1 /* GStreamer
2 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
3 * Copyright (C) <2005> Edgard Lima <edgard.lima@gmail.com>
4 * Copyright (C) <2005> Zeeshan Ali <zeenix@gmail.com>
5 *
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.
10 *
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.
15 *
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.
20 */
21
22 #ifdef HAVE_CONFIG_H
23 # include "config.h"
24 #endif
25
26 #include <string.h>
27 #include <gst/rtp/gstrtpbuffer.h>
28 #include <gst/audio/audio.h>
29 #include "gstrtppcmudepay.h"
30 #include "gstrtputils.h"
31
32 /* RtpPcmuDepay signals and args */
33 enum
34 {
35 /* FILL ME */
36 LAST_SIGNAL
37 };
38
39 enum
40 {
41 PROP_0
42 };
43
44 static GstStaticPadTemplate gst_rtp_pcmu_depay_sink_template =
45 GST_STATIC_PAD_TEMPLATE ("sink",
46 GST_PAD_SINK,
47 GST_PAD_ALWAYS,
48 GST_STATIC_CAPS ("application/x-rtp, "
49 "media = (string) \"audio\", "
50 "payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", "
51 "clock-rate = (int) 8000; "
52 "application/x-rtp, "
53 "media = (string) \"audio\", "
54 "encoding-name = (string) \"PCMU\", clock-rate = (int) [1, MAX ]")
55 );
56
57 static GstStaticPadTemplate gst_rtp_pcmu_depay_src_template =
58 GST_STATIC_PAD_TEMPLATE ("src",
59 GST_PAD_SRC,
60 GST_PAD_ALWAYS,
61 GST_STATIC_CAPS ("audio/x-mulaw, "
62 "channels = (int) 1, rate = (int) [1, MAX ]")
63 );
64
65 static GstBuffer *gst_rtp_pcmu_depay_process (GstRTPBaseDepayload * depayload,
66 GstRTPBuffer * rtp);
67 static gboolean gst_rtp_pcmu_depay_setcaps (GstRTPBaseDepayload * depayload,
68 GstCaps * caps);
69
70 #define gst_rtp_pcmu_depay_parent_class parent_class
71 G_DEFINE_TYPE (GstRtpPcmuDepay, gst_rtp_pcmu_depay,
72 GST_TYPE_RTP_BASE_DEPAYLOAD);
73
74 static void
gst_rtp_pcmu_depay_class_init(GstRtpPcmuDepayClass * klass)75 gst_rtp_pcmu_depay_class_init (GstRtpPcmuDepayClass * klass)
76 {
77 GstElementClass *gstelement_class;
78 GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
79
80 gstelement_class = (GstElementClass *) klass;
81 gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
82
83 gst_element_class_add_static_pad_template (gstelement_class,
84 &gst_rtp_pcmu_depay_src_template);
85 gst_element_class_add_static_pad_template (gstelement_class,
86 &gst_rtp_pcmu_depay_sink_template);
87
88 gst_element_class_set_static_metadata (gstelement_class,
89 "RTP PCMU depayloader", "Codec/Depayloader/Network/RTP",
90 "Extracts PCMU audio from RTP packets",
91 "Edgard Lima <edgard.lima@gmail.com>, Zeeshan Ali <zeenix@gmail.com>");
92
93 gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_pcmu_depay_process;
94 gstrtpbasedepayload_class->set_caps = gst_rtp_pcmu_depay_setcaps;
95 }
96
97 static void
gst_rtp_pcmu_depay_init(GstRtpPcmuDepay * rtppcmudepay)98 gst_rtp_pcmu_depay_init (GstRtpPcmuDepay * rtppcmudepay)
99 {
100 GstRTPBaseDepayload *depayload;
101
102 depayload = GST_RTP_BASE_DEPAYLOAD (rtppcmudepay);
103
104 gst_pad_use_fixed_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload));
105 }
106
107 static gboolean
gst_rtp_pcmu_depay_setcaps(GstRTPBaseDepayload * depayload,GstCaps * caps)108 gst_rtp_pcmu_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
109 {
110 GstCaps *srccaps;
111 GstStructure *structure;
112 gboolean ret;
113 gint clock_rate;
114
115 structure = gst_caps_get_structure (caps, 0);
116
117 if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
118 clock_rate = 8000; /* default */
119 depayload->clock_rate = clock_rate;
120
121 srccaps = gst_caps_new_simple ("audio/x-mulaw",
122 "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, clock_rate, NULL);
123 ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
124 gst_caps_unref (srccaps);
125
126 return ret;
127 }
128
129 static GstBuffer *
gst_rtp_pcmu_depay_process(GstRTPBaseDepayload * depayload,GstRTPBuffer * rtp)130 gst_rtp_pcmu_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
131 {
132 GstBuffer *outbuf = NULL;
133 guint len;
134 gboolean marker;
135
136 marker = gst_rtp_buffer_get_marker (rtp);
137
138 GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
139 gst_buffer_get_size (rtp->buffer), marker,
140 gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
141
142 len = gst_rtp_buffer_get_payload_len (rtp);
143 outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
144
145 if (outbuf) {
146 GST_BUFFER_DURATION (outbuf) =
147 gst_util_uint64_scale_int (len, GST_SECOND, depayload->clock_rate);
148
149 if (marker) {
150 /* mark start of talkspurt with RESYNC */
151 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
152 }
153
154 gst_rtp_drop_non_audio_meta (depayload, outbuf);
155 }
156
157 return outbuf;
158 }
159
160 gboolean
gst_rtp_pcmu_depay_plugin_init(GstPlugin * plugin)161 gst_rtp_pcmu_depay_plugin_init (GstPlugin * plugin)
162 {
163 return gst_element_register (plugin, "rtppcmudepay",
164 GST_RANK_SECONDARY, GST_TYPE_RTP_PCMU_DEPAY);
165 }
166