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.
27 struct AVResampleContext;
29 struct ReSampleContext {
30 struct AVResampleContext *resample_context;
35 int input_channels, output_channels, filter_channels;
38 /* n1: number of samples */
39 static void stereo_to_mono(short *output, short *input, int n1)
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;
56 q[0] = (p[0] + p[1]) >> 1;
63 /* n1: number of samples */
64 static void mono_to_stereo(short *output, short *input, int n1)
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;
82 v = p[0]; q[0] = v; q[1] = v;
89 /* XXX: should use more abstract 'N' channels system */
90 static void stereo_split(short *output1, short *output2, short *input, int n)
95 *output1++ = *input++;
96 *output2++ = *input++;
100 static void stereo_mux(short *output, short *input1, short *input2, int n)
105 *output++ = *input1++;
106 *output++ = *input2++;
110 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
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 */
127 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
128 int output_rate, int input_rate)
132 if ( input_channels > 2)
134 av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.");
138 s = av_mallocz(sizeof(ReSampleContext));
141 av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.");
145 s->ratio = (float)output_rate / (float)input_rate;
147 s->input_channels = input_channels;
148 s->output_channels = output_channels;
150 s->filter_channels = s->input_channels;
151 if (s->output_channels < s->filter_channels)
152 s->filter_channels = s->output_channels;
155 * ac3 output is the only case where filter_channels could be greater than 2.
156 * input channels can't be greater than 2, so resample the 2 channels and then
157 * expand to 6 channels after the resampling.
159 if(s->filter_channels>2)
160 s->filter_channels = 2;
162 s->resample_context= av_resample_init(output_rate, input_rate, 16, 10, 0, 1.0);
167 /* resample audio. 'nb_samples' is the number of input samples */
168 /* XXX: optimize it ! */
169 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
174 short *buftmp2[2], *buftmp3[2];
177 if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
179 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
183 /* XXX: move those malloc to resample init code */
184 for(i=0; i<s->filter_channels; i++){
185 bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
186 memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
187 buftmp2[i] = bufin[i] + s->temp_len;
190 /* make some zoom to avoid round pb */
191 lenout= (int)(nb_samples * s->ratio) + 16;
192 bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
193 bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
195 if (s->input_channels == 2 &&
196 s->output_channels == 1) {
198 stereo_to_mono(buftmp2[0], input, nb_samples);
199 } else if (s->output_channels >= 2 && s->input_channels == 1) {
200 buftmp3[0] = bufout[0];
201 memcpy(buftmp2[0], input, nb_samples*sizeof(short));
202 } else if (s->output_channels >= 2) {
203 buftmp3[0] = bufout[0];
204 buftmp3[1] = bufout[1];
205 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
208 memcpy(buftmp2[0], input, nb_samples*sizeof(short));
211 nb_samples += s->temp_len;
213 /* resample each channel */
214 nb_samples1 = 0; /* avoid warning */
215 for(i=0;i<s->filter_channels;i++) {
217 int is_last= i+1 == s->filter_channels;
219 nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
220 s->temp_len= nb_samples - consumed;
221 s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
222 memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
225 if (s->output_channels == 2 && s->input_channels == 1) {
226 mono_to_stereo(output, buftmp3[0], nb_samples1);
227 } else if (s->output_channels == 2) {
228 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
229 } else if (s->output_channels == 6) {
230 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
233 for(i=0; i<s->filter_channels; i++)
241 void audio_resample_close(ReSampleContext *s)
243 av_resample_close(s->resample_context);
244 av_freep(&s->temp[0]);
245 av_freep(&s->temp[1]);