Mercurial > libavcodec.hg
annotate audioconvert.c @ 9530:dbb16aa52d43 libavcodec
mlpdec: Restart header sync must be 0x31ea for MLP.
| author | ramiro |
|---|---|
| date | Tue, 21 Apr 2009 21:57:23 +0000 |
| parents | c2dba7ed94dc |
| children | 63dbab428e01 |
| rev | line source |
|---|---|
| 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 }, |
| 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 if (channel_layout==0) | |
| 129 channel_layout = avcodec_guess_channel_layout(nb_channels, CODEC_ID_NONE, NULL); | |
| 130 | |
| 131 for (i=0; channel_layout_map[i].name; i++) | |
| 132 if (nb_channels == channel_layout_map[i].nb_channels && | |
| 133 channel_layout == channel_layout_map[i].layout) { | |
| 8784 | 134 av_strlcpy(buf, channel_layout_map[i].name, buf_size); |
| 8098 | 135 return; |
| 136 } | |
| 137 | |
| 138 snprintf(buf, buf_size, "%d channels", nb_channels); | |
| 139 if (channel_layout) { | |
| 140 int i,ch; | |
| 141 av_strlcat(buf, " (", buf_size); | |
| 142 for(i=0,ch=0; i<64; i++) { | |
| 143 if ((channel_layout & (1L<<i))) { | |
| 144 const char *name = get_channel_name(i); | |
| 145 if (name) { | |
| 146 if (ch>0) av_strlcat(buf, "|", buf_size); | |
| 147 av_strlcat(buf, name, buf_size); | |
| 148 } | |
| 149 ch++; | |
| 150 } | |
| 151 } | |
| 152 av_strlcat(buf, ")", buf_size); | |
| 153 } | |
| 154 } | |
| 155 | |
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156 int avcodec_channel_layout_num_channels(int64_t channel_layout) |
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157 { |
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158 int count; |
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159 uint64_t x = channel_layout; |
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160 for (count = 0; x; count++) |
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161 x &= x-1; // unset lowest set bit |
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162 return count; |
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163 } |
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164 |
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165 struct AVAudioConvert { |
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166 int in_channels, out_channels; |
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167 int fmt_pair; |
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168 }; |
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169 |
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170 AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels, |
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171 enum SampleFormat in_fmt, int in_channels, |
| 7984 | 172 const float *matrix, int flags) |
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173 { |
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174 AVAudioConvert *ctx; |
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175 if (in_channels!=out_channels) |
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176 return NULL; /* FIXME: not supported */ |
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177 ctx = av_malloc(sizeof(AVAudioConvert)); |
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178 if (!ctx) |
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179 return NULL; |
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180 ctx->in_channels = in_channels; |
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181 ctx->out_channels = out_channels; |
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182 ctx->fmt_pair = out_fmt + SAMPLE_FMT_NB*in_fmt; |
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183 return ctx; |
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184 } |
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185 |
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186 void av_audio_convert_free(AVAudioConvert *ctx) |
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187 { |
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188 av_free(ctx); |
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189 } |
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190 |
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191 int av_audio_convert(AVAudioConvert *ctx, |
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192 void * const out[6], const int out_stride[6], |
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193 const void * const in[6], const int in_stride[6], int len) |
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194 { |
| 3594 | 195 int ch; |
| 196 | |
| 197 //FIXME optimize common cases | |
| 198 | |
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199 for(ch=0; ch<ctx->out_channels; ch++){ |
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200 const int is= in_stride[ch]; |
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201 const int os= out_stride[ch]; |
| 8635 | 202 const uint8_t *pi= in[ch]; |
| 3594 | 203 uint8_t *po= out[ch]; |
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204 uint8_t *end= po + os*len; |
| 3594 | 205 if(!out[ch]) |
| 206 continue; | |
| 207 | |
| 208 #define CONV(ofmt, otype, ifmt, expr)\ | |
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209 if(ctx->fmt_pair == ofmt + SAMPLE_FMT_NB*ifmt){\ |
| 3594 | 210 do{\ |
| 211 *(otype*)po = expr; pi += is; po += os;\ | |
| 212 }while(po < end);\ | |
| 213 } | |
| 214 | |
| 5127 | 215 //FIXME put things below under ifdefs so we do not waste space for cases no codec will need |
| 216 //FIXME rounding and clipping ? | |
| 3594 | 217 |
| 8635 | 218 CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_U8 , *(const uint8_t*)pi) |
| 219 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8) | |
| 220 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24) | |
| 221 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7))) | |
| 222 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7))) | |
| 223 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80) | |
| 224 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S16, *(const int16_t*)pi) | |
| 225 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S16, *(const int16_t*)pi<<16) | |
| 226 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15))) | |
| 227 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15))) | |
| 228 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80) | |
| 229 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S32, *(const int32_t*)pi>>16) | |
| 230 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S32, *(const int32_t*)pi) | |
| 231 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31))) | |
| 232 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31))) | |
| 233 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_FLT, lrintf(*(const float*)pi * (1<<7)) + 0x80) | |
| 234 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_FLT, lrintf(*(const float*)pi * (1<<15))) | |
| 235 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_FLT, lrintf(*(const float*)pi * (1<<31))) | |
| 236 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_FLT, *(const float*)pi) | |
| 237 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_FLT, *(const float*)pi) | |
| 238 else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_DBL, lrint(*(const double*)pi * (1<<7)) + 0x80) | |
| 239 else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_DBL, lrint(*(const double*)pi * (1<<15))) | |
| 240 else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_DBL, lrint(*(const double*)pi * (1<<31))) | |
| 241 else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_DBL, *(const double*)pi) | |
| 242 else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_DBL, *(const double*)pi) | |
| 3594 | 243 else return -1; |
| 244 } | |
| 245 return 0; | |
| 246 } |
