2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
3 * Copyright (C) <2015> GE Intelligent Platforms Embedded Systems, Inc.
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Library General Public License for more details.
15 * You should have received a copy of the GNU Library General Public
16 * License along with this library; if not, write to the
17 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
18 * Boston, MA 02110-1301, USA.
22 * SECTION:element-rtpL8depay
25 * Extract raw audio from RTP packets according to RFC 3551.
26 * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
31 * gst-launch udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L8, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL8depay ! pulsesink
32 * ]| This example pipeline will depayload an RTP raw audio stream. Refer to
33 * the rtpL8pay example to create the RTP stream.
43 #include <gst/audio/audio.h>
45 #include "gstrtpelements.h"
46 #include "gstrtpL8depay.h"
47 #include "gstrtpchannels.h"
49 GST_DEBUG_CATEGORY_STATIC (rtpL8depay_debug);
50 #define GST_CAT_DEFAULT (rtpL8depay_debug)
52 static GstStaticPadTemplate gst_rtp_L8_depay_src_template =
53 GST_STATIC_PAD_TEMPLATE ("src",
56 GST_STATIC_CAPS ("audio/x-raw, "
57 "format = (string) U8, "
58 "layout = (string) interleaved, "
59 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
62 static GstStaticPadTemplate gst_rtp_L8_depay_sink_template =
63 GST_STATIC_PAD_TEMPLATE ("sink",
66 GST_STATIC_CAPS ("application/x-rtp, "
67 "media = (string) audio, clock-rate = (int) [ 1, MAX ], "
68 /* "channels = (int) [1, MAX]" */
69 /* "emphasis = (string) ANY" */
70 /* "channel-order = (string) ANY" */
71 "encoding-name = (string) L8;")
74 #define gst_rtp_L8_depay_parent_class parent_class
75 G_DEFINE_TYPE (GstRtpL8Depay, gst_rtp_L8_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
78 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpL8depay, "rtpL8depay",
79 GST_RANK_SECONDARY, GST_TYPE_RTP_L8_DEPAY, rtp_element_init (plugin));
81 static gboolean gst_rtp_L8_depay_setcaps (GstRTPBaseDepayload * depayload,
83 static GstBuffer *gst_rtp_L8_depay_process (GstRTPBaseDepayload * depayload,
87 gst_rtp_L8_depay_class_init (GstRtpL8DepayClass * klass)
89 GstElementClass *gstelement_class;
90 GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
92 gstelement_class = (GstElementClass *) klass;
93 gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
95 gstrtpbasedepayload_class->set_caps = gst_rtp_L8_depay_setcaps;
96 gstrtpbasedepayload_class->process = gst_rtp_L8_depay_process;
98 gst_element_class_add_pad_template (gstelement_class,
99 gst_static_pad_template_get (&gst_rtp_L8_depay_src_template));
100 gst_element_class_add_pad_template (gstelement_class,
101 gst_static_pad_template_get (&gst_rtp_L8_depay_sink_template));
103 gst_element_class_set_static_metadata (gstelement_class,
104 "RTP audio depayloader", "Codec/Depayloader/Network/RTP",
105 "Extracts raw audio from RTP packets",
106 "Zeeshan Ali <zak147@yahoo.com>," "Wim Taymans <wim.taymans@gmail.com>, "
107 "GE Intelligent Platforms Embedded Systems, Inc.");
109 GST_DEBUG_CATEGORY_INIT (rtpL8depay_debug, "rtpL8depay", 0,
110 "Raw Audio RTP Depayloader");
114 gst_rtp_L8_depay_init (GstRtpL8Depay * rtpL8depay)
119 gst_rtp_L8_depay_parse_int (GstStructure * structure, const gchar * field,
125 if ((str = gst_structure_get_string (structure, field)))
128 if (gst_structure_get_int (structure, field, &res))
135 gst_rtp_L8_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
137 GstStructure *structure;
138 GstRtpL8Depay *rtpL8depay;
143 const gchar *channel_order;
144 const GstRTPChannelOrder *order;
147 rtpL8depay = GST_RTP_L8_DEPAY (depayload);
149 structure = gst_caps_get_structure (caps, 0);
151 /* no fixed mapping, we need clock-rate */
155 /* caps can overwrite defaults */
156 clock_rate = gst_rtp_L8_depay_parse_int (structure, "clock-rate", clock_rate);
161 gst_rtp_L8_depay_parse_int (structure, "encoding-params", channels);
163 channels = gst_rtp_L8_depay_parse_int (structure, "channels", channels);
165 /* channels defaults to 1 otherwise */
170 depayload->clock_rate = clock_rate;
172 info = &rtpL8depay->info;
173 gst_audio_info_init (info);
174 info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_U8);
175 info->rate = clock_rate;
176 info->channels = channels;
177 info->bpf = (info->finfo->width / 8) * channels;
179 /* add channel positions */
180 channel_order = gst_structure_get_string (structure, "channel-order");
182 order = gst_rtp_channels_get_by_order (channels, channel_order);
183 rtpL8depay->order = order;
185 memcpy (info->position, order->pos,
186 sizeof (GstAudioChannelPosition) * channels);
187 gst_audio_channel_positions_to_valid_order (info->position, info->channels);
189 GST_ELEMENT_WARNING (rtpL8depay, STREAM, DECODE,
190 (NULL), ("Unknown channel order '%s' for %d channels",
191 GST_STR_NULL (channel_order), channels));
192 /* create default NONE layout */
193 gst_rtp_channels_create_default (channels, info->position);
194 info->flags |= GST_AUDIO_FLAG_UNPOSITIONED;
197 srccaps = gst_audio_info_to_caps (info);
198 res = gst_pad_set_caps (depayload->srcpad, srccaps);
199 gst_caps_unref (srccaps);
206 GST_ERROR_OBJECT (depayload, "no clock-rate specified");
212 gst_rtp_L8_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
214 GstRtpL8Depay *rtpL8depay;
218 GstRTPBuffer rtp = { NULL };
220 rtpL8depay = GST_RTP_L8_DEPAY (depayload);
222 gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
223 payload_len = gst_rtp_buffer_get_payload_len (&rtp);
225 if (payload_len <= 0)
228 GST_DEBUG_OBJECT (rtpL8depay, "got payload of %d bytes", payload_len);
230 outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
231 marker = gst_rtp_buffer_get_marker (&rtp);
234 /* mark talk spurt with RESYNC */
235 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
238 outbuf = gst_buffer_make_writable (outbuf);
239 if (rtpL8depay->order &&
240 !gst_audio_buffer_reorder_channels (outbuf,
241 rtpL8depay->info.finfo->format, rtpL8depay->info.channels,
242 rtpL8depay->info.position, rtpL8depay->order->pos)) {
246 gst_rtp_buffer_unmap (&rtp);
253 GST_ELEMENT_WARNING (rtpL8depay, STREAM, DECODE,
254 ("Empty Payload."), (NULL));
255 gst_rtp_buffer_unmap (&rtp);
260 GST_ELEMENT_ERROR (rtpL8depay, STREAM, DECODE,
261 ("Channel reordering failed."), (NULL));
262 gst_rtp_buffer_unmap (&rtp);