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gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now

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Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_dithering_get_type),
(gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
(gst_audio_convert_init), (gst_audio_convert_set_caps),
(gst_audio_convert_set_property), (gst_audio_convert_get_property):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_noise_shaping),
(gst_audio_quantize_free_noise_shaping),
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither),
(gst_audio_quantize_setup_quantize_func),
(gst_audio_quantize_setup), (gst_audio_quantize_free):
* gst/audioconvert/gstaudioquantize.h:
Implement dithering and noise shaping in audioconvert. By default now
TPDF dithering (and no noise shaping) will be used when converting
from a higher bit depth to 20 bit depth or smaller, otherwise
everything will be as it is now.
For the last audioconvert in a pipeline it would make sense to
use some kind of noise shaping, enabling it by default for all
conversions would give undesired results though. Fixes #360246.
* tests/check/elements/audioconvert.c: (setup_audioconvert),
(GST_START_TEST):
Adjust unit test for the new audioconvert.
This commit is contained in:
Sebastian Dröge 2007-06-28 20:37:58 +00:00
parent 8c05f2ebc9
commit dbb857b93b
9 changed files with 920 additions and 84 deletions
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32
ChangeLog
View file

@ -1,3 +1,35 @@
2007-06-28 Sebastian Dröge <slomo@circular-chaos.org>
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_dithering_get_type),
(gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
(gst_audio_convert_init), (gst_audio_convert_set_caps),
(gst_audio_convert_set_property), (gst_audio_convert_get_property):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_noise_shaping),
(gst_audio_quantize_free_noise_shaping),
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither),
(gst_audio_quantize_setup_quantize_func),
(gst_audio_quantize_setup), (gst_audio_quantize_free):
* gst/audioconvert/gstaudioquantize.h:
Implement dithering and noise shaping in audioconvert. By default now
TPDF dithering (and no noise shaping) will be used when converting
from a higher bit depth to 20 bit depth or smaller, otherwise
everything will be as it is now.
For the last audioconvert in a pipeline it would make sense to
use some kind of noise shaping, enabling it by default for all
conversions would give undesired results though. Fixes #360246.
* tests/check/elements/audioconvert.c: (setup_audioconvert),
(GST_START_TEST):
Adjust unit test for the new audioconvert.
2007-06-28 Wim Taymans <wim@fluendo.com>
* gst/playback/gstqueue2.c: (apply_segment), (update_buffering):

2
gst/audioconvert/Makefile.am
View file

@ -4,6 +4,7 @@ libgstaudioconvert_la_SOURCES = \
gstaudioconvert.c \
audioconvert.c \
gstchannelmix.c \
gstaudioquantize.c \
plugin.c
libgstaudioconvert_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
@ -16,6 +17,7 @@ noinst_HEADERS = \
gstaudioconvert.h \
audioconvert.h \
gstchannelmix.h \
gstaudioquantize.h \
plugin.h
#TESTS = channelmixtest

