+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):
gstaudioconvert.c \
audioconvert.c \
gstchannelmix.c \
+ gstaudioquantize.c \
plugin.c
libgstaudioconvert_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
gstaudioconvert.h \
audioconvert.h \
gstchannelmix.h \
+ gstaudioquantize.h \
plugin.h
#TESTS = channelmixtest
#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)
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) \
\
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); \
} \
}
*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)
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. */
* 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 */
*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))
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)
(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[] = {
(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;
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 {
gboolean
audio_convert_prepare_context (AudioConvertCtx * ctx, AudioConvertFmt * in,
- AudioConvertFmt * out)
+ AudioConvertFmt * out, DitherType dither, NoiseShapingType ns)
{
gint idx_in, idx_out;
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);
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;
}
{
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);
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);
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);
#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;
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
{
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__ */
#include "gstaudioconvert.h"
#include "gstchannelmix.h"
+#include "gstaudioquantize.h"
#include "plugin.h"
GST_DEBUG_CATEGORY (audio_convert_debug);
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
enum
{
ARG_0,
- ARG_AGGRESSIVE
+ ARG_DITHERING,
+ ARG_NOISE_SHAPING,
};
#define DEBUG_INIT(bla) \
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
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 =
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));
}
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;
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;
+ }
+}
GstBaseTransform element;
AudioConvertCtx ctx;
+
+ DitherType dither;
+ NoiseShapingType ns;
};
struct _GstAudioConvertClass
--- /dev/null
+/* 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);
+}
--- /dev/null
+/* 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__ */
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
gint16 out[] = { 0, G_MININT16, G_MAXINT16,
32, 33, 32,
33, 31,
- -32, -33,
+ -31, -32,
-31, -33,
-32
};
{
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",
{
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",
projects / platform / upstream / gstreamer.git / commitdiff