2 * samplerate conversion for both audio and video
3 * Copyright (c) 2000 Fabrice Bellard
5 * This file is part of Libav.
7 * Libav 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 * Libav 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 Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24 * samplerate conversion for both audio and video
28 #include "audioconvert.h"
29 #include "libavutil/opt.h"
30 #include "libavutil/samplefmt.h"
32 struct AVResampleContext;
34 static const char *context_to_name(void *ptr)
36 return "audioresample";
39 static const AVOption options[] = {{NULL}};
40 static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT };
42 struct ReSampleContext {
43 struct AVResampleContext *resample_context;
48 int input_channels, output_channels, filter_channels;
49 AVAudioConvert *convert_ctx[2];
50 enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
51 unsigned sample_size[2]; ///< size of one sample in sample_fmt
52 short *buffer[2]; ///< buffers used for conversion to S16
53 unsigned buffer_size[2]; ///< sizes of allocated buffers
56 /* n1: number of samples */
57 static void stereo_to_mono(short *output, short *input, int n1)
65 q[0] = (p[0] + p[1]) >> 1;
66 q[1] = (p[2] + p[3]) >> 1;
67 q[2] = (p[4] + p[5]) >> 1;
68 q[3] = (p[6] + p[7]) >> 1;
74 q[0] = (p[0] + p[1]) >> 1;
81 /* n1: number of samples */
82 static void mono_to_stereo(short *output, short *input, int n1)
91 v = p[0]; q[0] = v; q[1] = v;
92 v = p[1]; q[2] = v; q[3] = v;
93 v = p[2]; q[4] = v; q[5] = v;
94 v = p[3]; q[6] = v; q[7] = v;
100 v = p[0]; q[0] = v; q[1] = v;
107 /* XXX: should use more abstract 'N' channels system */
108 static void stereo_split(short *output1, short *output2, short *input, int n)
113 *output1++ = *input++;
114 *output2++ = *input++;
118 static void stereo_mux(short *output, short *input1, short *input2, int n)
123 *output++ = *input1++;
124 *output++ = *input2++;
128 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
136 *output++ = l; /* left */
137 *output++ = (l/2)+(r/2); /* center */
138 *output++ = r; /* right */
139 *output++ = 0; /* left surround */
140 *output++ = 0; /* right surroud */
141 *output++ = 0; /* low freq */
145 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
146 int output_rate, int input_rate,
147 enum AVSampleFormat sample_fmt_out,
148 enum AVSampleFormat sample_fmt_in,
149 int filter_length, int log2_phase_count,
150 int linear, double cutoff)
154 if ( input_channels > 2)
156 av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n");
160 s = av_mallocz(sizeof(ReSampleContext));
163 av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
167 s->ratio = (float)output_rate / (float)input_rate;
169 s->input_channels = input_channels;
170 s->output_channels = output_channels;
172 s->filter_channels = s->input_channels;
173 if (s->output_channels < s->filter_channels)
174 s->filter_channels = s->output_channels;
176 s->sample_fmt [0] = sample_fmt_in;
177 s->sample_fmt [1] = sample_fmt_out;
178 s->sample_size[0] = av_get_bits_per_sample_fmt(s->sample_fmt[0])>>3;
179 s->sample_size[1] = av_get_bits_per_sample_fmt(s->sample_fmt[1])>>3;
181 if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
182 if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
183 s->sample_fmt[0], 1, NULL, 0))) {
184 av_log(s, AV_LOG_ERROR,
185 "Cannot convert %s sample format to s16 sample format\n",
186 av_get_sample_fmt_name(s->sample_fmt[0]));
192 if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
193 if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
194 AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
195 av_log(s, AV_LOG_ERROR,
196 "Cannot convert s16 sample format to %s sample format\n",
197 av_get_sample_fmt_name(s->sample_fmt[1]));
198 av_audio_convert_free(s->convert_ctx[0]);
205 * AC-3 output is the only case where filter_channels could be greater than 2.
