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Original commit message from CVS: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: Fix element descriptions.
165 lines
5 KiB
C
165 lines
5 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
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* Copyright (C) <2005> Zeeshan Ali <zeenix@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtppcmudepay.h"
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/* elementfactory information */
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static const GstElementDetails gst_rtp_pcmudepay_details =
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GST_ELEMENT_DETAILS ("RTP packet depayloader",
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"Codec/Depayloader/Network",
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"Extracts PCMU audio from RTP packets",
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"Edgard Lima <edgard.lima@indt.org.br>, Zeeshan Ali <zeenix@gmail.com>");
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/* RtpPcmuDepay signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0
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};
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static GstStaticPadTemplate gst_rtp_pcmu_depay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMU\"")
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);
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static GstStaticPadTemplate gst_rtp_pcmu_depay_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-mulaw, " "channels = (int) 1")
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);
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static GstBuffer *gst_rtp_pcmu_depay_process (GstBaseRTPDepayload * depayload,
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GstBuffer * buf);
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static gboolean gst_rtp_pcmu_depay_setcaps (GstBaseRTPDepayload * depayload,
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GstCaps * caps);
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GST_BOILERPLATE (GstRtpPcmuDepay, gst_rtp_pcmu_depay, GstBaseRTPDepayload,
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GST_TYPE_BASE_RTP_DEPAYLOAD);
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static void
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gst_rtp_pcmu_depay_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_pcmu_depay_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_pcmu_depay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_pcmudepay_details);
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}
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static void
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gst_rtp_pcmu_depay_class_init (GstRtpPcmuDepayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gstbasertpdepayload_class->process = gst_rtp_pcmu_depay_process;
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gstbasertpdepayload_class->set_caps = gst_rtp_pcmu_depay_setcaps;
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}
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static void
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gst_rtp_pcmu_depay_init (GstRtpPcmuDepay * rtppcmudepay,
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GstRtpPcmuDepayClass * klass)
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{
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GstBaseRTPDepayload *depayload;
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depayload = GST_BASE_RTP_DEPAYLOAD (rtppcmudepay);
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depayload->clock_rate = 8000;
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gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
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}
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static gboolean
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gst_rtp_pcmu_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
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{
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GstCaps *srccaps;
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gboolean ret;
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srccaps = gst_caps_new_simple ("audio/x-mulaw",
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"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);
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ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
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gst_caps_unref (srccaps);
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return ret;
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}
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static GstBuffer *
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gst_rtp_pcmu_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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{
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GstCaps *srccaps;
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GstBuffer *outbuf = NULL;
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gint payload_len;
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guint8 *payload;
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GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
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GST_BUFFER_SIZE (buf),
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gst_rtp_buffer_get_marker (buf),
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gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
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srccaps = GST_PAD_CAPS (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
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if (!srccaps) {
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/* Set the default caps */
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srccaps = gst_caps_new_simple ("audio/x-mulaw",
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"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);
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gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
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gst_caps_unref (srccaps);
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}
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payload_len = gst_rtp_buffer_get_payload_len (buf);
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payload = gst_rtp_buffer_get_payload (buf);
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outbuf = gst_buffer_new_and_alloc (payload_len);
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memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len);
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return outbuf;
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}
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gboolean
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gst_rtp_pcmu_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtppcmudepay",
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GST_RANK_MARGINAL, GST_TYPE_RTP_PCMU_DEPAY);
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}
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