gstreamer/gst/rtp/gstrtpgsmpay.c
Hyunjun Ko 5a17572119 rtppayload: set standard payload type as default
Initialize the PT to the default value of the codec and check if
it is still the default before declaring the pt to be dynamic or
not when setting the caps.

Also use the PT constants from the rtp lib when possible

https://bugzilla.gnome.org/show_bug.cgi?id=747965
2015-08-06 01:38:43 -03:00

177 lines
5.3 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 "gstrtpgsmpay.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_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_gsm_pay_sink_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&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_buffer_new_allocate (payload_len, 0, 0);
/* copy timestamp and duration */
GST_BUFFER_PTS (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = duration;
/* 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);
}