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 * Sample rate convertion for both audio and video.
26 #include "os_support.h"
29 /* fractional resampling */
30 uint32_t incr; /* fractional increment */
33 /* integer down sample */
34 int iratio; /* integer divison ratio */
37 } ReSampleChannelContext;
39 struct ReSampleContext {
40 ReSampleChannelContext channel_ctx[2];
43 int input_channels, output_channels, filter_channels;
48 #define FRAC (1 << FRAC_BITS)
50 static void init_mono_resample(ReSampleChannelContext *s, float ratio)
53 s->iratio = (int)floorf(ratio);
56 s->incr = (int)((ratio / s->iratio) * FRAC);
59 s->icount = s->iratio;
61 s->inv = (FRAC / s->iratio);
64 /* fractional audio resampling */
65 static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
67 unsigned int frac, incr;
76 pend = input + nb_samples;
82 *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
83 frac = frac + s->incr;
84 while (frac >= FRAC) {
98 static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
104 pend = input + nb_samples;
113 *q++ = (sum * s->inv) >> FRAC_BITS;
125 /* n1: number of samples */
126 static void stereo_to_mono(short *output, short *input, int n1)
134 q[0] = (p[0] + p[1]) >> 1;
135 q[1] = (p[2] + p[3]) >> 1;
136 q[2] = (p[4] + p[5]) >> 1;
137 q[3] = (p[6] + p[7]) >> 1;
143 q[0] = (p[0] + p[1]) >> 1;
150 /* n1: number of samples */
151 static void mono_to_stereo(short *output, short *input, int n1)
160 v = p[0]; q[0] = v; q[1] = v;
161 v = p[1]; q[2] = v; q[3] = v;
162 v = p[2]; q[4] = v; q[5] = v;
163 v = p[3]; q[6] = v; q[7] = v;
169 v = p[0]; q[0] = v; q[1] = v;
176 /* XXX: should use more abstract 'N' channels system */
177 static void stereo_split(short *output1, short *output2, short *input, int n)
182 *output1++ = *input++;
183 *output2++ = *input++;
187 static void stereo_mux(short *output, short *input1, short *input2, int n)
192 *output++ = *input1++;
193 *output++ = *input2++;
197 static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
202 buf1= (short*)av_malloc( nb_samples * sizeof(short) );
204 /* first downsample by an integer factor with averaging filter */
207 nb_samples = integer_downsample(s, buftmp, input, nb_samples);
212 /* then do a fractional resampling with linear interpolation */
213 if (s->incr != FRAC) {
214 nb_samples = fractional_resample(s, output, buftmp, nb_samples);
216 memcpy(output, buftmp, nb_samples * sizeof(short));
222 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
223 int output_rate, int input_rate)
228 if (output_channels > 2 || input_channels > 2)
231 s = av_mallocz(sizeof(ReSampleContext));
235 s->ratio = (float)output_rate / (float)input_rate;
237 s->input_channels = input_channels;
238 s->output_channels = output_channels;
240 s->filter_channels = s->input_channels;
241 if (s->output_channels < s->filter_channels)
242 s->filter_channels = s->output_channels;
244 for(i=0;i<s->filter_channels;i++) {
245 init_mono_resample(&s->channel_ctx[i], s->ratio);
250 /* resample audio. 'nb_samples' is the number of input samples */
251 /* XXX: optimize it ! */
252 /* XXX: do it with polyphase filters, since the quality here is
253 HORRIBLE. Return the number of samples available in output */
254 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
259 short *buftmp2[2], *buftmp3[2];
262 if (s->input_channels == s->output_channels && s->ratio == 1.0) {
264 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
268 /* XXX: move those malloc to resample init code */
269 bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
270 bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
272 /* make some zoom to avoid round pb */
273 lenout= (int)(nb_samples * s->ratio) + 16;
274 bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
275 bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
277 if (s->input_channels == 2 &&
278 s->output_channels == 1) {
279 buftmp2[0] = bufin[0];
281 stereo_to_mono(buftmp2[0], input, nb_samples);
282 } else if (s->output_channels == 2 && s->input_channels == 1) {
284 buftmp3[0] = bufout[0];
285 } else if (s->output_channels == 2) {
286 buftmp2[0] = bufin[0];
287 buftmp2[1] = bufin[1];
288 buftmp3[0] = bufout[0];
289 buftmp3[1] = bufout[1];
290 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
296 /* resample each channel */
297 nb_samples1 = 0; /* avoid warning */
298 for(i=0;i<s->filter_channels;i++) {
299 nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
302 if (s->output_channels == 2 && s->input_channels == 1) {
303 mono_to_stereo(output, buftmp3[0], nb_samples1);
304 } else if (s->output_channels == 2) {
305 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
316 void audio_resample_close(ReSampleContext *s)