gstreamer/tests/check/elements/audioconvert.c

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/* GStreamer
*
* unit test for audioconvert
*
* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
*
* 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 <unistd.h>
#include <gst/floatcast/floatcast.h>
#include <gst/check/gstcheck.h>
#include <gst/audio/multichannel.h>
gboolean have_eos = FALSE;
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
GstPad *mysrcpad, *mysinkpad;
#define CONVERT_CAPS_TEMPLATE_STRING \
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) { 32, 64 };" \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 32, " \
"depth = (int) [ 1, 32 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 24, " \
"depth = (int) [ 1, 24 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 16, " \
"depth = (int) [ 1, 16 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 8, " \
"depth = (int) [ 1, 8 ], " \
"signed = (boolean) { true, false } "
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (CONVERT_CAPS_TEMPLATE_STRING)
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (CONVERT_CAPS_TEMPLATE_STRING)
);
/* takes over reference for outcaps */
GstElement *
setup_audioconvert (GstCaps * outcaps)
{
GstElement *audioconvert;
GST_DEBUG ("setup_audioconvert with caps %" GST_PTR_FORMAT, outcaps);
audioconvert = gst_check_setup_element ("audioconvert");
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now 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.
2007-06-28 20:37:58 +00:00
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
* later */
gst_pad_use_fixed_caps (mysinkpad);
gst_pad_set_caps (mysinkpad, outcaps);
gst_caps_unref (outcaps);
outcaps = gst_pad_get_negotiated_caps (mysinkpad);
fail_unless (gst_caps_is_fixed (outcaps));
gst_caps_unref (outcaps);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
return audioconvert;
}
void
cleanup_audioconvert (GstElement * audioconvert)
{
GST_DEBUG ("cleanup_audioconvert");
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (audioconvert);
gst_check_teardown_sink_pad (audioconvert);
gst_check_teardown_element (audioconvert);
}
/* returns a newly allocated caps */
static GstCaps *
get_int_caps (guint channels, gchar * endianness, guint width,
guint depth, gboolean signedness)
{
GstCaps *caps;
gchar *string;
string = g_strdup_printf ("audio/x-raw-int, "
"rate = (int) 44100, "
"channels = (int) %d, "
"endianness = (int) %s, "
"width = (int) %d, "
"depth = (int) %d, "
"signed = (boolean) %s ",
channels, endianness, width, depth, signedness ? "true" : "false");
GST_DEBUG ("creating caps from %s", string);
caps = gst_caps_from_string (string);
g_free (string);
fail_unless (caps != NULL);
GST_DEBUG ("returning caps %p", caps);
return caps;
}
/* returns a newly allocated caps */
static GstCaps *
get_float_caps (guint channels, gchar * endianness, guint width)
{
GstCaps *caps;
gchar *string;
string = g_strdup_printf ("audio/x-raw-float, "
"rate = (int) 44100, "
"channels = (int) %d, "
"endianness = (int) %s, "
"width = (int) %d ", channels, endianness, width);
GST_DEBUG ("creating caps from %s", string);
caps = gst_caps_from_string (string);
g_free (string);
fail_unless (caps != NULL);
GST_DEBUG ("returning caps %p", caps);
return caps;
}
/* Copied from vorbis; the particular values used don't matter */
static GstAudioChannelPosition channelpositions[][6] = {
{ /* Mono */
GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
{ /* Stereo */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
{ /* Stereo + Centre */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
{ /* Quadraphonic */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
},
{ /* Stereo + Centre + rear stereo */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
},
{ /* Full 5.1 Surround */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE,
}
};
/* these are a bunch of random positions, they are mostly just
* different from the ones above, don't use elsewhere */
static GstAudioChannelPosition mixed_up_positions[][6] = {
{
GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_LFE,
}
};
static void
set_channel_positions (GstCaps * caps, int channels,
GstAudioChannelPosition * channelpositions)
{
GValue chanpos = { 0 };
GValue pos = { 0 };
GstStructure *structure = gst_caps_get_structure (caps, 0);
int c;
g_value_init (&chanpos, GST_TYPE_ARRAY);
g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
for (c = 0; c < channels; c++) {
g_value_set_enum (&pos, channelpositions[c]);
gst_value_array_append_value (&chanpos, &pos);
}
g_value_unset (&pos);
gst_structure_set_value (structure, "channel-positions", &chanpos);
g_value_unset (&chanpos);
}
/* For channels > 2, caps have to have channel positions. This adds some simple
* ones. Only implemented for channels between 1 and 6.
