mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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2ba1de92a4
Original commit message from CVS: Patch by: Olivier Crete <tester at tester dot ca> * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps): Make the delivery-method mandatory on the caps and only accept inline for now. Reverse strcmp checks for delivery-method. * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps): Make delivery method optional when parsing caps and note this in the caps. Reverse strcmp checks for delivery-method. * gst/rtp/gstrtpvorbispay.c: Update a comment to note that the delivery-method is optional, Fixes #537675.
707 lines
21 KiB
C
707 lines
21 KiB
C
/* GStreamer
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* Copyright (C) <2006> Wim Taymans <wim@fluendo.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 <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "fnv1hash.h"
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#include "gstrtpvorbispay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpvorbispay_debug);
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#define GST_CAT_DEFAULT (rtpvorbispay_debug)
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/* references:
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* http://svn.xiph.org/trunk/vorbis/doc/draft-ietf-avt-rtp-vorbis-04.txt
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*/
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/* elementfactory information */
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static const GstElementDetails gst_rtp_vorbispay_details =
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GST_ELEMENT_DETAILS ("RTP packet depayloader",
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"Codec/Payloader/Network",
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"Payload-encode Vorbis audio into RTP packets (draft-01 RFC XXXX)",
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"Wim Taymans <wim@fluendo.com>");
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static GstStaticPadTemplate gst_rtp_vorbis_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_DYNAMIC_STRING ", "
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"clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"VORBIS\""
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/* All required parameters
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*
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* "encoding-params = (string) <num channels>"
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* "configuration = (string) ANY"
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*/
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/* All optional parameters
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*
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* "delivery-method = (string) { inline, in_band, out_band/<specific_name> } "
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* "configuration-uri ="
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*/
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)
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);
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static GstStaticPadTemplate gst_rtp_vorbis_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-vorbis")
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);
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GST_BOILERPLATE (GstRtpVorbisPay, gst_rtp_vorbis_pay, GstBaseRTPPayload,
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GST_TYPE_BASE_RTP_PAYLOAD);
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static gboolean gst_rtp_vorbis_pay_setcaps (GstBaseRTPPayload * basepayload,
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GstCaps * caps);
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static GstStateChangeReturn gst_rtp_vorbis_pay_change_state (GstElement *
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element, GstStateChange transition);
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static GstFlowReturn gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * pad,
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GstBuffer * buffer);
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static void
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gst_rtp_vorbis_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_vorbis_pay_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_vorbis_pay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_vorbispay_details);
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}
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static void
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gst_rtp_vorbis_pay_class_init (GstRtpVorbisPayClass * 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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gstelement_class->change_state = gst_rtp_vorbis_pay_change_state;
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gstbasertppayload_class->set_caps = gst_rtp_vorbis_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_vorbis_pay_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (rtpvorbispay_debug, "rtpvorbispay", 0,
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"Vorbis RTP Payloader");
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}
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static void
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gst_rtp_vorbis_pay_init (GstRtpVorbisPay * rtpvorbispay,
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GstRtpVorbisPayClass * klass)
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{
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/* needed because of GST_BOILERPLATE */
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}
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static void
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gst_rtp_vorbis_pay_cleanup (GstRtpVorbisPay * rtpvorbispay)
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{
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g_list_foreach (rtpvorbispay->headers, (GFunc) gst_mini_object_unref, NULL);
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g_list_free (rtpvorbispay->headers);
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rtpvorbispay->headers = NULL;
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if (rtpvorbispay->packet)
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gst_buffer_unref (rtpvorbispay->packet);
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rtpvorbispay->packet = NULL;
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}
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static gboolean
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gst_rtp_vorbis_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
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{
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GstRtpVorbisPay *rtpvorbispay;
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rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
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rtpvorbispay->need_headers = TRUE;
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return TRUE;
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}
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static void
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gst_rtp_vorbis_pay_reset_packet (GstRtpVorbisPay * rtpvorbispay, guint8 VDT)
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{
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guint payload_len;
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GST_LOG_OBJECT (rtpvorbispay, "reset packet");
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rtpvorbispay->payload_pos = 4;
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payload_len = gst_rtp_buffer_get_payload_len (rtpvorbispay->packet);
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rtpvorbispay->payload_left = payload_len - 4;
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rtpvorbispay->payload_duration = 0;
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rtpvorbispay->payload_F = 0;
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rtpvorbispay->payload_VDT = VDT;
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rtpvorbispay->payload_pkts = 0;
