Mercurial > libavcodec.hg
annotate audioconvert.c @ 12223:93e27a5401de libavcodec
Convert VP8 macroblock structures to a ring buffer.
Uses a slightly nonintuitive ring buffer size of (width+height*2) to simplify
addressing logic.
Also split out the segmentation map to a separate structure, necessary to
implement the ring buffer.
author | darkshikari |
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date | Thu, 22 Jul 2010 11:45:18 +0000 |
parents | b37112450ec0 |
children | d23e48091c51 |
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 |
3596 | 24 * audio conversion |
3594 | 25 * @author Michael Niedermayer <michaelni@gmx.at> |
26 */ | |
27 | |
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28 #include "libavutil/avstring.h" |
11657 | 29 #include "libavutil/libm.h" |
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30 #include "avcodec.h" |
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31 #include "audioconvert.h" |
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32 |
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33 typedef struct SampleFmtInfo { |
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34 const char *name; |
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35 int bits; |
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36 } SampleFmtInfo; |
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37 |
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38 /** this table gives more information about formats */ |
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39 static const SampleFmtInfo sample_fmt_info[SAMPLE_FMT_NB] = { |
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40 [SAMPLE_FMT_U8] = { .name = "u8", .bits = 8 }, |
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41 [SAMPLE_FMT_S16] = { .name = "s16", .bits = 16 }, |
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42 [SAMPLE_FMT_S32] = { .name = "s32", .bits = 32 }, |
7612 | 43 [SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32 }, |
44 [SAMPLE_FMT_DBL] = { .name = "dbl", .bits = 64 }, | |
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45 }; |
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46 |
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47 const char *avcodec_get_sample_fmt_name(int sample_fmt) |
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48 { |
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49 if (sample_fmt < 0 || sample_fmt >= SAMPLE_FMT_NB) |
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50 return NULL; |
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51 return sample_fmt_info[sample_fmt].name; |
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52 } |
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53 |
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54 enum SampleFormat avcodec_get_sample_fmt(const char* name) |
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55 { |
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56 int i; |
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57 |
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58 for (i=0; i < SAMPLE_FMT_NB; i++) |
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59 if (!strcmp(sample_fmt_info[i].name, name)) |
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60 return i; |
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61 return SAMPLE_FMT_NONE; |
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62 } |
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63 |
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64 void avcodec_sample_fmt_string (char *buf, int buf_size, int sample_fmt) |
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65 { |
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66 /* print header */ |
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67 if (sample_fmt < 0) |
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68 snprintf (buf, buf_size, "name " " depth"); |
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69 else if (sample_fmt < SAMPLE_FMT_NB) { |
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70 SampleFmtInfo info= sample_fmt_info[sample_fmt]; |
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71 snprintf (buf, buf_size, "%-6s" " %2d ", info.name, info.bits); |
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72 } |
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73 } |
3594 | 74 |
8098 | 75 static const char* const channel_names[]={ |
76 "FL", "FR", "FC", "LFE", "BL", "BR", "FLC", "FRC", | |
77 "BC", "SL", "SR", "TC", "TFL", "TFC", "TFR", "TBL", | |
78 "TBC", "TBR", | |
79 [29] = "DL", | |
80 [30] = "DR", | |
81 }; | |
82 | |
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83 static const char *get_channel_name(int channel_id) |
8098 | 84 { |
85 if (channel_id<0 || channel_id>=FF_ARRAY_ELEMS(channel_names)) | |
86 return NULL; | |
87 return channel_names[channel_id]; | |
88 } | |
89 | |
90 int64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name) | |
91 { | |
92 switch(nb_channels) { | |
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93 case 1: return CH_LAYOUT_MONO; |
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94 case 2: return CH_LAYOUT_STEREO; |
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95 case 3: return CH_LAYOUT_SURROUND; |
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96 case 4: return CH_LAYOUT_QUAD; |
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97 case 5: return CH_LAYOUT_5POINT0; |
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98 case 6: return CH_LAYOUT_5POINT1; |
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99 case 8: return CH_LAYOUT_7POINT1; |
8098 | 100 default: return 0; |
101 } | |
102 } | |
103 | |
104 static const struct { | |
105 const char *name; | |
106 int nb_channels; | |
107 int64_t layout; | |
8155 | 108 } channel_layout_map[] = { |
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109 { "mono", 1, CH_LAYOUT_MONO }, |
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110 { "stereo", 2, CH_LAYOUT_STEREO }, |
9502 | 111 { "4.0", 4, CH_LAYOUT_4POINT0 }, |
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112 { "quad", 4, CH_LAYOUT_QUAD }, |
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113 { "5.0", 5, CH_LAYOUT_5POINT0 }, |
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114 { "5.0", 5, CH_LAYOUT_5POINT0_BACK }, |
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115 { "5.1", 6, CH_LAYOUT_5POINT1 }, |
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116 { "5.1", 6, CH_LAYOUT_5POINT1_BACK }, |
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117 { "5.1+downmix", 8, CH_LAYOUT_5POINT1|CH_LAYOUT_STEREO_DOWNMIX, }, |
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118 { "7.1", 8, CH_LAYOUT_7POINT1 }, |
