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* Copyright (C) <2010> Wim Taymans <wim.taymans@gmail.com>
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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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* 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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* 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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#include <gst/audio/audio.h>
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#include <gst/audio/multichannel.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpg722pay.h"
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#include "gstrtpchannels.h"
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GST_DEBUG_CATEGORY_STATIC (rtpg722pay_debug);
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#define GST_CAT_DEFAULT (rtpg722pay_debug)
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static GstStaticPadTemplate gst_rtp_g722_pay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_STATIC_CAPS ("audio/G722, " "rate = (int) 16000, " "channels = (int) 1")
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static GstStaticPadTemplate gst_rtp_g722_pay_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"encoding-name = (string) \"G722\", "
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"payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", "
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"clock-rate = (int) 8000")
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static gboolean gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload,
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static GstCaps *gst_rtp_g722_pay_getcaps (GstBaseRTPPayload * rtppayload,
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GST_BOILERPLATE (GstRtpG722Pay, gst_rtp_g722_pay, GstBaseRTPAudioPayload,
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
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gst_rtp_g722_pay_base_init (gpointer klass)
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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_g722_pay_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_g722_pay_sink_template));
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gst_element_class_set_details_simple (element_class, "RTP audio payloader",
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"Codec/Payloader/Network/RTP",
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"Payload-encode Raw audio into RTP packets (RFC 3551)",
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"Wim Taymans <wim.taymans@gmail.com>");
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gst_rtp_g722_pay_class_init (GstRtpG722PayClass * klass)
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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_g722_pay_setcaps;
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gstbasertppayload_class->get_caps = gst_rtp_g722_pay_getcaps;
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GST_DEBUG_CATEGORY_INIT (rtpg722pay_debug, "rtpg722pay", 0,
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"G722 RTP Payloader");
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gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay, GstRtpG722PayClass * klass)
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GstBaseRTPAudioPayload *basertpaudiopayload;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg722pay);
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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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gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
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GstRtpG722Pay *rtpg722pay;
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GstStructure *structure;
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gint rate, channels, clock_rate;
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GstAudioChannelPosition *pos;
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const GstRTPChannelOrder *order;
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GstBaseRTPAudioPayload *basertpaudiopayload;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
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rtpg722pay = GST_RTP_G722_PAY (basepayload);
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structure = gst_caps_get_structure (caps, 0);
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/* first parse input caps */
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if (!gst_structure_get_int (structure, "rate", &rate))
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if (!gst_structure_get_int (structure, "channels", &channels))
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/* get the channel order */
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pos = gst_audio_get_channel_positions (structure);
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order = gst_rtp_channels_get_by_pos (channels, pos);
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/* Clock rate is always 8000 Hz for G722 according to
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* RFC 3551 although the sampling rate is 16000 Hz */
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gst_basertppayload_set_options (basepayload, "audio", TRUE, "G722",
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params = g_strdup_printf ("%d", channels);
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if (!order && channels > 2) {
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GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE,
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(NULL), ("Unknown channel order for %d channels", channels));
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if (order && order->name) {
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res = gst_basertppayload_set_outcaps (basepayload,
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"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
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channels, "channel-order", G_TYPE_STRING, order->name, NULL);
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res = gst_basertppayload_set_outcaps (basepayload,
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"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
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rtpg722pay->rate = rate;
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rtpg722pay->channels = channels;
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/* octet-per-sample is 1 * channels for G722 */
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gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
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4 * rtpg722pay->channels);
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GST_DEBUG_OBJECT (rtpg722pay, "no rate given");
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GST_DEBUG_OBJECT (rtpg722pay, "no channels given");
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gst_rtp_g722_pay_getcaps (GstBaseRTPPayload * rtppayload, GstPad * pad)
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GstCaps *otherpadcaps;
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otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
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caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
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if (!gst_caps_is_empty (otherpadcaps)) {
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gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
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gst_caps_set_simple (caps, "rate", G_TYPE_INT, 16000, NULL);
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gst_caps_unref (otherpadcaps);
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gst_rtp_g722_pay_plugin_init (GstPlugin * plugin)
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return gst_element_register (plugin, "rtpg722pay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_G722_PAY);