mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-30 04:00:37 +00:00
e119cdee3b
While this is slightly more expensive (~48% slower per random number) it does not cause any measurable difference when running through a complete audio conversion pipeline. On the other hand its random numbers are of much higher quality and on spectrograms for 32 bit to 24 bit conversion the difference is clearly visible. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1729>
541 lines
15 KiB
C
541 lines
15 KiB
C
/* 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
|
|
*/
|
|
#ifdef HAVE_CONFIG_H
|
|
#include "config.h"
|
|
#endif
|
|
|
|
#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;
|
|
guint32 random_state;
|
|
/* 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);
|
|
}
|
|
|
|
/* 32 bit xorshift PRNG, see https://en.wikipedia.org/wiki/Xorshift */
|
|
static inline guint32
|
|
gst_fast_random_uint32 (guint32 * state)
|
|
{
|
|
guint64 x = *state;
|
|
x ^= x << 13;
|
|
x ^= x >> 17;
|
|
x ^= x << 5;
|
|
return (*state = x);
|
|
}
|
|
|
|
static inline gint32
|
|
gst_fast_random_int32 (guint32 * state)
|
|
{
|
|
return (gint32) gst_fast_random_uint32 (state);
|
|
}
|
|
|
|
/* Assuming dither == 2^n,
|
|
* returns one of 2^(n+1) possible random values:
|
|
* -dither <= retval < dither */
|
|
#define RANDOM_INT_DITHER(state, dither) \
|
|
(- dither + (gst_fast_random_int32 (state) & ((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 (&quant->random_state, dither);
|
|
break;
|
|
|
|
case GST_AUDIO_DITHER_TPDF:
|
|
dither = 1 << (shift - 1);
|
|
for (i = 0; i < len; i++)
|
|
d[i] =
|
|
bias + RANDOM_INT_DITHER (&quant->random_state,
|
|
dither) + RANDOM_INT_DITHER (&quant->random_state, 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 (&quant->random_state, 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)
|
|
{
|
|
/* Some non-zero number */
|
|
quant->random_state = 0xc2d6038f;
|
|
|
|
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);
|
|
}
|