--- /dev/null
+/* GStreamer
+ * Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
+ *
+ * gstrtcpbuffer.h: various helper functions to manipulate buffers
+ * with RTCP payload.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:gstrtcpbuffer
+ * @short_description: Helper methods for dealing with RTCP buffers
+ * @see_also: #GstBaseRTPPayload, #GstBaseRTPDepayload, #gstrtpbuffer
+ *
+ * Note: The API in this module is not yet declared stable.
+ *
+ * <refsect2>
+ * <para>
+ * The GstRTPCBuffer helper functions makes it easy to parse and create regular
+ * #GstBuffer objects that contain compound RTCP packets. These buffers are typically
+ * of 'application/x-rtcp' #GstCaps.
+ * </para>
+ * <para>
+ * An RTCP buffer consists of 1 or more #GstRTCPPacket structures that you can
+ * retrieve with gst_rtcp_buffer_get_first_packet(). #GstRTCPPacket acts as a pointer
+ * into the RTCP buffer; you can move to the next packet with
+ * gst_rtcp_packet_move_to_next().
+ * </para>
+ * </refsect2>
+ *
+ * Last reviewed on 2007-03-26 (0.10.13)
+ *
+ * Since: 0.10.13
+ */
+
+#include <string.h>
+
+#include "gstrtcpbuffer.h"
+
+/**
+ * gst_rtcp_buffer_new_take_data:
+ * @data: data for the new buffer
+ * @len: the length of data
+ *
+ * Create a new buffer and set the data and size of the buffer to @data and @len
+ * respectively. @data will be freed when the buffer is unreffed, so this
+ * function transfers ownership of @data to the new buffer.
+ *
+ * Returns: A newly allocated buffer with @data and of size @len.
+ */
+GstBuffer *
+gst_rtcp_buffer_new_take_data (gpointer data, guint len)
+{
+ GstBuffer *result;
+
+ g_return_val_if_fail (data != NULL, NULL);
+ g_return_val_if_fail (len > 0, NULL);
+
+ result = gst_buffer_new ();
+
+ GST_BUFFER_MALLOCDATA (result) = data;
+ GST_BUFFER_DATA (result) = data;
+ GST_BUFFER_SIZE (result) = len;
+
+ return result;
+}
+
+/**
+ * gst_rtcp_buffer_new_copy_data:
+ * @data: data for the new buffer
+ * @len: the length of data
+ *
+ * Create a new buffer and set the data to a copy of @len
+ * bytes of @data and the size to @len. The data will be freed when the buffer
+ * is freed.
+ *
+ * Returns: A newly allocated buffer with a copy of @data and of size @len.
+ */
+GstBuffer *
+gst_rtcp_buffer_new_copy_data (gpointer data, guint len)
+{
+ return gst_rtcp_buffer_new_take_data (g_memdup (data, len), len);
+}
+
+/**
+ * gst_rtcp_buffer_validate_data:
+ * @data: the data to validate
+ * @len: the length of @data to validate
+ *
+ * Check if the @data and @size point to the data of a valid RTCP (compound)
+ * packet.
+ * Use this function to validate a packet before using the other functions in
+ * this module.
+ *
+ * Returns: TRUE if the data points to a valid RTCP packet.
