2 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
3 * Copyright (C) <2005> Zeeshan Ali <zeenix@gmail.com>
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.
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.
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.
27 #include <gst/rtp/gstrtpbuffer.h>
29 #include "gstrtpgsmpay.h"
31 GST_DEBUG_CATEGORY_STATIC (rtpgsmpay_debug);
32 #define GST_CAT_DEFAULT (rtpgsmpay_debug)
34 static GstStaticPadTemplate gst_rtp_gsm_pay_sink_template =
35 GST_STATIC_PAD_TEMPLATE ("sink",
38 GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1")
41 static GstStaticPadTemplate gst_rtp_gsm_pay_src_template =
42 GST_STATIC_PAD_TEMPLATE ("src",
45 GST_STATIC_CAPS ("application/x-rtp, "
46 "media = (string) \"audio\", "
47 "payload = (int) " GST_RTP_PAYLOAD_GSM_STRING ", "
48 "clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"; "
50 "media = (string) \"audio\", "
51 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
52 "clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"")
55 static gboolean gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload,
57 static GstFlowReturn gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * payload,
60 #define gst_rtp_gsm_pay_parent_class parent_class
61 G_DEFINE_TYPE (GstRTPGSMPay, gst_rtp_gsm_pay, GST_TYPE_RTP_BASE_PAYLOAD);
64 gst_rtp_gsm_pay_class_init (GstRTPGSMPayClass * klass)
66 GstElementClass *gstelement_class;
67 GstRTPBasePayloadClass *gstrtpbasepayload_class;
69 GST_DEBUG_CATEGORY_INIT (rtpgsmpay_debug, "rtpgsmpay", 0,
70 "GSM Audio RTP Payloader");
72 gstelement_class = (GstElementClass *) klass;
73 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
75 gst_element_class_add_pad_template (gstelement_class,
76 gst_static_pad_template_get (&gst_rtp_gsm_pay_sink_template));
77 gst_element_class_add_pad_template (gstelement_class,
78 gst_static_pad_template_get (&gst_rtp_gsm_pay_src_template));
80 gst_element_class_set_static_metadata (gstelement_class, "RTP GSM payloader",
81 "Codec/Payloader/Network/RTP",
82 "Payload-encodes GSM audio into a RTP packet",
83 "Zeeshan Ali <zeenix@gmail.com>");
85 gstrtpbasepayload_class->set_caps = gst_rtp_gsm_pay_setcaps;
86 gstrtpbasepayload_class->handle_buffer = gst_rtp_gsm_pay_handle_buffer;
90 gst_rtp_gsm_pay_init (GstRTPGSMPay * rtpgsmpay)
92 GST_RTP_BASE_PAYLOAD (rtpgsmpay)->clock_rate = 8000;
93 GST_RTP_BASE_PAYLOAD_PT (rtpgsmpay) = GST_RTP_PAYLOAD_GSM;
97 gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
100 GstStructure *structure;
103 structure = gst_caps_get_structure (caps, 0);
105 stname = gst_structure_get_name (structure);
107 if (strcmp ("audio/x-gsm", stname))
110 gst_rtp_base_payload_set_options (payload, "audio", FALSE, "GSM", 8000);
111 res = gst_rtp_base_payload_set_outcaps (payload, NULL);
118 GST_WARNING_OBJECT (payload, "invalid media type received");
124 gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * basepayload,
127 GstRTPGSMPay *rtpgsmpay;
132 GstClockTime timestamp, duration;
134 GstRTPBuffer rtp = { NULL };
136 rtpgsmpay = GST_RTP_GSM_PAY (basepayload);
138 gst_buffer_map (buffer, &map, GST_MAP_READ);
140 timestamp = GST_BUFFER_TIMESTAMP (buffer);
141 duration = GST_BUFFER_DURATION (buffer);
143 /* FIXME, only one GSM frame per RTP packet for now */
144 payload_len = map.size;
146 /* FIXME, just error out for now */
147 if (payload_len > GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay))
150 outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
152 /* copy timestamp and duration */
153 GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
154 GST_BUFFER_DURATION (outbuf) = duration;
157 gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
159 /* copy data in payload */
160 payload = gst_rtp_buffer_get_payload (&rtp);
161 memcpy (payload, map.data, map.size);
163 gst_rtp_buffer_unmap (&rtp);
165 gst_buffer_unmap (buffer, &map);
166 gst_buffer_unref (buffer);
168 GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %" G_GSIZE_FORMAT,
169 gst_buffer_get_size (outbuf));
171 ret = gst_rtp_base_payload_push (basepayload, outbuf);
178 GST_ELEMENT_ERROR (rtpgsmpay, STREAM, ENCODE, (NULL),
179 ("payload_len %u > mtu %u", payload_len,
180 GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay)));
181 gst_buffer_unmap (buffer, &map);
182 return GST_FLOW_ERROR;
187 gst_rtp_gsm_pay_plugin_init (GstPlugin * plugin)
189 return gst_element_register (plugin, "rtpgsmpay",
190 GST_RANK_SECONDARY, GST_TYPE_RTP_GSM_PAY);