mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-09 10:59:39 +00:00
da3e23d375
Original commit message from CVS: * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_process): * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_process): * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_process): * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process): * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process): * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_flush): * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process): * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush): * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_process): * gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_flush): * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process): * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_process): * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process): * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process): Use more efficient adapter and rtpbuffer methods when possible.
274 lines
8 KiB
C
274 lines
8 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 payloader",
|
|
"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; "
|
|
"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;
|
|
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;
|
|
|
|
gst_adapter_copy (rtpmpapay->adapter, &payload[4], 0, 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);
|
|
}
|