mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-25 03:01:03 +00:00
de648b8832
Copy timestamps to payloaded buffer. Avoid input buffer memory leak. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692929
125 lines
3.8 KiB
C
125 lines
3.8 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., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, 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 (GstRTPBasePayload * payload,
|
|
GstCaps * caps);
|
|
static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload *
|
|
payload, GstBuffer * buffer);
|
|
|
|
G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD);
|
|
|
|
static void
|
|
gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
|
|
{
|
|
GstRTPBasePayloadClass *gstbasertppayload_class;
|
|
GstElementClass *element_class;
|
|
|
|
gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;
|
|
element_class = GST_ELEMENT_CLASS (klass);
|
|
|
|
gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
|
|
gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
|
|
|
|
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_static_metadata (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>");
|
|
|
|
GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0,
|
|
"Opus RTP Payloader");
|
|
}
|
|
|
|
static void
|
|
gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay)
|
|
{
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
|
|
{
|
|
gboolean res;
|
|
|
|
gst_rtp_base_payload_set_options (payload, "audio", FALSE,
|
|
"X-GST-OPUS-DRAFT-SPITTKA-00", 48000);
|
|
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstBuffer *outbuf;
|
|
GstClockTime pts, dts, duration;
|
|
|
|
pts = GST_BUFFER_PTS (buffer);
|
|
dts = GST_BUFFER_DTS (buffer);
|
|
duration = GST_BUFFER_DURATION (buffer);
|
|
|
|
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
|
|
outbuf = gst_buffer_append (outbuf, buffer);
|
|
|
|
GST_BUFFER_PTS (outbuf) = pts;
|
|
GST_BUFFER_DTS (outbuf) = dts;
|
|
GST_BUFFER_DURATION (outbuf) = duration;
|
|
|
|
/* Push out */
|
|
return gst_rtp_base_payload_push (basepayload, outbuf);
|
|
}
|