mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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32ba17cd0f
where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there.
122 lines
3.6 KiB
C
122 lines
3.6 KiB
C
/*
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* Opus Payloader Gst Element
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*
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* @author: Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
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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 <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpopuspay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug);
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#define GST_CAT_DEFAULT (rtpopuspay_debug)
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static GstStaticPadTemplate gst_rtp_opus_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-opus, multistream = (boolean) FALSE")
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);
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static GstStaticPadTemplate gst_rtp_opus_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) 48000, "
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"encoding-name = (string) \"X-GST-OPUS-DRAFT-SPITTKA-00\"")
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);
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static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload *
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payload, GstBuffer * buffer);
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G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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static void
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gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
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{
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GstRTPBasePayloadClass *gstbasertppayload_class;
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GstElementClass *element_class;
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gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;
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element_class = GST_ELEMENT_CLASS (klass);
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gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_opus_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_opus_pay_sink_template));
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gst_element_class_set_static_metadata (element_class,
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"RTP Opus payloader",
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"Codec/Payloader/Network/RTP",
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"Puts Opus audio in RTP packets",
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"Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
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GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0,
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"Opus RTP Payloader");
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}
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static void
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gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay)
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{
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}
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static gboolean
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gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
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{
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gboolean res;
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gchar *capsstr;
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capsstr = gst_caps_to_string (caps);
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gst_rtp_base_payload_set_options (payload, "audio", FALSE,
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"X-GST-OPUS-DRAFT-SPITTKA-00", 48000);
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res =
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gst_rtp_base_payload_set_outcaps (payload, "caps", G_TYPE_STRING, capsstr,
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NULL);
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g_free (capsstr);
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return res;
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}
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static GstFlowReturn
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gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
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GstBuffer * buffer)
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{
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GstBuffer *outbuf;
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outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
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outbuf = gst_buffer_append (outbuf, gst_buffer_ref (buffer));
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/* Push out */
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return gst_rtp_base_payload_push (basepayload, outbuf);
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}
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