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