1 /* GStreamer
2 *
3 * unit test for RTP RFC 6464 Header Extensions
4 *
5 * Copyright (C) <2020-2021> Guillaume Desmottes <guillaume.desmottes@collabora.com>
6 *
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.
11 *
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.
16 *
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.
21 */
22
23 #include <gst/check/gstcheck.h>
24 #include <gst/rtp/rtp.h>
25 #include <gst/sdp/gstsdpmessage.h>
26 #include <gst/audio/audio.h>
27 #include <gst/check/gstharness.h>
28
29 #define URN "urn:ietf:params:rtp-hdrext:ssrc-audio-level"
30
31 #define SDP "v=0\r\n" \
32 "o=- 123456 2 IN IP4 127.0.0.1 \r\n" \
33 "s=-\r\n" \
34 "t=0 0\r\n" \
35 "a=maxptime:60\r\n" \
36 "a=sendrecv\r\n" \
37 "m=audio 55815 RTP/SAVPF 100\r\n" \
38 "c=IN IP4 1.1.1.1\r\n" \
39 "a=rtpmap:100 opus/48000/2\r\n"
40
41 #define SDP_NO_VAD SDP \
42 "a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\n"
43 #define SDP_VAD_ON SDP \
44 "a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=on\r\n"
45 #define SDP_VAD_OFF SDP \
46 "a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=off\r\n"
47 #define SDP_VAD_WRONG SDP \
48 "a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=badger\r\n"
49
50 static GstCaps *
create_caps(const gchar * sdp)51 create_caps (const gchar * sdp)
52 {
53 GstSDPMessage *message;
54 glong length = -1;
55 const GstSDPMedia *media;
56 GstCaps *caps;
57
58 gst_sdp_message_new (&message);
59 gst_sdp_message_parse_buffer ((guint8 *) sdp, length, message);
60 media = gst_sdp_message_get_media (message, 0);
61 fail_unless (media != NULL);
62
63 caps = gst_sdp_media_get_caps_from_media (media, 100);
64 gst_sdp_media_attributes_to_caps (media, caps);
65 gst_sdp_message_free (message);
66 return caps;
67 }
68
69 static void
check_caps(GstRTPHeaderExtension * ext,gboolean vad)70 check_caps (GstRTPHeaderExtension * ext, gboolean vad)
71 {
72 GstCaps *caps;
73 GstStructure *s;
74 const GValue *arr, *val;
75
76 caps = gst_caps_new_empty_simple ("application/x-rtp");
77 fail_unless (gst_rtp_header_extension_set_caps_from_attributes (ext, caps));
78 s = gst_caps_get_structure (caps, 0);
79
80 arr = gst_structure_get_value (s, "extmap-1");
81 fail_unless (arr != NULL);
82 fail_unless (GST_VALUE_HOLDS_ARRAY (arr));
83 fail_unless (gst_value_array_get_size (arr) == 3);
84
85 val = gst_value_array_get_value (arr, 0);
86 fail_unless_equals_string (g_value_get_string (val), "");
87
88 val = gst_value_array_get_value (arr, 1);
89 fail_unless_equals_string (g_value_get_string (val), URN);
90
91 val = gst_value_array_get_value (arr, 2);
92 if (vad) {
93 fail_unless_equals_string (g_value_get_string (val), "vad=on");
94 } else {
95 fail_unless_equals_string (g_value_get_string (val), "vad=off");
96 }
97
98 gst_caps_unref (caps);
99 }
100
GST_START_TEST(rtphdrext_client_audio_level_sdp)101 GST_START_TEST (rtphdrext_client_audio_level_sdp)
102 {
103 GstRTPHeaderExtension *ext;
104 GstCaps *caps;
105 gboolean vad = FALSE;
106
107 ext = gst_rtp_header_extension_create_from_uri (URN);
108 fail_unless (ext != NULL);
109 gst_rtp_header_extension_set_id (ext, 1);
110
111 /* vad default to on */
112 caps = create_caps (SDP_NO_VAD);
113 fail_unless (gst_rtp_header_extension_set_attributes_from_caps (ext, caps));
114 gst_caps_unref (caps);
115 g_object_get (ext, "vad", &vad, NULL);
116 fail_unless (vad);
117 check_caps (ext, TRUE);
118
119 /* vad is disabled */
120 caps = create_caps (SDP_VAD_OFF);
121 fail_unless (gst_rtp_header_extension_set_attributes_from_caps (ext, caps));
122 gst_caps_unref (caps);
123 g_object_get (ext, "vad", &vad, NULL);
124 fail_if (vad);
125
126 /* vad is enabled */
127 caps = create_caps (SDP_VAD_ON);
128 fail_unless (gst_rtp_header_extension_set_attributes_from_caps (ext, caps));
129 gst_caps_unref (caps);
