lavr: resampling: add filter type and Kaiser window beta to AVOptions
[libav.git] / libavresample / utils.c
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1/*
2 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
3 *
4 * This file is part of Libav.
5 *
6 * Libav is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
10 *
11 * Libav is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
15 *
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with Libav; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19 */
20
21#include "libavutil/dict.h"
22#include "libavutil/error.h"
23#include "libavutil/log.h"
24#include "libavutil/mem.h"
25#include "libavutil/opt.h"
26
27#include "avresample.h"
28#include "audio_data.h"
29#include "internal.h"
30
31int avresample_open(AVAudioResampleContext *avr)
32{
33 int ret;
34
35 /* set channel mixing parameters */
36 avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout);
37 if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) {
38 av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n",
39 avr->in_channel_layout);
40 return AVERROR(EINVAL);
41 }
42 avr->out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout);
43 if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) {
44 av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n",
45 avr->out_channel_layout);
46 return AVERROR(EINVAL);
47 }
48 avr->resample_channels = FFMIN(avr->in_channels, avr->out_channels);
49 avr->downmix_needed = avr->in_channels > avr->out_channels;
50 avr->upmix_needed = avr->out_channels > avr->in_channels ||
51 avr->am->matrix ||
52 (avr->out_channels == avr->in_channels &&
53 avr->in_channel_layout != avr->out_channel_layout);
54 avr->mixing_needed = avr->downmix_needed || avr->upmix_needed;
55
56 /* set resampling parameters */
57 avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate ||
58 avr->force_resampling;
59
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60 /* select internal sample format if not specified by the user */
61 if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE &&
62 (avr->mixing_needed || avr->resample_needed)) {
63 enum AVSampleFormat in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
64 enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
65 int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
66 av_get_bytes_per_sample(out_fmt));
67 if (avr->resample_needed || max_bps <= 2) {
68 avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
69 } else if (avr->mixing_needed) {
70 avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
71 }
72 av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
73 av_get_sample_fmt_name(avr->internal_sample_fmt));
c8af852b 74 }
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75
76 /* set sample format conversion parameters */
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77 if (avr->in_channels == 1)
78 avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
79 if (avr->out_channels == 1)
80 avr->out_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
81 avr->in_convert_needed = (avr->resample_needed || avr->mixing_needed) &&
82 avr->in_sample_fmt != avr->internal_sample_fmt;
83 if (avr->resample_needed || avr->mixing_needed)
84 avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt;
85 else
86 avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt;
87
88 /* allocate buffers */
89 if (avr->mixing_needed || avr->in_convert_needed) {
90 avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels),
91 0, avr->internal_sample_fmt,
92 "in_buffer");
93 if (!avr->in_buffer) {
94 ret = AVERROR(EINVAL);
95 goto error;
96 }
97 }
98 if (avr->resample_needed) {
99 avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels,
100 0, avr->internal_sample_fmt,
101 "resample_out_buffer");
102 if (!avr->resample_out_buffer) {
103 ret = AVERROR(EINVAL);
104 goto error;
105 }
106 }
107 if (avr->out_convert_needed) {
108 avr->out_buffer = ff_audio_data_alloc(avr->out_channels, 0,
109 avr->out_sample_fmt, "out_buffer");
110 if (!avr->out_buffer) {
111 ret = AVERROR(EINVAL);
112 goto error;
113 }
114 }
115 avr->out_fifo = av_audio_fifo_alloc(avr->out_sample_fmt, avr->out_channels,
116 1024);
117 if (!avr->out_fifo) {
118 ret = AVERROR(ENOMEM);
119 goto error;
120 }
121
122 /* setup contexts */
123 if (avr->in_convert_needed) {
124 avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt,