*/
static GstCaps *
get_float_mc_caps (guint channels, gchar * endianness, guint width,
gboolean mixed_up_layout)
{
GstCaps *caps = get_float_caps (channels, endianness, width);
if (channels <= 6) {
if (mixed_up_layout)
set_channel_positions (caps, channels, mixed_up_positions[channels - 1]);
else
set_channel_positions (caps, channels, channelpositions[channels - 1]);
}
return caps;
}
static GstCaps *
get_int_mc_caps (guint channels, gchar * endianness, guint width,
guint depth, gboolean signedness, gboolean mixed_up_layout)
{
GstCaps *caps = get_int_caps (channels, endianness, width, depth, signedness);
if (channels <= 6) {
if (mixed_up_layout)
set_channel_positions (caps, channels, mixed_up_positions[channels - 1]);
else
set_channel_positions (caps, channels, channelpositions[channels - 1]);
}
return caps;
}
/* eats the refs to the caps */
static void
verify_convert (const gchar * which, void *in, int inlength,
GstCaps * incaps, void *out, int outlength, GstCaps * outcaps)
{
GstBuffer *inbuffer, *outbuffer;
GstElement *audioconvert;
GST_DEBUG ("verifying conversion %s", which);
GST_DEBUG ("incaps: %" GST_PTR_FORMAT, incaps);
GST_DEBUG ("outcaps: %" GST_PTR_FORMAT, outcaps);
ASSERT_CAPS_REFCOUNT (incaps, "incaps", 1);
ASSERT_CAPS_REFCOUNT (outcaps, "outcaps", 1);
audioconvert = setup_audioconvert (outcaps);
ASSERT_CAPS_REFCOUNT (outcaps, "outcaps", 1);
fail_unless (gst_element_set_state (audioconvert,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
GST_DEBUG ("Creating buffer of %d bytes", inlength);
inbuffer = gst_buffer_new_and_alloc (inlength);
memcpy (GST_BUFFER_DATA (inbuffer), in, inlength);
gst_buffer_set_caps (inbuffer, incaps);
ASSERT_CAPS_REFCOUNT (incaps, "incaps", 2);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
GST_DEBUG ("push it");
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
GST_DEBUG ("pushed it");
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) == 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), outlength);
if (memcmp (GST_BUFFER_DATA (outbuffer), out, outlength) != 0) {
g_print ("\nInput data:\n");
gst_util_dump_mem (in, inlength);
g_print ("\nConverted data:\n");
gst_util_dump_mem (GST_BUFFER_DATA (outbuffer), outlength);
g_print ("\nExpected data:\n");
gst_util_dump_mem (out, outlength);
}
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, outlength) == 0,
"failed converting %s", which);
buffers = g_list_remove (buffers, outbuffer);
gst_buffer_unref (outbuffer);
fail_unless (gst_element_set_state (audioconvert,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to null");
/* cleanup */
GST_DEBUG ("cleanup audioconvert");
cleanup_audioconvert (audioconvert);
GST_DEBUG ("cleanup, unref incaps");
ASSERT_CAPS_REFCOUNT (incaps, "incaps", 1);
gst_caps_unref (incaps);
}
#define RUN_CONVERSION(which, inarray, in_get_caps, outarray, out_get_caps) \
