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