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Original commit message from CVS: fix descriptions and license blocks cut and paste anyone ?
161 lines
5 KiB
C
161 lines
5 KiB
C
/* GStreamer
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* Copyright (C) <2006> Philippe Khalaf <burger@speedy.org>
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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 "gstrtpilbcpay.h"
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#include <gst/rtp/gstrtpbuffer.h>
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/* elementfactory information */
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static GstElementDetails gst_rtpilbcpay_details = {
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"RTP Payloader for iLBC Audio",
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"Codec/Payloader/Network",
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"Packetize iLBC audio streams into RTP packets",
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"Philippe Kalaf <philippe.kalaf@collabora.co.uk>"
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};
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GST_DEBUG_CATEGORY (rtpilbcpay_debug);
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#define GST_CAT_DEFAULT (rtpilbcpay_debug)
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static GstStaticPadTemplate gst_rtpilbcpay_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-iLBC, " "mode = (int) {20, 30}")
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);
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static GstStaticPadTemplate gst_rtpilbcpay_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, "
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"encoding-name = (string) \"iLBC\", " "mode = (int) {20, 30}")
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);
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static gboolean gst_rtpilbcpay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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GST_BOILERPLATE (GstRTPILBCPay, gst_rtpilbcpay, GstBaseRTPAudioPayload,
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
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static void
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gst_rtpilbcpay_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_rtpilbcpay_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtpilbcpay_src_template));
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gst_element_class_set_details (element_class, &gst_rtpilbcpay_details);
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}
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static void
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gst_rtpilbcpay_class_init (GstRTPILBCPayClass * 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_rtpilbcpay_setcaps;
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GST_DEBUG_CATEGORY_INIT (rtpilbcpay_debug, "rtpilbcpay", 0,
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"iLBC audio RTP payloader");
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}
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static void
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gst_rtpilbcpay_init (GstRTPILBCPay * rtpilbcpay, GstRTPILBCPayClass * 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 (rtpilbcpay);
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpilbcpay);
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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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rtpilbcpay->mode = -1;
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/* tell basertpaudiopayload that this is a frame based codec */
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gst_basertpaudiopayload_set_frame_based (basertpaudiopayload);
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}
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static gboolean
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gst_rtpilbcpay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
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{
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GstRTPILBCPay *rtpilbcpay;
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GstBaseRTPAudioPayload *basertpaudiopayload;
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gboolean ret;
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gint mode;
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GstStructure *structure;
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const char *payload_name;
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rtpilbcpay = GST_RTP_ILBC_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, "mode", &mode);
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if (mode != 20 && mode != 30) {
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return FALSE;
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}
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payload_name = gst_structure_get_name (structure);
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if (g_strcasecmp ("audio/x-iLBC", payload_name) == 0) {
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gst_basertppayload_set_options (basertppayload, "audio", TRUE, "iLBC",
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8000);
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/* set options for this frame based audio codec */
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gst_basertpaudiopayload_set_frame_options (basertpaudiopayload,
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mode, mode == 30 ? 50 : 38);
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} else {
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return FALSE;
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}
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ret =
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gst_basertppayload_set_outcaps (basertppayload, "mode", G_TYPE_INT, mode,
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NULL);
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if (mode != rtpilbcpay->mode && rtpilbcpay->mode != -1) {
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GST_ERROR_OBJECT (rtpilbcpay, "Mode has changed from %d to %d! \
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Mode cannot change while streaming", rtpilbcpay->mode, mode);
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return FALSE;
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}
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rtpilbcpay->mode = mode;
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return ret;
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}
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gboolean
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gst_rtp_ilbc_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpilbcpay",
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GST_RANK_NONE, GST_TYPE_RTP_ILBC_PAY);
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}
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