gstreamer/gst/rtp/gstrtpmpapay.c
Wim Taymans 75a0669d5f gst/rtp/: Various class and caps fixes from Andre Magalhaes (andrunko)
Original commit message from CVS:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_setcaps):
* gst/rtp/gstrtpgsmparse.c:
* gst/rtp/gstrtph263penc.c:
* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
(gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer),
(gst_rtpmp4venc_set_property):
* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_handle_buffer):
Various class and caps fixes from Andre Magalhaes (andrunko)
2005-09-30 16:36:49 +00:00

268 lines
7.6 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
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpmpaenc.h"
/* elementfactory information */
static GstElementDetails gst_rtp_mpaenc_details = {
"RTP packet parser",
"Codec/Encoder/Network",
"Encode MPEG audio as RTP packets (RFC 2038)",
"Wim Taymans <wim@fluendo.com>"
};
static GstStaticPadTemplate gst_rtpmpaenc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg")
);
static GstStaticPadTemplate gst_rtpmpaenc_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) [ 96, 255 ], "
"clock-rate = (int) 90000, " "encoding-name = (string) \"MPA\"")
);
static void gst_rtpmpaenc_class_init (GstRtpMPAEncClass * klass);
static void gst_rtpmpaenc_base_init (GstRtpMPAEncClass * klass);
static void gst_rtpmpaenc_init (GstRtpMPAEnc * rtpmpaenc);
static void gst_rtpmpaenc_finalize (GObject * object);
static gboolean gst_rtpmpaenc_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtpmpaenc_handle_buffer (GstBaseRTPPayload * payload,
GstBuffer * buffer);
static GstBaseRTPPayloadClass *parent_class = NULL;
static GType
gst_rtpmpaenc_get_type (void)
{
static GType rtpmpaenc_type = 0;
if (!rtpmpaenc_type) {
static const GTypeInfo rtpmpaenc_info = {
sizeof (GstRtpMPAEncClass),
(GBaseInitFunc) gst_rtpmpaenc_base_init,
NULL,
(GClassInitFunc) gst_rtpmpaenc_class_init,
NULL,
NULL,
sizeof (GstRtpMPAEnc),
0,
(GInstanceInitFunc) gst_rtpmpaenc_init,
};
rtpmpaenc_type =
g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMPAEnc",
&rtpmpaenc_info, 0);
}
return rtpmpaenc_type;
}
static void
gst_rtpmpaenc_base_init (GstRtpMPAEncClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpmpaenc_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpmpaenc_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_mpaenc_details);
}
static void
gst_rtpmpaenc_class_init (GstRtpMPAEncClass * 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_ref (GST_TYPE_BASE_RTP_PAYLOAD);
gobject_class->finalize = gst_rtpmpaenc_finalize;
gstbasertppayload_class->set_caps = gst_rtpmpaenc_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtpmpaenc_handle_buffer;
}
static void
gst_rtpmpaenc_init (GstRtpMPAEnc * rtpmpaenc)
{
rtpmpaenc->adapter = gst_adapter_new ();
}
static void
gst_rtpmpaenc_finalize (GObject * object)
{
GstRtpMPAEnc *rtpmpaenc;
rtpmpaenc = GST_RTP_MPA_ENC (object);
g_object_unref (rtpmpaenc->adapter);
rtpmpaenc->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtpmpaenc_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_rtpmpaenc_flush (GstRtpMPAEnc * rtpmpaenc)
{
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 (rtpmpaenc->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_rtpbuffer_calc_packet_len (4 + avail, 0, 0);
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpmpaenc));
/* this is the payload length */
payload_len = gst_rtpbuffer_calc_payload_len (towrite, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
payload_len -= 4;
gst_rtpbuffer_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_rtpbuffer_get_payload (outbuf);
payload[0] = 0;
payload[1] = 0;
payload[2] = frag_offset >> 8;
payload[3] = frag_offset & 0xff;
data = (guint8 *) gst_adapter_peek (rtpmpaenc->adapter, payload_len);
memcpy (&payload[4], data, payload_len);
gst_adapter_flush (rtpmpaenc->adapter, payload_len);
avail -= payload_len;
frag_offset += payload_len;
if (avail == 0)
gst_rtpbuffer_set_marker (outbuf, TRUE);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmpaenc->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmpaenc->duration;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmpaenc), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtpmpaenc_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpMPAEnc *rtpmpaenc;
GstFlowReturn ret;
guint size, avail;
guint packet_len;
GstClockTime duration;
rtpmpaenc = GST_RTP_MPA_ENC (basepayload);
size = GST_BUFFER_SIZE (buffer);
duration = GST_BUFFER_DURATION (buffer);
avail = gst_adapter_available (rtpmpaenc->adapter);
if (avail == 0) {
rtpmpaenc->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtpmpaenc->duration = 0;
}
/* get packet length of previous data and this new data,
* payload length includes a 4 byte header */
packet_len = gst_rtpbuffer_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, rtpmpaenc->duration + duration)) {
ret = gst_rtpmpaenc_flush (rtpmpaenc);
rtpmpaenc->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtpmpaenc->duration = 0;
} else {
ret = GST_FLOW_OK;
}
gst_adapter_push (rtpmpaenc->adapter, buffer);
rtpmpaenc->duration += duration;
return ret;
}
gboolean
gst_rtpmpaenc_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmpaenc",
GST_RANK_NONE, GST_TYPE_RTP_MPA_ENC);
}