/* GStreamer
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
* (C) 2015 Wim Taymans <wim.taymans@gmail.com>
*
* gstaudioquantize.c: quantizes audio to the target format and optionally
* applies dithering and noise shaping.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/* TODO: - Maybe drop 5-pole noise shaping and use coefficients
* generated by dmaker
* http://shibatch.sf.net
*/
#include <gst/gst.h>
#include <string.h>
#include <math.h>
#include "gstaudiopack.h"
#include "audio-quantize.h"
typedef void (*QuantizeFunc) (GstAudioQuantize * quant, const gpointer src,
gpointer dst, gint count);
struct _GstAudioQuantize
{
GstAudioDitherMethod dither;
GstAudioNoiseShapingMethod ns;
GstAudioQuantizeFlags flags;
GstAudioFormat format;
guint quantizer;
guint stride;
guint blocks;
guint shift;
guint32 mask, bias;
/* last random number generated per channel for hifreq TPDF dither */
gpointer last_random;
/* contains the past quantization errors, error[channels][count] */
guint error_size;
gpointer error_buf;
/* buffer with dither values */
guint dither_size;
gpointer dither_buf;
/* noise shaping coefficients */
gpointer coeffs;
gint n_coeffs;
QuantizeFunc quantize;
};
#define ADDSS(res,val) \
if (val > 0 && res > 0 && G_MAXINT32 - res <= val){ \
res = G_MAXINT32; \
} else if (val < 0 && res < 0 && G_MININT32 - res >= val){ \
res = G_MININT32; \
} else \
res += val;
static void
gst_audio_quantize_quantize_memcpy (GstAudioQuantize * quant,
const gpointer src, gpointer dst, gint samples)
{
if (src != dst)
memcpy (dst, src, samples * sizeof (gint32) * quant->stride);
}
/* Quantize functions for gint32 as intermediate format */
static void
gst_audio_quantize_quantize_int_none_none (GstAudioQuantize * quant,
const gpointer src, gpointer dst, gint samples)
{
audio_orc_int_bias (dst, src, quant->bias, ~quant->mask,
samples * quant->stride);
}
/* This is the base function, implementing a linear congruential generator
* and returning a pseudo random number between 0 and 2^32 - 1.
*/
static inline guint32
gst_fast_random_uint32 (void)
{
static guint32 state = 0xdeadbeef;
return (state = state * 1103515245 + 12345);
}
static inline gint32
gst_fast_random_int32 (void)
{
return (gint32) gst_fast_random_uint32 ();
}
/* Assuming dither == 2^n,
* returns one of 2^(n+1) possible random values:
* -dither <= retval < dither */
#define RANDOM_INT_DITHER(dither) \
(- dither + (gst_fast_random_int32 () & ((dither << 1) - 1)))
static void
setup_dither_buf (GstAudioQuantize * quant, gint samples)
{
gboolean need_init = FALSE;
gint stride = quant->stride;
gint i, len = samples * stride;
guint shift = quant->shift;
guint32 bias;
gint32 dither, *d;
if (quant->dither_size < len) {
quant->dither_size = len;
quant->dither_buf = g_realloc (quant->dither_buf, len * sizeof (gint32));
need_init = TRUE;
}
bias = quant->bias;
d = quant->dither_buf;
switch (quant->dither) {
case GST_AUDIO_DITHER_NONE:
if (need_init) {
for (i = 0; i < len; i++)
d[i] = 0;
}
break;
case GST_AUDIO_DITHER_RPDF:
dither = 1 << (shift);
for (i = 0; i < len; i++)
d[i] = bias + RANDOM_INT_DITHER (dither);
break;
case GST_AUDIO_DITHER_TPDF:
dither = 1 << (shift - 1);
for (i = 0; i < len; i++)
d[i] = bias + RANDOM_INT_DITHER (dither) + RANDOM_INT_DITHER (dither);
break;
case GST_AUDIO_DITHER_TPDF_HF:
{
gint32 tmp, *last_random = quant->last_random;
dither = 1 << (shift - 1);
for (i = 0; i < len; i++) {
tmp = RANDOM_INT_DITHER (dither);
d[i] = bias + tmp - last_random[i % stride];
last_random[i % stride] = tmp;
}
break;
}
}
}
static void
gst_audio_quantize_quantize_int_dither_none (GstAudioQuantize * quant,
const gpointer src, gpointer dst, gint samples)
{
setup_dither_buf (quant, samples);
audio_orc_int_dither (dst, src, quant->dither_buf, ~quant->mask,
samples * quant->stride);
}
static void
setup_error_buf (GstAudioQuantize * quant, gint samples, gint extra)
{
gint stride = quant->stride;
gint len = (samples + extra) * stride;
if (quant->error_size < len) {
quant->error_buf = g_realloc (quant->error_buf, len * sizeof (gint32));
