3 * unit test for mpg123audiodec
5 * Copyright (c) 2012 Carlos Rafael Giani <dv@pseudoterminal.org>
7 * This library is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Library General Public
9 * License as published by the Free Software Foundation; either
10 * version 2 of the License, or (at your option) any later version.
12 * This library is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Library General Public License for more details.
17 * You should have received a copy of the GNU Library General Public
18 * License along with this library; if not, write to the
19 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
20 * Boston, MA 02110-1301, USA.
25 #include <gst/check/gstcheck.h>
26 #include <gst/audio/audio.h>
28 #include <gst/fft/gstfft.h>
29 #include <gst/fft/gstffts16.h>
30 #include <gst/fft/gstffts32.h>
31 #include <gst/fft/gstfftf32.h>
32 #include <gst/fft/gstfftf64.h>
34 #include <gst/app/gstappsink.h>
36 /* For ease of programming we use globals to keep refs for our floating
37 * src and sink pads we create; otherwise we always have to do get_pad,
38 * get_peer, and then remove references in every test function */
39 static GstPad *mysrcpad, *mysinkpad;
42 #define MP2_STREAM_FILENAME "stream.mp2"
43 #define MP3_CBR_STREAM_FILENAME "cbr_stream.mp3"
44 #define MP3_VBR_STREAM_FILENAME "vbr_stream.mp3"
47 /* mpeg 1 layer 2 stream created with:
48 * gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
49 * "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
50 * avenc_mp2 bitrate=32000 ! tee name=t \
51 * t. ! queue ! fakesink silent=false \
52 * t. ! queue ! filesink location=test.mp2
54 * mpeg 1 layer 3 CBR stream created with:
55 * gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
56 * "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
57 * lamemp3enc encoding-engine-quality=high cbr=true target=bitrate bitrate=32 ! \
58 * "audio/mpeg, rate=(int)44100, channels=(int)1" ! tee name=t \
59 * t. ! queue ! fakesink silent=false \
60 * t. ! queue ! filesink location=test.mp3
62 * mpeg 1 layer 3 VBR stream created with:
63 * gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
64 * "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
65 * lamemp3enc encoding-engine-quality=high cbr=false target=quality quality=7 ! \
66 * "audio/mpeg, rate=(int)44100, channels=(int)1" ! tee name=t \
67 * t. ! queue ! fakesink silent=false \
68 * t. ! queue ! filesink location=test.mp3
72 /* FFT test helpers taken from gst-plugins-base tests/check/audioresample.c */
74 #define FFT_HELPERS(type,ffttag,ffttag2,scale) \
75 static gdouble magnitude##ffttag (const GstFFT##ffttag##Complex *c) \
77 gdouble mag = (gdouble) c->r * (gdouble) c->r; \
78 mag += (gdouble) c->i * (gdouble) c->i; \
79 mag /= scale * scale; \
80 mag = 10.0 * log10 (mag); \
83 static gdouble find_main_frequency_spot_##ffttag ( \
84 const GstFFT##ffttag##Complex *v, int elements) \
87 gdouble maxmag = -9999; \
89 for (i=0; i<elements; ++i) { \
90 gdouble mag = magnitude##ffttag (v+i); \
96 return maxidx / (gdouble) elements; \
98 static gboolean is_zero_except_##ffttag (const GstFFT##ffttag##Complex *v, \
99 int elements, gdouble spot) \
102 for (i=0; i<elements; ++i) { \
103 gdouble pos = i / (gdouble) elements; \
104 gdouble mag = magnitude##ffttag (v+i); \
105 if (fabs (pos - spot) > 0.01) { \
107 GST_LOG("Found magnitude at %f : %f (peak at %f)\n", pos, mag, spot); \
114 static void check_main_frequency_spot_##ffttag (GstBuffer *buffer, gdouble \
119 gdouble actual_spot; \
120 GstFFT##ffttag *ctx; \
121 GstFFT##ffttag##Complex *fftdata; \
123 gst_buffer_map (buffer, &map, GST_MAP_READ); \
125 num_samples = map.size / sizeof(type) & ~1; \
126 ctx = gst_fft_##ffttag2##_new (num_samples, FALSE); \
127 fftdata = g_new (GstFFT##ffttag##Complex, num_samples / 2 + 1); \
