1 /* GStreamer
2 *
3 * unit test for audioresample, based on the audioresample unit test
4 *
5 * Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
6 * Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
7 *
8 * This library is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Library General Public
10 * License as published by the Free Software Foundation; either
11 * version 2 of the License, or (at your option) any later version.
12 *
13 * This library is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Library General Public License for more details.
17 *
18 * You should have received a copy of the GNU Library General Public
19 * License along with this library; if not, write to the
20 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
21 * Boston, MA 02110-1301, USA.
22 */
23 #ifdef HAVE_CONFIG_H
24 #include "config.h"
25 #endif
26
27 #include <gst/check/gstcheck.h>
28
29 #include <gst/audio/audio.h>
30
31 #include <gst/fft/gstfft.h>
32 #include <gst/fft/gstffts16.h>
33 #include <gst/fft/gstffts32.h>
34 #include <gst/fft/gstfftf32.h>
35 #include <gst/fft/gstfftf64.h>
36
37 /* For ease of programming we use globals to keep refs for our floating
38 * src and sink pads we create; otherwise we always have to do get_pad,
39 * get_peer, and then remove references in every test function */
40 static GstPad *mysrcpad, *mysinkpad;
41
42 #if G_BYTE_ORDER == G_LITTLE_ENDIAN
43 #define FORMATS "{ F32LE, F64LE, S16LE, S32LE }"
44 #else
45 #define FORMATS "{ F32BE, F64BE, S16BE, S32BE }"
46 #endif
47
48 #define RESAMPLE_CAPS \
49 "audio/x-raw, " \
50 "format = (string) "FORMATS", " \
51 "channels = (int) [ 1, MAX ], " \
52 "rate = (int) [ 1, MAX ], " \
53 "layout = (string) interleaved"
54
55 static GstElement *
setup_audioresample(int channels,guint64 mask,int inrate,int outrate,const gchar * format)56 setup_audioresample (int channels, guint64 mask, int inrate, int outrate,
57 const gchar * format)
58 {
59 GstPadTemplate *sinktemplate;
60 static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
61 GST_PAD_SRC,
62 GST_PAD_ALWAYS,
63 GST_STATIC_CAPS (RESAMPLE_CAPS)
64 );
65 GstElement *audioresample;
66 GstCaps *caps;
67 GstStructure *structure;
68
69 GST_DEBUG ("setup_audioresample");
70 audioresample = gst_check_setup_element ("audioresample");
71
72 caps = gst_caps_from_string (RESAMPLE_CAPS);
73 structure = gst_caps_get_structure (caps, 0);
74 gst_structure_set (structure, "channels", G_TYPE_INT, channels,
75 "rate", G_TYPE_INT, inrate, "format", G_TYPE_STRING, format,
76 "channel-mask", GST_TYPE_BITMASK, mask, NULL);
77 fail_unless (gst_caps_is_fixed (caps));
78
79 fail_unless (gst_element_set_state (audioresample,
80 GST_STATE_PAUSED) == GST_STATE_CHANGE_SUCCESS,
81 "could not set to paused");
82
83 mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate);
84 gst_pad_set_active (mysrcpad, TRUE);
85 gst_check_setup_events (mysrcpad, audioresample, caps, GST_FORMAT_TIME);
86 gst_caps_unref (caps);
87
88 caps = gst_caps_from_string (RESAMPLE_CAPS);
89 structure = gst_caps_get_structure (caps, 0);
90 gst_structure_set (structure, "channels", G_TYPE_INT, channels,
91 "rate", G_TYPE_INT, outrate, "format", G_TYPE_STRING, format, NULL);
92 fail_unless (gst_caps_is_fixed (caps));
93 sinktemplate =
94 gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, caps);
95
96 mysinkpad =
97 gst_check_setup_sink_pad_from_template (audioresample, sinktemplate);
98 gst_pad_set_active (mysinkpad, TRUE);
99 /* this installs a getcaps func that will always return the caps we set
100 * later */
101 gst_pad_use_fixed_caps (mysinkpad);
102
103 gst_caps_unref (caps);
104 gst_object_unref (sinktemplate);
105
106 return audioresample;
107 }
108
109 static void
cleanup_audioresample(GstElement * audioresample)110 cleanup_audioresample (GstElement * audioresample)
111 {
112 GST_DEBUG ("cleanup_audioresample");
113
114 fail_unless (gst_element_set_state (audioresample,
115 GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
116
117 gst_pad_set_active (mysrcpad, FALSE);
118 gst_pad_set_active (mysinkpad, FALSE);
119 gst_check_teardown_src_pad (audioresample);
120 gst_check_teardown_sink_pad (audioresample);
121 gst_check_teardown_element (audioresample);
122 gst_check_drop_buffers ();
123 }
124
125 static void
fail_unless_perfect_stream(void)126 fail_unless_perfect_stream (void)
127 {
128 guint64 timestamp = 0L, duration = 0L;
129 guint64 offset = 0L, offset_end = 0L;
130
131 GList *l;
132 GstBuffer *buffer;
133
134 for (l = buffers; l; l = l->next) {
135 buffer = GST_BUFFER (l->data);
136 ASSERT_BUFFER_REFCOUNT (buffer, "buffer", 1);
137 GST_DEBUG ("buffer timestamp %" G_GUINT64_FORMAT ", duration %"
138 G_GUINT64_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %"
139 G_GUINT64_FORMAT,
140 GST_BUFFER_TIMESTAMP (buffer),
141 GST_BUFFER_DURATION (buffer),
142 GST_BUFFER_OFFSET (buffer), GST_BUFFER_OFFSET_END (buffer));
143
