gstreamer/gst/rtp/gstrtpgsmpay.c
Olivier Crête 7effe918d1 rtp*pay: Allocate using the base class for audio codecs
This is required to add RTP header extensions from the
meta automatically.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/674>
2020-07-17 16:53:40 -04:00

181 lines
5.4 KiB
C

/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2005> Zeeshan Ali <zeenix@gmail.com>
*
* 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 <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpgsmpay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpgsmpay_debug);
#define GST_CAT_DEFAULT (rtpgsmpay_debug)
static GstStaticPadTemplate gst_rtp_gsm_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1")
);
static GstStaticPadTemplate gst_rtp_gsm_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_GSM_STRING ", "
"clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"")
);
static gboolean gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * payload,
GstBuffer * buffer);
#define gst_rtp_gsm_pay_parent_class parent_class
G_DEFINE_TYPE (GstRTPGSMPay, gst_rtp_gsm_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
gst_rtp_gsm_pay_class_init (GstRTPGSMPayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
GST_DEBUG_CATEGORY_INIT (rtpgsmpay_debug, "rtpgsmpay", 0,
"GSM Audio RTP Payloader");
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_gsm_pay_sink_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_gsm_pay_src_template);
gst_element_class_set_static_metadata (gstelement_class, "RTP GSM payloader",
"Codec/Payloader/Network/RTP",
"Payload-encodes GSM audio into a RTP packet",
"Zeeshan Ali <zeenix@gmail.com>");
gstrtpbasepayload_class->set_caps = gst_rtp_gsm_pay_setcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_gsm_pay_handle_buffer;
}
static void
gst_rtp_gsm_pay_init (GstRTPGSMPay * rtpgsmpay)
{
GST_RTP_BASE_PAYLOAD (rtpgsmpay)->clock_rate = 8000;
GST_RTP_BASE_PAYLOAD_PT (rtpgsmpay) = GST_RTP_PAYLOAD_GSM;
}
static gboolean
gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
const char *stname;
GstStructure *structure;
gboolean res;
structure = gst_caps_get_structure (caps, 0);
stname = gst_structure_get_name (structure);
if (strcmp ("audio/x-gsm", stname))
goto invalid_type;
gst_rtp_base_payload_set_options (payload, "audio",
payload->pt != GST_RTP_PAYLOAD_GSM, "GSM", 8000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;
/* ERRORS */
invalid_type:
{
GST_WARNING_OBJECT (payload, "invalid media type received");
return FALSE;
}
}
static GstFlowReturn
gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRTPGSMPay *rtpgsmpay;
guint payload_len;
GstBuffer *outbuf;
GstClockTime timestamp, duration;
GstFlowReturn ret;
rtpgsmpay = GST_RTP_GSM_PAY (basepayload);
timestamp = GST_BUFFER_PTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
/* FIXME, only one GSM frame per RTP packet for now */
payload_len = gst_buffer_get_size (buffer);
/* FIXME, just error out for now */
if (payload_len > GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay))
goto too_big;
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
/* copy timestamp and duration */
GST_BUFFER_PTS (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = duration;
gst_rtp_copy_audio_meta (rtpgsmpay, outbuf, buffer);
/* append payload */
outbuf = gst_buffer_append (outbuf, buffer);
GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (outbuf));
ret = gst_rtp_base_payload_push (basepayload, outbuf);
return ret;
/* ERRORS */
too_big:
{
GST_ELEMENT_ERROR (rtpgsmpay, STREAM, ENCODE, (NULL),
("payload_len %u > mtu %u", payload_len,
GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay)));
return GST_FLOW_ERROR;
}
}
gboolean
gst_rtp_gsm_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpgsmpay",
GST_RANK_SECONDARY, GST_TYPE_RTP_GSM_PAY);
}