130 g_object_get (ext, "vad", &vad, NULL);
131 fail_unless (vad);
132
133 /* invalid vad */
134 caps = create_caps (SDP_VAD_WRONG);
135 fail_if (gst_rtp_header_extension_set_attributes_from_caps (ext, caps));
136 gst_caps_unref (caps);
137
138 gst_object_unref (ext);
139 }
140
141 GST_END_TEST;
142
GST_START_TEST(rtphdrext_client_audio_level_one_byte)143 GST_START_TEST (rtphdrext_client_audio_level_one_byte)
144 {
145 GstRTPHeaderExtension *ext;
146 GstRTPHeaderExtensionFlags flags;
147 GstBuffer *buffer;
148 guint8 *data;
149 gsize size, written;
150 GstAudioLevelMeta *meta;
151 guint8 level = 12;
152 gboolean voice = TRUE;
153
154 ext = gst_rtp_header_extension_create_from_uri (URN);
155 fail_unless (ext != NULL);
156 gst_rtp_header_extension_set_id (ext, 1);
157
158 flags = gst_rtp_header_extension_get_supported_flags (ext);
159 fail_unless (flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE);
160
161 buffer = gst_buffer_new ();
162 meta = gst_buffer_add_audio_level_meta (buffer, level, voice);
163
164 size = gst_rtp_header_extension_get_max_size (ext, buffer);
165 fail_unless (size > 0);
166 data = g_malloc0 (size);
167 fail_unless (data != NULL);
168
169 /* Write extension */
170 written =
171 gst_rtp_header_extension_write (ext, buffer,
172 GST_RTP_HEADER_EXTENSION_ONE_BYTE, buffer, data, size);
173 fail_unless (written == 1);
174
175 /* Read it back */
176 fail_unless (gst_buffer_remove_meta (buffer, (GstMeta *) meta));
177 fail_unless (gst_rtp_header_extension_read (ext,
178 GST_RTP_HEADER_EXTENSION_ONE_BYTE, data, size, buffer));
179 meta = gst_buffer_get_audio_level_meta (buffer);
180 fail_unless (meta != NULL);
181 fail_unless_equals_int (meta->level, level);
182 fail_unless (meta->voice_activity == voice);
183
184 g_free (data);
185 gst_buffer_unref (buffer);
186 gst_object_unref (ext);
187 }
188
189 GST_END_TEST;
190
GST_START_TEST(rtphdrext_client_audio_level_two_bytes)191 GST_START_TEST (rtphdrext_client_audio_level_two_bytes)
192 {
193 GstRTPHeaderExtension *ext;
194 GstRTPHeaderExtensionFlags flags;
195 GstBuffer *buffer;
196 guint8 *data;
197 gsize size, written;
198 GstAudioLevelMeta *meta;
199 guint8 level = 12;
200 gboolean voice = TRUE;
201
202 ext = gst_rtp_header_extension_create_from_uri (URN);
203 fail_unless (ext != NULL);
204 gst_rtp_header_extension_set_id (ext, 1);
205
206 flags = gst_rtp_header_extension_get_supported_flags (ext);
207 fail_unless (flags & GST_RTP_HEADER_EXTENSION_TWO_BYTE);
208
209 buffer = gst_buffer_new ();
210 meta = gst_buffer_add_audio_level_meta (buffer, level, voice);
211
212 size = gst_rtp_header_extension_get_max_size (ext, buffer);
213 fail_unless (size > 0);
214 data = g_malloc0 (size);
215 fail_unless (data != NULL);
216
217 /* Write extension */
218 written =
219 gst_rtp_header_extension_write (ext, buffer,
220 GST_RTP_HEADER_EXTENSION_TWO_BYTE, buffer, data, size);
221 fail_unless (written == 2);
222
223 /* Read it back */
224 fail_unless (gst_buffer_remove_meta (buffer, (GstMeta *) meta));
225 fail_unless (gst_rtp_header_extension_read (ext,
226 GST_RTP_HEADER_EXTENSION_TWO_BYTE, data, size, buffer));
227 meta = gst_buffer_get_audio_level_meta (buffer);
228 fail_unless (meta != NULL);
229 fail_unless_equals_int (meta->level, level);
230 fail_unless (meta->voice_activity == voice);
231
232 g_free (data);
233 gst_buffer_unref (buffer);
234 gst_object_unref (ext);
235 }
236
237 GST_END_TEST;
238
GST_START_TEST(rtphdrext_client_audio_level_no_meta)239 GST_START_TEST (rtphdrext_client_audio_level_no_meta)
240 {
241 GstRTPHeaderExtension *ext;
242 GstBuffer *buffer;
243 guint8 *data;
244 gsize size, written;
245
246 ext = gst_rtp_header_extension_create_from_uri (URN);
247 fail_unless (ext != NULL);
248 gst_rtp_header_extension_set_id (ext, 1);
249
