/* GStreamer

 *

 *

 * Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>

 * Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>

 *

 * 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 <gst/check/gstcheck.h>

#define LOG_FILE "c:\\logs\\audioresample_logs.txt" 

#include "std_log_result.h" 

#define LOG_FILENAME_LINE __FILE__, __LINE__

void create_xml(int result)

{

    if(result)

        assert_failed = 1;

    testResultXml(xmlfile);

    close_log_file();

}

#include "libgstreamer_wsd_solution.h" 

#if EMULATOR

static GET_GLOBAL_VAR_FROM_TLS(threads_running,gstcheck,gboolean)

#define _gst_check_threads_running (*GET_GSTREAMER_WSD_VAR_NAME(threads_running,gstcheck,g)())

#else 

#endif

#if EMULATOR

static GET_GLOBAL_VAR_FROM_TLS(raised_critical,gstcheck,gboolean)

#define _gst_check_raised_critical (*GET_GSTREAMER_WSD_VAR_NAME(raised_critical,gstcheck,g)())

#else 

#endif

//gboolean _gst_check_raised_warning = FALSE;

#if EMULATOR

static GET_GLOBAL_VAR_FROM_TLS(raised_warning,gstcheck,gboolean)

#define _gst_check_raised_warning (*GET_GSTREAMER_WSD_VAR_NAME(raised_warning,gstcheck,g)())

#else 

#endif

//gboolean _gst_check_expecting_log = FALSE;

#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 

#endif

#if EMULATOR

GET_GLOBAL_VAR_FROM_TLS(buffers,gstcheck,GList*)

#define buffers (*GET_GSTREAMER_WSD_VAR_NAME(buffers,gstcheck,g)())

#else 

#endif

/* 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;

    "audio/x-raw-int, "                 \

    "channels = (int) [ 1, MAX ], "     \

    "rate = (int) [ 1,  MAX ], "        \

    "endianness = (int) BYTE_ORDER, "   \

    "width = (int) 16, "                \

    "depth = (int) 16, "                \

    "signed = (bool) TRUE"

static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)

    );

static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",

    GST_PAD_SRC,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)

    );

static GstElement *

{

  GstElement *audioresample;

  GstCaps *caps;

  GstStructure *structure;

  GST_DEBUG ("setup_audioresample");

  audioresample = gst_check_setup_element ("audioresample");

  structure = gst_caps_get_structure (caps, 0);

  gst_structure_set (structure, "channels", G_TYPE_INT, channels,

  fail_unless (gst_caps_is_fixed (caps));

  fail_unless (gst_element_set_state (audioresample,

          GST_STATE_PAUSED) == GST_STATE_CHANGE_SUCCESS,

      "could not set to paused");

  mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate, caps);

  gst_caps_unref (caps);

  structure = gst_caps_get_structure (caps, 0);

  gst_structure_set (structure, "channels", G_TYPE_INT, channels,

  fail_unless (gst_caps_is_fixed (caps));

  mysinkpad = gst_check_setup_sink_pad (audioresample, &sinktemplate, caps);

  /* this installs a getcaps func that will always return the caps we set

   * later */

  gst_pad_use_fixed_caps (mysinkpad);

  gst_caps_unref (caps);

  return audioresample;

}

static void

cleanup_audioresample (GstElement * audioresample)

{

  GST_DEBUG ("cleanup_audioresample");

  fail_unless (gst_element_set_state (audioresample,

          GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");

  gst_pad_set_active (mysrcpad, FALSE);

  gst_pad_set_active (mysinkpad, FALSE);

  gst_check_teardown_src_pad (audioresample);

  gst_check_teardown_sink_pad (audioresample);

  gst_check_teardown_element (audioresample);

}

static void

fail_unless_perfect_stream (void)

{

  guint64 timestamp = 0L, duration = 0L;

  guint64 offset = 0L, offset_end = 0L;

  GList *l;

  GstBuffer *buffer;

  for (l = buffers; l; l = l->next) {

    buffer = GST_BUFFER (l->data);

    ASSERT_BUFFER_REFCOUNT (buffer, "buffer", 1);

    GST_DEBUG ("buffer timestamp %" G_GUINT64_FORMAT ", duration %"

    fail_unless_equals_uint64 (timestamp, GST_BUFFER_TIMESTAMP (buffer));

    fail_unless_equals_uint64 (offset, GST_BUFFER_OFFSET (buffer));

    duration = GST_BUFFER_DURATION (buffer);

    offset_end = GST_BUFFER_OFFSET_END (buffer);

    timestamp += duration;

    offset = offset_end;

    gst_buffer_unref (buffer);

