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  * @see_also: rtpamrpay
23  *
24  * Extract AMR audio from 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 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  * </refsect2>
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 "gstrtpamrdepay.h"
52 #include "gstrtputils.h"
53 
54 GST_DEBUG_CATEGORY_STATIC (rtpamrdepay_debug);
55 #define GST_CAT_DEFAULT (rtpamrdepay_debug)
56 
57 /* RtpAMRDepay signals and args */
58 enum
59 {
60   /* FILL ME */
61   LAST_SIGNAL
62 };
63 
64 enum
65 {
66   PROP_0
67 };
68 
69 /* input is an RTP packet
70  *
71  * params see RFC 3267, section 8.1
72  */
73 static GstStaticPadTemplate gst_rtp_amr_depay_sink_template =
74     GST_STATIC_PAD_TEMPLATE ("sink",
75     GST_PAD_SINK,
76     GST_PAD_ALWAYS,
77     GST_STATIC_CAPS ("application/x-rtp, "
78         "media = (string) \"audio\", "
79         "clock-rate = (int) 8000, " "encoding-name = (string) \"AMR\", "
80         /* This is the default, so the peer doesn't have to specify it
81          * "encoding-params = (string) \"1\", " */
82         /* NOTE that all values must be strings in orde to be able to do SDP <->
83          * GstCaps mapping. */
84         "octet-align = (string) \"1\";"
85         /* following options are not needed for a decoder
86          *
87          "crc = (string) { \"0\", \"1\" }, "
88          "robust-sorting = (string) \"0\", "
89          "interleaving = (string) \"0\";"
90          "mode-set = (int) [ 0, 7 ], "
91          "mode-change-period = (int) [ 1, MAX ], "
92          "mode-change-neighbor = (boolean) { TRUE, FALSE }, "
93          "maxptime = (int) [ 20, MAX ], "
94          "ptime = (int) [ 20, MAX ]"
95          */
96         "application/x-rtp, "
97         "media = (string) \"audio\", "
98         "clock-rate = (int) 16000, " "encoding-name = (string) \"AMR-WB\", "
99         /* This is the default, so the peer doesn't have to specify it
100          * "encoding-params = (string) \"1\", " */
101         /* NOTE that all values must be strings in orde to be able to do SDP <->
102          * GstCaps mapping. */
103         "octet-align = (string) \"1\";"
104         /* following options are not needed for a decoder
105          *
106          "crc = (string) { \"0\", \"1\" }, "
107          "robust-sorting = (string) \"0\", "
108          "interleaving = (string) \"0\""
109          "mode-set = (int) [ 0, 7 ], "
110          "mode-change-period = (int) [ 1, MAX ], "
111          "mode-change-neighbor = (boolean) { TRUE, FALSE }, "
112          "maxptime = (int) [ 20, MAX ], "
113          "ptime = (int) [ 20, MAX ]"
114          */
115     )
116     );
117 
118 static GstStaticPadTemplate gst_rtp_amr_depay_src_template =
119     GST_STATIC_PAD_TEMPLATE ("src",
120     GST_PAD_SRC,
121     GST_PAD_ALWAYS,
122     GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000;"
123         "audio/AMR-WB, " "channels = (int) 1," "rate = (int) 16000")
124     );
125 
126 static gboolean gst_rtp_amr_depay_setcaps (GstRTPBaseDepayload * depayload,
127     GstCaps * caps);
128 static GstBuffer *gst_rtp_amr_depay_process (GstRTPBaseDepayload * depayload,
129     GstRTPBuffer * rtp);
130 
131 #define gst_rtp_amr_depay_parent_class parent_class
132 G_DEFINE_TYPE (GstRtpAMRDepay, gst_rtp_amr_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
133 
134 static void
gst_rtp_amr_depay_class_init(GstRtpAMRDepayClass * klass)135 gst_rtp_amr_depay_class_init (GstRtpAMRDepayClass * klass)
136 {