129 gst_fft_##ffttag2##_window (ctx, (type*)map.data, \
130 GST_FFT_WINDOW_HAMMING); \
131 gst_fft_##ffttag2##_fft (ctx, (type*)map.data, fftdata); \
133 actual_spot = find_main_frequency_spot_##ffttag (fftdata, \
134 num_samples / 2 + 1); \
135 GST_LOG ("Expected spot: %.3f actual: %.3f %f", expected_spot, actual_spot, \
136 fabs (expected_spot - actual_spot)); \
137 fail_unless (fabs (expected_spot - actual_spot) < 0.05, \
138 "Actual main frequency spot is too far away from expected one"); \
139 fail_unless (is_zero_except_##ffttag (fftdata, num_samples / 2 + 1, \
140 actual_spot), "One secondary peak in spectrum exceeds threshold"); \
142 gst_buffer_unmap (buffer, &map); \
144 gst_fft_##ffttag2##_free (ctx); \
147 FFT_HELPERS (gint32, S32, s32, 2147483647.0);
150 static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
153 GST_STATIC_CAPS ("audio/x-raw, format = (string) " GST_AUDIO_NE (S32))
155 static GstStaticPadTemplate layer2_srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
158 GST_STATIC_CAPS_ANY);
159 static GstStaticPadTemplate layer3_srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
162 GST_STATIC_CAPS_ANY);
166 setup_input_pipeline (gchar const *stream_filename, GstElement ** pipeline,
167 GstElement ** appsink)
169 GstElement *source, *parser;
171 *pipeline = gst_pipeline_new (NULL);
172 source = gst_element_factory_make ("filesrc", NULL);
173 parser = gst_element_factory_make ("mpegaudioparse", NULL);
174 *appsink = gst_element_factory_make ("appsink", NULL);
176 gst_bin_add_many (GST_BIN (*pipeline), source, parser, *appsink, NULL);
177 gst_element_link_many (source, parser, *appsink, NULL);
180 char *full_filename =
181 g_build_filename (GST_TEST_FILES_PATH, stream_filename, NULL);
182 g_object_set (G_OBJECT (source), "location", full_filename, NULL);
183 g_free (full_filename);
186 gst_element_set_state (*pipeline, GST_STATE_PLAYING);
190 cleanup_input_pipeline (GstElement * pipeline)
192 gst_element_set_state (pipeline, GST_STATE_NULL);
193 gst_object_unref (pipeline);
197 setup_mpeg1layer2dec (void)
199 GstElement *mpg123audiodec;
202 GST_DEBUG ("setup_mpeg1layer2dec");
203 mpg123audiodec = gst_check_setup_element ("mpg123audiodec");
204 mysrcpad = gst_check_setup_src_pad (mpg123audiodec, &layer2_srctemplate);
205 mysinkpad = gst_check_setup_sink_pad (mpg123audiodec, &sinktemplate);
206 gst_pad_set_active (mysrcpad, TRUE);
207 gst_pad_set_active (mysinkpad, TRUE);
209 /* This is necessary to trigger a set_format call in the decoder;
210 * fixed caps don't trigger it */
211 caps = gst_caps_new_simple ("audio/mpeg",
212 "mpegversion", G_TYPE_INT, 1,
213 "layer", G_TYPE_INT, 2,
214 "rate", G_TYPE_INT, 44100,
215 "channels", G_TYPE_INT, 1, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
216 gst_check_setup_events (mysrcpad, mpg123audiodec, caps, GST_FORMAT_TIME);
217 gst_caps_unref (caps);
219 return mpg123audiodec;
223 setup_mpeg1layer3dec (void)
225 GstElement *mpg123audiodec;
228 GST_DEBUG ("setup_mpeg1layer3dec");
229 mpg123audiodec = gst_check_setup_element ("mpg123audiodec");
230 mysrcpad = gst_check_setup_src_pad (mpg123audiodec, &layer3_srctemplate);
231 mysinkpad = gst_check_setup_sink_pad (mpg123audiodec, &sinktemplate);
232 gst_pad_set_active (mysrcpad, TRUE);
233 gst_pad_set_active (mysinkpad, TRUE);
235 /* This is necessary to trigger a set_format call in the decoder;
236 * fixed caps don't trigger it */
237 caps = gst_caps_new_simple ("audio/mpeg",
238 "mpegversion", G_TYPE_INT, 1,
239 "layer", G_TYPE_INT, 3,
240 "rate", G_TYPE_INT, 44100,
241 "channels", G_TYPE_INT, 1, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
242 gst_check_setup_events (mysrcpad, mpg123audiodec, caps, GST_FORMAT_TIME);
243 gst_caps_unref (caps);
245 return mpg123audiodec;