125 avr->in_sample_fmt, avr->in_channels);
126 if (!avr->ac_in) {
127 ret = AVERROR(ENOMEM);
128 goto error;
129 }
130 }
131 if (avr->out_convert_needed) {
132 enum AVSampleFormat src_fmt;
133 if (avr->in_convert_needed)
134 src_fmt = avr->internal_sample_fmt;
135 else
136 src_fmt = avr->in_sample_fmt;
137 avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt,
138 avr->out_channels);
139 if (!avr->ac_out) {
140 ret = AVERROR(ENOMEM);
141 goto error;
142 }
143 }
144 if (avr->resample_needed) {
145 avr->resample = ff_audio_resample_init(avr);
146 if (!avr->resample) {
147 ret = AVERROR(ENOMEM);
148 goto error;
149 }
150 }
151 if (avr->mixing_needed) {
152 ret = ff_audio_mix_init(avr);
153 if (ret < 0)
154 goto error;
155 }
156
157 return 0;
158
159error:
160 avresample_close(avr);
161 return ret;
162}
163
164void avresample_close(AVAudioResampleContext *avr)
165{
166 ff_audio_data_free(&avr->in_buffer);
167 ff_audio_data_free(&avr->resample_out_buffer);
168 ff_audio_data_free(&avr->out_buffer);
169 av_audio_fifo_free(avr->out_fifo);
170 avr->out_fifo = NULL;
171 av_freep(&avr->ac_in);
172 av_freep(&avr->ac_out);
173 ff_audio_resample_free(&avr->resample);
174 ff_audio_mix_close(avr->am);
175 return;
176}
177
178void avresample_free(AVAudioResampleContext **avr)
179{
180 if (!*avr)
181 return;
182 avresample_close(*avr);
183 av_freep(&(*avr)->am);
184 av_opt_free(*avr);
185 av_freep(avr);
186}
187
188static int handle_buffered_output(AVAudioResampleContext *avr,
189 AudioData *output, AudioData *converted)
190{
191 int ret;
192
193 if (!output || av_audio_fifo_size(avr->out_fifo) > 0 ||
194 (converted && output->allocated_samples < converted->nb_samples)) {
195 if (converted) {
196 /* if there are any samples in the output FIFO or if the
197 user-supplied output buffer is not large enough for all samples,
198 we add to the output FIFO */
199 av_dlog(avr, "[FIFO] add %s to out_fifo\n", converted->name);
200 ret = ff_audio_data_add_to_fifo(avr->out_fifo, converted, 0,
201 converted->nb_samples);
202 if (ret < 0)
203 return ret;
204 }
205
206 /* if the user specified an output buffer, read samples from the output
207 FIFO to the user output */
208 if (output && output->allocated_samples > 0) {
209 av_dlog(avr, "[FIFO] read from out_fifo to output\n");
210 av_dlog(avr, "[end conversion]\n");
211 return ff_audio_data_read_from_fifo(avr->out_fifo, output,
212 output->allocated_samples);
213 }
214 } else if (converted) {
215 /* copy directly to output if it is large enough or there is not any
216 data in the output FIFO */
217 av_dlog(avr, "[copy] %s to output\n", converted->name);
218 output->nb_samples = 0;
219 ret = ff_audio_data_copy(output, converted);
220 if (ret < 0)
221 return ret;
222 av_dlog(avr, "[end conversion]\n");
223 return output->nb_samples;
224 }
225 av_dlog(avr, "[end conversion]\n");
226 return 0;
227}
228
229int avresample_convert(AVAudioResampleContext *avr, void **output,
230 int out_plane_size, int out_samples, void **input,
231 int in_plane_size, int in_samples)
232{
233 AudioData input_buffer;
234 AudioData output_buffer;
235 AudioData *current_buffer;
236 int ret;
237
238 /* reset internal buffers */
239 if (avr->in_buffer) {
240 avr->in_buffer->nb_samples = 0;
241 ff_audio_data_set_channels(avr->in_buffer,
242 avr->in_buffer->allocated_channels);
243 }
244 if (avr->resample_out_buffer) {
245 avr->resample_out_buffer->nb_samples = 0;
246 ff_audio_data_set_channels(avr->resample_out_buffer,
247 avr->resample_out_buffer->allocated_channels);
248 }
249 if (avr->out_buffer) {
250 avr->out_buffer->nb_samples = 0;
251 ff_audio_data_set_channels(avr->out_buffer,
252 avr->out_buffer->allocated_channels);
253 }
254
255 av_dlog(avr, "[start conversion]\n");
256
257 /* initialize output_buffer with output data */
258 if (output) {
259 ret = ff_audio_data_init(&output_buffer, output, out_plane_size,
260 avr->out_channels, out_samples,
261 avr->out_sample_fmt, 0, "output");
262 if (ret < 0)
263 return ret;
264 output_buffer.nb_samples = 0;
265 }
266
267 if (input) {
268 /* initialize input_buffer with input data */
269 ret = ff_audio_data_init(&input_buffer, input, in_plane_size,
270 avr->in_channels, in_samples,
271 avr->in_sample_fmt, 1, "input");
272 if (ret < 0)
273 return ret;
274 current_buffer = &input_buffer;
275
276 if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed &&
277 !avr->out_convert_needed && output && out_samples >= in_samples) {