144 fail_unless_equals_uint64 (timestamp, GST_BUFFER_TIMESTAMP (buffer));
145 fail_unless_equals_uint64 (offset, GST_BUFFER_OFFSET (buffer));
146 duration = GST_BUFFER_DURATION (buffer);
147 offset_end = GST_BUFFER_OFFSET_END (buffer);
148
149 timestamp += duration;
150 offset = offset_end;
151 gst_buffer_unref (buffer);
152 }
153 g_list_free (buffers);
154 buffers = NULL;
155 }
156
157 /* this tests that the output is a perfect stream if the input is */
158 static void
test_perfect_stream_instance(int inrate,int outrate,int samples,int numbuffers)159 test_perfect_stream_instance (int inrate, int outrate, int samples,
160 int numbuffers)
161 {
162 GstElement *audioresample;
163 GstBuffer *inbuffer, *outbuffer;
164 GstCaps *caps;
165 guint64 offset = 0;
166 int i, j;
167 GstMapInfo map;
168 gint16 *p;
169
170 audioresample =
171 setup_audioresample (2, 0x3, inrate, outrate, GST_AUDIO_NE (S16));
172 caps = gst_pad_get_current_caps (mysrcpad);
173 fail_unless (gst_caps_is_fixed (caps));
174
175 fail_unless (gst_element_set_state (audioresample,
176 GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
177 "could not set to playing");
178
179 for (j = 1; j <= numbuffers; ++j) {
180
181 inbuffer = gst_buffer_new_and_alloc (samples * 4);
182 GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (samples, inrate);
183 GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);
184 GST_BUFFER_OFFSET (inbuffer) = offset;
185 offset += samples;
186 GST_BUFFER_OFFSET_END (inbuffer) = offset;
187
188 gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
189 p = (gint16 *) map.data;
190
191 /* create a 16 bit signed ramp */
192 for (i = 0; i < samples; ++i) {
193 *p = -32767 + i * (65535 / samples);
194 ++p;
195 *p = -32767 + i * (65535 / samples);
196 ++p;
197 }
198 gst_buffer_unmap (inbuffer, &map);
199
200 /* pushing gives away my reference ... */
201 fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
202 /* ... but it ends up being collected on the global buffer list */
203 fail_unless_equals_int (g_list_length (buffers), j);
204 }
205
206 /* FIXME: we should make audioresample handle eos by flushing out the last
207 * samples, which will give us one more, small, buffer */
208 fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
209 ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
210
211 fail_unless_perfect_stream ();
212
213 /* cleanup */
214 gst_caps_unref (caps);
215 cleanup_audioresample (audioresample);
216 }
217
218
219 /* make sure that outgoing buffers are contiguous in timestamp/duration and
220 * offset/offsetend
221 */
GST_START_TEST(test_perfect_stream)222 GST_START_TEST (test_perfect_stream)
223 {
224 /* integral scalings */
225 test_perfect_stream_instance (48000, 24000, 500, 20);
226 test_perfect_stream_instance (48000, 12000, 500, 20);
227 test_perfect_stream_instance (12000, 24000, 500, 20);
228 test_perfect_stream_instance (12000, 48000, 500, 20);
229
230 /* non-integral scalings */
231 test_perfect_stream_instance (44100, 8000, 500, 20);
232 test_perfect_stream_instance (8000, 44100, 500, 20);
233
234 /* wacky scalings */
235 test_perfect_stream_instance (12345, 54321, 500, 20);
236 test_perfect_stream_instance (101, 99, 500, 20);
237 }
238
239 GST_END_TEST;
240
241 /* this tests that the output is a correct discontinuous stream
242 * if the input is; ie input drops in time come out the same way */
243 static void
test_discont_stream_instance(int inrate,int outrate,int samples,int numbuffers)244 test_discont_stream_instance (int inrate, int outrate, int samples,
245 int numbuffers)
246 {
247 GstElement *audioresample;
248 GstBuffer *inbuffer, *outbuffer;
249 GstCaps *caps;
250 GstClockTime ints;
251
252 int i, j;
253 GstMapInfo map;
254 gint16 *p;
255
256 GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
257 inrate, outrate, samples, numbuffers);
258
259 audioresample =
260 setup_audioresample (2, 3, inrate, outrate, GST_AUDIO_NE (S16));
261 caps = gst_pad_get_current_caps (mysrcpad);
262 fail_unless (gst_caps_is_fixed (caps));
263
264 fail_unless (gst_element_set_state (audioresample,
265 GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
266 "could not set to playing");
267
268 for (j = 1; j <= numbuffers; ++j) {
269
270 inbuffer = gst_buffer_new_and_alloc (samples * 4);
271 GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;
272 /* "drop" half the buffers */
273 ints = GST_BUFFER_DURATION (inbuffer) * 2 * (j - 1);
274 GST_BUFFER_TIMESTAMP (inbuffer) = ints;
275 GST_BUFFER_OFFSET (inbuffer) = (j - 1) * 2 * samples;
276 GST_BUFFER_OFFSET_END (inbuffer) = j * 2 * samples + samples;
277
278 gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
279 p = (gint16 *) map.data;
280 /* create a 16 bit signed ramp */
281 for (i = 0; i < samples; ++i) {
282 *p = -32767 + i * (65535 / samples);
283 ++p;
284 *p = -32767 + i * (65535 / samples);
285 ++p;
286 }
287 gst_buffer_unmap (inbuffer, &map);
288
289 GST_DEBUG ("Sending Buffer time:%" G_GUINT64_FORMAT " duration:%"
