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.
26 #include <gst/rtp/gstrtpbuffer.h>
28 #include "gstrtpamrpay.h"
30 GST_DEBUG_CATEGORY_STATIC (rtpamrpay_debug);
31 #define GST_CAT_DEFAULT (rtpamrpay_debug)
35 * RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
36 * Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive
37 * Multi-Rate Wideband (AMR-WB) Audio Codecs.
39 * ETSI TS 126 201 V6.0.0 (2004-12) - Digital cellular telecommunications system (Phase 2+);
40 * Universal Mobile Telecommunications System (UMTS);
41 * AMR speech codec, wideband;
43 * (3GPP TS 26.201 version 6.0.0 Release 6)
46 static GstStaticPadTemplate gst_rtp_amr_pay_sink_template =
47 GST_STATIC_PAD_TEMPLATE ("sink",
50 GST_STATIC_CAPS ("audio/AMR, channels=(int)1, rate=(int)8000; "
51 "audio/AMR-WB, channels=(int)1, rate=(int)16000")
54 static GstStaticPadTemplate gst_rtp_amr_pay_src_template =
55 GST_STATIC_PAD_TEMPLATE ("src",
58 GST_STATIC_CAPS ("application/x-rtp, "
59 "media = (string) \"audio\", "
60 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
61 "clock-rate = (int) 8000, "
62 "encoding-name = (string) \"AMR\", "
63 "encoding-params = (string) \"1\", "
64 "octet-align = (string) \"1\", "
65 "crc = (string) \"0\", "
66 "robust-sorting = (string) \"0\", "
67 "interleaving = (string) \"0\", "
68 "mode-set = (int) [ 0, 7 ], "
69 "mode-change-period = (int) [ 1, MAX ], "
70 "mode-change-neighbor = (string) { \"0\", \"1\" }, "
71 "maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ];"
73 "media = (string) \"audio\", "
74 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
75 "clock-rate = (int) 16000, "
76 "encoding-name = (string) \"AMR-WB\", "
77 "encoding-params = (string) \"1\", "
78 "octet-align = (string) \"1\", "
79 "crc = (string) \"0\", "
80 "robust-sorting = (string) \"0\", "
81 "interleaving = (string) \"0\", "
82 "mode-set = (int) [ 0, 7 ], "
83 "mode-change-period = (int) [ 1, MAX ], "
84 "mode-change-neighbor = (string) { \"0\", \"1\" }, "
85 "maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ]")
88 static gboolean gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload,
90 static GstFlowReturn gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * pad,
93 static GstStateChangeReturn
94 gst_rtp_amr_pay_change_state (GstElement * element, GstStateChange transition);
96 GST_BOILERPLATE (GstRtpAMRPay, gst_rtp_amr_pay, GstBaseRTPPayload,
97 GST_TYPE_BASE_RTP_PAYLOAD);
100 gst_rtp_amr_pay_base_init (gpointer klass)
102 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
104 gst_element_class_add_static_pad_template (element_class,
105 &gst_rtp_amr_pay_src_template);
106 gst_element_class_add_static_pad_template (element_class,
107 &gst_rtp_amr_pay_sink_template);
109 gst_element_class_set_details_simple (element_class, "RTP AMR payloader",
110 "Codec/Payloader/Network/RTP",
111 "Payload-encode AMR or AMR-WB audio into RTP packets (RFC 3267)",
112 "Wim Taymans <wim.taymans@gmail.com>");
116 gst_rtp_amr_pay_class_init (GstRtpAMRPayClass * klass)
118 GstBaseRTPPayloadClass *gstbasertppayload_class;
119 GstElementClass *gstelement_class;
121 gstelement_class = (GstElementClass *) klass;
122 gstelement_class->change_state = gst_rtp_amr_pay_change_state;
124 gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
126 gstbasertppayload_class->set_caps = gst_rtp_amr_pay_setcaps;
127 gstbasertppayload_class->handle_buffer = gst_rtp_amr_pay_handle_buffer;
129 GST_DEBUG_CATEGORY_INIT (rtpamrpay_debug, "rtpamrpay", 0,