+ */
+gboolean
+gst_rtcp_buffer_validate_data (guint8 * data, guint len)
+{
+ guint16 header_mask;
+ guint16 header_len;
+ guint8 version;
+ guint data_len;
+ gboolean padding;
+ guint8 pad_bytes;
+
+ g_return_val_if_fail (data != NULL, FALSE);
+
+ /* we need 4 bytes for the type and length */
+ if (G_UNLIKELY (len < 4))
+ goto wrong_length;
+
+ /* first packet must be RR or SR and version must be 2 */
+ header_mask = ((data[0] << 8) | data[1]) & GST_RTCP_VALID_MASK;
+ if (G_UNLIKELY (header_mask != GST_RTCP_VALID_VALUE))
+ goto wrong_mask;
+
+ /* no padding when mask succeeds */
+ padding = FALSE;
+
+ /* store len */
+ data_len = len;
+
+ while (TRUE) {
+ /* get packet length */
+ header_len = (((data[2] << 8) | data[3]) + 1) << 2;
+ if (data_len < header_len)
+ goto wrong_length;
+
+ /* move to next compount packet */
+ data += header_len;
+ data_len -= header_len;
+
+ /* we are at the end now */
+ if (data_len < 4)
+ break;
+
+ /* check version of new packet */
+ version = data[0] & 0xc0;
+ if (version != (GST_RTCP_VERSION << 6))
+ goto wrong_version;
+
+ /* padding only allowed on last packet */
+ if ((padding = data[0] & 0x20))
+ break;
+ }
+ if (data_len > 0) {
+ /* some leftover bytes, check padding */
+ if (!padding)
+ goto wrong_length;
+
+ /* get padding */
+ pad_bytes = data[data_len - 1];
+ if (data_len != pad_bytes)
+ goto wrong_padding;
+ }
+ return TRUE;
+
+ /* ERRORS */
+wrong_length:
+ {
+ GST_DEBUG ("len check failed");
+ return FALSE;
+ }
+wrong_mask:
+ {
+ GST_DEBUG ("mask check failed (%04x != %04x)", header_mask,
+ GST_RTCP_VALID_VALUE);
+ return FALSE;
+ }
+wrong_version:
+ {
+ GST_DEBUG ("wrong version (%d < 2)", version >> 6);
+ return FALSE;
+ }
+wrong_padding:
+ {
+ GST_DEBUG ("padding check failed");
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtcp_buffer_validate:
+ * @buffer: the buffer to validate
+ *
+ * Check if the data pointed to by @buffer is a valid RTCP packet using
+ * gst_rtcp_buffer_validate_data().
+ *
+ * Returns: TRUE if @buffer is a valid RTCP packet.
+ */
+gboolean
+gst_rtcp_buffer_validate (GstBuffer * buffer)
+{
+ guint8 *data;
+ guint len;
+
+ g_return_val_if_fail (GST_IS_BUFFER (buffer), FALSE);
+
+ data = GST_BUFFER_DATA (buffer);
+ len = GST_BUFFER_SIZE (buffer);
+
+ return gst_rtcp_buffer_validate_data (data, len);
+}
+
+/**
+ * gst_rtcp_buffer_new:
+ * @mtu: the maximum mtu size.
+ *
+ * Create a new buffer for constructing RTCP packets. The packet will have a
+ * maximum size of @mtu.
+ *
+ * Returns: A newly allocated buffer.
+ */
+GstBuffer *
+gst_rtcp_buffer_new (guint mtu)
+{
+ GstBuffer *result;
+
+ g_return_val_if_fail (mtu > 0, NULL);
+
+ result = gst_buffer_new ();
+ GST_BUFFER_MALLOCDATA (result) = g_malloc0 (mtu);
+ GST_BUFFER_DATA (result) = GST_BUFFER_MALLOCDATA (result);
+ GST_BUFFER_SIZE (result) = mtu;
+
+ return result;
+}
+
+/**
+ * gst_rtcp_buffer_end:
+ * @buffer: a buffer with an RTCP packet
+ *
+ * Finish @buffer after being constructured. This function is usually called
+ * after gst_rtcp_buffer_new() and after adding the RTCP items to the new buffer.
+ *
+ * The function adjusts the size of @buffer with the total length of all the
+ * added packets.
+ */
+void
+gst_rtcp_buffer_end (GstBuffer * buffer)
+{
+ GstRTCPPacket packet;
+
+ g_return_if_fail (GST_IS_BUFFER (buffer));
+
+ /* move to the first free space */
+ if (gst_rtcp_buffer_get_first_packet (buffer, &packet))
+ while (gst_rtcp_packet_move_to_next (&packet));
+
+ /* shrink size */
+ GST_BUFFER_SIZE (buffer) = packet.offset;
+}
+
+/**
+ * gst_rtcp_buffer_get_packet_count:
+ * @buffer: a valid RTCP buffer
+ *
+ * Get the number of RTCP packets in @buffer.
+ *
+ * Returns: the number of RTCP packets in @buffer.
+ */
+guint
+gst_rtcp_buffer_get_packet_count (GstBuffer * buffer)
+{
+ GstRTCPPacket packet;
+ guint count;
+
+ g_return_val_if_fail (GST_IS_BUFFER (buffer), 0);
+
+ count = 0;
+ if (gst_rtcp_buffer_get_first_packet (buffer, &packet)) {
+ do {
+ count++;
+ } while (gst_rtcp_packet_move_to_next (&packet));
+ }
+
+ return count;
+}
+
+/**
+ * read_packet_header:
+ * @packet: a packet
+ *
+ * Read the packet headers for the packet pointed to by @packet.