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119 { "7.1(wide)", 8, CH_LAYOUT_7POINT1_WIDE }, |
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120 { "7.1+downmix", 10, CH_LAYOUT_7POINT1|CH_LAYOUT_STEREO_DOWNMIX, }, |
8098 | 121 { 0 } |
122 }; | |
123 | |
124 void avcodec_get_channel_layout_string(char *buf, int buf_size, int nb_channels, int64_t channel_layout) | |
125 { | |
126 int i; | |
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 int avcodec_channel_layout_num_channels(int64_t channel_layout) |
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154 { |
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155 int count; |
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156 uint64_t x = channel_layout; |
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157 for (count = 0; x; count++) |
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158 x &= x-1; // unset lowest set bit |
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159 return count; |
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160 } |
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161 |
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162 struct AVAudioConvert { |
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163 int in_channels, out_channels; |
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164 int fmt_pair; |
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165 }; |
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166 |
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167 AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels, |
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168 enum SampleFormat in_fmt, int in_channels, |
7984 | 169 const float *matrix, int flags) |
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170 { |
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171 AVAudioConvert *ctx; |
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172 if (in_channels!=out_channels) |
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173 return NULL; /* FIXME: not supported */ |
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174 ctx = av_malloc(sizeof(AVAudioConvert)); |
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175 if (!ctx) |
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176 return NULL; |
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177 ctx->in_channels = in_channels; |
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178 ctx->out_channels = out_channels; |
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179 ctx->fmt_pair = out_fmt + SAMPLE_FMT_NB*in_fmt; |
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180 return ctx; |
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181 } |
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182 |
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183 void av_audio_convert_free(AVAudioConvert *ctx) |
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184 { |
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185 av_free(ctx); |
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186 } |
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187 |
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188 int av_audio_convert(AVAudioConvert *ctx, |
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189 void * const out[6], const int out_stride[6], |
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190 const void * const in[6], const int in_stride[6], int len) |
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191 { |
3594 | 192 int ch; |
193 | |
194 //FIXME optimize common cases | |
195 | |
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196 for(ch=0; ch<ctx->out_channels; ch++){ |
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197 const int is= in_stride[ch]; |
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198 const int os= out_stride[ch]; |
8635 | 199 const uint8_t *pi= in[ch]; |
3594 | 200 uint8_t *po= out[ch]; |
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201 uint8_t *end= po + os*len; |
3594 | 202 if(!out[ch]) |
203 continue; | |
204 | |
205 #define CONV(ofmt, otype, ifmt, expr)\ | |
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206 if(ctx->fmt_pair == ofmt + SAMPLE_FMT_NB*ifmt){\ |
3594 | 207 do{\ |
208 *(otype*)po = expr; pi += is; po += os;\ | |
209 }while(po < end);\ | |
210 } | |
211 | |
5127 | 212 //FIXME put things below under ifdefs so we do not waste space for cases no codec will need |
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213 //FIXME rounding ? |
3594 | 214 |
8635 | 215 CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_U8 , *(const uint8_t*)pi) |
216 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8) | |
217 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24) | |
218 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7))) | |
219 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7))) | |
220 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80) | |
221 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S16, *(const int16_t*)pi) | |
222 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S16, *(const int16_t*)pi<<16) | |
223 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15))) | |
224 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15))) | |
225 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80) | |
226 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S32, *(const int32_t*)pi>>16) | |
227 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S32, *(const int32_t*)pi) | |
228 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31))) | |
229 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31))) | |
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230 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80)) |
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231 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15)))) |
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232 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31)))) |
8635 | 233 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_FLT, *(const float*)pi) |
234 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_FLT, *(const float*)pi) | |
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235 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80)) |
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236 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15)))) |
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237 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31)))) |
8635 | 238 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_DBL, *(const double*)pi) |
239 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_DBL, *(const double*)pi) | |
3594 | 240 else return -1; |
241 } | |
242 return 0; | |
243 } |