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- /*
- * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
- *
- * Triangular with Noise Shaping is based on opusfile.
- * Copyright (c) 1994-2012 by the Xiph.Org Foundation and contributors
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
- /**
- * @file
- * Dithered Audio Sample Quantization
- *
- * Converts from dbl, flt, or s32 to s16 using dithering.
- */
-
- #include <math.h>
- #include <stdint.h>
-
- #include "libavutil/attributes.h"
- #include "libavutil/common.h"
- #include "libavutil/lfg.h"
- #include "libavutil/mem.h"
- #include "libavutil/samplefmt.h"
- #include "audio_convert.h"
- #include "dither.h"
- #include "internal.h"
-
- typedef struct DitherState {
- int mute;
- unsigned int seed;
- AVLFG lfg;
- float *noise_buf;
- int noise_buf_size;
- int noise_buf_ptr;
- float dither_a[4];
- float dither_b[4];
- } DitherState;
-
- struct DitherContext {
- DitherDSPContext ddsp;
- enum AVResampleDitherMethod method;
- int apply_map;
- ChannelMapInfo *ch_map_info;
-
- int mute_dither_threshold; // threshold for disabling dither
- int mute_reset_threshold; // threshold for resetting noise shaping
- const float *ns_coef_b; // noise shaping coeffs
- const float *ns_coef_a; // noise shaping coeffs
-
- int channels;
- DitherState *state; // dither states for each channel
-
- AudioData *flt_data; // input data in fltp
- AudioData *s16_data; // dithered output in s16p
- AudioConvert *ac_in; // converter for input to fltp
- AudioConvert *ac_out; // converter for s16p to s16 (if needed)
-
- void (*quantize)(int16_t *dst, const float *src, float *dither, int len);
- int samples_align;
- };
-
- /* mute threshold, in seconds */
- #define MUTE_THRESHOLD_SEC 0.000333
-
- /* scale factor for 16-bit output.
- The signal is attenuated slightly to avoid clipping */
- #define S16_SCALE 32753.0f
-
- /* scale to convert lfg from INT_MIN/INT_MAX to -0.5/0.5 */
- #define LFG_SCALE (1.0f / (2.0f * INT32_MAX))
-
- /* noise shaping coefficients */
-
- static const float ns_48_coef_b[4] = {
- 2.2374f, -0.7339f, -0.1251f, -0.6033f
- };
-
- static const float ns_48_coef_a[4] = {
- 0.9030f, 0.0116f, -0.5853f, -0.2571f
- };
-
- static const float ns_44_coef_b[4] = {
- 2.2061f, -0.4707f, -0.2534f, -0.6213f
- };
-
- static const float ns_44_coef_a[4] = {
- 1.0587f, 0.0676f, -0.6054f, -0.2738f
- };
-
- static void dither_int_to_float_rectangular_c(float *dst, int *src, int len)
- {
- int i;
- for (i = 0; i < len; i++)
- dst[i] = src[i] * LFG_SCALE;
- }
-
- static void dither_int_to_float_triangular_c(float *dst, int *src0, int len)
- {
- int i;
- int *src1 = src0 + len;
-
- for (i = 0; i < len; i++) {
- float r = src0[i] * LFG_SCALE;
- r += src1[i] * LFG_SCALE;
- dst[i] = r;
- }
- }
-
- static void quantize_c(int16_t *dst, const float *src, float *dither, int len)
- {
- int i;
- for (i = 0; i < len; i++)
- dst[i] = av_clip_int16(lrintf(src[i] * S16_SCALE + dither[i]));
- }
-
- #define SQRT_1_6 0.40824829046386301723f
-
- static void dither_highpass_filter(float *src, int len)
- {
- int i;
-
- /* filter is from libswresample in FFmpeg */
- for (i = 0; i < len - 2; i++)
- src[i] = (-src[i] + 2 * src[i + 1] - src[i + 2]) * SQRT_1_6;
- }
-
- static int generate_dither_noise(DitherContext *c, DitherState *state,
- int min_samples)
- {
- int i;
- int nb_samples = FFALIGN(min_samples, 16) + 16;
- int buf_samples = nb_samples *
- (c->method == AV_RESAMPLE_DITHER_RECTANGULAR ? 1 : 2);
- unsigned int *noise_buf_ui;
-
- av_freep(&state->noise_buf);
- state->noise_buf_size = state->noise_buf_ptr = 0;
-
- state->noise_buf = av_malloc(buf_samples * sizeof(*state->noise_buf));
- if (!state->noise_buf)
- return AVERROR(ENOMEM);
- state->noise_buf_size = FFALIGN(min_samples, 16);
- noise_buf_ui = (unsigned int *)state->noise_buf;
-
- av_lfg_init(&state->lfg, state->seed);
- for (i = 0; i < buf_samples; i++)
