1 /* 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
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
18 */
19
20 /**
21 * SECTION:element-rtpL24depay
22 * @title: rtpL24depay
23 * @see_also: rtpL24pay
24 *
25 * Extract raw audio from RTP packets according to RFC 3190, section 4.
26 * For detailed information see: http://www.rfc-editor.org/rfc/rfc3190.txt
27 *
28 * ## Example pipeline
29 * |[
30 * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L24, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL24depay ! pulsesink
31 * ]| This example pipeline will depayload an RTP raw audio stream. Refer to
32 * the rtpL24pay example to create the RTP stream.
33 *
34 */
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 "gstrtpelements.h"
46 #include "gstrtpL24depay.h"
47 #include "gstrtpchannels.h"
48 #include "gstrtputils.h"
49
50 GST_DEBUG_CATEGORY_STATIC (rtpL24depay_debug);
51 #define GST_CAT_DEFAULT (rtpL24depay_debug)
52
53 static GstStaticPadTemplate gst_rtp_L24_depay_src_template =
54 GST_STATIC_PAD_TEMPLATE ("src",
55 GST_PAD_SRC,
56 GST_PAD_ALWAYS,
57 GST_STATIC_CAPS ("audio/x-raw, "
58 "format = (string) S24BE, "
59 "layout = (string) interleaved, "
60 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
61 );
62
63 static GstStaticPadTemplate gst_rtp_L24_depay_sink_template =
64 GST_STATIC_PAD_TEMPLATE ("sink",
65 GST_PAD_SINK,
66 GST_PAD_ALWAYS,
67 GST_STATIC_CAPS ("application/x-rtp, "
68 "media = (string) \"audio\", " "clock-rate = (int) [ 1, MAX ], "
69 "encoding-name = (string) \"L24\"")
70 );
71
72 #define gst_rtp_L24_depay_parent_class parent_class
73 G_DEFINE_TYPE (GstRtpL24Depay, gst_rtp_L24_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
74 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpL24depay, "rtpL24depay",
75 GST_RANK_SECONDARY, GST_TYPE_RTP_L24_DEPAY, rtp_element_init (plugin));
76
77 static gboolean gst_rtp_L24_depay_setcaps (GstRTPBaseDepayload * depayload,
78 GstCaps * caps);
79 static GstBuffer *gst_rtp_L24_depay_process (GstRTPBaseDepayload * depayload,
80 GstRTPBuffer * rtp);
81
82 static void
gst_rtp_L24_depay_class_init(GstRtpL24DepayClass * klass)83 gst_rtp_L24_depay_class_init (GstRtpL24DepayClass * klass)
84 {
85 GstElementClass *gstelement_class;
86 GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
87
88 gstelement_class = (GstElementClass *) klass;
89 gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
90
91 gstrtpbasedepayload_class->set_caps = gst_rtp_L24_depay_setcaps;
92 gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_L24_depay_process;
93
94 gst_element_class_add_static_pad_template (gstelement_class,
95 &gst_rtp_L24_depay_src_template);
96 gst_element_class_add_static_pad_template (gstelement_class,
97 &gst_rtp_L24_depay_sink_template);
98
99 gst_element_class_set_static_metadata (gstelement_class,
100 "RTP audio depayloader", "Codec/Depayloader/Network/RTP",
101 "Extracts raw 24-bit audio from RTP packets",
102 "Zeeshan Ali <zak147@yahoo.com>," "Wim Taymans <wim.taymans@gmail.com>,"
103 "David Holroyd <dave@badgers-in-foil.co.uk>");
104
105 GST_DEBUG_CATEGORY_INIT (rtpL24depay_debug, "rtpL24depay", 0,
106 "Raw Audio RTP Depayloader");
107 }
108
109 static void
gst_rtp_L24_depay_init(GstRtpL24Depay * rtpL24depay)110 gst_rtp_L24_depay_init (GstRtpL24Depay * rtpL24depay)
111 {
112 }
113
114 static gint
gst_rtp_L24_depay_parse_int(GstStructure * structure,const gchar * field,gint def)115 gst_rtp_L24_depay_parse_int (GstStructure * structure, const gchar * field,
116 gint def)
117 {
118 const gchar *str;
119 gint res;
120
121 if ((str = gst_structure_get_string (structure, field)))
122 return atoi (str);
123
124 if (gst_structure_get_int (structure, field, &res))
125 return res;
126
127 return def;
128 }
129
130 static gboolean
