--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_base/tsrc/check/elements/audioconvert/src/audioconvert.c Thu Dec 17 08:53:32 2009 +0200
@@ -0,0 +1,951 @@
+/* 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 <gst/gst_global.h>
+#include <unistd.h>
+
+//#include <glib_global.h>
+
+#define LOG_FILE "c:\\logs\\audioconvert_logs.txt"
+#include "std_log_result.h"
+#define LOG_FILENAME_LINE __FILE__, __LINE__
+
+#include <gst/floatcast/floatcast.h>
+#include <gst/check/gstcheck.h>
+
+#include <gst/audio/multichannel.h>
+
+
+#if EMULATOR
+GET_GLOBAL_VAR_FROM_TLS(buffers,gstcheck,GList*)
+#define buffers (*GET_GSTREAMER_WSD_VAR_NAME(buffers,gstcheck,g)())
+#else
+extern GList *buffers;
+#endif
+
+#if EMULATOR
+static GET_GLOBAL_VAR_FROM_TLS(expecting_log,gstcheck,gboolean)
+#define _gst_check_expecting_log (*GET_GSTREAMER_WSD_VAR_NAME(expecting_log,gstcheck,g)())
+#else
+gboolean _gst_check_expecting_log = FALSE;
+#endif
+
+#include <libgstreamer_wsd_solution.h>
+
+void create_xml(int result)
+{
+ if(result)
+ assert_failed = 1;
+
+ testResultXml(xmlfile);
+ close_log_file();
+}
+
+
+/* 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 */
+static 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 */
+static GstElement *
+setup_audioconvert (GstCaps * outcaps)
+{
+ GstElement *audioconvert;
+
+ 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
+ * 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;
+}
+
+static 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);
+
+ /* make sure that the channel positions are not lost */
+ {
+ GstStructure *in_s, *out_s;
+ gint out_chans;
+
+ in_s = gst_caps_get_structure (incaps, 0);
+ out_s = gst_caps_get_structure (GST_BUFFER_CAPS (outbuffer), 0);
+ fail_unless (gst_structure_get_int (out_s, "channels", &out_chans));
+
+ /* positions for 1 and 2 channels are implicit if not provided */
+ if (out_chans > 2 && gst_structure_has_field (in_s, "channel-positions")) {
+ if (!gst_structure_has_field (out_s, "channel-positions")) {
+ g_error ("Channel layout got lost somewhere:\n\nIns : %s\nOuts: %s\n",
+ gst_structure_to_string (in_s), gst_structure_to_string (out_s));
+ }
+ }
+ }
+
+ 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)
+
+
+void test_int16()
+{
+ xmlfile = "test_int16";
+ std_log(LOG_FILENAME_LINE, "Test Started 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));
+ }
+ std_log(LOG_FILENAME_LINE, "Test Successful");
+ create_xml(0);
+}
+
+
+
+
+void test_float32()
+{
+ xmlfile = "test_float32";
+ std_log(LOG_FILENAME_LINE, "Test Started 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));
+ }
+ std_log(LOG_FILENAME_LINE, "Test Successful");
+ create_xml(0);
+}
+
+
+
+
+void test_int_conversion()
+{
+ xmlfile = "test_int_conversion";
+ std_log(LOG_FILENAME_LINE, "Test Started 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,
+ -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)
+ );
+ }
+ std_log(LOG_FILENAME_LINE, "Test Successful");
+ create_xml(0);
+}
+
+void create_array_float(gfloat in_be[], float temp[], int len)
+ {
+ int i;
+ for(i=0; i<len; i++)
+ in_be[i] = GFLOAT_TO_BE (temp[i]);
+ }
+void create_array_double(gdouble in_be[], float temp[], int len)
+ {
+ int i;
+ for(i=0; i<len; i++)
+ in_be[i] = GDOUBLE_TO_BE (temp[i]);
+ }
+
+
+void test_float_conversion()
+{
+ std_log(LOG_FILENAME_LINE, "Test Started 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[7];
+ float temp[] = {0.0, 1.0, -1.0, 0.5, -0.5, 1.1, -1.1};
+
+ gint16 out[] = { 0, 32767, -32768, 16384, -16384, 32767, -32768 };
+
+ create_array_float(in_be, temp, 7);
+
+ /* 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[7];
+ float temp[] = {0.0, 1.0, -1.0, 0.5, -0.5, 1.1, -1.1};
+
+ gint16 out[] = { 0, 32767, -32768, 16384, -16384, 32767, -32768 };
+ create_array_double(in_be, temp, 7);
+
+ /* 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,
+ (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,
+ (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[5];
+ float temp[] = {0.0, 1.0, -1.0, 0.5, -0.5};
+
+ create_array_float(be, temp, 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[5];
+ float temp[] = {0.0, 1.0, -1.0, 0.5, -0.5};
+
+ create_array_double(be, temp, 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));
+ }
+
+ std_log(LOG_FILENAME_LINE, "Test Successful");
+ create_xml(0);
+}
+
+
+void test_multichannel_conversion()
+{
+ xmlfile = "test_multichannel_conversion";
+ std_log(LOG_FILENAME_LINE, "Test Started 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));
+ }
+ std_log(LOG_FILENAME_LINE, "Test Successful");
+ create_xml(0);
+}
+
+
+
+
+void test_channel_remapping()
+{
+ xmlfile = "test_channel_remapping";
+ std_log(LOG_FILENAME_LINE, "Test Started 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 */
+ std_log(LOG_FILENAME_LINE, "Test Successful");
+ create_xml(0);
+}
+
+
+
+void test_caps_negotiation()
+{
+ GstElement *src, *ac1, *ac2, *ac3, *sink;
+ GstElement *pipeline;
+ GstPad *ac3_src;
+ GstCaps *caps1, *caps2;
+xmlfile = "test_caps_negotiation";
+ std_log(LOG_FILENAME_LINE, "Test Started test_caps_negotiation");
+ 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);
+ std_log(LOG_FILENAME_LINE, "Test Successful");
+ create_xml(0);
+}
+
+
+
+
+void (*fn[]) (void) = {
+test_int16,
+test_float32,
+test_int_conversion,
+test_float_conversion,
+test_multichannel_conversion,
+test_channel_remapping,
+test_caps_negotiation
+};
+
+char *args[] = {
+"test_int16",
+"test_float32",
+"test_int_conversion",
+"test_float_conversion",
+"test_multichannel_conversion",
+"test_channel_remapping",
+"test_caps_negotiation"
+};
+
+GST_CHECK_MAIN(audioconvert)
+
+