--- /dev/null
+/* 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;
+}
projects / platform / upstream / gst-plugins-ugly.git / commitdiff