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}
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static void
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gst_rtp_vorbis_pay_init_packet (GstRtpVorbisPay * rtpvorbispay, guint8 VDT,
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GstClockTime timestamp)
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{
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GST_LOG_OBJECT (rtpvorbispay, "starting new packet, VDT: %d", VDT);
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if (rtpvorbispay->packet)
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gst_buffer_unref (rtpvorbispay->packet);
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/* new packet allocate max packet size */
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rtpvorbispay->packet =
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gst_rtp_buffer_new_allocate_len (GST_BASE_RTP_PAYLOAD_MTU
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(rtpvorbispay), 0, 0);
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gst_rtp_vorbis_pay_reset_packet (rtpvorbispay, VDT);
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GST_BUFFER_TIMESTAMP (rtpvorbispay->packet) = timestamp;
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}
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static GstFlowReturn
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gst_rtp_vorbis_pay_flush_packet (GstRtpVorbisPay * rtpvorbispay)
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{
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GstFlowReturn ret;
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guint8 *payload;
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guint hlen;
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/* check for empty packet */
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if (!rtpvorbispay->packet || rtpvorbispay->payload_pos <= 4)
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return GST_FLOW_OK;
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GST_LOG_OBJECT (rtpvorbispay, "flushing packet");
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/* fix header */
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payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet);
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/*
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* 0 1 2 3
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* 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
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | Ident | F |VDT|# pkts.|
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*
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* F: Fragment type (0=none, 1=start, 2=cont, 3=end)
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* VDT: Vorbis data type (0=vorbis, 1=config, 2=comment, 3=reserved)
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* pkts: number of packets.
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*/
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payload[0] = (rtpvorbispay->payload_ident >> 16) & 0xff;
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payload[1] = (rtpvorbispay->payload_ident >> 8) & 0xff;
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payload[2] = (rtpvorbispay->payload_ident) & 0xff;
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payload[3] = (rtpvorbispay->payload_F & 0x3) << 6 |
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(rtpvorbispay->payload_VDT & 0x3) << 4 |
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(rtpvorbispay->payload_pkts & 0xf);
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/* shrink the buffer size to the last written byte */
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hlen = gst_rtp_buffer_calc_header_len (0);
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GST_BUFFER_SIZE (rtpvorbispay->packet) = hlen + rtpvorbispay->payload_pos;
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GST_BUFFER_DURATION (rtpvorbispay->packet) = rtpvorbispay->payload_duration;
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/* push, this gives away our ref to the packet, so clear it. */
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ret =
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gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpvorbispay),
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rtpvorbispay->packet);
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rtpvorbispay->packet = NULL;
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return ret;
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}
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static gchar *
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encode_base64 (const guint8 * in, guint size, guint * len)
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{
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gchar *ret, *d;
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static const gchar v[] =
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"ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/";
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*len = ((size + 2) / 3) * 4;
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d = ret = (gchar *) g_malloc (*len + 1);
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for (; size; in += 3) { /* process tuplets */
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*d++ = v[in[0] >> 2]; /* byte 1: high 6 bits (1) */
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/* byte 2: low 2 bits (1), high 4 bits (2) */
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*d++ = v[((in[0] << 4) + (--size ? (in[1] >> 4) : 0)) & 0x3f];
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/* byte 3: low 4 bits (2), high 2 bits (3) */
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*d++ = size ? v[((in[1] << 2) + (--size ? (in[2] >> 6) : 0)) & 0x3f] : '=';
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/* byte 4: low 6 bits (3) */
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*d++ = size ? v[in[2] & 0x3f] : '=';
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if (size)
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size--; /* count third character if processed */
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}
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*d = '\0'; /* tie off string */
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return ret; /* return the resulting string */
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}
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static gboolean
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gst_rtp_vorbis_pay_finish_headers (GstBaseRTPPayload * basepayload)
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{
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GstRtpVorbisPay *rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
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GList *walk;
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guint length, size, n_headers, configlen;
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gchar *cstr, *configuration;
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guint8 *data, *config;
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guint32 ident;
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GST_DEBUG_OBJECT (rtpvorbispay, "finish headers");
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if (!rtpvorbispay->headers)
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goto no_headers;
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/* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | Number of packed headers |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | Packed header |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | Packed header |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | .... |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*
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* We only construct a config containing 1 packed header like this:
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*
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* 0 1 2 3
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* 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
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | Ident | length ..