- noise_buf_ui[i] = av_lfg_get(&state->lfg);
-
- c->ddsp.dither_int_to_float(state->noise_buf, noise_buf_ui, nb_samples);
-
- if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_HP)
- dither_highpass_filter(state->noise_buf, nb_samples);
-
- return 0;
- }
-
- static void quantize_triangular_ns(DitherContext *c, DitherState *state,
- int16_t *dst, const float *src,
- int nb_samples)
- {
- int i, j;
- float *dither = &state->noise_buf[state->noise_buf_ptr];
-
- if (state->mute > c->mute_reset_threshold)
- memset(state->dither_a, 0, sizeof(state->dither_a));
-
- for (i = 0; i < nb_samples; i++) {
- float err = 0;
- float sample = src[i] * S16_SCALE;
-
- for (j = 0; j < 4; j++) {
- err += c->ns_coef_b[j] * state->dither_b[j] -
- c->ns_coef_a[j] * state->dither_a[j];
- }
- for (j = 3; j > 0; j--) {
- state->dither_a[j] = state->dither_a[j - 1];
- state->dither_b[j] = state->dither_b[j - 1];
- }
- state->dither_a[0] = err;
- sample -= err;
-
- if (state->mute > c->mute_dither_threshold) {
- dst[i] = av_clip_int16(lrintf(sample));
- state->dither_b[0] = 0;
- } else {
- dst[i] = av_clip_int16(lrintf(sample + dither[i]));
- state->dither_b[0] = av_clipf(dst[i] - sample, -1.5f, 1.5f);
- }
-
- state->mute++;
- if (src[i])
- state->mute = 0;
- }
- }
-
- static int convert_samples(DitherContext *c, int16_t **dst, float * const *src,
- int channels, int nb_samples)
- {
- int ch, ret;
- int aligned_samples = FFALIGN(nb_samples, 16);
-
- for (ch = 0; ch < channels; ch++) {
- DitherState *state = &c->state[ch];
-
- if (state->noise_buf_size < aligned_samples) {
- ret = generate_dither_noise(c, state, nb_samples);
- if (ret < 0)
- return ret;
- } else if (state->noise_buf_size - state->noise_buf_ptr < aligned_samples) {
- state->noise_buf_ptr = 0;
- }
-
- if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
- quantize_triangular_ns(c, state, dst[ch], src[ch], nb_samples);
- } else {
- c->quantize(dst[ch], src[ch],
- &state->noise_buf[state->noise_buf_ptr],
- FFALIGN(nb_samples, c->samples_align));
- }
-
- state->noise_buf_ptr += aligned_samples;
- }
-
- return 0;
- }
-
- int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src)
- {
- int ret;
- AudioData *flt_data;
-
- /* output directly to dst if it is planar */
- if (dst->sample_fmt == AV_SAMPLE_FMT_S16P)
- c->s16_data = dst;
- else {
- /* make sure s16_data is large enough for the output */
- ret = ff_audio_data_realloc(c->s16_data, src->nb_samples);
- if (ret < 0)
- return ret;
- }
-
- if (src->sample_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) {
- /* make sure flt_data is large enough for the input */
- ret = ff_audio_data_realloc(c->flt_data, src->nb_samples);
- if (ret < 0)
- return ret;
- flt_data = c->flt_data;
- }
-
- if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) {
- /* convert input samples to fltp and scale to s16 range */
- ret = ff_audio_convert(c->ac_in, flt_data, src);
- if (ret < 0)
- return ret;
- } else if (c->apply_map) {
- ret = ff_audio_data_copy(flt_data, src, c->ch_map_info);
- if (ret < 0)
- return ret;
- } else {
- flt_data = src;
- }
-
- /* check alignment and padding constraints */
- if (c->method != AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
- int ptr_align = FFMIN(flt_data->ptr_align, c->s16_data->ptr_align);
- int samples_align = FFMIN(flt_data->samples_align, c->s16_data->samples_align);
- int aligned_len = FFALIGN(src->nb_samples, c->ddsp.samples_align);
-
- if (!(ptr_align % c->ddsp.ptr_align) && samples_align >= aligned_len) {
- c->quantize = c->ddsp.quantize;
- c->samples_align = c->ddsp.samples_align;
- } else {
- c->quantize = quantize_c;
- c->samples_align = 1;
- }
- }
-
- ret = convert_samples(c, (int16_t **)c->s16_data->data,
- (float * const *)flt_data->data, src->channels,
- src->nb_samples);
- if (ret < 0)
- return ret;
-
- c->s16_data->nb_samples = src->nb_samples;
-