verify_convert (which, inarray, sizeof (inarray), \
in_get_caps, outarray, sizeof (outarray), out_get_caps)
GST_START_TEST (test_int16)
{
/* stereo to mono */
{
gint16 in[] = { 16384, -256, 1024, 1024 };
gint16 out[] = { 8064, 1024 };
RUN_CONVERSION ("int16 stereo to mono",
in, get_int_caps (2, "BYTE_ORDER", 16, 16, TRUE),
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE));
}
/* mono to stereo */
{
gint16 in[] = { 512, 1024 };
gint16 out[] = { 512, 512, 1024, 1024 };
RUN_CONVERSION ("int16 mono to stereo",
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
out, get_int_caps (2, "BYTE_ORDER", 16, 16, TRUE));
}
/* signed -> unsigned */
{
gint16 in[] = { 0, -32767, 32767, -32768 };
guint16 out[] = { 32768, 1, 65535, 0 };
RUN_CONVERSION ("int16 signed to unsigned",
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
out, get_int_caps (1, "BYTE_ORDER", 16, 16, FALSE));
RUN_CONVERSION ("int16 unsigned to signed",
out, get_int_caps (1, "BYTE_ORDER", 16, 16, FALSE),
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE));
}
}
GST_END_TEST;
GST_START_TEST (test_float32)
{
/* stereo to mono */
{
gfloat in[] = { 0.6, -0.0078125, 0.03125, 0.03125 };
gfloat out[] = { 0.29609375, 0.03125 };
RUN_CONVERSION ("float32 stereo to mono",
in, get_float_caps (2, "BYTE_ORDER", 32),
out, get_float_caps (1, "BYTE_ORDER", 32));
}
/* mono to stereo */
{
gfloat in[] = { 0.015625, 0.03125 };
gfloat out[] = { 0.015625, 0.015625, 0.03125, 0.03125 };
RUN_CONVERSION ("float32 mono to stereo",
in, get_float_caps (1, "BYTE_ORDER", 32),
out, get_float_caps (2, "BYTE_ORDER", 32));
}
}
GST_END_TEST;
GST_START_TEST (test_int_conversion)
{
/* 8 <-> 16 signed */
/* NOTE: if audioconvert was doing dithering we'd have a problem */
{
gint8 in[] = { 0, 1, 2, 127, -127 };
gint16 out[] = { 0, 256, 512, 32512, -32512 };
RUN_CONVERSION ("int 8bit to 16bit signed",
in, get_int_caps (1, "BYTE_ORDER", 8, 8, TRUE),
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE)
);
RUN_CONVERSION ("int 16bit signed to 8bit",
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
in, get_int_caps (1, "BYTE_ORDER", 8, 8, TRUE)
);
}
/* 16 -> 8 signed */
{
gint16 in[] = { 0, 127, 128, 256, 256 + 127, 256 + 128 };
gint8 out[] = { 0, 0, 1, 1, 1, 2 };
RUN_CONVERSION ("16 bit to 8 signed",
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
out, get_int_caps (1, "BYTE_ORDER", 8, 8, TRUE)
);
}
/* 8 unsigned <-> 16 signed */
/* NOTE: if audioconvert was doing dithering we'd have a problem */
{
guint8 in[] = { 128, 129, 130, 255, 1 };
gint16 out[] = { 0, 256, 512, 32512, -32512 };
GstCaps *incaps, *outcaps;
/* exploded for easier valgrinding */
incaps = get_int_caps (1, "BYTE_ORDER", 8, 8, FALSE);
outcaps = get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE);
GST_DEBUG ("incaps: %" GST_PTR_FORMAT, incaps);
GST_DEBUG ("outcaps: %" GST_PTR_FORMAT, outcaps);
