1 /* GStreamer
2 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
3 * Copyright (C) <2005> Zeeshan Ali <zeenix@gmail.com>
4 *
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.
9 *
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.
14 *
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.
19 */
20
21 #ifdef HAVE_CONFIG_H
22 # include "config.h"
23 #endif
24
25 #include <stdlib.h>
26 #include <string.h>
27 #include <gst/rtp/gstrtpbuffer.h>
28 #include <gst/audio/audio.h>
29
30 #include "gstrtpgsmpay.h"
31 #include "gstrtputils.h"
32
33 GST_DEBUG_CATEGORY_STATIC (rtpgsmpay_debug);
34 #define GST_CAT_DEFAULT (rtpgsmpay_debug)
35
36 static GstStaticPadTemplate gst_rtp_gsm_pay_sink_template =
37 GST_STATIC_PAD_TEMPLATE ("sink",
38 GST_PAD_SINK,
39 GST_PAD_ALWAYS,
40 GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1")
41 );
42
43 static GstStaticPadTemplate gst_rtp_gsm_pay_src_template =
44 GST_STATIC_PAD_TEMPLATE ("src",
45 GST_PAD_SRC,
46 GST_PAD_ALWAYS,
47 GST_STATIC_CAPS ("application/x-rtp, "
48 "media = (string) \"audio\", "
49 "payload = (int) " GST_RTP_PAYLOAD_GSM_STRING ", "
50 "clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"; "
51 "application/x-rtp, "
52 "media = (string) \"audio\", "
53 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
54 "clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"")
55 );
56
57 static gboolean gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload,
58 GstCaps * caps);
59 static GstFlowReturn gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * payload,
60 GstBuffer * buffer);
61
62 #define gst_rtp_gsm_pay_parent_class parent_class
63 G_DEFINE_TYPE (GstRTPGSMPay, gst_rtp_gsm_pay, GST_TYPE_RTP_BASE_PAYLOAD);
64
65 static void
gst_rtp_gsm_pay_class_init(GstRTPGSMPayClass * klass)66 gst_rtp_gsm_pay_class_init (GstRTPGSMPayClass * klass)
67 {
68 GstElementClass *gstelement_class;
69 GstRTPBasePayloadClass *gstrtpbasepayload_class;
70
71 GST_DEBUG_CATEGORY_INIT (rtpgsmpay_debug, "rtpgsmpay", 0,
72 "GSM Audio RTP Payloader");
73
74 gstelement_class = (GstElementClass *) klass;
75 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
76
77 gst_element_class_add_static_pad_template (gstelement_class,
78 &gst_rtp_gsm_pay_sink_template);
79 gst_element_class_add_static_pad_template (gstelement_class,
80 &gst_rtp_gsm_pay_src_template);
81
82 gst_element_class_set_static_metadata (gstelement_class, "RTP GSM payloader",
83 "Codec/Payloader/Network/RTP",
84 "Payload-encodes GSM audio into a RTP packet",
85 "Zeeshan Ali <zeenix@gmail.com>");
86
87 gstrtpbasepayload_class->set_caps = gst_rtp_gsm_pay_setcaps;
88 gstrtpbasepayload_class->handle_buffer = gst_rtp_gsm_pay_handle_buffer;
89 }
90
91 static void
gst_rtp_gsm_pay_init(GstRTPGSMPay * rtpgsmpay)92 gst_rtp_gsm_pay_init (GstRTPGSMPay * rtpgsmpay)
93 {
94 GST_RTP_BASE_PAYLOAD (rtpgsmpay)->clock_rate = 8000;
95 GST_RTP_BASE_PAYLOAD_PT (rtpgsmpay) = GST_RTP_PAYLOAD_GSM;
96 }
97
98 static gboolean
gst_rtp_gsm_pay_setcaps(GstRTPBasePayload * payload,GstCaps * caps)99 gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
100 {
101 const char *stname;
102 GstStructure *structure;
103 gboolean res;
104
105 structure = gst_caps_get_structure (caps, 0);
106
107 stname = gst_structure_get_name (structure);
108
109 if (strcmp ("audio/x-gsm", stname))
110 goto invalid_type;
111
112 gst_rtp_base_payload_set_options (payload, "audio",
113 payload->pt != GST_RTP_PAYLOAD_GSM, "GSM", 8000);
114 res = gst_rtp_base_payload_set_outcaps (payload, NULL);
115
116 return res;
117
118 /* ERRORS */
119 invalid_type:
120 {
121 GST_WARNING_OBJECT (payload, "invalid media type received");
122 return FALSE;
123 }
124 }
125
126 static GstFlowReturn
gst_rtp_gsm_pay_handle_buffer(GstRTPBasePayload * basepayload,GstBuffer * buffer)127 gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * basepayload,
128 GstBuffer * buffer)
129 {
130 GstRTPGSMPay *rtpgsmpay;
131 guint payload_len;
132 GstBuffer *outbuf;
133 GstClockTime timestamp, duration;
134 GstFlowReturn ret;
135
136 rtpgsmpay = GST_RTP_GSM_PAY (basepayload);
137
138 timestamp = GST_BUFFER_PTS (buffer);
139 duration = GST_BUFFER_DURATION (buffer);
140
141 /* FIXME, only one GSM frame per RTP packet for now */
142 payload_len = gst_buffer_get_size (buffer);
143
144 /* FIXME, just error out for now */
145 if (payload_len > GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay))
146 goto too_big;
147
148 outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
149
150 /* copy timestamp and duration */
151 GST_BUFFER_PTS (outbuf) = timestamp;
152 GST_BUFFER_DURATION (outbuf) = duration;
153
154 gst_rtp_copy_audio_meta (rtpgsmpay, outbuf, buffer);
155
156 /* append payload */
157 outbuf = gst_buffer_append (outbuf, buffer);
158
159 GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %" G_GSIZE_FORMAT,
160 gst_buffer_get_size (outbuf));
161
162 ret = gst_rtp_base_payload_push (basepayload, outbuf);
163
164 return ret;
165
166 /* ERRORS */
167 too_big:
168 {
169 GST_ELEMENT_ERROR (rtpgsmpay, STREAM, ENCODE, (NULL),
170 ("payload_len %u > mtu %u", payload_len,
171 GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay)));
172 return GST_FLOW_ERROR;
173 }
174 }
175
176 gboolean
gst_rtp_gsm_pay_plugin_init(GstPlugin * plugin)177 gst_rtp_gsm_pay_plugin_init (GstPlugin * plugin)
178 {
179 return gst_element_register (plugin, "rtpgsmpay",
180 GST_RANK_SECONDARY, GST_TYPE_RTP_GSM_PAY);
181 }
182