tets: add unit test for mpg123audiodec

https://bugzilla.gnome.org/show_bug.cgi?id=686595

5 files changed, 586 insertions, 0 deletions

diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index c1a87298e..f9556a6c5 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -212,6 +212,7 @@ check_PROGRAMS = \
 	elements/mpegtsmux \
 	elements/mpegvideoparse \
 	elements/mpeg4videoparse \
+	elements/mpg123audiodec \
 	elements/mxfdemux \
 	elements/mxfmux \
 	elements/id3mux \
@@ -351,6 +352,10 @@ elements_assrender_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_API_VERSION
 elements_mpegtsmux_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
 elements_mpegtsmux_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_API_VERSION) $(GST_BASE_LIBS) $(LDADD)
 
+elements_mpg123audiodec_LDADD = \
+	$(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) $(LDADD) \
+	-lgstaudio-@GST_API_VERSION@ -lgstfft-@GST_API_VERSION@ -lgstapp-@GST_API_VERSION@
+
 elements_uvch264demux_CFLAGS = -DUVCH264DEMUX_DATADIR="$(srcdir)/elements/uvch264demux_data" \
 				$(AM_CFLAGS)
 
diff --git a/tests/check/elements/mpg123audiodec.c b/tests/check/elements/mpg123audiodec.c
new file mode 100644
index 000000000..fd7c620e3
--- /dev/null
+++ b/tests/check/elements/mpg123audiodec.c
@@ -0,0 +1,581 @@
+/* GStreamer
+ *
+ * unit test for mpg123audiodec
+ *
+ * Copyright (c) 2012 Carlos Rafael Giani <dv@pseudoterminal.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/audio.h>
+
+#include <gst/fft/gstfft.h>
+#include <gst/fft/gstffts16.h>
+#include <gst/fft/gstffts32.h>
+#include <gst/fft/gstfftf32.h>
+#include <gst/fft/gstfftf64.h>
+
+#include <gst/app/gstappsink.h>
+
+/* 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 MP2_STREAM_FILENAME "stream.mp2"
+#define MP3_CBR_STREAM_FILENAME "cbr_stream.mp3"
+#define MP3_VBR_STREAM_FILENAME "vbr_stream.mp3"
+
+
+/* mpeg 1 layer 2 stream created with:
+ * gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
+ *   "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
+ *   avenc_mp2 bitrate=32000 ! tee name=t \
+ *   t. ! queue ! fakesink silent=false \
+ *   t. ! queue ! filesink location=test.mp2
+ *
+ * mpeg 1 layer 3 CBR stream created with:
+ * gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
+ *   "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
+ *   lamemp3enc encoding-engine-quality=high cbr=true target=bitrate bitrate=32 ! \
+ *   "audio/mpeg, rate=(int)44100, channels=(int)1" ! tee name=t \
+ *   t. ! queue ! fakesink silent=false \
+ *   t. ! queue ! filesink location=test.mp3
+ *
+ * mpeg 1 layer 3 VBR stream created with:
+ * gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
+ *   "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
+ *   lamemp3enc encoding-engine-quality=high cbr=false target=quality quality=7 ! \
+ *   "audio/mpeg, rate=(int)44100, channels=(int)1" ! tee name=t \
+ *   t. ! queue ! fakesink silent=false \
+ *   t. ! queue ! filesink location=test.mp3
+ */
+
+
+/* FFT test helpers taken from gst-plugins-base tests/check/audioresample.c */
+
+#define FFT_HELPERS(type,ffttag,ffttag2,scale)                                \
+static gdouble magnitude##ffttag (const GstFFT##ffttag##Complex *c)           \
+{                                                                             \
+  gdouble mag = (gdouble) c->r * (gdouble) c->r;                              \
+  mag += (gdouble) c->i * (gdouble) c->i;                                     \
+  mag /= scale * scale;                                                       \
+  mag = 10.0 * log10 (mag);                                                   \