if (quant->error_size == 0)
memset ((gint32 *) quant->error_buf, 0, stride * extra * sizeof (gint32));
quant->error_size = len;
}
}
static void
gst_audio_quantize_quantize_int_dither_feedback (GstAudioQuantize * quant,
const gpointer src, gpointer dst, gint samples)
{
guint32 mask;
gint i, len, stride;
const gint32 *s = src;
gint32 *dith, *d = dst, v, o, *e, err;
setup_dither_buf (quant, samples);
setup_error_buf (quant, samples, 1);
stride = quant->stride;
len = samples * stride;
dith = quant->dither_buf;
e = quant->error_buf;
mask = ~quant->mask;
for (i = 0; i < len; i++) {
o = v = s[i];
/* add dither */
err = dith[i];
/* remove error */
err -= e[i];
ADDSS (v, err);
v &= mask;
/* store new error */
e[i + stride] = e[i] + (v - o);
/* store result */
d[i] = v;
}
memmove (e, &e[len], sizeof (gint32) * stride);
}
#define SHIFT 10
#define REDUCE 8
#define RROUND (1<<(REDUCE-1))
#define SREDUCE 2
#define SROUND (1<<(SREDUCE-1))
static void
gst_audio_quantize_quantize_int_dither_noise_shape (GstAudioQuantize * quant,
const gpointer src, gpointer dst, gint samples)
{
guint32 mask;
gint i, j, k, len, stride, nc;
const gint32 *s = src;
gint32 *c, *dith, *d = dst, v, o, *e, err;
nc = quant->n_coeffs;
setup_dither_buf (quant, samples);
setup_error_buf (quant, samples, nc);
stride = quant->stride;
len = samples * stride;
dith = quant->dither_buf;
e = quant->error_buf;
c = quant->coeffs;
mask = ~quant->mask;
for (i = 0; i < len; i++) {
v = s[i];
/* combine and remove error */
err = 0;
for (j = 0, k = i; j < nc; j++, k += stride)
err -= e[k] * c[j];
err = (err + SROUND) >> (SREDUCE);
ADDSS (v, err);
o = v;
/* add dither */
err = dith[i];
ADDSS (v, err);
/* quantize */
v &= mask;
/* store new error with reduced precision */
e[k] = (v - o + RROUND) >> REDUCE;
/* store result */
d[i] = v;
}
memmove (e, &e[len], sizeof (gint32) * stride * nc);
}
#define MAKE_QUANTIZE_FUNC_NAME(name) \
gst_audio_quantize_quantize_##name
static const QuantizeFunc quantize_funcs[] = {
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_none_none),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_feedback),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_none),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_feedback),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_none),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_feedback),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_none),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_feedback),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape),
(QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (int_dither_noise_shape),
};
/* Same as error feedback but also add 1/2 of the previous error value.
* This moves the noise a bit more into the higher frequencies. */
static const gdouble ns_simple_coeffs[] = {
-0.5, 1.0
};
/* Noise shaping coefficients from[1], moves most power of the
* error noise into inaudible frequency ranges.
*
* [1]
* "Minimally Audible Noise Shaping", Stanley P. Lipshitz,
* John Vanderkooy, and Robert A. Wannamaker,
* J. Audio Eng. Soc., Vol. 39, No. 11, November 1991. */
static const gdouble ns_medium_coeffs[] = {
0.6149, -1.590, 1.959, -2.165, 2.033
};
/* Noise shaping coefficients by David Schleef, moves most power of the
* error noise into inaudible frequency ranges */
static const gdouble ns_high_coeffs[] = {
-0.340122, 0.876066, -1.72008, 2.61339, -3.31399, 3.27918, -2.92975, 2.08484,
};
static void
gst_audio_quantize_setup_noise_shaping (GstAudioQuantize * quant)
{
gint i, n_coeffs = 0;
gint32 *q;
const gdouble *coeffs;
switch (quant->ns) {
case GST_AUDIO_NOISE_SHAPING_HIGH:
n_coeffs = 8;
coeffs = ns_high_coeffs;
break;
case GST_AUDIO_NOISE_SHAPING_MEDIUM:
n_coeffs = 5;
coeffs = ns_medium_coeffs;
break;
case GST_AUDIO_NOISE_SHAPING_SIMPLE:
n_coeffs = 2;
coeffs = ns_simple_coeffs;
break;
case GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK:
break;
case GST_AUDIO_NOISE_SHAPING_NONE:
default:
break;
}
if (n_coeffs) {
quant->n_coeffs = n_coeffs;