206 * input channels can't be greater than 2, so resample the 2 channels and then
207 * expand to 6 channels after the resampling.
209 if(s->filter_channels>2)
210 s->filter_channels = 2;
213 s->resample_context= av_resample_init(output_rate, input_rate,
214 filter_length, log2_phase_count, linear, cutoff);
216 *(const AVClass**)s->resample_context = &audioresample_context_class;
221 /* resample audio. 'nb_samples' is the number of input samples */
222 /* XXX: optimize it ! */
223 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
228 short *buftmp2[2], *buftmp3[2];
229 short *output_bak = NULL;
232 if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
234 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
238 if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
239 int istride[1] = { s->sample_size[0] };
240 int ostride[1] = { 2 };
241 const void *ibuf[1] = { input };
243 unsigned input_size = nb_samples*s->input_channels*2;
245 if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
246 av_free(s->buffer[0]);
247 s->buffer_size[0] = input_size;
248 s->buffer[0] = av_malloc(s->buffer_size[0]);
250 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
255 obuf[0] = s->buffer[0];
257 if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
258 ibuf, istride, nb_samples*s->input_channels) < 0) {
259 av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format conversion failed\n");
263 input = s->buffer[0];
266 lenout= 4*nb_samples * s->ratio + 16;
268 if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
271 if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
272 av_free(s->buffer[1]);
273 s->buffer_size[1] = lenout;
274 s->buffer[1] = av_malloc(s->buffer_size[1]);
276 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
281 output = s->buffer[1];
284 /* XXX: move those malloc to resample init code */
285 for(i=0; i<s->filter_channels; i++){
286 bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
287 memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
288 buftmp2[i] = bufin[i] + s->temp_len;
291 /* make some zoom to avoid round pb */
292 bufout[0]= av_malloc( lenout * sizeof(short) );
293 bufout[1]= av_malloc( lenout * sizeof(short) );
295 if (s->input_channels == 2 &&
296 s->output_channels == 1) {
298 stereo_to_mono(buftmp2[0], input, nb_samples);
299 } else if (s->output_channels >= 2 && s->input_channels == 1) {
300 buftmp3[0] = bufout[0];
301 memcpy(buftmp2[0], input, nb_samples*sizeof(short));
302 } else if (s->output_channels >= 2) {
303 buftmp3[0] = bufout[0];
304 buftmp3[1] = bufout[1];
305 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
308 memcpy(buftmp2[0], input, nb_samples*sizeof(short));
311 nb_samples += s->temp_len;
313 /* resample each channel */
314 nb_samples1 = 0; /* avoid warning */
315 for(i=0;i<s->filter_channels;i++) {
317 int is_last= i+1 == s->filter_channels;
319 nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
320 s->temp_len= nb_samples - consumed;
321 s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
322 memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
325 if (s->output_channels == 2 && s->input_channels == 1) {
326 mono_to_stereo(output, buftmp3[0], nb_samples1);
327 } else if (s->output_channels == 2) {
328 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
329 } else if (s->output_channels == 6) {
330 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
333 if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
334 int istride[1] = { 2 };
335 int ostride[1] = { s->sample_size[1] };
336 const void *ibuf[1] = { output };
337 void *obuf[1] = { output_bak };
339 if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
340 ibuf, istride, nb_samples1*s->output_channels) < 0) {
341 av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format convertion failed\n");
346 for(i=0; i<s->filter_channels; i++)
354 void audio_resample_close(ReSampleContext *s)
356 av_resample_close(s->resample_context);
357 av_freep(&s->temp[0]);
358 av_freep(&s->temp[1]);
359 av_freep(&s->buffer[0]);
360 av_freep(&s->buffer[1]);
361 av_audio_convert_free(s->convert_ctx[0]);
362 av_audio_convert_free(s->convert_ctx[1]);