290 G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
291 G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (inbuffer),
292 GST_BUFFER_DURATION (inbuffer), GST_BUFFER_IS_DISCONT (inbuffer),
293 GST_BUFFER_OFFSET (inbuffer), GST_BUFFER_OFFSET_END (inbuffer));
294 /* pushing gives away my reference ... */
295 fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
296
297 /* check if the timestamp of the pushed buffer matches the incoming one */
298 outbuffer = g_list_nth_data (buffers, g_list_length (buffers) - 1);
299 fail_if (outbuffer == NULL);
300 fail_unless_equals_uint64 (ints, GST_BUFFER_TIMESTAMP (outbuffer));
301 GST_DEBUG ("Got Buffer time:%" G_GUINT64_FORMAT " duration:%"
302 G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
303 G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (outbuffer),
304 GST_BUFFER_DURATION (outbuffer), GST_BUFFER_IS_DISCONT (outbuffer),
305 GST_BUFFER_OFFSET (outbuffer), GST_BUFFER_OFFSET_END (outbuffer));
306 if (j > 1) {
307 fail_unless (GST_BUFFER_IS_DISCONT (outbuffer),
308 "expected discont for buffer #%d", j);
309 }
310 }
311
312 /* cleanup */
313 gst_caps_unref (caps);
314 cleanup_audioresample (audioresample);
315 }
316
GST_START_TEST(test_discont_stream)317 GST_START_TEST (test_discont_stream)
318 {
319 /* integral scalings */
320 test_discont_stream_instance (48000, 24000, 5000, 20);
321 test_discont_stream_instance (48000, 12000, 5000, 20);
322 test_discont_stream_instance (12000, 24000, 5000, 20);
323 test_discont_stream_instance (12000, 48000, 5000, 20);
324
325 /* non-integral scalings */
326 test_discont_stream_instance (44100, 8000, 5000, 20);
327 test_discont_stream_instance (8000, 44100, 5000, 20);
328
329 /* wacky scalings */
330 test_discont_stream_instance (12345, 54321, 5000, 20);
331 test_discont_stream_instance (101, 99, 5000, 20);
332 }
333
334 GST_END_TEST;
335
336
337
GST_START_TEST(test_reuse)338 GST_START_TEST (test_reuse)
339 {
340 GstElement *audioresample;
341 GstEvent *newseg;
342 GstBuffer *inbuffer;
343 GstCaps *caps;
344 GstSegment segment;
345
346 audioresample = setup_audioresample (1, 0, 9343, 48000, GST_AUDIO_NE (S16));
347 caps = gst_pad_get_current_caps (mysrcpad);
348 fail_unless (gst_caps_is_fixed (caps));
349
350 fail_unless (gst_element_set_state (audioresample,
351 GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
352 "could not set to playing");
353
354 gst_segment_init (&segment, GST_FORMAT_TIME);
355 newseg = gst_event_new_segment (&segment);
356 fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
357
358 inbuffer = gst_buffer_new_and_alloc (9343 * 4);
359 gst_buffer_memset (inbuffer, 0, 0, 9343 * 4);
360 GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
361 GST_BUFFER_TIMESTAMP (inbuffer) = 0;
362 GST_BUFFER_OFFSET (inbuffer) = 0;
363
364 /* pushing gives away my reference ... */
365 fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
366
367 /* ... but it ends up being collected on the global buffer list */
368 fail_unless_equals_int (g_list_length (buffers), 1);
369
370 /* now reset and try again ... */
371 fail_unless (gst_element_set_state (audioresample,
372 GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
373
374 fail_unless (gst_element_set_state (audioresample,
375 GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
376 "could not set to playing");
377
378 newseg = gst_event_new_segment (&segment);
379 fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
380
381 inbuffer = gst_buffer_new_and_alloc (9343 * 4);
382 gst_buffer_memset (inbuffer, 0, 0, 9343 * 4);
383 GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
384 GST_BUFFER_TIMESTAMP (inbuffer) = 0;
385 GST_BUFFER_OFFSET (inbuffer) = 0;
386
387 fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
388
389 /* ... it also ends up being collected on the global buffer list. If we
390 * now have more than 2 buffers, then audioresample probably didn't clean
391 * up its internal buffer properly and tried to push the remaining samples
392 * when it got the second NEWSEGMENT event */
393 fail_unless_equals_int (g_list_length (buffers), 2);
394
395 cleanup_audioresample (audioresample);
396 gst_caps_unref (caps);
397 }
398
399 GST_END_TEST;
400
GST_START_TEST(test_shutdown)401 GST_START_TEST (test_shutdown)
402 {
403 GstElement *pipeline, *src, *cf1, *ar, *cf2, *sink;
404 GstCaps *caps;
405 guint i;
406
407 /* create pipeline, force audioresample to actually resample */
408 pipeline = gst_pipeline_new (NULL);
409
410 src = gst_check_setup_element ("audiotestsrc");
411 cf1 = gst_check_setup_element ("capsfilter");
412 ar = gst_check_setup_element ("audioresample");
413 cf2 = gst_check_setup_element ("capsfilter");
414 g_object_set (cf2, "name", "capsfilter2", NULL);
415 sink = gst_check_setup_element ("fakesink");
416
417 caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, 11025, NULL);
418 g_object_set (cf1, "caps", caps, NULL);