137   GstElementClass *gstelement_class;
138   GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
139 
140   gstelement_class = (GstElementClass *) klass;
141   gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
142 
143   gst_element_class_add_static_pad_template (gstelement_class,
144       &gst_rtp_amr_depay_src_template);
145   gst_element_class_add_static_pad_template (gstelement_class,
146       &gst_rtp_amr_depay_sink_template);
147 
148   gst_element_class_set_static_metadata (gstelement_class,
149       "RTP AMR depayloader", "Codec/Depayloader/Network/RTP",
150       "Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)",
151       "Wim Taymans <wim.taymans@gmail.com>");
152 
153   gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_amr_depay_process;
154   gstrtpbasedepayload_class->set_caps = gst_rtp_amr_depay_setcaps;
155 
156   GST_DEBUG_CATEGORY_INIT (rtpamrdepay_debug, "rtpamrdepay", 0,
157       "AMR/AMR-WB RTP Depayloader");
158 }
159 
160 static void
gst_rtp_amr_depay_init(GstRtpAMRDepay * rtpamrdepay)161 gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay)
162 {
163   GstRTPBaseDepayload *depayload;
164 
165   depayload = GST_RTP_BASE_DEPAYLOAD (rtpamrdepay);
166 
167   gst_pad_use_fixed_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload));
168 }
169 
170 static gboolean
gst_rtp_amr_depay_setcaps(GstRTPBaseDepayload * depayload,GstCaps * caps)171 gst_rtp_amr_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
172 {
173   GstStructure *structure;
174   GstCaps *srccaps;
175   GstRtpAMRDepay *rtpamrdepay;
176   const gchar *params;
177   const gchar *str, *type;
178   gint clock_rate, need_clock_rate;
179   gboolean res;
180 
181   rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
182 
183   structure = gst_caps_get_structure (caps, 0);
184 
185   /* figure out the mode first and set the clock rates */
186   if ((str = gst_structure_get_string (structure, "encoding-name"))) {
187     if (strcmp (str, "AMR") == 0) {
188       rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_NB;
189       need_clock_rate = 8000;
190       type = "audio/AMR";
191     } else if (strcmp (str, "AMR-WB") == 0) {
192       rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_WB;
193       need_clock_rate = 16000;
194       type = "audio/AMR-WB";
195     } else
196       goto invalid_mode;
197   } else
198     goto invalid_mode;
199 
200   if (!(str = gst_structure_get_string (structure, "octet-align")))
201     rtpamrdepay->octet_align = FALSE;
202   else
203     rtpamrdepay->octet_align = (atoi (str) == 1);
204 
205   if (!(str = gst_structure_get_string (structure, "crc")))
206     rtpamrdepay->crc = FALSE;
207   else
208     rtpamrdepay->crc = (atoi (str) == 1);
209 
210   if (rtpamrdepay->crc) {
211     /* crc mode implies octet aligned mode */
212     rtpamrdepay->octet_align = TRUE;
213   }
214 
215   if (!(str = gst_structure_get_string (structure, "robust-sorting")))
216     rtpamrdepay->robust_sorting = FALSE;
217   else
218     rtpamrdepay->robust_sorting = (atoi (str) == 1);
219 
220   if (rtpamrdepay->robust_sorting) {
221     /* robust_sorting mode implies octet aligned mode */
222     rtpamrdepay->octet_align = TRUE;
223   }
224 
225   if (!(str = gst_structure_get_string (structure, "interleaving")))
226     rtpamrdepay->interleaving = FALSE;
227   else
228     rtpamrdepay->interleaving = (atoi (str) == 1);
229 
230   if (rtpamrdepay->interleaving) {
231     /* interleaving mode implies octet aligned mode */
232     rtpamrdepay->octet_align = TRUE;