130 "AMR/AMR-WB RTP Payloader");
134 gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay, GstRtpAMRPayClass * klass)
136 /* needed because of GST_BOILERPLATE */
140 gst_rtp_amr_pay_reset (GstRtpAMRPay * pay)
142 pay->next_rtp_time = 0;
143 pay->first_ts = GST_CLOCK_TIME_NONE;
144 pay->first_rtp_time = 0;
148 gst_rtp_amr_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
150 GstRtpAMRPay *rtpamrpay;
152 const GstStructure *s;
155 rtpamrpay = GST_RTP_AMR_PAY (basepayload);
157 /* figure out the mode Narrow or Wideband */
158 s = gst_caps_get_structure (caps, 0);
159 if ((str = gst_structure_get_name (s))) {
160 if (strcmp (str, "audio/AMR") == 0)
161 rtpamrpay->mode = GST_RTP_AMR_P_MODE_NB;
162 else if (strcmp (str, "audio/AMR-WB") == 0)
163 rtpamrpay->mode = GST_RTP_AMR_P_MODE_WB;
169 if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB)
170 gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
172 gst_basertppayload_set_options (basepayload, "audio", TRUE, "AMR-WB",
175 res = gst_basertppayload_set_outcaps (basepayload,
176 "encoding-params", G_TYPE_STRING, "1", "octet-align", G_TYPE_STRING, "1",
177 /* don't set the defaults
179 * "crc", G_TYPE_STRING, "0",
180 * "robust-sorting", G_TYPE_STRING, "0",
181 * "interleaving", G_TYPE_STRING, "0",
190 GST_ERROR_OBJECT (rtpamrpay, "unsupported media type '%s'",
197 gst_rtp_amr_pay_recalc_rtp_time (GstRtpAMRPay * rtpamrpay,
198 GstClockTime timestamp)
200 /* re-sync rtp time */
201 if (GST_CLOCK_TIME_IS_VALID (rtpamrpay->first_ts) &&
202 GST_CLOCK_TIME_IS_VALID (timestamp) && timestamp >= rtpamrpay->first_ts) {
206 /* interpolate to reproduce gap from start, rather than intermediate
207 * intervals to avoid roundup accumulation errors */
208 diff = timestamp - rtpamrpay->first_ts;
209 rtpdiff = ((diff / GST_MSECOND) * 8) <<
210 (rtpamrpay->mode == GST_RTP_AMR_P_MODE_WB);
211 rtpamrpay->next_rtp_time = rtpamrpay->first_rtp_time + rtpdiff;
212 GST_DEBUG_OBJECT (rtpamrpay,
213 "elapsed time %" GST_TIME_FORMAT ", rtp %" G_GUINT32_FORMAT ", "
214 "new offset %" G_GUINT32_FORMAT, GST_TIME_ARGS (diff), rtpdiff,
215 rtpamrpay->next_rtp_time);
220 static const gint nb_frame_size[16] = {
221 12, 13, 15, 17, 19, 20, 26, 31,
222 5, -1, -1, -1, -1, -1, -1, 0
225 static const gint wb_frame_size[16] = {
226 17, 23, 32, 36, 40, 46, 50, 58,
227 60, 5, -1, -1, -1, -1, -1, 0
231 gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * basepayload,
234 GstRtpAMRPay *rtpamrpay;
235 const gint *frame_size;
237 guint size, payload_len;
239 guint8 *payload, *data, *payload_amr;
240 GstClockTime timestamp, duration;
241 guint packet_len, mtu;
242 gint i, num_packets, num_nonempty_packets;
244 gboolean sid = FALSE;
246 rtpamrpay = GST_RTP_AMR_PAY (basepayload);
247 mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpamrpay);
249 size = GST_BUFFER_SIZE (buffer);
250 data = GST_BUFFER_DATA (buffer);
251 timestamp = GST_BUFFER_TIMESTAMP (buffer);
252 duration = GST_BUFFER_DURATION (buffer);
254 /* setup frame size pointer */
255 if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB)
256 frame_size = nb_frame_size;
258 frame_size = wb_frame_size;
260 GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
263 * octet aligned, no interleaving, single channel, no CRC,
264 * no robust-sorting. To fix this you need to implement the downstream
265 * negotiation function. */
267 /* first count number of packets and total amr frame size */
268 amr_len = num_packets = num_nonempty_packets = 0;
269 for (i = 0; i < size; i++) {