240
gst/audioconvert/audioconvert.c
View file

@ -26,13 +26,10 @@
#include <string.h>
#include "gstchannelmix.h"
#include "gstaudioquantize.h"
#include "audioconvert.h"
#include "gst/floatcast/floatcast.h"
/* int to float/double conversion: int2xxx(i) = 1 / (2^31-1) * i */
#define INT2FLOAT(i) (4.6566128752457969e-10 * ((gfloat)i))
#define INT2DOUBLE(i) (4.6566128752457969e-10 * ((gdouble)i))
/* sign bit in the intermediate format */
#define SIGNED (1U<<31)
@ -43,7 +40,7 @@
audio_convert_unpack_##name
/* unpack from integer to signed integer 32 */
#define MAKE_UNPACK_FUNC_II(name, stride, sign, READ_FUNC) \
#define MAKE_UNPACK_FUNC_II(name, stride, sign, READ_FUNC) \
static void \
MAKE_UNPACK_FUNC_NAME (name) (guint8 *src, gint32 *dst, \
gint scale, gint count) \
@ -63,7 +60,7 @@ MAKE_UNPACK_FUNC_NAME (name) (type * src, gint32 * dst, gint s, gint count) \
\
for (; count; count--) { \
/* blow up to 32 bit */ \
temp = (READ_FUNC (*src++) * 2147483647.0) + 0.5; \
temp = floor ((READ_FUNC (*src++) * 2147483647.0) + 0.5); \
*dst++ = (gint32) CLAMP (temp, G_MININT32, G_MAXINT32); \
} \
}
@ -78,6 +75,20 @@ MAKE_UNPACK_FUNC_NAME (name) (type * src, gdouble * dst, gint s, \
*dst++ = (gdouble) FUNC (*src++); \
}
/* unpack from int to float 64 (double) */
#define MAKE_UNPACK_FUNC_IF(name, stride, sign, READ_FUNC) \
static void \
MAKE_UNPACK_FUNC_NAME (name) (guint8 * src, gdouble * dst, gint scale, \
gint count) \
{ \
gdouble tmp; \
for (; count; count--) { \
tmp = (gdouble) ((((gint32) READ_FUNC (src)) << scale) ^ (sign)); \
*dst++ = tmp * (1.0 / 2147483647.0); \
src += stride; \
} \
}
#define READ8(p) GST_READ_UINT8(p)
#define READ16_FROM_LE(p) GST_READ_UINT16_LE (p)
#define READ16_FROM_BE(p) GST_READ_UINT16_BE (p)
@ -108,6 +119,20 @@ MAKE_UNPACK_FUNC_FF (float_hq_le, gfloat, GFLOAT_FROM_LE);
MAKE_UNPACK_FUNC_FF (float_hq_be, gfloat, GFLOAT_FROM_BE);
MAKE_UNPACK_FUNC_FF (double_hq_le, gdouble, GDOUBLE_FROM_LE);
MAKE_UNPACK_FUNC_FF (double_hq_be, gdouble, GDOUBLE_FROM_BE);
MAKE_UNPACK_FUNC_IF (u8_float, 1, SIGNED, READ8);
MAKE_UNPACK_FUNC_IF (s8_float, 1, 0, READ8);
MAKE_UNPACK_FUNC_IF (u16_le_float, 2, SIGNED, READ16_FROM_LE);
MAKE_UNPACK_FUNC_IF (s16_le_float, 2, 0, READ16_FROM_LE);
MAKE_UNPACK_FUNC_IF (u16_be_float, 2, SIGNED, READ16_FROM_BE);
MAKE_UNPACK_FUNC_IF (s16_be_float, 2, 0, READ16_FROM_BE);
MAKE_UNPACK_FUNC_IF (u24_le_float, 3, SIGNED, READ24_FROM_LE);
MAKE_UNPACK_FUNC_IF (s24_le_float, 3, 0, READ24_FROM_LE);
MAKE_UNPACK_FUNC_IF (u24_be_float, 3, SIGNED, READ24_FROM_BE);
MAKE_UNPACK_FUNC_IF (s24_be_float, 3, 0, READ24_FROM_BE);
MAKE_UNPACK_FUNC_IF (u32_le_float, 4, SIGNED, READ32_FROM_LE);
MAKE_UNPACK_FUNC_IF (s32_le_float, 4, 0, READ32_FROM_LE);
MAKE_UNPACK_FUNC_IF (u32_be_float, 4, SIGNED, READ32_FROM_BE);
MAKE_UNPACK_FUNC_IF (s32_be_float, 4, 0, READ32_FROM_BE);
/* One of the double_hq_* functions generated above is ineffecient, but it's
* never used anyway. The same is true for one of the s32_* functions. */
@ -122,64 +147,36 @@ audio_convert_pack_##name
* These functions convert the signed 32 bit integers to the
* target format. For this to work the following steps are done:
*
* 1) If the output format is smaller than 32 bit we add 0.5LSB of
* the target format (i.e. 1<<(scale-1)) to get proper rounding.
* Shifting will result in rounding towards negative infinity (for
* signed values) or zero (for unsigned values). As we might overflow
* an overflow check is performed.
* Additionally, if our target format is signed and the value is smaller
* than zero we decrease it by one to round -X.5 downwards.
* This leads to the following rounding:
* -1.2 => -1 1.2 => 1
* -1.5 => -2 1.5 => 2
* -1.7 => -2 1.7 => 2
* 2) If the output format is unsigned we will XOR the sign bit. This
* 1) If the output format is unsigned we will XOR the sign bit. This
* will do the same as if we add 1<<31.
* 3) Afterwards we shift to the target depth. It's necessary to left-shift
* 2) Afterwards we shift to the target depth. It's necessary to left-shift
* on signed values here to get arithmetical shifting.
* 4) This is then written into our target array by the corresponding write
* 3) This is then written into our target array by the corresponding write
* function for the target width.
*/
/* pack from signed integer 32 to integer */
#define MAKE_PACK_FUNC_II(name, stride, sign, WRITE_FUNC) \
static void \
MAKE_PACK_FUNC_NAME (name) (gint32 *src, gpointer dst, \
MAKE_PACK_FUNC_NAME (name) (gint32 *src, guint8 * dst, \
gint scale, gint count) \
{ \
guint8 *p = (guint8 *)dst; \
gint32 tmp; \
if (scale > 0) { \
guint32 bias = 1 << (scale - 1); \
for (;count; count--) { \
tmp = *src++; \
if (tmp > 0 && G_MAXINT32 - tmp < bias) \
tmp = G_MAXINT32; \
else \
tmp += bias; \
if (sign == 0 && tmp < 0) \
tmp--; \
tmp = ((tmp) ^ (sign)) >> scale; \
WRITE_FUNC (p, tmp); \
p+=stride; \
} \
} else { \
for (;count; count--) { \
tmp = (*src++ ^ (sign)); \
WRITE_FUNC (p, tmp); \
p+=stride; \
} \
for (;count; count--) { \
tmp = (*src++ ^ (sign)) >> scale; \
WRITE_FUNC (dst, tmp); \
dst += stride; \
} \
}
/* pack from signed integer 32 to float */
#define MAKE_PACK_FUNC_IF(name, type, FUNC, FUNC2) \
#define MAKE_PACK_FUNC_IF(name, type, FUNC) \
static void \
MAKE_PACK_FUNC_NAME (name) (gint32 * src, type * dst, gint scale, \
gint count) \
{ \
for (; count; count--) \
*dst++ = FUNC (FUNC2 (*src++)); \
*dst++ = FUNC ((type) ((*src++) * (1.0 / 2147483647.0))); \
}
/* pack from float 64 (double) to float */
@ -192,6 +189,42 @@ MAKE_PACK_FUNC_NAME (name) (gdouble * src, type * dst, gint s, \
*dst++ = FUNC ((type) (*src++)); \
}
/* pack from float 64 (double) to signed int.
* the floats are already in the correct range. Only a cast is needed.
*/
#define MAKE_PACK_FUNC_FI_S(name, stride, WRITE_FUNC) \
static void \
MAKE_PACK_FUNC_NAME (name) (gdouble * src, guint8 * dst, gint scale, \
gint count) \
{ \
gint32 tmp; \
for (; count; count--) { \
tmp = (gint32) (*src); \
WRITE_FUNC (dst, tmp); \
src++; \
dst += stride; \
} \
}
/* pack from float 64 (double) to unsigned int.
* the floats are already in the correct range. Only a cast is needed
* and an addition of 2^(target_depth-1) to get in the correct unsigned
* range. */
#define MAKE_PACK_FUNC_FI_U(name, stride, WRITE_FUNC) \
static void \
MAKE_PACK_FUNC_NAME (name) (gdouble * src, guint8 * dst, gint scale, \
gint count) \
{ \
guint32 tmp; \
gdouble limit = (1U<<(32-scale-1)); \
for (; count; count--) { \
tmp = (guint32) (*src + limit); \
WRITE_FUNC (dst, tmp); \
src++; \
dst += stride; \
} \
}
#define WRITE8(p, v) GST_WRITE_UINT8 (p, v)
#define WRITE16_TO_LE(p,v) GST_WRITE_UINT16_LE (p, (guint16)(v))
#define WRITE16_TO_BE(p,v) GST_WRITE_UINT16_BE (p, (guint16)(v))
@ -214,12 +247,27 @@ MAKE_PACK_FUNC_II (u32_le, 4, SIGNED, WRITE32_TO_LE);
MAKE_PACK_FUNC_II (s32_le, 4, 0, WRITE32_TO_LE);
MAKE_PACK_FUNC_II (u32_be, 4, SIGNED, WRITE32_TO_BE);
MAKE_PACK_FUNC_II (s32_be, 4, 0, WRITE32_TO_BE);
MAKE_PACK_FUNC_IF (float_le, gfloat, GFLOAT_TO_LE, INT2FLOAT);
MAKE_PACK_FUNC_IF (float_be, gfloat, GFLOAT_TO_BE, INT2FLOAT);
MAKE_PACK_FUNC_IF (double_le, gdouble, GDOUBLE_TO_LE, INT2DOUBLE);
MAKE_PACK_FUNC_IF (double_be, gdouble, GDOUBLE_TO_BE, INT2DOUBLE);
MAKE_PACK_FUNC_IF (float_le, gfloat, GFLOAT_TO_LE);
MAKE_PACK_FUNC_IF (float_be, gfloat, GFLOAT_TO_BE);
MAKE_PACK_FUNC_IF (double_le, gdouble, GDOUBLE_TO_LE);
MAKE_PACK_FUNC_IF (double_be, gdouble, GDOUBLE_TO_BE);
MAKE_PACK_FUNC_FF (float_hq_le, gfloat, GFLOAT_TO_LE);
MAKE_PACK_FUNC_FF (float_hq_be, gfloat, GFLOAT_TO_BE);
MAKE_PACK_FUNC_FI_U (u8_float, 1, WRITE8);
MAKE_PACK_FUNC_FI_S (s8_float, 1, WRITE8);
MAKE_PACK_FUNC_FI_U (u16_le_float, 2, WRITE16_TO_LE);
MAKE_PACK_FUNC_FI_S (s16_le_float, 2, WRITE16_TO_LE);
MAKE_PACK_FUNC_FI_U (u16_be_float, 2, WRITE16_TO_BE);