+  return mag;                                                                 \
+}                                                                             \
+static gdouble find_main_frequency_spot_##ffttag (                            \
+    const GstFFT##ffttag##Complex *v, int elements)                           \
+{                                                                             \
+  int i;                                                                      \
+  gdouble maxmag = -9999;                                                     \
+  int maxidx = 0;                                                             \
+  for (i=0; i<elements; ++i) {                                                \
+    gdouble mag = magnitude##ffttag (v+i);                                    \
+    if (mag > maxmag) {                                                       \
+      maxmag = mag;                                                           \
+      maxidx = i;                                                             \
+    }                                                                         \
+  }                                                                           \
+  return maxidx / (gdouble) elements;                                         \
+}                                                                             \
+static gboolean is_zero_except_##ffttag (const GstFFT##ffttag##Complex *v,    \
+    int elements, gdouble spot)                                               \
+{                                                                             \
+  int i;                                                                      \
+  for (i=0; i<elements; ++i) {                                                \
+    gdouble pos = i / (gdouble) elements;                                     \
+    gdouble mag = magnitude##ffttag (v+i);                                    \
+    if (fabs (pos - spot) > 0.01) {                                           \
+      if (mag > -35.0) {                                                      \
+        GST_LOG("Found magnitude at %f : %f (peak at %f)\n", pos, mag, spot); \
+        return FALSE;                                                         \
+      }                                                                       \
+    }                                                                         \
+  }                                                                           \
+  return TRUE;                                                                \
+}                                                                             \
+static void check_main_frequency_spot_##ffttag (GstBuffer *buffer, gdouble    \
+    expected_spot)                                                            \
+{                                                                             \
+  GstMapInfo map;                                                             \
+  int num_samples;                                                            \
+  gdouble actual_spot;                                                        \
+  GstFFT##ffttag *ctx;                                                        \
+  GstFFT##ffttag##Complex *fftdata;                                           \
+                                                                              \
+  gst_buffer_map (buffer, &map, GST_MAP_READ);                                \
+                                                                              \