278 /* in some rare cases we can copy input to output and upmix
279 directly in the output buffer */
280 av_dlog(avr, "[copy] %s to output\n", current_buffer->name);
281 ret = ff_audio_data_copy(&output_buffer, current_buffer);
282 if (ret < 0)
283 return ret;
284 current_buffer = &output_buffer;
285 } else if (avr->mixing_needed || avr->in_convert_needed) {
286 /* if needed, copy or convert input to in_buffer, and downmix if
287 applicable */
288 if (avr->in_convert_needed) {
289 ret = ff_audio_data_realloc(avr->in_buffer,
290 current_buffer->nb_samples);
291 if (ret < 0)
292 return ret;
293 av_dlog(avr, "[convert] %s to in_buffer\n", current_buffer->name);
294 ret = ff_audio_convert(avr->ac_in, avr->in_buffer, current_buffer,
295 current_buffer->nb_samples);
296 if (ret < 0)
297 return ret;
298 } else {
299 av_dlog(avr, "[copy] %s to in_buffer\n", current_buffer->name);
300 ret = ff_audio_data_copy(avr->in_buffer, current_buffer);
301 if (ret < 0)
302 return ret;
303 }
304 ff_audio_data_set_channels(avr->in_buffer, avr->in_channels);
305 if (avr->downmix_needed) {
306 av_dlog(avr, "[downmix] in_buffer\n");
307 ret = ff_audio_mix(avr->am, avr->in_buffer);
308 if (ret < 0)
309 return ret;
310 }
311 current_buffer = avr->in_buffer;
312 }
313 } else {
314 /* flush resampling buffer and/or output FIFO if input is NULL */
315 if (!avr->resample_needed)
316 return handle_buffered_output(avr, output ? &output_buffer : NULL,
317 NULL);
318 current_buffer = NULL;
319 }
320
321 if (avr->resample_needed) {
322 AudioData *resample_out;
323 int consumed = 0;
324
325 if (!avr->out_convert_needed && output && out_samples > 0)
326 resample_out = &output_buffer;
327 else
328 resample_out = avr->resample_out_buffer;
329 av_dlog(avr, "[resample] %s to %s\n", current_buffer->name,
330 resample_out->name);
331 ret = ff_audio_resample(avr->resample, resample_out,
332 current_buffer, &consumed);
333 if (ret < 0)
334 return ret;
335
336 /* if resampling did not produce any samples, just return 0 */
337 if (resample_out->nb_samples == 0) {
338 av_dlog(avr, "[end conversion]\n");
339 return 0;
340 }
341
342 current_buffer = resample_out;
343 }
344
345 if (avr->upmix_needed) {
346 av_dlog(avr, "[upmix] %s\n", current_buffer->name);
347 ret = ff_audio_mix(avr->am, current_buffer);
348 if (ret < 0)
349 return ret;
350 }
351
352 /* if we resampled or upmixed directly to output, return here */
353 if (current_buffer == &output_buffer) {
354 av_dlog(avr, "[end conversion]\n");
355 return current_buffer->nb_samples;
356 }
357
358 if (avr->out_convert_needed) {
359 if (output && out_samples >= current_buffer->nb_samples) {
360 /* convert directly to output */
361 av_dlog(avr, "[convert] %s to output\n", current_buffer->name);
362 ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer,
363 current_buffer->nb_samples);
364 if (ret < 0)
365 return ret;
366
367 av_dlog(avr, "[end conversion]\n");
368 return output_buffer.nb_samples;
369 } else {
370 ret = ff_audio_data_realloc(avr->out_buffer,
371 current_buffer->nb_samples);
372 if (ret < 0)
373 return ret;
374 av_dlog(avr, "[convert] %s to out_buffer\n", current_buffer->name);
375 ret = ff_audio_convert(avr->ac_out, avr->out_buffer,
376 current_buffer, current_buffer->nb_samples);
377 if (ret < 0)
378 return ret;
379 current_buffer = avr->out_buffer;
380 }
381 }
382
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383 return handle_buffered_output(avr, output ? &output_buffer : NULL,
384 current_buffer);
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385}
386
387int avresample_available(AVAudioResampleContext *avr)
388{
389 return av_audio_fifo_size(avr->out_fifo);
390}
391
392int avresample_read(AVAudioResampleContext *avr, void **output, int nb_samples)
393{
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394 if (!output)
395 return av_audio_fifo_drain(avr->out_fifo, nb_samples);
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396 return av_audio_fifo_read(avr->out_fifo, output, nb_samples);
397}
398
399unsigned avresample_version(void)
400{
401 return LIBAVRESAMPLE_VERSION_INT;
402}
403
404const char *avresample_license(void)
405{
406#define LICENSE_PREFIX "libavresample license: "
407 return LICENSE_PREFIX LIBAV_LICENSE + sizeof(LICENSE_PREFIX) - 1;
408}
409
410const char *avresample_configuration(void)
411{
412 return LIBAV_CONFIGURATION;
413}