  }

  g_list_free (buffers);

  buffers = NULL;

}

/* this tests that the output is a perfect stream if the input is */

static void

test_perfect_stream_instance (int inrate, int outrate, int samples,

    int numbuffers)

{

  GstElement *audioresample;

  GstBuffer *inbuffer, *outbuffer;

  GstCaps *caps;

  int i, j;

  gint16 *p;

  caps = gst_pad_get_negotiated_caps (mysrcpad);

  fail_unless (gst_caps_is_fixed (caps));

  fail_unless (gst_element_set_state (audioresample,

          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,

      "could not set to playing");

  for (j = 1; j <= numbuffers; ++j) {

    inbuffer = gst_buffer_new_and_alloc (samples * 4);

    GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);

    gst_buffer_set_caps (inbuffer, caps);

    p = (gint16 *) GST_BUFFER_DATA (inbuffer);

    /* create a 16 bit signed ramp */

    for (i = 0; i < samples; ++i) {

      *p = -32767 + i * (65535 / samples);

      ++p;

      *p = -32767 + i * (65535 / samples);

      ++p;

    }

    /* pushing gives away my reference ... */

    fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);

    /* ... but it ends up being collected on the global buffer list */

    fail_unless_equals_int (g_list_length (buffers), j);

  }

  /* FIXME: we should make audioresample handle eos by flushing out the last

   * samples, which will give us one more, small, buffer */

  fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);

  ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);

  fail_unless_perfect_stream ();

  /* cleanup */

  gst_caps_unref (caps);

  cleanup_audioresample (audioresample);

}

/* make sure that outgoing buffers are contiguous in timestamp/duration and

 * offset/offsetend

 */

void test_perfect_stream()

{

  std_log(LOG_FILENAME_LINE, "Test Started test_perfect_stream");

  test_perfect_stream_instance (48000, 24000, 500, 20);

  test_perfect_stream_instance (48000, 12000, 500, 20);

  test_perfect_stream_instance (12000, 24000, 500, 20);

  test_perfect_stream_instance (12000, 48000, 500, 20);

  /* non-integral scalings */

  test_perfect_stream_instance (44100, 8000, 500, 20);

  test_perfect_stream_instance (8000, 44100, 500, 20);

  /* wacky scalings */

  test_perfect_stream_instance (12345, 54321, 500, 20);

  test_perfect_stream_instance (101, 99, 500, 20);

  std_log(LOG_FILENAME_LINE, "Test Successful");

  create_xml(0);

}

/* this tests that the output is a correct discontinuous stream

 * if the input is; ie input drops in time come out the same way */

static void

test_discont_stream_instance (int inrate, int outrate, int samples,

    int numbuffers)

{

  GstElement *audioresample;

  GstBuffer *inbuffer, *outbuffer;

  GstCaps *caps;

  GstClockTime ints;

  int i, j;

  gint16 *p;

  caps = gst_pad_get_negotiated_caps (mysrcpad);

  fail_unless (gst_caps_is_fixed (caps));

  fail_unless (gst_element_set_state (audioresample,

          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,

      "could not set to playing");

  for (j = 1; j <= numbuffers; ++j) {

    inbuffer = gst_buffer_new_and_alloc (samples * 4);

    GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;

    /* "drop" half the buffers */

    ints = GST_BUFFER_DURATION (inbuffer) * 2 * (j - 1);

    GST_BUFFER_TIMESTAMP (inbuffer) = ints;

    GST_BUFFER_OFFSET (inbuffer) = (j - 1) * 2 * samples;

    GST_BUFFER_OFFSET_END (inbuffer) = j * 2 * samples + samples;

    gst_buffer_set_caps (inbuffer, caps);

    p = (gint16 *) GST_BUFFER_DATA (inbuffer);

    /* create a 16 bit signed ramp */

    for (i = 0; i < samples; ++i) {

      *p = -32767 + i * (65535 / samples);

      ++p;

      *p = -32767 + i * (65535 / samples);

      ++p;

    }

    /* pushing gives away my reference ... */

    fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);