MAKE_PACK_FUNC_FI_S (s16_be_float, 2, WRITE16_TO_BE);
MAKE_PACK_FUNC_FI_U (u24_le_float, 3, WRITE24_TO_LE);
MAKE_PACK_FUNC_FI_S (s24_le_float, 3, WRITE24_TO_LE);
MAKE_PACK_FUNC_FI_U (u24_be_float, 3, WRITE24_TO_BE);
MAKE_PACK_FUNC_FI_S (s24_be_float, 3, WRITE24_TO_BE);
MAKE_PACK_FUNC_FI_U (u32_le_float, 4, WRITE32_TO_LE);
MAKE_PACK_FUNC_FI_S (s32_le_float, 4, WRITE32_TO_LE);
MAKE_PACK_FUNC_FI_U (u32_be_float, 4, WRITE32_TO_BE);
MAKE_PACK_FUNC_FI_S (s32_be_float, 4, WRITE32_TO_BE);
/* For double_hq, packing and unpacking is the same, so we reuse the unpacking
* functions here. */
#define audio_convert_pack_double_hq_le MAKE_UNPACK_FUNC_NAME (double_hq_le)
@ -250,6 +298,22 @@ static AudioConvertUnpack unpack_funcs[] = {
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (float_hq_be),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (double_hq_le),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (double_hq_be),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (u8_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (s8_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (u8_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (s8_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (u16_le_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (s16_le_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (u16_be_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (s16_be_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (u24_le_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (s24_le_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (u24_be_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (s24_be_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (u32_le_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (s32_le_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (u32_be_float),
(AudioConvertUnpack) MAKE_UNPACK_FUNC_NAME (s32_be_float),
};
static AudioConvertPack pack_funcs[] = {
@ -277,10 +341,29 @@ static AudioConvertPack pack_funcs[] = {
(AudioConvertPack) MAKE_PACK_FUNC_NAME (float_hq_be),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (double_hq_le),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (double_hq_be),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (u8_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (s8_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (u8_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (s8_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (u16_le_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (s16_le_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (u16_be_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (s16_be_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (u24_le_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (s24_le_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (u24_be_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (s24_be_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (u32_le_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (s32_le_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (u32_be_float),
(AudioConvertPack) MAKE_PACK_FUNC_NAME (s32_be_float),
};
#define DOUBLE_INTERMEDIATE_FORMAT(ctx) \
((!ctx->in.is_int && !ctx->out.is_int) || (ctx->ns != NOISE_SHAPING_NONE))
static gint
audio_convert_get_func_index (AudioConvertFmt * fmt)
audio_convert_get_func_index (AudioConvertCtx * ctx, AudioConvertFmt * fmt)
{
gint index = 0;
@ -288,19 +371,22 @@ audio_convert_get_func_index (AudioConvertFmt * fmt)
index += (fmt->width / 8 - 1) * 4;
index += fmt->endianness == G_LITTLE_ENDIAN ? 0 : 2;
index += fmt->sign ? 1 : 0;
index += (ctx->ns == NOISE_SHAPING_NONE) ? 0 : 24;
} else {
/* this is float/double */
index = 16;
index += (fmt->width == 32) ? 0 : 2;
index += (fmt->endianness == G_LITTLE_ENDIAN) ? 0 : 1;
index += (DOUBLE_INTERMEDIATE_FORMAT (ctx)) ? 4 : 0;
}
return index;
}
static gboolean
static inline gboolean
check_default (AudioConvertCtx * ctx, AudioConvertFmt * fmt)
{
if (ctx->in.is_int || ctx->out.is_int) {
if (!DOUBLE_INTERMEDIATE_FORMAT (ctx)) {
return (fmt->width == 32 && fmt->depth == 32 &&
fmt->endianness == G_BYTE_ORDER && fmt->sign == TRUE);
} else {
@ -322,7 +408,7 @@ audio_convert_clean_fmt (AudioConvertFmt * fmt)
gboolean
audio_convert_prepare_context (AudioConvertCtx * ctx, AudioConvertFmt * in,
AudioConvertFmt * out)
AudioConvertFmt * out, DitherType dither, NoiseShapingType ns)
{
gint idx_in, idx_out;
@ -336,26 +422,40 @@ audio_convert_prepare_context (AudioConvertCtx * ctx, AudioConvertFmt * in,
ctx->in = *in;
ctx->out = *out;
/* Don't dither or apply noise shaping if out depth is bigger than 20 bits
* as DA converters only can do a SNR up to 20 bits in reality.
* Also don't dither or apply noise shaping if target depth is larger than
* source depth. */
if (ctx->out.depth <= 20 && (!ctx->in.is_int
|| ctx->in.depth >= ctx->out.depth)) {
ctx->dither = dither;
ctx->ns = ns;
} else {
ctx->dither = DITHER_NONE;
ctx->ns = NOISE_SHAPING_NONE;
}
/* Use simple error feedback when output sample rate is smaller than
* 32000 as the other methods might move the noise to audible ranges */
if (ctx->ns > NOISE_SHAPING_ERROR_FEEDBACK && ctx->out.rate < 32000)
ctx->ns = NOISE_SHAPING_ERROR_FEEDBACK;
gst_channel_mix_setup_matrix (ctx);
idx_in = audio_convert_get_func_index (in);
idx_in = audio_convert_get_func_index (ctx, in);
ctx->unpack = unpack_funcs[idx_in];
idx_out = audio_convert_get_func_index (out);
idx_out = audio_convert_get_func_index (ctx, out);
ctx->pack = pack_funcs[idx_out];
/* if both formats are float/double use double as intermediate format and
* and switch mixing */
if (in->is_int || out->is_int) {
/* if both formats are float/double or we use noise shaping use double as
* intermediate format and and switch mixing */
if (!DOUBLE_INTERMEDIATE_FORMAT (ctx)) {
GST_INFO ("use int mixing");
ctx->channel_mix = (AudioConvertMix) gst_channel_mix_mix_int;
} else {
GST_INFO ("use float mixing");
ctx->channel_mix = (AudioConvertMix) gst_channel_mix_mix_float;
/* Bump the pack/unpack function indices by 4 to use double as intermediary
* format (float_hq_*, double_hq_* functions).*/
ctx->unpack = unpack_funcs[idx_in + 4];
ctx->pack = pack_funcs[idx_out + 4];
}
GST_INFO ("unitsizes: %d -> %d", in->unit_size, out->unit_size);
@ -372,6 +472,8 @@ audio_convert_prepare_context (AudioConvertCtx * ctx, AudioConvertFmt * in,
ctx->in_scale = (in->is_int) ? (32 - in->depth) : 0;
ctx->out_scale = (out->is_int) ? (32 - out->depth) : 0;
gst_audio_quantize_setup (ctx);
return TRUE;
}
@ -380,6 +482,7 @@ audio_convert_clean_context (AudioConvertCtx * ctx)
{
g_return_val_if_fail (ctx != NULL, FALSE);
gst_audio_quantize_free (ctx);
audio_convert_clean_fmt (&ctx->in);
audio_convert_clean_fmt (&ctx->out);
gst_channel_mix_unset_matrix (ctx);
@ -425,12 +528,12 @@ audio_convert_convert (AudioConvertCtx * ctx, gpointer src,
outsize = ctx->out.unit_size * samples;
/* find biggest temp buffer size */
size = (ctx->in.is_int || ctx->out.is_int) ?
sizeof (gint32) : sizeof (gdouble);
size = (DOUBLE_INTERMEDIATE_FORMAT (ctx)) ? sizeof (gdouble)
: sizeof (gint32);
if (!ctx->in_default)
intemp = insize * size * 8 / ctx->in.width;
if (!ctx->mix_passthrough)
if (!ctx->mix_passthrough || !ctx->out_default)
outtemp = outsize * size * 8 / ctx->out.width;
biggest = MAX (intemp, outtemp);
@ -474,6 +577,15 @@ audio_convert_convert (AudioConvertCtx * ctx, gpointer src,
src = outbuf;
}
/* we only need to quantize if output format is int */
if (ctx->out.is_int) {
if (ctx->out_default)
outbuf = dst;
else
outbuf = tmpbuf;
ctx->quantize (ctx, src, outbuf, samples);
}
if (!ctx->out_default) {
/* pack default format into dst */
ctx->pack (src, dst, ctx->out_scale, samples * ctx->out.channels);

