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
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.
42 #include <gst/audio/audio.h>
43 #include <gst/rtp/gstrtpbuffer.h>
45 #include "gstrtpL8pay.h"
46 #include "gstrtpchannels.h"
48 GST_DEBUG_CATEGORY_STATIC (rtpL8pay_debug);
49 #define GST_CAT_DEFAULT (rtpL8pay_debug)
51 static GstStaticPadTemplate gst_rtp_L8_pay_sink_template =
52 GST_STATIC_PAD_TEMPLATE ("sink",
55 GST_STATIC_CAPS ("audio/x-raw, "
56 "format = (string) U8, "
57 "layout = (string) interleaved, "
58 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
61 static GstStaticPadTemplate gst_rtp_L8_pay_src_template =
62 GST_STATIC_PAD_TEMPLATE ("src",
65 GST_STATIC_CAPS ("application/x-rtp, "
66 "media = (string) audio, "
67 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
68 "clock-rate = (int) [ 1, MAX ], "
69 "encoding-name = (string) L8, " "channels = (int) [ 1, MAX ];")
72 static gboolean gst_rtp_L8_pay_setcaps (GstRTPBasePayload * basepayload,
74 static GstCaps *gst_rtp_L8_pay_getcaps (GstRTPBasePayload * rtppayload,
75 GstPad * pad, GstCaps * filter);
77 gst_rtp_L8_pay_handle_buffer (GstRTPBasePayload * basepayload,
80 #define gst_rtp_L8_pay_parent_class parent_class
81 G_DEFINE_TYPE (GstRtpL8Pay, gst_rtp_L8_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
84 gst_rtp_L8_pay_class_init (GstRtpL8PayClass * 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_L8_pay_setcaps;
93 gstrtpbasepayload_class->get_caps = gst_rtp_L8_pay_getcaps;
94 gstrtpbasepayload_class->handle_buffer = gst_rtp_L8_pay_handle_buffer;
96 gst_element_class_add_pad_template (gstelement_class,
97 gst_static_pad_template_get (&gst_rtp_L8_pay_src_template));
98 gst_element_class_add_pad_template (gstelement_class,
99 gst_static_pad_template_get (&gst_rtp_L8_pay_sink_template));
101 gst_element_class_set_static_metadata (gstelement_class,
102 "RTP audio payloader", "Codec/Payloader/Network/RTP",
103 "Payload-encode Raw audio into RTP packets (RFC 3551)",
104 "Wim Taymans <wim.taymans@gmail.com>, "
105 "GE Intelligent Platforms Embedded Systems, Inc.");
107 GST_DEBUG_CATEGORY_INIT (rtpL8pay_debug, "rtpL8pay", 0, "L8 RTP Payloader");
111 gst_rtp_L8_pay_init (GstRtpL8Pay * rtpL8pay)
113 GstRTPBaseAudioPayload *rtpbaseaudiopayload;
115 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL8pay);
117 /* tell rtpbaseaudiopayload that this is a sample based codec */
118 gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
122 gst_rtp_L8_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
124 GstRtpL8Pay *rtpL8pay;
128 const GstRTPChannelOrder *order;
129 GstRTPBaseAudioPayload *rtpbaseaudiopayload;
131 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
132 rtpL8pay = GST_RTP_L8_PAY (basepayload);
134 info = &rtpL8pay->info;
135 gst_audio_info_init (info);
136 if (!gst_audio_info_from_caps (info, caps))
139 order = gst_rtp_channels_get_by_pos (info->channels, info->position);
140 rtpL8pay->order = order;
142 gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L8",
144 params = g_strdup_printf ("%d", info->channels);
146 if (!order && info->channels > 2) {
147 GST_ELEMENT_WARNING (rtpL8pay, STREAM, DECODE,
148 (NULL), ("Unknown channel order for %d channels", info->channels));
151 if (order && order->name) {
152 res = gst_rtp_base_payload_set_outcaps (basepayload,
153 "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
154 info->channels, "channel-order", G_TYPE_STRING, order->name, NULL);
156 res = gst_rtp_base_payload_set_outcaps (basepayload,
157 "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
158 info->channels, NULL);
163 /* octet-per-sample is # channels for L8 */
164 gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload,
172 GST_DEBUG_OBJECT (rtpL8pay, "invalid caps");
178 gst_rtp_L8_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
181 GstCaps *otherpadcaps;
184 caps = gst_pad_get_pad_template_caps (pad);
186 otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
188 if (!gst_caps_is_empty (otherpadcaps)) {
189 GstStructure *structure;
193 structure = gst_caps_get_structure (otherpadcaps, 0);
194 caps = gst_caps_make_writable (caps);
196 if (gst_structure_get_int (structure, "channels", &channels)) {
197 gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);
200 if (gst_structure_get_int (structure, "clock-rate", &rate)) {
201 gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL);
205 gst_caps_unref (otherpadcaps);
209 GstCaps *tcaps = caps;
211 caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST);
212 gst_caps_unref (tcaps);
219 gst_rtp_L8_pay_handle_buffer (GstRTPBasePayload * basepayload,
222 GstRtpL8Pay *rtpL8pay;
224 rtpL8pay = GST_RTP_L8_PAY (basepayload);
225 buffer = gst_buffer_make_writable (buffer);
227 if (rtpL8pay->order &&
228 !gst_audio_buffer_reorder_channels (buffer, rtpL8pay->info.finfo->format,
229 rtpL8pay->info.channels, rtpL8pay->info.position,
230 rtpL8pay->order->pos)) {
231 return GST_FLOW_ERROR;
234 return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload,
239 gst_rtp_L8_pay_plugin_init (GstPlugin * plugin)
241 return gst_element_register (plugin, "rtpL8pay",
242 GST_RANK_SECONDARY, GST_TYPE_RTP_L8_PAY);