Mercurial > mplayer.hg
annotate libaf/af_format.c @ 36309:583523331410
vf_screenshot: slice handling fix.
We can only support slices if the next filter does so.
Fixes bugzilla issue #2140.
author | reimar |
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date | Mon, 05 Aug 2013 18:36:57 +0000 |
parents | d206960484fe |
children | e79bac50b920 |
rev | line source |
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1 /* |
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2 * This audio filter changes the format of a data block. Valid |
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3 * formats are: AFMT_U8, AFMT_S8, AFMT_S16_LE, AFMT_S16_BE |
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4 * AFMT_U16_LE, AFMT_U16_BE, AFMT_S32_LE and AFMT_S32_BE. |
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5 * |
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6 * This file is part of MPlayer. |
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7 * |
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8 * MPlayer is free software; you can redistribute it and/or modify |
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9 * it under the terms of the GNU General Public License as published by |
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10 * the Free Software Foundation; either version 2 of the License, or |
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11 * (at your option) any later version. |
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12 * |
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13 * MPlayer is distributed in the hope that it will be useful, |
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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16 * GNU General Public License for more details. |
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17 * |
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18 * You should have received a copy of the GNU General Public License along |
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19 * with MPlayer; if not, write to the Free Software Foundation, Inc., |
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20 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. |
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21 */ |
7568 | 22 |
23 #include <stdio.h> | |
24 #include <stdlib.h> | |
25 #include <string.h> | |
26 #include <inttypes.h> | |
27 #include <limits.h> | |
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28 #include <math.h> |
7568 | 29 |
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30 #include "config.h" |
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31 #include "af.h" |
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32 #include "mp_msg.h" |
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33 #include "mpbswap.h" |
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34 #include "libvo/fastmemcpy.h" |
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35 |
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36 #include "libavutil/avutil.h" |
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37 #include "libavutil/common.h" |
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38 |
8167 | 39 /* Functions used by play to convert the input audio to the correct |
40 format */ | |
7568 | 41 |
24561 | 42 /* The below includes retrieves functions for converting to and from |
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43 ulaw and alaw */ |
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44 #include "af_format_ulaw.h" |
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45 #include "af_format_alaw.h" |
7568 | 46 |
24561 | 47 // Switch endianness |
8167 | 48 static void endian(void* in, void* out, int len, int bps); |
24561 | 49 // From signed to unsigned and the other way |
16664 | 50 static void si2us(void* data, int len, int bps); |
8167 | 51 // Change the number of bits per sample |
52 static void change_bps(void* in, void* out, int len, int inbps, int outbps); | |
53 // From float to int signed | |
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54 static void float2int(float* in, void* out, int len, int bps); |
