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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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116 lines
3.8 KiB
C
116 lines
3.8 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> Nokia Corporation <kai.vehmanen@nokia.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 <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtppcmupay.h"
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static GstStaticPadTemplate gst_rtp_pcmu_pay_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 ("audio/x-mulaw, channels=(int)1, rate=(int)8000")
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);
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static GstStaticPadTemplate gst_rtp_pcmu_pay_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 ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", "
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"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMU\"; "
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMU\"")
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);
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static gboolean gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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#define gst_rtp_pcmu_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpPcmuPay, gst_rtp_pcmu_pay,
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
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static void
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gst_rtp_pcmu_pay_class_init (GstRtpPcmuPayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gstelement_class = (GstElementClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_pcmu_pay_sink_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_pcmu_pay_src_template));
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gst_element_class_set_details_simple (gstelement_class, "RTP PCMU payloader",
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"Codec/Payloader/Network/RTP",
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"Payload-encodes PCMU audio into a RTP packet",
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"Edgard Lima <edgard.lima@indt.org.br>");
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gstbasertppayload_class->set_caps = gst_rtp_pcmu_pay_setcaps;
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}
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static void
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gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay)
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{
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GstBaseRTPAudioPayload *basertpaudiopayload;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtppcmupay);
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GST_BASE_RTP_PAYLOAD (rtppcmupay)->clock_rate = 8000;
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/* tell basertpaudiopayload that this is a sample based codec */
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gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
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/* octet-per-sample is 1 for PCM */
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gst_base_rtp_audio_payload_set_sample_options (basertpaudiopayload, 1);
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}
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static gboolean
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gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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{
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gboolean res;
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payload->pt = GST_RTP_PAYLOAD_PCMU;
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gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
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res = gst_basertppayload_set_outcaps (payload, NULL);
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return res;
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}
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gboolean
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gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtppcmupay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_PCMU_PAY);
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}
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