RUN_CONVERSION ("8 unsigned to 16 signed", in, incaps, out, outcaps);
RUN_CONVERSION ("16 signed to 8 unsigned", out, get_int_caps (1,
"BYTE_ORDER", 16, 16, TRUE), in, get_int_caps (1, "BYTE_ORDER", 8,
8, FALSE)
);
}
/* 8 <-> 24 signed */
/* NOTE: if audioconvert was doing dithering we'd have a problem */
{
gint8 in[] = { 0, 1, 127 };
guint8 out[] = { 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x7f };
/* out has the bytes in little-endian, so that's how they should be
* interpreted during conversion */
RUN_CONVERSION ("8 to 24 signed", in, get_int_caps (1, "BYTE_ORDER", 8, 8,
TRUE), out, get_int_caps (1, "LITTLE_ENDIAN", 24, 24, TRUE)
);
RUN_CONVERSION ("24 signed to 8", out, get_int_caps (1, "LITTLE_ENDIAN", 24,
24, TRUE), in, get_int_caps (1, "BYTE_ORDER", 8, 8, TRUE)
);
}
/* 16 bit signed <-> unsigned */
{
gint16 in[] = { 0, 128, -128 };
guint16 out[] = { 32768, 32896, 32640 };
RUN_CONVERSION ("16 signed to 16 unsigned",
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
out, get_int_caps (1, "BYTE_ORDER", 16, 16, FALSE)
);
RUN_CONVERSION ("16 unsigned to 16 signed",
out, get_int_caps (1, "BYTE_ORDER", 16, 16, FALSE),
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE)
);
}
/* 16 bit signed <-> 8 in 16 bit signed */
/* NOTE: if audioconvert was doing dithering we'd have a problem */
{
gint16 in[] = { 0, 64 << 8, -64 << 8 };
gint16 out[] = { 0, 64, -64 };
RUN_CONVERSION ("16 signed to 8 in 16 signed",
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
out, get_int_caps (1, "BYTE_ORDER", 16, 8, TRUE)
);
RUN_CONVERSION ("8 in 16 signed to 16 signed",
out, get_int_caps (1, "BYTE_ORDER", 16, 8, TRUE),
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE)
);
}
/* 16 bit unsigned <-> 8 in 16 bit unsigned */
/* NOTE: if audioconvert was doing dithering we'd have a problem */
{
guint16 in[] = { 1 << 15, (1 << 15) - (64 << 8), (1 << 15) + (64 << 8) };
guint16 out[] = { 1 << 7, (1 << 7) - 64, (1 << 7) + 64 };
RUN_CONVERSION ("16 unsigned to 8 in 16 unsigned",
in, get_int_caps (1, "BYTE_ORDER", 16, 16, FALSE),
out, get_int_caps (1, "BYTE_ORDER", 16, 8, FALSE)
);
RUN_CONVERSION ("8 in 16 unsigned to 16 unsigned",
out, get_int_caps (1, "BYTE_ORDER", 16, 8, FALSE),
in, get_int_caps (1, "BYTE_ORDER", 16, 16, FALSE)
);
}
/* 32 bit signed -> 16 bit signed for rounding check */
/* NOTE: if audioconvert was doing dithering we'd have a problem */
{
gint32 in[] = { 0, G_MININT32, G_MAXINT32,
(32 << 16), (32 << 16) + (1 << 15), (32 << 16) - (1 << 15),
(32 << 16) + (2 << 15), (32 << 16) - (2 << 15),
(-32 << 16) + (1 << 15), (-32 << 16) - (1 << 15),
(-32 << 16) + (2 << 15), (-32 << 16) - (2 << 15),
(-32 << 16)
};
gint16 out[] = { 0, G_MININT16, G_MAXINT16,
32, 33, 32,
33, 31,
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now 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.
