polyphase kaiser windowed sinc and blackman nuttall windowed sinc audio resample...
[libav.git] / libavcodec / resample.c
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1/*
2 * Sample rate convertion for both audio and video
ff4ec49e 3 * Copyright (c) 2000 Fabrice Bellard.
de6d9b64 4 *
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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.
de6d9b64 9 *
ff4ec49e 10 * This library is distributed in the hope that it will be useful,
de6d9b64 11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Lesser General Public License for more details.
de6d9b64 14 *
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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
de6d9b64 18 */
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19
20/**
21 * @file resample.c
22 * Sample rate convertion for both audio and video.
23 */
24
de6d9b64 25#include "avcodec.h"
69db4e10 26
aaaf1635 27struct AVResampleContext;
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28
29struct ReSampleContext {
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30 struct AVResampleContext *resample_context;
31 short *temp[2];
32 int temp_len;
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33 float ratio;
34 /* channel convert */
35 int input_channels, output_channels, filter_channels;
36};
37
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38/* n1: number of samples */
39static 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 */
64static 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 */
90static 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
100static 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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110static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
111{
112 int i;
113 short l,r;
114
115 for(i=0;i<n;i++) {
116 l=*input1++;
117 r=*input2++;
118 *output++ = l; /* left */
119 *output++ = (l/2)+(r/2); /* center */
120 *output++ = r; /* right */
121 *output++ = 0; /* left surround */
122 *output++ = 0; /* right surroud */
123 *output++ = 0; /* low freq */
124 }
125}
126
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127ReSampleContext *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)
134 {
9b879566 135 av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.");
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136 return NULL;
137 }
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138
139 s = av_mallocz(sizeof(ReSampleContext));
140 if (!s)
743739d2 141 {
9b879566 142 av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.");
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143 return NULL;
144 }
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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/*
156 * ac3 output is the only case where filter_channels could be greater than 2.
157 * input channels can't be greater than 2, so resample the 2 channels and then
158 * expand to 6 channels after the resampling.
159 */
160 if(s->filter_channels>2)
161 s->filter_channels = 2;
162
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163 s->resample_context= av_resample_init(output_rate, input_rate);
164
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165 return s;
166}
167
168/* resample audio. 'nb_samples' is the number of input samples */
169/* XXX: optimize it ! */
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170int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
171{
172 int i, nb_samples1;
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173 short *bufin[2];
174 short *bufout[2];
de6d9b64 175 short *buftmp2[2], *buftmp3[2];
1a565432 176 int lenout;
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177
178 if (s->input_channels == s->output_channels && s->ratio == 1.0) {
179 /* nothing to do */
180 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
181 return nb_samples;
182 }
183
1a565432 184 /* XXX: move those malloc to resample init code */
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185 for(i=0; i<s->filter_channels; i++){
186 bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
187 memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
188 buftmp2[i] = bufin[i] + s->temp_len;
189 }
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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) );
194 bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
1a565432 195
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196 if (s->input_channels == 2 &&
197 s->output_channels == 1) {
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198 buftmp3[0] = output;
199 stereo_to_mono(buftmp2[0], input, nb_samples);
743739d2 200 } else if (s->output_channels >= 2 && s->input_channels == 1) {
de6d9b64 201 buftmp3[0] = bufout[0];
aaaf1635 202 memcpy(buftmp2[0], input, nb_samples*sizeof(short));
743739d2 203 } else if (s->output_channels >= 2) {
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204 buftmp3[0] = bufout[0];
205 buftmp3[1] = bufout[1];
206 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
207 } else {
de6d9b64 208 buftmp3[0] = output;
aaaf1635 209 memcpy(buftmp2[0], input, nb_samples*sizeof(short));
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210 }
211
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212 nb_samples += s->temp_len;
213
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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;
218 int is_last= i+1 == s->filter_channels;
219
220 nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
221 s->temp_len= nb_samples - consumed;
222 s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
223 memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
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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) {
231 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
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232 }
233
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234 av_free(bufin[0]);
235 av_free(bufin[1]);
1a565432 236
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237 av_free(bufout[0]);
238 av_free(bufout[1]);
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239 return nb_samples1;
240}
241
242void audio_resample_close(ReSampleContext *s)
243{
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244 av_resample_close(s->resample_context);
245 av_freep(&s->temp[0]);
246 av_freep(&s->temp[1]);
6000abfa 247 av_free(s);
de6d9b64 248}