Mercurial > libavcodec.hg
annotate resample.c @ 10720:bcc62edd9b41 libavcodec
16-bit Interplay video does not need palette control
author | kostya |
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date | Sun, 27 Dec 2009 14:07:21 +0000 |
parents | ff9ae6249d18 |
children | 7dd2a45249a9 |
rev | line source |
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0 | 1 /* |
5355 | 2 * samplerate conversion for both audio and video |
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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 /** | |
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23 * @file libavcodec/resample.c |
5355 | 24 * samplerate conversion for both audio and video |
1106 | 25 */ |
26 | |
64 | 27 #include "avcodec.h" |
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28 #include "audioconvert.h" |
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29 #include "opt.h" |
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30 |
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31 struct AVResampleContext; |
0 | 32 |
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33 static const char *context_to_name(void *ptr) |
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34 { |
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35 return "audioresample"; |
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36 } |
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37 |
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38 static const AVOption options[] = {{NULL}}; |
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39 static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options }; |
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40 |
0 | 41 struct ReSampleContext { |
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42 struct AVResampleContext *resample_context; |
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43 short *temp[2]; |
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44 int temp_len; |
0 | 45 float ratio; |
46 /* channel convert */ | |
47 int input_channels, output_channels, filter_channels; | |
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48 AVAudioConvert *convert_ctx[2]; |
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49 enum SampleFormat sample_fmt[2]; ///< input and output sample format |
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50 unsigned sample_size[2]; ///< size of one sample in sample_fmt |
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51 short *buffer[2]; ///< buffers used for conversion to S16 |
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52 unsigned buffer_size[2]; ///< sizes of allocated buffers |
0 | 53 }; |
54 | |
55 /* n1: number of samples */ | |
56 static void stereo_to_mono(short *output, short *input, int n1) | |
57 { | |
58 short *p, *q; | |
59 int n = n1; | |
60 | |
61 p = input; | |
62 q = output; | |
63 while (n >= 4) { | |
64 q[0] = (p[0] + p[1]) >> 1; | |
65 q[1] = (p[2] + p[3]) >> 1; | |
66 q[2] = (p[4] + p[5]) >> 1; | |
67 q[3] = (p[6] + p[7]) >> 1; | |
68 q += 4; | |
69 p += 8; | |
70 n -= 4; | |
71 } | |
72 while (n > 0) { | |
73 q[0] = (p[0] + p[1]) >> 1; | |
74 q++; | |
75 p += 2; | |
76 n--; | |
77 } | |
78 } | |
79 | |
80 /* n1: number of samples */ | |
81 static void mono_to_stereo(short *output, short *input, int n1) | |
82 { | |
83 short *p, *q; | |
84 int n = n1; | |
85 int v; | |
86 | |
87 p = input; | |
88 q = output; | |
89 while (n >= 4) { | |
90 v = p[0]; q[0] = v; q[1] = v; | |
91 v = p[1]; q[2] = v; q[3] = v; | |
92 v = p[2]; q[4] = v; q[5] = v; | |
93 v = p[3]; q[6] = v; q[7] = v; | |
94 q += 8; | |
95 p += 4; | |
96 n -= 4; | |
97 } | |
98 while (n > 0) { | |
99 v = p[0]; q[0] = v; q[1] = v; | |
100 q += 2; | |
101 p += 1; | |
102 n--; | |
103 } | |
104 } | |
105 | |
106 /* XXX: should use more abstract 'N' channels system */ | |
107 static void stereo_split(short *output1, short *output2, short *input, int n) | |
108 { | |
109 int i; | |
110 | |
111 for(i=0;i<n;i++) { | |
112 *output1++ = *input++; | |
113 *output2++ = *input++; | |
114 } | |
115 } | |
116 | |
117 static void stereo_mux(short *output, short *input1, short *input2, int n) | |
118 { | |
119 int i; | |
120 | |
121 for(i=0;i<n;i++) { | |
122 *output++ = *input1++; | |
123 *output++ = *input2++; | |
124 } | |
125 } | |
126 | |
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127 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) |
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128 { |
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129 int i; |
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130 short l,r; |
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131 |
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132 for(i=0;i<n;i++) { |
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133 l=*input1++; |
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134 r=*input2++; |
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135 *output++ = l; /* left */ |
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136 *output++ = (l/2)+(r/2); /* center */ |
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137 *output++ = r; /* right */ |
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138 *output++ = 0; /* left surround */ |
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139 *output++ = 0; /* right surroud */ |
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140 *output++ = 0; /* low freq */ |
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141 } |
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142 } |
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143 |
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144 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, |
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145 int output_rate, int input_rate, |
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146 enum SampleFormat sample_fmt_out, |
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147 enum SampleFormat sample_fmt_in, |
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148 int filter_length, int log2_phase_count, |
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149 int linear, double cutoff) |
0 | 150 { |
151 ReSampleContext *s; | |
2967 | 152 |
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153 if ( input_channels > 2) |
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154 { |
6023 | 155 av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n"); |
2979 | 156 return NULL; |
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157 } |
0 | 158 |
159 s = av_mallocz(sizeof(ReSampleContext)); | |
160 if (!s) | |
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161 { |
6023 | 162 av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n"); |
2979 | 163 return NULL; |
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164 } |
0 | 165 |
166 s->ratio = (float)output_rate / (float)input_rate; | |
2967 | 167 |
0 | 168 s->input_channels = input_channels; |
169 s->output_channels = output_channels; | |
2967 | 170 |
0 | 171 s->filter_channels = s->input_channels; |
172 if (s->output_channels < s->filter_channels) | |
173 s->filter_channels = s->output_channels; | |