249 cleanup_mpg123audiodec (GstElement * mpg123audiodec)
251 GST_DEBUG ("cleanup_mpeg1layer2dec");
252 gst_element_set_state (mpg123audiodec, GST_STATE_NULL);
254 gst_pad_set_active (mysrcpad, FALSE);
255 gst_pad_set_active (mysinkpad, FALSE);
256 gst_check_teardown_src_pad (mpg123audiodec);
257 gst_check_teardown_sink_pad (mpg123audiodec);
258 gst_check_teardown_element (mpg123audiodec);
262 run_decoding_test (GstElement * mpg123audiodec, gchar const *filename)
265 unsigned int num_input_buffers, num_decoded_buffers;
267 GstCaps *out_caps, *caps;
268 GstAudioInfo audioinfo;
269 GstElement *input_pipeline, *input_appsink;
271 GstBuffer *outbuffer;
273 /* 440 Hz = frequency of sine wave in audio data
274 * 44100 Hz = sample rate
275 * (44100 / 2) Hz = Nyquist frequency */
276 static double const expected_frequency_spot = 440.0 / (44100.0 / 2.0);
278 fail_unless (gst_element_set_state (mpg123audiodec,
279 GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
280 "could not set to playing");
281 bus = gst_bus_new ();
283 gst_element_set_bus (mpg123audiodec, bus);
285 setup_input_pipeline (filename, &input_pipeline, &input_appsink);
287 num_input_buffers = 0;
290 GstBuffer *input_buffer;
292 sample = gst_app_sink_pull_sample (GST_APP_SINK (input_appsink));
296 fail_unless (GST_IS_SAMPLE (sample));
298 input_buffer = gst_sample_get_buffer (sample);
299 fail_if (input_buffer == NULL);
301 /* This is done to be on the safe side - docs say lifetime of the input buffer
302 * depends *solely* on the sample */
303 input_buffer = gst_buffer_copy (input_buffer);
305 fail_unless_equals_int (gst_pad_push (mysrcpad, input_buffer), GST_FLOW_OK);
309 gst_sample_unref (sample);
312 num_decoded_buffers = g_list_length (buffers);
314 /* check number of decoded buffers */
315 fail_unless_equals_int (num_decoded_buffers, num_input_buffers - 2);
317 caps = gst_pad_get_current_caps (mysinkpad);
318 GST_LOG ("output caps %" GST_PTR_FORMAT, caps);
319 fail_unless (gst_audio_info_from_caps (&audioinfo, caps),
320 "Getting audio info from caps failed");
323 out_caps = gst_caps_new_simple ("audio/x-raw",
324 "format", G_TYPE_STRING, GST_AUDIO_NE (S32),
325 "layout", G_TYPE_STRING, "interleaved",
326 "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1, NULL);
328 fail_unless (gst_caps_is_equal_fixed (caps, out_caps), "Incorrect out caps");
330 gst_caps_unref (out_caps);
331 gst_caps_unref (caps);
333 /* here, test if decoded data is a sine tone, and if the sine frequency is at the
334 * right spot in the spectrum */
335 for (i = 0; i < num_decoded_buffers; ++i) {
336 outbuffer = GST_BUFFER (buffers->data);
337 fail_if (outbuffer == NULL, "Invalid buffer retrieved");
339 /* MPEG 1 layer 2 uses 1152 samples per frame */
340 expected_size = 1152 * GST_AUDIO_INFO_BPF (&audioinfo);
341 fail_unless_equals_int (gst_buffer_get_size (outbuffer), expected_size);
343 check_main_frequency_spot_S32 (outbuffer, expected_frequency_spot);
345 buffers = g_list_remove (buffers, outbuffer);
346 gst_buffer_unref (outbuffer);
350 g_list_free (buffers);
353 cleanup_input_pipeline (input_pipeline);
354 gst_bus_set_flushing (bus, TRUE);
355 gst_element_set_bus (mpg123audiodec, NULL);
356 gst_object_unref (GST_OBJECT (bus));
360 GST_START_TEST (test_decode_mpeg1layer2)
362 GstElement *mpg123audiodec;
363 mpg123audiodec = setup_mpeg1layer2dec ();
364 run_decoding_test (mpg123audiodec, MP2_STREAM_FILENAME);
365 cleanup_mpg123audiodec (mpg123audiodec);
366 mpg123audiodec = NULL;
372 GST_START_TEST (test_decode_mpeg1layer3_cbr)
374 GstElement *mpg123audiodec;
375 mpg123audiodec = setup_mpeg1layer3dec ();
376 run_decoding_test (mpg123audiodec, MP3_CBR_STREAM_FILENAME);
377 cleanup_mpg123audiodec (mpg123audiodec);
383 GST_START_TEST (test_decode_mpeg1layer3_vbr)
385 GstElement *mpg123audiodec;
