Mercurial > libavcodec.hg
annotate resample.c @ 2359:a3c029107a88 libavcodec
fixing rv20201002
author | michael |
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date | Fri, 19 Nov 2004 18:19:36 +0000 |
parents | de38526a1f3f |
children | e25782262d7d |
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 * |
429 | 5 * This library is free software; you can redistribute it and/or |
6 * modify it under the terms of the GNU Lesser General Public | |
7 * License as published by the Free Software Foundation; either | |
8 * version 2 of the License, or (at your option) any later version. | |
0 | 9 * |
429 | 10 * This library is distributed in the hope that it will be useful, |
0 | 11 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
429 | 12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
13 * Lesser General Public License for more details. | |
0 | 14 * |
429 | 15 * You should have received a copy of the GNU Lesser General Public |
16 * License along with this library; if not, write to the Free Software | |
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
0 | 18 */ |
1106 | 19 |
20 /** | |
21 * @file resample.c | |
22 * Sample rate convertion for both audio and video. | |
23 */ | |
24 | |
64 | 25 #include "avcodec.h" |
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26 |
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27 struct AVResampleContext; |
0 | 28 |
29 struct ReSampleContext { | |
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30 struct AVResampleContext *resample_context; |
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31 short *temp[2]; |
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32 int temp_len; |
0 | 33 float ratio; |
34 /* channel convert */ | |
35 int input_channels, output_channels, filter_channels; | |
36 }; | |
37 | |
38 /* n1: number of samples */ | |
39 static void stereo_to_mono(short *output, short *input, int n1) | |
40 { | |
41 short *p, *q; | |
42 int n = n1; | |
43 | |
44 p = input; | |
45 q = output; | |
46 while (n >= 4) { | |
47 q[0] = (p[0] + p[1]) >> 1; | |
48 q[1] = (p[2] + p[3]) >> 1; | |
49 q[2] = (p[4] + p[5]) >> 1; | |
50 q[3] = (p[6] + p[7]) >> 1; | |
51 q += 4; | |
52 p += 8; | |
53 n -= 4; | |
54 } | |
55 while (n > 0) { | |
56 q[0] = (p[0] + p[1]) >> 1; | |
57 q++; | |
58 p += 2; | |
59 n--; | |
60 } | |
61 } | |
62 | |
63 /* n1: number of samples */ | |
64 static void mono_to_stereo(short *output, short *input, int n1) | |
65 { | |
66 short *p, *q; | |
67 int n = n1; | |
68 int v; | |
69 | |
70 p = input; | |
71 q = output; | |
72 while (n >= 4) { | |
73 v = p[0]; q[0] = v; q[1] = v; | |
74 v = p[1]; q[2] = v; q[3] = v; | |
75 v = p[2]; q[4] = v; q[5] = v; | |
76 v = p[3]; q[6] = v; q[7] = v; | |
77 q += 8; | |
78 p += 4; | |
79 n -= 4; | |
80 } | |
81 while (n > 0) { | |
82 v = p[0]; q[0] = v; q[1] = v; | |
83 q += 2; | |
84 p += 1; | |
85 n--; | |
86 } | |
87 } | |
88 | |
89 /* XXX: should use more abstract 'N' channels system */ | |
90 static void stereo_split(short *output1, short *output2, short *input, int n) | |
91 { | |
92 int i; | |
93 | |
94 for(i=0;i<n;i++) { | |
95 *output1++ = *input++; | |
96 *output2++ = *input++; | |
97 } | |
98 } | |
99 | |
100 static void stereo_mux(short *output, short *input1, short *input2, int n) | |
101 { | |
102 int i; | |
103 | |
104 for(i=0;i<n;i++) { | |
105 *output++ = *input1++; | |
106 *output++ = *input2++; | |
107 } | |
108 } | |
109 | |
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110 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) |
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111 { |
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112 int i; |
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113 short l,r; |
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114 |
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115 for(i=0;i<n;i++) { |
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116 l=*input1++; |
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117 r=*input2++; |
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118 *output++ = l; /* left */ |
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119 *output++ = (l/2)+(r/2); /* center */ |
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120 *output++ = r; /* right */ |
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121 *output++ = 0; /* left surround */ |
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122 *output++ = 0; /* right surroud */ |
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123 *output++ = 0; /* low freq */ |
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124 } |
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125 } |
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126 |
0 | 127 ReSampleContext *audio_resample_init(int output_channels, int input_channels, |
128 int output_rate, int input_rate) | |
129 { | |
130 ReSampleContext *s; | |
131 int i; | |
132 | |
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133 if ( input_channels > 2) |
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134 { |
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135 av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported."); |
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136 return NULL; |
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137 } |
0 | 138 |
139 s = av_mallocz(sizeof(ReSampleContext)); | |
140 if (!s) | |
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141 { |
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142 av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context."); |
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143 return NULL; |
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144 } |
