2 * Sample rate convertion for both audio and video
3 * Copyright (c) 2000 Gerard Lantau.
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; either version 2 of the License, or
8 * (at your option) any later version.
10 * This program is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 * GNU General Public License for more details.
15 * You should have received a copy of the GNU General Public License
16 * along with this program; if not, write to the Free Software
17 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
23 /* fractional resampling */
24 UINT32 incr; /* fractional increment */
27 /* integer down sample */
28 int iratio; /* integer divison ratio */
31 } ReSampleChannelContext;
33 struct ReSampleContext {
34 ReSampleChannelContext channel_ctx[2];
37 int input_channels, output_channels, filter_channels;
42 #define FRAC (1 << FRAC_BITS)
44 static void init_mono_resample(ReSampleChannelContext *s, float ratio)
47 s->iratio = (int)floor(ratio);
50 s->incr = (int)((ratio / s->iratio) * FRAC);
53 s->icount = s->iratio;
55 s->inv = (FRAC / s->iratio);
58 /* fractional audio resampling */
59 static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
61 unsigned int frac, incr;
70 pend = input + nb_samples;
76 *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
77 frac = frac + s->incr;
78 while (frac >= FRAC) {
92 static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
98 pend = input + nb_samples;
107 *q++ = (sum * s->inv) >> FRAC_BITS;
119 /* n1: number of samples */
120 static void stereo_to_mono(short *output, short *input, int n1)
128 q[0] = (p[0] + p[1]) >> 1;
129 q[1] = (p[2] + p[3]) >> 1;
130 q[2] = (p[4] + p[5]) >> 1;
131 q[3] = (p[6] + p[7]) >> 1;
137 q[0] = (p[0] + p[1]) >> 1;
144 /* n1: number of samples */
145 static void mono_to_stereo(short *output, short *input, int n1)
154 v = p[0]; q[0] = v; q[1] = v;
155 v = p[1]; q[2] = v; q[3] = v;
156 v = p[2]; q[4] = v; q[5] = v;
157 v = p[3]; q[6] = v; q[7] = v;
163 v = p[0]; q[0] = v; q[1] = v;
170 /* XXX: should use more abstract 'N' channels system */
171 static void stereo_split(short *output1, short *output2, short *input, int n)
176 *output1++ = *input++;
177 *output2++ = *input++;
181 static void stereo_mux(short *output, short *input1, short *input2, int n)
186 *output++ = *input1++;
187 *output++ = *input2++;
191 static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
196 buf1= (short*) malloc( nb_samples * sizeof(short) );
198 /* first downsample by an integer factor with averaging filter */
201 nb_samples = integer_downsample(s, buftmp, input, nb_samples);
206 /* then do a fractional resampling with linear interpolation */
207 if (s->incr != FRAC) {
208 nb_samples = fractional_resample(s, output, buftmp, nb_samples);
210 memcpy(output, buftmp, nb_samples * sizeof(short));
216 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
217 int output_rate, int input_rate)
222 if (output_channels > 2 || input_channels > 2)
225 s = av_mallocz(sizeof(ReSampleContext));
229 s->ratio = (float)output_rate / (float)input_rate;
231 s->input_channels = input_channels;
232 s->output_channels = output_channels;
234 s->filter_channels = s->input_channels;
235 if (s->output_channels < s->filter_channels)
236 s->filter_channels = s->output_channels;
238 for(i=0;i<s->filter_channels;i++) {
239 init_mono_resample(&s->channel_ctx[i], s->ratio);
244 /* resample audio. 'nb_samples' is the number of input samples */
245 /* XXX: optimize it ! */
246 /* XXX: do it with polyphase filters, since the quality here is
247 HORRIBLE. Return the number of samples available in output */
248 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
253 short *buftmp2[2], *buftmp3[2];
256 if (s->input_channels == s->output_channels && s->ratio == 1.0) {
258 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
262 /* XXX: move those malloc to resample init code */
263 bufin[0]= (short*) malloc( nb_samples * sizeof(short) );
264 bufin[1]= (short*) malloc( nb_samples * sizeof(short) );
266 /* make some zoom to avoid round pb */
267 lenout= (int)(nb_samples * s->ratio) + 16;
268 bufout[0]= (short*) malloc( lenout * sizeof(short) );
269 bufout[1]= (short*) malloc( lenout * sizeof(short) );
271 if (s->input_channels == 2 &&
272 s->output_channels == 1) {
273 buftmp2[0] = bufin[0];
275 stereo_to_mono(buftmp2[0], input, nb_samples);
276 } else if (s->output_channels == 2 && s->input_channels == 1) {
278 buftmp3[0] = bufout[0];
279 } else if (s->output_channels == 2) {
280 buftmp2[0] = bufin[0];
281 buftmp2[1] = bufin[1];
282 buftmp3[0] = bufout[0];
283 buftmp3[1] = bufout[1];
284 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
290 /* resample each channel */
291 nb_samples1 = 0; /* avoid warning */
292 for(i=0;i<s->filter_channels;i++) {
293 nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
296 if (s->output_channels == 2 && s->input_channels == 1) {
297 mono_to_stereo(output, buftmp3[0], nb_samples1);
298 } else if (s->output_channels == 2) {
299 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
310 void audio_resample_close(ReSampleContext *s)