gstreamer/gst-libs/gst/rtp/gstrtpbuffer.c
Wim Taymans 9ce042e2a7 Avoid overflows in the padding checks by doing the check slightly
differently.
Add a unit test to check for correct behaviour.
2009-01-21 13:09:29 +01:00

1029 lines
27 KiB
C

/* GStreamer
* Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
* Copyright (C) <2006> Wim Taymans <wim@fluendo.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:gstrtpbuffer
* @short_description: Helper methods for dealing with RTP buffers
* @see_also: #GstBaseRTPPayload, #GstBaseRTPDepayload, gstrtcpbuffer
*
* <refsect2>
* <para>
* The GstRTPBuffer helper functions makes it easy to parse and create regular
* #GstBuffer objects that contain RTP payloads. These buffers are typically of
* 'application/x-rtp' #GstCaps.
* </para>
* </refsect2>
*
* Last reviewed on 2006-07-17 (0.10.10)
*/
#include "gstrtpbuffer.h"
#include <stdlib.h>
#include <string.h>
#define GST_RTP_HEADER_LEN 12
/* Note: we use bitfields here to make sure the compiler doesn't add padding
* between fields on certain architectures; can't assume aligned access either
*/
typedef struct _GstRTPHeader
{
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
unsigned int csrc_count:4; /* CSRC count */
unsigned int extension:1; /* header extension flag */
unsigned int padding:1; /* padding flag */
unsigned int version:2; /* protocol version */
unsigned int payload_type:7; /* payload type */
unsigned int marker:1; /* marker bit */
#elif G_BYTE_ORDER == G_BIG_ENDIAN
unsigned int version:2; /* protocol version */
unsigned int padding:1; /* padding flag */
unsigned int extension:1; /* header extension flag */
unsigned int csrc_count:4; /* CSRC count */
unsigned int marker:1; /* marker bit */
unsigned int payload_type:7; /* payload type */
#else
#error "G_BYTE_ORDER should be big or little endian."
#endif
unsigned int seq:16; /* sequence number */
unsigned int timestamp:32; /* timestamp */
unsigned int ssrc:32; /* synchronization source */
guint8 csrclist[4]; /* optional CSRC list, 32 bits each */
} GstRTPHeader;
#define GST_RTP_HEADER_VERSION(buf) (((GstRTPHeader *)(GST_BUFFER_DATA (buf)))->version)
#define GST_RTP_HEADER_PADDING(buf) (((GstRTPHeader *)(GST_BUFFER_DATA (buf)))->padding)
#define GST_RTP_HEADER_EXTENSION(buf) (((GstRTPHeader *)(GST_BUFFER_DATA (buf)))->extension)
#define GST_RTP_HEADER_CSRC_COUNT(buf) (((GstRTPHeader *)(GST_BUFFER_DATA (buf)))->csrc_count)
#define GST_RTP_HEADER_MARKER(buf) (((GstRTPHeader *)(GST_BUFFER_DATA (buf)))->marker)
#define GST_RTP_HEADER_PAYLOAD_TYPE(buf)(((GstRTPHeader *)(GST_BUFFER_DATA (buf)))->payload_type)
#define GST_RTP_HEADER_SEQ(buf) (((GstRTPHeader *)(GST_BUFFER_DATA (buf)))->seq)
#define GST_RTP_HEADER_TIMESTAMP(buf) (((GstRTPHeader *)(GST_BUFFER_DATA (buf)))->timestamp)
#define GST_RTP_HEADER_SSRC(buf) (((GstRTPHeader *)(GST_BUFFER_DATA (buf)))->ssrc)
#define GST_RTP_HEADER_CSRC_LIST_OFFSET(buf,i) \
GST_BUFFER_DATA (buf) + \
G_STRUCT_OFFSET(GstRTPHeader, csrclist) + \
((i) * sizeof(guint32))
#define GST_RTP_HEADER_CSRC_SIZE(buf) (GST_RTP_HEADER_CSRC_COUNT(buf) * sizeof (guint32))
/**
* gst_rtp_buffer_allocate_data:
* @buffer: a #GstBuffer
* @payload_len: the length of the payload
* @pad_len: the amount of padding
* @csrc_count: the number of CSRC entries
*
* Allocate enough data in @buffer to hold an RTP packet with @csrc_count CSRCs,
* a payload length of @payload_len and padding of @pad_len.
* MALLOCDATA of @buffer will be overwritten and will not be freed.
* All other RTP header fields will be set to 0/FALSE.
