2
* samplerate conversion 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
* samplerate conversion for both audio and video
28
#include "audioconvert.h"
31
struct AVResampleContext;
33
static const char *context_to_name(void *ptr)
35
return "audioresample";
38
static const AVOption options[] = {{NULL}};
39
static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT };
41
struct ReSampleContext {
42
struct AVResampleContext *resample_context;
47
int input_channels, output_channels, filter_channels;
48
AVAudioConvert *convert_ctx[2];
49
enum SampleFormat sample_fmt[2]; ///< input and output sample format
50
unsigned sample_size[2]; ///< size of one sample in sample_fmt
51
short *buffer[2]; ///< buffers used for conversion to S16
52
unsigned buffer_size[2]; ///< sizes of allocated buffers
55
/* n1: number of samples */
56
static void stereo_to_mono(short *output, short *input, int n1)
64
q[0] = (p[0] + p[1]) >> 1;
65
q[1] = (p[2] + p[3]) >> 1;
66
q[2] = (p[4] + p[5]) >> 1;
67
q[3] = (p[6] + p[7]) >> 1;
73
q[0] = (p[0] + p[1]) >> 1;
80
/* n1: number of samples */
81
static void mono_to_stereo(short *output, short *input, int n1)
90
v = p[0]; q[0] = v; q[1] = v;
91
v = p[1]; q[2] = v; q[3] = v;
92
v = p[2]; q[4] = v; q[5] = v;
93
v = p[3]; q[6] = v; q[7] = v;
99
v = p[0]; q[0] = v; q[1] = v;
106
/* XXX: should use more abstract 'N' channels system */
107
static void stereo_split(short *output1, short *output2, short *input, int n)
112
*output1++ = *input++;
113
*output2++ = *input++;
117
static void stereo_mux(short *output, short *input1, short *input2, int n)
122
*output++ = *input1++;
123
*output++ = *input2++;
127
static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
135
*output++ = l; /* left */
136
*output++ = (l/2)+(r/2); /* center */
137
*output++ = r; /* right */
138
*output++ = 0; /* left surround */
139
*output++ = 0; /* right surroud */
140
*output++ = 0; /* low freq */
144
ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
145
int output_rate, int input_rate,
146
enum SampleFormat sample_fmt_out,
147
enum SampleFormat sample_fmt_in,
148
int filter_length, int log2_phase_count,
149
int linear, double cutoff)
153
if ( input_channels > 2)
155
av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n");
159
s = av_mallocz(sizeof(ReSampleContext));
162
av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
166
s->ratio = (float)output_rate / (float)input_rate;
168
s->input_channels = input_channels;
169
s->output_channels = output_channels;
171
s->filter_channels = s->input_channels;
172
if (s->output_channels < s->filter_channels)
173
s->filter_channels = s->output_channels;
175
s->sample_fmt [0] = sample_fmt_in;
176
s->sample_fmt [1] = sample_fmt_out;
177
s->sample_size[0] = av_get_bits_per_sample_format(s->sample_fmt[0])>>3;
178
s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3;
180
if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
181
if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
182
s->sample_fmt[0], 1, NULL, 0))) {
183
av_log(s, AV_LOG_ERROR,
184
"Cannot convert %s sample format to s16 sample format\n",
185
avcodec_get_sample_fmt_name(s->sample_fmt[0]));
191
if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
192
if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
193
SAMPLE_FMT_S16, 1, NULL, 0))) {
194
av_log(s, AV_LOG_ERROR,
195
"Cannot convert s16 sample format to %s sample format\n",
196
avcodec_get_sample_fmt_name(s->sample_fmt[1]));
197
av_audio_convert_free(s->convert_ctx[0]);
204
* AC-3 output is the only case where filter_channels could be greater than 2.
205
* input channels can't be greater than 2, so resample the 2 channels and then
206
* expand to 6 channels after the resampling.