0 | 145 |
146 s->ratio = (float)output_rate / (float)input_rate; | |
147 | |
148 s->input_channels = input_channels; | |
149 s->output_channels = output_channels; | |
150 | |
151 s->filter_channels = s->input_channels; | |
152 if (s->output_channels < s->filter_channels) | |
153 s->filter_channels = s->output_channels; | |
154 | |
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155 /* |
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156 * ac3 output is the only case where filter_channels could be greater than 2. |
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157 * input channels can't be greater than 2, so resample the 2 channels and then |
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158 * expand to 6 channels after the resampling. |
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159 */ |
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160 if(s->filter_channels>2) |
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161 s->filter_channels = 2; |
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162 |
2308 | 163 s->resample_context= av_resample_init(output_rate, input_rate, 16, 10, 0, 1.0); |
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164 |
0 | 165 return s; |
166 } | |
167 | |
168 /* resample audio. 'nb_samples' is the number of input samples */ | |
169 /* XXX: optimize it ! */ | |
170 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) | |
171 { | |
172 int i, nb_samples1; | |
64 | 173 short *bufin[2]; |
174 short *bufout[2]; | |
0 | 175 short *buftmp2[2], *buftmp3[2]; |
64 | 176 int lenout; |
0 | 177 |
2109 | 178 if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) { |
0 | 179 /* nothing to do */ |
180 memcpy(output, input, nb_samples * s->input_channels * sizeof(short)); | |
181 return nb_samples; | |
182 } | |
183 | |
64 | 184 /* XXX: move those malloc to resample init code */ |
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185 for(i=0; i<s->filter_channels; i++){ |
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186 bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) ); |
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187 memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short)); |
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188 buftmp2[i] = bufin[i] + s->temp_len; |
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189 } |
64 | 190 |
191 /* make some zoom to avoid round pb */ | |
192 lenout= (int)(nb_samples * s->ratio) + 16; | |
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193 bufout[0]= (short*) av_malloc( lenout * sizeof(short) ); |
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194 bufout[1]= (short*) av_malloc( lenout * sizeof(short) ); |
64 | 195 |
0 | 196 if (s->input_channels == 2 && |
197 s->output_channels == 1) { | |
198 buftmp3[0] = output; | |
199 stereo_to_mono(buftmp2[0], input, nb_samples); | |
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200 } else if (s->output_channels >= 2 && s->input_channels == 1) { |
0 | 201 buftmp3[0] = bufout[0]; |
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202 memcpy(buftmp2[0], input, nb_samples*sizeof(short)); |
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203 } else if (s->output_channels >= 2) { |
0 | 204 buftmp3[0] = bufout[0]; |
205 buftmp3[1] = bufout[1]; | |
206 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples); | |
207 } else { | |
208 buftmp3[0] = output; | |
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209 memcpy(buftmp2[0], input, nb_samples*sizeof(short)); |
0 | 210 } |
211 | |
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212 nb_samples += s->temp_len; |
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213 |
0 | 214 /* resample each channel */ |
215 nb_samples1 = 0; /* avoid warning */ | |
216 for(i=0;i<s->filter_channels;i++) { | |
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217 int consumed; |
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218 int is_last= i+1 == s->filter_channels; |
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219 |
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220 nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last); |
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221 s->temp_len= nb_samples - consumed; |
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222 s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short)); |
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223 memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short)); |
0 | 224 } |
225 | |
226 if (s->output_channels == 2 && s->input_channels == 1) { | |
227 mono_to_stereo(output, buftmp3[0], nb_samples1); | |
228 } else if (s->output_channels == 2) { | |
229 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); | |
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230 } else if (s->output_channels == 6) { |
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231 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); |
0 | 232 } |
233 | |
2084 | 234 for(i=0; i<s->filter_channels; i++) |
235 av_free(bufin[i]); | |
64 | 236 |
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237 av_free(bufout[0]); |
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238 av_free(bufout[1]); |
0 | 239 return nb_samples1; |
240 } | |
241 | |
242 void audio_resample_close(ReSampleContext *s) | |
243 { | |
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244 av_resample_close(s->resample_context); |
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245 av_freep(&s->temp[0]); |
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246 av_freep(&s->temp[1]); |
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247 av_free(s); |
0 | 248 } |