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/* GStreamer
* Copyright (C) <2010> 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/audio/audio.h>
#include <gst/audio/multichannel.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpg722pay.h"
#include "gstrtpchannels.h"
GST_DEBUG_CATEGORY_STATIC (rtpg722pay_debug);
#define GST_CAT_DEFAULT (rtpg722pay_debug)
static GstStaticPadTemplate gst_rtp_g722_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/G722, " "rate = (int) 16000, " "channels = (int) 1")
);
static GstStaticPadTemplate gst_rtp_g722_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"encoding-name = (string) \"G722\", "
"payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", "
"clock-rate = (int) 8000")
);
static gboolean gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload,
GstCaps * caps);
static GstCaps *gst_rtp_g722_pay_getcaps (GstBaseRTPPayload * rtppayload,
GstPad * pad);
GST_BOILERPLATE (GstRtpG722Pay, gst_rtp_g722_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_g722_pay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_g722_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_g722_pay_sink_template));
gst_element_class_set_details_simple (element_class, "RTP audio payloader",
"Codec/Payloader/Network/RTP",
"Payload-encode Raw audio into RTP packets (RFC 3551)",
"Wim Taymans <wim.taymans@gmail.com>");
}
static void
gst_rtp_g722_pay_class_init (GstRtpG722PayClass * klass)
{
GstBaseRTPPayloadClass *gstbasertppayload_class;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
gstbasertppayload_class->set_caps = gst_rtp_g722_pay_setcaps;
gstbasertppayload_class->get_caps = gst_rtp_g722_pay_getcaps;
GST_DEBUG_CATEGORY_INIT (rtpg722pay_debug, "rtpg722pay", 0,
"G722 RTP Payloader");
}
static void
gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay, GstRtpG722PayClass * klass)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg722pay);
/* tell basertpaudiopayload that this is a sample based codec */
gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
}
static gboolean
gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
{
GstRtpG722Pay *rtpg722pay;
GstStructure *structure;
gint rate, channels, clock_rate;
gboolean res;
gchar *params;
GstAudioChannelPosition *pos;
const GstRTPChannelOrder *order;
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
rtpg722pay = GST_RTP_G722_PAY (basepayload);
structure = gst_caps_get_structure (caps, 0);
/* first parse input caps */
if (!gst_structure_get_int (structure, "rate", &rate))
goto no_rate;
if (!gst_structure_get_int (structure, "channels", &channels))
goto no_channels;
/* get the channel order */
pos = gst_audio_get_channel_positions (structure);
if (pos)
order = gst_rtp_channels_get_by_pos (channels, pos);
else
order = NULL;
/* Clock rate is always 8000 Hz for G722 according to
* RFC 3551 although the sampling rate is 16000 Hz */
clock_rate = 8000;
gst_basertppayload_set_options (basepayload, "audio", TRUE, "G722",
clock_rate);
params = g_strdup_printf ("%d", channels);
if (!order && channels > 2) {
GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE,
(NULL), ("Unknown channel order for %d channels", channels));
}
if (order && order->name) {
res = gst_basertppayload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, "channel-order", G_TYPE_STRING, order->name, NULL);
} else {
res = gst_basertppayload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, NULL);
}
g_free (params);
g_free (pos);
rtpg722pay->rate = rate;
rtpg722pay->channels = channels;
/* octet-per-sample is 1 * channels for G722 */
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
4 * rtpg722pay->channels);
return res;
/* ERRORS */
no_rate:
{
GST_DEBUG_OBJECT (rtpg722pay, "no rate given");
return FALSE;
}
no_channels:
{
GST_DEBUG_OBJECT (rtpg722pay, "no channels given");
return FALSE;
}
}
static GstCaps *
gst_rtp_g722_pay_getcaps (GstBaseRTPPayload * rtppayload, GstPad * pad)
{
GstCaps *otherpadcaps;
GstCaps *caps;
otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
if (otherpadcaps) {
if (!gst_caps_is_empty (otherpadcaps)) {
gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
gst_caps_set_simple (caps, "rate", G_TYPE_INT, 16000, NULL);
}
gst_caps_unref (otherpadcaps);
}
return caps;
}
gboolean
gst_rtp_g722_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg722pay",
GST_RANK_SECONDARY, GST_TYPE_RTP_G722_PAY);
}
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