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* .. | n. of headers | length1 | length2 ..
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* .. | Identification Header ..
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* .................................................................
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* .. | Comment Header ..
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* .................................................................
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* .. Comment Header |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | Setup Header ..
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* .................................................................
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* .. Setup Header |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*/
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/* we need 4 bytes for the number of headers (which is always 1), 3 bytes for
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* the ident, 2 bytes for length, 1 byte for n. of headers. */
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size = 4 + 3 + 2 + 1;
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/* count the size of the headers first and update the hash */
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length = 0;
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n_headers = 0;
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ident = fnv1_hash_32_new ();
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for (walk = rtpvorbispay->headers; walk; walk = g_list_next (walk)) {
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GstBuffer *buf = GST_BUFFER_CAST (walk->data);
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guint bsize;
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bsize = GST_BUFFER_SIZE (buf);
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length += bsize;
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n_headers++;
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/* count number of bytes needed for length fields, we don't need this for
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* the last header. */
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if (g_list_next (walk)) {
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do {
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size++;
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bsize >>= 7;
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} while (bsize);
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}
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/* update hash */
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ident = fnv1_hash_32_update (ident, GST_BUFFER_DATA (buf),
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GST_BUFFER_SIZE (buf));
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}
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/* packet length is header size + packet length */
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configlen = size + length;
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config = data = g_malloc (configlen);
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/* number of packed headers, we only pack 1 header */
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data[0] = 0;
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data[1] = 0;
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data[2] = 0;
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data[3] = 1;
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ident = fnv1_hash_32_to_24 (ident);
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rtpvorbispay->payload_ident = ident;
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GST_DEBUG_OBJECT (rtpvorbispay, "ident 0x%08x", ident);
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/* take lower 3 bytes */
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data[4] = (ident >> 16) & 0xff;
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data[5] = (ident >> 8) & 0xff;
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data[6] = ident & 0xff;
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/* store length of all vorbis headers */
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data[7] = ((length) >> 8) & 0xff;
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data[8] = (length) & 0xff;
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/* store number of headers minus one. */
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data[9] = n_headers - 1;
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data += 10;
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/* store length for each header */
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for (walk = rtpvorbispay->headers; walk; walk = g_list_next (walk)) {
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GstBuffer *buf = GST_BUFFER_CAST (walk->data);
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guint bsize, size, temp;
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/* only need to store the length when it's not the last header */
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if (!g_list_next (walk))
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break;
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bsize = GST_BUFFER_SIZE (buf);
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/* calc size */
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size = 0;
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do {
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size++;
