diff -r 000000000000 -r 0e761a78d257 gst_plugins_base/tests/check/elements/audioconvert.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/gst_plugins_base/tests/check/elements/audioconvert.c Thu Dec 17 08:53:32 2009 +0200 @@ -0,0 +1,884 @@ +/* GStreamer + * + * unit test for audioconvert + * + * Copyright (C) <2005> Thomas Vander Stichele + * + * 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 + +#include +#include +#include + +/* 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) + + +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, + -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, + (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[] = { 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; + + +static 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; +}