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 static void
live_switch_push(gint pts,gint rate,GstCaps * caps)447 live_switch_push (gint pts, gint rate, GstCaps * caps)
448 {
449 GstBuffer *inbuffer;
450 GstCaps *desired;
451
452 desired = gst_caps_copy (caps);
453 gst_caps_set_simple (desired, "rate", G_TYPE_INT, rate, NULL);
454 gst_pad_set_caps (mysrcpad, desired);
455
456 inbuffer = gst_buffer_new_and_alloc (rate * 4 * 2);
457 gst_buffer_memset (inbuffer, 0, 0, rate * 4 * 2);
458
459 GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
460 GST_BUFFER_TIMESTAMP (inbuffer) = pts * GST_SECOND;
461 GST_BUFFER_OFFSET (inbuffer) = 0;
462 GST_BUFFER_OFFSET_END (inbuffer) = rate - 1;
463
464 /* pushing gives away my reference ... */
465 fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
466
467 /* ... but it ends up being collected on the global buffer list */
468
469 gst_caps_unref (desired);
470 }
471
472 #if !GLIB_CHECK_VERSION(2,58,0)
473 #define G_APPROX_VALUE(a, b, epsilon) \
474 (((a) > (b) ? (a) - (b) : (b) - (a)) < (epsilon))
475 #endif
476
GST_START_TEST(test_live_switch)477 GST_START_TEST (test_live_switch)
478 {
479 GstElement *audioresample;
480 GstEvent *newseg;
481 GstCaps *caps;
482 GstSegment segment;
483 GList *l;
484 guint i;
485
486 audioresample =
487 setup_audioresample (4, 0xf, 48000, 48000, GST_AUDIO_NE (S16));
488
489 caps = gst_pad_get_current_caps (mysrcpad);
490 fail_unless (gst_caps_is_fixed (caps));
491
492 fail_unless (gst_element_set_state (audioresample,
493 GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
494 "could not set to playing");
495
496 gst_segment_init (&segment, GST_FORMAT_TIME);
497 newseg = gst_event_new_segment (&segment);
498 fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
499
500 /* downstream can accept the requested rate */
501 live_switch_push (0, 48000, caps);
502
503 /* buffer is directly passed through */
504 fail_unless_equals_int (g_list_length (buffers), 1);
505
506 /* Downstream can never accept this rate */
507 live_switch_push (1, 40000, caps);
508
509 /* one additional buffer is provided with the new sample rate */
510 fail_unless_equals_int (g_list_length (buffers), 2);
511
512 /* Downstream can never accept this rate */
513 live_switch_push (2, 50000, caps);
514
515 /* two additional buffers are provided. One is the drained remainder of
516 * the previous sample rate, the second is the buffer with the new sample
517 * rate */
518 fail_unless_equals_int (g_list_length (buffers), 4);
519
520 /* Send EOS to drain the remaining samples */
521 fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
522 fail_unless_equals_int (g_list_length (buffers), 5);
523
524 /* Now test that each buffer has the expected samples. We simply check this
525 * by checking whether the timestamps, durations and sizes are matching */
526 for (l = buffers, i = 0; l; l = l->next, i++) {
527 GstBuffer *buffer = GST_BUFFER (l->data);
528
529 switch (i) {
530 case 0:
531 fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 0 * GST_SECOND);
532 fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer),
533 1 * GST_SECOND);
534 fail_unless_equals_int (gst_buffer_get_size (buffer), 48000 * 4 * 2);
535 break;
536 case 1:
537 fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 1 * GST_SECOND);
538 fail_unless_equals_int (gst_buffer_get_size (buffer), 47961 * 4 * 2);
539 break;
540 case 2:
541 fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) +
542 GST_BUFFER_DURATION (buffer), 2 * GST_SECOND,
543 GST_SECOND / 48000 + 1));
544 fail_unless_equals_int (gst_buffer_get_size (buffer), 38 * 4 * 2);
545 break;
546 case 3:
547 fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 2 * GST_SECOND);
