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f0f72088bc
Fixes #594258
1855 lines
45 KiB
C
1855 lines
45 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
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*
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* gstrtcpbuffer.h: various helper functions to manipulate buffers
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* with RTCP payload.
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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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/**
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* SECTION:gstrtcpbuffer
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* @short_description: Helper methods for dealing with RTCP buffers
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* @see_also: #GstBaseRTPPayload, #GstBaseRTPDepayload, #gstrtpbuffer
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*
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* Note: The API in this module is not yet declared stable.
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*
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* <refsect2>
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* <para>
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* The GstRTPCBuffer helper functions makes it easy to parse and create regular
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* #GstBuffer objects that contain compound RTCP packets. These buffers are typically
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* of 'application/x-rtcp' #GstCaps.
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* </para>
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* <para>
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* An RTCP buffer consists of 1 or more #GstRTCPPacket structures that you can
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* retrieve with gst_rtcp_buffer_get_first_packet(). #GstRTCPPacket acts as a pointer
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* into the RTCP buffer; you can move to the next packet with
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* gst_rtcp_packet_move_to_next().
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2007-03-26 (0.10.13)
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*
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* Since: 0.10.13
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*/
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#include <string.h>
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#include "gstrtcpbuffer.h"
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/**
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* gst_rtcp_buffer_new_take_data:
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* @data: data for the new buffer
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* @len: the length of data
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*
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* Create a new buffer and set the data and size of the buffer to @data and @len
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* respectively. @data will be freed when the buffer is unreffed, so this
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* function transfers ownership of @data to the new buffer.
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*
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* Returns: A newly allocated buffer with @data and of size @len.
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*/
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GstBuffer *
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gst_rtcp_buffer_new_take_data (gpointer data, guint len)
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{
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GstBuffer *result;
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g_return_val_if_fail (data != NULL, NULL);
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g_return_val_if_fail (len > 0, NULL);
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result = gst_buffer_new ();
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GST_BUFFER_MALLOCDATA (result) = data;
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GST_BUFFER_DATA (result) = data;
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GST_BUFFER_SIZE (result) = len;
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return result;
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}
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/**
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* gst_rtcp_buffer_new_copy_data:
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* @data: data for the new buffer
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* @len: the length of data
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*
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* Create a new buffer and set the data to a copy of @len
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* bytes of @data and the size to @len. The data will be freed when the buffer
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* is freed.
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*
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* Returns: A newly allocated buffer with a copy of @data and of size @len.
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*/
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GstBuffer *
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gst_rtcp_buffer_new_copy_data (gpointer data, guint len)
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{
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return gst_rtcp_buffer_new_take_data (g_memdup (data, len), len);
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}
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/**
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* gst_rtcp_buffer_validate_data:
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* @data: the data to validate
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* @len: the length of @data to validate
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*
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* Check if the @data and @size point to the data of a valid RTCP (compound)
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* packet.
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* Use this function to validate a packet before using the other functions in
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* this module.
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*
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* Returns: TRUE if the data points to a valid RTCP packet.
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*/
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gboolean
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gst_rtcp_buffer_validate_data (guint8 * data, guint len)
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{
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guint16 header_mask;
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guint16 header_len;
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guint8 version;
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guint data_len;
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gboolean padding;
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guint8 pad_bytes;
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g_return_val_if_fail (data != NULL, FALSE);
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/* we need 4 bytes for the type and length */
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if (G_UNLIKELY (len < 4))
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goto wrong_length;
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/* first packet must be RR or SR and version must be 2 */
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header_mask = ((data[0] << 8) | data[1]) & GST_RTCP_VALID_MASK;
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if (G_UNLIKELY (header_mask != GST_RTCP_VALID_VALUE))
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goto wrong_mask;
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/* no padding when mask succeeds */
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padding = FALSE;
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/* store len */
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data_len = len;
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while (TRUE) {
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/* get packet length */
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header_len = (((data[2] << 8) | data[3]) + 1) << 2;
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if (data_len < header_len)
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goto wrong_length;
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/* move to next compount packet */
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data += header_len;
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data_len -= header_len;
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/* we are at the end now */
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if (data_len < 4)
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break;
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/* check version of new packet */
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version = data[0] & 0xc0;
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if (version != (GST_RTCP_VERSION << 6))
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goto wrong_version;
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/* padding only allowed on last packet */
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if ((padding = data[0] & 0x20))
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break;
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}
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if (data_len > 0) {
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/* some leftover bytes, check padding */
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if (!padding)
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goto wrong_length;
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/* get padding */
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pad_bytes = data[len - 1];
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if (data_len != pad_bytes)
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goto wrong_padding;
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}
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return TRUE;
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/* ERRORS */
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wrong_length:
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{
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GST_DEBUG ("len check failed");
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return FALSE;
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}
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wrong_mask:
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{
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GST_DEBUG ("mask check failed (%04x != %04x)", header_mask,
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GST_RTCP_VALID_VALUE);
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return FALSE;
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}
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wrong_version:
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{
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GST_DEBUG ("wrong version (%d < 2)", version >> 6);
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return FALSE;
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}
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wrong_padding:
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{
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GST_DEBUG ("padding check failed");
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return FALSE;
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}
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}
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/**
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* gst_rtcp_buffer_validate:
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* @buffer: the buffer to validate
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*
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* Check if the data pointed to by @buffer is a valid RTCP packet using
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* gst_rtcp_buffer_validate_data().
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*
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* Returns: TRUE if @buffer is a valid RTCP packet.
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*/
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gboolean
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gst_rtcp_buffer_validate (GstBuffer * buffer)
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{
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guint8 *data;
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guint len;
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g_return_val_if_fail (GST_IS_BUFFER (buffer), FALSE);
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data = GST_BUFFER_DATA (buffer);
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len = GST_BUFFER_SIZE (buffer);
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return gst_rtcp_buffer_validate_data (data, len);
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}
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/**
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* gst_rtcp_buffer_new:
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* @mtu: the maximum mtu size.
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*
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* Create a new buffer for constructing RTCP packets. The packet will have a
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* maximum size of @mtu.
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*
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* Returns: A newly allocated buffer.
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*/
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GstBuffer *
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gst_rtcp_buffer_new (guint mtu)
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{
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GstBuffer *result;
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g_return_val_if_fail (mtu > 0, NULL);
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result = gst_buffer_new ();
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GST_BUFFER_MALLOCDATA (result) = g_malloc0 (mtu);
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GST_BUFFER_DATA (result) = GST_BUFFER_MALLOCDATA (result);
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GST_BUFFER_SIZE (result) = mtu;
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return result;
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}
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/**
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* gst_rtcp_buffer_end:
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* @buffer: a buffer with an RTCP packet
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*
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* Finish @buffer after being constructured. This function is usually called
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* after gst_rtcp_buffer_new() and after adding the RTCP items to the new buffer.
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*
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* The function adjusts the size of @buffer with the total length of all the
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* added packets.
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*/
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void
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gst_rtcp_buffer_end (GstBuffer * buffer)
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{
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GstRTCPPacket packet;
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g_return_if_fail (GST_IS_BUFFER (buffer));
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/* move to the first free space */
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if (gst_rtcp_buffer_get_first_packet (buffer, &packet))
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while (gst_rtcp_packet_move_to_next (&packet));
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/* shrink size */
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GST_BUFFER_SIZE (buffer) = packet.offset;
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}
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/**
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* gst_rtcp_buffer_get_packet_count:
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* @buffer: a valid RTCP buffer
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*
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* Get the number of RTCP packets in @buffer.
