blob: 2ae8d338becba6a12f021b4a98c4251504945efb
1 | /* |
2 | * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> |
3 | * |
4 | * Triangular with Noise Shaping is based on opusfile. |
5 | * Copyright (c) 1994-2012 by the Xiph.Org Foundation and contributors |
6 | * |
7 | * This file is part of FFmpeg. |
8 | * |
9 | * FFmpeg is free software; you can redistribute it and/or |
10 | * modify it under the terms of the GNU Lesser General Public |
11 | * License as published by the Free Software Foundation; either |
12 | * version 2.1 of the License, or (at your option) any later version. |
13 | * |
14 | * FFmpeg is distributed in the hope that it will be useful, |
15 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
16 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
17 | * Lesser General Public License for more details. |
18 | * |
19 | * You should have received a copy of the GNU Lesser General Public |
20 | * License along with FFmpeg; if not, write to the Free Software |
21 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
22 | */ |
23 | |
24 | /** |
25 | * @file |
26 | * Dithered Audio Sample Quantization |
27 | * |
28 | * Converts from dbl, flt, or s32 to s16 using dithering. |
29 | */ |
30 | |
31 | #include <math.h> |
32 | #include <stdint.h> |
33 | |
34 | #include "libavutil/attributes.h" |
35 | #include "libavutil/common.h" |
36 | #include "libavutil/lfg.h" |
37 | #include "libavutil/mem.h" |
38 | #include "libavutil/samplefmt.h" |
39 | #include "audio_convert.h" |
40 | #include "dither.h" |
41 | #include "internal.h" |
42 | |
43 | typedef struct DitherState { |
44 | int mute; |
45 | unsigned int seed; |
46 | AVLFG lfg; |
47 | float *noise_buf; |
48 | int noise_buf_size; |
49 | int noise_buf_ptr; |
50 | float dither_a[4]; |
51 | float dither_b[4]; |
52 | } DitherState; |
53 | |
54 | struct DitherContext { |
55 | DitherDSPContext ddsp; |
56 | enum AVResampleDitherMethod method; |
57 | int apply_map; |
58 | ChannelMapInfo *ch_map_info; |
59 | |
60 | int mute_dither_threshold; // threshold for disabling dither |
61 | int mute_reset_threshold; // threshold for resetting noise shaping |
62 | const float *ns_coef_b; // noise shaping coeffs |
63 | const float *ns_coef_a; // noise shaping coeffs |
64 | |
65 | int channels; |
66 | DitherState *state; // dither states for each channel |
67 | |
68 | AudioData *flt_data; // input data in fltp |
69 | AudioData *s16_data; // dithered output in s16p |
70 | AudioConvert *ac_in; // converter for input to fltp |
71 | AudioConvert *ac_out; // converter for s16p to s16 (if needed) |
72 | |
73 | void (*quantize)(int16_t *dst, const float *src, float *dither, int len); |
74 | int samples_align; |
75 | }; |
76 | |
77 | /* mute threshold, in seconds */ |
78 | #define MUTE_THRESHOLD_SEC 0.000333 |
79 | |
80 | /* scale factor for 16-bit output. |
81 | The signal is attenuated slightly to avoid clipping */ |
82 | #define S16_SCALE 32753.0f |
83 | |
84 | /* scale to convert lfg from INT_MIN/INT_MAX to -0.5/0.5 */ |
85 | #define LFG_SCALE (1.0f / (2.0f * INT32_MAX)) |
86 | |
87 | /* noise shaping coefficients */ |
88 | |
89 | static const float ns_48_coef_b[4] = { |
90 | 2.2374f, -0.7339f, -0.1251f, -0.6033f |
91 | }; |
92 | |
93 | static const float ns_48_coef_a[4] = { |
94 | 0.9030f, 0.0116f, -0.5853f, -0.2571f |
95 | }; |
96 | |
97 | static const float ns_44_coef_b[4] = { |
98 | 2.2061f, -0.4707f, -0.2534f, -0.6213f |
99 | }; |
100 | |
101 | static const float ns_44_coef_a[4] = { |
102 | 1.0587f, 0.0676f, -0.6054f, -0.2738f |
103 | }; |
104 | |
105 | static void dither_int_to_float_rectangular_c(float *dst, int *src, int len) |
106 | { |
107 | int i; |
108 | for (i = 0; i < len; i++) |