548 fail_unless_equals_int (gst_buffer_get_size (buffer), 47969 * 4 * 2);
549 break;
550 case 4:
551 fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) +
552 GST_BUFFER_DURATION (buffer), 3 * GST_SECOND,
553 GST_SECOND / 48000 + 1));
554 fail_unless_equals_int (gst_buffer_get_size (buffer), 30 * 4 * 2);
555 break;
556 default:
557 g_assert_not_reached ();
558 break;
559 }
560
561 gst_buffer_unref (buffer);
562 }
563
564 g_list_free (buffers);
565 buffers = NULL;
566
567 cleanup_audioresample (audioresample);
568 gst_caps_unref (caps);
569 }
570
571 GST_END_TEST;
572
573 static gint current_rate = 0;
574
575 static gboolean
live_switch_sink_query(GstPad * pad,GstObject * parent,GstQuery * query)576 live_switch_sink_query (GstPad * pad, GstObject * parent, GstQuery * query)
577 {
578 switch (GST_QUERY_TYPE (query)) {
579 case GST_QUERY_ACCEPT_CAPS:{
580 GstCaps *acceptable_caps;
581 GstCaps *caps;
582
583 acceptable_caps = gst_pad_get_current_caps (mysrcpad);
584 acceptable_caps = gst_caps_make_writable (acceptable_caps);
585 gst_caps_set_simple (acceptable_caps, "rate", G_TYPE_INT, current_rate,
586 NULL);
587
588 gst_query_parse_accept_caps (query, &caps);
589
590 gst_query_set_accept_caps_result (query, gst_caps_can_intersect (caps,
591 acceptable_caps));
592
593 gst_caps_unref (acceptable_caps);
594
595 return TRUE;
596 }
597 case GST_QUERY_CAPS:{
598 GstCaps *acceptable_caps;
599 GstCaps *filter;
600 GstCaps *caps;
601
602 acceptable_caps = gst_pad_get_current_caps (mysrcpad);
603 acceptable_caps = gst_caps_make_writable (acceptable_caps);
604 gst_caps_set_simple (acceptable_caps, "rate", G_TYPE_INT, current_rate,
605 NULL);
606
607 gst_query_parse_caps (query, &filter);
608
609 if (filter)
610 caps =
611 gst_caps_intersect_full (filter, acceptable_caps,
612 GST_CAPS_INTERSECT_FIRST);
613 else
614 caps = gst_caps_ref (acceptable_caps);
615
616 gst_query_set_caps_result (query, caps);
617
618 gst_caps_unref (caps);
619 gst_caps_unref (acceptable_caps);
620
621 return TRUE;
622 }
623 default:
624 return gst_pad_query_default (pad, parent, query);
625 }
626 }
627
628 static void
live_switch_push_downstream(gint pts,gint rate)629 live_switch_push_downstream (gint pts, gint rate)
630 {
631 GstBuffer *inbuffer;
632
633 current_rate = rate;
634 gst_pad_push_event (mysinkpad, gst_event_new_reconfigure ());
635
636 inbuffer = gst_buffer_new_and_alloc (48000 * 4 * 2);
637 gst_buffer_memset (inbuffer, 0, 0, 48000 * 4 * 2);
638
639 GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
640 GST_BUFFER_TIMESTAMP (inbuffer) = pts * GST_SECOND;
641 GST_BUFFER_OFFSET (inbuffer) = 0;
642 GST_BUFFER_OFFSET_END (inbuffer) = 47999;
643
644 /* pushing gives away my reference ... */
645 fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
646
647 /* ... but it ends up being collected on the global buffer list */
648 }
649
GST_START_TEST(test_live_switch_downstream)650 GST_START_TEST (test_live_switch_downstream)
651 {
652 GstElement *audioresample;
653 GstEvent *newseg;
654 GstCaps *caps;
655 GstSegment segment;
656 GList *l;
657 guint i;
658
659 audioresample =
660 setup_audioresample (4, 0xf, 48000, 48000, GST_AUDIO_NE (S16));
661
662 gst_pad_set_query_function (mysinkpad, live_switch_sink_query);
663
664 caps = gst_pad_get_current_caps (mysrcpad);
665 fail_unless (gst_caps_is_fixed (caps));
666
667 fail_unless (gst_element_set_state (audioresample,
668 GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
669 "could not set to playing");
670
671 gst_segment_init (&segment, GST_FORMAT_TIME);
672 newseg = gst_event_new_segment (&segment);
673 fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
674
675 /* buffer is directly passed through */
676 live_switch_push_downstream (0, 48000);