233   }
234 
235   if (!(params = gst_structure_get_string (structure, "encoding-params")))
236     rtpamrdepay->channels = 1;
237   else {
238     rtpamrdepay->channels = atoi (params);
239   }
240 
241   if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
242     clock_rate = need_clock_rate;
243   depayload->clock_rate = clock_rate;
244 
245   /* we require 1 channel, 8000 Hz, octet aligned, no CRC,
246    * no robust sorting, no interleaving for now */
247   if (rtpamrdepay->channels != 1)
248     return FALSE;
249   if (clock_rate != need_clock_rate)
250     return FALSE;
251   if (rtpamrdepay->octet_align != TRUE)
252     return FALSE;
253   if (rtpamrdepay->robust_sorting != FALSE)
254     return FALSE;
255   if (rtpamrdepay->interleaving != FALSE)
256     return FALSE;
257 
258   srccaps = gst_caps_new_simple (type,
259       "channels", G_TYPE_INT, rtpamrdepay->channels,
260       "rate", G_TYPE_INT, clock_rate, NULL);
261   res = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
262   gst_caps_unref (srccaps);
263 
264   return res;
265 
266   /* ERRORS */
267 invalid_mode:
268   {
269     GST_ERROR_OBJECT (rtpamrdepay, "invalid encoding-name");
270     return FALSE;
271   }
272 }
273 
274 /* -1 is invalid */
275 static const gint nb_frame_size[16] = {
276   12, 13, 15, 17, 19, 20, 26, 31,
277   5, -1, -1, -1, -1, -1, -1, 0
278 };
279 
280 static const gint wb_frame_size[16] = {
281   17, 23, 32, 36, 40, 46, 50, 58,
282   60, 5, -1, -1, -1, -1, -1, 0
283 };
284 
285 static GstBuffer *
gst_rtp_amr_depay_process(GstRTPBaseDepayload * depayload,GstRTPBuffer * rtp)286 gst_rtp_amr_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
287 {
288   GstRtpAMRDepay *rtpamrdepay;
289   const gint *frame_size;
290   GstBuffer *outbuf = NULL;
291   gint payload_len;
292   GstMapInfo map;
293 
294   rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
295 
296   /* setup frame size pointer */
297   if (rtpamrdepay->mode == GST_RTP_AMR_DP_MODE_NB)
298     frame_size = nb_frame_size;
299   else
300     frame_size = wb_frame_size;
301 
302   /* when we get here, 1 channel, 8000/16000 Hz, octet aligned, no CRC,
303    * no robust sorting, no interleaving data is to be depayloaded */
304   {
305     guint8 *payload, *p, *dp;
306     gint i, num_packets, num_nonempty_packets;
307     gint amr_len;
308     gint ILL, ILP;
309 
310     payload_len = gst_rtp_buffer_get_payload_len (rtp);
311 
312     /* need at least 2 bytes for the header */
313     if (payload_len < 2)
314       goto too_small;
315 
316     payload = gst_rtp_buffer_get_payload (rtp);
317 
318     /* depay CMR. The CMR is used by the sender to request
319      * a new encoding mode.
320      *
321      *  0 1 2 3 4 5 6 7
322      * +-+-+-+-+-+-+-+-+
323      * | CMR   |R|R|R|R|
324      * +-+-+-+-+-+-+-+-+
325      */
326     /* CMR = (payload[0] & 0xf0) >> 4; */
327 
328     /* strip CMR header now, pack FT and the data for the decoder */
329     payload_len -= 1;
330     payload += 1;
331 
332     GST_DEBUG_OBJECT (rtpamrdepay, "payload len %d", payload_len);
333 
334     if (rtpamrdepay->interleaving) {
335       ILL = (payload[0] & 0xf0) >> 4;
336       ILP = (payload[0] & 0x0f);
337 
338       payload_len -= 1;
339       payload += 1;
340 
341       if (ILP > ILL)
342         goto wrong_interleaving;
343     }
344 
345     /*
346      *  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
347      * +-+-+-+-+-+-+-+-+..
348      * |F|  FT   |Q|P|P| more FT..
349      * +-+-+-+-+-+-+-+-+..
