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Original commit message from CVS: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: Fix element descriptions.
248 lines
7.8 KiB
C
248 lines
7.8 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
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* Copyright (C) <2005> Nokia Corporation <kai.vehmanen@nokia.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtppcmupay.h"
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/* elementfactory information */
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static const GstElementDetails gst_rtp_pcmu_pay_details =
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GST_ELEMENT_DETAILS ("RTP packet payloader",
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"Codec/Payloader/Network",
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"Payload-encodes PCMU audio into a RTP packet",
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"Edgard Lima <edgard.lima@indt.org.br>");
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static GstStaticPadTemplate gst_rtp_pcmu_pay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-mulaw, channels=(int)1, rate=(int)8000")
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);
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static GstStaticPadTemplate gst_rtp_pcmu_pay_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", "
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"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMU\"; "
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMU\"")
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);
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static gboolean gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_pcmu_pay_handle_buffer (GstBaseRTPPayload *
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payload, GstBuffer * buffer);
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static void gst_rtp_pcmu_pay_finalize (GObject * object);
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GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPPayload,
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GST_TYPE_BASE_RTP_PAYLOAD);
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/* The lower limit for number of octet to put in one packet
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* (clock-rate=8000, octet-per-sample=1). The default 80 is equal
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* to to 10msec (see RFC3551) */
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#define GST_RTP_PCMU_MIN_PTIME_OCTETS 80
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static void
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gst_rtp_pcmu_pay_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_pcmu_pay_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_pcmu_pay_src_template));
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gst_element_class_set_details (element_class, &gst_rtp_pcmu_pay_details);
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}
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static void
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gst_rtp_pcmu_pay_class_init (GstRtpPcmuPayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_rtp_pcmu_pay_finalize;
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gstbasertppayload_class->set_caps = gst_rtp_pcmu_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_pcmu_pay_handle_buffer;
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}
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static void
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gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay, GstRtpPcmuPayClass * klass)
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{
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rtppcmupay->adapter = gst_adapter_new ();
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GST_BASE_RTP_PAYLOAD (rtppcmupay)->clock_rate = 8000;
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}
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static void
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gst_rtp_pcmu_pay_finalize (GObject * object)
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{
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GstRtpPcmuPay *rtppcmupay;
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rtppcmupay = GST_RTP_PCMU_PAY (object);
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g_object_unref (rtppcmupay->adapter);
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rtppcmupay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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{
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payload->pt = GST_RTP_PAYLOAD_PCMU;
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gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
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gst_basertppayload_set_outcaps (payload, NULL);
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return TRUE;
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}
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static GstFlowReturn
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gst_rtp_pcmu_pay_flush (GstRtpPcmuPay * rtppcmupay, guint32 clock_rate)
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{
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guint avail;
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GstBuffer *outbuf;
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GstFlowReturn ret;
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guint maxptime_octets = G_MAXUINT;
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guint minptime_octets = GST_RTP_PCMU_MIN_PTIME_OCTETS;
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if (GST_BASE_RTP_PAYLOAD (rtppcmupay)->max_ptime > 0) {
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/* calculate octet count with:
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maxptime-nsec * samples-per-sec / nsecs-per-sec * octets-per-sample */
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maxptime_octets =
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gst_util_uint64_scale_int (GST_BASE_RTP_PAYLOAD (rtppcmupay)->max_ptime,
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clock_rate, GST_SECOND);
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}
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/* the data available in the adapter is either smaller
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* than the MTU or bigger. In the case it is smaller, the complete
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* adapter contents can be put in one packet. */
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avail = gst_adapter_available (rtppcmupay->adapter);
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ret = GST_FLOW_OK;
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while (avail >= minptime_octets) {
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guint8 *payload;
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guint8 *data;
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guint payload_len;
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guint packet_len;
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/* fill one MTU or all available bytes */
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payload_len =
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MIN (MIN (GST_BASE_RTP_PAYLOAD_MTU (rtppcmupay), maxptime_octets),
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avail);
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/* this will be the total lenght of the packet */
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packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
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/* create buffer to hold the payload */
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outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
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/* copy payload */
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gst_rtp_buffer_set_payload_type (outbuf,
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GST_BASE_RTP_PAYLOAD_PT (rtppcmupay));
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payload = gst_rtp_buffer_get_payload (outbuf);
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data = (guint8 *) gst_adapter_peek (rtppcmupay->adapter, payload_len);
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memcpy (payload, data, payload_len);
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gst_adapter_flush (rtppcmupay->adapter, payload_len);
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avail -= payload_len;
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GST_BUFFER_TIMESTAMP (outbuf) = rtppcmupay->first_ts;
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ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtppcmupay), outbuf);
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/* increase count (in ts) of data pushed to basertppayload */
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rtppcmupay->first_ts +=
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gst_util_uint64_scale_int (payload_len, GST_SECOND, clock_rate);
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/* store amount of unpushed data (in ts) */
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rtppcmupay->duration =
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gst_util_uint64_scale_int (avail, GST_SECOND, clock_rate);
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}
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return ret;
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}
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static GstFlowReturn
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gst_rtp_pcmu_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpPcmuPay *rtppcmupay;
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guint size, packet_len, avail;
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GstFlowReturn ret;
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GstClockTime duration;
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guint32 clock_rate;
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rtppcmupay = GST_RTP_PCMU_PAY (basepayload);
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clock_rate = basepayload->clock_rate;
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size = GST_BUFFER_SIZE (buffer);
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duration = gst_util_uint64_scale_int (size, GST_SECOND, clock_rate);
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avail = gst_adapter_available (rtppcmupay->adapter);
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if (avail == 0) {
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rtppcmupay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
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rtppcmupay->duration = 0;
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}
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/* get packet length of data and see if we exceeded MTU. */
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packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
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/* if this buffer is going to overflow the packet, flush what we
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* have. */
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if (gst_basertppayload_is_filled (basepayload,
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packet_len, rtppcmupay->duration + duration)) {
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/* note: first_ts and duration updated in ...pay_flush() */
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ret = gst_rtp_pcmu_pay_flush (rtppcmupay, clock_rate);
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} else {
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ret = GST_FLOW_OK;
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}
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gst_adapter_push (rtppcmupay->adapter, buffer);
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rtppcmupay->duration += duration;
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return ret;
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}
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gboolean
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gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtppcmupay",
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GST_RANK_NONE, GST_TYPE_RTP_PCMU_PAY);
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}
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