677 fail_unless_equals_int (g_list_length (buffers), 1);
678
679 /* Reconfigure downstream to 40000 Hz */
680 live_switch_push_downstream (1, 40000);
681
682 /* one additional buffer is provided with the new sample rate */
683 fail_unless_equals_int (g_list_length (buffers), 2);
684
685 /* Reconfigure downstream to 50000 Hz */
686 live_switch_push_downstream (2, 50000);
687
688 /* two additional buffers are provided. One is the drained remainder of
689 * the previous sample rate, the second is the buffer with the new sample
690 * rate */
691 fail_unless_equals_int (g_list_length (buffers), 4);
692
693 /* Send EOS to drain the remaining samples */
694 fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
695 fail_unless_equals_int (g_list_length (buffers), 5);
696
697 /* Now test that each buffer has the expected samples. We simply check this
698 * by checking whether the timestamps, durations and sizes are matching */
699 for (l = buffers, i = 0; l; l = l->next, i++) {
700 GstBuffer *buffer = GST_BUFFER (l->data);
701
702 switch (i) {
703 case 0:
704 fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 0 * GST_SECOND);
705 fail_unless_equals_uint64 (GST_BUFFER_DURATION (buffer),
706 1 * GST_SECOND);
707 fail_unless_equals_int (gst_buffer_get_size (buffer), 48000 * 4 * 2);
708 break;
709 case 1:
710 fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 1 * GST_SECOND);
711 fail_unless_equals_int (gst_buffer_get_size (buffer), 39966 * 4 * 2);
712 break;
713 case 2:
714 fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) +
715 GST_BUFFER_DURATION (buffer), 2 * GST_SECOND,
716 GST_SECOND / 40000 + 1));
717 fail_unless_equals_int (gst_buffer_get_size (buffer), 34 * 4 * 2);
718 break;
719 case 3:
720 fail_unless_equals_uint64 (GST_BUFFER_PTS (buffer), 2 * GST_SECOND);
721 fail_unless_equals_int (gst_buffer_get_size (buffer), 49966 * 4 * 2);
722 break;
723 case 4:
724 fail_unless (G_APPROX_VALUE (GST_BUFFER_PTS (buffer) +
725 GST_BUFFER_DURATION (buffer), 3 * GST_SECOND,
726 GST_SECOND / 50000 + 1));
727 fail_unless_equals_int (gst_buffer_get_size (buffer), 33 * 4 * 2);
728 break;
729 default:
730 g_assert_not_reached ();
731 break;
732 }
733
734 gst_buffer_unref (buffer);
735 }
736
737 g_list_free (buffers);
738 buffers = NULL;
739
740 cleanup_audioresample (audioresample);
741 gst_caps_unref (caps);
742 }
743
744 GST_END_TEST;
745
746 #ifndef GST_DISABLE_PARSE
747
748 static GMainLoop *loop;
749 static gint messages = 0;
750
751 static void
element_message_cb(GstBus * bus,GstMessage * message,gpointer user_data)752 element_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
753 {
754 gchar *s;
755
756 s = gst_structure_to_string (gst_message_get_structure (message));
757 GST_DEBUG ("Received message: %s", s);
758 g_free (s);
759
760 messages++;
761 }
762
763 static void
eos_message_cb(GstBus * bus,GstMessage * message,gpointer user_data)764 eos_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
765 {
766 GST_DEBUG ("Received eos");
767 g_main_loop_quit (loop);
768 }
769
770 static void
test_pipeline(const gchar * format,gint inrate,gint outrate,gint quality)771 test_pipeline (const gchar * format, gint inrate, gint outrate, gint quality)
772 {
773 GstElement *pipeline;
774 GstBus *bus;
775 GError *error = NULL;
776 gchar *pipe_str;
777
778 pipe_str =
779 g_strdup_printf
780 ("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",
781 format, inrate, quality, format, outrate);
782
783 pipeline = gst_parse_launch (pipe_str, &error);
784 fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
785 error ? error->message : "(invalid error)");
786 g_free (pipe_str);
787
788 bus = gst_element_get_bus (pipeline);
789 fail_if (bus == NULL);
790 gst_bus_add_signal_watch (bus);
