gstreamer/tests/check/elements/audioconvert.c
Sebastian Dröge 293a9c09b8 gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
Original commit message from CVS:
* gst/audioconvert/audioconvert.c:
Add docs to the integer pack functions and implement proper
rounding. Before we had rounding towards negative infinity, i.e.
always the smaller number was taken. Now we use natural rounding,
i.e. rounding to the nearest integer and to the one with the largest
absolute value for X.5. The old rounding introduced some minor
distortions. Fixes #420079
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Fix one unit test that assumed the old rounding and added unit tests
for checking signed/unsigned int16 <-> signed/unsigned int16 with
depth 8, one for signed int16 <-> unsigned int16 and one for the new
rounding from signed int32 to signed/unsigned int16.
2007-03-27 12:44:14 +00:00

796 lines
26 KiB
C

/* 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/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) BYTE_ORDER, " \
"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");
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 ("\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 */
{
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 */
{
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 */
{
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,
-32, -33,
-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 */
{
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[] = { 0.0, 1.0, -1.0, 0.5, -0.5, 1.1, -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 to 16 signed",
in, get_float_caps (1, "BYTE_ORDER", 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[] = { 0.0, 1.0, -1.0, 0.5, -0.5, 1.1, -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 to 16 signed",
in, get_float_caps (1, "BYTE_ORDER", 64),
out, get_int_caps (1, "BYTE_ORDER", 16, 16, TRUE));
}
{
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 */
};
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,
4.6566128752457969e-10 * (gdouble) (-1L << 31), /* ~ -1.0 */
4.6566128752457969e-10 * (gdouble) (1L << 30), /* ~ 0.5 */
4.6566128752457969e-10 * (gdouble) (-1L << 30), /* ~ -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));
}
}
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;
}