419 gst_caps_unref (caps);
420
421 caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, 48000, NULL);
422 g_object_set (cf2, "caps", caps, NULL);
423 gst_caps_unref (caps);
424
425 /* don't want to sync against the clock, the more throughput the better */
426 g_object_set (src, "is-live", FALSE, NULL);
427 g_object_set (sink, "sync", FALSE, NULL);
428
429 gst_bin_add_many (GST_BIN (pipeline), src, cf1, ar, cf2, sink, NULL);
430 fail_if (!gst_element_link_many (src, cf1, ar, cf2, sink, NULL));
431
432 /* now, wait until pipeline is running and then shut it down again; repeat */
433 for (i = 0; i < 20; ++i) {
434 gst_element_set_state (pipeline, GST_STATE_PAUSED);
435 gst_element_get_state (pipeline, NULL, NULL, -1);
436 gst_element_set_state (pipeline, GST_STATE_PLAYING);
437 g_usleep (100);
438 gst_element_set_state (pipeline, GST_STATE_NULL);
439 }
440
441 gst_object_unref (pipeline);
442 }
443
444 GST_END_TEST;
445
446 #if 0
447 static GstFlowReturn
448 live_switch_alloc_only_48000 (GstPad * pad, guint64 offset,
449 guint size, GstCaps * caps, GstBuffer ** buf)
450 {
451 GstStructure *structure;
452 gint rate;
453 gint channels;
454 GstCaps *desired;
455
456 structure = gst_caps_get_structure (caps, 0);
457 fail_unless (gst_structure_get_int (structure, "rate", &rate));
458 fail_unless (gst_structure_get_int (structure, "channels", &channels));
459
460 if (rate < 48000)
461 return GST_FLOW_NOT_NEGOTIATED;
462
463 desired = gst_caps_copy (caps);
464 gst_caps_set_simple (desired, "rate", G_TYPE_INT, 48000, NULL);
465
466 *buf = gst_buffer_new_and_alloc (channels * 48000);
467 gst_buffer_set_caps (*buf, desired);
468 gst_caps_unref (desired);
469
470 return GST_FLOW_OK;
471 }
472
473 static GstCaps *
474 live_switch_get_sink_caps (GstPad * pad)
475 {
476 GstCaps *result;
477
478 result = gst_caps_make_writable (gst_pad_get_current_caps (pad));
479
480 gst_caps_set_simple (result,
481 "rate", GST_TYPE_INT_RANGE, 48000, G_MAXINT, NULL);
482
483 return result;
484 }
485 #endif
486
487 static void
live_switch_push(int rate,GstCaps * caps)488 live_switch_push (int rate, GstCaps * caps)
489 {
490 GstBuffer *inbuffer;
491 GstCaps *desired;
492 GList *l;
493
494 desired = gst_caps_copy (caps);
495 gst_caps_set_simple (desired, "rate", G_TYPE_INT, rate, NULL);
496 gst_pad_set_caps (mysrcpad, desired);
497
498 #if 0
499 fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad,
500 GST_BUFFER_OFFSET_NONE, rate * 4, desired, &inbuffer) == GST_FLOW_OK);
501 #endif
502 inbuffer = gst_buffer_new_and_alloc (rate * 4);
503 gst_buffer_memset (inbuffer, 0, 0, rate * 4);
504
505 GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
506 GST_BUFFER_TIMESTAMP (inbuffer) = 0;
507 GST_BUFFER_OFFSET (inbuffer) = 0;
508
509 /* pushing gives away my reference ... */
510 fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
511
512 /* ... but it ends up being collected on the global buffer list */
513 fail_unless_equals_int (g_list_length (buffers), 1);
514
515 for (l = buffers; l; l = l->next) {
516 GstBuffer *buffer = GST_BUFFER (l->data);
517
518 gst_buffer_unref (buffer);
519 }
520
521 g_list_free (buffers);
522 buffers = NULL;
523
524 gst_caps_unref (desired);
525 }
526
GST_START_TEST(test_live_switch)527 GST_START_TEST (test_live_switch)
528 {
529 GstElement *audioresample;
530 GstEvent *newseg;
531 GstCaps *caps;
532 GstSegment segment;
533
534 audioresample =
535 setup_audioresample (4, 0xf, 48000, 48000, GST_AUDIO_NE (S16));
536
537 /* Let the sinkpad act like something that can only handle things of
538 * rate 48000- and can only allocate buffers for that rate, but if someone
539 * tries to get a buffer with a rate higher then 48000 tries to renegotiate
540 * */
541 //gst_pad_set_bufferalloc_function (mysinkpad, live_switch_alloc_only_48000);
542 //gst_pad_set_getcaps_function (mysinkpad, live_switch_get_sink_caps);
543
544 gst_pad_use_fixed_caps (mysrcpad);
545
546 caps = gst_pad_get_current_caps (mysrcpad);
547 fail_unless (gst_caps_is_fixed (caps));
548
549 fail_unless (gst_element_set_state (audioresample,
550 GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
551 "could not set to playing");
552
553 gst_segment_init (&segment, GST_FORMAT_TIME);
554 newseg = gst_event_new_segment (&segment);
555 fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
556
557 /* downstream can provide the requested rate, a buffer alloc will be passed
558 * on */
559 live_switch_push (48000, caps);
560
561 /* Downstream can never accept this rate, buffer alloc isn't passed on */
562 live_switch_push (40000, caps);
563
564 /* Downstream can provide the requested rate but will re-negotiate */
565 live_switch_push (50000, caps);
566
567 cleanup_audioresample (audioresample);
568 gst_caps_unref (caps);
569 }
570
571 GST_END_TEST;
572
573 #ifndef GST_DISABLE_PARSE
574
575 static GMainLoop *loop;
576 static gint messages = 0;
577
578 static void