54
gst/audioconvert/audioconvert.h
View file

@ -25,6 +25,23 @@
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
typedef enum
{
DITHER_NONE = 0,
DITHER_RPDF,
DITHER_TPDF,
DITHER_TPDF_HF
} DitherType;
typedef enum
{
NOISE_SHAPING_NONE = 0,
NOISE_SHAPING_ERROR_FEEDBACK,
NOISE_SHAPING_SIMPLE,
NOISE_SHAPING_MEDIUM,
NOISE_SHAPING_HIGH
} NoiseShapingType;
typedef struct _AudioConvertCtx AudioConvertCtx;
typedef struct _AudioConvertFmt AudioConvertFmt;
@ -45,10 +62,14 @@ struct _AudioConvertFmt
gint unit_size;
};
typedef void (*AudioConvertUnpack) (gpointer src, gpointer dst, gint scale, gint count);
typedef void (*AudioConvertPack) (gpointer src, gpointer dst, gint scale, gint count);
typedef void (*AudioConvertUnpack) (gpointer src, gpointer dst, gint scale,
gint count);
typedef void (*AudioConvertPack) (gpointer src, gpointer dst, gint scale,
gint count);
typedef void (*AudioConvertMix) (AudioConvertCtx *, gpointer, gpointer, gint);
typedef void (*AudioConvertQuantize) (AudioConvertCtx * ctx, gpointer src,
gpointer dst, gint count);
struct _AudioConvertCtx
{
@ -73,20 +94,31 @@ struct _AudioConvertCtx
gint in_scale;
gint out_scale;
AudioConvertMix channel_mix;
AudioConvertQuantize quantize;
DitherType dither;
NoiseShapingType ns;
/* random number generate for dither noise */
GRand *dither_random;
/* last random number generated per channel for hifreq TPDF dither */
gpointer last_random;
/* contains the past quantization errors, error[out_channels][count] */
gdouble *error_buf;
};
gboolean audio_convert_clean_fmt (AudioConvertFmt *fmt);
gboolean audio_convert_clean_fmt (AudioConvertFmt * fmt);
gboolean audio_convert_prepare_context (AudioConvertCtx *ctx, AudioConvertFmt *in,
AudioConvertFmt *out);
gboolean audio_convert_get_sizes (AudioConvertCtx *ctx, gint samples, gint *srcsize,
gint *dstsize);
gboolean audio_convert_prepare_context (AudioConvertCtx * ctx,
AudioConvertFmt * in, AudioConvertFmt * out, DitherType dither,
NoiseShapingType ns);
gboolean audio_convert_get_sizes (AudioConvertCtx * ctx, gint samples,
gint * srcsize, gint * dstsize);
gboolean audio_convert_clean_context (AudioConvertCtx *ctx);
gboolean audio_convert_clean_context (AudioConvertCtx * ctx);
gboolean audio_convert_convert (AudioConvertCtx *ctx, gpointer src,
gpointer dst, gint samples, gboolean src_writable);
gboolean audio_convert_convert (AudioConvertCtx * ctx, gpointer src,
gpointer dst, gint samples, gboolean src_writable);
#endif /* __AUDIO_CONVERT_H__ */

