2 * Copyright (C) <2009> Wim Taymans <wim.taymans@gmail.com>
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.
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.
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.
21 * SECTION:element-rtpbvpay
23 * @see_also: rtpbvdepay
25 * Payload BroadcomVoice audio into RTP packets according to RFC 4298.
26 * For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt
36 #include <gst/rtp/gstrtpbuffer.h>
37 #include "gstrtpbvpay.h"
39 GST_DEBUG_CATEGORY_STATIC (rtpbvpay_debug);
40 #define GST_CAT_DEFAULT (rtpbvpay_debug)
42 static GstStaticPadTemplate gst_rtp_bv_pay_sink_template =
43 GST_STATIC_PAD_TEMPLATE ("sink",
46 GST_STATIC_CAPS ("audio/x-bv, " "mode = (int) {16, 32}")
49 static GstStaticPadTemplate gst_rtp_bv_pay_src_template =
50 GST_STATIC_PAD_TEMPLATE ("src",
53 GST_STATIC_CAPS ("application/x-rtp, "
54 "media = (string) \"audio\", "
55 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
56 "clock-rate = (int) 8000, "
57 "encoding-name = (string) \"BV16\";"
59 "media = (string) \"audio\", "
60 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
61 "clock-rate = (int) 16000, " "encoding-name = (string) \"BV32\"")
65 static GstCaps *gst_rtp_bv_pay_sink_getcaps (GstRTPBasePayload * payload,
66 GstPad * pad, GstCaps * filter);
67 static gboolean gst_rtp_bv_pay_sink_setcaps (GstRTPBasePayload * payload,
70 #define gst_rtp_bv_pay_parent_class parent_class
71 G_DEFINE_TYPE (GstRTPBVPay, gst_rtp_bv_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
74 gst_rtp_bv_pay_class_init (GstRTPBVPayClass * klass)
76 GstElementClass *gstelement_class;
77 GstRTPBasePayloadClass *gstrtpbasepayload_class;
79 GST_DEBUG_CATEGORY_INIT (rtpbvpay_debug, "rtpbvpay", 0,
80 "BroadcomVoice audio RTP payloader");
82 gstelement_class = (GstElementClass *) klass;
83 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
85 gst_element_class_add_static_pad_template (gstelement_class,
86 &gst_rtp_bv_pay_sink_template);
87 gst_element_class_add_static_pad_template (gstelement_class,
88 &gst_rtp_bv_pay_src_template);
90 gst_element_class_set_static_metadata (gstelement_class, "RTP BV Payloader",
91 "Codec/Payloader/Network/RTP",
92 "Packetize BroadcomVoice audio streams into RTP packets (RFC 4298)",
93 "Wim Taymans <wim.taymans@collabora.co.uk>");
95 gstrtpbasepayload_class->set_caps = gst_rtp_bv_pay_sink_setcaps;
96 gstrtpbasepayload_class->get_caps = gst_rtp_bv_pay_sink_getcaps;
100 gst_rtp_bv_pay_init (GstRTPBVPay * rtpbvpay)
102 GstRTPBaseAudioPayload *rtpbaseaudiopayload;
104 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbvpay);
108 /* tell rtpbaseaudiopayload that this is a frame based codec */
109 gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload);
113 gst_rtp_bv_pay_sink_setcaps (GstRTPBasePayload * rtpbasepayload, GstCaps * caps)
115 GstRTPBVPay *rtpbvpay;
116 GstRTPBaseAudioPayload *rtpbaseaudiopayload;
118 GstStructure *structure;
119 const char *payload_name;
121 rtpbvpay = GST_RTP_BV_PAY (rtpbasepayload);
122 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload);
124 structure = gst_caps_get_structure (caps, 0);
126 payload_name = gst_structure_get_name (structure);
127 if (g_ascii_strcasecmp ("audio/x-bv", payload_name))
130 if (!gst_structure_get_int (structure, "mode", &mode))
133 if (mode != 16 && mode != 32)
137 gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "BV16",
139 rtpbasepayload->clock_rate = 8000;
141 gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "BV32",
143 rtpbasepayload->clock_rate = 16000;
146 /* set options for this frame based audio codec */
147 gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload,
148 mode, mode == 16 ? 10 : 20);
150 if (mode != rtpbvpay->mode && rtpbvpay->mode != -1)
153 rtpbvpay->mode = mode;
160 GST_ERROR_OBJECT (rtpbvpay, "expected audio/x-bv, received %s",
166 GST_ERROR_OBJECT (rtpbvpay, "did not receive a mode");
171 GST_ERROR_OBJECT (rtpbvpay, "mode must be 16 or 32, received %d", mode);
176 GST_ERROR_OBJECT (rtpbvpay, "Mode has changed from %d to %d! "
177 "Mode cannot change while streaming", rtpbvpay->mode, mode);
182 /* we return the padtemplate caps with the mode field fixated to a value if we
185 gst_rtp_bv_pay_sink_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
188 GstCaps *otherpadcaps;
191 caps = gst_pad_get_pad_template_caps (pad);
193 otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
195 if (!gst_caps_is_empty (otherpadcaps)) {
196 GstStructure *structure;
197 const gchar *mode_str;
200 structure = gst_caps_get_structure (otherpadcaps, 0);
202 /* construct mode, if we can */
203 mode_str = gst_structure_get_string (structure, "encoding-name");
205 if (!strcmp (mode_str, "BV16"))
207 else if (!strcmp (mode_str, "BV32"))
212 if (mode == 16 || mode == 32) {
213 caps = gst_caps_make_writable (caps);
214 structure = gst_caps_get_structure (caps, 0);
215 gst_structure_set (structure, "mode", G_TYPE_INT, mode, NULL);
219 gst_caps_unref (otherpadcaps);
225 GST_DEBUG_OBJECT (rtppayload, "Intersect %" GST_PTR_FORMAT " and filter %"
226 GST_PTR_FORMAT, caps, filter);
227 tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
228 gst_caps_unref (caps);
236 gst_rtp_bv_pay_plugin_init (GstPlugin * plugin)
238 return gst_element_register (plugin, "rtpbvpay",
239 GST_RANK_SECONDARY, GST_TYPE_RTP_BV_PAY);