109 | dst[i] = src[i] * LFG_SCALE; |
110 | } |
111 | |
112 | static void dither_int_to_float_triangular_c(float *dst, int *src0, int len) |
113 | { |
114 | int i; |
115 | int *src1 = src0 + len; |
116 | |
117 | for (i = 0; i < len; i++) { |
118 | float r = src0[i] * LFG_SCALE; |
119 | r += src1[i] * LFG_SCALE; |
120 | dst[i] = r; |
121 | } |
122 | } |
123 | |
124 | static void quantize_c(int16_t *dst, const float *src, float *dither, int len) |
125 | { |
126 | int i; |
127 | for (i = 0; i < len; i++) |
128 | dst[i] = av_clip_int16(lrintf(src[i] * S16_SCALE + dither[i])); |
129 | } |
130 | |
131 | #define SQRT_1_6 0.40824829046386301723f |
132 | |
133 | static void dither_highpass_filter(float *src, int len) |
134 | { |
135 | int i; |
136 | |
137 | /* filter is from libswresample in FFmpeg */ |
138 | for (i = 0; i < len - 2; i++) |
139 | src[i] = (-src[i] + 2 * src[i + 1] - src[i + 2]) * SQRT_1_6; |
140 | } |
141 | |
142 | static int generate_dither_noise(DitherContext *c, DitherState *state, |
143 | int min_samples) |
144 | { |
145 | int i; |
146 | int nb_samples = FFALIGN(min_samples, 16) + 16; |
147 | int buf_samples = nb_samples * |
148 | (c->method == AV_RESAMPLE_DITHER_RECTANGULAR ? 1 : 2); |
149 | unsigned int *noise_buf_ui; |
150 | |
151 | av_freep(&state->noise_buf); |
152 | state->noise_buf_size = state->noise_buf_ptr = 0; |
153 | |
154 | state->noise_buf = av_malloc(buf_samples * sizeof(*state->noise_buf)); |
155 | if (!state->noise_buf) |
156 | return AVERROR(ENOMEM); |
157 | state->noise_buf_size = FFALIGN(min_samples, 16); |
158 | noise_buf_ui = (unsigned int *)state->noise_buf; |
159 | |
160 | av_lfg_init(&state->lfg, state->seed); |
161 | for (i = 0; i < buf_samples; i++) |
162 | noise_buf_ui[i] = av_lfg_get(&state->lfg); |
163 | |
164 | c->ddsp.dither_int_to_float(state->noise_buf, noise_buf_ui, nb_samples); |
165 | |
166 | if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_HP) |
167 | dither_highpass_filter(state->noise_buf, nb_samples); |
168 | |
169 | return 0; |
170 | } |
171 | |
172 | static void quantize_triangular_ns(DitherContext *c, DitherState *state, |
173 | int16_t *dst, const float *src, |
174 | int nb_samples) |
175 | { |
176 | int i, j; |
177 | float *dither = &state->noise_buf[state->noise_buf_ptr]; |
178 | |
179 | if (state->mute > c->mute_reset_threshold) |
180 | memset(state->dither_a, 0, sizeof(state->dither_a)); |
181 | |
182 | for (i = 0; i < nb_samples; i++) { |
183 | float err = 0; |
184 | float sample = src[i] * S16_SCALE; |
185 | |
186 | for (j = 0; j < 4; j++) { |
187 | err += c->ns_coef_b[j] * state->dither_b[j] - |
188 | c->ns_coef_a[j] * state->dither_a[j]; |
189 | } |
190 | for (j = 3; j > 0; j--) { |
191 | state->dither_a[j] = state->dither_a[j - 1]; |
192 | state->dither_b[j] = state->dither_b[j - 1]; |
193 | } |
194 | state->dither_a[0] = err; |
195 | sample -= err; |
196 | |
197 | if (state->mute > c->mute_dither_threshold) { |
198 | dst[i] = av_clip_int16(lrintf(sample)); |
199 | state->dither_b[0] = 0; |
200 | } else { |
201 | dst[i] = av_clip_int16(lrintf(sample + dither[i])); |
202 | state->dither_b[0] = av_clipf(dst[i] - sample, -1.5f, 1.5f); |
203 | } |
204 | |
205 | state->mute++; |
206 | if (src[i]) |
207 | state->mute = 0; |
208 | } |
209 | } |
210 | |
211 | static int convert_samples(DitherContext *c, int16_t **dst, float * const *src, |
212 | int channels, int nb_samples) |
213 | { |
214 | int ch, ret; |
215 | int aligned_samples = FFALIGN(nb_samples, 16); |
216 | |
217 | for (ch = 0; ch < channels; ch++) { |
218 | DitherState *state = &c->state[ch]; |
219 | |
220 | if (state->noise_buf_size < aligned_samples) { |
221 | ret = generate_dither_noise(c, state, nb_samples); |
