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ffmpeg/libavresample/dither.c

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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 Libav.
*
* Libav 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.
*
* Libav 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 Libav; 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/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 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;
}
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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