]; | |
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112 uint8_t *end= po + os*len; |
3594 | 113 if(!out[ch]) |
114 continue; | |
115 | |
116 #define CONV(ofmt, otype, ifmt, expr)\ | |
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117 if(ctx->fmt_pair == ofmt + SAMPLE_FMT_NB*ifmt){\ |
3594 | 118 do{\ |
119 *(otype*)po = expr; pi += is; po += os;\ | |
120 }while(po < end);\ | |
121 } | |
122 | |
5127 | 123 //FIXME put things below under ifdefs so we do not waste space for cases no codec will need |
124 //FIXME rounding and clipping ? | |
3594 | 125 |
126 CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_U8 , *(uint8_t*)pi) | |
127 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_U8 , (*(uint8_t*)pi - 0x80)<<8) | |
128 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_U8 , (*(uint8_t*)pi - 0x80)<<24) | |
129 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_U8 , (*(uint8_t*)pi - 0x80)*(1.0 / (1<<7))) | |
130 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S16, (*(int16_t*)pi>>8) + 0x80) | |
131 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S16, *(int16_t*)pi) | |
132 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S16, *(int16_t*)pi<<16) | |
133 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S16, *(int16_t*)pi*(1.0 / (1<<15))) | |
134 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S32, (*(int32_t*)pi>>24) + 0x80) | |
135 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S32, *(int32_t*)pi>>16) | |
136 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S32, *(int32_t*)pi) | |
137 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S32, *(int32_t*)pi*(1.0 / (1<<31))) | |
138 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_FLT, lrintf(*(float*)pi * (1<<7)) + 0x80) | |
139 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_FLT, lrintf(*(float*)pi * (1<<15))) | |
140 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_FLT, lrintf(*(float*)pi * (1<<31))) | |
141 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_FLT, *(float*)pi) | |
142 else return -1; | |
143 } | |
144 return 0; | |
145 } |