element_message_cb(GstBus * bus,GstMessage * message,gpointer user_data)579 element_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
580 {
581 gchar *s;
582
583 s = gst_structure_to_string (gst_message_get_structure (message));
584 GST_DEBUG ("Received message: %s", s);
585 g_free (s);
586
587 messages++;
588 }
589
590 static void
eos_message_cb(GstBus * bus,GstMessage * message,gpointer user_data)591 eos_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
592 {
593 GST_DEBUG ("Received eos");
594 g_main_loop_quit (loop);
595 }
596
597 static void
test_pipeline(const gchar * format,gint inrate,gint outrate,gint quality)598 test_pipeline (const gchar * format, gint inrate, gint outrate, gint quality)
599 {
600 GstElement *pipeline;
601 GstBus *bus;
602 GError *error = NULL;
603 gchar *pipe_str;
604
605 pipe_str =
606 g_strdup_printf
607 ("audiotestsrc num-buffers=10 ! audioconvert ! audio/x-raw,format=%s,rate=%d,channels=2 ! audioresample quality=%d ! audio/x-raw,format=%s,rate=%d ! identity check-imperfect-timestamp=TRUE ! fakesink",
608 format, inrate, quality, format, outrate);
609
610 pipeline = gst_parse_launch (pipe_str, &error);
611 fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
612 error ? error->message : "(invalid error)");
613 g_free (pipe_str);
614
615 bus = gst_element_get_bus (pipeline);
616 fail_if (bus == NULL);
617 gst_bus_add_signal_watch (bus);
618 g_signal_connect (bus, "message::element", (GCallback) element_message_cb,
619 NULL);
620 g_signal_connect (bus, "message::eos", (GCallback) eos_message_cb, NULL);
621
622 gst_element_set_state (pipeline, GST_STATE_PLAYING);
623
624 /* run until we receive EOS */
625 loop = g_main_loop_new (NULL, FALSE);
626
627 g_main_loop_run (loop);
628
629 g_main_loop_unref (loop);
630 loop = NULL;
631
632 gst_element_set_state (pipeline, GST_STATE_NULL);
633
634 gst_bus_remove_signal_watch (bus);
635 gst_object_unref (bus);
636
637 fail_if (messages > 0, "Received imperfect timestamp messages");
638 gst_object_unref (pipeline);
639 }
640
GST_START_TEST(test_pipelines)641 GST_START_TEST (test_pipelines)
642 {
643 gint quality;
644
645 /* Test qualities 0, 5 and 10 */
646 for (quality = 0; quality < 11; quality += 5) {
647 GST_DEBUG ("Checking with quality %d", quality);
648
649 test_pipeline ("S8", 44100, 48000, quality);
650 test_pipeline ("S8", 48000, 44100, quality);
651
652 test_pipeline (GST_AUDIO_NE (S16), 44100, 48000, quality);
653 test_pipeline (GST_AUDIO_NE (S16), 48000, 44100, quality);
654
655 test_pipeline (GST_AUDIO_NE (S24), 44100, 48000, quality);
656 test_pipeline (GST_AUDIO_NE (S24), 48000, 44100, quality);
657
658 test_pipeline (GST_AUDIO_NE (S32), 44100, 48000, quality);
659 test_pipeline (GST_AUDIO_NE (S32), 48000, 44100, quality);
660
661 test_pipeline (GST_AUDIO_NE (F32), 44100, 48000, quality);
662 test_pipeline (GST_AUDIO_NE (F32), 48000, 44100, quality);
663
664 test_pipeline (GST_AUDIO_NE (F64), 44100, 48000, quality);
665 test_pipeline (GST_AUDIO_NE (F64), 48000, 44100, quality);
666 }
667 }
668
669 GST_END_TEST;
670
GST_START_TEST(test_preference_passthrough)671 GST_START_TEST (test_preference_passthrough)
672 {
673 GstStateChangeReturn ret;
674 GstElement *pipeline, *src;
675 GstStructure *s;
676 GstMessage *msg;
677 GstCaps *caps;
678 GstPad *pad;
679 GstBus *bus;
680 GError *error = NULL;
681 gint rate = 0;
682
683 pipeline = gst_parse_launch ("audiotestsrc num-buffers=1 name=src ! "
684 "audioresample ! audio/x-raw,format=" GST_AUDIO_NE (S16) ",channels=1,"
685 "rate=8000 ! fakesink can-activate-pull=false", &error);
686 fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
687 error ? error->message : "(invalid error)");
688
689 ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
690 fail_unless_equals_int (ret, GST_STATE_CHANGE_ASYNC);
691
692 /* run until we receive EOS */
693 bus = gst_element_get_bus (pipeline);
694 fail_if (bus == NULL);
695 msg = gst_bus_timed_pop_filtered (bus, -1, GST_MESSAGE_EOS);
696 gst_message_unref (msg);
697 gst_object_unref (bus);
698
699 src = gst_bin_get_by_name (GST_BIN (pipeline), "src");
700 fail_unless (src != NULL);
701 pad = gst_element_get_static_pad (src, "src");
702 fail_unless (pad != NULL);
703 caps = gst_pad_get_current_caps (pad);
704 GST_LOG ("current audiotestsrc caps: %" GST_PTR_FORMAT, caps);
705 fail_unless (caps != NULL);
706 s = gst_caps_get_structure (caps, 0);
707 fail_unless (gst_structure_get_int (s, "rate", &rate));
708 /* there's no need to resample, audiotestsrc supports any rate, so make
709 * sure audioresample provided upstream with the right caps to negotiate
710 * this correctly */
711 fail_unless_equals_int (rate, 8000);
712 gst_caps_unref (caps);
713 gst_object_unref (pad);
714 gst_object_unref (src);
715
716 gst_element_set_state (pipeline, GST_STATE_NULL);
717 gst_object_unref (pipeline);
718 }
719
720 GST_END_TEST;
721
722 #endif
723
724 static void
_message_cb(GstBus * bus,GstMessage * message,gpointer user_data)725 _message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