    /* check if the timestamp of the pushed buffer matches the incoming one */

    outbuffer = g_list_nth_data (buffers, g_list_length (buffers) - 1);

    fail_if (outbuffer == NULL);

    fail_unless_equals_uint64 (ints, GST_BUFFER_TIMESTAMP (outbuffer));

    if (j > 1) {

      fail_unless (GST_BUFFER_IS_DISCONT (outbuffer),

    }

  }

  /* cleanup */

  gst_caps_unref (caps);

  cleanup_audioresample (audioresample);

}

void test_discont_stream()

{

  std_log(LOG_FILENAME_LINE, "Test Started test_discont_stream");

  test_discont_stream_instance (48000, 24000, 500, 20);

  test_discont_stream_instance (48000, 12000, 500, 20);

  test_discont_stream_instance (12000, 24000, 500, 20);

  test_discont_stream_instance (12000, 48000, 500, 20);

  /* non-integral scalings */

  test_discont_stream_instance (44100, 8000, 500, 20);

  test_discont_stream_instance (8000, 44100, 500, 20);

  /* wacky scalings */

  test_discont_stream_instance (12345, 54321, 500, 20);

  test_discont_stream_instance (101, 99, 500, 20);

  std_log(LOG_FILENAME_LINE, "Test Successful");

  create_xml(0);

}

void test_reuse()

{

  GstElement *audioresample;

  GstEvent *newseg;

  GstBuffer *inbuffer;

  GstCaps *caps;

  xmlfile = "test_reuse";

  caps = gst_pad_get_negotiated_caps (mysrcpad);

  fail_unless (gst_caps_is_fixed (caps));

  fail_unless (gst_element_set_state (audioresample,

          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,

      "could not set to playing");

  newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);

  fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);

  inbuffer = gst_buffer_new_and_alloc (9343 * 4);

  memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));

  GST_BUFFER_DURATION (inbuffer) = GST_SECOND;

  GST_BUFFER_TIMESTAMP (inbuffer) = 0;

  GST_BUFFER_OFFSET (inbuffer) = 0;

  gst_buffer_set_caps (inbuffer, caps);

  /* pushing gives away my reference ... */

  fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);

  /* ... but it ends up being collected on the global buffer list */

  fail_unless_equals_int (g_list_length (buffers), 1);

  /* now reset and try again ... */

  fail_unless (gst_element_set_state (audioresample,

          GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");

  fail_unless (gst_element_set_state (audioresample,

          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,

      "could not set to playing");

  newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);

  fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);

  inbuffer = gst_buffer_new_and_alloc (9343 * 4);

  memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));

  GST_BUFFER_DURATION (inbuffer) = GST_SECOND;

  GST_BUFFER_TIMESTAMP (inbuffer) = 0;

  GST_BUFFER_OFFSET (inbuffer) = 0;

  gst_buffer_set_caps (inbuffer, caps);

  fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);

  /* ... it also ends up being collected on the global buffer list. If we

   * now have more than 2 buffers, then audioresample probably didn't clean

   * up its internal buffer properly and tried to push the remaining samples

   * when it got the second NEWSEGMENT event */

  fail_unless_equals_int (g_list_length (buffers), 2);

  cleanup_audioresample (audioresample);

  gst_caps_unref (caps);

  std_log(LOG_FILENAME_LINE, "Test Successful");

  create_xml(0);

}

void test_shutdown()

{

  GstElement *pipeline, *src, *cf1, *ar, *cf2, *sink;

  GstCaps *caps;

  guint i;

  xmlfile = "test_shutdown";

  /* create pipeline, force audioresample to actually resample */

  pipeline = gst_pipeline_new (NULL);

  src = gst_check_setup_element ("audiotestsrc");

  cf1 = gst_check_setup_element ("capsfilter");

  ar = gst_check_setup_element ("audioresample");

  cf2 = gst_check_setup_element ("capsfilter");

  g_object_set (cf2, "name", "capsfilter2", NULL);

  sink = gst_check_setup_element ("fakesink");

  caps =

      gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, 11025, NULL);

  g_object_set (cf1, "caps", caps, NULL);

  gst_caps_unref (caps);

  caps =

      gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, 48000, NULL);

  g_object_set (cf2, "caps", caps, NULL);

  gst_caps_unref (caps);

  /* don't want to sync against the clock, the more throughput the better */

  g_object_set (src, "is-live", FALSE, NULL);

  g_object_set (sink, "sync", FALSE, NULL);

  gst_bin_add_many (GST_BIN (pipeline), src, cf1, ar, cf2, sink, NULL);

  fail_if (!gst_element_link_many (src, cf1, ar, cf2, sink, NULL));