250 buffer = gst_buffer_new ();
251
252 size = gst_rtp_header_extension_get_max_size (ext, buffer);
253 fail_unless (size > 0);
254 data = g_malloc0 (size);
255 fail_unless (data != NULL);
256
257 written =
258 gst_rtp_header_extension_write (ext, buffer,
259 GST_RTP_HEADER_EXTENSION_ONE_BYTE, buffer, data, size);
260 fail_unless (written == 0);
261
262 written =
263 gst_rtp_header_extension_write (ext, buffer,
264 GST_RTP_HEADER_EXTENSION_TWO_BYTE, buffer, data, size);
265 fail_unless (written == 0);
266
267 g_free (data);
268 gst_buffer_unref (buffer);
269 gst_object_unref (ext);
270 }
271
272 GST_END_TEST;
273
GST_START_TEST(rtphdrext_client_audio_level_payloader_depayloader)274 GST_START_TEST (rtphdrext_client_audio_level_payloader_depayloader)
275 {
276 GstHarness *h;
277 GstBuffer *b;
278 GstFlowReturn fret;
279 GstAudioLevelMeta *meta;
280
281 h = gst_harness_new_parse ("rtpL16pay ! "
282 "application/x-rtp, extmap-1=(string)< \"\", " URN " , \"vad=on\" >"
283 " ! rtpL16depay");
284
285 gst_harness_set_src_caps_str (h, "audio/x-raw, rate=44100, channels=1,"
286 " layout=interleaved, format=S16BE");
287
288 b = gst_buffer_new_allocate (NULL, 100, NULL);
289 gst_buffer_add_audio_level_meta (b, 12, TRUE);
290 fret = gst_harness_push (h, b);
291 fail_unless (fret == GST_FLOW_OK);
292
293 b = gst_harness_pull (h);
294 meta = gst_buffer_get_audio_level_meta (b);
295
296 fail_unless (meta != NULL);
297 fail_unless (meta->level == 12);
298 fail_unless (meta->voice_activity == TRUE);
299
300 gst_buffer_unref (b);
301 gst_harness_teardown (h);
302 }
303
304 GST_END_TEST;
305
306
GST_START_TEST(rtphdrext_client_audio_level_payloader_api)307 GST_START_TEST (rtphdrext_client_audio_level_payloader_api)
308 {
309 GstHarness *h;
310 GstRTPHeaderExtension *ext;
311 GstBuffer *b;
312 GstFlowReturn fret;
313 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
314 guint8 *data;
315 guint size;
316 guint8 level;
317 gboolean voice_activity;
318
319 h = gst_harness_new ("rtpL16pay");
320 gst_harness_set_src_caps_str (h, "audio/x-raw, rate=44100, channels=1,"
321 " layout=interleaved, format=S16BE");
322
323 ext = gst_rtp_header_extension_create_from_uri (URN);
324 gst_rtp_header_extension_set_id (ext, 2);
325 fail_unless (ext);
326 g_signal_emit_by_name (h->element, "add-extension", ext);
327
328 b = gst_buffer_new_allocate (NULL, 100, NULL);
329 gst_buffer_add_audio_level_meta (b, 12, TRUE);
330 fret = gst_harness_push (h, b);
331 fail_unless (fret == GST_FLOW_OK);
332
333 b = gst_harness_pull (h);
334 fail_unless (gst_rtp_buffer_map (b, GST_MAP_READ, &rtp));
335 fail_unless (gst_rtp_buffer_get_extension_onebyte_header (&rtp, 2, 0,
336 (gpointer *) & data, &size));
337 fail_unless (size == 1);
338 level = data[0] & 0x7F;
339 voice_activity = (data[0] & 0x80) >> 7;
340 fail_unless (level == 12);
341 fail_unless (voice_activity == TRUE);
342 gst_rtp_buffer_unmap (&rtp);
343 gst_buffer_unref (b);
344
345 gst_object_unref (ext);
346 gst_harness_teardown (h);
347 }
348
349 GST_END_TEST;
350
351
352 static Suite *
rtphdrext_client_audio_level_suite(void)353 rtphdrext_client_audio_level_suite (void)
354 {
355 Suite *s = suite_create ("rtphdrext_client_audio_level");
356 TCase *tc_chain = tcase_create ("general");
357
358 suite_add_tcase (s, tc_chain);
359
360 tcase_add_test (tc_chain, rtphdrext_client_audio_level_sdp);
361 tcase_add_test (tc_chain, rtphdrext_client_audio_level_one_byte);
362 tcase_add_test (tc_chain, rtphdrext_client_audio_level_two_bytes);
363 tcase_add_test (tc_chain, rtphdrext_client_audio_level_no_meta);
364 tcase_add_test (tc_chain, rtphdrext_client_audio_level_payloader_depayloader);
365 tcase_add_test (tc_chain, rtphdrext_client_audio_level_payloader_api);
366
367 return s;
368 }
369
370 GST_CHECK_MAIN (rtphdrext_client_audio_level)
371