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*
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* Returns: the number of RTCP packets in @buffer.
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*/
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guint
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gst_rtcp_buffer_get_packet_count (GstBuffer * buffer)
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{
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GstRTCPPacket packet;
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guint count;
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g_return_val_if_fail (GST_IS_BUFFER (buffer), 0);
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count = 0;
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if (gst_rtcp_buffer_get_first_packet (buffer, &packet)) {
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do {
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count++;
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} while (gst_rtcp_packet_move_to_next (&packet));
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}
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return count;
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}
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/**
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* read_packet_header:
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* @packet: a packet
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*
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* Read the packet headers for the packet pointed to by @packet.
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*
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* Returns: TRUE if @packet pointed to a valid header.
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*/
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static gboolean
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read_packet_header (GstRTCPPacket * packet)
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{
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guint8 *data;
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guint size;
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guint offset;
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g_return_val_if_fail (packet != NULL, FALSE);
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g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
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data = GST_BUFFER_DATA (packet->buffer);
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size = GST_BUFFER_SIZE (packet->buffer);
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offset = packet->offset;
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/* check if we are at the end of the buffer, we add 4 because we also want to
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* ensure we can read the header. */
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if (offset + 4 > size)
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return FALSE;
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if ((data[offset] & 0xc0) != (GST_RTCP_VERSION << 6))
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return FALSE;
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/* read count, type and length */
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packet->padding = (data[offset] & 0x20) == 0x20;
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packet->count = data[offset] & 0x1f;
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packet->type = data[offset + 1];
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packet->length = (data[offset + 2] << 8) | data[offset + 3];
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packet->item_offset = 4;
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packet->item_count = 0;
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packet->entry_offset = 4;
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return TRUE;
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}
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/**
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* gst_rtcp_buffer_get_first_packet:
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* @buffer: a valid RTCP buffer
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* @packet: a #GstRTCPPacket
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*
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* Initialize a new #GstRTCPPacket pointer that points to the first packet in
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* @buffer.
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*
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* Returns: TRUE if the packet existed in @buffer.
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*/
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gboolean
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gst_rtcp_buffer_get_first_packet (GstBuffer * buffer, GstRTCPPacket * packet)
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{
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g_return_val_if_fail (GST_IS_BUFFER (buffer), FALSE);
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g_return_val_if_fail (packet != NULL, FALSE);
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/* init to 0 */
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packet->buffer = buffer;
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packet->offset = 0;
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packet->type = GST_RTCP_TYPE_INVALID;
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if (!read_packet_header (packet))
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return FALSE;
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return TRUE;
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}
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/**
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* gst_rtcp_packet_move_to_next:
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* @packet: a #GstRTCPPacket
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*
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* Move the packet pointer @packet to the next packet in the payload.
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* Use gst_rtcp_buffer_get_first_packet() to initialize @packet.
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*
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* Returns: TRUE if @packet is pointing to a valid packet after calling this
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* function.
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*/
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gboolean
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gst_rtcp_packet_move_to_next (GstRTCPPacket * packet)
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{
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g_return_val_if_fail (packet != NULL, FALSE);
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g_return_val_if_fail (packet->type != GST_RTCP_TYPE_INVALID, FALSE);
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g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
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/* if we have a padding or invalid packet, it must be the last,
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* return FALSE */
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if (packet->type == GST_RTCP_TYPE_INVALID || packet->padding)
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goto end;
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/* move to next packet. Add 4 because the header is not included in length */
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packet->offset += (packet->length << 2) + 4;
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/* try to read new header */
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if (!read_packet_header (packet))
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goto end;
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return TRUE;
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/* ERRORS */
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end:
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{
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packet->type = GST_RTCP_TYPE_INVALID;
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return FALSE;
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}
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}
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/**
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* gst_rtcp_buffer_add_packet:
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* @buffer: a valid RTCP buffer
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* @type: the #GstRTCPType of the new packet
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* @packet: pointer to new packet
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*
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* Add a new packet of @type to @buffer. @packet will point to the newly created
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* packet.
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*
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* Returns: %TRUE if the packet could be created. This function returns %FALSE
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* if the max mtu is exceeded for the buffer.
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*/
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gboolean
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gst_rtcp_buffer_add_packet (GstBuffer * buffer, GstRTCPType type,
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GstRTCPPacket * packet)
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{
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guint len, size;
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guint8 *data;
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gboolean result;
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g_return_val_if_fail (GST_IS_BUFFER (buffer), FALSE);
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g_return_val_if_fail (type != GST_RTCP_TYPE_INVALID, FALSE);
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g_return_val_if_fail (packet != NULL, FALSE);
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/* find free space */
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if (gst_rtcp_buffer_get_first_packet (buffer, packet))
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while (gst_rtcp_packet_move_to_next (packet));
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size = GST_BUFFER_SIZE (buffer);
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/* packet->offset is now pointing to the next free offset in the buffer to
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* start a compount packet. Next we figure out if we have enough free space in
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* the buffer to continue. */
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switch (type) {
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case GST_RTCP_TYPE_SR:
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len = 28;
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break;
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case GST_RTCP_TYPE_RR:
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len = 8;
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break;
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case GST_RTCP_TYPE_SDES:
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len = 4;
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break;
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case GST_RTCP_TYPE_BYE:
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len = 4;
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break;
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case GST_RTCP_TYPE_APP:
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len = 12;
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break;
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case GST_RTCP_TYPE_RTPFB:
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len = 12;
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break;
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case GST_RTCP_TYPE_PSFB:
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len = 12;
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break;
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default:
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goto unknown_type;
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}
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if (packet->offset + len >= size)
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goto no_space;
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data = GST_BUFFER_DATA (buffer) + packet->offset;
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data[0] = (GST_RTCP_VERSION << 6);
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data[1] = type;
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/* length is stored in multiples of 32 bit words minus the length of the
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* header */
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len = (len - 4) >> 2;
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data[2] = len >> 8;
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data[3] = len & 0xff;
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|
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/* now try to position to the packet */
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result = read_packet_header (packet);
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return result;
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/* ERRORS */
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unknown_type:
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{
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g_warning ("unknown type %d", type);
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return FALSE;
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}
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no_space:
|
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{
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return FALSE;
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}
|
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}
|
|
|
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/**
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* gst_rtcp_packet_remove:
|
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* @packet: a #GstRTCPPacket
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*
|
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* Removes the packet pointed to by @packet and moves pointer to the next one
|
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*
|
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* Returns: TRUE if @packet is pointing to a valid packet after calling this
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* function.
