2 * Sample rate convertion for both audio and video
3 * Copyright (c) 2000 Fabrice Bellard.
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.
10 * This library 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 GNU
13 * Lesser General Public License for more details.
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
22 /* fractional resampling */
23 UINT32 incr; /* fractional increment */
26 /* integer down sample */
27 int iratio; /* integer divison ratio */
30 } ReSampleChannelContext;
32 struct ReSampleContext {
33 ReSampleChannelContext channel_ctx[2];
36 int input_channels, output_channels, filter_channels;
41 #define FRAC (1 << FRAC_BITS)
43 static void init_mono_resample(ReSampleChannelContext *s, float ratio)
46 s->iratio = (int)floor(ratio);
49 s->incr = (int)((ratio / s->iratio) * FRAC);
52 s->icount = s->iratio;
54 s->inv = (FRAC / s->iratio);
57 /* fractional audio resampling */
58 static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
60 unsigned int frac, incr;
69 pend = input + nb_samples;
75 *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
76 frac = frac + s->incr;
77 while (frac >= FRAC) {
91 static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
97 pend = input + nb_samples;
106 *q++ = (sum * s->inv) >> FRAC_BITS;
118 /* n1: number of samples */
119 static void stereo_to_mono(short *output, short *input, int n1)
127 q[0] = (p[0] + p[1]) >> 1;
128 q[1] = (p[2] + p[3]) >> 1;
129 q[2] = (p[4] + p[5]) >> 1;
130 q[3] = (p[6] + p[7]) >> 1;
136 q[0] = (p[0] + p[1]) >> 1;
143 /* n1: number of samples */
144 static void mono_to_stereo(short *output, short *input, int n1)
153 v = p[0]; q[0] = v; q[1] = v;
154 v = p[1]; q[2] = v; q[3] = v;
155 v = p[2]; q[4] = v; q[5] = v;
156 v = p[3]; q[6] = v; q[7] = v;
162 v = p[0]; q[0] = v; q[1] = v;
169 /* XXX: should use more abstract 'N' channels system */
170 static void stereo_split(short *output1, short *output2, short *input, int n)
175 *output1++ = *input++;
176 *output2++ = *input++;
180 static void stereo_mux(short *output, short *input1, short *input2, int n)
185 *output++ = *input1++;
186 *output++ = *input2++;
190 static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
195 buf1= (short*)av_malloc( nb_samples * sizeof(short) );
197 /* first downsample by an integer factor with averaging filter */
200 nb_samples = integer_downsample(s, buftmp, input, nb_samples);
205 /* then do a fractional resampling with linear interpolation */
206 if (s->incr != FRAC) {
207 nb_samples = fractional_resample(s, output, buftmp, nb_samples);
209 memcpy(output, buftmp, nb_samples * sizeof(short));
215 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
216 int output_rate, int input_rate)
221 if (output_channels > 2 || input_channels > 2)
224 s = av_mallocz(sizeof(ReSampleContext));
228 s->ratio = (float)output_rate / (float)input_rate;
230 s->input_channels = input_channels;
231 s->output_channels = output_channels;
233 s->filter_channels = s->input_channels;
234 if (s->output_channels < s->filter_channels)
235 s->filter_channels = s->output_channels;
237 for(i=0;i<s->filter_channels;i++) {
238 init_mono_resample(&s->channel_ctx[i], s->ratio);
243 /* resample audio. 'nb_samples' is the number of input samples */
244 /* XXX: optimize it ! */
245 /* XXX: do it with polyphase filters, since the quality here is
246 HORRIBLE. Return the number of samples available in output */
247 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
252 short *buftmp2[2], *buftmp3[2];
255 if (s->input_channels == s->output_channels && s->ratio == 1.0) {
257 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
261 /* XXX: move those malloc to resample init code */
262 bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
263 bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
265 /* make some zoom to avoid round pb */
266 lenout= (int)(nb_samples * s->ratio) + 16;
267 bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
268 bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
270 if (s->input_channels == 2 &&
271 s->output_channels == 1) {
272 buftmp2[0] = bufin[0];
274 stereo_to_mono(buftmp2[0], input, nb_samples);
275 } else if (s->output_channels == 2 && s->input_channels == 1) {
277 buftmp3[0] = bufout[0];
278 } else if (s->output_channels == 2) {
279 buftmp2[0] = bufin[0];
280 buftmp2[1] = bufin[1];
281 buftmp3[0] = bufout[0];
282 buftmp3[1] = bufout[1];
283 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
289 /* resample each channel */
290 nb_samples1 = 0; /* avoid warning */
291 for(i=0;i<s->filter_channels;i++) {
292 nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
295 if (s->output_channels == 2 && s->input_channels == 1) {
296 mono_to_stereo(output, buftmp3[0], nb_samples1);
297 } else if (s->output_channels == 2) {
298 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
309 void audio_resample_close(ReSampleContext *s)