386 mpg123audiodec = setup_mpeg1layer3dec ();
387 run_decoding_test (mpg123audiodec, MP3_VBR_STREAM_FILENAME);
388 cleanup_mpg123audiodec (mpg123audiodec);
394 GST_START_TEST (test_decode_garbage_mpeg1layer2)
396 GstElement *mpg123audiodec;
402 mpg123audiodec = setup_mpeg1layer2dec ();
404 fail_unless (gst_element_set_state (mpg123audiodec,
405 GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
406 "could not set to playing");
407 bus = gst_bus_new ();
409 /* initialize the buffer with something that is no mpeg2 */
410 tmpbuf = g_new (guint32, 4096);
411 for (i = 0; i < 4096; i++) {
414 inbuffer = gst_buffer_new_wrapped (tmpbuf, 4096 * sizeof (guint32));
416 ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
418 gst_element_set_bus (mpg123audiodec, bus);
420 /* should be possible to push without problems but nothing gets decoded */
421 fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
423 num_buffers = g_list_length (buffers);
425 /* should be 0 buffers as decoding should've been impossible */
426 fail_unless_equals_int (num_buffers, 0);
428 g_list_free (buffers);
431 gst_bus_set_flushing (bus, TRUE);
432 gst_element_set_bus (mpg123audiodec, NULL);
433 gst_object_unref (GST_OBJECT (bus));
434 cleanup_mpg123audiodec (mpg123audiodec);
435 mpg123audiodec = NULL;
441 GST_START_TEST (test_decode_garbage_mpeg1layer3)
443 GstElement *mpg123audiodec;
449 mpg123audiodec = setup_mpeg1layer3dec ();
451 fail_unless (gst_element_set_state (mpg123audiodec,
452 GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
453 "could not set to playing");
454 bus = gst_bus_new ();
456 /* initialize the buffer with something that is no mpeg2 */
457 tmpbuf = g_new (guint32, 4096);
458 for (i = 0; i < 4096; i++) {
461 inbuffer = gst_buffer_new_wrapped (tmpbuf, 4096 * sizeof (guint32));
463 ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
465 gst_element_set_bus (mpg123audiodec, bus);
467 /* should be possible to push without problems but nothing gets decoded */
468 fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
470 num_buffers = g_list_length (buffers);
472 /* should be 0 buffers as decoding should've been impossible */
473 fail_unless_equals_int (num_buffers, 0);
475 g_list_free (buffers);
478 gst_bus_set_flushing (bus, TRUE);
479 gst_element_set_bus (mpg123audiodec, NULL);
480 gst_object_unref (GST_OBJECT (bus));
481 cleanup_mpg123audiodec (mpg123audiodec);
482 mpg123audiodec = NULL;
489 is_test_file_available (gchar const *filename)
492 gchar *full_filename;
495 cwd = g_get_current_dir ();
496 full_filename = g_build_filename (cwd, GST_TEST_FILES_PATH, filename, NULL);
498 g_file_test (full_filename, G_FILE_TEST_IS_REGULAR | G_FILE_TEST_EXISTS);
499 g_free (full_filename);
505 mpg123audiodec_suite (void)
507 GstRegistry *registry;
508 Suite *s = suite_create ("mpg123audiodec");
509 TCase *tc_chain = tcase_create ("general");
511 registry = gst_registry_get ();
513 suite_add_tcase (s, tc_chain);
514 if (gst_registry_check_feature_version (registry, "filesrc",
515 GST_VERSION_MAJOR, GST_VERSION_MINOR, 0) &&
516 gst_registry_check_feature_version (registry, "mpegaudioparse",
517 GST_VERSION_MAJOR, GST_VERSION_MINOR, 0) &&
518 gst_registry_check_feature_version (registry, "appsrc",
519 GST_VERSION_MAJOR, GST_VERSION_MINOR, 0)) {
520 if (is_test_file_available (MP2_STREAM_FILENAME))
521 tcase_add_test (tc_chain, test_decode_mpeg1layer2);
522 if (is_test_file_available (MP3_CBR_STREAM_FILENAME))
523 tcase_add_test (tc_chain, test_decode_mpeg1layer3_cbr);
524 if (is_test_file_available (MP3_VBR_STREAM_FILENAME))
525 tcase_add_test (tc_chain, test_decode_mpeg1layer3_vbr);
527 tcase_add_test (tc_chain, test_decode_garbage_mpeg1layer2);
528 tcase_add_test (tc_chain, test_decode_garbage_mpeg1layer3);
534 GST_CHECK_MAIN (mpg123audiodec)