gstreamer/ext/opus/gstrtpopuspay.c
Vincent Penquerc'h f40ccb3811 opusenc: add upstream negotiation for multistream ability
This will help elements that cannot deal with multistream,
such as the RTP payloader.

The caps now do not include a "streams" field anymore, but
a "multistream" boolean, since we have no real use for knowing
the exact amount of streams.

https://bugzilla.gnome.org/show_bug.cgi?id=665078
2011-12-10 10:39:11 +00:00

140 lines
4 KiB
C

/*
* Opus Payloader Gst Element
*
* @author: Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpopuspay.h"
GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug);
#define GST_CAT_DEFAULT (rtpopuspay_debug)
static GstStaticPadTemplate gst_rtp_opus_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-opus, multistream = (boolean) FALSE")
);
static GstStaticPadTemplate gst_rtp_opus_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 48000, "
"encoding-name = (string) \"X-GST-OPUS-DRAFT-SPITTKA-00\"")
);
static gboolean gst_rtp_opus_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
GST_BOILERPLATE (GstRtpOPUSPay, gst_rtp_opus_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtp_opus_pay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_opus_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_opus_pay_sink_template));
gst_element_class_set_details_simple (element_class,
"RTP Opus payloader",
"Codec/Payloader/Network/RTP",
"Puts Opus audio in RTP packets",
"Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
}
static void
gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
{
GstBaseRTPPayloadClass *gstbasertppayload_class;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0,
"Opus RTP Payloader");
}
static void
gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay, GstRtpOPUSPayClass * klass)
{
}
static gboolean
gst_rtp_opus_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
gboolean res;
gchar *capsstr;
capsstr = gst_caps_to_string (caps);
gst_basertppayload_set_options (payload, "audio", FALSE,
"X-GST-OPUS-DRAFT-SPITTKA-00", 48000);
res =
gst_basertppayload_set_outcaps (payload, "caps", G_TYPE_STRING, capsstr,
NULL);
g_free (capsstr);
return res;
}
static GstFlowReturn
gst_rtp_opus_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstBuffer *outbuf;
GstClockTime timestamp;
guint size;
guint8 *data;
guint8 *payload;
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
outbuf = gst_rtp_buffer_new_allocate (size, 0, 0);
payload = gst_rtp_buffer_get_payload (outbuf);
memcpy (payload, data, size);
gst_rtp_buffer_set_marker (outbuf, FALSE);
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
return gst_basertppayload_push (basepayload, outbuf);
}