791 g_signal_connect (bus, "message::element", (GCallback) element_message_cb,
792 NULL);
793 g_signal_connect (bus, "message::eos", (GCallback) eos_message_cb, NULL);
794
795 gst_element_set_state (pipeline, GST_STATE_PLAYING);
796
797 /* run until we receive EOS */
798 loop = g_main_loop_new (NULL, FALSE);
799
800 g_main_loop_run (loop);
801
802 g_main_loop_unref (loop);
803 loop = NULL;
804
805 gst_element_set_state (pipeline, GST_STATE_NULL);
806
807 gst_bus_remove_signal_watch (bus);
808 gst_object_unref (bus);
809
810 fail_if (messages > 0, "Received imperfect timestamp messages");
811 gst_object_unref (pipeline);
812 }
813
GST_START_TEST(test_pipelines)814 GST_START_TEST (test_pipelines)
815 {
816 gint quality;
817
818 /* Test qualities 0, 5 and 10 */
819 for (quality = 0; quality < 11; quality += 5) {
820 GST_DEBUG ("Checking with quality %d", quality);
821
822 test_pipeline ("S8", 44100, 48000, quality);
823 test_pipeline ("S8", 48000, 44100, quality);
824
825 test_pipeline (GST_AUDIO_NE (S16), 44100, 48000, quality);
826 test_pipeline (GST_AUDIO_NE (S16), 48000, 44100, quality);
827
828 test_pipeline (GST_AUDIO_NE (S24), 44100, 48000, quality);
829 test_pipeline (GST_AUDIO_NE (S24), 48000, 44100, quality);
830
831 test_pipeline (GST_AUDIO_NE (S32), 44100, 48000, quality);
832 test_pipeline (GST_AUDIO_NE (S32), 48000, 44100, quality);
833
834 test_pipeline (GST_AUDIO_NE (F32), 44100, 48000, quality);
835 test_pipeline (GST_AUDIO_NE (F32), 48000, 44100, quality);
836
837 test_pipeline (GST_AUDIO_NE (F64), 44100, 48000, quality);
838 test_pipeline (GST_AUDIO_NE (F64), 48000, 44100, quality);
839 }
840 }
841
842 GST_END_TEST;
843
GST_START_TEST(test_preference_passthrough)844 GST_START_TEST (test_preference_passthrough)
845 {
846 GstStateChangeReturn ret;
847 GstElement *pipeline, *src;
848 GstStructure *s;
849 GstMessage *msg;
850 GstCaps *caps;
851 GstPad *pad;
852 GstBus *bus;
853 GError *error = NULL;
854 gint rate = 0;
855
856 pipeline = gst_parse_launch ("audiotestsrc num-buffers=1 name=src ! "
857 "audioresample ! audio/x-raw,format=" GST_AUDIO_NE (S16) ",channels=1,"
858 "rate=8000 ! fakesink can-activate-pull=false", &error);
859 fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
860 error ? error->message : "(invalid error)");
861
862 ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
863 fail_unless_equals_int (ret, GST_STATE_CHANGE_ASYNC);
864
865 /* run until we receive EOS */
866 bus = gst_element_get_bus (pipeline);
867 fail_if (bus == NULL);
868 msg = gst_bus_timed_pop_filtered (bus, -1, GST_MESSAGE_EOS);
869 gst_message_unref (msg);
870 gst_object_unref (bus);
871
872 src = gst_bin_get_by_name (GST_BIN (pipeline), "src");
873 fail_unless (src != NULL);
874 pad = gst_element_get_static_pad (src, "src");
875 fail_unless (pad != NULL);
876 caps = gst_pad_get_current_caps (pad);
877 GST_LOG ("current audiotestsrc caps: %" GST_PTR_FORMAT, caps);
878 fail_unless (caps != NULL);
879 s = gst_caps_get_structure (caps, 0);
880 fail_unless (gst_structure_get_int (s, "rate", &rate));
881 /* there's no need to resample, audiotestsrc supports any rate, so make
882 * sure audioresample provided upstream with the right caps to negotiate
883 * this correctly */
884 fail_unless_equals_int (rate, 8000);
885 gst_caps_unref (caps);
886 gst_object_unref (pad);
887 gst_object_unref (src);
888
889 gst_element_set_state (pipeline, GST_STATE_NULL);
890 gst_object_unref (pipeline);
891 }
892
893 GST_END_TEST;
894
895 #endif
896
897 static void
_message_cb(GstBus * bus,GstMessage * message,gpointer user_data)898 _message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
899 {
900 GMainLoop *loop = user_data;
901
902 switch (GST_MESSAGE_TYPE (message)) {
