2 * Copyright (C) <2005> Wim Taymans <wim.taymans@gmail.com>
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.
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.
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., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
24 #include <gst/rtp/gstrtpbuffer.h>
28 #include "gstrtpamrdepay.h"
30 GST_DEBUG_CATEGORY_STATIC (rtpamrdepay_debug);
31 #define GST_CAT_DEFAULT (rtpamrdepay_debug)
35 * RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
36 * Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
37 * Wideband (AMR-WB) Audio Codecs.
40 /* elementfactory information */
41 static const GstElementDetails gst_rtp_amrdepay_details =
42 GST_ELEMENT_DETAILS ("RTP AMR depayloader",
43 "Codec/Depayloader/Network",
44 "Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)",
45 "Wim Taymans <wim.taymans@gmail.com>");
47 /* RtpAMRDepay signals and args */
59 /* input is an RTP packet
61 * params see RFC 3267, section 8.1
63 static GstStaticPadTemplate gst_rtp_amr_depay_sink_template =
64 GST_STATIC_PAD_TEMPLATE ("sink",
67 GST_STATIC_CAPS ("application/x-rtp, "
68 "media = (string) \"audio\", "
69 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
70 "clock-rate = (int) 8000, "
71 "encoding-name = (string) \"AMR\", "
72 "encoding-params = (string) \"1\", "
73 /* NOTE that all values must be strings in orde to be able to do SDP <->
75 "octet-align = (string) \"1\", "
76 "crc = (string) { \"0\", \"1\" }, "
77 "robust-sorting = (string) \"0\", " "interleaving = (string) \"0\";"
78 /* following options are not needed for a decoder
80 "mode-set = (int) [ 0, 7 ], "
81 "mode-change-period = (int) [ 1, MAX ], "
82 "mode-change-neighbor = (boolean) { TRUE, FALSE }, "
83 "maxptime = (int) [ 20, MAX ], "
84 "ptime = (int) [ 20, MAX ]"
87 "media = (string) \"audio\", "
88 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
89 "clock-rate = (int) 16000, "
90 "encoding-name = (string) \"AMR-WB\", "
91 "encoding-params = (string) \"1\", "
92 /* NOTE that all values must be strings in orde to be able to do SDP <->
94 "octet-align = (string) \"1\", "
95 "crc = (string) { \"0\", \"1\" }, "
96 "robust-sorting = (string) \"0\", " "interleaving = (string) \"0\""
97 /* following options are not needed for a decoder
99 "mode-set = (int) [ 0, 7 ], "
100 "mode-change-period = (int) [ 1, MAX ], "
101 "mode-change-neighbor = (boolean) { TRUE, FALSE }, "
102 "maxptime = (int) [ 20, MAX ], "
103 "ptime = (int) [ 20, MAX ]"
108 static GstStaticPadTemplate gst_rtp_amr_depay_src_template =
109 GST_STATIC_PAD_TEMPLATE ("src",
112 GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000;"
113 "audio/AMR-WB, " "channels = (int) 1," "rate = (int) 16000")
116 static gboolean gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload,
118 static GstBuffer *gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload,
121 GST_BOILERPLATE (GstRtpAMRDepay, gst_rtp_amr_depay, GstBaseRTPDepayload,
122 GST_TYPE_BASE_RTP_DEPAYLOAD);
125 gst_rtp_amr_depay_base_init (gpointer klass)
127 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
129 gst_element_class_add_pad_template (element_class,
130 gst_static_pad_template_get (&gst_rtp_amr_depay_src_template));
131 gst_element_class_add_pad_template (element_class,
132 gst_static_pad_template_get (&gst_rtp_amr_depay_sink_template));
134 gst_element_class_set_details (element_class, &gst_rtp_amrdepay_details);
138 gst_rtp_amr_depay_class_init (GstRtpAMRDepayClass * klass)
140 GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
142 gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
144 gstbasertpdepayload_class->process = gst_rtp_amr_depay_process;
145 gstbasertpdepayload_class->set_caps = gst_rtp_amr_depay_setcaps;
147 GST_DEBUG_CATEGORY_INIT (rtpamrdepay_debug, "rtpamrdepay", 0,
148 "AMR/AMR-WB RTP Depayloader");