208
if(s->filter_channels>2)
209
s->filter_channels = 2;
212
s->resample_context= av_resample_init(output_rate, input_rate,
213
filter_length, log2_phase_count, linear, cutoff);
215
*(const AVClass**)s->resample_context = &audioresample_context_class;
220
#if LIBAVCODEC_VERSION_MAJOR < 53
221
ReSampleContext *audio_resample_init(int output_channels, int input_channels,
222
int output_rate, int input_rate)
224
return av_audio_resample_init(output_channels, input_channels,
225
output_rate, input_rate,
226
SAMPLE_FMT_S16, SAMPLE_FMT_S16,
231
/* resample audio. 'nb_samples' is the number of input samples */
232
/* XXX: optimize it ! */
233
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
238
short *buftmp2[2], *buftmp3[2];
239
short *output_bak = NULL;
242
if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
244
memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
248
if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
249
int istride[1] = { s->sample_size[0] };
250
int ostride[1] = { 2 };
251
const void *ibuf[1] = { input };
253
unsigned input_size = nb_samples*s->input_channels*2;
255
if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
256
av_free(s->buffer[0]);
257
s->buffer_size[0] = input_size;
258
s->buffer[0] = av_malloc(s->buffer_size[0]);
260
av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
265
obuf[0] = s->buffer[0];
267
if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
268
ibuf, istride, nb_samples*s->input_channels) < 0) {
269
av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format conversion failed\n");
273
input = s->buffer[0];
276
lenout= 4*nb_samples * s->ratio + 16;
278
if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
281
if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
282
av_free(s->buffer[1]);
283
s->buffer_size[1] = lenout;
284
s->buffer[1] = av_malloc(s->buffer_size[1]);
286
av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
291
output = s->buffer[1];
294
/* XXX: move those malloc to resample init code */
295
for(i=0; i<s->filter_channels; i++){
296
bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
297
memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
298
buftmp2[i] = bufin[i] + s->temp_len;
301
/* make some zoom to avoid round pb */
302
bufout[0]= av_malloc( lenout * sizeof(short) );
303
bufout[1]= av_malloc( lenout * sizeof(short) );
305
if (s->input_channels == 2 &&
306
s->output_channels == 1) {
308
stereo_to_mono(buftmp2[0], input, nb_samples);
309
} else if (s->output_channels >= 2 && s->input_channels == 1) {
310
buftmp3[0] = bufout[0];
311
memcpy(buftmp2[0], input, nb_samples*sizeof(short));
312
} else if (s->output_channels >= 2) {
313
buftmp3[0] = bufout[0];
314
buftmp3[1] = bufout[1];
315
stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
318
memcpy(buftmp2[0], input, nb_samples*sizeof(short));
321
nb_samples += s->temp_len;
323
/* resample each channel */
324
nb_samples1 = 0; /* avoid warning */
325
for(i=0;i<s->filter_channels;i++) {
327
int is_last= i+1 == s->filter_channels;
329
nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
330
s->temp_len= nb_samples - consumed;
331
s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
332
memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
335
if (s->output_channels == 2 && s->input_channels == 1) {
336
mono_to_stereo(output, buftmp3[0], nb_samples1);
337
} else if (s->output_channels == 2) {
338
stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
339
} else if (s->output_channels == 6) {
340
ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
343
if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
344
int istride[1] = { 2 };
345
int ostride[1] = { s->sample_size[1] };
346
const void *ibuf[1] = { output };
347
void *obuf[1] = { output_bak };
349
if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
350
ibuf, istride, nb_samples1*s->output_channels) < 0) {
351
av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format convertion failed\n");
356
for(i=0; i<s->filter_channels; i++)
364
void audio_resample_close(ReSampleContext *s)
366
av_resample_close(s->resample_context);
367
av_freep(&s->temp[0]);
368
av_freep(&s->temp[1]);
369
av_freep(&s->buffer[0]);
370
av_freep(&s->buffer[1]);
371
av_audio_convert_free(s->convert_ctx[0]);
372
av_audio_convert_free(s->convert_ctx[1]);