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*/
|
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gboolean
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gst_rtcp_packet_remove (GstRTCPPacket * packet)
|
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{
|
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gboolean ret = FALSE;
|
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guint offset = 0;
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|
|
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g_return_val_if_fail (packet != NULL, FALSE);
|
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g_return_val_if_fail (packet->type != GST_RTCP_TYPE_INVALID, FALSE);
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|
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/* The next packet starts at offset + length + 4 (the header) */
|
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offset = packet->offset + (packet->length << 2) + 4;
|
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|
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/* Overwrite this packet with the rest of the data */
|
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memmove (GST_BUFFER_DATA (packet->buffer) + packet->offset,
|
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GST_BUFFER_DATA (packet->buffer) + offset,
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GST_BUFFER_SIZE (packet->buffer) - offset);
|
|
|
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/* try to read next header */
|
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ret = read_packet_header (packet);
|
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if (!ret)
|
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packet->type = GST_RTCP_TYPE_INVALID;
|
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|
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return ret;
|
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}
|
|
|
|
/**
|
|
* gst_rtcp_packet_get_padding:
|
|
* @packet: a valid #GstRTCPPacket
|
|
*
|
|
* Get the packet padding of the packet pointed to by @packet.
|
|
*
|
|
* Returns: If the packet has the padding bit set.
|
|
*/
|
|
gboolean
|
|
gst_rtcp_packet_get_padding (GstRTCPPacket * packet)
|
|
{
|
|
g_return_val_if_fail (packet != NULL, FALSE);
|
|
g_return_val_if_fail (packet->type != GST_RTCP_TYPE_INVALID, FALSE);
|
|
|
|
return packet->padding;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_get_type:
|
|
* @packet: a valid #GstRTCPPacket
|
|
*
|
|
* Get the packet type of the packet pointed to by @packet.
|
|
*
|
|
* Returns: The packet type or GST_RTCP_TYPE_INVALID when @packet is not
|
|
* pointing to a valid packet.
|
|
*/
|
|
GstRTCPType
|
|
gst_rtcp_packet_get_type (GstRTCPPacket * packet)
|
|
{
|
|
g_return_val_if_fail (packet != NULL, GST_RTCP_TYPE_INVALID);
|
|
|
|
return packet->type;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_get_count:
|
|
* @packet: a valid #GstRTCPPacket
|
|
*
|
|
* Get the count field in @packet.
|
|
*
|
|
* Returns: The count field in @packet or -1 if @packet does not point to a
|
|
* valid packet.
|
|
*/
|
|
guint8
|
|
gst_rtcp_packet_get_count (GstRTCPPacket * packet)
|
|
{
|
|
g_return_val_if_fail (packet != NULL, -1);
|
|
g_return_val_if_fail (packet->type != GST_RTCP_TYPE_INVALID, -1);
|
|
|
|
return packet->count;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_get_length:
|
|
* @packet: a valid #GstRTCPPacket
|
|
*
|
|
* Get the length field of @packet. This is the length of the packet in
|
|
* 32-bit words minus one.
|
|
*
|
|
* Returns: The length field of @packet.
|
|
*/
|
|
guint16
|
|
gst_rtcp_packet_get_length (GstRTCPPacket * packet)
|
|
{
|
|
g_return_val_if_fail (packet != NULL, 0);
|
|
g_return_val_if_fail (packet->type != GST_RTCP_TYPE_INVALID, 0);
|
|
|
|
return packet->length;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_sr_get_sender_info:
|
|
* @packet: a valid SR #GstRTCPPacket
|
|
* @ssrc: result SSRC
|
|
* @ntptime: result NTP time
|
|
* @rtptime: result RTP time
|
|
* @packet_count: result packet count
|
|
* @octet_count: result octect count
|
|
*
|
|
* Parse the SR sender info and store the values.
|
|
*/
|
|
void
|
|
gst_rtcp_packet_sr_get_sender_info (GstRTCPPacket * packet, guint32 * ssrc,
|
|
guint64 * ntptime, guint32 * rtptime, guint32 * packet_count,
|
|
guint32 * octet_count)
|
|
{
|
|
guint8 *data;
|
|
|
|
g_return_if_fail (packet != NULL);
|
|
g_return_if_fail (packet->type == GST_RTCP_TYPE_SR);
|
|
g_return_if_fail (GST_IS_BUFFER (packet->buffer));
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
|
|
/* skip header */
|
|
data += packet->offset + 4;
|
|
if (ssrc)
|
|
*ssrc = GST_READ_UINT32_BE (data);
|
|
data += 4;
|
|
if (ntptime)
|
|
*ntptime = GST_READ_UINT64_BE (data);
|
|
data += 8;
|
|
if (rtptime)
|
|
*rtptime = GST_READ_UINT32_BE (data);
|
|
data += 4;
|
|
if (packet_count)
|
|
*packet_count = GST_READ_UINT32_BE (data);
|
|
data += 4;
|
|
if (octet_count)
|
|
*octet_count = GST_READ_UINT32_BE (data);
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_sr_set_sender_info:
|
|
* @packet: a valid SR #GstRTCPPacket
|
|
* @ssrc: the SSRC
|
|
* @ntptime: the NTP time
|
|
* @rtptime: the RTP time
|
|
* @packet_count: the packet count
|
|
* @octet_count: the octect count
|
|
*
|
|
* Set the given values in the SR packet @packet.
|
|
*/
|
|
void
|
|
gst_rtcp_packet_sr_set_sender_info (GstRTCPPacket * packet, guint32 ssrc,
|
|
guint64 ntptime, guint32 rtptime, guint32 packet_count, guint32 octet_count)
|
|
{
|
|
guint8 *data;
|
|
|
|
g_return_if_fail (packet != NULL);
|
|
g_return_if_fail (packet->type == GST_RTCP_TYPE_SR);
|
|
g_return_if_fail (GST_IS_BUFFER (packet->buffer));
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
|
|
/* skip header */
|
|
data += packet->offset + 4;
|
|
GST_WRITE_UINT32_BE (data, ssrc);
|
|
data += 4;
|
|
GST_WRITE_UINT64_BE (data, ntptime);
|
|
data += 8;
|
|
GST_WRITE_UINT32_BE (data, rtptime);
|
|
data += 4;
|
|
GST_WRITE_UINT32_BE (data, packet_count);
|
|
data += 4;
|
|
GST_WRITE_UINT32_BE (data, octet_count);
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_rr_get_ssrc:
|
|
* @packet: a valid RR #GstRTCPPacket
|
|
*
|
|
* Get the ssrc field of the RR @packet.
|
|
*
|
|
* Returns: the ssrc.
|
|
*/
|
|
guint32
|
|
gst_rtcp_packet_rr_get_ssrc (GstRTCPPacket * packet)
|
|
{
|
|
guint8 *data;
|
|
guint32 ssrc;
|
|
|
|
g_return_val_if_fail (packet != NULL, 0);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_RR, 0);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), 0);
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
|
|
/* skip header */
|
|
data += packet->offset + 4;
|
|
ssrc = GST_READ_UINT32_BE (data);
|
|
|
|
return ssrc;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_rr_set_ssrc:
|
|
* @packet: a valid RR #GstRTCPPacket
|
|
* @ssrc: the SSRC to set
|
|
*
|
|
* Set the ssrc field of the RR @packet.
|
|
*/
|
|
void
|
|
gst_rtcp_packet_rr_set_ssrc (GstRTCPPacket * packet, guint32 ssrc)
|
|
{
|
|
guint8 *data;
|
|
|
|
g_return_if_fail (packet != NULL);
|
|
g_return_if_fail (packet->type == GST_RTCP_TYPE_RR);
|
|
g_return_if_fail (GST_IS_BUFFER (packet->buffer));
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
|
|
/* skip header */
|
|
data += packet->offset + 4;
|
|
GST_WRITE_UINT32_BE (data, ssrc);
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_get_rb_count:
|
|
* @packet: a valid SR or RR #GstRTCPPacket
|
|
*
|
|
* Get the number of report blocks in @packet.
|
|
*
|
|
* Returns: The number of report blocks in @packet.
