Mercurial > libavcodec.hg
annotate resample.c @ 4970:28b67304e5d0 libavcodec
Fix r8963
author | ramiro |
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date | Thu, 10 May 2007 15:00:09 +0000 |
parents | 5c0c96d437f2 |
children | 45d083bbbbe7 |
rev | line source |
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0 | 1 /* |
2 * Sample rate convertion for both audio and video | |
429 | 3 * Copyright (c) 2000 Fabrice Bellard. |
0 | 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 |
429 | 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. |
0 | 11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
0 | 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
429 | 14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
15 * Lesser General Public License for more details. | |
0 | 16 * |
429 | 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 |
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
0 | 20 */ |
1106 | 21 |
22 /** | |
23 * @file resample.c | |
24 * Sample rate convertion for both audio and video. | |
25 */ | |
26 | |
64 | 27 #include "avcodec.h" |
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28 |
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29 struct AVResampleContext; |
0 | 30 |
31 struct ReSampleContext { | |
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32 struct AVResampleContext *resample_context; |
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33 short *temp[2]; |
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34 int temp_len; |
0 | 35 float ratio; |
36 /* channel convert */ | |
37 int input_channels, output_channels, filter_channels; | |
38 }; | |
39 | |
40 /* n1: number of samples */ | |
41 static void stereo_to_mono(short *output, short *input, int n1) | |
42 { | |
43 short *p, *q; | |
44 int n = n1; | |
45 | |
46 p = input; | |
47 q = output; | |
48 while (n >= 4) { | |
49 q[0] = (p[0] + p[1]) >> 1; | |
50 q[1] = (p[2] + p[3]) >> 1; | |
51 q[2] = (p[4] + p[5]) >> 1; | |
52 q[3] = (p[6] + p[7]) >> 1; | |
53 q += 4; | |
54 p += 8; | |
55 n -= 4; | |
56 } | |
57 while (n > 0) { | |
58 q[0] = (p[0] + p[1]) >> 1; | |
59 q++; | |
60 p += 2; | |
61 n--; | |
62 } | |
63 } | |
64 | |
65 /* n1: number of samples */ | |
66 static void mono_to_stereo(short *output, short *input, int n1) | |
67 { | |
68 short *p, *q; | |
69 int n = n1; | |
70 int v; | |
71 | |
72 p = input; | |
73 q = output; | |
74 while (n >= 4) { | |
75 v = p[0]; q[0] = v; q[1] = v; | |
76 v = p[1]; q[2] = v; q[3] = v; | |
77 v = p[2]; q[4] = v; q[5] = v; | |
78 v = p[3]; q[6] = v; q[7] = v; | |
79 q += 8; | |
80 p += 4; | |
81 n -= 4; | |
82 } | |
83 while (n > 0) { | |
84 v = p[0]; q[0] = v; q[1] = v; | |
85 q += 2; | |
86 p += 1; | |
87 n--; | |
88 } | |
89 } | |
90 | |
91 /* XXX: should use more abstract 'N' channels system */ | |
92 static void stereo_split(short *output1, short *output2, short *input, int n) | |
93 { | |
94 int i; | |
95 | |
96 for(i=0;i<n;i++) { | |
97 *output1++ = *input++; | |
98 *output2++ = *input++; | |
99 } | |
100 } | |
101 | |
102 static void stereo_mux(short *output, short *input1, short *input2, int n) | |
103 { | |
104 int i; | |
105 | |
106 for(i=0;i<n;i++) { | |
107 *output++ = *input1++; | |
108 *output++ = *input2++; | |
109 } | |
110 } | |
111 | |
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112 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) |
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113 { |
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114 int i; |
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115 short l,r; |
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116 |
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117 for(i=0;i<n;i++) { |
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118 l=*input1++; |
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119 r=*input2++; |
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120 *output++ = l; /* left */ |
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121 *output++ = (l/2)+(r/2); /* center */ |
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122 *output++ = r; /* right */ |
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123 *output++ = 0; /* left surround */ |
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124 *output++ = 0; /* right surroud */ |
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125 *output++ = 0; /* low freq */ |
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126 } |
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127 } |
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128 |
2967 | 129 ReSampleContext *audio_resample_init(int output_channels, int input_channels, |
0 | 130 int output_rate, int input_rate) |
131 { | |
132 ReSampleContext *s; | |
2967 | 133 |
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134 if ( input_channels > 2) |
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135 { |
2979 | 136 av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported."); |
137 return NULL; | |
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138 } |
0 | 139 |
140 s = av_mallocz(sizeof(ReSampleContext)); | |
141 if (!s) | |
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142 { |
2979 | 143 av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context."); |
144 return NULL; | |
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145 } |
0 | 146 |
147 s->ratio = (float)output_rate / (float)input_rate; | |
2967 | 148 |
0 | 149 s->input_channels = input_channels; |
150 s->output_channels = output_channels; | |
2967 | 151 |