2007-06-28 20:37:58 +00:00
-31, -32,
-31, -33,
-32
};
RUN_CONVERSION ("32 signed to 16 signed for rounding",
in, get_int_caps (1, "BYTE_ORDER", 32, 32, TRUE),
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE)
);
}
/* 32 bit signed -> 16 bit unsigned for rounding check */
/* NOTE: if audioconvert was doing dithering we'd have a problem */
{
gint32 in[] = { 0, G_MININT32, G_MAXINT32,
(32 << 16), (32 << 16) + (1 << 15), (32 << 16) - (1 << 15),
(32 << 16) + (2 << 15), (32 << 16) - (2 << 15),
(-32 << 16) + (1 << 15), (-32 << 16) - (1 << 15),
(-32 << 16) + (2 << 15), (-32 << 16) - (2 << 15),
(-32 << 16)
};
guint16 out[] = { (1 << 15), 0, G_MAXUINT16,
(1 << 15) + 32, (1 << 15) + 33, (1 << 15) + 32,
(1 << 15) + 33, (1 << 15) + 31,
(1 << 15) - 31, (1 << 15) - 32,
(1 << 15) - 31, (1 << 15) - 33,
(1 << 15) - 32
};
RUN_CONVERSION ("32 signed to 16 unsigned for rounding",
in, get_int_caps (1, "BYTE_ORDER", 32, 32, TRUE),
out, get_int_caps (1, "BYTE_ORDER", 16, 16, FALSE)
);
}
}
GST_END_TEST;
GST_START_TEST (test_float_conversion)
{
/* 32 float <-> 16 signed */
/* NOTE: if audioconvert was doing dithering we'd have a problem */
{
gfloat in_le[] =
{ GFLOAT_TO_LE (0.0), GFLOAT_TO_LE (1.0), GFLOAT_TO_LE (-1.0),
GFLOAT_TO_LE (0.5), GFLOAT_TO_LE (-0.5), GFLOAT_TO_LE (1.1),
GFLOAT_TO_LE (-1.1)
};
gfloat in_be[] =
{ GFLOAT_TO_BE (0.0), GFLOAT_TO_BE (1.0), GFLOAT_TO_BE (-1.0),
GFLOAT_TO_BE (0.5), GFLOAT_TO_BE (-0.5), GFLOAT_TO_BE (1.1),
GFLOAT_TO_BE (-1.1)
};
gint16 out[] = { 0, 32767, -32768, 16384, -16384, 32767, -32768 };
/* only one direction conversion, the other direction does
* not produce exactly the same as the input due to floating
* point rounding errors etc. */
RUN_CONVERSION ("32 float le to 16 signed",
in_le, get_float_caps (1, "LITTLE_ENDIAN", 32),
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE));
RUN_CONVERSION ("32 float be to 16 signed",
in_be, get_float_caps (1, "BIG_ENDIAN", 32),
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE));
}
{
gint16 in[] = { 0, -32768, 16384, -16384 };
gfloat out[] = { 0.0, -1.0, 0.5, -0.5 };
RUN_CONVERSION ("16 signed to 32 float",
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
out, get_float_caps (1, "BYTE_ORDER", 32));
}
/* 64 float <-> 16 signed */
/* NOTE: if audioconvert was doing dithering we'd have a problem */
{
gdouble in_le[] =
{ GDOUBLE_TO_LE (0.0), GDOUBLE_TO_LE (1.0), GDOUBLE_TO_LE (-1.0),
GDOUBLE_TO_LE (0.5), GDOUBLE_TO_LE (-0.5), GDOUBLE_TO_LE (1.1),
GDOUBLE_TO_LE (-1.1)
};
gdouble in_be[] =
{ GDOUBLE_TO_BE (0.0), GDOUBLE_TO_BE (1.0), GDOUBLE_TO_BE (-1.0),
GDOUBLE_TO_BE (0.5), GDOUBLE_TO_BE (-0.5), GDOUBLE_TO_BE (1.1),
GDOUBLE_TO_BE (-1.1)
};
gint16 out[] = { 0, 32767, -32768, 16384, -16384, 32767, -32768 };
/* only one direction conversion, the other direction does
* not produce exactly the same as the input due to floating
* point rounding errors etc. */
RUN_CONVERSION ("64 float LE to 16 signed",
in_le, get_float_caps (1, "LITTLE_ENDIAN", 64),
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE));
RUN_CONVERSION ("64 float BE to 16 signed",
in_be, get_float_caps (1, "BIG_ENDIAN", 64),
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE));
}
{
gint16 in[] = { 0, -32768, 16384, -16384 };
gdouble out[] = { 0.0,
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now 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.
2007-06-28 20:37:58 +00:00
(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",
in, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE),
out, get_float_caps (1, "BYTE_ORDER", 64));
}
{
gint32 in[] = { 0, (-1L << 31), (1L << 30), (-1L << 30) };
gdouble out[] = { 0.0,
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now 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.