109
gst/audioconvert/gstaudioconvert.c
View file

@ -74,6 +74,7 @@
#include "gstaudioconvert.h"
#include "gstchannelmix.h"
#include "gstaudioquantize.h"
#include "plugin.h"
GST_DEBUG_CATEGORY (audio_convert_debug);
@ -102,6 +103,11 @@ static GstFlowReturn gst_audio_convert_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
static GstFlowReturn gst_audio_convert_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static void gst_audio_convert_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audio_convert_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
/* AudioConvert signals and args */
enum
@ -113,7 +119,8 @@ enum
enum
{
ARG_0,
ARG_AGGRESSIVE
ARG_DITHERING,
ARG_NOISE_SHAPING,
};
#define DEBUG_INIT(bla) \
@ -179,6 +186,50 @@ GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_ALWAYS,
STATIC_CAPS);
#define GST_TYPE_AUDIO_CONVERT_DITHERING (gst_audio_convert_dithering_get_type ())
static GType
gst_audio_convert_dithering_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{DITHER_NONE, "No dithering",
"none"},
{DITHER_RPDF, "Rectangular dithering", "rpdf"},
{DITHER_TPDF, "Triangular dithering (default)", "tpdf"},
{DITHER_TPDF_HF, "High frequency triangular dithering", "tpdf-hf"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioConvertDithering", values);
}
return gtype;
}
#define GST_TYPE_AUDIO_CONVERT_NOISE_SHAPING (gst_audio_convert_ns_get_type ())
static GType
gst_audio_convert_ns_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{NOISE_SHAPING_NONE, "No noise shaping (default)",
"none"},
{NOISE_SHAPING_ERROR_FEEDBACK, "Error feedback", "error-feedback"},
{NOISE_SHAPING_SIMPLE, "Simple 2-pole noise shaping", "simple"},
{NOISE_SHAPING_MEDIUM, "Medium 5-pole noise shaping", "medium"},
{NOISE_SHAPING_HIGH, "High 8-pole noise shaping", "high"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioConvertNoiseShaping", values);
}
return gtype;
}
/*** TYPE FUNCTIONS ***********************************************************/
static void
@ -201,12 +252,25 @@ gst_audio_convert_class_init (GstAudioConvertClass * klass)
gint i;
gobject_class->dispose = gst_audio_convert_dispose;
gobject_class->set_property = gst_audio_convert_set_property;
gobject_class->get_property = gst_audio_convert_get_property;
supported_positions = g_new0 (GstAudioChannelPosition,
GST_AUDIO_CHANNEL_POSITION_NUM);
for (i = 0; i < GST_AUDIO_CHANNEL_POSITION_NUM; i++)
supported_positions[i] = i;
g_object_class_install_property (gobject_class, ARG_DITHERING,
g_param_spec_enum ("dithering", "Dithering",
"Selects between different dithering methods.",
GST_TYPE_AUDIO_CONVERT_DITHERING, DITHER_TPDF, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_NOISE_SHAPING,
g_param_spec_enum ("noise-shaping", "Noise shaping",
"Selects between different noise shaping methods.",
GST_TYPE_AUDIO_CONVERT_NOISE_SHAPING, NOISE_SHAPING_NONE,
G_PARAM_READWRITE));
basetransform_class->get_unit_size =
GST_DEBUG_FUNCPTR (gst_audio_convert_get_unit_size);
basetransform_class->transform_caps =
@ -226,6 +290,8 @@ gst_audio_convert_class_init (GstAudioConvertClass * klass)
static void
gst_audio_convert_init (GstAudioConvert * this, GstAudioConvertClass * g_class)
{
this->dither = DITHER_TPDF;
this->ns = NOISE_SHAPING_NONE;
memset (&this->ctx, 0, sizeof (AudioConvertCtx));
}
@ -672,7 +738,8 @@ gst_audio_convert_set_caps (GstBaseTransform * base, GstCaps * incaps,
if (!gst_audio_convert_parse_caps (outcaps, &out_ac_caps))
return FALSE;
if (!audio_convert_prepare_context (&this->ctx, &in_ac_caps, &out_ac_caps))
if (!audio_convert_prepare_context (&this->ctx, &in_ac_caps, &out_ac_caps,
this->dither, this->ns))
goto no_converter;
return TRUE;
@ -753,3 +820,41 @@ convert_error:
return GST_FLOW_ERROR;
}
}
static void
gst_audio_convert_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioConvert *this = GST_AUDIO_CONVERT (object);
switch (prop_id) {
case ARG_DITHERING:
this->dither = g_value_get_enum (value);
break;
case ARG_NOISE_SHAPING:
this->ns = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_convert_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioConvert *this = GST_AUDIO_CONVERT (object);
switch (prop_id) {
case ARG_DITHERING:
g_value_set_enum (value, this->dither);
break;
case ARG_NOISE_SHAPING:
g_value_set_enum (value, this->ns);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}

3
gst/audioconvert/gstaudioconvert.h
View file

@ -47,6 +47,9 @@ struct _GstAudioConvert
GstBaseTransform element;
AudioConvertCtx ctx;
DitherType dither;
NoiseShapingType ns;
};
struct _GstAudioConvertClass

