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-rtpamrpay
22 * @see_also: rtpamrdepay
23 *
24 * Payload AMR audio into RTP packets according to RFC 3267.
25 * For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.txt
26 *
27 * <refsect2>
28 * <title>Example pipeline</title>
29 * |[
30 * gst-launch-1.0 -v audiotestsrc ! amrnbenc ! rtpamrpay ! udpsink
31 * ]| This example pipeline will encode and payload an AMR stream. Refer to
32 * the rtpamrdepay example to depayload and decode the RTP stream.
33 * </refsect2>
34 */
35
36 /* references:
37 *
38 * RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
39 * Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive
40 * Multi-Rate Wideband (AMR-WB) Audio Codecs.
41 *
42 * ETSI TS 126 201 V6.0.0 (2004-12) - Digital cellular telecommunications system (Phase 2+);
43 * Universal Mobile Telecommunications System (UMTS);
44 * AMR speech codec, wideband;
45 * Frame structure
46 * (3GPP TS 26.201 version 6.0.0 Release 6)
47 */
48
49 #ifdef HAVE_CONFIG_H
50 # include "config.h"
51 #endif
52
53 #include <string.h>
54
55 #include <gst/rtp/gstrtpbuffer.h>
56 #include <gst/audio/audio.h>
57
58 #include "gstrtpamrpay.h"
59 #include "gstrtputils.h"
60
61 GST_DEBUG_CATEGORY_STATIC (rtpamrpay_debug);
62 #define GST_CAT_DEFAULT (rtpamrpay_debug)
63
64 static GstStaticPadTemplate gst_rtp_amr_pay_sink_template =
65 GST_STATIC_PAD_TEMPLATE ("sink",
66 GST_PAD_SINK,
67 GST_PAD_ALWAYS,
68 GST_STATIC_CAPS ("audio/AMR, channels=(int)1, rate=(int)8000; "
69 "audio/AMR-WB, channels=(int)1, rate=(int)16000")
70 );
71
72 static GstStaticPadTemplate gst_rtp_amr_pay_src_template =
73 GST_STATIC_PAD_TEMPLATE ("src",
74 GST_PAD_SRC,
75 GST_PAD_ALWAYS,
76 GST_STATIC_CAPS ("application/x-rtp, "
77 "media = (string) \"audio\", "
78 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
79 "clock-rate = (int) 8000, "
80 "encoding-name = (string) \"AMR\", "
81 "encoding-params = (string) \"1\", "
82 "octet-align = (string) \"1\", "
83 "crc = (string) \"0\", "
84 "robust-sorting = (string) \"0\", "
85 "interleaving = (string) \"0\", "
86 "mode-set = (int) [ 0, 7 ], "
87 "mode-change-period = (int) [ 1, MAX ], "
88 "mode-change-neighbor = (string) { \"0\", \"1\" }, "
89 "maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ];"
90 "application/x-rtp, "
91 "media = (string) \"audio\", "
92 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
93 "clock-rate = (int) 16000, "
94 "encoding-name = (string) \"AMR-WB\", "
95 "encoding-params = (string) \"1\", "
96 "octet-align = (string) \"1\", "
97 "crc = (string) \"0\", "
98 "robust-sorting = (string) \"0\", "
99 "interleaving = (string) \"0\", "
100 "mode-set = (int) [ 0, 7 ], "
101 "mode-change-period = (int) [ 1, MAX ], "
102 "mode-change-neighbor = (string) { \"0\", \"1\" }, "
103 "maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ]")
104 );
105
106 static gboolean gst_rtp_amr_pay_setcaps (GstRTPBasePayload * basepayload,
107 GstCaps * caps);
108 static GstFlowReturn gst_rtp_amr_pay_handle_buffer (GstRTPBasePayload * pad,
109 GstBuffer * buffer);
110
111 static GstStateChangeReturn
112 gst_rtp_amr_pay_change_state (GstElement * element, GstStateChange transition);
113
114 #define gst_rtp_amr_pay_parent_class parent_class
115 G_DEFINE_TYPE (GstRtpAMRPay, gst_rtp_amr_pay, GST_TYPE_RTP_BASE_PAYLOAD);
116
117 static void
gst_rtp_amr_pay_class_init(GstRtpAMRPayClass * klass)118 gst_rtp_amr_pay_class_init (GstRtpAMRPayClass * klass)
