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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2005> Edgard Lima <edgard.lima@gmail.com>
* Copyright (C) <2005> Zeeshan Ali <zeenix@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.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpelements.h"
#include "gstrtppcmudepay.h"
#include "gstrtputils.h"
/* RtpPcmuDepay signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0
};
static GstStaticPadTemplate gst_rtp_pcmu_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", "
"clock-rate = (int) 8000; "
"application/x-rtp, "
"media = (string) \"audio\", "
"encoding-name = (string) \"PCMU\", clock-rate = (int) [1, MAX ]")
);
static GstStaticPadTemplate gst_rtp_pcmu_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-mulaw, "
"channels = (int) 1, rate = (int) [1, MAX ]")
);
static GstBuffer *gst_rtp_pcmu_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp);
static gboolean gst_rtp_pcmu_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
#define gst_rtp_pcmu_depay_parent_class parent_class
G_DEFINE_TYPE (GstRtpPcmuDepay, gst_rtp_pcmu_depay,
GST_TYPE_RTP_BASE_DEPAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtppcmudepay, "rtppcmudepay",
GST_RANK_SECONDARY, GST_TYPE_RTP_PCMU_DEPAY, rtp_element_init (plugin));
static void
gst_rtp_pcmu_depay_class_init (GstRtpPcmuDepayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gstelement_class = (GstElementClass *) klass;
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_pcmu_depay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_pcmu_depay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP PCMU depayloader", "Codec/Depayloader/Network/RTP",
"Extracts PCMU audio from RTP packets",
"Edgard Lima <edgard.lima@gmail.com>, Zeeshan Ali <zeenix@gmail.com>");
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_pcmu_depay_process;
gstrtpbasedepayload_class->set_caps = gst_rtp_pcmu_depay_setcaps;
}
static void
gst_rtp_pcmu_depay_init (GstRtpPcmuDepay * rtppcmudepay)
{
GstRTPBaseDepayload *depayload;
depayload = GST_RTP_BASE_DEPAYLOAD (rtppcmudepay);
gst_pad_use_fixed_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload));
}
static gboolean
gst_rtp_pcmu_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstCaps *srccaps;
GstStructure *structure;
gboolean ret;
gint clock_rate;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
clock_rate = 8000; /* default */
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("audio/x-mulaw",
"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, clock_rate, NULL);
ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
return ret;
}
static GstBuffer *
gst_rtp_pcmu_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstBuffer *outbuf = NULL;
guint len;
gboolean marker;
marker = gst_rtp_buffer_get_marker (rtp);
GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
gst_buffer_get_size (rtp->buffer), marker,
gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
len = gst_rtp_buffer_get_payload_len (rtp);
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
if (outbuf) {
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (len, GST_SECOND, depayload->clock_rate);
if (marker) {
/* mark start of talkspurt with RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
gst_rtp_drop_non_audio_meta (depayload, outbuf);
}
return outbuf;
}
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