Mercurial > libavcodec.hg
annotate audioconvert.c @ 8991:ca768cb2bfb6 libavcodec
Use last decoded SPS as current SPS in order to parse picture timing SEI
correctly. This works around an apparent H.264 standard deficiency.
Patch by Ivan Schreter, schreter gmx net
author | cehoyos |
---|---|
date | Fri, 20 Feb 2009 16:20:01 +0000 |
parents | 7c47da8e2cf8 |
children | f7bd7e90ef9f |
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 /** | |
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23 * @file libavcodec/audioconvert.c |
3596 | 24 * audio conversion |
3594 | 25 * @author Michael Niedermayer <michaelni@gmx.at> |
26 */ | |
27 | |
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28 #include "libavutil/avstring.h" |
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29 #include "avcodec.h" |
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30 #include "audioconvert.h" |
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31 |
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32 typedef struct SampleFmtInfo { |
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33 const char *name; |
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34 int bits; |
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35 } SampleFmtInfo; |
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36 |
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37 /** this table gives more information about formats */ |
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38 static const SampleFmtInfo sample_fmt_info[SAMPLE_FMT_NB] = { |
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39 [SAMPLE_FMT_U8] = { .name = "u8", .bits = 8 }, |
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40 [SAMPLE_FMT_S16] = { .name = "s16", .bits = 16 }, |
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41 [SAMPLE_FMT_S32] = { .name = "s32", .bits = 32 }, |
7612 | 42 [SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32 }, |
43 [SAMPLE_FMT_DBL] = { .name = "dbl", .bits = 64 }, | |
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44 }; |
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45 |
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46 const char *avcodec_get_sample_fmt_name(int sample_fmt) |
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47 { |
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48 if (sample_fmt < 0 || sample_fmt >= SAMPLE_FMT_NB) |
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49 return NULL; |
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50 return sample_fmt_info[sample_fmt].name; |
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51 } |
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52 |
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53 enum SampleFormat avcodec_get_sample_fmt(const char* name) |
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54 { |
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55 int i; |
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56 |
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57 for (i=0; i < SAMPLE_FMT_NB; i++) |
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58 if (!strcmp(sample_fmt_info[i].name, name)) |
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59 return i; |
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60 return SAMPLE_FMT_NONE; |
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61 } |
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62 |
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63 void avcodec_sample_fmt_string (char *buf, int buf_size, int sample_fmt) |
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64 { |
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65 /* print header */ |
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66 if (sample_fmt < 0) |
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67 snprintf (buf, buf_size, "name " " depth"); |
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68 else if (sample_fmt < SAMPLE_FMT_NB) { |
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69 SampleFmtInfo info= sample_fmt_info[sample_fmt]; |
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70 snprintf (buf, buf_size, "%-6s" " %2d ", info.name, info.bits); |
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71 } |
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72 } |
3594 | 73 |
8098 | 74 static const char* const channel_names[]={ |
75 "FL", "FR", "FC", "LFE", "BL", "BR", "FLC", "FRC", | |
76 "BC", "SL", "SR", "TC", "TFL", "TFC", "TFR", "TBL", | |
77 "TBC", "TBR", | |
78 [29] = "DL", | |
79 [30] = "DR", | |
80 }; | |
81 | |
82 const char *get_channel_name(int channel_id) | |
83 { | |
84 if (channel_id<0 || channel_id>=FF_ARRAY_ELEMS(channel_names)) | |
85 return NULL; | |
86 return channel_names[channel_id]; | |
87 } | |
88 | |
89 int64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name) | |
90 { | |
91 switch(nb_channels) { | |
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92 case 1: return CH_LAYOUT_MONO; |
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93 case 2: return CH_LAYOUT_STEREO; |
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94 case 3: return CH_LAYOUT_SURROUND; |
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95 case 4: return CH_LAYOUT_QUAD; |
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96 case 5: return CH_LAYOUT_5POINT0; |
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97 case 6: return CH_LAYOUT_5POINT1; |
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98 case 8: return CH_LAYOUT_7POINT1; |
8098 | 99 default: return 0; |
100 } | |
101 } | |
102 | |
103 static const struct { | |
104 const char *name; | |
105 int nb_channels; | |
106 int64_t layout; | |
8155 | 107 } channel_layout_map[] = { |
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108 { "mono", 1, CH_LAYOUT_MONO }, |
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109 { "stereo", 2, CH_LAYOUT_STEREO }, |
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110 { "surround", 3, CH_LAYOUT_SURROUND }, |
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111 { "quad", 4, CH_LAYOUT_QUAD }, |
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112 { "5.0", 5, CH_LAYOUT_5POINT0 }, |
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113 { "5.1", 6, CH_LAYOUT_5POINT1 }, |
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114 { "5.1+downmix", 8, CH_LAYOUT_5POINT1|CH_LAYOUT_STEREO_DOWNMIX, }, |
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115 { "7.1", 8, CH_LAYOUT_7POINT1 }, |
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116 { "7.1(wide)", 8, CH_LAYOUT_7POINT1_WIDE }, |
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117 { "7.1+downmix", 10, CH_LAYOUT_7POINT1|CH_LAYOUT_STEREO_DOWNMIX, }, |