119 {
120 GstElementClass *gstelement_class;
121 GstRTPBasePayloadClass *gstrtpbasepayload_class;
122
123 gstelement_class = (GstElementClass *) klass;
124 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
125
126 gstelement_class->change_state = gst_rtp_amr_pay_change_state;
127
128 gst_element_class_add_static_pad_template (gstelement_class,
129 &gst_rtp_amr_pay_src_template);
130 gst_element_class_add_static_pad_template (gstelement_class,
131 &gst_rtp_amr_pay_sink_template);
132
133 gst_element_class_set_static_metadata (gstelement_class, "RTP AMR payloader",
134 "Codec/Payloader/Network/RTP",
135 "Payload-encode AMR or AMR-WB audio into RTP packets (RFC 3267)",
136 "Wim Taymans <wim.taymans@gmail.com>");
137
138 gstrtpbasepayload_class->set_caps = gst_rtp_amr_pay_setcaps;
139 gstrtpbasepayload_class->handle_buffer = gst_rtp_amr_pay_handle_buffer;
140
141 GST_DEBUG_CATEGORY_INIT (rtpamrpay_debug, "rtpamrpay", 0,
142 "AMR/AMR-WB RTP Payloader");
143 }
144
145 static void
gst_rtp_amr_pay_init(GstRtpAMRPay * rtpamrpay)146 gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay)
147 {
148 }
149
150 static void
gst_rtp_amr_pay_reset(GstRtpAMRPay * pay)151 gst_rtp_amr_pay_reset (GstRtpAMRPay * pay)
152 {
153 pay->next_rtp_time = 0;
154 pay->first_ts = GST_CLOCK_TIME_NONE;
155 pay->first_rtp_time = 0;
156 }
157
158 static gboolean
gst_rtp_amr_pay_setcaps(GstRTPBasePayload * basepayload,GstCaps * caps)159 gst_rtp_amr_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
160 {
161 GstRtpAMRPay *rtpamrpay;
162 gboolean res;
163 const GstStructure *s;
164 const gchar *str;
165
166 rtpamrpay = GST_RTP_AMR_PAY (basepayload);
167
168 /* figure out the mode Narrow or Wideband */
169 s = gst_caps_get_structure (caps, 0);
170 if ((str = gst_structure_get_name (s))) {
171 if (strcmp (str, "audio/AMR") == 0)
172 rtpamrpay->mode = GST_RTP_AMR_P_MODE_NB;
173 else if (strcmp (str, "audio/AMR-WB") == 0)
174 rtpamrpay->mode = GST_RTP_AMR_P_MODE_WB;
175 else
176 goto wrong_type;
177 } else
178 goto wrong_type;
179
180 if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB)
181 gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
182 else
183 gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "AMR-WB",
184 16000);
185
186 res = gst_rtp_base_payload_set_outcaps (basepayload,
187 "encoding-params", G_TYPE_STRING, "1", "octet-align", G_TYPE_STRING, "1",
188 /* don't set the defaults
189 *
190 * "crc", G_TYPE_STRING, "0",
191 * "robust-sorting", G_TYPE_STRING, "0",
192 * "interleaving", G_TYPE_STRING, "0",
193 */
194 NULL);
195
196 return res;
197
198 /* ERRORS */
199 wrong_type:
200 {
201 GST_ERROR_OBJECT (rtpamrpay, "unsupported media type '%s'",
202 GST_STR_NULL (str));
203 return FALSE;
204 }
205 }
206
207 static void
gst_rtp_amr_pay_recalc_rtp_time(GstRtpAMRPay * rtpamrpay,GstClockTime timestamp)208 gst_rtp_amr_pay_recalc_rtp_time (GstRtpAMRPay * rtpamrpay,
209 GstClockTime timestamp)
210 {
211 /* re-sync rtp time */
212 if (GST_CLOCK_TIME_IS_VALID (rtpamrpay->first_ts) &&
213 GST_CLOCK_TIME_IS_VALID (timestamp) && timestamp >= rtpamrpay->first_ts) {
214 GstClockTime diff;
215 guint32 rtpdiff;
216
217 /* interpolate to reproduce gap from start, rather than intermediate
218 * intervals to avoid roundup accumulation errors */
219 diff = timestamp - rtpamrpay->first_ts;
220 rtpdiff = ((diff / GST_MSECOND) * 8) <<
221 (rtpamrpay->mode == GST_RTP_AMR_P_MODE_WB);