903 case GST_MESSAGE_ERROR:
904 case GST_MESSAGE_WARNING:
905 g_assert_not_reached ();
906 break;
907 case GST_MESSAGE_EOS:
908 g_main_loop_quit (loop);
909 break;
910 default:
911 break;
912 }
913 }
914
915 typedef struct
916 {
917 guint64 latency;
918 GstClockTime in_ts;
919
920 GstClockTime next_out_ts;
921 guint64 next_out_off;
922
923 guint64 in_buffer_count, out_buffer_count;
924 } TimestampDriftCtx;
925
926 static void
fakesink_handoff_cb(GstElement * object,GstBuffer * buffer,GstPad * pad,gpointer user_data)927 fakesink_handoff_cb (GstElement * object, GstBuffer * buffer, GstPad * pad,
928 gpointer user_data)
929 {
930 TimestampDriftCtx *ctx = user_data;
931
932 ctx->out_buffer_count++;
933 if (ctx->latency == GST_CLOCK_TIME_NONE) {
934 ctx->latency = 1000 - gst_buffer_get_size (buffer) / 8;
935 }
936
937 /* Check if we have a perfectly timestamped stream */
938 if (ctx->next_out_ts != GST_CLOCK_TIME_NONE)
939 fail_unless (ctx->next_out_ts == GST_BUFFER_TIMESTAMP (buffer),
940 "expected timestamp %" GST_TIME_FORMAT " got timestamp %"
941 GST_TIME_FORMAT, GST_TIME_ARGS (ctx->next_out_ts),
942 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
943
944 /* Check if we have a perfectly offsetted stream */
945 fail_unless (GST_BUFFER_OFFSET_END (buffer) ==
946 GST_BUFFER_OFFSET (buffer) + gst_buffer_get_size (buffer) / 8,
947 "expected offset end %" G_GUINT64_FORMAT " got offset end %"
948 G_GUINT64_FORMAT,
949 GST_BUFFER_OFFSET (buffer) + gst_buffer_get_size (buffer) / 8,
950 GST_BUFFER_OFFSET_END (buffer));
951 if (ctx->next_out_off != GST_BUFFER_OFFSET_NONE) {
952 fail_unless (GST_BUFFER_OFFSET (buffer) == ctx->next_out_off,
953 "expected offset %" G_GUINT64_FORMAT " got offset %" G_GUINT64_FORMAT,
954 ctx->next_out_off, GST_BUFFER_OFFSET (buffer));
955 }
956
957 if (ctx->in_buffer_count != ctx->out_buffer_count) {
958 GST_INFO ("timestamp %" GST_TIME_FORMAT,
959 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
960 }
961
962 if (ctx->in_ts != GST_CLOCK_TIME_NONE && ctx->in_buffer_count > 1
963 && ctx->in_buffer_count == ctx->out_buffer_count) {
964 fail_unless (GST_BUFFER_TIMESTAMP (buffer) ==
965 ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND,
966 4096),
967 "expected output timestamp %" GST_TIME_FORMAT " (%" G_GUINT64_FORMAT
968 ") got output timestamp %" GST_TIME_FORMAT " (%" G_GUINT64_FORMAT ")",
969 GST_TIME_ARGS (ctx->in_ts - gst_util_uint64_scale_round (ctx->latency,
970 GST_SECOND, 4096)),
971 ctx->in_ts - gst_util_uint64_scale_round (ctx->latency, GST_SECOND,
972 4096), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
973 GST_BUFFER_TIMESTAMP (buffer));
974 }
975
976 ctx->next_out_ts =
977 GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer);
978 ctx->next_out_off = GST_BUFFER_OFFSET_END (buffer);
979 }
980
981 static void
identity_handoff_cb(GstElement * object,GstBuffer * buffer,gpointer user_data)982 identity_handoff_cb (GstElement * object, GstBuffer * buffer,
983 gpointer user_data)
984 {
985 TimestampDriftCtx *ctx = user_data;
986
987 ctx->in_ts = GST_BUFFER_TIMESTAMP (buffer);
988 ctx->in_buffer_count++;
989 }
990
GST_START_TEST(test_timestamp_drift)991 GST_START_TEST (test_timestamp_drift)
992 {
993 TimestampDriftCtx ctx =
994 { GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE,
995 GST_BUFFER_OFFSET_NONE, 0, 0
996 };
997 GstElement *pipeline;
998 GstElement *audiotestsrc, *capsfilter1, *identity, *audioresample,
999 *capsfilter2, *fakesink;
1000 GstBus *bus;
1001 GMainLoop *loop;
1002 GstCaps *caps;
1003
1004 pipeline = gst_pipeline_new ("pipeline");
1005 fail_unless (pipeline != NULL);
1006
1007 audiotestsrc = gst_element_factory_make ("audiotestsrc", "src");
1008 fail_unless (audiotestsrc != NULL);
