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d6f37dadb5
Original commit message from CVS: Based on patch by: Olivier Crete <tester at tester dot ca> * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_base_init), (gst_rtp_g729_depay_class_init), (gst_rtp_g729_depay_init), (gst_rtp_g729_depay_setcaps), (gst_rtp_g729_depay_process), (gst_rtp_g729_depay_plugin_init): * gst/rtp/gstrtpg729depay.h: * gst/rtp/gstrtpg729pay.c: (gst_rtpg729pay_base_init), (gst_rtpg729pay_class_init), (gst_rtpg729pay_init), (gst_rtpg729pay_setcaps), (gst_rtp_g729_pay_plugin_init): * gst/rtp/gstrtpg729pay.h: Added G729 pay and depayloaders. Fixes #532409.
150 lines
4.7 KiB
C
150 lines
4.7 KiB
C
/* GStreamer
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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 "gstrtpg729pay.h"
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#include <gst/rtp/gstrtpbuffer.h>
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/* elementfactory information */
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static GstElementDetails gst_rtpg729pay_details = {
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"RTP Payloader for G729 Audio",
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"Codec/Payloader/Network",
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"Packetize G729 audio streams into RTP packets",
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"Laurent Glayal <spglegle@yahoo.fr>"
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};
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GST_DEBUG_CATEGORY_STATIC (rtpg729pay_debug);
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#define GST_CAT_DEFAULT (rtpg729pay_debug)
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static GstStaticPadTemplate gst_rtpg729pay_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/G729, channels=(int)1, rate=(int)8000")
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);
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static GstStaticPadTemplate gst_rtpg729pay_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) 8000, " "encoding-name = (string) \"G729\";"
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_G729_STRING ", "
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"clock-rate = (int) 8000")
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);
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static gboolean gst_rtpg729pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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GST_BOILERPLATE (GstRtpG729Pay, gst_rtpg729pay, GstBaseRTPAudioPayload,
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
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static void
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gst_rtpg729pay_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_rtpg729pay_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtpg729pay_src_template));
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gst_element_class_set_details (element_class, &gst_rtpg729pay_details);
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}
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static void
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gst_rtpg729pay_class_init (GstRtpG729PayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
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gstbasertppayload_class->set_caps = gst_rtpg729pay_setcaps;
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GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0,
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"G729 audio RTP payloader");
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}
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static void
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gst_rtpg729pay_init (GstRtpG729Pay * rtpg729pay, GstRtpG729PayClass * 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 (rtpg729pay);
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg729pay);
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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 = 8000;
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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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gst_basertppayload_set_options (basertppayload, "audio", FALSE, "G729", 8000);
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gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 10, 10);
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}
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static gboolean
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gst_rtpg729pay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
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{
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GstRtpG729Pay *rtpg729pay;
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GstBaseRTPAudioPayload *basertpaudiopayload;
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gboolean ret;
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GstStructure *structure;
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const char *payload_name;
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rtpg729pay = GST_RTP_G729_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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payload_name = gst_structure_get_name (structure);
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if (g_strcasecmp ("audio/G729", payload_name) != 0)
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goto wrong_name;
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ret = gst_basertppayload_set_outcaps (basertppayload, NULL);
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return ret;
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/* ERRORS */
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wrong_name:
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{
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GST_ERROR_OBJECT (rtpg729pay, "wrong name, expected 'audio/G729', got '%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_g729_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpg729pay",
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GST_RANK_NONE, GST_TYPE_RTP_G729_PAY);
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}
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