1 /* GStreamer
2 * Copyright (C) <2005> Wim Taymans <wim.taymans@gmail.com>
3 *
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
8 *
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
13 *
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
18 */
19
20 /**
21 * SECTION:element-rtpamrdepay
22 * @title: rtpamrdepay
23 * @see_also: rtpamrpay
24 *
25 * Extract AMR audio from RTP packets according to RFC 3267.
26 * For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.txt
27 *
28 * ## Example pipeline
29 * |[
30 * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)AMR, encoding-params=(string)1, octet-align=(string)1, payload=(int)96' ! rtpamrdepay ! amrnbdec ! pulsesink
31 * ]| This example pipeline will depayload and decode an RTP AMR stream. Refer to
32 * the rtpamrpay example to create the RTP stream.
33 *
34 */
35
36 /*
37 * RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
38 * Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
39 * Wideband (AMR-WB) Audio Codecs.
40 *
41 */
42 #ifdef HAVE_CONFIG_H
43 # include "config.h"
44 #endif
45
46 #include <gst/rtp/gstrtpbuffer.h>
47 #include <gst/audio/audio.h>
48
49 #include <stdlib.h>
50 #include <string.h>
51 #include "gstrtpelements.h"
52 #include "gstrtpamrdepay.h"
53 #include "gstrtputils.h"
54
55 GST_DEBUG_CATEGORY_STATIC (rtpamrdepay_debug);
56 #define GST_CAT_DEFAULT (rtpamrdepay_debug)
57
58 /* RtpAMRDepay signals and args */
59 enum
60 {
61 /* FILL ME */
62 LAST_SIGNAL
63 };
64
65 enum
66 {
67 PROP_0
68 };
69
70 /* input is an RTP packet
71 *
72 * params see RFC 3267, section 8.1
73 */
74 static GstStaticPadTemplate gst_rtp_amr_depay_sink_template =
75 GST_STATIC_PAD_TEMPLATE ("sink",
76 GST_PAD_SINK,
77 GST_PAD_ALWAYS,
78 GST_STATIC_CAPS ("application/x-rtp, "
79 "media = (string) \"audio\", "
80 "clock-rate = (int) 8000, " "encoding-name = (string) \"AMR\", "
81 /* This is the default, so the peer doesn't have to specify it
82 * "encoding-params = (string) \"1\", " */
83 /* NOTE that all values must be strings in orde to be able to do SDP <->
84 * GstCaps mapping. */
85 "octet-align = (string) \"1\";"
86 /* following options are not needed for a decoder
87 *
88 "crc = (string) { \"0\", \"1\" }, "
89 "robust-sorting = (string) \"0\", "
90 "interleaving = (string) \"0\";"
91 "mode-set = (int) [ 0, 7 ], "
92 "mode-change-period = (int) [ 1, MAX ], "
93 "mode-change-neighbor = (boolean) { TRUE, FALSE }, "
94 "maxptime = (int) [ 20, MAX ], "
95 "ptime = (int) [ 20, MAX ]"
96 */
97 "application/x-rtp, "
98 "media = (string) \"audio\", "
99 "clock-rate = (int) 16000, " "encoding-name = (string) \"AMR-WB\", "
100 /* This is the default, so the peer doesn't have to specify it
101 * "encoding-params = (string) \"1\", " */
102 /* NOTE that all values must be strings in orde to be able to do SDP <->
103 * GstCaps mapping. */
104 "octet-align = (string) \"1\";"
105 /* following options are not needed for a decoder
106 *
107 "crc = (string) { \"0\", \"1\" }, "
108 "robust-sorting = (string) \"0\", "
109 "interleaving = (string) \"0\""
110 "mode-set = (int) [ 0, 7 ], "
111 "mode-change-period = (int) [ 1, MAX ], "
112 "mode-change-neighbor = (boolean) { TRUE, FALSE }, "
113 "maxptime = (int) [ 20, MAX ], "
114 "ptime = (int) [ 20, MAX ]"
115 */
116 )
117 );
118
119 static GstStaticPadTemplate gst_rtp_amr_depay_src_template =
120 GST_STATIC_PAD_TEMPLATE ("src",
121 GST_PAD_SRC,
122 GST_PAD_ALWAYS,
123 GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000;"
124 "audio/AMR-WB, " "channels = (int) 1," "rate = (int) 16000")
125 );
126