|
|
*/
|
|
guint
|
|
gst_rtcp_packet_get_rb_count (GstRTCPPacket * packet)
|
|
{
|
|
g_return_val_if_fail (packet != NULL, 0);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_RR ||
|
|
packet->type == GST_RTCP_TYPE_SR, 0);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), 0);
|
|
|
|
return packet->count;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_get_rb:
|
|
* @packet: a valid SR or RR #GstRTCPPacket
|
|
* @nth: the nth report block in @packet
|
|
* @ssrc: result for data source being reported
|
|
* @fractionlost: result for fraction lost since last SR/RR
|
|
* @packetslost: result for the cumululative number of packets lost
|
|
* @exthighestseq: result for the extended last sequence number received
|
|
* @jitter: result for the interarrival jitter
|
|
* @lsr: result for the last SR packet from this source
|
|
* @dlsr: result for the delay since last SR packet
|
|
*
|
|
* Parse the values of the @nth report block in @packet and store the result in
|
|
* the values.
|
|
*/
|
|
void
|
|
gst_rtcp_packet_get_rb (GstRTCPPacket * packet, guint nth, guint32 * ssrc,
|
|
guint8 * fractionlost, gint32 * packetslost, guint32 * exthighestseq,
|
|
guint32 * jitter, guint32 * lsr, guint32 * dlsr)
|
|
{
|
|
guint8 *data;
|
|
guint32 tmp;
|
|
|
|
g_return_if_fail (packet != NULL);
|
|
g_return_if_fail (packet->type == GST_RTCP_TYPE_RR ||
|
|
packet->type == GST_RTCP_TYPE_SR);
|
|
g_return_if_fail (GST_IS_BUFFER (packet->buffer));
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
|
|
/* skip header */
|
|
data += packet->offset + 4;
|
|
if (packet->type == GST_RTCP_TYPE_RR)
|
|
data += 4;
|
|
else
|
|
data += 24;
|
|
|
|
/* move to requested index */
|
|
data += (nth * 24);
|
|
|
|
if (ssrc)
|
|
*ssrc = GST_READ_UINT32_BE (data);
|
|
data += 4;
|
|
tmp = GST_READ_UINT32_BE (data);
|
|
if (fractionlost)
|
|
*fractionlost = (tmp >> 24);
|
|
if (packetslost) {
|
|
/* sign extend */
|
|
if (tmp & 0x00800000)
|
|
tmp |= 0xff000000;
|
|
else
|
|
tmp &= 0x00ffffff;
|
|
*packetslost = (gint32) tmp;
|
|
}
|
|
data += 4;
|
|
if (exthighestseq)
|
|
*exthighestseq = GST_READ_UINT32_BE (data);
|
|
data += 4;
|
|
if (jitter)
|
|
*jitter = GST_READ_UINT32_BE (data);
|
|
data += 4;
|
|
if (lsr)
|
|
*lsr = GST_READ_UINT32_BE (data);
|
|
data += 4;
|
|
if (dlsr)
|
|
*dlsr = GST_READ_UINT32_BE (data);
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_add_rb:
|
|
* @packet: a valid SR or RR #GstRTCPPacket
|
|
* @ssrc: data source being reported
|
|
* @fractionlost: fraction lost since last SR/RR
|
|
* @packetslost: the cumululative number of packets lost
|
|
* @exthighestseq: the extended last sequence number received
|
|
* @jitter: the interarrival jitter
|
|
* @lsr: the last SR packet from this source
|
|
* @dlsr: the delay since last SR packet
|
|
*
|
|
* Add a new report block to @packet with the given values.
|
|
*
|
|
* Returns: %TRUE if the packet was created. This function can return %FALSE if
|
|
* the max MTU is exceeded or the number of report blocks is greater than
|
|
* #GST_RTCP_MAX_RB_COUNT.
|
|
*/
|
|
gboolean
|
|
gst_rtcp_packet_add_rb (GstRTCPPacket * packet, guint32 ssrc,
|
|
guint8 fractionlost, gint32 packetslost, guint32 exthighestseq,
|
|
guint32 jitter, guint32 lsr, guint32 dlsr)
|
|
{
|
|
guint8 *data;
|
|
guint size, offset;
|
|
|
|
g_return_val_if_fail (packet != NULL, FALSE);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_RR ||
|
|
packet->type == GST_RTCP_TYPE_SR, FALSE);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
|
|
|
|
if (packet->count >= GST_RTCP_MAX_RB_COUNT)
|
|
goto no_space;
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
size = GST_BUFFER_SIZE (packet->buffer);
|
|
|
|
/* skip header */
|
|
offset = packet->offset + 4;
|
|
if (packet->type == GST_RTCP_TYPE_RR)
|
|
offset += 4;
|
|
else
|
|
offset += 24;
|
|
|
|
/* move to current index */
|
|
offset += (packet->count * 24);
|
|
|
|
/* we need 24 free bytes now */
|
|
if (offset + 24 >= size)
|
|
goto no_space;
|
|
|
|
/* increment packet count and length */
|
|
packet->count++;
|
|
data[packet->offset]++;
|
|
packet->length += 6;
|
|
data[packet->offset + 2] = (packet->length) >> 8;
|
|
data[packet->offset + 3] = (packet->length) & 0xff;
|
|
|
|
/* move to new report block offset */
|
|
data += offset;
|
|
|
|
GST_WRITE_UINT32_BE (data, ssrc);
|
|
data += 4;
|
|
GST_WRITE_UINT32_BE (data, (fractionlost << 24) | (packetslost & 0xffffff));
|
|
data += 4;
|
|
GST_WRITE_UINT32_BE (data, exthighestseq);
|
|
data += 4;
|
|
GST_WRITE_UINT32_BE (data, jitter);
|
|
data += 4;
|
|
GST_WRITE_UINT32_BE (data, lsr);
|
|
data += 4;
|
|
GST_WRITE_UINT32_BE (data, dlsr);
|
|
|
|
return TRUE;
|
|
|
|
no_space:
|
|
{
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_set_rb:
|
|
* @packet: a valid SR or RR #GstRTCPPacket
|
|
* @nth: the nth report block to set
|
|
* @ssrc: data source being reported
|
|
* @fractionlost: fraction lost since last SR/RR
|
|
* @packetslost: the cumululative number of packets lost
|
|
* @exthighestseq: the extended last sequence number received
|
|
* @jitter: the interarrival jitter
|
|
* @lsr: the last SR packet from this source
|
|
* @dlsr: the delay since last SR packet
|
|
*
|
|
* Set the @nth new report block in @packet with the given values.
|
|
*
|
|
* Note: Not implemented.
|
|
*/
|
|
void
|
|
gst_rtcp_packet_set_rb (GstRTCPPacket * packet, guint nth, guint32 ssrc,
|
|
guint8 fractionlost, gint32 packetslost, guint32 exthighestseq,
|
|
guint32 jitter, guint32 lsr, guint32 dlsr)
|
|
{
|
|
g_return_if_fail (packet != NULL);
|
|
g_return_if_fail (packet->type == GST_RTCP_TYPE_RR ||
|
|
packet->type == GST_RTCP_TYPE_SR);
|
|
g_return_if_fail (GST_IS_BUFFER (packet->buffer));
|
|
|
|
g_warning ("not implemented");
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_rtcp_packet_sdes_get_item_count:
|
|
* @packet: a valid SDES #GstRTCPPacket
|
|
*
|
|
* Get the number of items in the SDES packet @packet.
|
|
*
|
|
* Returns: The number of items in @packet.
