2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
3 * Copyright (C) <2015> GE Intelligent Platforms Embedded Systems, Inc.
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.
22 * SECTION:element-rtpL8pay
23 * @see_also: rtpL8depay
25 * Payload raw audio into RTP packets according to RFC 3551.
26 * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
29 * <title>Example pipeline</title>
31 * gst-launch -v audiotestsrc ! audioconvert ! rtpL8pay ! udpsink
32 * ]| This example pipeline will payload raw audio. Refer to
33 * the rtpL8depay example to depayload and play the RTP stream.
43 #include <gst/audio/audio.h>
44 #include <gst/rtp/gstrtpbuffer.h>
46 #include "gstrtpL8pay.h"
47 #include "gstrtpchannels.h"
49 GST_DEBUG_CATEGORY_STATIC (rtpL8pay_debug);
50 #define GST_CAT_DEFAULT (rtpL8pay_debug)
52 static GstStaticPadTemplate gst_rtp_L8_pay_sink_template =
53 GST_STATIC_PAD_TEMPLATE ("sink",
56 GST_STATIC_CAPS ("audio/x-raw, "
57 "format = (string) U8, "
58 "layout = (string) interleaved, "
59 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
62 static GstStaticPadTemplate gst_rtp_L8_pay_src_template =
63 GST_STATIC_PAD_TEMPLATE ("src",
66 GST_STATIC_CAPS ("application/x-rtp, "
67 "media = (string) audio, "
68 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
69 "clock-rate = (int) [ 1, MAX ], "
70 "encoding-name = (string) L8, " "channels = (int) [ 1, MAX ];")
73 static gboolean gst_rtp_L8_pay_setcaps (GstRTPBasePayload * basepayload,
75 static GstCaps *gst_rtp_L8_pay_getcaps (GstRTPBasePayload * rtppayload,
76 GstPad * pad, GstCaps * filter);
78 gst_rtp_L8_pay_handle_buffer (GstRTPBasePayload * basepayload,
81 #define gst_rtp_L8_pay_parent_class parent_class
82 G_DEFINE_TYPE (GstRtpL8Pay, gst_rtp_L8_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
85 gst_rtp_L8_pay_class_init (GstRtpL8PayClass * klass)
87 GstElementClass *gstelement_class;
88 GstRTPBasePayloadClass *gstrtpbasepayload_class;
90 gstelement_class = (GstElementClass *) klass;
91 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
93 gstrtpbasepayload_class->set_caps = gst_rtp_L8_pay_setcaps;
94 gstrtpbasepayload_class->get_caps = gst_rtp_L8_pay_getcaps;
95 gstrtpbasepayload_class->handle_buffer = gst_rtp_L8_pay_handle_buffer;
97 gst_element_class_add_pad_template (gstelement_class,
98 gst_static_pad_template_get (&gst_rtp_L8_pay_src_template));
99 gst_element_class_add_pad_template (gstelement_class,
100 gst_static_pad_template_get (&gst_rtp_L8_pay_sink_template));
102 gst_element_class_set_static_metadata (gstelement_class,
103 "RTP audio payloader", "Codec/Payloader/Network/RTP",
104 "Payload-encode Raw audio into RTP packets (RFC 3551)",
105 "Wim Taymans <wim.taymans@gmail.com>, "
106 "GE Intelligent Platforms Embedded Systems, Inc.");
108 GST_DEBUG_CATEGORY_INIT (rtpL8pay_debug, "rtpL8pay", 0, "L8 RTP Payloader");
112 gst_rtp_L8_pay_init (GstRtpL8Pay * rtpL8pay)
114 GstRTPBaseAudioPayload *rtpbaseaudiopayload;
116 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL8pay);
118 /* tell rtpbaseaudiopayload that this is a sample based codec */
119 gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
123 gst_rtp_L8_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
125 GstRtpL8Pay *rtpL8pay;
129 const GstRTPChannelOrder *order;
130 GstRTPBaseAudioPayload *rtpbaseaudiopayload;
132 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
133 rtpL8pay = GST_RTP_L8_PAY (basepayload);
135 info = &rtpL8pay->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 rtpL8pay->order = order;
143 gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L8",
145 params = g_strdup_printf ("%d", info->channels);
147 if (!order && info->channels > 2) {
148 GST_ELEMENT_WARNING (rtpL8pay, 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 # channels for L8 */
165 gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload,
173 GST_DEBUG_OBJECT (rtpL8pay, "invalid caps");
179 gst_rtp_L8_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_L8_pay_handle_buffer (GstRTPBasePayload * basepayload,
223 GstRtpL8Pay *rtpL8pay;
225 rtpL8pay = GST_RTP_L8_PAY (basepayload);
226 buffer = gst_buffer_make_writable (buffer);
228 if (rtpL8pay->order &&
229 !gst_audio_buffer_reorder_channels (buffer, rtpL8pay->info.finfo->format,
230 rtpL8pay->info.channels, rtpL8pay->info.position,
231 rtpL8pay->order->pos)) {
232 return GST_FLOW_ERROR;
235 return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload,
240 gst_rtp_L8_pay_plugin_init (GstPlugin * plugin)
242 return gst_element_register (plugin, "rtpL8pay",
243 GST_RANK_SECONDARY, GST_TYPE_RTP_L8_PAY);