+ *
+ * Returns: TRUE if @packet pointed to a valid header.
+ */
+static gboolean
+read_packet_header (GstRTCPPacket * packet)
+{
+ guint8 *data;
+ guint size;
+ guint offset;
+
+ g_return_val_if_fail (packet != NULL, FALSE);
+ g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
+
+ data = GST_BUFFER_DATA (packet->buffer);
+ size = GST_BUFFER_SIZE (packet->buffer);
+
+ offset = packet->offset;
+
+ /* check if we are at the end of the buffer, we add 4 because we also want to
+ * ensure we can read the header. */
+ if (offset + 4 > size)
+ return FALSE;
+
+ if ((data[offset] & 0xc0) != (GST_RTCP_VERSION << 6))
+ return FALSE;
+
+ /* read count, type and length */
+ packet->padding = (data[offset] & 0x20) == 0x20;
+ packet->count = data[offset] & 0x1f;
+ packet->type = data[offset + 1];
+ packet->length = (data[offset + 2] << 8) | data[offset + 3];
+ packet->item_offset = 4;
+ packet->item_count = 0;
+ packet->entry_offset = 4;
+
+ return TRUE;
+}
+
+/**
+ * gst_rtcp_buffer_get_first_packet:
+ * @buffer: a valid RTCP buffer
+ * @packet: a #GstRTCPPacket
+ *
+ * Initialize a new #GstRTCPPacket pointer that points to the first packet in
+ * @buffer.
+ *
+ * Returns: TRUE if the packet existed in @buffer.
+ */
+gboolean
+gst_rtcp_buffer_get_first_packet (GstBuffer * buffer, GstRTCPPacket * packet)
+{
+ g_return_val_if_fail (GST_IS_BUFFER (buffer), FALSE);
+ g_return_val_if_fail (packet != NULL, FALSE);
+
+ /* init to 0 */
+ packet->buffer = buffer;
+ packet->offset = 0;
+ packet->type = GST_RTCP_TYPE_INVALID;
+
+ if (!read_packet_header (packet))
+ return FALSE;
+
+ return TRUE;
+}
+
+/**
+ * gst_rtcp_packet_move_to_next:
+ * @packet: a #GstRTCPPacket
+ *
+ * Move the packet pointer @packet to the next packet in the payload.
+ * Use gst_rtcp_buffer_get_first_packet() to initialize @packet.
+ *
+ * Returns: TRUE if @packet is pointing to a valid packet after calling this
+ * function.
+ */
+gboolean
+gst_rtcp_packet_move_to_next (GstRTCPPacket * packet)
+{
+ g_return_val_if_fail (packet != NULL, FALSE);
+ g_return_val_if_fail (packet->type != GST_RTCP_TYPE_INVALID, FALSE);
+ g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
+
+ /* if we have a padding or invalid packet, it must be the last,
+ * return FALSE */
+ if (packet->type == GST_RTCP_TYPE_INVALID || packet->padding)
+ goto end;
+
+ /* move to next packet. Add 4 because the header is not included in length */
+ packet->offset += (packet->length << 2) + 4;
+
+ /* try to read new header */
+ if (!read_packet_header (packet))
+ goto end;
+
+ return TRUE;
+
+ /* ERRORS */
+end:
+ {
+ packet->type = GST_RTCP_TYPE_INVALID;
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtcp_buffer_add_packet:
+ * @buffer: a valid RTCP buffer
+ * @type: the #GstRTCPType of the new packet
+ * @packet: pointer to new packet
+ *
+ * Add a new packet of @type to @buffer. @packet will point to the newly created
+ * packet.
+ *
+ * Returns: %TRUE if the packet could be created. This function returns %FALSE
+ * if the max mtu is exceeded for the buffer.
