2 * Copyright (C) <2007> Nokia Corporation
3 * Copyright (C) <2007> Collabora Ltd
4 * @author: Olivier Crete <olivier.crete@collabora.co.uk>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * This payloader assumes that the data will ALWAYS come as zero or more
24 * 10 bytes frame of audio followed by 0 or 1 2 byte frame of silence.
25 * Any other buffer format won't work
33 #include <gst/rtp/gstrtpbuffer.h>
34 #include <gst/base/gstadapter.h>
35 #include <gst/audio/audio.h>
37 #include "gstrtpelements.h"
38 #include "gstrtpg729pay.h"
39 #include "gstrtputils.h"
41 GST_DEBUG_CATEGORY_STATIC (rtpg729pay_debug);
42 #define GST_CAT_DEFAULT (rtpg729pay_debug)
44 #define G729_FRAME_SIZE 10
45 #define G729B_CN_FRAME_SIZE 2
46 #define G729_FRAME_DURATION (10 * GST_MSECOND)
47 #define G729_FRAME_DURATION_MS (10)
50 gst_rtp_g729_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps);
52 gst_rtp_g729_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf);
54 static GstStateChangeReturn
55 gst_rtp_g729_pay_change_state (GstElement * element, GstStateChange transition);
57 static GstStaticPadTemplate gst_rtp_g729_pay_sink_template =
58 GST_STATIC_PAD_TEMPLATE ("sink",
61 GST_STATIC_CAPS ("audio/G729, " /* according to RFC 3555 */
62 "channels = (int) 1, " "rate = (int) 8000")
65 static GstStaticPadTemplate gst_rtp_g729_pay_src_template =
66 GST_STATIC_PAD_TEMPLATE ("src",
69 GST_STATIC_CAPS ("application/x-rtp, "
70 "media = (string) \"audio\", "
71 "payload = (int) " GST_RTP_PAYLOAD_G729_STRING ", "
72 "clock-rate = (int) 8000, "
73 "encoding-name = (string) \"G729\"; "
75 "media = (string) \"audio\", "
76 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
77 "clock-rate = (int) 8000, " "encoding-name = (string) \"G729\"")
80 #define gst_rtp_g729_pay_parent_class parent_class
81 G_DEFINE_TYPE (GstRTPG729Pay, gst_rtp_g729_pay, GST_TYPE_RTP_BASE_PAYLOAD);
82 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpg729pay, "rtpg729pay",
83 GST_RANK_SECONDARY, GST_TYPE_RTP_G729_PAY, rtp_element_init (plugin));
86 gst_rtp_g729_pay_finalize (GObject * object)
88 GstRTPG729Pay *pay = GST_RTP_G729_PAY (object);
90 g_object_unref (pay->adapter);
92 G_OBJECT_CLASS (parent_class)->finalize (object);
96 gst_rtp_g729_pay_class_init (GstRTPG729PayClass * klass)
98 GObjectClass *gobject_class = (GObjectClass *) klass;
99 GstElementClass *gstelement_class = (GstElementClass *) klass;
100 GstRTPBasePayloadClass *payload_class = GST_RTP_BASE_PAYLOAD_CLASS (klass);
102 GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0,
103 "G.729 RTP Payloader");
105 gobject_class->finalize = gst_rtp_g729_pay_finalize;
107 gstelement_class->change_state = gst_rtp_g729_pay_change_state;
109 gst_element_class_add_static_pad_template (gstelement_class,
110 &gst_rtp_g729_pay_sink_template);
111 gst_element_class_add_static_pad_template (gstelement_class,
112 &gst_rtp_g729_pay_src_template);
114 gst_element_class_set_static_metadata (gstelement_class,
115 "RTP G.729 payloader", "Codec/Payloader/Network/RTP",
116 "Packetize G.729 audio into RTP packets",
117 "Olivier Crete <olivier.crete@collabora.co.uk>");
119 payload_class->set_caps = gst_rtp_g729_pay_set_caps;
120 payload_class->handle_buffer = gst_rtp_g729_pay_handle_buffer;
124 gst_rtp_g729_pay_init (GstRTPG729Pay * pay)
126 GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay);
128 payload->pt = GST_RTP_PAYLOAD_G729;
130 pay->adapter = gst_adapter_new ();
