2 * Copyright (C) <2007> 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-rtpL24pay
22 * @see_also: rtpL24depay
24 * Payload raw 24-bit audio into RTP packets according to RFC 3190, section 4.
25 * For detailed information see: http://www.rfc-editor.org/rfc/rfc3190.txt
28 * <title>Example pipeline</title>
30 * gst-launch -v audiotestsrc ! audioconvert ! rtpL24pay ! udpsink
31 * ]| This example pipeline will payload raw audio. Refer to
32 * the rtpL24depay example to depayload and play the RTP stream.
42 #include <gst/audio/audio.h>
43 #include <gst/rtp/gstrtpbuffer.h>
45 #include "gstrtpL24pay.h"
46 #include "gstrtpchannels.h"
48 GST_DEBUG_CATEGORY_STATIC (rtpL24pay_debug);
49 #define GST_CAT_DEFAULT (rtpL24pay_debug)
51 static GstStaticPadTemplate gst_rtp_L24_pay_sink_template =
52 GST_STATIC_PAD_TEMPLATE ("sink",
55 GST_STATIC_CAPS ("audio/x-raw, "
56 "format = (string) S24BE, "
57 "layout = (string) interleaved, "
58 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
61 static GstStaticPadTemplate gst_rtp_L24_pay_src_template =
62 GST_STATIC_PAD_TEMPLATE ("src",
65 GST_STATIC_CAPS ("application/x-rtp, "
66 "media = (string) \"audio\", "
67 "payload = (int) [ 96, 127 ], "
68 "clock-rate = (int) [ 1, MAX ], "
69 "encoding-name = (string) \"L24\", " "channels = (int) [ 1, MAX ];")
72 static gboolean gst_rtp_L24_pay_setcaps (GstRTPBasePayload * basepayload,
74 static GstCaps *gst_rtp_L24_pay_getcaps (GstRTPBasePayload * rtppayload,
75 GstPad * pad, GstCaps * filter);
77 gst_rtp_L24_pay_handle_buffer (GstRTPBasePayload * basepayload,
80 #define gst_rtp_L24_pay_parent_class parent_class
81 G_DEFINE_TYPE (GstRtpL24Pay, gst_rtp_L24_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
84 gst_rtp_L24_pay_class_init (GstRtpL24PayClass * klass)
86 GstElementClass *gstelement_class;
87 GstRTPBasePayloadClass *gstrtpbasepayload_class;
89 gstelement_class = (GstElementClass *) klass;
90 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
92 gstrtpbasepayload_class->set_caps = gst_rtp_L24_pay_setcaps;
93 gstrtpbasepayload_class->get_caps = gst_rtp_L24_pay_getcaps;
94 gstrtpbasepayload_class->handle_buffer = gst_rtp_L24_pay_handle_buffer;
96 gst_element_class_add_pad_template (gstelement_class,
97 gst_static_pad_template_get (&gst_rtp_L24_pay_src_template));
98 gst_element_class_add_pad_template (gstelement_class,
99 gst_static_pad_template_get (&gst_rtp_L24_pay_sink_template));
101 gst_element_class_set_static_metadata (gstelement_class,
102 "RTP audio payloader", "Codec/Payloader/Network/RTP",
103 "Payload-encode Raw 24-bit audio into RTP packets (RFC 3190)",
104 "Wim Taymans <wim.taymans@gmail.com>,"
105 "David Holroyd <dave@badgers-in-foil.co.uk>");
107 GST_DEBUG_CATEGORY_INIT (rtpL24pay_debug, "rtpL24pay", 0,
108 "L24 RTP Payloader");
112 gst_rtp_L24_pay_init (GstRtpL24Pay * rtpL24pay)
114 GstRTPBaseAudioPayload *rtpbaseaudiopayload;
116 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL24pay);
118 /* tell rtpbaseaudiopayload that this is a sample based codec */
119 gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
123 gst_rtp_L24_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
125 GstRtpL24Pay *rtpL24pay;
129 const GstRTPChannelOrder *order;
130 GstRTPBaseAudioPayload *rtpbaseaudiopayload;
132 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
133 rtpL24pay = GST_RTP_L24_PAY (basepayload);
135 info = &rtpL24pay->info;
136 gst_audio_info_init (info);
137 if (!gst_audio_info_from_caps (info, caps))
140 order = gst_rtp_channels_get_by_pos (info->channels, info->position);
141 rtpL24pay->order = order;
143 gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L24",
145 params = g_strdup_printf ("%d", info->channels);
147 if (!order && info->channels > 2) {
148 GST_ELEMENT_WARNING (rtpL24pay, STREAM, DECODE,
149 (NULL), ("Unknown channel order for %d channels", info->channels));
152 if (order && order->name) {
153 res = gst_rtp_base_payload_set_outcaps (basepayload,
154 "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
155 info->channels, "channel-order", G_TYPE_STRING, order->name, NULL);
157 res = gst_rtp_base_payload_set_outcaps (basepayload,
158 "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
159 info->channels, NULL);
164 /* octet-per-sample is 3 * channels for L24 */
165 gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload,
173 GST_DEBUG_OBJECT (rtpL24pay, "invalid caps");
179 gst_rtp_L24_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
182 GstCaps *otherpadcaps;
185 caps = gst_pad_get_pad_template_caps (pad);
187 otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
189 if (!gst_caps_is_empty (otherpadcaps)) {
190 GstStructure *structure;
194 structure = gst_caps_get_structure (otherpadcaps, 0);
195 caps = gst_caps_make_writable (caps);
197 if (gst_structure_get_int (structure, "channels", &channels)) {
198 gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);
201 if (gst_structure_get_int (structure, "clock-rate", &rate)) {
202 gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL);
206 gst_caps_unref (otherpadcaps);
210 GstCaps *tcaps = caps;
212 caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST);
213 gst_caps_unref (tcaps);
220 gst_rtp_L24_pay_handle_buffer (GstRTPBasePayload * basepayload,
223 GstRtpL24Pay *rtpL24pay;
225 rtpL24pay = GST_RTP_L24_PAY (basepayload);
226 buffer = gst_buffer_make_writable (buffer);
228 if (rtpL24pay->order &&
229 !gst_audio_buffer_reorder_channels (buffer, rtpL24pay->info.finfo->format,
230 rtpL24pay->info.channels, rtpL24pay->info.position,
231 rtpL24pay->order->pos)) {
232 return GST_FLOW_ERROR;
235 return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload,
240 gst_rtp_L24_pay_plugin_init (GstPlugin * plugin)
242 return gst_element_register (plugin, "rtpL24pay",
243 GST_RANK_SECONDARY, GST_TYPE_RTP_L24_PAY);