222 | if (ret < 0) |
223 | return ret; |
224 | } else if (state->noise_buf_size - state->noise_buf_ptr < aligned_samples) { |
225 | state->noise_buf_ptr = 0; |
226 | } |
227 | |
228 | if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) { |
229 | quantize_triangular_ns(c, state, dst[ch], src[ch], nb_samples); |
230 | } else { |
231 | c->quantize(dst[ch], src[ch], |
232 | &state->noise_buf[state->noise_buf_ptr], |
233 | FFALIGN(nb_samples, c->samples_align)); |
234 | } |
235 | |
236 | state->noise_buf_ptr += aligned_samples; |
237 | } |
238 | |
239 | return 0; |
240 | } |
241 | |
242 | int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src) |
243 | { |
244 | int ret; |
245 | AudioData *flt_data; |
246 | |
247 | /* output directly to dst if it is planar */ |
248 | if (dst->sample_fmt == AV_SAMPLE_FMT_S16P) |
249 | c->s16_data = dst; |
250 | else { |
251 | /* make sure s16_data is large enough for the output */ |
252 | ret = ff_audio_data_realloc(c->s16_data, src->nb_samples); |
253 | if (ret < 0) |
254 | return ret; |
255 | } |
256 | |
257 | if (src->sample_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) { |
258 | /* make sure flt_data is large enough for the input */ |
259 | ret = ff_audio_data_realloc(c->flt_data, src->nb_samples); |
260 | if (ret < 0) |
261 | return ret; |
262 | flt_data = c->flt_data; |
263 | } |
264 | |
265 | if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) { |
266 | /* convert input samples to fltp and scale to s16 range */ |
267 | ret = ff_audio_convert(c->ac_in, flt_data, src); |
268 | if (ret < 0) |
269 | return ret; |
270 | } else if (c->apply_map) { |
271 | ret = ff_audio_data_copy(flt_data, src, c->ch_map_info); |
272 | if (ret < 0) |
273 | return ret; |
274 | } else { |
275 | flt_data = src; |
276 | } |
277 | |
278 | /* check alignment and padding constraints */ |
279 | if (c->method != AV_RESAMPLE_DITHER_TRIANGULAR_NS) { |
280 | int ptr_align = FFMIN(flt_data->ptr_align, c->s16_data->ptr_align); |
281 | int samples_align = FFMIN(flt_data->samples_align, c->s16_data->samples_align); |
282 | int aligned_len = FFALIGN(src->nb_samples, c->ddsp.samples_align); |
283 | |
284 | if (!(ptr_align % c->ddsp.ptr_align) && samples_align >= aligned_len) { |
285 | c->quantize = c->ddsp.quantize; |
286 | c->samples_align = c->ddsp.samples_align; |
287 | } else { |
288 | c->quantize = quantize_c; |
289 | c->samples_align = 1; |
290 | } |
291 | } |
292 | |
293 | ret = convert_samples(c, (int16_t **)c->s16_data->data, |
294 | (float * const *)flt_data->data, src->channels, |
295 | src->nb_samples); |
296 | if (ret < 0) |
297 | return ret; |
298 | |
299 | c->s16_data->nb_samples = src->nb_samples; |
300 | |
301 | /* interleave output to dst if needed */ |
302 | if (dst->sample_fmt == AV_SAMPLE_FMT_S16) { |
303 | ret = ff_audio_convert(c->ac_out, dst, c->s16_data); |
304 | if (ret < 0) |
305 | return ret; |
306 | } else |
307 | c->s16_data = NULL; |
308 | |
309 | return 0; |
310 | } |
311 | |
312 | void ff_dither_free(DitherContext **cp) |
313 | { |
314 | DitherContext *c = *cp; |
315 | int ch; |
316 | |
317 | if (!c) |
318 | return; |
319 | ff_audio_data_free(&c->flt_data); |
320 | ff_audio_data_free(&c->s16_data); |
321 | ff_audio_convert_free(&c->ac_in); |
322 | ff_audio_convert_free(&c->ac_out); |
323 | for (ch = 0; ch < c->channels; ch++) |
324 | av_free(c->state[ch].noise_buf); |
325 | av_free(c->state); |
326 | av_freep(cp); |
327 | } |
328 | |
329 | static av_cold void dither_init(DitherDSPContext *ddsp, |
330 | enum AVResampleDitherMethod method) |
331 | { |
332 | ddsp->quantize = quantize_c; |
333 | ddsp->ptr_align = 1; |
334 | ddsp->samples_align = 1; |
335 | |
