gstreamer/gst/rtp/gstrtppcmupay.c
Kai Vehmanen ca00f98eb5 gst/rtp/: Fix timestamp calculation on outgoing RTP packets.
Original commit message from CVS:
Patch by: Kai Vehmanen <kv2004 at eca dot cx>
* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_flush),
(gst_rtp_pcma_pay_handle_buffer):
* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_flush),
(gst_rtp_pcmu_pay_handle_buffer):
Fix timestamp calculation on outgoing RTP packets.
Fixes #348675.
2006-07-26 16:36:59 +00:00

244 lines
7.6 KiB
C

/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
* Copyright (C) <2005> Nokia Corporation <kai.vehmanen@nokia.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 <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtppcmupay.h"
/* elementfactory information */
static const GstElementDetails gst_rtp_pcmu_pay_details =
GST_ELEMENT_DETAILS ("RTP packet parser",
"Codec/Payloader/Network",
"Payload-encodes PCMU audio into a RTP packet",
"Edgard Lima <edgard.lima@indt.org.br>");
static GstStaticPadTemplate gst_rtp_pcmu_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-mulaw, channels=(int)1, rate=(int)8000")
);
static GstStaticPadTemplate gst_rtp_pcmu_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_PCMU_STRING ", "
"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMU\"")
);
static gboolean gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_pcmu_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
static void gst_rtp_pcmu_pay_finalize (GObject * object);
GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
/* The lower limit for number of octet to put in one packet
* (clock-rate=8000, octet-per-sample=1). The default 80 is equal
* to to 10msec (see RFC3551) */
#define GST_RTP_PCMU_MIN_PTIME_OCTETS 80
static void
gst_rtp_pcmu_pay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_pcmu_pay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_pcmu_pay_src_template));
gst_element_class_set_details (element_class, &gst_rtp_pcmu_pay_details);
}
static void
gst_rtp_pcmu_pay_class_init (GstRtpPcmuPayClass * 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_pcmu_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_pcmu_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_pcmu_pay_handle_buffer;
}
static void
gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay, GstRtpPcmuPayClass * klass)
{
rtppcmupay->adapter = gst_adapter_new ();
GST_BASE_RTP_PAYLOAD (rtppcmupay)->clock_rate = 8000;
}
static void
gst_rtp_pcmu_pay_finalize (GObject * object)
{
GstRtpPcmuPay *rtppcmupay;
rtppcmupay = GST_RTP_PCMU_PAY (object);
g_object_unref (rtppcmupay->adapter);
rtppcmupay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
payload->pt = GST_RTP_PAYLOAD_PCMU;
gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
gst_basertppayload_set_outcaps (payload, NULL);
return TRUE;
}
static GstFlowReturn
gst_rtp_pcmu_pay_flush (GstRtpPcmuPay * rtppcmupay, guint32 clock_rate)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
guint maxptime_octets = G_MAXUINT;
guint minptime_octets = GST_RTP_PCMU_MIN_PTIME_OCTETS;
if (GST_BASE_RTP_PAYLOAD (rtppcmupay)->max_ptime > 0) {
/* calculate octet count with:
maxptime-nsec * samples-per-sec / nsecs-per-sec * octets-per-sample */
maxptime_octets =
gst_util_uint64_scale_int (GST_BASE_RTP_PAYLOAD (rtppcmupay)->max_ptime,
clock_rate, GST_SECOND);
}
/* 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. */
avail = gst_adapter_available (rtppcmupay->adapter);
ret = GST_FLOW_OK;
while (avail >= minptime_octets) {
guint8 *payload;
guint8 *data;
guint payload_len;
guint packet_len;
/* fill one MTU or all available bytes */
payload_len =
MIN (MIN (GST_BASE_RTP_PAYLOAD_MTU (rtppcmupay), maxptime_octets),
avail);
/* this will be the total lenght of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
gst_rtp_buffer_set_payload_type (outbuf,
GST_BASE_RTP_PAYLOAD_PT (rtppcmupay));
payload = gst_rtp_buffer_get_payload (outbuf);
data = (guint8 *) gst_adapter_peek (rtppcmupay->adapter, payload_len);
memcpy (payload, data, payload_len);
gst_adapter_flush (rtppcmupay->adapter, payload_len);
avail -= payload_len;
GST_BUFFER_TIMESTAMP (outbuf) = rtppcmupay->first_ts;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtppcmupay), outbuf);
/* increase count (in ts) of data pushed to basertppayload */
rtppcmupay->first_ts +=
gst_util_uint64_scale_int (payload_len, GST_SECOND, clock_rate);
/* store amount of unpushed data (in ts) */
rtppcmupay->duration =
gst_util_uint64_scale_int (avail, GST_SECOND, clock_rate);
}
return ret;
}
static GstFlowReturn
gst_rtp_pcmu_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpPcmuPay *rtppcmupay;
guint size, packet_len, avail;
GstFlowReturn ret;
GstClockTime duration;
guint32 clock_rate;
rtppcmupay = GST_RTP_PCMU_PAY (basepayload);
clock_rate = basepayload->clock_rate;
size = GST_BUFFER_SIZE (buffer);
duration = gst_util_uint64_scale_int (size, GST_SECOND, clock_rate);
avail = gst_adapter_available (rtppcmupay->adapter);
if (avail == 0) {
rtppcmupay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtppcmupay->duration = 0;
}
/* get packet length of data and see if we exceeded MTU. */
packet_len = gst_rtp_buffer_calc_packet_len (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, rtppcmupay->duration + duration)) {
/* note: first_ts and duration updated in ...pay_flush() */
ret = gst_rtp_pcmu_pay_flush (rtppcmupay, clock_rate);
} else {
ret = GST_FLOW_OK;
}
gst_adapter_push (rtppcmupay->adapter, buffer);
rtppcmupay->duration += duration;
return ret;
}
gboolean
gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtppcmupay",
GST_RANK_NONE, GST_TYPE_RTP_PCMU_PAY);
}