8167 | 55 // From signed int to float |
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56 static void int2float(void* in, float* out, int len, int bps); |
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57 |
14272 | 58 static af_data_t* play(struct af_instance_s* af, af_data_t* data); |
59 static af_data_t* play_swapendian(struct af_instance_s* af, af_data_t* data); | |
60 static af_data_t* play_float_s16(struct af_instance_s* af, af_data_t* data); | |
61 static af_data_t* play_s16_float(struct af_instance_s* af, af_data_t* data); | |
62 | |
8607 | 63 // Helper functions to check sanity for input arguments |
64 | |
65 // Sanity check for bytes per sample | |
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66 static int check_bps(int bps) |
8607 | 67 { |
12478 | 68 if(bps != 4 && bps != 3 && bps != 2 && bps != 1){ |
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69 mp_msg(MSGT_AFILTER, MSGL_ERR, "[format] The number of bytes per sample" |
12478 | 70 " must be 1, 2, 3 or 4. Current value is %i \n",bps); |
8607 | 71 return AF_ERROR; |
72 } | |
73 return AF_OK; | |
74 } | |
75 | |
76 // Check for unsupported formats | |
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77 static int check_format(int format) |
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78 { |
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79 char buf[256]; |
8607 | 80 switch(format & AF_FORMAT_SPECIAL_MASK){ |
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81 case 0: |
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82 case AF_FORMAT_MU_LAW: |
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83 case AF_FORMAT_A_LAW: |
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84 return AF_OK; |
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85 } |
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86 mp_msg(MSGT_AFILTER, MSGL_ERR, "[format] Sample format %s not yet supported \n", |
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87 af_fmt2str(format,buf,256)); |
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88 return AF_ERROR; |
8607 | 89 } |
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90 |
7568 | 91 // Initialization and runtime control |
92 static int control(struct af_instance_s* af, int cmd, void* arg) | |
93 { | |
94 switch(cmd){ | |
8167 | 95 case AF_CONTROL_REINIT:{ |
96 char buf1[256]; | |
97 char buf2[256]; | |
14272 | 98 af_data_t *data = arg; |
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99 int supported_ac3 = 0; |
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100 |
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101 // Make sure this filter isn't redundant |
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102 if(af->data->format == data->format && |
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103 af->data->bps == data->bps) |
7568 | 104 return AF_DETACH; |
8607 | 105 |
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106 // A bit complex because we can convert AC3 |
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107 // to generic iec61937 but not the other way |
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108 // round. |
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109 if (AF_FORMAT_IS_AC3(af->data->format)) |
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110 supported_ac3 = AF_FORMAT_IS_AC3(data->format); |
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111 else if (AF_FORMAT_IS_IEC61937(af->data->format)) |
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112 supported_ac3 = AF_FORMAT_IS_IEC61937(data->format); |
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113 |
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114 // Allow trivial AC3-endianness conversion |
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115 if (!supported_ac3) |
24521 | 116 // Check for errors in configuration |
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117 if((AF_OK != check_bps(data->bps)) || |
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118 (AF_OK != check_format(data->format)) || |
