/* 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 

#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 */

  {