|
|
*/
|
|
guint
|
|
gst_rtcp_packet_sdes_get_item_count (GstRTCPPacket * packet)
|
|
{
|
|
g_return_val_if_fail (packet != NULL, 0);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_SDES, 0);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), 0);
|
|
|
|
return packet->count;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_sdes_first_item:
|
|
* @packet: a valid SDES #GstRTCPPacket
|
|
*
|
|
* Move to the first SDES item in @packet.
|
|
*
|
|
* Returns: TRUE if there was a first item.
|
|
*/
|
|
gboolean
|
|
gst_rtcp_packet_sdes_first_item (GstRTCPPacket * packet)
|
|
{
|
|
g_return_val_if_fail (packet != NULL, FALSE);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_SDES, FALSE);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
|
|
|
|
packet->item_offset = 4;
|
|
packet->item_count = 0;
|
|
packet->entry_offset = 4;
|
|
|
|
if (packet->count == 0)
|
|
return FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_sdes_next_item:
|
|
* @packet: a valid SDES #GstRTCPPacket
|
|
*
|
|
* Move to the next SDES item in @packet.
|
|
*
|
|
* Returns: TRUE if there was a next item.
|
|
*/
|
|
gboolean
|
|
gst_rtcp_packet_sdes_next_item (GstRTCPPacket * packet)
|
|
{
|
|
guint8 *data;
|
|
guint offset;
|
|
guint len;
|
|
|
|
g_return_val_if_fail (packet != NULL, FALSE);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_SDES, FALSE);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
|
|
|
|
/* if we are at the last item, we are done */
|
|
if (packet->item_count == packet->count)
|
|
return FALSE;
|
|
|
|
/* move to SDES */
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
data += packet->offset;
|
|
/* move to item */
|
|
offset = packet->item_offset;
|
|
/* skip SSRC */
|
|
offset += 4;
|
|
|
|
/* don't overrun */
|
|
len = (packet->length << 2);
|
|
|
|
while (offset < len) {
|
|
if (data[offset] == 0) {
|
|
/* end of list, round to next 32-bit word */
|
|
offset = (offset + 3) & ~3;
|
|
break;
|
|
}
|
|
offset += data[offset + 1] + 2;
|
|
}
|
|
if (offset >= len)
|
|
return FALSE;
|
|
|
|
packet->item_offset = offset;
|
|
packet->item_count++;
|
|
packet->entry_offset = 4;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_sdes_get_ssrc:
|
|
* @packet: a valid SDES #GstRTCPPacket
|
|
*
|
|
* Get the SSRC of the current SDES item.
|
|
*
|
|
* Returns: the SSRC of the current item.
|
|
*/
|
|
guint32
|
|
gst_rtcp_packet_sdes_get_ssrc (GstRTCPPacket * packet)
|
|
{
|
|
guint32 ssrc;
|
|
guint8 *data;
|
|
|
|
g_return_val_if_fail (packet != NULL, 0);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_SDES, 0);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), 0);
|
|
|
|
/* move to SDES */
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
data += packet->offset;
|
|
/* move to item */
|
|
data += packet->item_offset;
|
|
|
|
ssrc = GST_READ_UINT32_BE (data);
|
|
|
|
return ssrc;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_sdes_first_entry:
|
|
* @packet: a valid SDES #GstRTCPPacket
|
|
*
|
|
* Move to the first SDES entry in the current item.
|
|
*
|
|
* Returns: %TRUE if there was a first entry.
|
|
*/
|
|
gboolean
|
|
gst_rtcp_packet_sdes_first_entry (GstRTCPPacket * packet)
|
|
{
|
|
guint8 *data;
|
|
guint len, offset;
|
|
|
|
g_return_val_if_fail (packet != NULL, FALSE);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_SDES, FALSE);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
|
|
|
|
/* move to SDES */
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
data += packet->offset;
|
|
/* move to item */
|
|
offset = packet->item_offset;
|
|
/* skip SSRC */
|
|
offset += 4;
|
|
|
|
packet->entry_offset = 4;
|
|
|
|
/* don't overrun */
|
|
len = (packet->length << 2);
|
|
if (offset >= len)
|
|
return FALSE;
|
|
|
|
if (data[offset] == 0)
|
|
return FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_sdes_next_entry:
|
|
* @packet: a valid SDES #GstRTCPPacket
|
|
*
|
|
* Move to the next SDES entry in the current item.
|
|
*
|
|
* Returns: %TRUE if there was a next entry.
|
|
*/
|
|
gboolean
|
|
gst_rtcp_packet_sdes_next_entry (GstRTCPPacket * packet)
|
|
{
|
|
guint8 *data;
|
|
guint len, offset, item_len;
|
|
|
|
g_return_val_if_fail (packet != NULL, FALSE);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_SDES, FALSE);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
|
|
|
|
/* move to SDES */
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
data += packet->offset;
|
|
/* move to item */
|
|
offset = packet->item_offset;
|
|
/* move to entry */
|
|
offset += packet->entry_offset;
|
|
|
|
item_len = data[offset + 1] + 2;
|
|
/* skip item */
|
|
offset += item_len;
|
|
|
|
/* don't overrun */
|
|
len = (packet->length << 2);
|
|
if (offset >= len)
|
|
return FALSE;
|
|
|
|
packet->entry_offset += item_len;
|
|
|
|
/* check for end of list */
|
|
if (data[offset] == 0)
|
|
return FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_sdes_get_entry:
|
|
* @packet: a valid SDES #GstRTCPPacket
|
|
* @type: result of the entry type
|
|
* @len: result length of the entry data
|
|
* @data: result entry data
|
|
*
|
|
* Get the data of the current SDES item entry. @type (when not NULL) will
|
|
* contain the type of the entry. @data (when not NULL) will point to @len
|
|
* bytes.
|
|
*
|
|
* When @type refers to a text item, @data will point to a UTF8 string. Note
|
|
* that this UTF8 string is NOT null-terminated. Use
|
|
* gst_rtcp_packet_sdes_copy_entry() to get a null-termined copy of the entry.
|
|
*
|
|
* Returns: %TRUE if there was valid data.
|
|
*/
|
|
gboolean
|
|
gst_rtcp_packet_sdes_get_entry (GstRTCPPacket * packet,
|
|
GstRTCPSDESType * type, guint8 * len, guint8 ** data)
|
|
{
|
|
guint8 *bdata;
|
|
guint offset;
|
|
|
|
g_return_val_if_fail (packet != NULL, FALSE);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_SDES, FALSE);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
|
|
|
|
/* move to SDES */
|
|
bdata = GST_BUFFER_DATA (packet->buffer);
|
|
bdata += packet->offset;
|
|
/* move to item */
|
|
offset = packet->item_offset;
|
|
/* move to entry */
|
|
offset += packet->entry_offset;
|
|
|
|
if (bdata[offset] == 0)
|
|
return FALSE;
|
|
|
|
if (type)
|
|
*type = bdata[offset];
|
|
if (len)
|
|
*len = bdata[offset + 1];
|
|
if (data)
|
|
*data = &bdata[offset + 2];
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_sdes_copy_entry:
|
|
* @packet: a valid SDES #GstRTCPPacket
|
|
* @type: result of the entry type
|
|
* @len: result length of the entry data
|
|
* @data: result entry data
|
|
*
|
|
* This function is like gst_rtcp_packet_sdes_get_entry() but it returns a
|
|
* null-terminated copy of the data instead. use g_free() after usage.
|
|
*
|
|
* Returns: %TRUE if there was valid data.