350      */
351     /* count number of packets by counting the FTs. Also
352      * count number of amr data bytes and number of non-empty
353      * packets (this is also the number of CRCs if present). */
354     amr_len = 0;
355     num_nonempty_packets = 0;
356     num_packets = 0;
357     for (i = 0; i < payload_len; i++) {
358       gint fr_size;
359       guint8 FT;
360 
361       FT = (payload[i] & 0x78) >> 3;
362 
363       fr_size = frame_size[FT];
364       GST_DEBUG_OBJECT (rtpamrdepay, "frame size %d", fr_size);
365       if (fr_size == -1)
366         goto wrong_framesize;
367 
368       if (fr_size > 0) {
369         amr_len += fr_size;
370         num_nonempty_packets++;
371       }
372       num_packets++;
373 
374       if ((payload[i] & 0x80) == 0)
375         break;
376     }
377 
378     if (rtpamrdepay->crc) {
379       /* data len + CRC len + header bytes should be smaller than payload_len */
380       if (num_packets + num_nonempty_packets + amr_len > payload_len)
381         goto wrong_length_1;
382     } else {
383       /* data len + header bytes should be smaller than payload_len */
384       if (num_packets + amr_len > payload_len)
385         goto wrong_length_2;
386     }
387 
388     outbuf = gst_buffer_new_and_alloc (payload_len);
389 
390     /* point to destination */
391     gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
392 
393     /* point to first data packet */
394     p = map.data;
395     dp = payload + num_packets;
396     if (rtpamrdepay->crc) {
397       /* skip CRC if present */
398       dp += num_nonempty_packets;
399     }
400 
401     for (i = 0; i < num_packets; i++) {
402       gint fr_size;
403 
404       /* copy FT, clear F bit */
405       *p++ = payload[i] & 0x7f;
406 
407       fr_size = frame_size[(payload[i] & 0x78) >> 3];
408       if (fr_size > 0) {
409         /* copy data packet, FIXME, calc CRC here. */
410         memcpy (p, dp, fr_size);
411 
412         p += fr_size;
413         dp += fr_size;
414       }
415     }
416     gst_buffer_unmap (outbuf, &map);
417 
418     /* we can set the duration because each packet is 20 milliseconds */
419     GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND;
420 
421     if (gst_rtp_buffer_get_marker (rtp)) {
422       /* marker bit marks a buffer after a talkspurt. */
423       GST_DEBUG_OBJECT (depayload, "marker bit was set");
424       GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
425     }
426 
427     GST_DEBUG_OBJECT (depayload, "pushing buffer of size %" G_GSIZE_FORMAT,
428         gst_buffer_get_size (outbuf));
429 
430     gst_rtp_copy_audio_meta (rtpamrdepay, outbuf, rtp->buffer);
431   }
432 
433   return outbuf;
434 
435   /* ERRORS */
436 too_small:
437   {
438     GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
439         (NULL), ("AMR RTP payload too small (%d)", payload_len));
440     goto bad_packet;
441   }
442 wrong_interleaving:
443   {
444     GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
445         (NULL), ("AMR RTP wrong interleaving"));
446     goto bad_packet;
447   }
448 wrong_framesize:
449   {
450     GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
451         (NULL), ("AMR RTP frame size == -1"));
452     goto bad_packet;
453   }
454 wrong_length_1:
455   {
456     GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
457         (NULL), ("AMR RTP wrong length 1"));
458     goto bad_packet;
459   }
460 wrong_length_2:
461   {
462     GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
463         (NULL), ("AMR RTP wrong length 2"));
464     goto bad_packet;
465   }
466 bad_packet:
467   {
468     /* no fatal error */
469     return NULL;
470   }
471 }
472 
473 gboolean
gst_rtp_amr_depay_plugin_init(GstPlugin * plugin)474 gst_rtp_amr_depay_plugin_init (GstPlugin * plugin)
475 {
476   return gst_element_register (plugin, "rtpamrdepay",
477       GST_RANK_SECONDARY, GST_TYPE_RTP_AMR_DEPAY);
478 }
479