8098 | 118 { 0 } |
119 }; | |
120 | |
121 void avcodec_get_channel_layout_string(char *buf, int buf_size, int nb_channels, int64_t channel_layout) | |
122 { | |
123 int i; | |
124 | |
125 if (channel_layout==0) | |
126 channel_layout = avcodec_guess_channel_layout(nb_channels, CODEC_ID_NONE, NULL); | |
127 | |
128 for (i=0; channel_layout_map[i].name; i++) | |
129 if (nb_channels == channel_layout_map[i].nb_channels && | |
130 channel_layout == channel_layout_map[i].layout) { | |
8784 | 131 av_strlcpy(buf, channel_layout_map[i].name, buf_size); |
8098 | 132 return; |
133 } | |
134 | |
135 snprintf(buf, buf_size, "%d channels", nb_channels); | |
136 if (channel_layout) { | |
137 int i,ch; | |
138 av_strlcat(buf, " (", buf_size); | |
139 for(i=0,ch=0; i<64; i++) { | |
140 if ((channel_layout & (1L<<i))) { | |
141 const char *name = get_channel_name(i); | |
142 if (name) { | |
143 if (ch>0) av_strlcat(buf, "|", buf_size); | |
144 av_strlcat(buf, name, buf_size); | |
145 } | |
146 ch++; | |
147 } | |
148 } | |
149 av_strlcat(buf, ")", buf_size); | |
150 } | |
151 } | |
152 | |
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153 struct AVAudioConvert { |
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154 int in_channels, out_channels; |
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155 int fmt_pair; |
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156 }; |
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157 |
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158 AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels, |
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159 enum SampleFormat in_fmt, int in_channels, |
7984 | 160 const float *matrix, int flags) |
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161 { |
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162 AVAudioConvert *ctx; |
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163 if (in_channels!=out_channels) |
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164 return NULL; /* FIXME: not supported */ |
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165 ctx = av_malloc(sizeof(AVAudioConvert)); |
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166 if (!ctx) |
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167 return NULL; |
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168 ctx->in_channels = in_channels; |
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169 ctx->out_channels = out_channels; |
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170 ctx->fmt_pair = out_fmt + SAMPLE_FMT_NB*in_fmt; |
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171 return ctx; |
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172 } |
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173 |
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174 void av_audio_convert_free(AVAudioConvert *ctx) |
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175 { |
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176 av_free(ctx); |
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177 } |
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178 |
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179 int av_audio_convert(AVAudioConvert *ctx, |
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180 void * const out[6], const int out_stride[6], |
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181 const void * const in[6], const int in_stride[6], int len) |
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182 { |
3594 | 183 int ch; |
184 | |
185 //FIXME optimize common cases | |
186 | |
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187 for(ch=0; ch<ctx->out_channels; ch++){ |
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188 const int is= in_stride[ch]; |
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189 const int os= out_stride[ch]; |
8635 | 190 const uint8_t *pi= in[ch]; |
3594 | 191 uint8_t *po= out[ch]; |
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192 uint8_t *end= po + os*len; |
3594 | 193 if(!out[ch]) |
194 continue; | |
195 | |
196 #define CONV(ofmt, otype, ifmt, expr)\ | |
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197 if(ctx->fmt_pair == ofmt + SAMPLE_FMT_NB*ifmt){\ |
3594 | 198 do{\ |
199 *(otype*)po = expr; pi += is; po += os;\ | |
200 }while(po < end);\ | |
201 } | |
202 | |
5127 | 203 //FIXME put things below under ifdefs so we do not waste space for cases no codec will need |
204 //FIXME rounding and clipping ? | |
3594 | 205 |
8635 | 206 CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_U8 , *(const uint8_t*)pi) |
207 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8) | |
208 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24) | |
209 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7))) | |
210 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7))) | |
211 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80) | |
212 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S16, *(const int16_t*)pi) | |
213 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S16, *(const int16_t*)pi<<16) | |
214 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15))) | |
215 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15))) | |
216 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80) | |
217 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S32, *(const int32_t*)pi>>16) | |
218 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S32, *(const int32_t*)pi) | |
219 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31))) | |
220 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31))) | |
221 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_FLT, lrintf(*(const float*)pi * (1<<7)) + 0x80) | |
222 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_FLT, lrintf(*(const float*)pi * (1<<15))) | |
223 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_FLT, lrintf(*(const float*)pi * (1<<31))) | |
224 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_FLT, *(const float*)pi) | |
225 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_FLT, *(const float*)pi) | |
226 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_DBL, lrint(*(const double*)pi * (1<<7)) + 0x80) | |
227 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_DBL, lrint(*(const double*)pi * (1<<15))) | |
228 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_DBL, lrint(*(const double*)pi * (1<<31))) | |
229 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_DBL, *(const double*)pi) | |
230 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_DBL, *(const double*)pi) | |
3594 | 231 else return -1; |
232 } | |
233 return 0; | |
234 } |