|
|
*/
|
|
gboolean
|
|
gst_rtcp_packet_sdes_copy_entry (GstRTCPPacket * packet,
|
|
GstRTCPSDESType * type, guint8 * len, guint8 ** data)
|
|
{
|
|
guint8 *tdata;
|
|
guint8 tlen;
|
|
|
|
g_return_val_if_fail (packet != NULL, FALSE);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_SDES, FALSE);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
|
|
|
|
if (!gst_rtcp_packet_sdes_get_entry (packet, type, &tlen, &tdata))
|
|
return FALSE;
|
|
|
|
if (len)
|
|
*len = tlen;
|
|
if (data)
|
|
*data = (guint8 *) g_strndup ((gchar *) tdata, tlen);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_sdes_add_item:
|
|
* @packet: a valid SDES #GstRTCPPacket
|
|
* @ssrc: the SSRC of the new item to add
|
|
*
|
|
* Add a new SDES item for @ssrc to @packet.
|
|
*
|
|
* Returns: %TRUE if the item could be added, %FALSE if the maximum amount of
|
|
* items has been exceeded for the SDES packet or the MTU has been reached.
|
|
*/
|
|
gboolean
|
|
gst_rtcp_packet_sdes_add_item (GstRTCPPacket * packet, guint32 ssrc)
|
|
{
|
|
guint8 *data;
|
|
guint offset, size;
|
|
|
|
g_return_val_if_fail (packet != NULL, FALSE);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_SDES, FALSE);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
|
|
|
|
/* increment item count when possible */
|
|
if (packet->count >= GST_RTCP_MAX_SDES_ITEM_COUNT)
|
|
goto no_space;
|
|
|
|
/* pretend there is a next packet for the next call */
|
|
packet->count++;
|
|
|
|
/* jump over current item */
|
|
gst_rtcp_packet_sdes_next_item (packet);
|
|
|
|
/* move to SDES */
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
size = GST_BUFFER_SIZE (packet->buffer);
|
|
data += packet->offset;
|
|
/* move to current item */
|
|
offset = packet->item_offset;
|
|
|
|
/* we need 2 free words now */
|
|
if (offset + 8 >= size)
|
|
goto no_next;
|
|
|
|
/* write SSRC */
|
|
GST_WRITE_UINT32_BE (&data[offset], ssrc);
|
|
/* write 0 entry with padding */
|
|
GST_WRITE_UINT32_BE (&data[offset + 4], 0);
|
|
|
|
/* update count */
|
|
data[0] = (data[0] & 0xe0) | packet->count;
|
|
/* update length, we added 2 words */
|
|
packet->length += 2;
|
|
data[2] = (packet->length) >> 8;
|
|
data[3] = (packet->length) & 0xff;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_space:
|
|
{
|
|
return FALSE;
|
|
}
|
|
no_next:
|
|
{
|
|
packet->count--;
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_sdes_add_entry:
|
|
* @packet: a valid SDES #GstRTCPPacket
|
|
* @type: the #GstRTCPSDESType of the SDES entry
|
|
* @len: the data length
|
|
* @data: the data
|
|
*
|
|
* Add a new SDES entry to the current item in @packet.
|
|
*
|
|
* Returns: %TRUE if the item could be added, %FALSE if the MTU has been
|
|
* reached.
|
|
*/
|
|
gboolean
|
|
gst_rtcp_packet_sdes_add_entry (GstRTCPPacket * packet, GstRTCPSDESType type,
|
|
guint8 len, const guint8 * data)
|
|
{
|
|
guint8 *bdata;
|
|
guint offset, size, padded;
|
|
|
|
g_return_val_if_fail (packet != NULL, FALSE);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_SDES, FALSE);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
|
|
|
|
/* move to SDES */
|
|
bdata = GST_BUFFER_DATA (packet->buffer);
|
|
size = GST_BUFFER_SIZE (packet->buffer);
|
|
bdata += packet->offset;
|
|
/* move to item */
|
|
offset = packet->item_offset;
|
|
/* move to entry */
|
|
offset += packet->entry_offset;
|
|
|
|
/* add 1 byte end and up to 3 bytes padding to fill a full 32 bit word */
|
|
padded = (offset + 2 + len + 1 + 3) & ~3;
|
|
|
|
/* we need enough space for type, len, data and padding */
|
|
if (packet->offset + padded >= size)
|
|
goto no_space;
|
|
|
|
bdata[offset] = type;
|
|
bdata[offset + 1] = len;
|
|
memcpy (&bdata[offset + 2], data, len);
|
|
bdata[offset + 2 + len] = 0;
|
|
|
|
/* calculate new packet length */
|
|
packet->length = (padded - 4) >> 2;
|
|
bdata[2] = (packet->length) >> 8;
|
|
bdata[3] = (packet->length) & 0xff;
|
|
|
|
/* position to new next entry */
|
|
packet->entry_offset += 2 + len;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_space:
|
|
{
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_bye_get_ssrc_count:
|
|
* @packet: a valid BYE #GstRTCPPacket
|
|
*
|
|
* Get the number of SSRC fields in @packet.
|
|
*
|
|
* Returns: The number of SSRC fields in @packet.
|
|
*/
|
|
guint
|
|
gst_rtcp_packet_bye_get_ssrc_count (GstRTCPPacket * packet)
|
|
{
|
|
g_return_val_if_fail (packet != NULL, -1);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_BYE, -1);
|
|
|
|
return packet->count;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_bye_get_nth_ssrc:
|
|
* @packet: a valid BYE #GstRTCPPacket
|
|
* @nth: the nth SSRC to get
|
|
*
|
|
* Get the @nth SSRC of the BYE @packet.
|
|
*
|
|
* Returns: The @nth SSRC of @packet.
|
|
*/
|
|
guint32
|
|
gst_rtcp_packet_bye_get_nth_ssrc (GstRTCPPacket * packet, guint nth)
|
|
{
|
|
guint8 *data;
|
|
guint offset;
|
|
guint32 ssrc;
|
|
guint8 sc;
|
|
|
|
g_return_val_if_fail (packet != NULL, 0);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_BYE, 0);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), 0);
|
|
|
|
/* get amount of sources and check that we don't read too much */
|
|
sc = packet->count;
|
|
if (nth >= sc)
|
|
return 0;
|
|
|
|
/* get offset in 32-bits words into packet, skip the header */
|
|
offset = 1 + nth;
|
|
/* check that we don't go past the packet length */
|
|
if (offset > packet->length)
|
|
return 0;
|
|
|
|
/* scale to bytes */
|
|
offset <<= 2;
|
|
offset += packet->offset;
|
|
|
|
/* check if the packet is valid */
|
|
if (offset + 4 > GST_BUFFER_SIZE (packet->buffer))
|
|
return 0;
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
data += offset;
|
|
|
|
ssrc = GST_READ_UINT32_BE (data);
|
|
|
|
return ssrc;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_bye_add_ssrc:
|
|
* @packet: a valid BYE #GstRTCPPacket
|
|
* @ssrc: an SSRC to add
|
|
*
|
|
* Add @ssrc to the BYE @packet.
|
|
*
|
|
* Returns: %TRUE if the ssrc was added. This function can return %FALSE if
|
|
* the max MTU is exceeded or the number of sources blocks is greater than
|
|
* #GST_RTCP_MAX_BYE_SSRC_COUNT.
