2 * Sample rate convertion for both audio and video
3 * Copyright (c) 2000 Fabrice Bellard.
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24 * Sample rate convertion for both audio and video.
29 struct AVResampleContext;
31 struct ReSampleContext {
32 struct AVResampleContext *resample_context;
37 int input_channels, output_channels, filter_channels;
40 /* n1: number of samples */
41 static void stereo_to_mono(short *output, short *input, int n1)
49 q[0] = (p[0] + p[1]) >> 1;
50 q[1] = (p[2] + p[3]) >> 1;
51 q[2] = (p[4] + p[5]) >> 1;
52 q[3] = (p[6] + p[7]) >> 1;
58 q[0] = (p[0] + p[1]) >> 1;
65 /* n1: number of samples */
66 static void mono_to_stereo(short *output, short *input, int n1)
75 v = p[0]; q[0] = v; q[1] = v;
76 v = p[1]; q[2] = v; q[3] = v;
77 v = p[2]; q[4] = v; q[5] = v;
78 v = p[3]; q[6] = v; q[7] = v;
84 v = p[0]; q[0] = v; q[1] = v;
91 /* XXX: should use more abstract 'N' channels system */
92 static void stereo_split(short *output1, short *output2, short *input, int n)
97 *output1++ = *input++;
98 *output2++ = *input++;
102 static void stereo_mux(short *output, short *input1, short *input2, int n)
107 *output++ = *input1++;
108 *output++ = *input2++;
112 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
120 *output++ = l; /* left */
121 *output++ = (l/2)+(r/2); /* center */
122 *output++ = r; /* right */
123 *output++ = 0; /* left surround */
124 *output++ = 0; /* right surroud */
125 *output++ = 0; /* low freq */
129 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
130 int output_rate, int input_rate)
134 if ( input_channels > 2)
136 av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.");
140 s = av_mallocz(sizeof(ReSampleContext));
143 av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.");
147 s->ratio = (float)output_rate / (float)input_rate;
149 s->input_channels = input_channels;
150 s->output_channels = output_channels;
152 s->filter_channels = s->input_channels;
153 if (s->output_channels < s->filter_channels)
154 s->filter_channels = s->output_channels;
157 * ac3 output is the only case where filter_channels could be greater than 2.
158 * input channels can't be greater than 2, so resample the 2 channels and then
159 * expand to 6 channels after the resampling.
161 if(s->filter_channels>2)
162 s->filter_channels = 2;
164 s->resample_context= av_resample_init(output_rate, input_rate, 16, 10, 0, 1.0);
169 /* resample audio. 'nb_samples' is the number of input samples */
170 /* XXX: optimize it ! */
171 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
176 short *buftmp2[2], *buftmp3[2];
179 if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
181 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
185 /* XXX: move those malloc to resample init code */
186 for(i=0; i<s->filter_channels; i++){
187 bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
188 memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
189 buftmp2[i] = bufin[i] + s->temp_len;
192 /* make some zoom to avoid round pb */
193 lenout= (int)(nb_samples * s->ratio) + 16;
194 bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
195 bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
197 if (s->input_channels == 2 &&
198 s->output_channels == 1) {
200 stereo_to_mono(buftmp2[0], input, nb_samples);
201 } else if (s->output_channels >= 2 && s->input_channels == 1) {
202 buftmp3[0] = bufout[0];
203 memcpy(buftmp2[0], input, nb_samples*sizeof(short));
204 } else if (s->output_channels >= 2) {
205 buftmp3[0] = bufout[0];
206 buftmp3[1] = bufout[1];
207 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
210 memcpy(buftmp2[0], input, nb_samples*sizeof(short));
213 nb_samples += s->temp_len;
215 /* resample each channel */
216 nb_samples1 = 0; /* avoid warning */
217 for(i=0;i<s->filter_channels;i++) {
219 int is_last= i+1 == s->filter_channels;
221 nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
222 s->temp_len= nb_samples - consumed;
223 s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
224 memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
227 if (s->output_channels == 2 && s->input_channels == 1) {
228 mono_to_stereo(output, buftmp3[0], nb_samples1);
229 } else if (s->output_channels == 2) {
230 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
231 } else if (s->output_channels == 6) {
232 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
235 for(i=0; i<s->filter_channels; i++)
243 void audio_resample_close(ReSampleContext *s)
245 av_resample_close(s->resample_context);
246 av_freep(&s->temp[0]);
247 av_freep(&s->temp[1]);