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
43typedef 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
54struct 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
89static const float ns_48_coef_b[4] = {
90    2.2374f, -0.7339f, -0.1251f, -0.6033f
91};
92
93static const float ns_48_coef_a[4] = {
94    0.9030f, 0.0116f, -0.5853f, -0.2571f
95};
96
97static const float ns_44_coef_b[4] = {
98    2.2061f, -0.4707f, -0.2534f, -0.6213f
99};
100
101static const float ns_44_coef_a[4] = {
102    1.0587f, 0.0676f, -0.6054f, -0.2738f
103};
104
105static 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
112static 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
124static 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
133static 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
142static 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
172static 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
211static 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
242int 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
312void 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
329static 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
345DitherContext *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
437fail:
438    ff_dither_free(&c);
439    return NULL;
440}
441