mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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152 lines
4.8 KiB
C
152 lines
4.8 KiB
C
/*
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* Siren Payloader Gst Element
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*
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* @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
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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 "gstrtpsirenpay.h"
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#include <gst/rtp/gstrtpbuffer.h>
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GST_DEBUG_CATEGORY_STATIC (rtpsirenpay_debug);
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#define GST_CAT_DEFAULT (rtpsirenpay_debug)
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static GstStaticPadTemplate gst_rtp_siren_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-siren, " "dct-length = (int) 320")
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);
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static GstStaticPadTemplate gst_rtp_siren_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_DYNAMIC_STRING ", "
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"clock-rate = (int) 16000, "
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"encoding-name = (string) \"SIREN\", "
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"bitrate = (string) \"16000\", " "dct-length = (int) 320")
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);
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static gboolean gst_rtp_siren_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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GST_BOILERPLATE (GstRTPSirenPay, gst_rtp_siren_pay, GstBaseRTPAudioPayload,
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
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static void
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gst_rtp_siren_pay_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_siren_pay_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_siren_pay_src_template));
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gst_element_class_set_details_simple (element_class,
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"RTP Payloader for Siren Audio", "Codec/Payloader/Network/RTP",
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"Packetize Siren audio streams into RTP packets",
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"Youness Alaoui <kakaroto@kakaroto.homelinux.net>");
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}
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static void
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gst_rtp_siren_pay_class_init (GstRTPSirenPayClass * klass)
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{
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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gstbasertppayload_class->set_caps = gst_rtp_siren_pay_setcaps;
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GST_DEBUG_CATEGORY_INIT (rtpsirenpay_debug, "rtpsirenpay", 0,
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"siren audio RTP payloader");
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}
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static void
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gst_rtp_siren_pay_init (GstRTPSirenPay * rtpsirenpay,
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GstRTPSirenPayClass * klass)
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{
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GstBaseRTPPayload *basertppayload;
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GstBaseRTPAudioPayload *basertpaudiopayload;
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basertppayload = GST_BASE_RTP_PAYLOAD (rtpsirenpay);
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpsirenpay);
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/* we don't set the payload type, it should be set by the application using
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* the pt property or the default 96 will be used */
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basertppayload->clock_rate = 16000;
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/* tell basertpaudiopayload that this is a frame based codec */
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gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
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}
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static gboolean
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gst_rtp_siren_pay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
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{
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GstRTPSirenPay *rtpsirenpay;
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GstBaseRTPAudioPayload *basertpaudiopayload;
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gint dct_length;
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GstStructure *structure;
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const char *payload_name;
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rtpsirenpay = GST_RTP_SIREN_PAY (basertppayload);
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (structure, "dct-length", &dct_length);
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if (dct_length != 320)
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goto wrong_dct;
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payload_name = gst_structure_get_name (structure);
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if (g_ascii_strcasecmp ("audio/x-siren", payload_name))
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goto wrong_caps;
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gst_basertppayload_set_options (basertppayload, "audio", TRUE, "SIREN",
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16000);
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/* set options for this frame based audio codec */
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gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 20, 40);
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return gst_basertppayload_set_outcaps (basertppayload, NULL);
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/* ERRORS */
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wrong_dct:
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{
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GST_ERROR_OBJECT (rtpsirenpay, "dct-length must be 320, received %d",
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dct_length);
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return FALSE;
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}
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wrong_caps:
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{
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GST_ERROR_OBJECT (rtpsirenpay, "expected audio/x-siren, received %s",
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payload_name);
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return FALSE;
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}
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}
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gboolean
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gst_rtp_siren_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpsirenpay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_SIREN_PAY);
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}
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