273 FT = (data[i] & 0x78) >> 3;
275 fr_size = frame_size[FT];
276 GST_DEBUG_OBJECT (basepayload, "frame type %d, frame size %d", FT, fr_size);
277 /* FIXME, we don't handle this yet.. */
285 num_nonempty_packets++;
290 goto incomplete_frame;
292 /* we need one extra byte for the CMR, the ToC is in the input
294 payload_len = size + 1;
296 /* get packet len to check against MTU */
297 packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
298 if (packet_len > mtu)
301 /* now alloc output buffer */
302 outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
305 GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
307 if (duration != GST_CLOCK_TIME_NONE)
308 GST_BUFFER_DURATION (outbuf) = duration;
310 GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND;
313 if (GST_BUFFER_IS_DISCONT (buffer)) {
314 GST_DEBUG_OBJECT (basepayload, "discont, setting marker bit");
315 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
316 gst_rtp_buffer_set_marker (outbuf, TRUE);
317 gst_rtp_amr_pay_recalc_rtp_time (rtpamrpay, timestamp);
320 if (G_UNLIKELY (sid)) {
321 gst_rtp_amr_pay_recalc_rtp_time (rtpamrpay, timestamp);
324 /* perfect rtptime */
325 if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (rtpamrpay->first_ts))) {
326 rtpamrpay->first_ts = timestamp;
327 rtpamrpay->first_rtp_time = rtpamrpay->next_rtp_time;
329 GST_BUFFER_OFFSET (outbuf) = rtpamrpay->next_rtp_time;
330 rtpamrpay->next_rtp_time +=
331 (num_packets * 160) << (rtpamrpay->mode == GST_RTP_AMR_P_MODE_WB);
333 /* get payload, this is now writable */
334 payload = gst_rtp_buffer_get_payload (outbuf);
341 payload[0] = 0xF0; /* CMR, no specific mode requested */
343 /* this is where we copy the AMR data, after num_packets FTs and the
345 payload_amr = payload + num_packets + 1;
347 /* copy data in payload, first we copy all the FTs then all
348 * the AMR data. The last FT has to have the F flag cleared. */
349 for (i = 1; i <= num_packets; i++) {
355 * |F| FT |Q|P|P| more FT...
358 FT = (*data & 0x78) >> 3;
360 fr_size = frame_size[FT];
362 if (i == num_packets)
363 /* last packet, clear F flag */
364 payload[i] = *data & 0x7f;
367 payload[i] = *data | 0x80;
369 memcpy (payload_amr, &data[1], fr_size);
371 /* all sizes are > 0 since we checked for that above */
373 payload_amr += fr_size;
376 gst_buffer_unref (buffer);
378 ret = gst_basertppayload_push (basepayload, outbuf);
385 GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
386 (NULL), ("received AMR frame with size <= 0"));
387 gst_buffer_unref (buffer);
389 return GST_FLOW_ERROR;
393 GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
394 (NULL), ("received incomplete AMR frames"));
395 gst_buffer_unref (buffer);
397 return GST_FLOW_ERROR;
401 GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
402 (NULL), ("received too many AMR frames for MTU"));
403 gst_buffer_unref (buffer);
405 return GST_FLOW_ERROR;
409 static GstStateChangeReturn
410 gst_rtp_amr_pay_change_state (GstElement * element, GstStateChange transition)
412 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
414 /* handle upwards state changes here */
415 switch (transition) {
420 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
422 /* handle downwards state changes */
423 switch (transition) {
424 case GST_STATE_CHANGE_PAUSED_TO_READY:
425 gst_rtp_amr_pay_reset (GST_RTP_AMR_PAY (element));
435 gst_rtp_amr_pay_plugin_init (GstPlugin * plugin)
437 return gst_element_register (plugin, "rtpamrpay",
438 GST_RANK_SECONDARY, GST_TYPE_RTP_AMR_PAY);