511
gst/audioconvert/gstaudioquantize.c Normal file
View file

@ -0,0 +1,511 @@
/* GStreamer
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/*
* FIXME: When doing dithering with int as intermediate format
* one gets audible harmonics while the noise floor is
* constant for double as intermediate format!
*/
/* 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 "audioconvert.h"
#include "gstaudioquantize.h"
#define MAKE_QUANTIZE_FUNC_NAME(name) \
gst_audio_quantize_quantize_##name
/* Quantize functions for gint32 as intermediate format */
#define MAKE_QUANTIZE_FUNC_I(name, DITHER_INIT_FUNC, ADD_DITHER_FUNC, \
ROUND_FUNC) \
static void \
MAKE_QUANTIZE_FUNC_NAME (name) (AudioConvertCtx *ctx, gint32 *src, \
gint32 *dst, gint count) \
{ \
gint scale = ctx->out_scale; \
gint channels = ctx->out.channels; \
gint chan_pos; \
\
if (scale > 0) { \
gint32 tmp; \
guint32 mask = 0xffffffff & (0xffffffff << scale); \
guint32 bias = 1U << (scale - 1); \
DITHER_INIT_FUNC(); \
\
for (;count;count--) { \
for (chan_pos = 0; chan_pos < channels; chan_pos++) { \
tmp = *src++; \
ADD_DITHER_FUNC(); \
ROUND_FUNC(); \
*dst = tmp & mask; \
dst++; \
} \
} \
} else { \
for (;count;count--) { \
for (chan_pos = 0; chan_pos < channels; chan_pos++) { \
*dst = *src++; \
dst++; \
} \
} \
} \
}
/* Quantize functions for gdouble as intermediate format with
* int as target */
#define MAKE_QUANTIZE_FUNC_F(name, DITHER_INIT_FUNC, NS_INIT_FUNC, \
ADD_NS_FUNC, ADD_DITHER_FUNC, \
UPDATE_ERROR_FUNC) \
static void \
MAKE_QUANTIZE_FUNC_NAME (name) (AudioConvertCtx *ctx, gdouble *src, \
gdouble *dst, gint count) \
{ \
gint scale = ctx->out_scale; \
gint channels = ctx->out.channels; \
gint chan_pos; \
gdouble factor = (1U<<(32-scale-1)) - 1; \
\
if (scale > 0) { \
gdouble tmp; \
DITHER_INIT_FUNC(); \
NS_INIT_FUNC(); \
\
for (;count;count--) { \
for (chan_pos = 0; chan_pos < channels; chan_pos++) { \
tmp = *src++; \
ADD_NS_FUNC(); \
ADD_DITHER_FUNC(); \
tmp = floor(tmp * factor + 0.5); \
*dst = CLAMP (tmp, -factor - 1, factor); \
UPDATE_ERROR_FUNC(); \
dst++; \
} \
} \
} else { \
for (;count;count--) { \
for (chan_pos = 0; chan_pos < channels; chan_pos++) { \
*dst = *src++ * 2147483647.0; \
dst++; \
} \
} \
} \
}
/* Rounding functions for int as intermediate format, only used when
* not using dithering. With dithering we include this offset in our
* dither noise instead. */
#define ROUND() \
if (tmp > 0 && G_MAXINT32 - tmp <= bias) \
tmp = G_MAXINT32; \
else \
tmp += bias;
#define NONE_FUNC()
/* Dithering definitions
* See http://en.wikipedia.org/wiki/Dithering or
* http://www.cadenzarecording.com/Dither.html for explainations.
*
* We already add the rounding offset to the dither noise here
* to have only one overflow check instead of two. */
#define INIT_DITHER_RPDF_I() \
gint32 rand; \
gint32 dither = (1<<(scale));
#define ADD_DITHER_RPDF_I() \
rand = g_rand_int_range (ctx->dither_random, bias - dither, \
bias + dither); \
if (rand > 0 && tmp > 0 && G_MAXINT32 - tmp <= rand) \
tmp = G_MAXINT32; \
else if (rand < 0 && tmp < 0 && G_MININT32 - tmp >= rand) \
tmp = G_MININT32; \
else \
tmp += rand;
#define INIT_DITHER_RPDF_F() \
gdouble dither = 1.0/(1U<<(32 - scale - 1));
#define ADD_DITHER_RPDF_F() \
tmp += g_rand_double_range (ctx->dither_random, - dither, \
dither);
#define INIT_DITHER_TPDF_I() \
gint32 rand; \
gint32 dither = (1<<(scale - 1)); \
bias = bias >> 1;
#define ADD_DITHER_TPDF_I() \
rand = g_rand_int_range (ctx->dither_random, bias - dither, \
bias + dither - 1) \
+ g_rand_int_range (ctx->dither_random, bias - dither, \
bias + dither - 1); \
if (rand > 0 && tmp > 0 && G_MAXINT32 - tmp <= rand) \
tmp = G_MAXINT32; \
else if (rand < 0 && tmp < 0 && G_MININT32 - tmp >= rand) \
tmp = G_MININT32; \
else \
tmp += rand;
#define INIT_DITHER_TPDF_F() \
gdouble dither = 1.0/(1U<<(32 - scale));
#define ADD_DITHER_TPDF_F() \
tmp += g_rand_double_range (ctx->dither_random, - dither, \
dither) \
+ g_rand_double_range (ctx->dither_random, - dither, \
dither);
#define INIT_DITHER_TPDF_HF_I() \
gint32 rand; \
gint32 dither = (1<<(scale-1)); \
gint32 *last_random = (gint32 *) ctx->last_random, tmp_rand; \
bias = bias >> 1;
#define ADD_DITHER_TPDF_HF_I() \
tmp_rand = g_rand_int_range (ctx->dither_random, bias - dither, \
bias + dither); \
rand = tmp_rand - last_random[chan_pos]; \
last_random[chan_pos] = tmp_rand; \
if (rand > 0 && tmp > 0 && G_MAXINT32 - tmp <= rand) \
tmp = G_MAXINT32; \
else if (rand < 0 && tmp < 0 && G_MININT32 - tmp >= rand) \
tmp = G_MININT32; \
else \
tmp += rand;
/* Like TPDF dither but the dither noise is oriented more to the
* higher frequencies */
#define INIT_DITHER_TPDF_HF_F() \
gdouble rand; \
gdouble dither = 1.0/(1U<<(32 - scale)); \
gdouble *last_random = (gdouble *) ctx->last_random, tmp_rand;
#define ADD_DITHER_TPDF_HF_F() \
tmp_rand = g_rand_double_range (ctx->dither_random, - dither, \
dither); \
rand = tmp_rand - last_random[chan_pos]; \
last_random[chan_pos] = tmp_rand; \
tmp += rand;
/* Noise shaping definitions.
* See http://en.wikipedia.org/wiki/Noise_shaping for explainations. */
/* Simple error feedback: Just accumulate the dithering and quantization
* error and remove it from each sample. */
#define INIT_NS_ERROR_FEEDBACK() \
gdouble orig; \
gdouble *errors = ctx->error_buf;
#define ADD_NS_ERROR_FEEDBACK() \
orig = tmp; \
tmp -= errors[chan_pos];
#define UPDATE_ERROR_ERROR_FEEDBACK() \
errors[chan_pos] += (*dst)/factor - orig;
/* 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. */
#define INIT_NS_SIMPLE() \
gdouble orig; \
gdouble *errors = ctx->error_buf, cur_error;
#define ADD_NS_SIMPLE() \
cur_error = errors[chan_pos*2] - 0.5 * errors[chan_pos*2 + 1]; \
tmp -= cur_error; \
orig = tmp;
#define UPDATE_ERROR_SIMPLE() \
errors[chan_pos*2 + 1] = errors[chan_pos*2]; \
errors[chan_pos*2] = (*dst)/factor - orig;
/* 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[] = {
2.033, -2.165, 1.959, -1.590, 0.6149
};
#define INIT_NS_MEDIUM() \
gdouble orig; \
gdouble *errors = ctx->error_buf, cur_error; \
int j;
#define ADD_NS_MEDIUM() \
cur_error = 0.0; \
for (j = 0; j < 5; j++) \
cur_error += errors[chan_pos*5 + j] * ns_medium_coeffs[j]; \
tmp -= cur_error; \
orig = tmp;
#define UPDATE_ERROR_MEDIUM() \
for (j = 4; j > 0; j--) \
errors[chan_pos*5 + j] = errors[chan_pos*5 + j-1]; \
errors[chan_pos*5] = (*dst)/factor - orig;
/* Noise shaping coefficients by David Schleef, moves most power of the
* error noise into inaudible frequency ranges */
static const gdouble ns_high_coeffs[] = {
2.08484, -2.92975, 3.27918, -3.31399, 2.61339, -1.72008, 0.876066, -0.340122
};
#define INIT_NS_HIGH() \
gdouble orig; \
gdouble *errors = ctx->error_buf, cur_error; \
int j;
#define ADD_NS_HIGH() \
cur_error = 0.0; \
for (j = 0; j < 8; j++) \
cur_error += errors[chan_pos + j] * ns_high_coeffs[j]; \
tmp -= cur_error; \
orig = tmp;
#define UPDATE_ERROR_HIGH() \
for (j = 7; j > 0; j--) \
errors[chan_pos + j] = errors[chan_pos + j-1]; \
