gstreamer/gst/rtp/gstrtppcmudepay.c
Wim Taymans 34f916abbd gst/rtp/gstrtph263pdepay.c: Add some more debug info and guard against small payloads.
Original commit message from CVS:
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process):
Add some more debug info and guard against small payloads.
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process):
Set duration on outgoing buffers because we can.
2008-05-02 12:44:18 +00:00

174 lines
5.4 KiB
C

/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtppcmudepay.h"
/* elementfactory information */
static const GstElementDetails gst_rtp_pcmudepay_details =
GST_ELEMENT_DETAILS ("RTP packet depayloader",
"Codec/Depayloader/Network",
"Extracts PCMU audio from RTP packets",
"Edgard Lima <edgard.lima@indt.org.br>, Zeeshan Ali <zeenix@gmail.com>");
/* RtpPcmuDepay signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_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_DYNAMIC_STRING ", "
"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMU\";"
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", "
"clock-rate = (int) 8000")
);
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) 8000")
);
static GstBuffer *gst_rtp_pcmu_depay_process (GstBaseRTPDepayload * depayload,
GstBuffer * buf);
static gboolean gst_rtp_pcmu_depay_setcaps (GstBaseRTPDepayload * depayload,
GstCaps * caps);
GST_BOILERPLATE (GstRtpPcmuDepay, gst_rtp_pcmu_depay, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
static void
gst_rtp_pcmu_depay_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_pcmu_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_pcmu_depay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_pcmudepay_details);
}
static void
gst_rtp_pcmu_depay_class_init (GstRtpPcmuDepayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gstbasertpdepayload_class->process = gst_rtp_pcmu_depay_process;
gstbasertpdepayload_class->set_caps = gst_rtp_pcmu_depay_setcaps;
}
static void
gst_rtp_pcmu_depay_init (GstRtpPcmuDepay * rtppcmudepay,
GstRtpPcmuDepayClass * klass)
{
GstBaseRTPDepayload *depayload;
depayload = GST_BASE_RTP_DEPAYLOAD (rtppcmudepay);
gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
}
static gboolean
gst_rtp_pcmu_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstCaps *srccaps;
GstStructure *structure;
gboolean ret;
gint clock_rate = 8000; /* default */
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "clock-rate", &clock_rate);
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_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
return ret;
}
static GstBuffer *
gst_rtp_pcmu_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstCaps *srccaps;
GstBuffer *outbuf = NULL;
guint len;
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
GST_BUFFER_SIZE (buf),
gst_rtp_buffer_get_marker (buf),
gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
srccaps = GST_PAD_CAPS (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
if (!srccaps) {
/* Set the default caps */
srccaps = gst_caps_new_simple ("audio/x-mulaw",
"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);
gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
}
len = gst_rtp_buffer_get_payload_len (buf);
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (len, GST_SECOND, depayload->clock_rate);
return outbuf;
}
gboolean
gst_rtp_pcmu_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtppcmudepay",
GST_RANK_MARGINAL, GST_TYPE_RTP_PCMU_DEPAY);
}