+  num_samples = map.size / sizeof(type) & ~1;                                 \
+  ctx = gst_fft_##ffttag2##_new (num_samples, FALSE);                         \
+  fftdata = g_new (GstFFT##ffttag##Complex, num_samples / 2 + 1);             \
+                                                                              \
+  gst_fft_##ffttag2##_window (ctx, (type*)map.data,                           \
+    GST_FFT_WINDOW_HAMMING);                                                  \
+  gst_fft_##ffttag2##_fft (ctx, (type*)map.data, fftdata);                    \
+                                                                              \
+  actual_spot = find_main_frequency_spot_##ffttag (fftdata,                   \
+    num_samples / 2 + 1);                                                     \
+  GST_LOG ("Expected spot: %.3f actual: %.3f %f", expected_spot, actual_spot, \
+    fabs (expected_spot - actual_spot));                                      \
+  fail_unless (fabs (expected_spot - actual_spot) < 0.05,                     \
+    "Actual main frequency spot is too far away from expected one");          \
+  fail_unless (is_zero_except_##ffttag (fftdata, num_samples / 2 + 1,         \
+    actual_spot), "One secondary peak in spectrum exceeds threshold");        \
+                                                                              \
+  gst_buffer_unmap (buffer, &map);                                            \
+                                                                              \
+  gst_fft_##ffttag2##_free (ctx);                                             \
+  g_free (fftdata);                                                           \
+}
+FFT_HELPERS (gint32, S32, s32, 2147483647.0);
+
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw, format = (string) S32LE ")
+    );
+static GstStaticPadTemplate layer2_srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS_ANY);
+static GstStaticPadTemplate layer3_srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS_ANY);
+
+
+static void
+setup_input_pipeline (gchar const *stream_filename, GstElement ** pipeline,
+    GstElement ** appsink)
+{
+  GstElement *source, *parser;
+
+  *pipeline = gst_pipeline_new (NULL);
+  source = gst_element_factory_make ("filesrc", NULL);
+  parser = gst_element_factory_make ("mpegaudioparse", NULL);
+  *appsink = gst_element_factory_make ("appsink", NULL);
+
+  gst_bin_add_many (GST_BIN (*pipeline), source, parser, *appsink, NULL);
+  gst_element_link_many (source, parser, *appsink, NULL);
+
+  {
+    char *full_filename =
+        g_build_filename (GST_TEST_FILES_PATH, stream_filename, NULL);
+    g_object_set (G_OBJECT (source), "location", full_filename, NULL);
+    g_free (full_filename);
+  }
+
+  gst_element_set_state (*pipeline, GST_STATE_PLAYING);
+}
+
+static void
+cleanup_input_pipeline (GstElement * pipeline)
+{
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_object_unref (pipeline);
+}
+
+static GstElement *
+setup_mpeg1layer2dec (void)
+{
+  GstElement *mpg123audiodec;
+  GstSegment seg;
+  GstCaps *caps;
+
+  GST_DEBUG ("setup_mpeg1layer2dec");
+  mpg123audiodec = gst_check_setup_element ("mpg123audiodec");
+  mysrcpad = gst_check_setup_src_pad (mpg123audiodec, &layer2_srctemplate);
+  mysinkpad = gst_check_setup_sink_pad (mpg123audiodec, &sinktemplate);
+  gst_pad_set_active (mysrcpad, TRUE);
+  gst_pad_set_active (mysinkpad, TRUE);
+
+  gst_segment_init (&seg, GST_FORMAT_TIME);
+  gst_pad_push_event (mysrcpad, gst_event_new_segment (&seg));
+
+  /* This is necessary to trigger a set_format call in the decoder;
+   * fixed caps don't trigger it */