127 static gboolean gst_rtp_amr_depay_setcaps (GstRTPBaseDepayload * depayload,
128 GstCaps * caps);
129 static GstBuffer *gst_rtp_amr_depay_process (GstRTPBaseDepayload * depayload,
130 GstRTPBuffer * rtp);
131
132 #define gst_rtp_amr_depay_parent_class parent_class
133 G_DEFINE_TYPE (GstRtpAMRDepay, gst_rtp_amr_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
134 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpamrdepay, "rtpamrdepay",
135 GST_RANK_SECONDARY, GST_TYPE_RTP_AMR_DEPAY, rtp_element_init (plugin));
136
137 static void
gst_rtp_amr_depay_class_init(GstRtpAMRDepayClass * klass)138 gst_rtp_amr_depay_class_init (GstRtpAMRDepayClass * klass)
139 {
140 GstElementClass *gstelement_class;
141 GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
142
143 gstelement_class = (GstElementClass *) klass;
144 gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
145
146 gst_element_class_add_static_pad_template (gstelement_class,
147 &gst_rtp_amr_depay_src_template);
148 gst_element_class_add_static_pad_template (gstelement_class,
149 &gst_rtp_amr_depay_sink_template);
150
151 gst_element_class_set_static_metadata (gstelement_class,
152 "RTP AMR depayloader", "Codec/Depayloader/Network/RTP",
153 "Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)",
154 "Wim Taymans <wim.taymans@gmail.com>");
155
156 gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_amr_depay_process;
157 gstrtpbasedepayload_class->set_caps = gst_rtp_amr_depay_setcaps;
158
159 GST_DEBUG_CATEGORY_INIT (rtpamrdepay_debug, "rtpamrdepay", 0,
160 "AMR/AMR-WB RTP Depayloader");
161 }
162
163 static void
gst_rtp_amr_depay_init(GstRtpAMRDepay * rtpamrdepay)164 gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay)
165 {
166 GstRTPBaseDepayload *depayload;
167
168 depayload = GST_RTP_BASE_DEPAYLOAD (rtpamrdepay);
169
170 gst_pad_use_fixed_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload));
171 }
172
173 static gboolean
gst_rtp_amr_depay_setcaps(GstRTPBaseDepayload * depayload,GstCaps * caps)174 gst_rtp_amr_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
175 {
176 GstStructure *structure;
177 GstCaps *srccaps;
178 GstRtpAMRDepay *rtpamrdepay;
179 const gchar *params;
180 const gchar *str, *type;
181 gint clock_rate, need_clock_rate;
182 gboolean res;
183
184 rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
185
186 structure = gst_caps_get_structure (caps, 0);
187
188 /* figure out the mode first and set the clock rates */
189 if ((str = gst_structure_get_string (structure, "encoding-name"))) {
190 if (strcmp (str, "AMR") == 0) {
191 rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_NB;
192 need_clock_rate = 8000;
193 type = "audio/AMR";
194 } else if (strcmp (str, "AMR-WB") == 0) {
195 rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_WB;
196 need_clock_rate = 16000;
197 type = "audio/AMR-WB";
198 } else
199 goto invalid_mode;
200 } else
201 goto invalid_mode;
202
203 if (!(str = gst_structure_get_string (structure, "octet-align")))
204 rtpamrdepay->octet_align = FALSE;
205 else
206 rtpamrdepay->octet_align = (atoi (str) == 1);
207
208 if (!(str = gst_structure_get_string (structure, "crc")))
209 rtpamrdepay->crc = FALSE;
210 else
211 rtpamrdepay->crc = (atoi (str) == 1);
212
213 if (rtpamrdepay->crc) {
214 /* crc mode implies octet aligned mode */
215 rtpamrdepay->octet_align = TRUE;
216 }
217
218 if (!(str = gst_structure_get_string (structure, "robust-sorting")))
219 rtpamrdepay->robust_sorting = FALSE;
220 else
221 rtpamrdepay->robust_sorting = (atoi (str) == 1);
222