726 {
727 GMainLoop *loop = user_data;
728
729 switch (GST_MESSAGE_TYPE (message)) {
730 case GST_MESSAGE_ERROR:
731 case GST_MESSAGE_WARNING:
732 g_assert_not_reached ();
733 break;
734 case GST_MESSAGE_EOS:
735 g_main_loop_quit (loop);
736 break;
737 default:
738 break;
739 }
740 }
741
742 typedef struct
743 {
744 guint64 latency;
745 GstClockTime in_ts;
746
747 GstClockTime next_out_ts;
748 guint64 next_out_off;
749
750 guint64 in_buffer_count, out_buffer_count;
751 } TimestampDriftCtx;
752
753 static void
fakesink_handoff_cb(GstElement * object,GstBuffer * buffer,GstPad * pad,gpointer user_data)754 fakesink_handoff_cb (GstElement * object, GstBuffer * buffer, GstPad * pad,
755 gpointer user_data)
756 {
757 TimestampDriftCtx *ctx = user_data;
758
759 ctx->out_buffer_count++;
760 if (ctx->latency == GST_CLOCK_TIME_NONE) {
761 ctx->latency = 1000 - gst_buffer_get_size (buffer) / 8;
762 }
763
764 /* Check if we have a perfectly timestamped stream */
765 if (ctx->next_out_ts != GST_CLOCK_TIME_NONE)
766 fail_unless (ctx->next_out_ts == GST_BUFFER_TIMESTAMP (buffer),
767 "expected timestamp %" GST_TIME_FORMAT " got timestamp %"
768 GST_TIME_FORMAT, GST_TIME_ARGS (ctx->next_out_ts),
769 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
770
771 /* Check if we have a perfectly offsetted stream */
772 fail_unless (GST_BUFFER_OFFSET_END (buffer) ==
773 GST_BUFFER_OFFSET (buffer) + gst_buffer_get_size (buffer) / 8,
774 "expected offset end %" G_GUINT64_FORMAT " got offset end %"
775 G_GUINT64_FORMAT,
776 GST_BUFFER_OFFSET (buffer) + gst_buffer_get_size (buffer) / 8,
777 GST_BUFFER_OFFSET_END (buffer));
778 if (ctx->next_out_off != GST_BUFFER_OFFSET_NONE) {
779 fail_unless (GST_BUFFER_OFFSET (buffer) == ctx->next_out_off,
780 "expected offset %" G_GUINT64_FORMAT " got offset %" G_GUINT64_FORMAT,
781 ctx->next_out_off, GST_BUFFER_OFFSET (buffer));
782 }
783
784 if (ctx->in_buffer_count != ctx->out_buffer_count) {
785 GST_INFO ("timestamp %" GST_TIME_FORMAT,
786 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
787 }
788
789 if (ctx->in_ts != GST_CLOCK_TIME_NONE && ctx->in_buffer_count > 1
790 && ctx->in_buffer_count == ctx->out_buffer_count) {
791 fail_unless (GST_BUFFER_TIMESTAMP (buffer) ==
792 ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND,
793 4096),
794 "expected output timestamp %" GST_TIME_FORMAT " (%" G_GUINT64_FORMAT
795 ") got output timestamp %" GST_TIME_FORMAT " (%" G_GUINT64_FORMAT ")",
796 GST_TIME_ARGS (ctx->in_ts - gst_util_uint64_scale_round (ctx->latency,
797 GST_SECOND, 4096)),
798 ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND,
799 4096), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
800 GST_BUFFER_TIMESTAMP (buffer));
801 }
802
803 ctx->next_out_ts =
804 GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer);
805 ctx->next_out_off = GST_BUFFER_OFFSET_END (buffer);
806 }
807
808 static void
identity_handoff_cb(GstElement * object,GstBuffer * buffer,gpointer user_data)809 identity_handoff_cb (GstElement * object, GstBuffer * buffer,
810 gpointer user_data)
811 {
812 TimestampDriftCtx *ctx = user_data;
813
814 ctx->in_ts = GST_BUFFER_TIMESTAMP (buffer);
815 ctx->in_buffer_count++;
816 }
817
GST_START_TEST(test_timestamp_drift)818 GST_START_TEST (test_timestamp_drift)
819 {
820 TimestampDriftCtx ctx =
821 { GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE,
822 GST_BUFFER_OFFSET_NONE, 0, 0
823 };
824 GstElement *pipeline;
825 GstElement *audiotestsrc, *capsfilter1, *identity, *audioresample,
826 *capsfilter2, *fakesink;
827 GstBus *bus;
828 GMainLoop *loop;
829 GstCaps *caps;
830
831 pipeline = gst_pipeline_new ("pipeline");
832 fail_unless (pipeline != NULL);
833
834 audiotestsrc = gst_element_factory_make ("audiotestsrc", "src");
835 fail_unless (audiotestsrc != NULL);
836 g_object_set (G_OBJECT (audiotestsrc), "num-buffers", 10000,
837 "samplesperbuffer", 4000, NULL);
838
839 capsfilter1 = gst_element_factory_make ("capsfilter", "capsfilter1");
840 fail_unless (capsfilter1 != NULL);
841 caps = gst_caps_from_string ("audio/x-raw, format=" GST_AUDIO_NE (F64)
842 ", channels=1, rate=16384");
843 g_object_set (G_OBJECT (capsfilter1), "caps", caps, NULL);
844 gst_caps_unref (caps);
845
846 identity = gst_element_factory_make ("identity", "identity");
847 fail_unless (identity != NULL);
848 g_object_set (G_OBJECT (identity), "sync", FALSE, "signal-handoffs", TRUE,
849 NULL);
850 g_signal_connect (identity, "handoff", (GCallback) identity_handoff_cb, &ctx);
851
852 audioresample = gst_element_factory_make ("audioresample", "resample");
853 fail_unless (audioresample != NULL);
854 capsfilter2 = gst_element_factory_make ("capsfilter", "capsfilter2");
855 fail_unless (capsfilter2 != NULL);
856 caps = gst_caps_from_string ("audio/x-raw, format=" GST_AUDIO_NE (F64)
857 ", channels=1, rate=4096");
858 g_object_set (G_OBJECT (capsfilter2), "caps", caps, NULL);