errors[chan_pos] = (*dst)/factor - orig;
MAKE_QUANTIZE_FUNC_I (signed_none_none, NONE_FUNC, NONE_FUNC, ROUND);
MAKE_QUANTIZE_FUNC_I (signed_rpdf_none, INIT_DITHER_RPDF_I, ADD_DITHER_RPDF_I,
NONE_FUNC);
MAKE_QUANTIZE_FUNC_I (signed_tpdf_none, INIT_DITHER_TPDF_I, ADD_DITHER_TPDF_I,
NONE_FUNC);
MAKE_QUANTIZE_FUNC_I (signed_tpdf_hf_none, INIT_DITHER_TPDF_HF_I,
ADD_DITHER_TPDF_HF_I, NONE_FUNC);
MAKE_QUANTIZE_FUNC_I (unsigned_none_none, NONE_FUNC, NONE_FUNC, ROUND);
MAKE_QUANTIZE_FUNC_I (unsigned_rpdf_none, INIT_DITHER_RPDF_I, ADD_DITHER_RPDF_I,
NONE_FUNC);
MAKE_QUANTIZE_FUNC_I (unsigned_tpdf_none, INIT_DITHER_TPDF_I, ADD_DITHER_TPDF_I,
NONE_FUNC);
MAKE_QUANTIZE_FUNC_I (unsigned_tpdf_hf_none, INIT_DITHER_TPDF_HF_I,
ADD_DITHER_TPDF_HF_I, NONE_FUNC);
MAKE_QUANTIZE_FUNC_F (float_none_error_feedback, NONE_FUNC,
INIT_NS_ERROR_FEEDBACK, ADD_NS_ERROR_FEEDBACK, NONE_FUNC,
UPDATE_ERROR_ERROR_FEEDBACK);
MAKE_QUANTIZE_FUNC_F (float_none_simple, NONE_FUNC, INIT_NS_SIMPLE,
ADD_NS_SIMPLE, NONE_FUNC, UPDATE_ERROR_SIMPLE);
MAKE_QUANTIZE_FUNC_F (float_none_medium, NONE_FUNC, INIT_NS_MEDIUM,
ADD_NS_MEDIUM, NONE_FUNC, UPDATE_ERROR_MEDIUM);
MAKE_QUANTIZE_FUNC_F (float_none_high, NONE_FUNC, INIT_NS_HIGH, ADD_NS_HIGH,
NONE_FUNC, UPDATE_ERROR_HIGH);
MAKE_QUANTIZE_FUNC_F (float_rpdf_error_feedback, INIT_DITHER_RPDF_F,
INIT_NS_ERROR_FEEDBACK, ADD_NS_ERROR_FEEDBACK, ADD_DITHER_RPDF_F,
UPDATE_ERROR_ERROR_FEEDBACK);
MAKE_QUANTIZE_FUNC_F (float_rpdf_simple, INIT_DITHER_RPDF_F, INIT_NS_SIMPLE,
ADD_NS_SIMPLE, ADD_DITHER_RPDF_F, UPDATE_ERROR_SIMPLE);
MAKE_QUANTIZE_FUNC_F (float_rpdf_medium, INIT_DITHER_RPDF_F, INIT_NS_MEDIUM,
ADD_NS_MEDIUM, ADD_DITHER_RPDF_F, UPDATE_ERROR_MEDIUM);
MAKE_QUANTIZE_FUNC_F (float_rpdf_high, INIT_DITHER_RPDF_F, INIT_NS_HIGH,
ADD_NS_HIGH, ADD_DITHER_RPDF_F, UPDATE_ERROR_HIGH);
MAKE_QUANTIZE_FUNC_F (float_tpdf_error_feedback, INIT_DITHER_TPDF_F,
INIT_NS_ERROR_FEEDBACK, ADD_NS_ERROR_FEEDBACK, ADD_DITHER_TPDF_F,
UPDATE_ERROR_ERROR_FEEDBACK);
MAKE_QUANTIZE_FUNC_F (float_tpdf_simple, INIT_DITHER_TPDF_F, INIT_NS_SIMPLE,
ADD_NS_SIMPLE, ADD_DITHER_TPDF_F, UPDATE_ERROR_SIMPLE);
MAKE_QUANTIZE_FUNC_F (float_tpdf_medium, INIT_DITHER_TPDF_F, INIT_NS_MEDIUM,
ADD_NS_MEDIUM, ADD_DITHER_TPDF_F, UPDATE_ERROR_MEDIUM);
MAKE_QUANTIZE_FUNC_F (float_tpdf_high, INIT_DITHER_TPDF_F, INIT_NS_HIGH,
ADD_NS_HIGH, ADD_DITHER_TPDF_F, UPDATE_ERROR_HIGH);
MAKE_QUANTIZE_FUNC_F (float_tpdf_hf_error_feedback, INIT_DITHER_TPDF_HF_F,
INIT_NS_ERROR_FEEDBACK, ADD_NS_ERROR_FEEDBACK, ADD_DITHER_TPDF_HF_F,
UPDATE_ERROR_ERROR_FEEDBACK);
MAKE_QUANTIZE_FUNC_F (float_tpdf_hf_simple, INIT_DITHER_TPDF_HF_F,
INIT_NS_SIMPLE, ADD_NS_SIMPLE, ADD_DITHER_TPDF_HF_F, UPDATE_ERROR_SIMPLE);
MAKE_QUANTIZE_FUNC_F (float_tpdf_hf_medium, INIT_DITHER_TPDF_HF_F,
INIT_NS_MEDIUM, ADD_NS_MEDIUM, ADD_DITHER_TPDF_HF_F, UPDATE_ERROR_MEDIUM);
MAKE_QUANTIZE_FUNC_F (float_tpdf_hf_high, INIT_DITHER_TPDF_HF_F, INIT_NS_HIGH,
ADD_NS_HIGH, ADD_DITHER_TPDF_HF_F, UPDATE_ERROR_HIGH);
static AudioConvertQuantize quantize_funcs[] = {
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (signed_none_none),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (signed_rpdf_none),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (signed_tpdf_none),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (signed_tpdf_hf_none),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (unsigned_none_none),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (unsigned_rpdf_none),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (unsigned_tpdf_none),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (unsigned_tpdf_hf_none),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_none_error_feedback),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_none_simple),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_none_medium),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_none_high),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_rpdf_error_feedback),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_rpdf_simple),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_rpdf_medium),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_rpdf_high),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_tpdf_error_feedback),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_tpdf_simple),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_tpdf_medium),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_tpdf_high),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_tpdf_hf_error_feedback),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_tpdf_hf_simple),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_tpdf_hf_medium),
(AudioConvertQuantize) MAKE_QUANTIZE_FUNC_NAME (float_tpdf_hf_high),
};
static void
gst_audio_quantize_setup_noise_shaping (AudioConvertCtx * ctx)
{
switch (ctx->ns) {
case NOISE_SHAPING_HIGH:{
ctx->error_buf = g_new0 (gdouble, ctx->out.channels * 8);
break;
}
case NOISE_SHAPING_MEDIUM:{
ctx->error_buf = g_new0 (gdouble, ctx->out.channels * 5);
break;
}
case NOISE_SHAPING_SIMPLE:{
ctx->error_buf = g_new0 (gdouble, ctx->out.channels * 2);
break;
}
case NOISE_SHAPING_ERROR_FEEDBACK:
ctx->error_buf = g_new0 (gdouble, ctx->out.channels);
break;
case NOISE_SHAPING_NONE:
default:
ctx->error_buf = NULL;
break;
}
return;
}
static void
gst_audio_quantize_free_noise_shaping (AudioConvertCtx * ctx)
{
switch (ctx->ns) {
case NOISE_SHAPING_HIGH:
case NOISE_SHAPING_MEDIUM:
case NOISE_SHAPING_SIMPLE:
case NOISE_SHAPING_ERROR_FEEDBACK:
case NOISE_SHAPING_NONE:
default:
break;
}
return;
g_free (ctx->error_buf);
ctx->error_buf = NULL;
return;
}
static void
gst_audio_quantize_setup_dither (AudioConvertCtx * ctx)
{
switch (ctx->dither) {
case DITHER_TPDF_HF:
if (ctx->out.is_int)
ctx->last_random = g_new0 (gint32, ctx->out.channels);
else
ctx->last_random = g_new0 (gdouble, ctx->out.channels);
ctx->dither_random = g_rand_new ();
break;
case DITHER_RPDF:
case DITHER_TPDF:
ctx->dither_random = g_rand_new ();
ctx->last_random = NULL;
break;
case DITHER_NONE:
default:
ctx->dither_random = NULL;
ctx->last_random = NULL;
break;
}
return;
}
static void
gst_audio_quantize_free_dither (AudioConvertCtx * ctx)
{
g_free (ctx->last_random);
if (ctx->dither_random)
g_rand_free (ctx->dither_random);
return;
}
static void
gst_audio_quantize_setup_quantize_func (AudioConvertCtx * ctx)
{
gint index = 0;
if (!ctx->out.is_int) {
ctx->quantize = NULL;
return;
}
if (ctx->ns == NOISE_SHAPING_NONE) {
index += ctx->dither;
index += (ctx->out.sign) ? 0 : 4;
} else {
index += 8 + (4 * ctx->dither);
index += ctx->ns - 1;
}
ctx->quantize = quantize_funcs[index];
}
gboolean
gst_audio_quantize_setup (AudioConvertCtx * ctx)
{
gst_audio_quantize_setup_dither (ctx);
gst_audio_quantize_setup_noise_shaping (ctx);
gst_audio_quantize_setup_quantize_func (ctx);
return TRUE;
}
void
gst_audio_quantize_free (AudioConvertCtx * ctx)
{
gst_audio_quantize_free_dither (ctx);
gst_audio_quantize_free_noise_shaping (ctx);
}