223 if (rtpamrdepay->robust_sorting) {
224 /* robust_sorting mode implies octet aligned mode */
225 rtpamrdepay->octet_align = TRUE;
226 }
227
228 if (!(str = gst_structure_get_string (structure, "interleaving")))
229 rtpamrdepay->interleaving = FALSE;
230 else
231 rtpamrdepay->interleaving = (atoi (str) == 1);
232
233 if (rtpamrdepay->interleaving) {
234 /* interleaving mode implies octet aligned mode */
235 rtpamrdepay->octet_align = TRUE;
236 }
237
238 if (!(params = gst_structure_get_string (structure, "encoding-params")))
239 rtpamrdepay->channels = 1;
240 else {
241 rtpamrdepay->channels = atoi (params);
242 }
243
244 if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
245 clock_rate = need_clock_rate;
246 depayload->clock_rate = clock_rate;
247
248 /* we require 1 channel, 8000 Hz, octet aligned, no CRC,
249 * no robust sorting, no interleaving for now */
250 if (rtpamrdepay->channels != 1)
251 return FALSE;
252 if (clock_rate != need_clock_rate)
253 return FALSE;
254 if (rtpamrdepay->octet_align != TRUE)
255 return FALSE;
256 if (rtpamrdepay->robust_sorting != FALSE)
257 return FALSE;
258 if (rtpamrdepay->interleaving != FALSE)
259 return FALSE;
260
261 srccaps = gst_caps_new_simple (type,
262 "channels", G_TYPE_INT, rtpamrdepay->channels,
263 "rate", G_TYPE_INT, clock_rate, NULL);
264 res = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
265 gst_caps_unref (srccaps);
266
267 return res;
268
269 /* ERRORS */
270 invalid_mode:
271 {
272 GST_ERROR_OBJECT (rtpamrdepay, "invalid encoding-name");
273 return FALSE;
274 }
275 }
276
277 /* -1 is invalid */
278 static const gint nb_frame_size[16] = {
279 12, 13, 15, 17, 19, 20, 26, 31,
280 5, -1, -1, -1, -1, -1, -1, 0
281 };
282
283 static const gint wb_frame_size[16] = {
284 17, 23, 32, 36, 40, 46, 50, 58,
285 60, 5, -1, -1, -1, -1, -1, 0
286 };
287
288 static GstBuffer *
gst_rtp_amr_depay_process(GstRTPBaseDepayload * depayload,GstRTPBuffer * rtp)289 gst_rtp_amr_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
290 {
291 GstRtpAMRDepay *rtpamrdepay;
292 const gint *frame_size;
293 GstBuffer *outbuf = NULL;
294 gint payload_len;
295 GstMapInfo map;
296
297 rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
298
299 /* setup frame size pointer */
300 if (rtpamrdepay->mode == GST_RTP_AMR_DP_MODE_NB)
301 frame_size = nb_frame_size;
302 else
303 frame_size = wb_frame_size;
304
305 /* when we get here, 1 channel, 8000/16000 Hz, octet aligned, no CRC,
306 * no robust sorting, no interleaving data is to be depayloaded */
307 {
308 guint8 *payload, *p, *dp;
309 gint i, num_packets, num_nonempty_packets;
310 gint amr_len;
311 gint ILL, ILP;
312
313 payload_len = gst_rtp_buffer_get_payload_len (rtp);
314
315 /* need at least 2 bytes for the header */
316 if (payload_len < 2)
317 goto too_small;
318
319 payload = gst_rtp_buffer_get_payload (rtp);
320
321 /* depay CMR. The CMR is used by the sender to request
322 * a new encoding mode.
323 *
324 * 0 1 2 3 4 5 6 7
325 * +-+-+-+-+-+-+-+-+
326 * | CMR |R|R|R|R|
327 * +-+-+-+-+-+-+-+-+
328 */
329 /* CMR = (payload[0] & 0xf0) >> 4; */
330
331 /* strip CMR header now, pack FT and the data for the decoder */
332 payload_len -= 1;
333 payload += 1;
334
335 GST_DEBUG_OBJECT (rtpamrdepay, "payload len %d", payload_len);
336
337 if (rtpamrdepay->interleaving) {
338 ILL = (payload[0] & 0xf0) >> 4;
339 ILP = (payload[0] & 0x0f);
340
341 payload_len -= 1;
342 payload += 1;
343
344 if (ILP > ILL)
345 goto wrong_interleaving;
346 }
347
348 /*
349 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
350 * +-+-+-+-+-+-+-+-+..