859 gst_caps_unref (caps);
860
861 fakesink = gst_element_factory_make ("fakesink", "sink");
862 fail_unless (fakesink != NULL);
863 g_object_set (G_OBJECT (fakesink), "sync", FALSE, "async", FALSE,
864 "signal-handoffs", TRUE, NULL);
865 g_signal_connect (fakesink, "handoff", (GCallback) fakesink_handoff_cb, &ctx);
866
867
868 gst_bin_add_many (GST_BIN (pipeline), audiotestsrc, capsfilter1, identity,
869 audioresample, capsfilter2, fakesink, NULL);
870 fail_unless (gst_element_link_many (audiotestsrc, capsfilter1, identity,
871 audioresample, capsfilter2, fakesink, NULL));
872
873 loop = g_main_loop_new (NULL, FALSE);
874
875 bus = gst_element_get_bus (pipeline);
876 gst_bus_add_signal_watch (bus);
877 g_signal_connect (bus, "message", (GCallback) _message_cb, loop);
878
879 fail_unless (gst_element_set_state (pipeline,
880 GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
881 g_main_loop_run (loop);
882
883 fail_unless (gst_element_set_state (pipeline,
884 GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
885 g_main_loop_unref (loop);
886 gst_bus_remove_signal_watch (bus);
887 gst_object_unref (bus);
888
889 gst_object_unref (pipeline);
890
891 } GST_END_TEST;
892
893 #define FFT_HELPERS(type,ffttag,ffttag2,scale); \
894 static gdouble magnitude##ffttag (const GstFFT##ffttag##Complex *c) \
895 { \
896 gdouble mag = (gdouble) c->r * (gdouble) c->r; \
897 mag += (gdouble) c->i * (gdouble) c->i; \
898 mag /= scale * scale; \
899 mag = 10.0 * log10 (mag); \
900 return mag; \
901 } \
902 static gdouble find_main_frequency_spot_##ffttag (const GstFFT##ffttag##Complex *v, \
903 int elements) \
904 { \
905 int i; \
906 gdouble maxmag = -9999; \
907 int maxidx = 0; \
908 for (i=0; i<elements; ++i) { \
909 gdouble mag = magnitude##ffttag (v+i); \
910 if (mag > maxmag) { \
911 maxmag = mag; \
912 maxidx = i; \
913 } \
914 } \
915 return maxidx / (gdouble) elements; \
916 } \
917 static gboolean is_zero_except_##ffttag (const GstFFT##ffttag##Complex *v, int elements, \
918 gdouble spot) \
919 { \
920 int i; \
921 for (i=0; i<elements; ++i) { \
922 gdouble pos = i / (gdouble) elements; \
923 gdouble mag = magnitude##ffttag (v+i); \
924 if (fabs (pos - spot) > 0.01) { \
925 if (mag > -55.0) { \
926 return FALSE; \
927 } \
928 } \
929 } \
930 return TRUE; \
931 } \
932 static void compare_ffts_##ffttag (GstBuffer *inbuffer, GstBuffer *outbuffer) \
933 { \
934 GstMapInfo inmap, outmap; \
935 int insamples, outsamples; \
936 gdouble inspot, outspot; \
937 GstFFT##ffttag *inctx, *outctx; \
938 GstFFT##ffttag##Complex *in, *out; \
939 \
940 gst_buffer_map (inbuffer, &inmap, GST_MAP_READ); \
941 gst_buffer_map (outbuffer, &outmap, GST_MAP_READWRITE); \
942 \
943 insamples = inmap.size / sizeof(type) & ~1; \
944 outsamples = outmap.size / sizeof(type) & ~1; \
945 inctx = gst_fft_##ffttag2##_new (insamples, FALSE); \
946 outctx = gst_fft_##ffttag2##_new (outsamples, FALSE); \
947 in = g_new (GstFFT##ffttag##Complex, insamples / 2 + 1); \
948 out = g_new (GstFFT##ffttag##Complex, outsamples / 2 + 1); \
949 \
950 gst_fft_##ffttag2##_window (inctx, (type*)inmap.data, \
951 GST_FFT_WINDOW_HAMMING); \
952 gst_fft_##ffttag2##_fft (inctx, (type*)inmap.data, in); \
953 gst_fft_##ffttag2##_window (outctx, (type*)outmap.data, \
954 GST_FFT_WINDOW_HAMMING); \
955 gst_fft_##ffttag2##_fft (outctx, (type*)outmap.data, out); \
956 \
957 inspot = find_main_frequency_spot_##ffttag (in, insamples / 2 + 1); \
958 outspot = find_main_frequency_spot_##ffttag (out, outsamples / 2 + 1); \
959 GST_LOG ("Spots are %.3f and %.3f", inspot, outspot); \
960 fail_unless (fabs (outspot - inspot) < 0.05); \
961 fail_unless (is_zero_except_##ffttag (in, insamples / 2 + 1, inspot)); \
962 fail_unless (is_zero_except_##ffttag (out, outsamples / 2 + 1, outspot)); \
963 \
964 gst_buffer_unmap (inbuffer, &inmap); \
965 gst_buffer_unmap (outbuffer, &outmap); \
966 \
967 gst_fft_##ffttag2##_free (inctx); \
968 gst_fft_##ffttag2##_free (outctx); \
969 g_free (in); \
970 g_free (out); \
971 }
972 FFT_HELPERS (float, F32, f32, 2048.0f);
973 FFT_HELPERS (double, F64, f64, 2048.0);
974 FFT_HELPERS (gint16, S16, s16, 32767.0);
975 FFT_HELPERS (gint32, S32, s32, 2147483647.0);
976
977 #define FILL_BUFFER(type, desc, value); \
978 static void init_##type##_##desc (GstBuffer *buffer) \
979 { \
980 GstMapInfo map; \
981 type *ptr; \
982 int i, nsamples; \
983 gst_buffer_map (buffer, &map, GST_MAP_WRITE); \
984 ptr = (type *)map.data; \
985 nsamples = map.size / sizeof (type); \
986 for (i = 0; i < nsamples; ++i) { \
987 *ptr++ = value; \
988 } \
989 gst_buffer_unmap (buffer, &map); \
990 }
991
992 FILL_BUFFER (float, silence, 0.0f);
993 FILL_BUFFER (double, silence, 0.0);
994 FILL_BUFFER (gint16, silence, 0);
995 FILL_BUFFER (gint32, silence, 0);
996 FILL_BUFFER (float, sine, sinf (i * 0.01f));
997 FILL_BUFFER (float, sine2, sinf (i * 1.8f));
998 FILL_BUFFER (double, sine, sin (i * 0.01));