|
|
*/
|
|
gboolean
|
|
gst_rtcp_packet_bye_add_ssrc (GstRTCPPacket * packet, guint32 ssrc)
|
|
{
|
|
guint8 *data;
|
|
guint size, offset;
|
|
|
|
g_return_val_if_fail (packet != NULL, FALSE);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_BYE, FALSE);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
|
|
|
|
if (packet->count >= GST_RTCP_MAX_BYE_SSRC_COUNT)
|
|
goto no_space;
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
size = GST_BUFFER_SIZE (packet->buffer);
|
|
|
|
/* skip header */
|
|
offset = packet->offset + 4;
|
|
|
|
/* move to current index */
|
|
offset += (packet->count * 4);
|
|
|
|
if (offset + 4 >= size)
|
|
goto no_space;
|
|
|
|
/* increment packet count and length */
|
|
packet->count++;
|
|
data[packet->offset]++;
|
|
packet->length += 1;
|
|
data[packet->offset + 2] = (packet->length) >> 8;
|
|
data[packet->offset + 3] = (packet->length) & 0xff;
|
|
|
|
/* move to new SSRC offset and write ssrc */
|
|
data += offset;
|
|
GST_WRITE_UINT32_BE (data, ssrc);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_space:
|
|
{
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_bye_add_ssrcs:
|
|
* @packet: a valid BYE #GstRTCPPacket
|
|
* @ssrc: an array of SSRCs to add
|
|
* @len: number of elements in @ssrc
|
|
*
|
|
* Adds @len SSRCs in @ssrc to BYE @packet.
|
|
*
|
|
* Returns: %TRUE if the all the SSRCs were added. This function can return %FALSE if
|
|
* the max MTU is exceeded or the number of sources blocks is greater than
|
|
* #GST_RTCP_MAX_BYE_SSRC_COUNT.
|
|
*/
|
|
gboolean
|
|
gst_rtcp_packet_bye_add_ssrcs (GstRTCPPacket * packet, guint32 * ssrc,
|
|
guint len)
|
|
{
|
|
guint i;
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (packet != NULL, FALSE);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_BYE, FALSE);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
|
|
|
|
res = TRUE;
|
|
for (i = 0; i < len && res; i++) {
|
|
res = gst_rtcp_packet_bye_add_ssrc (packet, ssrc[i]);
|
|
}
|
|
return res;
|
|
}
|
|
|
|
/* get the offset in packet of the reason length */
|
|
static guint
|
|
get_reason_offset (GstRTCPPacket * packet)
|
|
{
|
|
guint offset;
|
|
|
|
/* get amount of sources plus header */
|
|
offset = 1 + packet->count;
|
|
|
|
/* check that we don't go past the packet length */
|
|
if (offset > packet->length)
|
|
return 0;
|
|
|
|
/* scale to bytes */
|
|
offset <<= 2;
|
|
offset += packet->offset;
|
|
|
|
/* check if the packet is valid */
|
|
if (offset + 1 > GST_BUFFER_SIZE (packet->buffer))
|
|
return 0;
|
|
|
|
return offset;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_bye_get_reason_len:
|
|
* @packet: a valid BYE #GstRTCPPacket
|
|
*
|
|
* Get the length of the reason string.
|
|
*
|
|
* Returns: The length of the reason string or 0 when there is no reason string
|
|
* present.
|
|
*/
|
|
guint8
|
|
gst_rtcp_packet_bye_get_reason_len (GstRTCPPacket * packet)
|
|
{
|
|
guint8 *data;
|
|
guint roffset;
|
|
|
|
g_return_val_if_fail (packet != NULL, 0);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_BYE, 0);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), 0);
|
|
|
|
roffset = get_reason_offset (packet);
|
|
if (roffset == 0)
|
|
return 0;
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
|
|
return data[roffset];
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_bye_get_reason:
|
|
* @packet: a valid BYE #GstRTCPPacket
|
|
*
|
|
* Get the reason in @packet.
|
|
*
|
|
* Returns: The reason for the BYE @packet or NULL if the packet did not contain
|
|
* a reason string. The string must be freed with g_free() after usage.
|
|
*/
|
|
gchar *
|
|
gst_rtcp_packet_bye_get_reason (GstRTCPPacket * packet)
|
|
{
|
|
guint8 *data;
|
|
guint roffset;
|
|
guint8 len;
|
|
|
|
g_return_val_if_fail (packet != NULL, NULL);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_BYE, NULL);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), NULL);
|
|
|
|
roffset = get_reason_offset (packet);
|
|
if (roffset == 0)
|
|
return NULL;
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
|
|
/* get length of reason string */
|
|
len = data[roffset];
|
|
if (len == 0)
|
|
return NULL;
|
|
|
|
/* move to string */
|
|
roffset += 1;
|
|
|
|
/* check if enough data to copy */
|
|
if (roffset + len > GST_BUFFER_SIZE (packet->buffer))
|
|
return NULL;
|
|
|
|
return g_strndup ((gconstpointer) (data + roffset), len);
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_bye_set_reason:
|
|
* @packet: a valid BYE #GstRTCPPacket
|
|
* @reason: a reason string
|
|
*
|
|
* Set the reason string to @reason in @packet.
|
|
*
|
|
* Returns: TRUE if the string could be set.
|
|
*/
|
|
gboolean
|
|
gst_rtcp_packet_bye_set_reason (GstRTCPPacket * packet, const gchar * reason)
|
|
{
|
|
guint8 *data;
|
|
guint roffset, size;
|
|
guint8 len, padded;
|
|
|
|
g_return_val_if_fail (packet != NULL, FALSE);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_BYE, FALSE);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
|
|
|
|
if (reason == NULL)
|
|
return TRUE;
|
|
|
|
len = strlen (reason);
|
|
if (len == 0)
|
|
return TRUE;
|
|
|
|
/* make room for the string before we get the offset */
|
|
packet->length++;
|
|
|
|
roffset = get_reason_offset (packet);
|
|
if (roffset == 0)
|
|
goto no_space;
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
size = GST_BUFFER_SIZE (packet->buffer);
|
|
|
|
/* we have 1 byte length and we need to pad to 4 bytes */
|
|
padded = ((len + 1) + 3) & ~3;
|
|
|
|
/* we need enough space for the padded length */
|
|
if (roffset + padded >= size)
|
|
goto no_space;
|
|
|
|
data[roffset] = len;
|
|
memcpy (&data[roffset + 1], reason, len);
|
|
|
|
/* update packet length, we made room for 1 double word already */
|
|
packet->length += (padded >> 2) - 1;
|
|
data[packet->offset + 2] = (packet->length) >> 8;
|
|
data[packet->offset + 3] = (packet->length) & 0xff;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_space:
|
|
{
|
|
packet->length--;
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_fb_get_sender_ssrc:
|
|
* @packet: a valid RTPFB or PSFB #GstRTCPPacket
|
|
*
|
|
* Get the sender SSRC field of the RTPFB or PSFB @packet.
|
|
*
|
|
* Returns: the sender SSRC.
|
|
*
|
|
* Since: 0.10.23
|
|
*/
|
|
guint32
|
|
gst_rtcp_packet_fb_get_sender_ssrc (GstRTCPPacket * packet)
|
|
{
|
|
guint8 *data;
|
|
guint32 ssrc;
|
|
|
|
g_return_val_if_fail (packet != NULL, 0);
|
|
g_return_val_if_fail ((packet->type == GST_RTCP_TYPE_RTPFB ||
|
|
packet->type == GST_RTCP_TYPE_PSFB), 0);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), 0);
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
|
|
/* skip header */
|
|
data += packet->offset + 4;
|
|
ssrc = GST_READ_UINT32_BE (data);
|
|
|
|
return ssrc;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_fb_set_sender_ssrc:
|
|
* @packet: a valid RTPFB or PSFB #GstRTCPPacket
|
|
* @ssrc: a sender SSRC
|
|
*
|
|
* Set the sender SSRC field of the RTPFB or PSFB @packet.