152 gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay,
153 GstRtpAMRDepayClass * klass)
155 GstBaseRTPDepayload *depayload;
157 depayload = GST_BASE_RTP_DEPAYLOAD (rtpamrdepay);
159 gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
163 gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
165 GstStructure *structure;
167 GstRtpAMRDepay *rtpamrdepay;
169 const gchar *str, *type;
170 gint clock_rate, need_clock_rate;
173 rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
175 structure = gst_caps_get_structure (caps, 0);
177 /* figure out the mode first and set the clock rates */
178 if ((str = gst_structure_get_string (structure, "encoding-name"))) {
179 if (strcmp (str, "AMR") == 0) {
180 rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_NB;
181 need_clock_rate = 8000;
183 } else if (strcmp (str, "AMR-WB") == 0) {
184 rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_WB;
185 need_clock_rate = 16000;
186 type = "audio/AMR-WB";
192 if (!(str = gst_structure_get_string (structure, "octet-align")))
193 rtpamrdepay->octet_align = FALSE;
195 rtpamrdepay->octet_align = (atoi (str) == 1);
197 if (!(str = gst_structure_get_string (structure, "crc")))
198 rtpamrdepay->crc = FALSE;
200 rtpamrdepay->crc = (atoi (str) == 1);
202 if (rtpamrdepay->crc) {
203 /* crc mode implies octet aligned mode */
204 rtpamrdepay->octet_align = TRUE;
207 if (!(str = gst_structure_get_string (structure, "robust-sorting")))
208 rtpamrdepay->robust_sorting = FALSE;
210 rtpamrdepay->robust_sorting = (atoi (str) == 1);
212 if (rtpamrdepay->robust_sorting) {
213 /* robust_sorting mode implies octet aligned mode */
214 rtpamrdepay->octet_align = TRUE;
217 if (!(str = gst_structure_get_string (structure, "interleaving")))
218 rtpamrdepay->interleaving = FALSE;
220 rtpamrdepay->interleaving = (atoi (str) == 1);
222 if (rtpamrdepay->interleaving) {
223 /* interleaving mode implies octet aligned mode */
224 rtpamrdepay->octet_align = TRUE;
227 if (!(params = gst_structure_get_string (structure, "encoding-params")))
228 rtpamrdepay->channels = 1;
230 rtpamrdepay->channels = atoi (params);
233 if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
234 clock_rate = need_clock_rate;
235 depayload->clock_rate = clock_rate;
237 /* we require 1 channel, 8000 Hz, octet aligned, no CRC,
238 * no robust sorting, no interleaving for now */
239 if (rtpamrdepay->channels != 1)
241 if (clock_rate != need_clock_rate)
243 if (rtpamrdepay->octet_align != TRUE)
245 if (rtpamrdepay->robust_sorting != FALSE)
247 if (rtpamrdepay->interleaving != FALSE)
250 srccaps = gst_caps_new_simple (type,
251 "channels", G_TYPE_INT, rtpamrdepay->channels,
252 "rate", G_TYPE_INT, clock_rate, NULL);
253 res = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
254 gst_caps_unref (srccaps);
261 GST_ERROR_OBJECT (rtpamrdepay, "invalid encoding-name");
267 static gint nb_frame_size[16] = {
268 12, 13, 15, 17, 19, 20, 26, 31,
269 5, -1, -1, -1, -1, -1, -1, 0
272 static gint wb_frame_size[16] = {
273 17, 23, 32, 36, 40, 46, 50, 58,
274 60, -1, -1, -1, -1, -1, -1, 0
278 gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
280 GstRtpAMRDepay *rtpamrdepay;
281 GstBuffer *outbuf = NULL;
285 rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
287 /* setup frame size pointer */
288 if (rtpamrdepay->mode == GST_RTP_AMR_DP_MODE_NB)
289 frame_size = nb_frame_size;
291 frame_size = wb_frame_size;
293 /* when we get here, 1 channel, 8000/16000 Hz, octet aligned, no CRC,
294 * no robust sorting, no interleaving data is to be depayloaded */
296 guint8 *payload, *p, *dp;
298 gint i, num_packets, num_nonempty_packets;
302 payload_len = gst_rtp_buffer_get_payload_len (buf);
304 /* need at least 2 bytes for the header */
308 payload = gst_rtp_buffer_get_payload (buf);
310 /* depay CMR. The CMR is used by the sender to request
311 * a new encoding mode.
