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.
22 #include <netinet/in.h>
30 #define FRAC (1 << FRAC_BITS)
32 static void init_mono_resample(ReSampleChannelContext *s, float ratio)
35 s->iratio = (int)floor(ratio);
38 s->incr = (int)((ratio / s->iratio) * FRAC);
41 s->icount = s->iratio;
43 s->inv = (FRAC / s->iratio);
46 /* fractional audio resampling */
47 static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
49 unsigned int frac, incr;
58 pend = input + nb_samples;
64 *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
65 frac = frac + s->incr;
66 while (frac >= FRAC) {
80 static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
86 pend = input + nb_samples;
95 *q++ = (sum * s->inv) >> FRAC_BITS;
107 /* n1: number of samples */
108 static void stereo_to_mono(short *output, short *input, int n1)
116 q[0] = (p[0] + p[1]) >> 1;
117 q[1] = (p[2] + p[3]) >> 1;
118 q[2] = (p[4] + p[5]) >> 1;
119 q[3] = (p[6] + p[7]) >> 1;
125 q[0] = (p[0] + p[1]) >> 1;
132 /* XXX: should use more abstract 'N' channels system */
133 static void stereo_split(short *output1, short *output2, short *input, int n)
138 *output1++ = *input++;
139 *output2++ = *input++;
143 static void stereo_mux(short *output, short *input1, short *input2, int n)
148 *output++ = *input1++;
149 *output++ = *input2++;
153 static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
155 short buf1[nb_samples];
158 /* first downsample by an integer factor with averaging filter */
161 nb_samples = integer_downsample(s, buftmp, input, nb_samples);
166 /* then do a fractional resampling with linear interpolation */
167 if (s->incr != FRAC) {
168 nb_samples = fractional_resample(s, output, buftmp, nb_samples);
170 memcpy(output, buftmp, nb_samples * sizeof(short));
175 /* ratio = output_rate / input_rate */
176 int audio_resample_init(ReSampleContext *s,
177 int output_channels, int input_channels,
178 int output_rate, int input_rate)
182 s->ratio = (float)output_rate / (float)input_rate;
184 if (output_channels > 2 || input_channels > 2)
186 s->input_channels = input_channels;
187 s->output_channels = output_channels;
189 for(i=0;i<output_channels;i++) {
190 init_mono_resample(&s->channel_ctx[i], s->ratio);
195 /* resample audio. 'nb_samples' is the number of input samples */
196 /* XXX: optimize it ! */
197 /* XXX: do it with polyphase filters, since the quality here is
198 HORRIBLE. Return the number of samples available in output */
199 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
202 short buf[5][nb_samples];
203 short *buftmp1, *buftmp2[2], *buftmp3[2];
205 if (s->input_channels == s->output_channels && s->ratio == 1.0) {
207 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
211 if (s->input_channels == 2 &&
212 s->output_channels == 1) {
214 stereo_to_mono(buftmp1, input, nb_samples);
215 } else if (s->input_channels == 1 &&
216 s->output_channels == 2) {
223 if (s->output_channels == 2) {
228 stereo_split(buftmp2[0], buftmp2[1], buftmp1, nb_samples);
230 buftmp2[0] = buftmp1;
234 /* resample each channel */
235 nb_samples1 = 0; /* avoid warning */
236 for(i=0;i<s->output_channels;i++) {
237 nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
240 if (s->output_channels == 2) {
241 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);