2007-06-28 20:37:58 +00:00
(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",
in, get_int_caps (1, "BYTE_ORDER", 32, 32, TRUE),
out, get_float_caps (1, "BYTE_ORDER", 64));
}
/* 64-bit float <-> 32-bit float */
{
gdouble in[] = { 0.0, 1.0, -1.0, 0.5, -0.5 };
gfloat out[] = { 0.0, 1.0, -1.0, 0.5, -0.5 };
RUN_CONVERSION ("64 float to 32 float",
in, get_float_caps (1, "BYTE_ORDER", 64),
out, get_float_caps (1, "BYTE_ORDER", 32));
RUN_CONVERSION ("32 float to 64 float",
out, get_float_caps (1, "BYTE_ORDER", 32),
in, get_float_caps (1, "BYTE_ORDER", 64));
}
/* 32-bit float little endian <-> big endian */
{
gfloat le[] = { GFLOAT_TO_LE (0.0), GFLOAT_TO_LE (1.0), GFLOAT_TO_LE (-1.0),
GFLOAT_TO_LE (0.5), GFLOAT_TO_LE (-0.5)
};
gfloat be[] = { GFLOAT_TO_BE (0.0), GFLOAT_TO_BE (1.0), GFLOAT_TO_BE (-1.0),
GFLOAT_TO_BE (0.5), GFLOAT_TO_BE (-0.5)
};
RUN_CONVERSION ("32 float LE to BE",
le, get_float_caps (1, "LITTLE_ENDIAN", 32),
be, get_float_caps (1, "BIG_ENDIAN", 32));
RUN_CONVERSION ("32 float BE to LE",
be, get_float_caps (1, "BIG_ENDIAN", 32),
le, get_float_caps (1, "LITTLE_ENDIAN", 32));
}
/* 64-bit float little endian <-> big endian */
{
gdouble le[] =
{ GDOUBLE_TO_LE (0.0), GDOUBLE_TO_LE (1.0), GDOUBLE_TO_LE (-1.0),
GDOUBLE_TO_LE (0.5), GDOUBLE_TO_LE (-0.5)
};
gdouble be[] =
{ GDOUBLE_TO_BE (0.0), GDOUBLE_TO_BE (1.0), GDOUBLE_TO_BE (-1.0),
GDOUBLE_TO_BE (0.5), GDOUBLE_TO_BE (-0.5)
};
RUN_CONVERSION ("64 float LE to BE",
le, get_float_caps (1, "LITTLE_ENDIAN", 64),
be, get_float_caps (1, "BIG_ENDIAN", 64));
RUN_CONVERSION ("64 float BE to LE",
be, get_float_caps (1, "BIG_ENDIAN", 64),
le, get_float_caps (1, "LITTLE_ENDIAN", 64));
}
}
GST_END_TEST;
GST_START_TEST (test_multichannel_conversion)
{
{
/* Ensure that audioconvert prefers to convert to integer, rather than mix
* to mono
*/
gfloat in[] = { 0.0, 0.0, 0.0, 0.0, 0.0, 0.0 };
gfloat out[] = { 0.0, 0.0 };
/* only one direction conversion, the other direction does
* not produce exactly the same as the input due to floating
* point rounding errors etc. */
RUN_CONVERSION ("3 channels to 1", in, get_float_mc_caps (3,
"BYTE_ORDER", 32, FALSE), out, get_float_caps (1, "BYTE_ORDER",
32));
}
}
GST_END_TEST;
GST_START_TEST (test_channel_remapping)
{
/* float */
{
gfloat in[] = { 0.0, 1.0, -0.5 };
gfloat out[] = { -0.5, 1.0, 0.0 };
GstCaps *in_caps = get_float_mc_caps (3, "BYTE_ORDER", 32, FALSE);
GstCaps *out_caps = get_float_mc_caps (3, "BYTE_ORDER", 32, TRUE);
RUN_CONVERSION ("3 channels layout remapping float", in, in_caps,
out, out_caps);
}
/* int */
{
guint16 in[] = { 0, 65535, 0x9999 };
guint16 out[] = { 0x9999, 65535, 0 };
GstCaps *in_caps = get_int_mc_caps (3, "BYTE_ORDER", 16, 16, FALSE, FALSE);
GstCaps *out_caps = get_int_mc_caps (3, "BYTE_ORDER", 16, 16, FALSE, TRUE);
RUN_CONVERSION ("3 channels layout remapping int", in, in_caps,
out, out_caps);
}
/* TODO: float => int conversion with remapping and vice versa,
* int => int conversion with remapping */
}
GST_END_TEST;
GST_START_TEST (test_caps_negotiation)
{