351 * |F| FT |Q|P|P| more FT..
352 * +-+-+-+-+-+-+-+-+..
353 */
354 /* count number of packets by counting the FTs. Also
355 * count number of amr data bytes and number of non-empty
356 * packets (this is also the number of CRCs if present). */
357 amr_len = 0;
358 num_nonempty_packets = 0;
359 num_packets = 0;
360 for (i = 0; i < payload_len; i++) {
361 gint fr_size;
362 guint8 FT;
363
364 FT = (payload[i] & 0x78) >> 3;
365
366 fr_size = frame_size[FT];
367 GST_DEBUG_OBJECT (rtpamrdepay, "frame size %d", fr_size);
368 if (fr_size == -1)
369 goto wrong_framesize;
370
371 if (fr_size > 0) {
372 amr_len += fr_size;
373 num_nonempty_packets++;
374 }
375 num_packets++;
376
377 if ((payload[i] & 0x80) == 0)
378 break;
379 }
380
381 if (rtpamrdepay->crc) {
382 /* data len + CRC len + header bytes should be smaller than payload_len */
383 if (num_packets + num_nonempty_packets + amr_len > payload_len)
384 goto wrong_length_1;
385 } else {
386 /* data len + header bytes should be smaller than payload_len */
387 if (num_packets + amr_len > payload_len)
388 goto wrong_length_2;
389 }
390
391 outbuf = gst_buffer_new_and_alloc (payload_len);
392
393 /* point to destination */
394 gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
395
396 /* point to first data packet */
397 p = map.data;
398 dp = payload + num_packets;
399 if (rtpamrdepay->crc) {
400 /* skip CRC if present */
401 dp += num_nonempty_packets;
402 }
403
404 for (i = 0; i < num_packets; i++) {
405 gint fr_size;
406
407 /* copy FT, clear F bit */
408 *p++ = payload[i] & 0x7f;
409
410 fr_size = frame_size[(payload[i] & 0x78) >> 3];
411 if (fr_size > 0) {
412 /* copy data packet, FIXME, calc CRC here. */
413 memcpy (p, dp, fr_size);
414
415 p += fr_size;
416 dp += fr_size;
417 }
418 }
419 gst_buffer_unmap (outbuf, &map);
420
421 /* we can set the duration because each packet is 20 milliseconds */
422 GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND;
423
424 if (gst_rtp_buffer_get_marker (rtp)) {
425 /* marker bit marks a buffer after a talkspurt. */
426 GST_DEBUG_OBJECT (depayload, "marker bit was set");
427 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
428 }
429
430 GST_DEBUG_OBJECT (depayload, "pushing buffer of size %" G_GSIZE_FORMAT,
431 gst_buffer_get_size (outbuf));
432
433 gst_rtp_copy_audio_meta (rtpamrdepay, outbuf, rtp->buffer);
434 }
435
436 return outbuf;
437
438 /* ERRORS */
439 too_small:
440 {
441 GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
442 (NULL), ("AMR RTP payload too small (%d)", payload_len));
443 goto bad_packet;
444 }
445 wrong_interleaving:
446 {
447 GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
448 (NULL), ("AMR RTP wrong interleaving"));
449 goto bad_packet;
450 }
451 wrong_framesize:
452 {
453 GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
454 (NULL), ("AMR RTP frame size == -1"));
455 goto bad_packet;
456 }
457 wrong_length_1:
458 {
459 GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
460 (NULL), ("AMR RTP wrong length 1"));
461 goto bad_packet;
462 }
463 wrong_length_2:
464 {
465 GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
466 (NULL), ("AMR RTP wrong length 2"));
467 goto bad_packet;
468 }
469 bad_packet:
470 {
471 /* no fatal error */
472 return NULL;
473 }
474 }
475