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/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* 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-rtpL24pay
* @title: rtpL24pay
* @see_also: rtpL24depay
*
* Payload raw 24-bit audio into RTP packets according to RFC 3190, section 4.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc3190.txt
*
* ## Example pipeline
* |[
* gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL24pay ! udpsink
* ]| This example pipeline will payload raw audio. Refer to
* the rtpL24depay example to depayload and play the RTP stream.
*
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/audio/audio.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpelements.h"
#include "gstrtpL24pay.h"
#include "gstrtpchannels.h"
GST_DEBUG_CATEGORY_STATIC (rtpL24pay_debug);
#define GST_CAT_DEFAULT (rtpL24pay_debug)
static GstStaticPadTemplate gst_rtp_L24_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) S24BE, "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
static GstStaticPadTemplate gst_rtp_L24_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) [ 96, 127 ], "
"clock-rate = (int) [ 1, MAX ], "
"encoding-name = (string) \"L24\", " "channels = (int) [ 1, MAX ];")
);
static gboolean gst_rtp_L24_pay_setcaps (GstRTPBasePayload * basepayload,
GstCaps * caps);
static GstCaps *gst_rtp_L24_pay_getcaps (GstRTPBasePayload * rtppayload,
GstPad * pad, GstCaps * filter);
static GstFlowReturn
gst_rtp_L24_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer);
#define gst_rtp_L24_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpL24Pay, gst_rtp_L24_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpL24pay, "rtpL24pay",
GST_RANK_SECONDARY, GST_TYPE_RTP_L24_PAY, rtp_element_init (plugin));
static void
gst_rtp_L24_pay_class_init (GstRtpL24PayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gstrtpbasepayload_class->set_caps = gst_rtp_L24_pay_setcaps;
gstrtpbasepayload_class->get_caps = gst_rtp_L24_pay_getcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_L24_pay_handle_buffer;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_L24_pay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_L24_pay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP audio payloader", "Codec/Payloader/Network/RTP",
"Payload-encode Raw 24-bit audio into RTP packets (RFC 3190)",
"Wim Taymans <wim.taymans@gmail.com>,"
"David Holroyd <dave@badgers-in-foil.co.uk>");
GST_DEBUG_CATEGORY_INIT (rtpL24pay_debug, "rtpL24pay", 0,
"L24 RTP Payloader");
}
static void
gst_rtp_L24_pay_init (GstRtpL24Pay * rtpL24pay)
{
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL24pay);
/* tell rtpbaseaudiopayload that this is a sample based codec */
gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
}
static gboolean
gst_rtp_L24_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
{
GstRtpL24Pay *rtpL24pay;
gboolean res;
gchar *params;
GstAudioInfo *info;
const GstRTPChannelOrder *order;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
rtpL24pay = GST_RTP_L24_PAY (basepayload);
info = &rtpL24pay->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);
rtpL24pay->order = order;
gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L24",
info->rate);
params = g_strdup_printf ("%d", info->channels);
if (!order && info->channels > 2) {
GST_ELEMENT_WARNING (rtpL24pay, 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 3 * channels for L24 */
gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload,
3 * info->channels);
return res;
/* ERRORS */
invalid_caps:
{
GST_DEBUG_OBJECT (rtpL24pay, "invalid caps");
return FALSE;
}
}
static GstCaps *
gst_rtp_L24_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);
}
if (gst_structure_get_int (structure, "clock-rate", &rate)) {
gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL);
}
}
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_L24_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpL24Pay *rtpL24pay;
rtpL24pay = GST_RTP_L24_PAY (basepayload);
buffer = gst_buffer_make_writable (buffer);
if (rtpL24pay->order &&
!gst_audio_buffer_reorder_channels (buffer, rtpL24pay->info.finfo->format,
rtpL24pay->info.channels, rtpL24pay->info.position,
rtpL24pay->order->pos)) {
return GST_FLOW_ERROR;
}
return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload,
buffer);
}
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