GstElement *src, *ac1, *ac2, *ac3, *sink;
GstElement *pipeline;
GstPad *ac3_src;
GstCaps *caps1, *caps2;
pipeline = gst_pipeline_new ("test");
/* create elements */
src = gst_element_factory_make ("audiotestsrc", "src");
ac1 = gst_element_factory_make ("audioconvert", "ac1");
ac2 = gst_element_factory_make ("audioconvert", "ac2");
ac3 = gst_element_factory_make ("audioconvert", "ac3");
sink = gst_element_factory_make ("fakesink", "sink");
ac3_src = gst_element_get_pad (ac3, "src");
/* test with 2 audioconvert elements */
gst_bin_add_many (GST_BIN (pipeline), src, ac1, ac3, sink, NULL);
gst_element_link_many (src, ac1, ac3, sink, NULL);
/* Set to PAUSED and wait for PREROLL */
fail_if (gst_element_set_state (pipeline, GST_STATE_PAUSED) ==
GST_STATE_CHANGE_FAILURE, "Failed to set test pipeline to PAUSED");
fail_if (gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE) !=
GST_STATE_CHANGE_SUCCESS, "Failed to set test pipeline to PAUSED");
caps1 = gst_pad_get_caps (ac3_src);
fail_if (caps1 == NULL, "gst_pad_get_caps returned NULL");
GST_DEBUG ("Caps size 1 : %d", gst_caps_get_size (caps1));
fail_if (gst_element_set_state (pipeline, GST_STATE_READY) ==
GST_STATE_CHANGE_FAILURE, "Failed to set test pipeline back to READY");
fail_if (gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE) !=
GST_STATE_CHANGE_SUCCESS, "Failed to set test pipeline back to READY");
/* test with 3 audioconvert elements */
gst_element_unlink (ac1, ac3);
gst_bin_add (GST_BIN (pipeline), ac2);
gst_element_link_many (ac1, ac2, ac3, NULL);
fail_if (gst_element_set_state (pipeline, GST_STATE_PAUSED) ==
GST_STATE_CHANGE_FAILURE, "Failed to set test pipeline back to PAUSED");
fail_if (gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE) !=
GST_STATE_CHANGE_SUCCESS, "Failed to set test pipeline back to PAUSED");
caps2 = gst_pad_get_caps (ac3_src);
fail_if (caps2 == NULL, "gst_pad_get_caps returned NULL");
GST_DEBUG ("Caps size 2 : %d", gst_caps_get_size (caps2));
fail_unless (gst_caps_get_size (caps1) == gst_caps_get_size (caps2));
gst_caps_unref (caps1);
gst_caps_unref (caps2);
fail_if (gst_element_set_state (pipeline, GST_STATE_NULL) ==
GST_STATE_CHANGE_FAILURE, "Failed to set test pipeline back to NULL");
fail_if (gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE) !=
GST_STATE_CHANGE_SUCCESS, "Failed to set test pipeline back to NULL");
gst_object_unref (ac3_src);
gst_object_unref (pipeline);
}
GST_END_TEST;
Suite *
audioconvert_suite (void)
{
Suite *s = suite_create ("audioconvert");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_int16);
tcase_add_test (tc_chain, test_float32);
tcase_add_test (tc_chain, test_int_conversion);
tcase_add_test (tc_chain, test_float_conversion);
tcase_add_test (tc_chain, test_multichannel_conversion);
tcase_add_test (tc_chain, test_channel_remapping);
tcase_add_test (tc_chain, test_caps_negotiation);
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = audioconvert_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_NORMAL);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}