+  caps = gst_caps_new_simple ("audio/mpeg",
+      "mpegversion", G_TYPE_INT, 1,
+      "layer", G_TYPE_INT, 2,
+      "rate", G_TYPE_INT, 44100,
+      "channels", G_TYPE_INT, 1, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
+  gst_pad_set_caps (mysrcpad, caps);
+  gst_caps_unref (caps);
+
+  return mpg123audiodec;
+}
+
+static GstElement *
+setup_mpeg1layer3dec (void)
+{
+  GstElement *mpg123audiodec;
+  GstSegment seg;
+  GstCaps *caps;
+
+  GST_DEBUG ("setup_mpeg1layer3dec");
+  mpg123audiodec = gst_check_setup_element ("mpg123audiodec");
+  mysrcpad = gst_check_setup_src_pad (mpg123audiodec, &layer3_srctemplate);
+  mysinkpad = gst_check_setup_sink_pad (mpg123audiodec, &sinktemplate);
+  gst_pad_set_active (mysrcpad, TRUE);
+  gst_pad_set_active (mysinkpad, TRUE);
+
+  gst_segment_init (&seg, GST_FORMAT_TIME);
+  gst_pad_push_event (mysrcpad, gst_event_new_segment (&seg));
+
+  /* This is necessary to trigger a set_format call in the decoder;
+   * fixed caps don't trigger it */
+  caps = gst_caps_new_simple ("audio/mpeg",
+      "mpegversion", G_TYPE_INT, 1,
+      "layer", G_TYPE_INT, 3,
+      "rate", G_TYPE_INT, 44100,
+      "channels", G_TYPE_INT, 1, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
+  gst_pad_set_caps (mysrcpad, caps);
+  gst_caps_unref (caps);
+
+  return mpg123audiodec;
+}
+
+static void
+cleanup_mpg123audiodec (GstElement * mpg123audiodec)
+{
+  GST_DEBUG ("cleanup_mpeg1layer2dec");
+  gst_element_set_state (mpg123audiodec, GST_STATE_NULL);
+
+  gst_pad_set_active (mysrcpad, FALSE);
+  gst_pad_set_active (mysinkpad, FALSE);
+  gst_check_teardown_src_pad (mpg123audiodec);
+  gst_check_teardown_sink_pad (mpg123audiodec);
+  gst_check_teardown_element (mpg123audiodec);
+}
+
+static void
+run_decoding_test (GstElement * mpg123audiodec, gchar const *filename)
+{
+  GstBus *bus;
+  unsigned int num_input_buffers, num_decoded_buffers;
+  gint expected_size;
+  GstCaps *out_caps, *caps;
+  GstAudioInfo audioinfo;
+  GstElement *input_pipeline, *input_appsink;
+  int i;
+  GstBuffer *outbuffer;
+
+  /* 440 Hz = frequency of sine wave in audio data
+   * 44100 Hz = sample rate
+   * (44100 / 2) Hz = Nyquist frequency */
+  static double const expected_frequency_spot = 440.0 / (44100.0 / 2.0);
+
+  fail_unless (gst_element_set_state (mpg123audiodec,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+  bus = gst_bus_new ();
+
+  gst_element_set_bus (mpg123audiodec, bus);
+
+  setup_input_pipeline (filename, &input_pipeline, &input_appsink);
+
+  num_input_buffers = 0;
+  while (TRUE) {
+    GstSample *sample;
+    GstBuffer *input_buffer;
+
+    sample = gst_app_sink_pull_sample (GST_APP_SINK (input_appsink));
+    if (sample == NULL)
+      break;
+
+    fail_unless (GST_IS_SAMPLE (sample));
+
+    input_buffer = gst_sample_get_buffer (sample);
+    fail_if (input_buffer == NULL);
+
+    /* This is done to be on the safe side - docs say lifetime of the input buffer
+     * depends *solely* on the sample */
+    input_buffer = gst_buffer_copy (input_buffer);
+
+    fail_unless_equals_int (gst_pad_push (mysrcpad, input_buffer), GST_FLOW_OK);
+
+    ++num_input_buffers;
+  }
+
+  num_decoded_buffers = g_list_length (buffers);
+
+  /* check number of decoded buffers */
+  fail_unless_equals_int (num_decoded_buffers, num_input_buffers - 2);
+
+  caps = gst_pad_get_current_caps (mysinkpad);
+  GST_LOG ("output caps %" GST_PTR_FORMAT, caps);
+  fail_unless (gst_audio_info_from_caps (&audioinfo, caps),
+      "Getting audio info from caps failed");
+
+  /* check caps */
+  out_caps = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, "S32LE",