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bsize >>= 7;
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} while (bsize);
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temp = size;
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bsize = GST_BUFFER_SIZE (buf);
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/* write the size backwards */
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while (size) {
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size--;
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data[size] = bsize & 0x7f;
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bsize >>= 7;
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}
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data += temp;
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}
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/* copy header data */
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for (walk = rtpvorbispay->headers; walk; walk = g_list_next (walk)) {
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GstBuffer *buf = GST_BUFFER_CAST (walk->data);
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memcpy (data, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
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data += GST_BUFFER_SIZE (buf);
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}
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/* serialize to base64 */
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configuration = encode_base64 (config, configlen, &size);
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g_free (config);
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/* configure payloader settings */
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cstr = g_strdup_printf ("%d", rtpvorbispay->channels);
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gst_basertppayload_set_options (basepayload, "audio", TRUE, "VORBIS",
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rtpvorbispay->rate);
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gst_basertppayload_set_outcaps (basepayload, "encoding-params", G_TYPE_STRING,
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cstr, "configuration", G_TYPE_STRING, configuration,
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"delivery-method", G_TYPE_STRING, "inline",
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/* don't set the other defaults
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*/
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NULL);
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g_free (cstr);
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g_free (configuration);
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return TRUE;
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/* ERRORS */
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no_headers:
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{
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GST_DEBUG_OBJECT (rtpvorbispay, "finish headers");
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return FALSE;
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}
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}
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static gboolean
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gst_rtp_vorbis_pay_parse_id (GstBaseRTPPayload * basepayload, guint8 * data,
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guint size)
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{
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GstRtpVorbisPay *rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
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guint8 channels;
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gint32 rate, version;
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if (G_UNLIKELY (size < 16))
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goto too_short;
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if (G_UNLIKELY (memcmp (data, "\001vorbis", 7)))
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goto invalid_start;
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data += 7;
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|
|
if (G_UNLIKELY ((version = GST_READ_UINT32_LE (data)) != 0))
|
|
goto invalid_version;
|
|
data += 4;
|
|
|
|
if (G_UNLIKELY ((channels = *data++) < 1))
|
|
goto invalid_channels;
|
|
|
|
if (G_UNLIKELY ((rate = GST_READ_UINT32_LE (data)) < 1))
|
|
goto invalid_rate;
|
|
|
|
/* all fine, store the values */
|
|
rtpvorbispay->channels = channels;
|
|
rtpvorbispay->rate = rate;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
too_short:
|
|
{
|
|
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
|
|
("Identification packet is too short, need at least 16, got %d", size),
|
|
(NULL));
|
|
return FALSE;
|
|
}
|
|
invalid_start:
|
|
{
|
|
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
|
|
("Invalid header start in identification packet"), (NULL));
|
|
return FALSE;
|
|
}
|
|
invalid_version:
|
|
{
|
|
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
|
|
("Invalid version, expected 0, got %d", version), (NULL));
|
|
return FALSE;
|
|
}
|
|
invalid_rate:
|
|
{
|
|
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
|
|
("Invalid rate %d", rate), (NULL));
|
|
return FALSE;
|
|
}
|
|
invalid_channels:
|
|
{
|
|
GST_ELEMENT_ERROR (basepayload, STREAM, DECODE,
|
|
("Invalid channels %d", channels), (NULL));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * basepayload,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpVorbisPay *rtpvorbispay;
|
|
