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Olivier Naudan988c6f02012-04-16 07:16:25 -04001/* GStreamer
2 * Copyright (C) <2007> 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
Sebastian Dröge5f0bab02013-07-14 11:42:29 +020016 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
18 */
19
20/**
21 * SECTION:element-rtpL16depay
22 * @see_also: rtpL16pay
23 *
24 * Extract raw audio from RTP packets according to RFC 3551.
25 * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
26 *
27 * <refsect2>
28 * <title>Example pipeline</title>
29 * |[
Sebastian Dröge010e0292015-05-13 13:24:03 +030030 * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL16depay ! pulsesink
Sebastian Dröge5f0bab02013-07-14 11:42:29 +020031 * ]| This example pipeline will depayload an RTP raw audio stream. Refer to
32 * the rtpL16pay example to create the RTP stream.
33 * </refsect2>
Olivier Naudan988c6f02012-04-16 07:16:25 -040034 */
35
36#ifdef HAVE_CONFIG_H
37#include "config.h"
38#endif
39
40#include <string.h>
41#include <stdlib.h>
42
43#include <gst/audio/audio.h>
44
45#include "gstrtpL16depay.h"
46#include "gstrtpchannels.h"
Sebastian Drögeb04f9722015-08-19 13:19:08 +030047#include "gstrtputils.h"
Olivier Naudan988c6f02012-04-16 07:16:25 -040048
49GST_DEBUG_CATEGORY_STATIC (rtpL16depay_debug);
50#define GST_CAT_DEFAULT (rtpL16depay_debug)
51
52static GstStaticPadTemplate gst_rtp_L16_depay_src_template =
53GST_STATIC_PAD_TEMPLATE ("src",
54 GST_PAD_SRC,
55 GST_PAD_ALWAYS,
56 GST_STATIC_CAPS ("audio/x-raw, "
57 "format = (string) S16BE, "
58 "layout = (string) interleaved, "
59 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
60 );
61
62static GstStaticPadTemplate gst_rtp_L16_depay_sink_template =
63 GST_STATIC_PAD_TEMPLATE ("sink",
64 GST_PAD_SINK,
65 GST_PAD_ALWAYS,
66 GST_STATIC_CAPS ("application/x-rtp, "
Sebastian Dröge5f0bab02013-07-14 11:42:29 +020067 "media = (string) \"audio\", " "clock-rate = (int) [ 1, MAX ], "
Olivier Naudan988c6f02012-04-16 07:16:25 -040068 /* "channels = (int) [1, MAX]" */
69 /* "emphasis = (string) ANY" */
70 /* "channel-order = (string) ANY" */
71 "encoding-name = (string) \"L16\";"
72 "application/x-rtp, "
73 "media = (string) \"audio\", "
74 "payload = (int) { " GST_RTP_PAYLOAD_L16_STEREO_STRING ", "
75 GST_RTP_PAYLOAD_L16_MONO_STRING " }," "clock-rate = (int) [ 1, MAX ]"
76 /* "channels = (int) [1, MAX]" */
77 /* "emphasis = (string) ANY" */
78 /* "channel-order = (string) ANY" */
79 )
80 );
81
82#define gst_rtp_L16_depay_parent_class parent_class
83G_DEFINE_TYPE (GstRtpL16Depay, gst_rtp_L16_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
84
85static gboolean gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload,
86 GstCaps * caps);
87static GstBuffer *gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload,
Sebastian Drögeb04f9722015-08-19 13:19:08 +030088 GstRTPBuffer * rtp);
Olivier Naudan988c6f02012-04-16 07:16:25 -040089
90static void
91gst_rtp_L16_depay_class_init (GstRtpL16DepayClass * klass)
92{
93 GstElementClass *gstelement_class;
94 GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
95
96 gstelement_class = (GstElementClass *) klass;
97 gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
98
99 gstrtpbasedepayload_class->set_caps = gst_rtp_L16_depay_setcaps;
Sebastian Drögeb04f9722015-08-19 13:19:08 +0300100 gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_L16_depay_process;
Olivier Naudan988c6f02012-04-16 07:16:25 -0400101
Sebastian Dröge3a50aa72016-07-06 12:38:23 +0300102 gst_element_class_add_static_pad_template (gstelement_class,
103 &gst_rtp_L16_depay_src_template);
104 gst_element_class_add_static_pad_template (gstelement_class,
105 &gst_rtp_L16_depay_sink_template);
Olivier Naudan988c6f02012-04-16 07:16:25 -0400106
107 gst_element_class_set_static_metadata (gstelement_class,
108 "RTP audio depayloader", "Codec/Depayloader/Network/RTP",
109 "Extracts raw audio from RTP packets",
110 "Zeeshan Ali <zak147@yahoo.com>," "Wim Taymans <wim.taymans@gmail.com>");
111
112 GST_DEBUG_CATEGORY_INIT (rtpL16depay_debug, "rtpL16depay", 0,
113 "Raw Audio RTP Depayloader");
114}
115
116static void
117gst_rtp_L16_depay_init (GstRtpL16Depay * rtpL16depay)
118{
Olivier Naudan988c6f02012-04-16 07:16:25 -0400119}
120
121static gint
122gst_rtp_L16_depay_parse_int (GstStructure * structure, const gchar * field,
123 gint def)
124{
125 const gchar *str;
126 gint res;
127
128 if ((str = gst_structure_get_string (structure, field)))
129 return atoi (str);
130
131 if (gst_structure_get_int (structure, field, &res))
132 return res;
133
134 return def;
135}
136
137static gboolean
138gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
139{
140 GstStructure *structure;