0 | 152 s->filter_channels = s->input_channels; |
153 if (s->output_channels < s->filter_channels) | |
154 s->filter_channels = s->output_channels; | |
155 | |
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156 /* |
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157 * ac3 output is the only case where filter_channels could be greater than 2. |
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158 * input channels can't be greater than 2, so resample the 2 channels and then |
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159 * expand to 6 channels after the resampling. |
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160 */ |
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161 if(s->filter_channels>2) |
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162 s->filter_channels = 2; |
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163 |
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164 #define TAPS 16 |
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165 s->resample_context= av_resample_init(output_rate, input_rate, TAPS, 10, 0, 0.8); |
2967 | 166 |
0 | 167 return s; |
168 } | |
169 | |
170 /* resample audio. 'nb_samples' is the number of input samples */ | |
171 /* XXX: optimize it ! */ | |
172 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) | |
173 { | |
174 int i, nb_samples1; | |
64 | 175 short *bufin[2]; |
176 short *bufout[2]; | |
0 | 177 short *buftmp2[2], *buftmp3[2]; |
64 | 178 int lenout; |
0 | 179 |
2109 | 180 if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) { |
0 | 181 /* nothing to do */ |
182 memcpy(output, input, nb_samples * s->input_channels * sizeof(short)); | |
183 return nb_samples; | |
184 } | |
185 | |
64 | 186 /* XXX: move those malloc to resample init code */ |
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187 for(i=0; i<s->filter_channels; i++){ |
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188 bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) ); |
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189 memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short)); |
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190 buftmp2[i] = bufin[i] + s->temp_len; |
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191 } |
2967 | 192 |
64 | 193 /* make some zoom to avoid round pb */ |
194 lenout= (int)(nb_samples * s->ratio) + 16; | |
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195 bufout[0]= (short*) av_malloc( lenout * sizeof(short) ); |
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196 bufout[1]= (short*) av_malloc( lenout * sizeof(short) ); |
64 | 197 |
0 | 198 if (s->input_channels == 2 && |
199 s->output_channels == 1) { | |
200 buftmp3[0] = output; | |
201 stereo_to_mono(buftmp2[0], input, nb_samples); | |
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202 } else if (s->output_channels >= 2 && s->input_channels == 1) { |
0 | 203 buftmp3[0] = bufout[0]; |
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204 memcpy(buftmp2[0], input, nb_samples*sizeof(short)); |
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205 } else if (s->output_channels >= 2) { |
0 | 206 buftmp3[0] = bufout[0]; |
207 buftmp3[1] = bufout[1]; | |
208 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples); | |
209 } else { | |
210 buftmp3[0] = output; | |
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211 memcpy(buftmp2[0], input, nb_samples*sizeof(short)); |
0 | 212 } |
213 | |
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214 nb_samples += s->temp_len; |
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215 |
0 | 216 /* resample each channel */ |
217 nb_samples1 = 0; /* avoid warning */ | |
218 for(i=0;i<s->filter_channels;i++) { | |
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219 int consumed; |
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220 int is_last= i+1 == s->filter_channels; |
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221 |
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222 nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last); |
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223 s->temp_len= nb_samples - consumed; |
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224 s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short)); |
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225 memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short)); |
0 | 226 } |
227 | |
228 if (s->output_channels == 2 && s->input_channels == 1) { | |
229 mono_to_stereo(output, buftmp3[0], nb_samples1); | |
230 } else if (s->output_channels == 2) { | |
231 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); | |
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232 } else if (s->output_channels == 6) { |
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233 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); |
0 | 234 } |
235 | |
2084 | 236 for(i=0; i<s->filter_channels; i++) |
237 av_free(bufin[i]); | |
64 | 238 |
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239 av_free(bufout[0]); |
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240 av_free(bufout[1]); |
0 | 241 return nb_samples1; |
242 } | |
243 | |
244 void audio_resample_close(ReSampleContext *s) | |
245 { | |
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246 av_resample_close(s->resample_context); |
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247 av_freep(&s->temp[0]); |
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248 av_freep(&s->temp[1]); |
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249 av_free(s); |
0 | 250 } |