8607 | 119 (AF_OK != check_bps(af->data->bps)) || |
120 (AF_OK != check_format(af->data->format))) | |
8167 | 121 return AF_ERROR; |
122 | |
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123 af_fmt2str(data->format,buf1,256); |
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124 af_fmt2str(af->data->format,buf2,256); |
29049 | 125 mp_msg(MSGT_AFILTER, MSGL_V, "[format] Changing sample format from %s to %s\n", |
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126 buf1, buf2); |
7568 | 127 |
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128 af->data->rate = data->rate; |
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129 af->data->nch = data->nch; |
24888 | 130 af->mul = (double)af->data->bps / data->bps; |
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131 |
14272 | 132 af->play = play; // set default |
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133 |
24561 | 134 // look whether only endianness differences are there |
14272 | 135 if ((af->data->format & ~AF_FORMAT_END_MASK) == |
136 (data->format & ~AF_FORMAT_END_MASK)) | |
137 { | |
29049 | 138 mp_msg(MSGT_AFILTER, MSGL_V, "[format] Accelerated endianness conversion only\n"); |
14272 | 139 af->play = play_swapendian; |
140 } | |
141 if ((data->format == AF_FORMAT_FLOAT_NE) && | |
142 (af->data->format == AF_FORMAT_S16_NE)) | |
143 { | |
29049 | 144 mp_msg(MSGT_AFILTER, MSGL_V, "[format] Accelerated %s to %s conversion\n", |
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145 buf1, buf2); |
14272 | 146 af->play = play_float_s16; |
147 } | |
148 if ((data->format == AF_FORMAT_S16_NE) && | |
149 (af->data->format == AF_FORMAT_FLOAT_NE)) | |
150 { | |
29049 | 151 mp_msg(MSGT_AFILTER, MSGL_V, "[format] Accelerated %s to %s conversion\n", |
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152 buf1, buf2); |
14272 | 153 af->play = play_s16_float; |
154 } | |
7568 | 155 return AF_OK; |
8167 | 156 } |
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157 case AF_CONTROL_COMMAND_LINE:{ |
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158 int format = af_str2fmt_short(arg); |
15311 | 159 if (format == -1) { |
29049 | 160 mp_msg(MSGT_AFILTER, MSGL_ERR, "[format] %s is not a valid format\n", (char *)arg); |
15311 | 161 return AF_ERROR; |
162 } | |
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163 if(AF_OK != af->control(af,AF_CONTROL_FORMAT_FMT | AF_CONTROL_SET,&format)) |
8607 | 164 return AF_ERROR; |
165 return AF_OK; | |
8167 | 166 } |
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167 case AF_CONTROL_FORMAT_FMT | AF_CONTROL_SET:{ |
24561 | 168 // Check for errors in configuration |
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169 if(!AF_FORMAT_IS_AC3(*(int*)arg) && AF_OK != check_format(*(int*)arg)) |
8607 | 170 return AF_ERROR; |
171 | |
172 af->data->format = *(int*)arg; | |
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173 af->data->bps = af_fmt2bits(af->data->format)/8; |
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174 |
7568 | 175 return AF_OK; |
176 } | |
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177 } |
7568 | 178 return AF_UNKNOWN; |
179 } | |
180 | |
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181 // Deallocate memory |
7568 | 182 static void uninit(struct af_instance_s* af) |
183 { | |
22179 | 184 if (af->data) |
185 free(af->data->audio); | |
186 free(af->data); | |
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187 af->setup = 0; |
7568 | 188 } |
189 | |
14272 | 190 static af_data_t* play_swapendian(struct af_instance_s* af, af_data_t* data) |
191 { | |
192 af_data_t* l = af->data; // Local data | |
193 af_data_t* c = data; // Current working data | |
24561 | 194 int len = c->len/c->bps; // Length in samples of current audio block |
14272 | 195 |
196 if(AF_OK != RESIZE_LOCAL_BUFFER(af,data)) | |
197 return NULL; | |
198 | |
199 endian(c->audio,l->audio,len,c->bps); | |
200 | |
201 c->audio = l->audio; | |
202 c->format = l->format; | |
203 | |
204 return c; | |
205 } | |
206 | |
207 static af_data_t* play_float_s16(struct af_instance_s* af, af_data_t* data) | |
208 { | |
209 af_data_t* l = af->data; // Local data | |
