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diff --git a/gst/rtp/gstrtpL8pay.c b/gst/rtp/gstrtpL8pay.c
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+/* GStreamer
+ * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
+ * Copyright (C) <2015> GE Intelligent Platforms Embedded Systems, Inc.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION:element-rtpL8pay
+ * @see_also: rtpL8depay
+ *
+ * Payload raw audio into RTP packets according to RFC 3551.
+ * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
+ *
+ * <refsect2>
+ * <title>Example pipeline</title>
+ * |[
+ * gst-launch -v audiotestsrc ! audioconvert ! rtpL8pay ! udpsink
+ * ]| This example pipeline will payload raw audio. Refer to
+ * the rtpL8depay example to depayload and play the RTP stream.
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <string.h>
+
+#include <gst/audio/audio.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "gstrtpL8pay.h"
+#include "gstrtpchannels.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpL8pay_debug);
+#define GST_CAT_DEFAULT (rtpL8pay_debug)
+
+static GstStaticPadTemplate gst_rtp_L8_pay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) U8, "
+ "layout = (string) interleaved, "
+ "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
+ );
+
+static GstStaticPadTemplate gst_rtp_L8_pay_src_template =
+ GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp, "
+ "media = (string) audio, "
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "clock-rate = (int) [ 1, MAX ], "
+ "encoding-name = (string) L8, " "channels = (int) [ 1, MAX ];")
+ );
+
+static gboolean gst_rtp_L8_pay_setcaps (GstRTPBasePayload * basepayload,
+ GstCaps * caps);
+static GstCaps *gst_rtp_L8_pay_getcaps (GstRTPBasePayload * rtppayload,
+ GstPad * pad, GstCaps * filter);
+static GstFlowReturn
+gst_rtp_L8_pay_handle_buffer (GstRTPBasePayload * basepayload,
+ GstBuffer * buffer);
+
+#define gst_rtp_L8_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpL8Pay, gst_rtp_L8_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
+
+static void
+gst_rtp_L8_pay_class_init (GstRtpL8PayClass * klass)
+{
+ GstElementClass *gstelement_class;
+ GstRTPBasePayloadClass *gstrtpbasepayload_class;
+
+ gstelement_class = (GstElementClass *) klass;
+ gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
+
+ gstrtpbasepayload_class->set_caps = gst_rtp_L8_pay_setcaps;
+ gstrtpbasepayload_class->get_caps = gst_rtp_L8_pay_getcaps;
+ gstrtpbasepayload_class->handle_buffer = gst_rtp_L8_pay_handle_buffer;
+
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtp_L8_pay_src_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtp_L8_pay_sink_template));
+
+ gst_element_class_set_static_metadata (gstelement_class,
+ "RTP audio payloader", "Codec/Payloader/Network/RTP",
+ "Payload-encode Raw audio into RTP packets (RFC 3551)",
+ "Wim Taymans <wim.taymans@gmail.com>, "
+ "GE Intelligent Platforms Embedded Systems, Inc.");
+
+ GST_DEBUG_CATEGORY_INIT (rtpL8pay_debug, "rtpL8pay", 0, "L8 RTP Payloader");
+}
+
+static void
+gst_rtp_L8_pay_init (GstRtpL8Pay * rtpL8pay)
+{
+ GstRTPBaseAudioPayload *rtpbaseaudiopayload;
+
+ rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL8pay);
+
+ /* tell rtpbaseaudiopayload that this is a sample based codec */
+ gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
+}
+
+static gboolean
+gst_rtp_L8_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
+{
+ GstRtpL8Pay *rtpL8pay;
+ gboolean res;
+ gchar *params;
+ GstAudioInfo *info;
+ const GstRTPChannelOrder *order;
+ GstRTPBaseAudioPayload *rtpbaseaudiopayload;
+
+ rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
+ rtpL8pay = GST_RTP_L8_PAY (basepayload);
+
+ info = &rtpL8pay->info;
+ gst_audio_info_init (info);
+ if (!gst_audio_info_from_caps (info, caps))
+ goto invalid_caps;
+
+ order = gst_rtp_channels_get_by_pos (info->channels, info->position);
+ rtpL8pay->order = order;
+
+ gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L8",
+ info->rate);
+ params = g_strdup_printf ("%d", info->channels);
+
+ if (!order && info->channels > 2) {
+ GST_ELEMENT_WARNING (rtpL8pay, STREAM, DECODE,
+ (NULL), ("Unknown channel order for %d channels", info->channels));
+ }
+
+ if (order && order->name) {
+ res = gst_rtp_base_payload_set_outcaps (basepayload,
+ "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
+ info->channels, "channel-order", G_TYPE_STRING, order->name, NULL);
+ } else {
+ res = gst_rtp_base_payload_set_outcaps (basepayload,
+ "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
+ info->channels, NULL);
+ }
+
+ g_free (params);
+
+ /* octet-per-sample is # channels for L8 */
+ gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload,
+ info->channels);
+
+ return res;
+
+ /* ERRORS */
+invalid_caps:
+ {
+ GST_DEBUG_OBJECT (rtpL8pay, "invalid caps");
+ return FALSE;
+ }
+}
+
+static GstCaps *
+gst_rtp_L8_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
+ GstCaps * filter)
+{
+ GstCaps *otherpadcaps;
+ GstCaps *caps;
+
+ caps = gst_pad_get_pad_template_caps (pad);
+
+ otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
+ if (otherpadcaps) {
+ if (!gst_caps_is_empty (otherpadcaps)) {
+ GstStructure *structure;
+ gint channels;
+ gint rate;
+
+ structure = gst_caps_get_structure (otherpadcaps, 0);
+ caps = gst_caps_make_writable (caps);
+
+ if (gst_structure_get_int (structure, "channels", &channels)) {
+ gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);
+ } else {
+ /* Support any number of channels, if not explicitly specified */
+ gst_structure_remove_field (structure, "channels");
+ }
+
+ if (gst_structure_get_int (structure, "clock-rate", &rate)) {
+ gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL);
+ } else {
+ /* Support any rate, if not explicitly specified */
+ gst_structure_remove_field (structure, "rate");
+ }
+
+ }
+ gst_caps_unref (otherpadcaps);
+ }
+
+ if (filter) {
+ GstCaps *tcaps = caps;
+
+ caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST);
+ gst_caps_unref (tcaps);
+ }
+
+ return caps;
+}
+
+static GstFlowReturn
+gst_rtp_L8_pay_handle_buffer (GstRTPBasePayload * basepayload,
+ GstBuffer * buffer)
+{
+ GstRtpL8Pay *rtpL8pay;
+
+ rtpL8pay = GST_RTP_L8_PAY (basepayload);
+ buffer = gst_buffer_make_writable (buffer);
+
+ if (rtpL8pay->order &&
+ !gst_audio_buffer_reorder_channels (buffer, rtpL8pay->info.finfo->format,
+ rtpL8pay->info.channels, rtpL8pay->info.position,
+ rtpL8pay->order->pos)) {
+ return GST_FLOW_ERROR;
+ }
+
+ return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload,
+ buffer);
+}
+
+gboolean
+gst_rtp_L8_pay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpL8pay",
+ GST_RANK_SECONDARY, GST_TYPE_RTP_L8_PAY);
+}
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