gstreamer/gst/rtp/gstrtpmpapay.c
Ville Syrjala 69ce71927e gst/rtp/: Add static payload numbers in addition to the dynamic ones.
Original commit message from CVS:
Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
* gst/rtp/gstrtpgsmpay.c:
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtpmpapay.c:
* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_flush),
(gst_rtp_pcma_pay_handle_buffer):
* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_flush):
Add static payload numbers in addition to the dynamic ones.
Fixes #361639.
2006-10-18 17:06:21 +00:00

276 lines
8.1 KiB
C

/* GStreamer
* Copyright (C) <2005> Wim Taymans <wim@fluendo.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., 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 "gstrtpmpapay.h"
/* elementfactory information */
static const GstElementDetails gst_rtp_mpapay_details =
GST_ELEMENT_DETAILS ("RTP packet parser",
"Codec/Payloader/Network",
"Payload MPEG audio as RTP packets (RFC 2038)",
"Wim Taymans <wim@fluendo.com>");
static GstStaticPadTemplate gst_rtp_mpa_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg")
);
static GstStaticPadTemplate gst_rtp_mpa_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_MPA_STRING ", "
"clock-rate = (int) 90000, " "encoding-name = (string) \"MPA\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 90000, " "encoding-name = (string) \"MPA\"")
);
static void gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * klass);
static void gst_rtp_mpa_pay_base_init (GstRtpMPAPayClass * klass);
static void gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay);
static void gst_rtp_mpa_pay_finalize (GObject * object);
static gboolean gst_rtp_mpa_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_mpa_pay_handle_buffer (GstBaseRTPPayload * payload,
GstBuffer * buffer);
static GstBaseRTPPayloadClass *parent_class = NULL;
static GType
gst_rtp_mpa_pay_get_type (void)
{
static GType rtpmpapay_type = 0;
if (!rtpmpapay_type) {
static const GTypeInfo rtpmpapay_info = {
sizeof (GstRtpMPAPayClass),
(GBaseInitFunc) gst_rtp_mpa_pay_base_init,
NULL,
(GClassInitFunc) gst_rtp_mpa_pay_class_init,
NULL,
NULL,
sizeof (GstRtpMPAPay),
0,
(GInstanceInitFunc) gst_rtp_mpa_pay_init,
};
rtpmpapay_type =
g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMPAPay",
&rtpmpapay_info, 0);
}
return rtpmpapay_type;
}
static void
gst_rtp_mpa_pay_base_init (GstRtpMPAPayClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mpa_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mpa_pay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_mpapay_details);
}
static void
gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_rtp_mpa_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_mpa_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_mpa_pay_handle_buffer;
}
static void
gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay)
{
rtpmpapay->adapter = gst_adapter_new ();
}
static void
gst_rtp_mpa_pay_finalize (GObject * object)
{
GstRtpMPAPay *rtpmpapay;
rtpmpapay = GST_RTP_MPA_PAY (object);
g_object_unref (rtpmpapay->adapter);
rtpmpapay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_mpa_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
gst_basertppayload_set_options (payload, "audio", TRUE, "MPA", 90000);
gst_basertppayload_set_outcaps (payload, NULL);
return TRUE;
}
static GstFlowReturn
gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
guint16 frag_offset;
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. In the case the
* adapter has more than one MTU, we need to split the MPA data
* over multiple packets. The frag_offset in each packet header
* needs to be updated with the position in the MPA frame. */
avail = gst_adapter_available (rtpmpapay->adapter);
ret = GST_FLOW_OK;
frag_offset = 0;
while (avail > 0) {
guint towrite;
guint8 *payload;
guint8 *data;
guint payload_len;
guint packet_len;
/* this will be the total lenght of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0);
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpmpapay));
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
payload_len -= 4;
gst_rtp_buffer_set_payload_type (outbuf, GST_RTP_PAYLOAD_MPA);
/*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | MBZ | Frag_offset |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
payload = gst_rtp_buffer_get_payload (outbuf);
payload[0] = 0;
payload[1] = 0;
payload[2] = frag_offset >> 8;
payload[3] = frag_offset & 0xff;
data = (guint8 *) gst_adapter_peek (rtpmpapay->adapter, payload_len);
memcpy (&payload[4], data, payload_len);
gst_adapter_flush (rtpmpapay->adapter, payload_len);
avail -= payload_len;
frag_offset += payload_len;
if (avail == 0)
gst_rtp_buffer_set_marker (outbuf, TRUE);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmpapay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmpapay->duration;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpapay), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtp_mpa_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpMPAPay *rtpmpapay;
GstFlowReturn ret;
guint size, avail;
guint packet_len;
GstClockTime duration;
rtpmpapay = GST_RTP_MPA_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
duration = GST_BUFFER_DURATION (buffer);
avail = gst_adapter_available (rtpmpapay->adapter);
if (avail == 0) {
rtpmpapay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtpmpapay->duration = 0;
}
/* get packet length of previous data and this new data,
* payload length includes a 4 byte header */
packet_len = gst_rtp_buffer_calc_packet_len (4 + avail + size, 0, 0);
/* if this buffer is going to overflow the packet, flush what we
* have. */
if (gst_basertppayload_is_filled (basepayload,
packet_len, rtpmpapay->duration + duration)) {
ret = gst_rtp_mpa_pay_flush (rtpmpapay);
rtpmpapay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtpmpapay->duration = 0;
} else {
ret = GST_FLOW_OK;
}
gst_adapter_push (rtpmpapay->adapter, buffer);
rtpmpapay->duration += duration;
return ret;
}
gboolean
gst_rtp_mpa_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmpapay",
GST_RANK_NONE, GST_TYPE_RTP_MPA_PAY);
}