210 af_data_t* c = data; // Current working data | |
24561 | 211 int len = c->len/4; // Length in samples of current audio block |
14272 | 212 |
213 if(AF_OK != RESIZE_LOCAL_BUFFER(af,data)) | |
214 return NULL; | |
215 | |
216 float2int(c->audio, l->audio, len, 2); | |
217 | |
218 c->audio = l->audio; | |
219 c->len = len*2; | |
220 c->bps = 2; | |
221 c->format = l->format; | |
222 | |
223 return c; | |
224 } | |
225 | |
226 static af_data_t* play_s16_float(struct af_instance_s* af, af_data_t* data) | |
227 { | |
228 af_data_t* l = af->data; // Local data | |
229 af_data_t* c = data; // Current working data | |
24561 | 230 int len = c->len/2; // Length in samples of current audio block |
14272 | 231 |
232 if(AF_OK != RESIZE_LOCAL_BUFFER(af,data)) | |
233 return NULL; | |
234 | |
235 int2float(c->audio, l->audio, len, 2); | |
236 | |
237 c->audio = l->audio; | |
238 c->len = len*4; | |
239 c->bps = 4; | |
240 c->format = l->format; | |
241 | |
242 return c; | |
243 } | |
244 | |
7568 | 245 // Filter data through filter |
246 static af_data_t* play(struct af_instance_s* af, af_data_t* data) | |
247 { | |
8167 | 248 af_data_t* l = af->data; // Local data |
249 af_data_t* c = data; // Current working data | |
24561 | 250 int len = c->len/c->bps; // Length in samples of current audio block |
7568 | 251 |
252 if(AF_OK != RESIZE_LOCAL_BUFFER(af,data)) | |
253 return NULL; | |
254 | |
8167 | 255 // Change to cpu native endian format |
256 if((c->format&AF_FORMAT_END_MASK)!=AF_FORMAT_NE) | |
257 endian(c->audio,c->audio,len,c->bps); | |
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258 |
8167 | 259 // Conversion table |
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260 if((c->format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_MU_LAW) { |
8167 | 261 from_ulaw(c->audio, l->audio, len, l->bps, l->format&AF_FORMAT_POINT_MASK); |
262 if(AF_FORMAT_A_LAW == (l->format&AF_FORMAT_SPECIAL_MASK)) | |
263 to_ulaw(l->audio, l->audio, len, 1, AF_FORMAT_SI); | |
264 if((l->format&AF_FORMAT_SIGN_MASK) == AF_FORMAT_US) | |
16664 | 265 si2us(l->audio,len,l->bps); |
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266 } else if((c->format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_A_LAW) { |
8167 | 267 from_alaw(c->audio, l->audio, len, l->bps, l->format&AF_FORMAT_POINT_MASK); |
268 if(AF_FORMAT_A_LAW == (l->format&AF_FORMAT_SPECIAL_MASK)) | |
269 to_alaw(l->audio, l->audio, len, 1, AF_FORMAT_SI); | |
270 if((l->format&AF_FORMAT_SIGN_MASK) == AF_FORMAT_US) | |
16664 | 271 si2us(l->audio,len,l->bps); |
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272 } else if((c->format & AF_FORMAT_POINT_MASK) == AF_FORMAT_F) { |
8167 | 273 switch(l->format&AF_FORMAT_SPECIAL_MASK){ |
274 case(AF_FORMAT_MU_LAW): | |
275 to_ulaw(c->audio, l->audio, len, c->bps, c->format&AF_FORMAT_POINT_MASK); | |
276 break; | |
277 case(AF_FORMAT_A_LAW): | |
278 to_alaw(c->audio, l->audio, len, c->bps, c->format&AF_FORMAT_POINT_MASK); | |
7568 | 279 break; |
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280 default: |
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281 float2int(c->audio, l->audio, len, l->bps); |
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282 if((l->format&AF_FORMAT_SIGN_MASK) == AF_FORMAT_US) |
16664 | 283 si2us(l->audio,len,l->bps); |
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284 break; |
8167 | 285 } |
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286 } else { |
8167 | 287 // Input must be int |
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288 |
8167 | 289 // Change signed/unsigned |
290 if((c->format&AF_FORMAT_SIGN_MASK) != (l->format&AF_FORMAT_SIGN_MASK)){ | |
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291 si2us(c->audio,len,c->bps); |
8167 | 292 } |
293 // Convert to special formats | |
8180
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294 switch(l->format&(AF_FORMAT_SPECIAL_MASK|AF_FORMAT_POINT_MASK)){ |
8167 | 295 case(AF_FORMAT_MU_LAW): |
296 to_ulaw(c->audio, l->audio, len, c->bps, c->format&AF_FORMAT_POINT_MASK); | |
297 break; | |
298 case(AF_FORMAT_A_LAW): | |