GstFlowReturn ret;
|
|
guint size, newsize;
|
|
guint8 *data;
|
|
guint packet_len;
|
|
GstClockTime duration, newduration, timestamp;
|
|
gboolean flush;
|
|
guint8 VDT;
|
|
guint plen;
|
|
guint8 *ppos, *payload;
|
|
gboolean fragmented;
|
|
|
|
rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload);
|
|
|
|
size = GST_BUFFER_SIZE (buffer);
|
|
data = GST_BUFFER_DATA (buffer);
|
|
duration = GST_BUFFER_DURATION (buffer);
|
|
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
|
|
|
GST_LOG_OBJECT (rtpvorbispay, "size %u, duration %" GST_TIME_FORMAT,
|
|
size, GST_TIME_ARGS (duration));
|
|
|
|
if (G_UNLIKELY (size < 1 || size > 0xffff))
|
|
goto wrong_size;
|
|
|
|
/* find packet type */
|
|
if (data[0] & 1) {
|
|
/* header */
|
|
if (data[0] == 1) {
|
|
/* identification, we need to parse this in order to get the clock rate. */
|
|
if (G_UNLIKELY (!gst_rtp_vorbis_pay_parse_id (basepayload, data, size)))
|
|
goto parse_id_failed;
|
|
VDT = 1;
|
|
} else if (data[0] == 3) {
|
|
/* comment */
|
|
VDT = 2;
|
|
} else if (data[0] == 5) {
|
|
/* setup */
|
|
VDT = 1;
|
|
} else
|
|
goto unknown_header;
|
|
} else
|
|
/* data */
|
|
VDT = 0;
|
|
|
|
if (rtpvorbispay->need_headers) {
|
|
/* we need to collect the headers and construct a config string from them */
|
|
if (VDT != 0) {
|
|
GST_DEBUG_OBJECT (rtpvorbispay, "collecting header");
|
|
/* append header to the list of headers */
|
|
rtpvorbispay->headers = g_list_append (rtpvorbispay->headers, buffer);
|
|
ret = GST_FLOW_OK;
|
|
goto done;
|
|
} else {
|
|
if (!gst_rtp_vorbis_pay_finish_headers (basepayload))
|
|
goto header_error;
|
|
rtpvorbispay->need_headers = FALSE;
|
|
}
|
|
}
|
|
|
|
/* size increases with packet length and 2 bytes size eader. */
|
|
newduration = rtpvorbispay->payload_duration;
|
|
if (duration != GST_CLOCK_TIME_NONE)
|
|
newduration += duration;
|
|
|
|
newsize = rtpvorbispay->payload_pos + 2 + size;
|
|
packet_len = gst_rtp_buffer_calc_packet_len (newsize, 0, 0);
|
|
|
|
/* check buffer filled against length and max latency */
|
|
flush = gst_basertppayload_is_filled (basepayload, packet_len, newduration);
|
|
/* we can store up to 15 vorbis packets in one RTP packet. */
|
|
flush |= (rtpvorbispay->payload_pkts == 15);
|
|
/* flush if we have a new VDT */
|
|
if (rtpvorbispay->packet)
|
|
flush |= (rtpvorbispay->payload_VDT != VDT);
|
|
if (flush)
|
|
ret = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay);
|
|
|
|
/* create new packet if we must */
|
|
if (!rtpvorbispay->packet) {
|
|
gst_rtp_vorbis_pay_init_packet (rtpvorbispay, VDT, timestamp);
|
|
}
|
|
|
|
payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet);
|
|
ppos = payload + rtpvorbispay->payload_pos;
|
|
fragmented = FALSE;
|
|
|
|
ret = GST_FLOW_OK;
|
|
|
|
/* put buffer in packet, it either fits completely or needs to be fragmented
|
|
* over multiple RTP packets. */
|
|
while (size) {
|
|
plen = MIN (rtpvorbispay->payload_left - 2, size);
|
|
|
|
GST_LOG_OBJECT (rtpvorbispay, "append %u bytes", plen);
|
|
|
|
/* data is copied in the payload with a 2 byte length header */
|
|
ppos[0] = (plen >> 8) & 0xff;
|
|
ppos[1] = (plen & 0xff);
|
|
memcpy (&ppos[2], data, plen);
|
|
|
|
size -= plen;
|
|
data += plen;
|
|
|
|
rtpvorbispay->payload_pos += plen + 2;
|
|
rtpvorbispay->payload_left -= plen + 2;
|
|
|
|
if (fragmented) {
|
|
if (size == 0)
|
|
/* last fragment, set F to 0x3. */
|
|
rtpvorbispay->payload_F = 0x3;
|
|
else
|
|
/* fragment continues, set F to 0x2. */
|
|
rtpvorbispay->payload_F = 0x2;
|
|
} else {
|
|
if (size > 0) {
|
|
/* fragmented packet starts, set F to 0x1, mark ourselves as
|
|
* fragmented. */
|
|
rtpvorbispay->payload_F = 0x1;
|
|
fragmented = TRUE;
|
|
}
|
|
}
|
|
if (fragmented) {
|
|
/* fragmented packets are always flushed and have ptks of 0 */
|
|
rtpvorbispay->payload_pkts = 0;
|
|
ret = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay);
|
|
|
|
if (size > 0) {
|
|
/* start new packet and get pointers. VDT stays the same. */
|
|
gst_rtp_vorbis_pay_init_packet (rtpvorbispay,
|
|
rtpvorbispay->payload_VDT, timestamp);
|
|
payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet);
|
|
ppos = payload + rtpvorbispay->payload_pos;
|
|
}
|
|
} else {
|
|
/* unfragmented packet, update stats for next packet, size == 0 and we
|
|
* exit the while loop */
|
|
rtpvorbispay->payload_pkts++;
|
|
if (duration != GST_CLOCK_TIME_NONE)
|
|
rtpvorbispay->payload_duration += duration;
|
|
}
|
|
}
|
|
gst_buffer_unref (buffer);
|
|
|
|
done:
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
wrong_size:
|
|
{
|
|
GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
|
|
("Invalid packet size (1 < %d <= 0xffff)", size), (NULL));
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
parse_id_failed:
|
|
{
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
unknown_header:
|
|
{
|
|
GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
|
|
(NULL), ("Ignoring unknown header received"));
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
header_error:
|
|
{
|
|
GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE,
|
|
(NULL), ("Error initializing header config"));
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_vorbis_pay_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstRtpVorbisPay *rtpvorbispay;
|
|
GstStateChangeReturn ret;
|
|
|
|
rtpvorbispay = GST_RTP_VORBIS_PAY (element);
|
|
|
|
switch (transition) {
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_rtp_vorbis_pay_cleanup (rtpvorbispay);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_vorbis_pay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpvorbispay",
|
|
GST_RANK_NONE, GST_TYPE_RTP_VORBIS_PAY);
|
|
}
|