gstreamer/gst/rtp/gstrtpL16pay.c
Wim Taymans 40d06b6a55 gst/rtp/gstrtpL16pay.*: Fill up to MTU using adapter.
Original commit message from CVS:
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps),
(gst_rtp_L16_pay_flush), (gst_rtp_L16_pay_handle_buffer):
* gst/rtp/gstrtpL16pay.h:
Fill up to MTU using adapter.
Timestamp rtp packets.
2007-01-25 10:54:19 +00:00

281 lines
7.9 KiB
C

/* GStreamer
* Copyright (C) <2007> 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 "gstrtpL16pay.h"
GST_DEBUG_CATEGORY_STATIC (rtpL16pay_debug);
#define GST_CAT_DEFAULT (rtpL16pay_debug)
/* elementfactory information */
static const GstElementDetails gst_rtp_L16_pay_details =
GST_ELEMENT_DETAILS ("RTP packet payloader",
"Codec/Payloader/Network",
"Payload-encode Raw audio into RTP packets (RFC 3551)",
"Wim Taymans <wim@fluendo.com>");
static GstStaticPadTemplate gst_rtp_L16_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) BIG_ENDIAN, "
"signed = (boolean) true, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
static GstStaticPadTemplate gst_rtp_L16_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) [ 96, 127 ], "
"clock-rate = (int) [ 1, MAX ], "
"encoding-name = (string) \"L16\", "
"channels = (int) [ 1, MAX ], "
"rate = (int) [ 1, MAX ];"
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) { " GST_RTP_PAYLOAD_L16_STEREO_STRING ", "
GST_RTP_PAYLOAD_L16_MONO_STRING " }," "clock-rate = (int) 44100")
);
static void gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass);
static void gst_rtp_L16_pay_base_init (GstRtpL16PayClass * klass);
static void gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay);
static void gst_rtp_L16_pay_finalize (GObject * object);
static gboolean gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload,
GstCaps * caps);
static GstFlowReturn gst_rtp_L16_pay_handle_buffer (GstBaseRTPPayload * pad,
GstBuffer * buffer);
static GstBaseRTPPayloadClass *parent_class = NULL;
static GType
gst_rtp_L16_pay_get_type (void)
{
static GType rtpL16pay_type = 0;
if (!rtpL16pay_type) {
static const GTypeInfo rtpL16pay_info = {
sizeof (GstRtpL16PayClass),
(GBaseInitFunc) gst_rtp_L16_pay_base_init,
NULL,
(GClassInitFunc) gst_rtp_L16_pay_class_init,
NULL,
NULL,
sizeof (GstRtpL16Pay),
0,
(GInstanceInitFunc) gst_rtp_L16_pay_init,
};
rtpL16pay_type =
g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpL16Pay",
&rtpL16pay_info, 0);
}
return rtpL16pay_type;
}
static void
gst_rtp_L16_pay_base_init (GstRtpL16PayClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_L16_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_L16_pay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_L16_pay_details);
}
static void
gst_rtp_L16_pay_class_init (GstRtpL16PayClass * 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_L16_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_L16_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_L16_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtpL16pay_debug, "rtpL16pay", 0,
"L16 RTP Payloader");
}
static void
gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay)
{
rtpL16pay->adapter = gst_adapter_new ();
}
static void
gst_rtp_L16_pay_finalize (GObject * object)
{
GstRtpL16Pay *rtpL16pay;
rtpL16pay = GST_RTP_L16_PAY (object);
g_object_unref (rtpL16pay->adapter);
rtpL16pay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
{
GstRtpL16Pay *rtpL16pay;
GstStructure *structure;
gint channels, rate;
rtpL16pay = GST_RTP_L16_PAY (basepayload);
structure = gst_caps_get_structure (caps, 0);
/* first parse input caps */
if (!gst_structure_get_int (structure, "rate", &rate))
goto no_rate;
if (!gst_structure_get_int (structure, "channels", &channels))
goto no_channels;
gst_basertppayload_set_options (basepayload, "audio", TRUE, "L16", rate);
gst_basertppayload_set_outcaps (basepayload,
"channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, rate, NULL);
rtpL16pay->rate = rate;
rtpL16pay->channels = channels;
return TRUE;
/* ERRORS */
no_rate:
{
GST_DEBUG_OBJECT (rtpL16pay, "no rate given");
return FALSE;
}
no_channels:
{
GST_DEBUG_OBJECT (rtpL16pay, "no channels given");
return FALSE;
}
}
static GstFlowReturn
gst_rtp_L16_pay_flush (GstRtpL16Pay * rtpL16pay, guint len)
{
GstBuffer *outbuf;
guint8 *payload;
GstFlowReturn ret;
guint samples;
GstClockTime duration;
/* now alloc output buffer */
outbuf = gst_rtp_buffer_new_allocate (len, 0, 0);
/* get payload, this is now writable */
payload = gst_rtp_buffer_get_payload (outbuf);
/* copy and flush data out of adapter into the RTP payload */
gst_adapter_copy (rtpL16pay->adapter, payload, 0, len);
gst_adapter_flush (rtpL16pay->adapter, len);
samples = len / (2 * rtpL16pay->channels);
duration = gst_util_uint64_scale_int (samples, GST_SECOND, rtpL16pay->rate);
GST_BUFFER_TIMESTAMP (outbuf) = rtpL16pay->first_ts;
GST_BUFFER_DURATION (outbuf) = duration;
/* increase count (in ts) of data pushed to basertppayload */
if (GST_CLOCK_TIME_IS_VALID (rtpL16pay->first_ts))
rtpL16pay->first_ts += duration;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpL16pay), outbuf);
return ret;
}
static GstFlowReturn
gst_rtp_L16_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpL16Pay *rtpL16pay;
GstFlowReturn ret = GST_FLOW_OK;
guint payload_len;
GstClockTime timestamp, duration;
guint mtu, avail;
rtpL16pay = GST_RTP_L16_PAY (basepayload);
mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpL16pay);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
if (GST_BUFFER_IS_DISCONT (buffer))
gst_adapter_clear (rtpL16pay->adapter);
avail = gst_adapter_available (rtpL16pay->adapter);
if (avail == 0) {
rtpL16pay->first_ts = timestamp;
}
/* push buffer in adapter */
gst_adapter_push (rtpL16pay->adapter, buffer);
/* get payload len for MTU */
payload_len = gst_rtp_buffer_calc_payload_len (mtu, 0, 0);
/* flush complete MTU while we have enough data in the adapter */
while (avail >= payload_len) {
/* flush payload_len bytes */
ret = gst_rtp_L16_pay_flush (rtpL16pay, payload_len);
if (ret != GST_FLOW_OK)
break;
avail = gst_adapter_available (rtpL16pay->adapter);
}
return ret;
}
gboolean
gst_rtp_L16_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpL16pay",
GST_RANK_NONE, GST_TYPE_RTP_L16_PAY);
}