1 /* GStreamer
2 * Copyright (C) <2005> 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 #ifdef HAVE_CONFIG_H
21 # include "config.h"
22 #endif
23
24 #include <gst/rtp/gstrtpbuffer.h>
25 #include <gst/audio/audio.h>
26
27 #include <string.h>
28 #include "gstrtpelements.h"
29 #include "gstrtpmpadepay.h"
30 #include "gstrtputils.h"
31
32 GST_DEBUG_CATEGORY_STATIC (rtpmpadepay_debug);
33 #define GST_CAT_DEFAULT (rtpmpadepay_debug)
34
35 static GstStaticPadTemplate gst_rtp_mpa_depay_src_template =
36 GST_STATIC_PAD_TEMPLATE ("src",
37 GST_PAD_SRC,
38 GST_PAD_ALWAYS,
39 GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1")
40 );
41
42 static GstStaticPadTemplate gst_rtp_mpa_depay_sink_template =
43 GST_STATIC_PAD_TEMPLATE ("sink",
44 GST_PAD_SINK,
45 GST_PAD_ALWAYS,
46 GST_STATIC_CAPS ("application/x-rtp, "
47 "media = (string) \"audio\", "
48 "payload = (int) " GST_RTP_PAYLOAD_MPA_STRING ", "
49 "clock-rate = (int) 90000 ;"
50 "application/x-rtp, "
51 "media = (string) \"audio\", "
52 "encoding-name = (string) \"MPA\", clock-rate = (int) [1, MAX]")
53 );
54
55 #define gst_rtp_mpa_depay_parent_class parent_class
56 G_DEFINE_TYPE (GstRtpMPADepay, gst_rtp_mpa_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
57 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpmpadepay, "rtpmpadepay",
58 GST_RANK_SECONDARY, GST_TYPE_RTP_MPA_DEPAY, rtp_element_init (plugin));
59
60 static gboolean gst_rtp_mpa_depay_setcaps (GstRTPBaseDepayload * depayload,
61 GstCaps * caps);
62 static GstBuffer *gst_rtp_mpa_depay_process (GstRTPBaseDepayload * depayload,
63 GstRTPBuffer * rtp);
64
65 static void
gst_rtp_mpa_depay_class_init(GstRtpMPADepayClass * klass)66 gst_rtp_mpa_depay_class_init (GstRtpMPADepayClass * klass)
67 {
68 GstElementClass *gstelement_class;
69 GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
70
71 GST_DEBUG_CATEGORY_INIT (rtpmpadepay_debug, "rtpmpadepay", 0,
72 "MPEG Audio RTP Depayloader");
73
74 gstelement_class = (GstElementClass *) klass;
75 gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
76
77 gst_element_class_add_static_pad_template (gstelement_class,
78 &gst_rtp_mpa_depay_src_template);
79 gst_element_class_add_static_pad_template (gstelement_class,
80 &gst_rtp_mpa_depay_sink_template);
81
82 gst_element_class_set_static_metadata (gstelement_class,
83 "RTP MPEG audio depayloader", "Codec/Depayloader/Network/RTP",
84 "Extracts MPEG audio from RTP packets (RFC 2038)",
85 "Wim Taymans <wim.taymans@gmail.com>");
86
87 gstrtpbasedepayload_class->set_caps = gst_rtp_mpa_depay_setcaps;
88 gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_mpa_depay_process;
89 }
90
91 static void
gst_rtp_mpa_depay_init(GstRtpMPADepay * rtpmpadepay)92 gst_rtp_mpa_depay_init (GstRtpMPADepay * rtpmpadepay)
93 {
94 }
95
96 static gboolean
gst_rtp_mpa_depay_setcaps(GstRTPBaseDepayload * depayload,GstCaps * caps)97 gst_rtp_mpa_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
98 {
99 GstStructure *structure;
100 GstCaps *outcaps;
101 gint clock_rate;
102 gboolean res;
103
104 structure = gst_caps_get_structure (caps, 0);
105
106 if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
107 clock_rate = 90000;
108 depayload->clock_rate = clock_rate;
109
110 outcaps =
111 gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 1, NULL);
112 res = gst_pad_set_caps (depayload->srcpad, outcaps);
113 gst_caps_unref (outcaps);
114
115 return res;
116 }
117
118 static GstBuffer *
gst_rtp_mpa_depay_process(GstRTPBaseDepayload * depayload,GstRTPBuffer * rtp)119 gst_rtp_mpa_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
120 {
121 GstRtpMPADepay *rtpmpadepay;
122 GstBuffer *outbuf;
123 gint payload_len;
124 #if 0
125 guint8 *payload;
126 guint16 frag_offset;
127 #endif
128 gboolean marker;
129
130 rtpmpadepay = GST_RTP_MPA_DEPAY (depayload);
131
132 payload_len = gst_rtp_buffer_get_payload_len (rtp);
133
134 if (payload_len <= 4)
135 goto empty_packet;
136
137 #if 0
138 payload = gst_rtp_buffer_get_payload (&rtp);
139 /* strip off header
140 *
141 * 0 1 2 3
142 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
143 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
144 * | MBZ | Frag_offset |
145 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
146 */
147 frag_offset = (payload[2] << 8) | payload[3];
148 #endif
149
150 /* subbuffer skipping the 4 header bytes */
151 outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, 4, -1);
152 marker = gst_rtp_buffer_get_marker (rtp);
153
154 if (marker) {
155 /* mark start of talkspurt with RESYNC */
156 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
157 }
158 GST_DEBUG_OBJECT (rtpmpadepay,
159 "gst_rtp_mpa_depay_chain: pushing buffer of size %" G_GSIZE_FORMAT "",
160 gst_buffer_get_size (outbuf));
161
162 if (outbuf) {
163 gst_rtp_drop_non_audio_meta (rtpmpadepay, outbuf);
164 }
165
166 /* FIXME, we can push half mpeg frames when they are split over multiple
167 * RTP packets */
168 return outbuf;
169
170 /* ERRORS */
171 empty_packet:
172 {
173 GST_ELEMENT_WARNING (rtpmpadepay, STREAM, DECODE,
174 ("Empty Payload."), (NULL));
175 return NULL;
176 }
177 }
178