299 to_alaw(c->audio, l->audio, len, c->bps, c->format&AF_FORMAT_POINT_MASK); | |
300 break; | |
8180
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301 case(AF_FORMAT_F): |
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302 int2float(c->audio, l->audio, len, c->bps); |
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303 break; |
8167 | 304 default: |
8180
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305 // Change the number of bits |
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306 if(c->bps != l->bps) |
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Fixing segfault bug and addnig support for lrintf() in format conversion
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307 change_bps(c->audio,l->audio,len,c->bps,l->bps); |
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Fixing segfault bug and addnig support for lrintf() in format conversion
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308 else |
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309 fast_memcpy(l->audio,c->audio,len*c->bps); |
7568 | 310 break; |
311 } | |
312 } | |
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Fix for audio filters on big endian cpus. It's working now on Solaris SPARC &
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313 |
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314 // Switch from cpu native endian to the correct endianness |
8167 | 315 if((l->format&AF_FORMAT_END_MASK)!=AF_FORMAT_NE) |
316 endian(l->audio,l->audio,len,l->bps); | |
7568 | 317 |
318 // Set output data | |
319 c->audio = l->audio; | |
8167 | 320 c->len = len*l->bps; |
7568 | 321 c->bps = l->bps; |
322 c->format = l->format; | |
323 return c; | |
324 } | |
325 | |
326 // Allocate memory and set function pointers | |
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327 static int af_open(af_instance_t* af){ |
7568 | 328 af->control=control; |
329 af->uninit=uninit; | |
330 af->play=play; | |
24888 | 331 af->mul=1; |
7568 | 332 af->data=calloc(1,sizeof(af_data_t)); |
333 if(af->data == NULL) | |
334 return AF_ERROR; | |
335 return AF_OK; | |
336 } | |
337 | |
338 // Description of this filter | |
339 af_info_t af_info_format = { | |
340 "Sample format conversion", | |
341 "format", | |
342 "Anders", | |
343 "", | |
7615 | 344 AF_FLAGS_REENTRANT, |
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345 af_open |
7568 | 346 }; |
8167 | 347 |
12478 | 348 static inline uint32_t load24bit(void* data, int pos) { |
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349 #if HAVE_BIGENDIAN |
12478 | 350 return (((uint32_t)((uint8_t*)data)[3*pos])<<24) | |
351 (((uint32_t)((uint8_t*)data)[3*pos+1])<<16) | | |
352 (((uint32_t)((uint8_t*)data)[3*pos+2])<<8); | |
353 #else | |
354 return (((uint32_t)((uint8_t*)data)[3*pos])<<8) | | |
355 (((uint32_t)((uint8_t*)data)[3*pos+1])<<16) | | |
356 (((uint32_t)((uint8_t*)data)[3*pos+2])<<24); | |
357 #endif | |
358 } | |
359 | |
360 static inline void store24bit(void* data, int pos, uint32_t expanded_value) { | |
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361 #if HAVE_BIGENDIAN |
12478 | 362 ((uint8_t*)data)[3*pos]=expanded_value>>24; |
363 ((uint8_t*)data)[3*pos+1]=expanded_value>>16; | |
364 ((uint8_t*)data)[3*pos+2]=expanded_value>>8; | |
365 #else | |
366 ((uint8_t*)data)[3*pos]=expanded_value>>8; | |
367 ((uint8_t*)data)[3*pos+1]=expanded_value>>16; | |
368 ((uint8_t*)data)[3*pos+2]=expanded_value>>24; | |
369 #endif | |
370 } | |
371 | |
8167 | 372 // Function implementations used by play |
373 static void endian(void* in, void* out, int len, int bps) | |
374 { | |
375 register int i; | |
376 switch(bps){ | |
377 case(2):{ | |
378 for(i=0;i<len;i++){ | |
12486 | 379 ((uint16_t*)out)[i]=bswap_16(((uint16_t*)in)[i]); |
8167 | 380 } |
381 break; | |
382 } | |
12478 | 383 case(3):{ |
384 register uint8_t s; | |
385 for(i=0;i<len;i++){ | |
386 s=((uint8_t*)in)[3*i]; | |
387 ((uint8_t*)out)[3*i]=((uint8_t*)in)[3*i+2]; | |
12481
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388 if (in != out) |
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389 ((uint8_t*)out)[3*i+1]=((uint8_t*)in)[3*i+1]; |
12478 | 390 ((uint8_t*)out)[3*i+2]=s; |