1009 g_object_set (G_OBJECT (audiotestsrc), "num-buffers", 10000,
1010 "samplesperbuffer", 4000, NULL);
1011
1012 capsfilter1 = gst_element_factory_make ("capsfilter", "capsfilter1");
1013 fail_unless (capsfilter1 != NULL);
1014 caps = gst_caps_from_string ("audio/x-raw, format=" GST_AUDIO_NE (F64)
1015 ", channels=1, rate=16384");
1016 g_object_set (G_OBJECT (capsfilter1), "caps", caps, NULL);
1017 gst_caps_unref (caps);
1018
1019 identity = gst_element_factory_make ("identity", "identity");
1020 fail_unless (identity != NULL);
1021 g_object_set (G_OBJECT (identity), "sync", FALSE, "signal-handoffs", TRUE,
1022 NULL);
1023 g_signal_connect (identity, "handoff", (GCallback) identity_handoff_cb, &ctx);
1024
1025 audioresample = gst_element_factory_make ("audioresample", "resample");
1026 fail_unless (audioresample != NULL);
1027 capsfilter2 = gst_element_factory_make ("capsfilter", "capsfilter2");
1028 fail_unless (capsfilter2 != NULL);
1029 caps = gst_caps_from_string ("audio/x-raw, format=" GST_AUDIO_NE (F64)
1030 ", channels=1, rate=4096");
1031 g_object_set (G_OBJECT (capsfilter2), "caps", caps, NULL);
1032 gst_caps_unref (caps);
1033
1034 fakesink = gst_element_factory_make ("fakesink", "sink");
1035 fail_unless (fakesink != NULL);
1036 g_object_set (G_OBJECT (fakesink), "sync", FALSE, "async", FALSE,
1037 "signal-handoffs", TRUE, NULL);
1038 g_signal_connect (fakesink, "handoff", (GCallback) fakesink_handoff_cb, &ctx);
1039
1040
1041 gst_bin_add_many (GST_BIN (pipeline), audiotestsrc, capsfilter1, identity,
1042 audioresample, capsfilter2, fakesink, NULL);
1043 fail_unless (gst_element_link_many (audiotestsrc, capsfilter1, identity,
1044 audioresample, capsfilter2, fakesink, NULL));
1045
1046 loop = g_main_loop_new (NULL, FALSE);
1047
1048 bus = gst_element_get_bus (pipeline);
1049 gst_bus_add_signal_watch (bus);
1050 g_signal_connect (bus, "message", (GCallback) _message_cb, loop);
1051
1052 fail_unless (gst_element_set_state (pipeline,
1053 GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
1054 g_main_loop_run (loop);
1055
1056 fail_unless (gst_element_set_state (pipeline,
1057 GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
1058 g_main_loop_unref (loop);
1059 gst_bus_remove_signal_watch (bus);
1060 gst_object_unref (bus);
1061
1062 gst_object_unref (pipeline);
1063
1064 } GST_END_TEST;
1065
1066 #define FFT_HELPERS(type,ffttag,ffttag2,scale); \
1067 static gdouble magnitude##ffttag (const GstFFT##ffttag##Complex *c) \
1068 { \
1069 gdouble mag = (gdouble) c->r * (gdouble) c->r; \
1070 mag += (gdouble) c->i * (gdouble) c->i; \
1071 mag /= scale * scale; \
1072 mag = 10.0 * log10 (mag); \
1073 return mag; \
1074 } \
1075 static gdouble find_main_frequency_spot_##ffttag (const GstFFT##ffttag##Complex *v, \
1076 int elements) \
1077 { \
1078 int i; \
1079 gdouble maxmag = -9999; \
1080 int maxidx = 0; \
1081 for (i=0; i<elements; ++i) { \
1082 gdouble mag = magnitude##ffttag (v+i); \
1083 if (mag > maxmag) { \
1084 maxmag = mag; \
1085 maxidx = i; \
1086 } \
1087 } \
1088 return maxidx / (gdouble) elements; \
1089 } \
1090 static gboolean is_zero_except_##ffttag (const GstFFT##ffttag##Complex *v, int elements, \
1091 gdouble spot) \
1092 { \
1093 int i; \
1094 for (i=0; i<elements; ++i) { \
1095 gdouble pos = i / (gdouble) elements; \
1096 gdouble mag = magnitude##ffttag (v+i); \
1097 if (fabs (pos - spot) > 0.01) { \
1098 if (mag > -55.0) { \
1099 return FALSE; \
1100 } \
1101 } \
1102 } \
1103 return TRUE; \
1104 } \
1105 static void compare_ffts_##ffttag (GstBuffer *inbuffer, GstBuffer *outbuffer) \
1106 { \
1107 GstMapInfo inmap, outmap; \
1108 int insamples, outsamples; \
1109 gdouble inspot, outspot; \
1110 GstFFT##ffttag *inctx, *outctx; \