134 gst_rtp_g729_pay_reset (GstRTPG729Pay * pay)
136 gst_adapter_clear (pay->adapter);
137 pay->discont = FALSE;
138 pay->next_rtp_time = 0;
139 pay->first_ts = GST_CLOCK_TIME_NONE;
140 pay->first_rtp_time = 0;
144 gst_rtp_g729_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
148 gst_rtp_base_payload_set_options (payload, "audio",
149 payload->pt != GST_RTP_PAYLOAD_G729, "G729", 8000);
151 res = gst_rtp_base_payload_set_outcaps (payload, NULL);
157 gst_rtp_g729_pay_push (GstRTPG729Pay * rtpg729pay, GstBuffer * buf)
159 GstRTPBasePayload *basepayload;
160 GstClockTime duration;
164 GstRTPBuffer rtp = { NULL };
165 guint payload_len = gst_buffer_get_size (buf);
167 basepayload = GST_RTP_BASE_PAYLOAD (rtpg729pay);
169 GST_DEBUG_OBJECT (rtpg729pay, "Pushing %d bytes ts %" GST_TIME_FORMAT,
170 payload_len, GST_TIME_ARGS (rtpg729pay->next_ts));
172 /* create buffer to hold the payload */
174 gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
175 (rtpg729pay), 0, 0, 0);
177 gst_rtp_buffer_map (outbuf, GST_MAP_READWRITE, &rtp);
181 (payload_len / G729_FRAME_SIZE) + ((payload_len % G729_FRAME_SIZE) >> 1);
182 duration = frames * G729_FRAME_DURATION;
183 GST_BUFFER_PTS (outbuf) = rtpg729pay->next_ts;
184 GST_BUFFER_DURATION (outbuf) = duration;
185 GST_BUFFER_OFFSET (outbuf) = rtpg729pay->next_rtp_time;
186 rtpg729pay->next_ts += duration;
187 rtpg729pay->next_rtp_time += frames * 80;
189 if (G_UNLIKELY (rtpg729pay->discont)) {
190 GST_DEBUG_OBJECT (basepayload, "discont, setting marker bit");
191 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
192 gst_rtp_buffer_set_marker (&rtp, TRUE);
193 rtpg729pay->discont = FALSE;
195 gst_rtp_buffer_unmap (&rtp);
198 gst_rtp_copy_audio_meta (basepayload, outbuf, buf);
199 outbuf = gst_buffer_append (outbuf, buf);
201 ret = gst_rtp_base_payload_push (basepayload, outbuf);
207 gst_rtp_g729_pay_recalc_rtp_time (GstRTPG729Pay * rtpg729pay, GstClockTime time)
209 if (GST_CLOCK_TIME_IS_VALID (rtpg729pay->first_ts)
210 && GST_CLOCK_TIME_IS_VALID (time) && time >= rtpg729pay->first_ts) {
214 diff = time - rtpg729pay->first_ts;
215 rtpdiff = (diff / GST_MSECOND) * 8;
216 rtpg729pay->next_rtp_time = rtpg729pay->first_rtp_time + rtpdiff;
217 GST_DEBUG_OBJECT (rtpg729pay,
218 "elapsed time %" GST_TIME_FORMAT ", rtp %" G_GUINT32_FORMAT ", "
219 "new offset %" G_GUINT32_FORMAT, GST_TIME_ARGS (diff), rtpdiff,
220 rtpg729pay->next_rtp_time);
225 gst_rtp_g729_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf)
227 GstFlowReturn ret = GST_FLOW_OK;
228 GstRTPG729Pay *rtpg729pay = GST_RTP_G729_PAY (payload);
229 GstAdapter *adapter = NULL;
232 guint maxptime_octets = G_MAXUINT;
233 guint minptime_octets = 0;
234 guint min_payload_len;
235 guint max_payload_len;
237 GstClockTime timestamp;
239 size = gst_buffer_get_size (buf);
241 if (size % G729_FRAME_SIZE != 0 &&
242 size % G729_FRAME_SIZE != G729B_CN_FRAME_SIZE)
245 /* max number of bytes based on given ptime, has to be multiple of
247 if (payload->max_ptime != -1) {
248 guint ptime_ms = payload->max_ptime / GST_MSECOND;
250 maxptime_octets = G729_FRAME_SIZE *
251 (int) (ptime_ms / G729_FRAME_DURATION_MS);
253 if (maxptime_octets < G729_FRAME_SIZE) {
254 GST_WARNING_OBJECT (payload, "Given ptime %" G_GINT64_FORMAT
255 " is smaller than minimum %d ns, overwriting to minimum",
256 payload->max_ptime, G729_FRAME_DURATION_MS);
257 maxptime_octets = G729_FRAME_SIZE;
261 max_payload_len = MIN (