+ */
+gboolean
+gst_rtcp_buffer_add_packet (GstBuffer * buffer, GstRTCPType type,
+ GstRTCPPacket * packet)
+{
+ guint len, size;
+ guint8 *data;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_BUFFER (buffer), FALSE);
+ g_return_val_if_fail (type != GST_RTCP_TYPE_INVALID, FALSE);
+ g_return_val_if_fail (packet != NULL, FALSE);
+
+ /* find free space */
+ if (gst_rtcp_buffer_get_first_packet (buffer, packet))
+ while (gst_rtcp_packet_move_to_next (packet));
+
+ size = GST_BUFFER_SIZE (buffer);
+
+ /* packet->offset is now pointing to the next free offset in the buffer to
+ * start a compount packet. Next we figure out if we have enough free space in
+ * the buffer to continue. */
+ switch (type) {
+ case GST_RTCP_TYPE_SR:
+ len = 28;
+ break;
+ case GST_RTCP_TYPE_RR:
+ len = 8;
+ break;
+ case GST_RTCP_TYPE_SDES:
+ len = 4;
+ break;
+ case GST_RTCP_TYPE_BYE:
+ len = 4;
+ break;
+ case GST_RTCP_TYPE_APP:
+ len = 12;
+ break;
+ case GST_RTCP_TYPE_RTPFB:
+ len = 12;
+ break;
+ case GST_RTCP_TYPE_PSFB:
+ len = 12;
+ break;
+ default:
+ goto unknown_type;
+ }
+ if (packet->offset + len >= size)
+ goto no_space;
+
+ data = GST_BUFFER_DATA (buffer) + packet->offset;
+
+ data[0] = (GST_RTCP_VERSION << 6);
+ data[1] = type;
+ /* length is stored in multiples of 32 bit words minus the length of the
+ * header */
+ len = (len - 4) >> 2;
+ data[2] = len >> 8;
+ data[3] = len & 0xff;
+
+ /* now try to position to the packet */
+ result = read_packet_header (packet);
+
+ return result;
+
+ /* ERRORS */
+unknown_type:
+ {
+ g_warning ("unknown type %d", type);
+ return FALSE;
+ }
+no_space:
+ {
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtcp_packet_remove:
+ * @packet: a #GstRTCPPacket
+ *
+ * Removes the packet pointed to by @packet and moves pointer to the next one
+ *
+ * Returns: TRUE if @packet is pointing to a valid packet after calling this
+ * function.
+ */
+gboolean
+gst_rtcp_packet_remove (GstRTCPPacket * packet)
+{
+ gboolean ret = FALSE;
+ guint offset = 0;
+
+ g_return_val_if_fail (packet != NULL, FALSE);
+ g_return_val_if_fail (packet->type != GST_RTCP_TYPE_INVALID, FALSE);
+
+ /* The next packet starts at offset + length + 4 (the header) */
+ offset = packet->offset + (packet->length << 2) + 4;
+
+ /* Overwrite this packet with the rest of the data */
+ memmove (GST_BUFFER_DATA (packet->buffer) + packet->offset,
+ GST_BUFFER_DATA (packet->buffer) + offset,
+ GST_BUFFER_SIZE (packet->buffer) - offset);
+
+ /* try to read next header */
+ ret = read_packet_header (packet);
+ if (!ret)
+ packet->type = GST_RTCP_TYPE_INVALID;
+
+ return ret;
+}
+
+/**
+ * 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 + 4) & ~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;
+}
+
+/**
+ * gst_rtcp_sdes_type_to_name:
+ * @type: a #GstRTCPSDESType
+ *
+ * Converts @type to the string equivalent. The string is typically used as a
+ * key in a #GstStructure containing SDES items.
+ *
+ * Returns: the string equivalent of @type
+ *
+ * Since: 0.10.26
+ */
+const gchar *
+gst_rtcp_sdes_type_to_name (GstRTCPSDESType type)
+{
+ const gchar *result;
+
+ switch (type) {
+ case GST_RTCP_SDES_CNAME:
+ result = "cname";
+ break;
+ case GST_RTCP_SDES_NAME:
+ result = "name";
+ break;
+ case GST_RTCP_SDES_EMAIL:
+ result = "email";
+ break;
+ case GST_RTCP_SDES_PHONE:
+ result = "phone";
+ break;
+ case GST_RTCP_SDES_LOC:
+ result = "location";
+ break;
+ case GST_RTCP_SDES_TOOL:
+ result = "tool";
+ break;
+ case GST_RTCP_SDES_NOTE:
+ result = "note";
+ break;
+ case GST_RTCP_SDES_PRIV:
+ result = "priv";
+ break;
+ default:
+ result = NULL;
+ break;
+ }
+ return result;
+}
+
+/**
+ * gst_rtcp_sdes_name_to_type:
+ * @name: a SDES name
+ *
+ * Convert @name into a @GstRTCPSDESType. @name is typically a key in a
+ * #GstStructure containing SDES items.