174 | |
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175 s->sample_fmt [0] = sample_fmt_in; |
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176 s->sample_fmt [1] = sample_fmt_out; |
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177 s->sample_size[0] = av_get_bits_per_sample_format(s->sample_fmt[0])>>3; |
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178 s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3; |
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179 |
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180 if (s->sample_fmt[0] != SAMPLE_FMT_S16) { |
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181 if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1, |
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182 s->sample_fmt[0], 1, NULL, 0))) { |
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183 av_log(s, AV_LOG_ERROR, |
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184 "Cannot convert %s sample format to s16 sample format\n", |
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185 avcodec_get_sample_fmt_name(s->sample_fmt[0])); |
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186 av_free(s); |
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187 return NULL; |
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188 } |
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189 } |
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190 |
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191 if (s->sample_fmt[1] != SAMPLE_FMT_S16) { |
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192 if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1, |
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193 SAMPLE_FMT_S16, 1, NULL, 0))) { |
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194 av_log(s, AV_LOG_ERROR, |
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195 "Cannot convert s16 sample format to %s sample format\n", |
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196 avcodec_get_sample_fmt_name(s->sample_fmt[1])); |
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197 av_audio_convert_free(s->convert_ctx[0]); |
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198 av_free(s); |
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199 return NULL; |
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200 } |
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201 } |
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202 |
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203 /* |
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204 * AC-3 output is the only case where filter_channels could be greater than 2. |
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205 * input channels can't be greater than 2, so resample the 2 channels and then |
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206 * expand to 6 channels after the resampling. |
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207 */ |
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208 if(s->filter_channels>2) |
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209 s->filter_channels = 2; |
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210 |
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211 #define TAPS 16 |
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212 s->resample_context= av_resample_init(output_rate, input_rate, |
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213 filter_length, log2_phase_count, linear, cutoff); |
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214 |
9835 | 215 *(const AVClass**)s->resample_context = &audioresample_context_class; |
2967 | 216 |
0 | 217 return s; |
218 } | |
219 | |
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220 #if LIBAVCODEC_VERSION_MAJOR < 53 |
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221 ReSampleContext *audio_resample_init(int output_channels, int input_channels, |
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222 int output_rate, int input_rate) |
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223 { |
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224 return av_audio_resample_init(output_channels, input_channels, |
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225 output_rate, input_rate, |
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226 SAMPLE_FMT_S16, SAMPLE_FMT_S16, |
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227 TAPS, 10, 0, 0.8); |
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228 } |
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229 #endif |
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230 |
0 | 231 /* resample audio. 'nb_samples' is the number of input samples */ |
232 /* XXX: optimize it ! */ | |
233 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) | |
234 { | |
235 int i, nb_samples1; | |
64 | 236 short *bufin[2]; |
237 short *bufout[2]; | |
0 | 238 short *buftmp2[2], *buftmp3[2]; |
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239 short *output_bak = NULL; |
64 | 240 int lenout; |
0 | 241 |
2109 | 242 if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) { |
0 | 243 /* nothing to do */ |
244 memcpy(output, input, nb_samples * s->input_channels * sizeof(short)); | |
245 return nb_samples; | |
246 } | |
247 | |
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248 if (s->sample_fmt[0] != SAMPLE_FMT_S16) { |
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249 int istride[1] = { s->sample_size[0] }; |
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250 int ostride[1] = { 2 }; |
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251 const void *ibuf[1] = { input }; |
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252 void *obuf[1]; |
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253 unsigned input_size = nb_samples*s->input_channels*2; |
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254 |
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255 if (!s->buffer_size[0] || s->buffer_size[0] < input_size) { |
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256 av_free(s->buffer[0]); |
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257 s->buffer_size[0] = input_size; |
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258 s->buffer[0] = av_malloc(s->buffer_size[0]); |
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259 if (!s->buffer[0]) { |
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260 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n"); |
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261 return 0; |
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262 } |
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263 } |
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264 |
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265 obuf[0] = s->buffer[0]; |
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266 |
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267 if (av_audio_convert(s->convert_ctx[0], obuf, ostride, |
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268 ibuf, istride, nb_samples*s->input_channels) < 0) { |
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269 av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format conversion failed\n"); |