1111 GstFFT##ffttag##Complex *in, *out; \
1112 \
1113 gst_buffer_map (inbuffer, &inmap, GST_MAP_READ); \
1114 gst_buffer_map (outbuffer, &outmap, GST_MAP_READWRITE); \
1115 \
1116 insamples = inmap.size / sizeof(type) & ~1; \
1117 outsamples = outmap.size / sizeof(type) & ~1; \
1118 inctx = gst_fft_##ffttag2##_new (insamples, FALSE); \
1119 outctx = gst_fft_##ffttag2##_new (outsamples, FALSE); \
1120 in = g_new (GstFFT##ffttag##Complex, insamples / 2 + 1); \
1121 out = g_new (GstFFT##ffttag##Complex, outsamples / 2 + 1); \
1122 \
1123 gst_fft_##ffttag2##_window (inctx, (type*)inmap.data, \
1124 GST_FFT_WINDOW_HAMMING); \
1125 gst_fft_##ffttag2##_fft (inctx, (type*)inmap.data, in); \
1126 gst_fft_##ffttag2##_window (outctx, (type*)outmap.data, \
1127 GST_FFT_WINDOW_HAMMING); \
1128 gst_fft_##ffttag2##_fft (outctx, (type*)outmap.data, out); \
1129 \
1130 inspot = find_main_frequency_spot_##ffttag (in, insamples / 2 + 1); \
1131 outspot = find_main_frequency_spot_##ffttag (out, outsamples / 2 + 1); \
1132 GST_LOG ("Spots are %.3f and %.3f", inspot, outspot); \
1133 fail_unless (fabs (outspot - inspot) < 0.05); \
1134 fail_unless (is_zero_except_##ffttag (in, insamples / 2 + 1, inspot)); \
1135 fail_unless (is_zero_except_##ffttag (out, outsamples / 2 + 1, outspot)); \
1136 \
1137 gst_buffer_unmap (inbuffer, &inmap); \
1138 gst_buffer_unmap (outbuffer, &outmap); \
1139 \
1140 gst_fft_##ffttag2##_free (inctx); \
1141 gst_fft_##ffttag2##_free (outctx); \
1142 g_free (in); \
1143 g_free (out); \
1144 }
1145 FFT_HELPERS (float, F32, f32, 2048.0f);
1146 FFT_HELPERS (double, F64, f64, 2048.0);
1147 FFT_HELPERS (gint16, S16, s16, 32767.0);
1148 FFT_HELPERS (gint32, S32, s32, 2147483647.0);
1149
1150 #define FILL_BUFFER(type, desc, value); \
1151 static void init_##type##_##desc (GstBuffer *buffer) \
1152 { \
1153 GstMapInfo map; \
1154 type *ptr; \
1155 int i, nsamples; \
1156 gst_buffer_map (buffer, &map, GST_MAP_WRITE); \
1157 ptr = (type *)map.data; \
1158 nsamples = map.size / sizeof (type); \
1159 for (i = 0; i < nsamples; ++i) { \
1160 *ptr++ = value; \
1161 } \
1162 gst_buffer_unmap (buffer, &map); \
1163 }
1164
1165 FILL_BUFFER (float, silence, 0.0f);
1166 FILL_BUFFER (double, silence, 0.0);
1167 FILL_BUFFER (gint16, silence, 0);
1168 FILL_BUFFER (gint32, silence, 0);
1169 FILL_BUFFER (float, sine, sinf (i * 0.01f));
1170 FILL_BUFFER (float, sine2, sinf (i * 1.8f));
1171 FILL_BUFFER (double, sine, sin (i * 0.01));
1172 FILL_BUFFER (double, sine2, sin (i * 1.8));
1173 FILL_BUFFER (gint16, sine, (gint16) (32767 * sinf (i * 0.01f)));
1174 FILL_BUFFER (gint16, sine2, (gint16) (32767 * sinf (i * 1.8f)));
1175 FILL_BUFFER (gint32, sine, (gint32) (2147483647.0 * sin (i * 0.01)));
1176 FILL_BUFFER (gint32, sine2, (gint32) (2147483647.0 * sin (i * 1.8)));
1177
1178 static void
run_fft_pipeline(int inrate,int outrate,int quality,int width,const gchar * format,void (* init)(GstBuffer *),void (* compare_ffts)(GstBuffer *,GstBuffer *))1179 run_fft_pipeline (int inrate, int outrate, int quality, int width,
1180 const gchar * format, void (*init) (GstBuffer *),
1181 void (*compare_ffts) (GstBuffer *, GstBuffer *))
1182 {
1183 GstElement *audioresample;
1184 GstBuffer *inbuffer, *outbuffer;
1185 const int nsamples = 2048;
1186
1187 audioresample = setup_audioresample (1, 0, inrate, outrate, format);
1188 fail_unless (audioresample != NULL);
1189 g_object_set (audioresample, "quality", quality, NULL);
1190
1191 fail_unless (gst_element_set_state (audioresample,
1192 GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
1193 "could not set to playing");
1194
1195 inbuffer = gst_buffer_new_and_alloc (nsamples * width / 8);
1196 GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (nsamples, inrate);