|
|
*
|
|
* Since: 0.10.23
|
|
*/
|
|
void
|
|
gst_rtcp_packet_fb_set_sender_ssrc (GstRTCPPacket * packet, guint32 ssrc)
|
|
{
|
|
guint8 *data;
|
|
|
|
g_return_if_fail (packet != NULL);
|
|
g_return_if_fail (packet->type == GST_RTCP_TYPE_RTPFB ||
|
|
packet->type == GST_RTCP_TYPE_PSFB);
|
|
g_return_if_fail (GST_IS_BUFFER (packet->buffer));
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
|
|
/* skip header */
|
|
data += packet->offset + 4;
|
|
GST_WRITE_UINT32_BE (data, ssrc);
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_fb_get_media_ssrc:
|
|
* @packet: a valid RTPFB or PSFB #GstRTCPPacket
|
|
*
|
|
* Get the media SSRC field of the RTPFB or PSFB @packet.
|
|
*
|
|
* Returns: the media SSRC.
|
|
*
|
|
* Since: 0.10.23
|
|
*/
|
|
guint32
|
|
gst_rtcp_packet_fb_get_media_ssrc (GstRTCPPacket * packet)
|
|
{
|
|
guint8 *data;
|
|
guint32 ssrc;
|
|
|
|
g_return_val_if_fail (packet != NULL, 0);
|
|
g_return_val_if_fail ((packet->type == GST_RTCP_TYPE_RTPFB ||
|
|
packet->type == GST_RTCP_TYPE_PSFB), 0);
|
|
g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), 0);
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
|
|
/* skip header and sender ssrc */
|
|
data += packet->offset + 8;
|
|
ssrc = GST_READ_UINT32_BE (data);
|
|
|
|
return ssrc;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_fb_set_media_ssrc:
|
|
* @packet: a valid RTPFB or PSFB #GstRTCPPacket
|
|
* @ssrc: a media SSRC
|
|
*
|
|
* Set the media SSRC field of the RTPFB or PSFB @packet.
|
|
*
|
|
* Since: 0.10.23
|
|
*/
|
|
void
|
|
gst_rtcp_packet_fb_set_media_ssrc (GstRTCPPacket * packet, guint32 ssrc)
|
|
{
|
|
guint8 *data;
|
|
|
|
g_return_if_fail (packet != NULL);
|
|
g_return_if_fail (packet->type == GST_RTCP_TYPE_RTPFB ||
|
|
packet->type == GST_RTCP_TYPE_PSFB);
|
|
g_return_if_fail (GST_IS_BUFFER (packet->buffer));
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
|
|
/* skip header and sender ssrc */
|
|
data += packet->offset + 8;
|
|
GST_WRITE_UINT32_BE (data, ssrc);
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_fb_get_type:
|
|
* @packet: a valid RTPFB or PSFB #GstRTCPPacket
|
|
*
|
|
* Get the feedback message type of the FB @packet.
|
|
*
|
|
* Returns: The feedback message type.
|
|
*
|
|
* Since: 0.10.23
|
|
*/
|
|
GstRTCPFBType
|
|
gst_rtcp_packet_fb_get_type (GstRTCPPacket * packet)
|
|
{
|
|
g_return_val_if_fail (packet != NULL, GST_RTCP_FB_TYPE_INVALID);
|
|
g_return_val_if_fail (packet->type == GST_RTCP_TYPE_RTPFB ||
|
|
packet->type == GST_RTCP_TYPE_PSFB, GST_RTCP_FB_TYPE_INVALID);
|
|
|
|
return packet->count;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_packet_fb_set_type:
|
|
* @packet: a valid RTPFB or PSFB #GstRTCPPacket
|
|
* @type: the #GstRTCPFBType to set
|
|
*
|
|
* Set the feedback message type of the FB @packet.
|
|
*
|
|
* Since: 0.10.23
|
|
*/
|
|
void
|
|
gst_rtcp_packet_fb_set_type (GstRTCPPacket * packet, GstRTCPFBType type)
|
|
{
|
|
guint8 *data;
|
|
|
|
g_return_if_fail (packet != NULL);
|
|
g_return_if_fail (packet->type == GST_RTCP_TYPE_RTPFB ||
|
|
packet->type == GST_RTCP_TYPE_PSFB);
|
|
g_return_if_fail (GST_IS_BUFFER (packet->buffer));
|
|
|
|
data = GST_BUFFER_DATA (packet->buffer);
|
|
|
|
data[packet->offset] = (data[packet->offset] & 0xe0) | type;
|
|
packet->count = type;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_ntp_to_unix:
|
|
* @ntptime: an NTP timestamp
|
|
*
|
|
* Converts an NTP time to UNIX nanoseconds. @ntptime can typically be
|
|
* the NTP time of an SR RTCP message and contains, in the upper 32 bits, the
|
|
* number of seconds since 1900 and, in the lower 32 bits, the fractional
|
|
* seconds. The resulting value will be the number of nanoseconds since 1970.
|
|
*
|
|
* Returns: the UNIX time for @ntptime in nanoseconds.
|
|
*/
|
|
guint64
|
|
gst_rtcp_ntp_to_unix (guint64 ntptime)
|
|
{
|
|
guint64 unixtime;
|
|
|
|
/* conversion from NTP timestamp (seconds since 1900) to seconds since
|
|
* 1970. */
|
|
unixtime = ntptime - (G_GUINT64_CONSTANT (2208988800) << 32);
|
|
/* conversion to nanoseconds */
|
|
unixtime =
|
|
gst_util_uint64_scale (unixtime, GST_SECOND,
|
|
(G_GINT64_CONSTANT (1) << 32));
|
|
|
|
return unixtime;
|
|
}
|
|
|
|
/**
|
|
* gst_rtcp_unix_to_ntp:
|
|
* @unixtime: an UNIX timestamp in nanoseconds
|
|
*
|
|
* Converts a UNIX timestamp in nanoseconds to an NTP time. The caller should
|
|
* pass a value with nanoseconds since 1970. The NTP time will, in the upper
|
|
* 32 bits, contain the number of seconds since 1900 and, in the lower 32
|
|
* bits, the fractional seconds. The resulting value can be used as an ntptime
|
|
* for constructing SR RTCP packets.
|
|
*
|
|
* Returns: the NTP time for @unixtime.
|
|
*/
|
|
guint64
|
|
gst_rtcp_unix_to_ntp (guint64 unixtime)
|
|
{
|
|
guint64 ntptime;
|
|
|
|
/* convert clock time to NTP time. upper 32 bits should contain the seconds
|
|
* and the lower 32 bits, the fractions of a second. */
|
|
ntptime =
|
|
gst_util_uint64_scale (unixtime, (G_GINT64_CONSTANT (1) << 32),
|
|
GST_SECOND);
|
|
/* conversion from UNIX timestamp (seconds since 1970) to NTP (seconds
|
|
* since 1900). */
|
|
ntptime += (G_GUINT64_CONSTANT (2208988800) << 32);
|
|
|
|
return ntptime;
|
|
}
|