263 (int) (gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU
264 (payload), 0, 0) / G729_FRAME_SIZE)
269 /* min number of bytes based on a given ptime, has to be a multiple
272 guint64 min_ptime = payload->min_ptime;
274 min_ptime = min_ptime / GST_MSECOND;
275 minptime_octets = G729_FRAME_SIZE *
276 (int) (min_ptime / G729_FRAME_DURATION_MS);
279 min_payload_len = MAX (minptime_octets, G729_FRAME_SIZE);
281 if (min_payload_len > max_payload_len) {
282 min_payload_len = max_payload_len;
285 /* If the ptime is specified in the caps, tried to adhere to it exactly */
286 if (payload->ptime) {
287 guint64 ptime = payload->ptime / GST_MSECOND;
288 guint ptime_in_bytes = G729_FRAME_SIZE *
289 (guint) (ptime / G729_FRAME_DURATION_MS);
291 /* clip to computed min and max lengths */
292 ptime_in_bytes = MAX (min_payload_len, ptime_in_bytes);
293 ptime_in_bytes = MIN (max_payload_len, ptime_in_bytes);
295 min_payload_len = max_payload_len = ptime_in_bytes;
298 GST_LOG_OBJECT (payload,
299 "Calculated min_payload_len %u and max_payload_len %u",
300 min_payload_len, max_payload_len);
302 adapter = rtpg729pay->adapter;
303 available = gst_adapter_available (adapter);
305 timestamp = GST_BUFFER_PTS (buf);
307 /* resync rtp time on discont or a discontinuous cn packet */
308 if (GST_BUFFER_IS_DISCONT (buf)) {
309 /* flush remainder */
311 gst_rtp_g729_pay_push (rtpg729pay,
312 gst_adapter_take_buffer_fast (adapter, available));
315 rtpg729pay->discont = TRUE;
316 gst_rtp_g729_pay_recalc_rtp_time (rtpg729pay, timestamp);
319 if (size < G729_FRAME_SIZE)
320 gst_rtp_g729_pay_recalc_rtp_time (rtpg729pay, timestamp);
322 if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (rtpg729pay->first_ts))) {
323 rtpg729pay->first_ts = timestamp;
324 rtpg729pay->first_rtp_time = rtpg729pay->next_rtp_time;
327 /* let's reset the base timestamp when the adapter is empty */
329 rtpg729pay->next_ts = timestamp;
331 if (available == 0 && size >= min_payload_len && size <= max_payload_len) {
332 ret = gst_rtp_g729_pay_push (rtpg729pay, buf);
336 gst_adapter_push (adapter, buf);
337 available = gst_adapter_available (adapter);
339 /* as long as we have full frames */
340 /* this loop will push all available buffers till the last frame */
341 while (available >= min_payload_len ||
342 available % G729_FRAME_SIZE == G729B_CN_FRAME_SIZE) {
343 /* We send as much as we can */
344 if (available <= max_payload_len) {
345 payload_len = available;
347 payload_len = MIN (max_payload_len,
348 (available / G729_FRAME_SIZE) * G729_FRAME_SIZE);
351 ret = gst_rtp_g729_pay_push (rtpg729pay,
352 gst_adapter_take_buffer_fast (adapter, payload_len));
353 available -= payload_len;
361 GST_ELEMENT_ERROR (payload, STREAM, WRONG_TYPE,
362 ("Invalid input buffer size"),
363 ("Invalid buffer size, should be a multiple of"
364 " G729_FRAME_SIZE(10) with an optional G729B_CN_FRAME_SIZE(2)"
365 " added to it, but it is %" G_GSIZE_FORMAT, size));
366 gst_buffer_unref (buf);
367 return GST_FLOW_ERROR;
371 static GstStateChangeReturn
372 gst_rtp_g729_pay_change_state (GstElement * element, GstStateChange transition)
374 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
376 /* handle upwards state changes here */
377 switch (transition) {
382 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
384 /* handle downwards state changes */
385 switch (transition) {
386 case GST_STATE_CHANGE_PAUSED_TO_READY:
387 gst_rtp_g729_pay_reset (GST_RTP_G729_PAY (element));