1197 GST_BUFFER_TIMESTAMP (inbuffer) = 0;
1198
1199 (*init) (inbuffer);
1200
1201 gst_buffer_ref (inbuffer);
1202 /* pushing gives away my reference ... */
1203 fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
1204 /* ... but it ends up being collected on the global buffer list */
1205 fail_unless_equals_int (g_list_length (buffers), 1);
1206 /* retrieve out buffer */
1207 fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
1208
1209 fail_unless (gst_element_set_state (audioresample,
1210 GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to null");
1211
1212 if (inbuffer == outbuffer)
1213 gst_buffer_unref (inbuffer);
1214
1215 (*compare_ffts) (inbuffer, outbuffer);
1216
1217 /* cleanup */
1218 cleanup_audioresample (audioresample);
1219 }
1220
GST_START_TEST(test_fft)1221 GST_START_TEST (test_fft)
1222 {
1223 int quality;
1224 size_t f0, f1;
1225 static const int frequencies[] =
1226 { 8000, 16000, 44100, 48000, 128000, 12345, 54321 };
1227
1228 /* audioresample uses a mixed float/double code path for floats with quality>8, make sure we test it */
1229 for (quality = 0; quality <= 10; quality += 5) {
1230 for (f0 = 0; f0 < G_N_ELEMENTS (frequencies); ++f0) {
1231 for (f1 = 0; f1 < G_N_ELEMENTS (frequencies); ++f1) {
1232 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
1233 GST_AUDIO_NE (F32), &init_float_silence, &compare_ffts_F32);
1234 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
1235 GST_AUDIO_NE (F32), &init_float_sine, &compare_ffts_F32);
1236 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
1237 GST_AUDIO_NE (F32), &init_float_sine2, &compare_ffts_F32);
1238 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64,
1239 GST_AUDIO_NE (F64), &init_double_silence, &compare_ffts_F64);
1240 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64,
1241 GST_AUDIO_NE (F64), &init_double_sine, &compare_ffts_F64);
1242 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 64,
1243 GST_AUDIO_NE (F64), &init_double_sine2, &compare_ffts_F64);
1244 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16,
1245 GST_AUDIO_NE (S16), &init_gint16_silence, &compare_ffts_S16);
1246 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16,
1247 GST_AUDIO_NE (S16), &init_gint16_sine, &compare_ffts_S16);
1248 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 16,
1249 GST_AUDIO_NE (S16), &init_gint16_sine2, &compare_ffts_S16);
1250 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
1251 GST_AUDIO_NE (S32), &init_gint32_silence, &compare_ffts_S32);
1252 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
1253 GST_AUDIO_NE (S32), &init_gint32_sine, &compare_ffts_S32);
1254 run_fft_pipeline (frequencies[f0], frequencies[f0], quality, 32,
1255 GST_AUDIO_NE (S32), &init_gint32_sine2, &compare_ffts_S32);
1256 }
1257 }
1258 }
1259 }
1260
1261 GST_END_TEST;
1262
1263 static Suite *
audioresample_suite(void)1264 audioresample_suite (void)
1265 {
1266 Suite *s = suite_create ("audioresample");
1267 TCase *tc_chain = tcase_create ("general");
1268
1269 suite_add_tcase (s, tc_chain);
1270 tcase_add_test (tc_chain, test_perfect_stream);
1271 tcase_add_test (tc_chain, test_discont_stream);
1272 tcase_add_test (tc_chain, test_reuse);
1273 tcase_add_test (tc_chain, test_shutdown);
1274 tcase_add_test (tc_chain, test_live_switch);
1275 tcase_add_test (tc_chain, test_live_switch_downstream);
1276 tcase_add_test (tc_chain, test_timestamp_drift);
1277 tcase_add_test (tc_chain, test_fft);
1278
1279 #ifndef GST_DISABLE_PARSE
1280 tcase_set_timeout (tc_chain, 360);
1281 tcase_add_test (tc_chain, test_pipelines);
1282 tcase_add_test (tc_chain, test_preference_passthrough);
1283 #endif
1284
1285 return s;
1286 }
1287
1288 GST_CHECK_MAIN (audioresample);
1289