+      "layout", G_TYPE_STRING, "interleaved",
+      "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1, NULL);
+
+  fail_unless (gst_caps_is_equal_fixed (caps, out_caps), "Incorrect out caps");
+
+  gst_caps_unref (out_caps);
+  gst_caps_unref (caps);
+
+  /* here, test if decoded data is a sine tone, and if the sine frequency is at the
+   * right spot in the spectrum */
+  for (i = 0; i < num_decoded_buffers; ++i) {
+    outbuffer = GST_BUFFER (buffers->data);
+    fail_if (outbuffer == NULL, "Invalid buffer retrieved");
+
+    /* MPEG 1 layer 2 uses 1152 samples per frame */
+    expected_size = 1152 * GST_AUDIO_INFO_BPF (&audioinfo);
+    fail_unless_equals_int (gst_buffer_get_size (outbuffer), expected_size);
+
+    check_main_frequency_spot_S32 (outbuffer, expected_frequency_spot);
+
+    buffers = g_list_remove (buffers, outbuffer);
+    gst_buffer_unref (outbuffer);
+    outbuffer = NULL;
+  }
+
+  g_list_free (buffers);
+  buffers = NULL;
+
+  cleanup_input_pipeline (input_pipeline);
+  gst_bus_set_flushing (bus, TRUE);
+  gst_element_set_bus (mpg123audiodec, NULL);
+  gst_object_unref (GST_OBJECT (bus));
+}
+
+
+GST_START_TEST (test_decode_mpeg1layer2)
+{
+  GstElement *mpg123audiodec;
+  mpg123audiodec = setup_mpeg1layer2dec ();
+  run_decoding_test (mpg123audiodec, MP2_STREAM_FILENAME);
+  cleanup_mpg123audiodec (mpg123audiodec);
+  mpg123audiodec = NULL;
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_decode_mpeg1layer3_cbr)
+{
+  GstElement *mpg123audiodec;
+  mpg123audiodec = setup_mpeg1layer3dec ();
+  run_decoding_test (mpg123audiodec, MP3_CBR_STREAM_FILENAME);
+  cleanup_mpg123audiodec (mpg123audiodec);
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_decode_mpeg1layer3_vbr)
+{
+  GstElement *mpg123audiodec;
+  mpg123audiodec = setup_mpeg1layer3dec ();
+  run_decoding_test (mpg123audiodec, MP3_VBR_STREAM_FILENAME);
+  cleanup_mpg123audiodec (mpg123audiodec);
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_decode_garbage_mpeg1layer2)
+{
+  GstElement *mpg123audiodec;
+  GstBuffer *inbuffer;
+  GstBus *bus;
+  int i, num_buffers;
+  guint32 *tmpbuf;
+
+  mpg123audiodec = setup_mpeg1layer2dec ();
+
+  fail_unless (gst_element_set_state (mpg123audiodec,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+  bus = gst_bus_new ();
+
+  /* initialize the buffer with something that is no mpeg2 */
+  tmpbuf = g_new (guint32, 4096);
+  for (i = 0; i < 4096; i++) {
+    tmpbuf[i] = i;
+  }
+  inbuffer = gst_buffer_new_wrapped (tmpbuf, 4096 * sizeof (guint32));
+
+  ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+  gst_element_set_bus (mpg123audiodec, bus);
+
+  /* should be possible to push without problems but nothing gets decoded */
+  fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
+
+  num_buffers = g_list_length (buffers);
+
+  /* should be 0 buffers as decoding should've been impossible */
+  fail_unless_equals_int (num_buffers, 0);
+
+  g_list_free (buffers);
+  buffers = NULL;
+
+  gst_bus_set_flushing (bus, TRUE);
+  gst_element_set_bus (mpg123audiodec, NULL);
+  gst_object_unref (GST_OBJECT (bus));
+  cleanup_mpg123audiodec (mpg123audiodec);
+  mpg123audiodec = NULL;
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_decode_garbage_mpeg1layer3)
+{
+  GstElement *mpg123audiodec;
+  GstBuffer *inbuffer;
+  GstBus *bus;
+  int i, num_buffers;
+  guint32 *tmpbuf;
+
+  mpg123audiodec = setup_mpeg1layer3dec ();
+
+  fail_unless (gst_element_set_state (mpg123audiodec,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+  bus = gst_bus_new ();
+
+  /* initialize the buffer with something that is no mpeg2 */