+ *
+ * Returns: the #GstRTCPSDESType for @name or #GST_RTCP_SDES_PRIV when @name
+ * is a private sdes item.
+ *
+ * Since: 0.10.26
+ */
+GstRTCPSDESType
+gst_rtcp_sdes_name_to_type (const gchar * name)
+{
+ if (name == NULL || strlen (name) == 0)
+ return GST_RTCP_SDES_INVALID;
+
+ if (strcmp ("cname", name) == 0)
+ return GST_RTCP_SDES_CNAME;
+
+ if (strcmp ("name", name) == 0)
+ return GST_RTCP_SDES_NAME;
+
+ if (strcmp ("email", name) == 0)
+ return GST_RTCP_SDES_EMAIL;
+
+ if (strcmp ("phone", name) == 0)
+ return GST_RTCP_SDES_PHONE;
+
+ if (strcmp ("location", name) == 0)
+ return GST_RTCP_SDES_LOC;
+
+ if (strcmp ("tool", name) == 0)
+ return GST_RTCP_SDES_TOOL;
+
+ if (strcmp ("note", name) == 0)
+ return GST_RTCP_SDES_NOTE;
+
+ return GST_RTCP_SDES_PRIV;
+}
+
+/**
+ * gst_rtcp_packet_fb_get_fci_length:
+ * @packet: a valid RTPFB or PSFB #GstRTCPPacket
+ *
+ * Get the length of the Feedback Control Information attached to a
+ * RTPFB or PSFB @packet.
+ *
+ * Returns: The length of the FCI in 32-bit words.
+ *
+ * Since: 0.10.31
+ */
+guint16
+gst_rtcp_packet_fb_get_fci_length (GstRTCPPacket * packet)
+{
+ guint8 *data;
+
+ 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) + packet->offset + 2;
+
+ return GST_READ_UINT16_BE (data) - 2;
+}
+
+/**
+ * gst_rtcp_packet_fb_set_fci_length:
+ * @packet: a valid RTPFB or PSFB #GstRTCPPacket
+ * @wordlen: Length of the FCI in 32-bit words
+ *
+ * Set the length of the Feedback Control Information attached to a
+ * RTPFB or PSFB @packet.
+ *
+ * Returns: %TRUE if there was enough space in the packet to add this much FCI
+ *
+ * Since: 0.10.31
+ */
+gboolean
+gst_rtcp_packet_fb_set_fci_length (GstRTCPPacket * packet, guint16 wordlen)
+{
+ guint8 *data;
+
+ g_return_val_if_fail (packet != NULL, FALSE);
+ g_return_val_if_fail (packet->type == GST_RTCP_TYPE_RTPFB ||
+ packet->type == GST_RTCP_TYPE_PSFB, FALSE);
+ g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), FALSE);
+
+ if (GST_BUFFER_SIZE (packet->buffer) < packet->offset + ((wordlen + 3) * 4))
+ return FALSE;
+
+ data = GST_BUFFER_DATA (packet->buffer) + packet->offset + 2;
+ wordlen += 2;
+ GST_WRITE_UINT16_BE (data, wordlen);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtcp_packet_fb_get_fci:
+ * @packet: a valid RTPFB or PSFB #GstRTCPPacket
+ *
+ * Get the Feedback Control Information attached to a RTPFB or PSFB @packet.
+ *
+ * Returns: a pointer to the FCI
+ *
+ * Since: 0.10.31
+ */
+guint8 *
+gst_rtcp_packet_fb_get_fci (GstRTCPPacket * packet)
+{
+ guint8 *data;
+
+ g_return_val_if_fail (packet != NULL, NULL);
+ g_return_val_if_fail (packet->type == GST_RTCP_TYPE_RTPFB ||
+ packet->type == GST_RTCP_TYPE_PSFB, NULL);
+ g_return_val_if_fail (GST_IS_BUFFER (packet->buffer), NULL);
+
+ data = GST_BUFFER_DATA (packet->buffer) + packet->offset;
+
+ if (GST_READ_UINT16_BE (data + 2) <= 2)
+ return NULL;
+
+ return data + 12;
+}
projects / mafwsubrenderer / blobdiff