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270 return 0; |
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271 } |
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272 |
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273 input = s->buffer[0]; |
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274 } |
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275 |
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276 lenout= 4*nb_samples * s->ratio + 16; |
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277 |
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278 if (s->sample_fmt[1] != SAMPLE_FMT_S16) { |
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279 output_bak = output; |
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280 |
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281 if (!s->buffer_size[1] || s->buffer_size[1] < lenout) { |
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282 av_free(s->buffer[1]); |
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283 s->buffer_size[1] = lenout; |
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284 s->buffer[1] = av_malloc(s->buffer_size[1]); |
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285 if (!s->buffer[1]) { |
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286 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n"); |
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287 return 0; |
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288 } |
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289 } |
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290 |
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291 output = s->buffer[1]; |
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292 } |
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293 |
64 | 294 /* XXX: move those malloc to resample init code */ |
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295 for(i=0; i<s->filter_channels; i++){ |
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296 bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) ); |
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297 memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short)); |
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298 buftmp2[i] = bufin[i] + s->temp_len; |
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299 } |
2967 | 300 |
64 | 301 /* make some zoom to avoid round pb */ |
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302 bufout[0]= av_malloc( lenout * sizeof(short) ); |
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303 bufout[1]= av_malloc( lenout * sizeof(short) ); |
64 | 304 |
0 | 305 if (s->input_channels == 2 && |
306 s->output_channels == 1) { | |
307 buftmp3[0] = output; | |
308 stereo_to_mono(buftmp2[0], input, nb_samples); | |
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309 } else if (s->output_channels >= 2 && s->input_channels == 1) { |
0 | 310 buftmp3[0] = bufout[0]; |
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311 memcpy(buftmp2[0], input, nb_samples*sizeof(short)); |
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312 } else if (s->output_channels >= 2) { |
0 | 313 buftmp3[0] = bufout[0]; |
314 buftmp3[1] = bufout[1]; | |
315 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples); | |
316 } else { | |
317 buftmp3[0] = output; | |
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318 memcpy(buftmp2[0], input, nb_samples*sizeof(short)); |
0 | 319 } |
320 | |
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321 nb_samples += s->temp_len; |
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322 |
0 | 323 /* resample each channel */ |
324 nb_samples1 = 0; /* avoid warning */ | |
325 for(i=0;i<s->filter_channels;i++) { | |
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326 int consumed; |
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327 int is_last= i+1 == s->filter_channels; |
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328 |
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329 nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last); |
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330 s->temp_len= nb_samples - consumed; |
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331 s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short)); |
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332 memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short)); |
0 | 333 } |
334 | |
335 if (s->output_channels == 2 && s->input_channels == 1) { | |
336 mono_to_stereo(output, buftmp3[0], nb_samples1); | |
337 } else if (s->output_channels == 2) { | |
338 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); | |
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339 } else if (s->output_channels == 6) { |
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340 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); |
0 | 341 } |
342 | |
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343 if (s->sample_fmt[1] != SAMPLE_FMT_S16) { |
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344 int istride[1] = { 2 }; |
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345 int ostride[1] = { s->sample_size[1] }; |
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346 const void *ibuf[1] = { output }; |
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347 void *obuf[1] = { output_bak }; |
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348 |
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349 if (av_audio_convert(s->convert_ctx[1], obuf, ostride, |
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350 ibuf, istride, nb_samples1*s->output_channels) < 0) { |
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351 av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format convertion failed\n"); |
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352 return 0; |
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353 } |
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354 } |
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355 |
2084 | 356 for(i=0; i<s->filter_channels; i++) |
357 av_free(bufin[i]); | |
64 | 358 |
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359 av_free(bufout[0]); |
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360 av_free(bufout[1]); |
0 | 361 return nb_samples1; |
362 } | |
363 | |
364 void audio_resample_close(ReSampleContext *s) | |
365 { | |
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366 av_resample_close(s->resample_context); |
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367 av_freep(&s->temp[0]); |
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368 av_freep(&s->temp[1]); |
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369 av_freep(&s->buffer[0]); |
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370 av_freep(&s->buffer[1]); |
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371 av_audio_convert_free(s->convert_ctx[0]); |
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372 av_audio_convert_free(s->convert_ctx[1]); |
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373 av_free(s); |
0 | 374 } |