+  tmpbuf = g_new (guint32, 4096);
+  for (i = 0; i < 4096; i++) {
+    tmpbuf[i] = i;
+  }
+  inbuffer = gst_buffer_new_wrapped (tmpbuf, 4096 * sizeof (guint32));
+
+  ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+  gst_element_set_bus (mpg123audiodec, bus);
+
+  /* should be possible to push without problems but nothing gets decoded */
+  fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
+
+  num_buffers = g_list_length (buffers);
+
+  /* should be 0 buffers as decoding should've been impossible */
+  fail_unless_equals_int (num_buffers, 0);
+
+  g_list_free (buffers);
+  buffers = NULL;
+
+  gst_bus_set_flushing (bus, TRUE);
+  gst_element_set_bus (mpg123audiodec, NULL);
+  gst_object_unref (GST_OBJECT (bus));
+  cleanup_mpg123audiodec (mpg123audiodec);
+  mpg123audiodec = NULL;
+}
+
+GST_END_TEST;
+
+
+static gboolean
+is_test_file_available (gchar const *filename)
+{
+  gboolean ret;
+  gchar *full_filename;
+  gchar *cwd;
+
+  cwd = g_get_current_dir ();
+  full_filename = g_build_filename (cwd, GST_TEST_FILES_PATH, filename, NULL);
+  ret =
+      g_file_test (full_filename, G_FILE_TEST_IS_REGULAR | G_FILE_TEST_EXISTS);
+  g_free (full_filename);
+  g_free (cwd);
+  return ret;
+}
+
+
+static Suite *
+mpg123audiodec_suite (void)
+{
+  gboolean has_necessary_elements = TRUE;
+  Suite *s = suite_create ("mpg123audiodec");
+  TCase *tc_chain = tcase_create ("general");
+
+  /* check if mpegaudioparse, appsink, and filesrc elments are available */
+  {
+    gchar const **element;
+    gchar const *elements[] = { "filesrc", "mpegaudioparse", "appsink", NULL };
+
+    for (element = elements; *element != NULL; ++element) {
+      GstElement *e;
+      GstStateChangeReturn ret;
+
+      e = gst_element_factory_make (*element, NULL);
+      if (e == NULL) {
+        has_necessary_elements = FALSE;
+        break;
+      }
+
+      ret = gst_element_set_state (e, GST_STATE_READY);
+      if (ret == GST_STATE_CHANGE_SUCCESS) {
+        gst_element_set_state (e, GST_STATE_NULL);
+        gst_object_unref (GST_OBJECT (e));
+      } else {
+        gst_object_unref (GST_OBJECT (e));
+        has_necessary_elements = FALSE;
+        break;
+      }
+    }
+  }
+
+  suite_add_tcase (s, tc_chain);
+  if (has_necessary_elements) {
+    if (is_test_file_available (MP2_STREAM_FILENAME))
+      tcase_add_test (tc_chain, test_decode_mpeg1layer2);
+    if (is_test_file_available (MP3_CBR_STREAM_FILENAME))
+      tcase_add_test (tc_chain, test_decode_mpeg1layer3_cbr);
+    if (is_test_file_available (MP3_VBR_STREAM_FILENAME))
+      tcase_add_test (tc_chain, test_decode_mpeg1layer3_vbr);
+  }
+  tcase_add_test (tc_chain, test_decode_garbage_mpeg1layer2);
+  tcase_add_test (tc_chain, test_decode_garbage_mpeg1layer3);
+
+  return s;
+}
+
+
+int
+main (int argc, char **argv)
+{
+  int nf;
+  Suite *s;
+  SRunner *sr;
+
+  gst_check_init (&argc, &argv);
+
+  s = mpg123audiodec_suite ();
+  if (s == NULL)
+    return 0;
+
+  sr = srunner_create (s);
+
+  srunner_run_all (sr, CK_NORMAL);
+  nf = srunner_ntests_failed (sr);
+  srunner_free (sr);
+
+  return nf;
+}
diff --git a/tests/files/cbr_stream.mp3 b/tests/files/cbr_stream.mp3
new file mode 100644
index 000000000..b1a5c439d
--- /dev/null
+++ b/tests/files/cbr_stream.mp3
diff --git a/tests/files/stream.mp2 b/tests/files/stream.mp2
new file mode 100644
index 000000000..ab6e900d4
--- /dev/null
+++ b/tests/files/stream.mp2
diff --git a/tests/files/vbr_stream.mp3 b/tests/files/vbr_stream.mp3
new file mode 100644
index 000000000..81fc38b59
--- /dev/null
+++ b/tests/files/vbr_stream.mp3