gst_rtp_L24_depay_setcaps(GstRTPBaseDepayload * depayload,GstCaps * caps)131 gst_rtp_L24_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
132 {
133 GstStructure *structure;
134 GstRtpL24Depay *rtpL24depay;
135 gint clock_rate, payload;
136 gint channels;
137 GstCaps *srccaps;
138 gboolean res;
139 const gchar *channel_order;
140 const GstRTPChannelOrder *order;
141 GstAudioInfo *info;
142
143 rtpL24depay = GST_RTP_L24_DEPAY (depayload);
144
145 structure = gst_caps_get_structure (caps, 0);
146
147 payload = 96;
148 gst_structure_get_int (structure, "payload", &payload);
149 /* no fixed mapping, we need clock-rate */
150 channels = 0;
151 clock_rate = 0;
152
153 /* caps can overwrite defaults */
154 clock_rate =
155 gst_rtp_L24_depay_parse_int (structure, "clock-rate", clock_rate);
156 if (clock_rate == 0)
157 goto no_clockrate;
158
159 channels =
160 gst_rtp_L24_depay_parse_int (structure, "encoding-params", channels);
161 if (channels == 0) {
162 channels = gst_rtp_L24_depay_parse_int (structure, "channels", channels);
163 if (channels == 0) {
164 /* channels defaults to 1 otherwise */
165 channels = 1;
166 }
167 }
168
169 depayload->clock_rate = clock_rate;
170
171 info = &rtpL24depay->info;
172 gst_audio_info_init (info);
173 info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S24BE);
174 info->rate = clock_rate;
175 info->channels = channels;
176 info->bpf = (info->finfo->width / 8) * channels;
177
178 /* add channel positions */
179 channel_order = gst_structure_get_string (structure, "channel-order");
180
181 order = gst_rtp_channels_get_by_order (channels, channel_order);
182 rtpL24depay->order = order;
183 if (order) {
184 memcpy (info->position, order->pos,
185 sizeof (GstAudioChannelPosition) * channels);
186 gst_audio_channel_positions_to_valid_order (info->position, info->channels);
187 } else {
188 GST_ELEMENT_WARNING (rtpL24depay, STREAM, DECODE,
189 (NULL), ("Unknown channel order '%s' for %d channels",
190 GST_STR_NULL (channel_order), channels));
191 /* create default NONE layout */
192 gst_rtp_channels_create_default (channels, info->position);
193 info->flags |= GST_AUDIO_FLAG_UNPOSITIONED;
194 }
195
196 srccaps = gst_audio_info_to_caps (info);
197 res = gst_pad_set_caps (depayload->srcpad, srccaps);
198 gst_caps_unref (srccaps);
199
200 return res;
201
202 /* ERRORS */
203 no_clockrate:
204 {
205 GST_ERROR_OBJECT (depayload, "no clock-rate specified");
206 return FALSE;
207 }
208 }
209
210 static GstBuffer *
gst_rtp_L24_depay_process(GstRTPBaseDepayload * depayload,GstRTPBuffer * rtp)211 gst_rtp_L24_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
212 {
213 GstRtpL24Depay *rtpL24depay;
214 GstBuffer *outbuf;
215 gint payload_len;
216 gboolean marker;
217
218 rtpL24depay = GST_RTP_L24_DEPAY (depayload);
219
220 payload_len = gst_rtp_buffer_get_payload_len (rtp);
221
222 if (payload_len <= 0)
223 goto empty_packet;
224
225 GST_DEBUG_OBJECT (rtpL24depay, "got payload of %d bytes", payload_len);
226
227 outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
228 marker = gst_rtp_buffer_get_marker (rtp);
229
230 if (marker) {
231 /* mark talk spurt with RESYNC */
232 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
233 }
234
235 outbuf = gst_buffer_make_writable (outbuf);
236 if (outbuf) {
237 gst_rtp_drop_non_audio_meta (rtpL24depay, outbuf);
238 }
239 if (rtpL24depay->order &&
240 !gst_audio_buffer_reorder_channels (outbuf,
241 rtpL24depay->info.finfo->format, rtpL24depay->info.channels,
242 rtpL24depay->info.position, rtpL24depay->order->pos)) {
243 goto reorder_failed;
244 }
245
246 return outbuf;
247
248 /* ERRORS */
249 empty_packet:
250 {
251 GST_ELEMENT_WARNING (rtpL24depay, STREAM, DECODE,
252 ("Empty Payload."), (NULL));
253 return NULL;
254 }
255 reorder_failed:
256 {
257 GST_ELEMENT_ERROR (rtpL24depay, STREAM, DECODE,
258 ("Channel reordering failed."), (NULL));
259 return NULL;
260 }
261 }
262