q = quant->coeffs = g_new0 (gint32, n_coeffs);
for (i = 0; i < n_coeffs; i++)
q[i] = floor (coeffs[i] * (1 << SHIFT) + 0.5);
}
return;
}
static void
gst_audio_quantize_setup_dither (GstAudioQuantize * quant)
{
switch (quant->dither) {
case GST_AUDIO_DITHER_TPDF_HF:
quant->last_random = g_new0 (gint32, quant->stride);
break;
case GST_AUDIO_DITHER_RPDF:
case GST_AUDIO_DITHER_TPDF:
quant->last_random = NULL;
break;
case GST_AUDIO_DITHER_NONE:
default:
quant->last_random = NULL;
break;
}
return;
}
static void
gst_audio_quantize_setup_quantize_func (GstAudioQuantize * quant)
{
gint index;
if (quant->shift == 0) {
quant->quantize = (QuantizeFunc) MAKE_QUANTIZE_FUNC_NAME (memcpy);
return;
}
index = 5 * quant->dither + quant->ns;
quant->quantize = quantize_funcs[index];
}
static gint
count_power (guint v)
{
gint res = 0;
while (v > 1) {
res++;
v >>= 1;
}
return res;
}
/**
* gst_audio_quantize_new: (skip):
* @dither: a #GstAudioDitherMethod
* @ns: a #GstAudioNoiseShapingMethod
* @flags: #GstAudioQuantizeFlags
* @format: the #GstAudioFormat of the samples
* @channels: the amount of channels in the samples
* @quantizer: the quantizer to use
*
* Create a new quantizer object with the given parameters.
*
* Output samples will be quantized to a multiple of @quantizer. Better
* performance is achieved when @quantizer is a power of 2.
*
* Dithering and noise-shaping can be performed during quantization with
* the @dither and @ns parameters.
*
* Returns: a new #GstAudioQuantize. Free with gst_audio_quantize_free().
*/
GstAudioQuantize *
gst_audio_quantize_new (GstAudioDitherMethod dither,
GstAudioNoiseShapingMethod ns, GstAudioQuantizeFlags flags,
GstAudioFormat format, guint channels, guint quantizer)
{
GstAudioQuantize *quant;
g_return_val_if_fail (format == GST_AUDIO_FORMAT_S32, NULL);
g_return_val_if_fail (channels > 0, NULL);
quant = g_slice_new0 (GstAudioQuantize);
quant->dither = dither;
quant->ns = ns;
quant->flags = flags;
quant->format = format;
if (flags & GST_AUDIO_QUANTIZE_FLAG_NON_INTERLEAVED) {
quant->stride = 1;
quant->blocks = channels;
} else {
quant->stride = channels;
quant->blocks = 1;
}
quant->quantizer = quantizer;
quant->shift = count_power (quantizer);
if (quant->shift > 0)
quant->bias = (1U << (quant->shift - 1));
else
quant->bias = 0;
quant->mask = (1U << quant->shift) - 1;
gst_audio_quantize_setup_dither (quant);
gst_audio_quantize_setup_noise_shaping (quant);
gst_audio_quantize_setup_quantize_func (quant);
return quant;
}
/**
* gst_audio_quantize_free:
* @quant: a #GstAudioQuantize
*
* Free a #GstAudioQuantize.
*/
void
gst_audio_quantize_free (GstAudioQuantize * quant)
{
g_return_if_fail (quant != NULL);
g_free (quant->error_buf);
g_free (quant->coeffs);
g_free (quant->last_random);
g_free (quant->dither_buf);
g_slice_free (GstAudioQuantize, quant);
}
/**
* gst_audio_quantize_reset:
* @quant: a #GstAudioQuantize
*
* Reset @quant to the state is was when created, clearing any
* history it might have.
*/
void
gst_audio_quantize_reset (GstAudioQuantize * quant)
{
g_free (quant->error_buf);
quant->error_buf = NULL;
quant->error_size = 0;
}
/**
* gst_audio_quantize_samples:
* @quant: a #GstAudioQuantize
* @in: input samples
* @out: output samples
* @samples: number of samples
*
* Perform quantization on @samples in @in and write the result to @out.
*
* In case the samples are interleaved, @in and @out must point to an
* array with a single element pointing to a block of interleaved samples.
*
* If non-interleaved samples are used, @in and @out must point to an
* array with pointers to memory blocks, one for each channel.
*
* @in and @out may point to the same memory location, in which case samples will be
* modified in-place.
*/
void
gst_audio_quantize_samples (GstAudioQuantize * quant,
const gpointer in[], gpointer out[], guint samples)
{
guint i;
g_return_if_fail (quant != NULL);
g_return_if_fail (out != NULL || samples == 0);
g_return_if_fail (in != NULL || samples == 0);
for (i = 0; i < quant->blocks; i++)
quant->quantize (quant, in[i], out[i], samples);
}