222 rtpamrpay->next_rtp_time = rtpamrpay->first_rtp_time + rtpdiff;
223 GST_DEBUG_OBJECT (rtpamrpay,
224 "elapsed time %" GST_TIME_FORMAT ", rtp %" G_GUINT32_FORMAT ", "
225 "new offset %" G_GUINT32_FORMAT, GST_TIME_ARGS (diff), rtpdiff,
226 rtpamrpay->next_rtp_time);
227 }
228 }
229
230 /* -1 is invalid */
231 static const gint nb_frame_size[16] = {
232 12, 13, 15, 17, 19, 20, 26, 31,
233 5, -1, -1, -1, -1, -1, -1, 0
234 };
235
236 static const gint wb_frame_size[16] = {
237 17, 23, 32, 36, 40, 46, 50, 58,
238 60, 5, -1, -1, -1, -1, -1, 0
239 };
240
241 static GstFlowReturn
gst_rtp_amr_pay_handle_buffer(GstRTPBasePayload * basepayload,GstBuffer * buffer)242 gst_rtp_amr_pay_handle_buffer (GstRTPBasePayload * basepayload,
243 GstBuffer * buffer)
244 {
245 GstRtpAMRPay *rtpamrpay;
246 const gint *frame_size;
247 GstFlowReturn ret;
248 guint payload_len;
249 GstMapInfo map;
250 GstBuffer *outbuf;
251 guint8 *payload, *ptr, *payload_amr;
252 GstClockTime timestamp, duration;
253 guint packet_len, mtu;
254 gint i, num_packets, num_nonempty_packets;
255 gint amr_len;
256 gboolean sid = FALSE;
257 GstRTPBuffer rtp = { NULL };
258
259 rtpamrpay = GST_RTP_AMR_PAY (basepayload);
260 mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpamrpay);
261
262 gst_buffer_map (buffer, &map, GST_MAP_READ);
263
264 timestamp = GST_BUFFER_PTS (buffer);
265 duration = GST_BUFFER_DURATION (buffer);
266
267 /* setup frame size pointer */
268 if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB)
269 frame_size = nb_frame_size;
270 else
271 frame_size = wb_frame_size;
272
273 GST_DEBUG_OBJECT (basepayload, "got %" G_GSIZE_FORMAT " bytes", map.size);
274
275 /* FIXME, only
276 * octet aligned, no interleaving, single channel, no CRC,
277 * no robust-sorting. To fix this you need to implement the downstream
278 * negotiation function. */
279
280 /* first count number of packets and total amr frame size */
281 amr_len = num_packets = num_nonempty_packets = 0;
282 for (i = 0; i < map.size; i++) {
283 guint8 FT;
284 gint fr_size;
285
286 FT = (map.data[i] & 0x78) >> 3;
287
288 fr_size = frame_size[FT];
289 GST_DEBUG_OBJECT (basepayload, "frame type %d, frame size %d", FT, fr_size);
290 /* FIXME, we don't handle this yet.. */
291 if (fr_size <= 0)
292 goto wrong_size;
293
294 if (fr_size == 5)
295 sid = TRUE;
296
297 amr_len += fr_size;
298 num_nonempty_packets++;
299 num_packets++;
300 i += fr_size;
301 }
302 if (amr_len > map.size)
303 goto incomplete_frame;
304
305 /* we need one extra byte for the CMR, the ToC is in the input
306 * data */
307 payload_len = map.size + 1;
308
309 /* get packet len to check against MTU */
310 packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
311 if (packet_len > mtu)
312 goto too_big;
313
314 /* now alloc output buffer */
315 outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
316
317 gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
318
319 /* copy timestamp */
320 GST_BUFFER_PTS (outbuf) = timestamp;
321
322 if (duration != GST_CLOCK_TIME_NONE)
323 GST_BUFFER_DURATION (outbuf) = duration;
324 else {
325 GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND;
326 }
327
328 if (GST_BUFFER_IS_DISCONT (buffer)) {
329 GST_DEBUG_OBJECT (basepayload, "discont, setting marker bit");
330 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
331 gst_rtp_buffer_set_marker (&rtp, TRUE);
332 gst_rtp_amr_pay_recalc_rtp_time (rtpamrpay, timestamp);
333 }
334
335 if (G_UNLIKELY (sid)) {