391 } | |
392 break; | |
393 } | |
8167 | 394 case(4):{ |
395 for(i=0;i<len;i++){ | |
12486 | 396 ((uint32_t*)out)[i]=bswap_32(((uint32_t*)in)[i]); |
8167 | 397 } |
398 break; | |
399 } | |
400 } | |
401 } | |
402 | |
16664 | 403 static void si2us(void* data, int len, int bps) |
8167 | 404 { |
16664 | 405 register long i = -(len * bps); |
406 register uint8_t *p = &((uint8_t *)data)[len * bps]; | |
407 #if AF_FORMAT_NE == AF_FORMAT_LE | |
408 p += bps - 1; | |
409 #endif | |
410 if (len <= 0) return; | |
411 do { | |
412 p[i] ^= 0x80; | |
413 } while (i += bps); | |
8167 | 414 } |
415 | |
416 static void change_bps(void* in, void* out, int len, int inbps, int outbps) | |
417 { | |
418 register int i; | |
419 switch(inbps){ | |
420 case(1): | |
421 switch(outbps){ | |
422 case(2): | |
423 for(i=0;i<len;i++) | |
424 ((uint16_t*)out)[i]=((uint16_t)((uint8_t*)in)[i])<<8; | |
425 break; | |
12478 | 426 case(3): |
427 for(i=0;i<len;i++) | |
428 store24bit(out, i, ((uint32_t)((uint8_t*)in)[i])<<24); | |
429 break; | |
8167 | 430 case(4): |
431 for(i=0;i<len;i++) | |
432 ((uint32_t*)out)[i]=((uint32_t)((uint8_t*)in)[i])<<24; | |
433 break; | |
434 } | |
435 break; | |
436 case(2): | |
437 switch(outbps){ | |
438 case(1): | |
439 for(i=0;i<len;i++) | |
440 ((uint8_t*)out)[i]=(uint8_t)((((uint16_t*)in)[i])>>8); | |
441 break; | |
12478 | 442 case(3): |
443 for(i=0;i<len;i++) | |
444 store24bit(out, i, ((uint32_t)((uint16_t*)in)[i])<<16); | |
445 break; | |
8167 | 446 case(4): |
447 for(i=0;i<len;i++) | |
448 ((uint32_t*)out)[i]=((uint32_t)((uint16_t*)in)[i])<<16; | |
449 break; | |
450 } | |
451 break; | |
12478 | 452 case(3): |
453 switch(outbps){ | |
454 case(1): | |
455 for(i=0;i<len;i++) | |
456 ((uint8_t*)out)[i]=(uint8_t)(load24bit(in, i)>>24); | |
457 break; | |
458 case(2): | |
459 for(i=0;i<len;i++) | |
460 ((uint16_t*)out)[i]=(uint16_t)(load24bit(in, i)>>16); | |
461 break; | |
462 case(4): | |
463 for(i=0;i<len;i++) | |
464 ((uint32_t*)out)[i]=(uint32_t)load24bit(in, i); | |
465 break; | |
466 } | |
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diff
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|
467 break; |
8167 | 468 case(4): |
469 switch(outbps){ | |
470 case(1): | |
471 for(i=0;i<len;i++) | |
472 ((uint8_t*)out)[i]=(uint8_t)((((uint32_t*)in)[i])>>24); | |
473 break; | |
474 case(2): | |
475 for(i=0;i<len;i++) | |
476 ((uint16_t*)out)[i]=(uint16_t)((((uint32_t*)in)[i])>>16); | |
477 break; | |
12478 | 478 case(3): |
479 for(i=0;i<len;i++) | |
480 store24bit(out, i, ((uint32_t*)in)[i]); | |
481 break; | |
8167 | 482 } |
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|
483 break; |
8167 | 484 } |
485 } | |
486 | |
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100l for me, lrintf is better. now fixed so it should be prototyped, and should work even if there is no prototype
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diff
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|
487 static void float2int(float* in, void* out, int len, int bps) |
8167 | 488 { |
33723
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iive
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diff
changeset
|
489 float f; |
8167 | 490 register int i; |
491 switch(bps){ | |
492 case(1): | |
493 for(i=0;i<len;i++) | |
33723
6c8743e5fa30
Audio format conversion from float to int is not checked for overflow.
iive
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diff
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|
494 ((int8_t *)out)[i] = av_clip_int8(lrintf(128.0 * in[i])); |
8167 | 495 break; |
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|
496 case(2): |
8167 | 497 for(i=0;i<len;i++) |
33723
6c8743e5fa30
Audio format conversion from float to int is not checked for overflow.
iive
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diff
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|
498 ((int16_t*)out)[i] = av_clip_int16(lrintf(32768.0 * in[i])); |
8167 | 499 break; |
12478 | 500 case(3): |
33723
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Audio format conversion from float to int is not checked for overflow.