999 FILL_BUFFER (double, sine2, sin (i * 1.8));
1000 FILL_BUFFER (gint16, sine, (gint16) (32767 * sinf (i * 0.01f)));
1001 FILL_BUFFER (gint16, sine2, (gint16) (32767 * sinf (i * 1.8f)));
1002 FILL_BUFFER (gint32, sine, (gint32) (2147483647 * sinf (i * 0.01f)));
1003 FILL_BUFFER (gint32, sine2, (gint32) (2147483647 * sinf (i * 1.8f)));
1004
1005 static void
run_fft_pipeline(int inrate,int outrate,int quality,int width,const gchar * format,void (* init)(GstBuffer *),void (* compare_ffts)(GstBuffer *,GstBuffer *))1006 run_fft_pipeline (int inrate, int outrate, int quality, int width,
1007 const gchar * format, void (*init) (GstBuffer *),
1008 void (*compare_ffts) (GstBuffer *, GstBuffer *))
1009 {
1010 GstElement *audioresample;
1011 GstBuffer *inbuffer, *outbuffer;
1012 const int nsamples = 2048;
1013
1014 audioresample = setup_audioresample (1, 0, inrate, outrate, format);
1015 fail_unless (audioresample != NULL);
1016 g_object_set (audioresample, "quality", quality, NULL);
1017
1018 fail_unless (gst_element_set_state (audioresample,
1019 GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
1020 "could not set to playing");
1021
1022 inbuffer = gst_buffer_new_and_alloc (nsamples * width / 8);
1023 GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (nsamples, inrate);
1024 GST_BUFFER_TIMESTAMP (inbuffer) = 0;
1025
1026 (*init) (inbuffer);
1027
1028 gst_buffer_ref (inbuffer);
1029 /* pushing gives away my reference ... */
1030 fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
1031 /* ... but it ends up being collected on the global buffer list */
1032 fail_unless_equals_int (g_list_length (buffers), 1);
1033 /* retrieve out buffer */
1034 fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
1035
1036 fail_unless (gst_element_set_state (audioresample,
1037 GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to null");
1038
1039 if (inbuffer == outbuffer)
1040 gst_buffer_unref (inbuffer);
1041
1042 (*compare_ffts) (inbuffer, outbuffer);
1043
1044 /* cleanup */
1045 cleanup_audioresample (audioresample);
1046 }
1047
GST_START_TEST(test_fft)1048 GST_START_TEST (test_fft)
1049 {
1050 int quality;
1051 size_t f0, f1;
1052 static const int frequencies[] =
1053 { 8000, 16000, 44100, 48000, 128000, 12345, 54321 };
1054
1055 /* audioresample uses a mixed float/double code path for floats with quality>8, make sure we test it */
1056 for (quality = 0; quality <= 10; quality += 5) {
1057 for (f0 = 0; f0 < G_N_ELEMENTS (frequencies); ++f0) {
1058 for (f1 = 0; f1 < G_N_ELEMENTS (frequencies); ++f1) {
1059 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
1060 GST_AUDIO_NE (F32), &init_float_silence, &compare_ffts_F32);
1061 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
1062 GST_AUDIO_NE (F32), &init_float_sine, &compare_ffts_F32);
1063 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
1064 GST_AUDIO_NE (F32), &init_float_sine2, &compare_ffts_F32);
1065 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64,
1066 GST_AUDIO_NE (F64), &init_double_silence, &compare_ffts_F64);
1067 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64,
1068 GST_AUDIO_NE (F64), &init_double_sine, &compare_ffts_F64);
1069 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64,
1070 GST_AUDIO_NE (F64), &init_double_sine2, &compare_ffts_F64);
1071 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16,
1072 GST_AUDIO_NE (S16), &init_gint16_silence, &compare_ffts_S16);
1073 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16,
1074 GST_AUDIO_NE (S16), &init_gint16_sine, &compare_ffts_S16);
1075 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16,
1076 GST_AUDIO_NE (S16), &init_gint16_sine2, &compare_ffts_S16);
1077 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
1078 GST_AUDIO_NE (S32), &init_gint32_silence, &compare_ffts_S32);
1079 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
1080 GST_AUDIO_NE (S32), &init_gint32_sine, &compare_ffts_S32);
1081 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
1082 GST_AUDIO_NE (S32), &init_gint32_sine2, &compare_ffts_S32);
1083 }
1084 }
1085 }
1086 }
1087
1088 GST_END_TEST;
1089
1090 static Suite *
audioresample_suite(void)1091 audioresample_suite (void)
1092 {
1093 Suite *s = suite_create ("audioresample");
1094 TCase *tc_chain = tcase_create ("general");
1095
1096 suite_add_tcase (s, tc_chain);
1097 tcase_add_test (tc_chain, test_perfect_stream);
1098 tcase_add_test (tc_chain, test_discont_stream);
1099 tcase_add_test (tc_chain, test_reuse);
1100 tcase_add_test (tc_chain, test_shutdown);
1101 tcase_add_test (tc_chain, test_live_switch);
1102 tcase_add_test (tc_chain, test_timestamp_drift);
1103 tcase_add_test (tc_chain, test_fft);
1104
1105 #ifndef GST_DISABLE_PARSE
1106 tcase_set_timeout (tc_chain, 360);
1107 tcase_add_test (tc_chain, test_pipelines);
1108 tcase_add_test (tc_chain, test_preference_passthrough);
1109 #endif
1110
1111 return s;
1112 }
1113
1114 GST_CHECK_MAIN (audioresample);
1115