37
gst/audioconvert/gstaudioquantize.h Normal file
View file

@ -0,0 +1,37 @@
/* GStreamer
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* gstaudioquantize.h: 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <gst/gst.h>
#include "audioconvert.h"
GST_DEBUG_CATEGORY_EXTERN (audio_convert_debug);
#define GST_CAT_DEFAULT (audio_convert_debug)
#ifndef __GST_AUDIO_QUANTIZE_H__
#define __GST_AUDIO_QUANTIZE_H__
gboolean gst_audio_quantize_setup (AudioConvertCtx * ctx);
void gst_audio_quantize_reset (AudioConvertCtx * ctx);
void gst_audio_quantize_free (AudioConvertCtx * ctx);
#endif /* __GST_AUDIO_QUANTIZE_H__ */

16
tests/check/elements/audioconvert.c
View file

@ -87,6 +87,8 @@ setup_audioconvert (GstCaps * outcaps)
GST_DEBUG ("setup_audioconvert with caps %" GST_PTR_FORMAT, outcaps);
audioconvert = gst_check_setup_element ("audioconvert");
g_object_set (G_OBJECT (audioconvert), "dithering", 0, NULL);
g_object_set (G_OBJECT (audioconvert), "noise-shaping", 0, NULL);
mysrcpad = gst_check_setup_src_pad (audioconvert, &srctemplate, NULL);
mysinkpad = gst_check_setup_sink_pad (audioconvert, &sinktemplate, NULL);
/* this installs a getcaps func that will always return the caps we set
@ -532,7 +534,7 @@ GST_START_TEST (test_int_conversion)
gint16 out[] = { 0, G_MININT16, G_MAXINT16,
32, 33, 32,
33, 31,
-32, -33,
-31, -32,
-31, -33,
-32
};
@ -633,9 +635,9 @@ GST_START_TEST (test_float_conversion)
{
gint16 in[] = { 0, -32768, 16384, -16384 };
gdouble out[] = { 0.0,
4.6566128752457969e-10 * (gdouble) (-32768L << 16), /* ~ -1.0 */
4.6566128752457969e-10 * (gdouble) (16384L << 16), /* ~ 0.5 */
4.6566128752457969e-10 * (gdouble) (-16384L << 16), /* ~ -0.5 */
(gdouble) (-32768L << 16) / 2147483647.0, /* ~ -1.0 */
(gdouble) (16384L << 16) / 2147483647.0, /* ~ 0.5 */
(gdouble) (-16384L << 16) / 2147483647.0, /* ~ -0.5 */
};
RUN_CONVERSION ("16 signed to 64 float",
@ -645,9 +647,9 @@ GST_START_TEST (test_float_conversion)
{
gint32 in[] = { 0, (-1L << 31), (1L << 30), (-1L << 30) };
gdouble out[] = { 0.0,
4.6566128752457969e-10 * (gdouble) (-1L << 31), /* ~ -1.0 */
4.6566128752457969e-10 * (gdouble) (1L << 30), /* ~ 0.5 */
4.6566128752457969e-10 * (gdouble) (-1L << 30), /* ~ -0.5 */
(gdouble) (-1L << 31) / 2147483647.0, /* ~ -1.0 */
(gdouble) (1L << 30) / 2147483647.0, /* ~ 0.5 */
(gdouble) (-1L << 30) / 2147483647.0, /* ~ -0.5 */
};
RUN_CONVERSION ("32 signed to 64 float",