iive
parents:
30241
diff
changeset
|
501 for(i=0;i<len;i++){ |
33736
a5a54c7a15ce
Fix the precision loss in float -> 32bit conversion case, introduced
iive
parents:
33723
diff
changeset
|
502 f = in[i] * 8388608; |
33723
6c8743e5fa30
Audio format conversion from float to int is not checked for overflow.
iive
parents:
30241
diff
changeset
|
503 store24bit(out, i, av_clip(lrintf(f), -1*(1<<23), (1<<23)-1) << 8); |
6c8743e5fa30
Audio format conversion from float to int is not checked for overflow.
iive
parents:
30241
diff
changeset
|
504 } |
12478 | 505 break; |
8167 | 506 case(4): |
33723
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Audio format conversion from float to int is not checked for overflow.
iive
parents:
30241
diff
changeset
|
507 for(i=0;i<len;i++){ |
33736
a5a54c7a15ce
Fix the precision loss in float -> 32bit conversion case, introduced
iive
parents:
33723
diff
changeset
|
508 f = in[i]; |
a5a54c7a15ce
Fix the precision loss in float -> 32bit conversion case, introduced
iive
parents:
33723
diff
changeset
|
509 if (f <= -1.0) |
a5a54c7a15ce
Fix the precision loss in float -> 32bit conversion case, introduced
iive
parents:
33723
diff
changeset
|
510 ((int32_t*)out)[i] = INT_MIN; |
a5a54c7a15ce
Fix the precision loss in float -> 32bit conversion case, introduced
iive
parents:
33723
diff
changeset
|
511 else |
a5a54c7a15ce
Fix the precision loss in float -> 32bit conversion case, introduced
iive
parents:
33723
diff
changeset
|
512 if (f >= 1.0)//no need to use corrected constant, rounding won't cause overflow |
a5a54c7a15ce
Fix the precision loss in float -> 32bit conversion case, introduced
iive
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diff
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|
513 ((int32_t*)out)[i] = INT_MAX; |
a5a54c7a15ce
Fix the precision loss in float -> 32bit conversion case, introduced
iive
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diff
changeset
|
514 else |
a5a54c7a15ce
Fix the precision loss in float -> 32bit conversion case, introduced
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diff
changeset
|
515 ((int32_t*)out)[i] = lrintf(f*2147483648.0); |
a5a54c7a15ce
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iive
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diff
changeset
|
516 |
33723
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diff
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|
517 } |
8167 | 518 break; |
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|
519 } |
8167 | 520 } |
521 | |
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14758
diff
changeset
|
522 static void int2float(void* in, float* out, int len, int bps) |
8167 | 523 { |
524 register int i; | |
525 switch(bps){ | |
526 case(1): | |
527 for(i=0;i<len;i++) | |
14791
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100l for me, lrintf is better. now fixed so it should be prototyped, and should work even if there is no prototype
rfelker
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14758
diff
changeset
|
528 out[i]=(1.0/128.0)*((int8_t*)in)[i]; |
8167 | 529 break; |
530 case(2): | |
531 for(i=0;i<len;i++) | |
14791
df515839c8a9
100l for me, lrintf is better. now fixed so it should be prototyped, and should work even if there is no prototype
rfelker
parents:
14758
diff
changeset
|
532 out[i]=(1.0/32768.0)*((int16_t*)in)[i]; |
8167 | 533 break; |
12478 | 534 case(3): |
535 for(i=0;i<len;i++) | |
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rfelker
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14758
diff
changeset
|
536 out[i]=(1.0/2147483648.0)*((int32_t)load24bit(in, i)); |
12478 | 537 break; |
8167 | 538 case(4): |
539 for(i=0;i<len;i++) | |
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100l for me, lrintf is better. now fixed so it should be prototyped, and should work even if there is no prototype
rfelker
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14758
diff
changeset
|
540 out[i]=(1.0/2147483648.0)*((int32_t*)in)[i]; |
8167 | 541 break; |
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diff
changeset
|
542 } |
8167 | 543 } |