318 CMR = (payload[0] & 0xf0) >> 4;
320 /* strip CMR header now, pack FT and the data for the decoder */
324 GST_DEBUG_OBJECT (rtpamrdepay, "payload len %d", payload_len);
326 if (rtpamrdepay->interleaving) {
327 ILL = (payload[0] & 0xf0) >> 4;
328 ILP = (payload[0] & 0x0f);
334 goto wrong_interleaving;
338 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
339 * +-+-+-+-+-+-+-+-+..
340 * |F| FT |Q|P|P| more FT..
341 * +-+-+-+-+-+-+-+-+..
343 /* count number of packets by counting the FTs. Also
344 * count number of amr data bytes and number of non-empty
345 * packets (this is also the number of CRCs if present). */
347 num_nonempty_packets = 0;
349 for (i = 0; i < payload_len; i++) {
353 FT = (payload[i] & 0x78) >> 3;
355 fr_size = frame_size[FT];
356 GST_DEBUG_OBJECT (rtpamrdepay, "frame size %d", fr_size);
358 goto wrong_framesize;
362 num_nonempty_packets++;
366 if ((payload[i] & 0x80) == 0)
370 if (rtpamrdepay->crc) {
371 /* data len + CRC len + header bytes should be smaller than payload_len */
372 if (num_packets + num_nonempty_packets + amr_len > payload_len)
375 /* data len + header bytes should be smaller than payload_len */
376 if (num_packets + amr_len > payload_len)
380 outbuf = gst_buffer_new_and_alloc (payload_len);
382 /* point to destination */
383 p = GST_BUFFER_DATA (outbuf);
384 /* point to first data packet */
385 dp = payload + num_packets;
386 if (rtpamrdepay->crc) {
387 /* skip CRC if present */
388 dp += num_nonempty_packets;
391 for (i = 0; i < num_packets; i++) {
394 /* copy FT, clear F bit */
395 *p++ = payload[i] & 0x7f;
397 fr_size = frame_size[(payload[i] & 0x78) >> 3];
399 /* copy data packet, FIXME, calc CRC here. */
400 memcpy (p, dp, fr_size);
406 /* we can set the duration because each packet is 20 milliseconds */
407 GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND;
409 if (gst_rtp_buffer_get_marker (buf)) {
410 /* marker bit marks a discont buffer after a talkspurt. */
411 GST_DEBUG_OBJECT (depayload, "marker bit was set");
412 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
415 GST_DEBUG_OBJECT (depayload, "pushing buffer of size %d",
416 GST_BUFFER_SIZE (outbuf));
423 GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
424 (NULL), ("AMR RTP payload too small (%d)", payload_len));
429 GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
430 (NULL), ("AMR RTP wrong interleaving"));
435 GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
436 (NULL), ("AMR RTP frame size == -1"));
441 GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
442 (NULL), ("AMR RTP wrong length 1"));
447 GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
448 (NULL), ("AMR RTP wrong length 2"));
459 gst_rtp_amr_depay_plugin_init (GstPlugin * plugin)
461 return gst_element_register (plugin, "rtpamrdepay",
462 GST_RANK_MARGINAL, GST_TYPE_RTP_AMR_DEPAY);