141 GstRtpL16Depay *rtpL16depay;
142 gint clock_rate, payload;
143 gint channels;
144 GstCaps *srccaps;
145 gboolean res;
146 const gchar *channel_order;
147 const GstRTPChannelOrder *order;
148 GstAudioInfo *info;
149
150 rtpL16depay = GST_RTP_L16_DEPAY (depayload);
151
152 structure = gst_caps_get_structure (caps, 0);
153
154 payload = 96;
155 gst_structure_get_int (structure, "payload", &payload);
156 switch (payload) {
157 case GST_RTP_PAYLOAD_L16_STEREO:
158 channels = 2;
159 clock_rate = 44100;
160 break;
161 case GST_RTP_PAYLOAD_L16_MONO:
162 channels = 1;
163 clock_rate = 44100;
164 break;
165 default:
166 /* no fixed mapping, we need clock-rate */
167 channels = 0;
168 clock_rate = 0;
169 break;
170 }
171
172 /* caps can overwrite defaults */
173 clock_rate =
174 gst_rtp_L16_depay_parse_int (structure, "clock-rate", clock_rate);
175 if (clock_rate == 0)
176 goto no_clockrate;
177
178 channels =
179 gst_rtp_L16_depay_parse_int (structure, "encoding-params", channels);
180 if (channels == 0) {
181 channels = gst_rtp_L16_depay_parse_int (structure, "channels", channels);
182 if (channels == 0) {
183 /* channels defaults to 1 otherwise */
184 channels = 1;
185 }
186 }
187
188 depayload->clock_rate = clock_rate;
189
190 info = &rtpL16depay->info;
191 gst_audio_info_init (info);
192 info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S16BE);
193 info->rate = clock_rate;
194 info->channels = channels;
195 info->bpf = (info->finfo->width / 8) * channels;
196
197 /* add channel positions */
198 channel_order = gst_structure_get_string (structure, "channel-order");
199
200 order = gst_rtp_channels_get_by_order (channels, channel_order);
201 rtpL16depay->order = order;
202 if (order) {
203 memcpy (info->position, order->pos,
204 sizeof (GstAudioChannelPosition) * channels);
205 gst_audio_channel_positions_to_valid_order (info->position, info->channels);
206 } else {
207 GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
208 (NULL), ("Unknown channel order '%s' for %d channels",
209 GST_STR_NULL (channel_order), channels));
210 /* create default NONE layout */
211 gst_rtp_channels_create_default (channels, info->position);
212 }
213
214 srccaps = gst_audio_info_to_caps (info);
215 res = gst_pad_set_caps (depayload->srcpad, srccaps);
216 gst_caps_unref (srccaps);
217
218 return res;
219
220 /* ERRORS */
221no_clockrate:
222 {
223 GST_ERROR_OBJECT (depayload, "no clock-rate specified");
224 return FALSE;
225 }
226}
227
228static GstBuffer *
Sebastian Drögeb04f9722015-08-19 13:19:08 +0300229gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
Olivier Naudan988c6f02012-04-16 07:16:25 -0400230{
231 GstRtpL16Depay *rtpL16depay;
232 GstBuffer *outbuf;
233 gint payload_len;
234 gboolean marker;
Olivier Naudan988c6f02012-04-16 07:16:25 -0400235
236 rtpL16depay = GST_RTP_L16_DEPAY (depayload);
237
Sebastian Drögeb04f9722015-08-19 13:19:08 +0300238 payload_len = gst_rtp_buffer_get_payload_len (rtp);
Olivier Naudan988c6f02012-04-16 07:16:25 -0400239
240 if (payload_len <= 0)
241 goto empty_packet;
242
243 GST_DEBUG_OBJECT (rtpL16depay, "got payload of %d bytes", payload_len);
244
Sebastian Drögeb04f9722015-08-19 13:19:08 +0300245 outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
246 marker = gst_rtp_buffer_get_marker (rtp);
Olivier Naudan988c6f02012-04-16 07:16:25 -0400247
248 if (marker) {
Sebastian Dröge5f0bab02013-07-14 11:42:29 +0200249 /* mark talk spurt with RESYNC */
250 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
Olivier Naudan988c6f02012-04-16 07:16:25 -0400251 }
252
253 outbuf = gst_buffer_make_writable (outbuf);
254 if (rtpL16depay->order &&
255 !gst_audio_buffer_reorder_channels (outbuf,
256 rtpL16depay->info.finfo->format, rtpL16depay->info.channels,
257 rtpL16depay->info.position, rtpL16depay->order->pos)) {
258 goto reorder_failed;
259 }
260
Sebastian Drögeb04f9722015-08-19 13:19:08 +0300261 gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpL16depay), outbuf,
262 g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
Olivier Naudan988c6f02012-04-16 07:16:25 -0400263
264 return outbuf;
265
266 /* ERRORS */
267empty_packet:
268 {
269 GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
270 ("Empty Payload."), (NULL));
Olivier Naudan988c6f02012-04-16 07:16:25 -0400271 return NULL;
272 }
273reorder_failed:
274 {
275 GST_ELEMENT_ERROR (rtpL16depay, STREAM, DECODE,
276 ("Channel reordering failed."), (NULL));
Olivier Naudan988c6f02012-04-16 07:16:25 -0400277 return NULL;
278 }
279}
280
281gboolean
282gst_rtp_L16_depay_plugin_init (GstPlugin * plugin)
283{
284 return gst_element_register (plugin, "rtpL16depay",
285 GST_RANK_SECONDARY, GST_TYPE_RTP_L16_DEPAY);
286}