336 | if (method == AV_RESAMPLE_DITHER_RECTANGULAR) |
337 | ddsp->dither_int_to_float = dither_int_to_float_rectangular_c; |
338 | else |
339 | ddsp->dither_int_to_float = dither_int_to_float_triangular_c; |
340 | |
341 | if (ARCH_X86) |
342 | ff_dither_init_x86(ddsp, method); |
343 | } |
344 | |
345 | DitherContext *ff_dither_alloc(AVAudioResampleContext *avr, |
346 | enum AVSampleFormat out_fmt, |
347 | enum AVSampleFormat in_fmt, |
348 | int channels, int sample_rate, int apply_map) |
349 | { |
350 | AVLFG seed_gen; |
351 | DitherContext *c; |
352 | int ch; |
353 | |
354 | if (av_get_packed_sample_fmt(out_fmt) != AV_SAMPLE_FMT_S16 || |
355 | av_get_bytes_per_sample(in_fmt) <= 2) { |
356 | av_log(avr, AV_LOG_ERROR, "dithering %s to %s is not supported\n", |
357 | av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt)); |
358 | return NULL; |
359 | } |
360 | |
361 | c = av_mallocz(sizeof(*c)); |
362 | if (!c) |
363 | return NULL; |
364 | |
365 | c->apply_map = apply_map; |
366 | if (apply_map) |
367 | c->ch_map_info = &avr->ch_map_info; |
368 | |
369 | if (avr->dither_method == AV_RESAMPLE_DITHER_TRIANGULAR_NS && |
370 | sample_rate != 48000 && sample_rate != 44100) { |
371 | av_log(avr, AV_LOG_WARNING, "sample rate must be 48000 or 44100 Hz " |
372 | "for triangular_ns dither. using triangular_hp instead.\n"); |
373 | avr->dither_method = AV_RESAMPLE_DITHER_TRIANGULAR_HP; |
374 | } |
375 | c->method = avr->dither_method; |
376 | dither_init(&c->ddsp, c->method); |
377 | |
378 | if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) { |
379 | if (sample_rate == 48000) { |
380 | c->ns_coef_b = ns_48_coef_b; |
381 | c->ns_coef_a = ns_48_coef_a; |
382 | } else { |
383 | c->ns_coef_b = ns_44_coef_b; |
384 | c->ns_coef_a = ns_44_coef_a; |
385 | } |
386 | } |
387 | |
388 | /* Either s16 or s16p output format is allowed, but s16p is used |
389 | internally, so we need to use a temp buffer and interleave if the output |
390 | format is s16 */ |
391 | if (out_fmt != AV_SAMPLE_FMT_S16P) { |
392 | c->s16_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_S16P, |
393 | "dither s16 buffer"); |
394 | if (!c->s16_data) |
395 | goto fail; |
396 | |
397 | c->ac_out = ff_audio_convert_alloc(avr, out_fmt, AV_SAMPLE_FMT_S16P, |
398 | channels, sample_rate, 0); |
399 | if (!c->ac_out) |
400 | goto fail; |
401 | } |
402 | |
403 | if (in_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) { |
404 | c->flt_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_FLTP, |
405 | "dither flt buffer"); |
406 | if (!c->flt_data) |
407 | goto fail; |
408 | } |
409 | if (in_fmt != AV_SAMPLE_FMT_FLTP) { |
410 | c->ac_in = ff_audio_convert_alloc(avr, AV_SAMPLE_FMT_FLTP, in_fmt, |
411 | channels, sample_rate, c->apply_map); |
412 | if (!c->ac_in) |
413 | goto fail; |
414 | } |
415 | |
416 | c->state = av_mallocz(channels * sizeof(*c->state)); |
417 | if (!c->state) |
418 | goto fail; |
419 | c->channels = channels; |
420 | |
421 | /* calculate thresholds for turning off dithering during periods of |
422 | silence to avoid replacing digital silence with quiet dither noise */ |
423 | c->mute_dither_threshold = lrintf(sample_rate * MUTE_THRESHOLD_SEC); |
424 | c->mute_reset_threshold = c->mute_dither_threshold * 4; |
425 | |
426 | /* initialize dither states */ |
427 | av_lfg_init(&seed_gen, 0xC0FFEE); |
428 | for (ch = 0; ch < channels; ch++) { |
429 | DitherState *state = &c->state[ch]; |
430 | state->mute = c->mute_reset_threshold + 1; |
431 | state->seed = av_lfg_get(&seed_gen); |
432 | generate_dither_noise(c, state, FFMAX(32768, sample_rate / 2)); |
433 | } |
434 | |
435 | return c; |
436 | |
437 | fail: |
438 | ff_dither_free(&c); |
439 | return NULL; |
440 | } |
441 |