- /* interleave output to dst if needed */
- if (dst->sample_fmt == AV_SAMPLE_FMT_S16) {
- ret = ff_audio_convert(c->ac_out, dst, c->s16_data);
- if (ret < 0)
- return ret;
- } else
- c->s16_data = NULL;
-
- return 0;
- }
-
- void ff_dither_free(DitherContext **cp)
- {
- DitherContext *c = *cp;
- int ch;
-
- if (!c)
- return;
- ff_audio_data_free(&c->flt_data);
- ff_audio_data_free(&c->s16_data);
- ff_audio_convert_free(&c->ac_in);
- ff_audio_convert_free(&c->ac_out);
- for (ch = 0; ch < c->channels; ch++)
- av_free(c->state[ch].noise_buf);
- av_free(c->state);
- av_freep(cp);
- }
-
- static av_cold void dither_init(DitherDSPContext *ddsp,
- enum AVResampleDitherMethod method)
- {
- ddsp->quantize = quantize_c;
- ddsp->ptr_align = 1;
- ddsp->samples_align = 1;
-
- if (method == AV_RESAMPLE_DITHER_RECTANGULAR)
- ddsp->dither_int_to_float = dither_int_to_float_rectangular_c;
- else
- ddsp->dither_int_to_float = dither_int_to_float_triangular_c;
-
- if (ARCH_X86)
- ff_dither_init_x86(ddsp, method);
- }
-
- DitherContext *ff_dither_alloc(AVAudioResampleContext *avr,
- enum AVSampleFormat out_fmt,
- enum AVSampleFormat in_fmt,
- int channels, int sample_rate, int apply_map)
- {
- AVLFG seed_gen;
- DitherContext *c;
- int ch;
-
- if (av_get_packed_sample_fmt(out_fmt) != AV_SAMPLE_FMT_S16 ||
- av_get_bytes_per_sample(in_fmt) <= 2) {
- av_log(avr, AV_LOG_ERROR, "dithering %s to %s is not supported\n",
- av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt));
- return NULL;
- }
-
- c = av_mallocz(sizeof(*c));
- if (!c)
- return NULL;
-
- c->apply_map = apply_map;
- if (apply_map)
- c->ch_map_info = &avr->ch_map_info;
-
- if (avr->dither_method == AV_RESAMPLE_DITHER_TRIANGULAR_NS &&
- sample_rate != 48000 && sample_rate != 44100) {
- av_log(avr, AV_LOG_WARNING, "sample rate must be 48000 or 44100 Hz "
- "for triangular_ns dither. using triangular_hp instead.\n");
- avr->dither_method = AV_RESAMPLE_DITHER_TRIANGULAR_HP;
- }
- c->method = avr->dither_method;
- dither_init(&c->ddsp, c->method);
-
- if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) {
- if (sample_rate == 48000) {
- c->ns_coef_b = ns_48_coef_b;
- c->ns_coef_a = ns_48_coef_a;
- } else {
- c->ns_coef_b = ns_44_coef_b;
- c->ns_coef_a = ns_44_coef_a;
- }
- }
-
- /* Either s16 or s16p output format is allowed, but s16p is used
- internally, so we need to use a temp buffer and interleave if the output
- format is s16 */
- if (out_fmt != AV_SAMPLE_FMT_S16P) {
- c->s16_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_S16P,
- "dither s16 buffer");
- if (!c->s16_data)
- goto fail;
-
- c->ac_out = ff_audio_convert_alloc(avr, out_fmt, AV_SAMPLE_FMT_S16P,
- channels, sample_rate, 0);
- if (!c->ac_out)
- goto fail;
- }
-
- if (in_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) {
- c->flt_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_FLTP,
- "dither flt buffer");
- if (!c->flt_data)
- goto fail;
- }
- if (in_fmt != AV_SAMPLE_FMT_FLTP) {
- c->ac_in = ff_audio_convert_alloc(avr, AV_SAMPLE_FMT_FLTP, in_fmt,
- channels, sample_rate, c->apply_map);
- if (!c->ac_in)
- goto fail;
- }
-
- c->state = av_mallocz(channels * sizeof(*c->state));
- if (!c->state)
- goto fail;
- c->channels = channels;
-
- /* calculate thresholds for turning off dithering during periods of
- silence to avoid replacing digital silence with quiet dither noise */
- c->mute_dither_threshold = lrintf(sample_rate * MUTE_THRESHOLD_SEC);
- c->mute_reset_threshold = c->mute_dither_threshold * 4;
-
- /* initialize dither states */
- av_lfg_init(&seed_gen, 0xC0FFEE);
- for (ch = 0; ch < channels; ch++) {
- DitherState *state = &c->state[ch];
- state->mute = c->mute_reset_threshold + 1;
- state->seed = av_lfg_get(&seed_gen);
- generate_dither_noise(c, state, FFMAX(32768, sample_rate / 2));
- }
-
- return c;
-
- fail:
- ff_dither_free(&c);
- return NULL;
- }
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