336 gst_rtp_amr_pay_recalc_rtp_time (rtpamrpay, timestamp);
337 }
338
339 /* perfect rtptime */
340 if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (rtpamrpay->first_ts))) {
341 rtpamrpay->first_ts = timestamp;
342 rtpamrpay->first_rtp_time = rtpamrpay->next_rtp_time;
343 }
344 GST_BUFFER_OFFSET (outbuf) = rtpamrpay->next_rtp_time;
345 rtpamrpay->next_rtp_time +=
346 (num_packets * 160) << (rtpamrpay->mode == GST_RTP_AMR_P_MODE_WB);
347
348 /* get payload, this is now writable */
349 payload = gst_rtp_buffer_get_payload (&rtp);
350
351 /* 0 1 2 3 4 5 6 7
352 * +-+-+-+-+-+-+-+-+
353 * | CMR |R|R|R|R|
354 * +-+-+-+-+-+-+-+-+
355 */
356 payload[0] = 0xF0; /* CMR, no specific mode requested */
357
358 /* this is where we copy the AMR data, after num_packets FTs and the
359 * CMR. */
360 payload_amr = payload + num_packets + 1;
361
362 /* copy data in payload, first we copy all the FTs then all
363 * the AMR data. The last FT has to have the F flag cleared. */
364 ptr = map.data;
365 for (i = 1; i <= num_packets; i++) {
366 guint8 FT;
367 gint fr_size;
368
369 /* 0 1 2 3 4 5 6 7
370 * +-+-+-+-+-+-+-+-+
371 * |F| FT |Q|P|P| more FT...
372 * +-+-+-+-+-+-+-+-+
373 */
374 FT = (*ptr & 0x78) >> 3;
375
376 fr_size = frame_size[FT];
377
378 if (i == num_packets)
379 /* last packet, clear F flag */
380 payload[i] = *ptr & 0x7f;
381 else
382 /* set F flag */
383 payload[i] = *ptr | 0x80;
384
385 memcpy (payload_amr, &ptr[1], fr_size);
386
387 /* all sizes are > 0 since we checked for that above */
388 ptr += fr_size + 1;
389 payload_amr += fr_size;
390 }
391
392 gst_buffer_unmap (buffer, &map);
393 gst_rtp_buffer_unmap (&rtp);
394
395 gst_rtp_copy_audio_meta (rtpamrpay, outbuf, buffer);
396
397 gst_buffer_unref (buffer);
398
399 ret = gst_rtp_base_payload_push (basepayload, outbuf);
400
401 return ret;
402
403 /* ERRORS */
404 wrong_size:
405 {
406 GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
407 (NULL), ("received AMR frame with size <= 0"));
408 gst_buffer_unmap (buffer, &map);
409 gst_buffer_unref (buffer);
410
411 return GST_FLOW_ERROR;
412 }
413 incomplete_frame:
414 {
415 GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
416 (NULL), ("received incomplete AMR frames"));
417 gst_buffer_unmap (buffer, &map);
418 gst_buffer_unref (buffer);
419
420 return GST_FLOW_ERROR;
421 }
422 too_big:
423 {
424 GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
425 (NULL), ("received too many AMR frames for MTU"));
426 gst_buffer_unmap (buffer, &map);
427 gst_buffer_unref (buffer);
428
429 return GST_FLOW_ERROR;
430 }
431 }
432
433 static GstStateChangeReturn
gst_rtp_amr_pay_change_state(GstElement * element,GstStateChange transition)434 gst_rtp_amr_pay_change_state (GstElement * element, GstStateChange transition)
435 {
436 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
437
438 /* handle upwards state changes here */
439 switch (transition) {
440 default:
441 break;
442 }
443
444 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
445
446 /* handle downwards state changes */
447 switch (transition) {
448 case GST_STATE_CHANGE_PAUSED_TO_READY:
449 gst_rtp_amr_pay_reset (GST_RTP_AMR_PAY (element));
450 break;
451 default:
452 break;
453 }
454
455 return ret;
456 }
457
458 gboolean
gst_rtp_amr_pay_plugin_init(GstPlugin * plugin)459 gst_rtp_amr_pay_plugin_init (GstPlugin * plugin)
460 {
461 return gst_element_register (plugin, "rtpamrpay",
462 GST_RANK_SECONDARY, GST_TYPE_RTP_AMR_PAY);
463 }
464