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>
28 #include <gst/audio/audio.h>
30 #include "gstrtpelements.h"
31 #include "gstrtpgsmpay.h"
32 #include "gstrtputils.h"
34 GST_DEBUG_CATEGORY_STATIC (rtpgsmpay_debug);
35 #define GST_CAT_DEFAULT (rtpgsmpay_debug)
37 static GstStaticPadTemplate gst_rtp_gsm_pay_sink_template =
38 GST_STATIC_PAD_TEMPLATE ("sink",
41 GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1")
44 static GstStaticPadTemplate gst_rtp_gsm_pay_src_template =
45 GST_STATIC_PAD_TEMPLATE ("src",
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\"; "
53 "media = (string) \"audio\", "
54 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
55 "clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"")
58 static gboolean gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload,
60 static GstFlowReturn gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * payload,
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));
69 gst_rtp_gsm_pay_class_init (GstRTPGSMPayClass * klass)
71 GstElementClass *gstelement_class;
72 GstRTPBasePayloadClass *gstrtpbasepayload_class;
74 GST_DEBUG_CATEGORY_INIT (rtpgsmpay_debug, "rtpgsmpay", 0,
75 "GSM Audio RTP Payloader");
77 gstelement_class = (GstElementClass *) klass;
78 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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);
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>");
90 gstrtpbasepayload_class->set_caps = gst_rtp_gsm_pay_setcaps;
91 gstrtpbasepayload_class->handle_buffer = gst_rtp_gsm_pay_handle_buffer;
95 gst_rtp_gsm_pay_init (GstRTPGSMPay * rtpgsmpay)
97 GST_RTP_BASE_PAYLOAD (rtpgsmpay)->clock_rate = 8000;
98 GST_RTP_BASE_PAYLOAD_PT (rtpgsmpay) = GST_RTP_PAYLOAD_GSM;
102 gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
105 GstStructure *structure;
108 structure = gst_caps_get_structure (caps, 0);
110 stname = gst_structure_get_name (structure);
112 if (strcmp ("audio/x-gsm", stname))
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);
124 GST_WARNING_OBJECT (payload, "invalid media type received");
130 gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * basepayload,
133 GstRTPGSMPay *rtpgsmpay;
136 GstClockTime timestamp, duration;
139 rtpgsmpay = GST_RTP_GSM_PAY (basepayload);
141 timestamp = GST_BUFFER_PTS (buffer);
142 duration = GST_BUFFER_DURATION (buffer);
144 /* FIXME, only one GSM frame per RTP packet for now */
145 payload_len = gst_buffer_get_size (buffer);
147 /* FIXME, just error out for now */
148 if (payload_len > GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay))
151 outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
153 /* copy timestamp and duration */
154 GST_BUFFER_PTS (outbuf) = timestamp;
155 GST_BUFFER_DURATION (outbuf) = duration;
157 gst_rtp_copy_audio_meta (rtpgsmpay, outbuf, buffer);
160 outbuf = gst_buffer_append (outbuf, buffer);
162 GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %" G_GSIZE_FORMAT,
163 gst_buffer_get_size (outbuf));
165 ret = gst_rtp_base_payload_push (basepayload, outbuf);
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;