*/
void
gst_rtp_buffer_allocate_data (GstBuffer * buffer, guint payload_len,
guint8 pad_len, guint8 csrc_count)
{
guint len;
g_return_if_fail (csrc_count <= 15);
g_return_if_fail (GST_IS_BUFFER (buffer));
len = GST_RTP_HEADER_LEN + csrc_count * sizeof (guint32)
+ payload_len + pad_len;
GST_BUFFER_MALLOCDATA (buffer) = g_malloc (len);
GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
GST_BUFFER_SIZE (buffer) = len;
/* fill in defaults */
GST_RTP_HEADER_VERSION (buffer) = GST_RTP_VERSION;
GST_RTP_HEADER_PADDING (buffer) = FALSE;
GST_RTP_HEADER_EXTENSION (buffer) = FALSE;
GST_RTP_HEADER_CSRC_COUNT (buffer) = csrc_count;
memset (GST_RTP_HEADER_CSRC_LIST_OFFSET (buffer, 0), 0,
csrc_count * sizeof (guint32));
GST_RTP_HEADER_MARKER (buffer) = FALSE;
GST_RTP_HEADER_PAYLOAD_TYPE (buffer) = 0;
GST_RTP_HEADER_SEQ (buffer) = 0;
GST_RTP_HEADER_TIMESTAMP (buffer) = 0;
GST_RTP_HEADER_SSRC (buffer) = 0;
}
/**
* gst_rtp_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_rtp_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_rtp_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_rtp_buffer_new_copy_data (gpointer data, guint len)
{
return gst_rtp_buffer_new_take_data (g_memdup (data, len), len);
}
/**
* gst_rtp_buffer_new_allocate:
* @payload_len: the length of the payload
* @pad_len: the amount of padding
* @csrc_count: the number of CSRC entries
*
* Allocate a new #Gstbuffer with enough data to hold an RTP packet with @csrc_count CSRCs,
* a payload length of @payload_len and padding of @pad_len.
* All other RTP header fields will be set to 0/FALSE.
*
* Returns: A newly allocated buffer that can hold an RTP packet with given
* parameters.
*/
GstBuffer *
gst_rtp_buffer_new_allocate (guint payload_len, guint8 pad_len,
guint8 csrc_count)
{
GstBuffer *result;
g_return_val_if_fail (csrc_count <= 15, NULL);
result = gst_buffer_new ();
gst_rtp_buffer_allocate_data (result, payload_len, pad_len, csrc_count);
return result;
}
/**
* gst_rtp_buffer_new_allocate_len:
* @packet_len: the total length of the packet
* @pad_len: the amount of padding
* @csrc_count: the number of CSRC entries
*
* Create a new #GstBuffer that can hold an RTP packet that is exactly
* @packet_len long. The length of the payload depends on @pad_len and
* @csrc_count and can be calculated with gst_rtp_buffer_calc_payload_len().
* All RTP header fields will be set to 0/FALSE.
*
* Returns: A newly allocated buffer that can hold an RTP packet of @packet_len.
*/
GstBuffer *
gst_rtp_buffer_new_allocate_len (guint packet_len, guint8 pad_len,
guint8 csrc_count)
{
guint len;
g_return_val_if_fail (csrc_count <= 15, NULL);
len = gst_rtp_buffer_calc_payload_len (packet_len, pad_len, csrc_count);
return gst_rtp_buffer_new_allocate (len, pad_len, csrc_count);
}
/**
* gst_rtp_buffer_calc_header_len:
* @csrc_count: the number of CSRC entries
*
* Calculate the header length of an RTP packet with @csrc_count CSRC entries.
* An RTP packet can have at most 15 CSRC entries.
*
* Returns: The length of an RTP header with @csrc_count CSRC entries.
*/
guint
gst_rtp_buffer_calc_header_len (guint8 csrc_count)
{
g_return_val_if_fail (csrc_count <= 15, 0);
return GST_RTP_HEADER_LEN + (csrc_count * sizeof (guint32));
}
/**
* gst_rtp_buffer_calc_packet_len:
* @payload_len: the length of the payload
* @pad_len: the amount of padding
* @csrc_count: the number of CSRC entries
*
* Calculate the total length of an RTP packet with a payload size of @payload_len,
* a padding of @pad_len and a @csrc_count CSRC entries.
*
* Returns: The total length of an RTP header with given parameters.
*/
guint
gst_rtp_buffer_calc_packet_len (guint payload_len, guint8 pad_len,
guint8 csrc_count)
{
g_return_val_if_fail (csrc_count <= 15, 0);
return payload_len + GST_RTP_HEADER_LEN + (csrc_count * sizeof (guint32))
+ pad_len;
}
/**
* gst_rtp_buffer_calc_payload_len:
* @packet_len: the length of the total RTP packet
* @pad_len: the amount of padding
* @csrc_count: the number of CSRC entries
*
* Calculate the length of the payload of an RTP packet with size @packet_len,
* a padding of @pad_len and a @csrc_count CSRC entries.
*
* Returns: The length of the payload of an RTP packet with given parameters.
*/
guint
gst_rtp_buffer_calc_payload_len (guint packet_len, guint8 pad_len,
guint8 csrc_count)
{
g_return_val_if_fail (csrc_count <= 15, 0);
return packet_len - GST_RTP_HEADER_LEN - (csrc_count * sizeof (guint32))
- pad_len;
}
/**
* gst_rtp_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 RTP packet.
* This function checks the length, version and padding of the packet data.
* Use this function to validate a packet before using the other functions in
* this module.
*
* Returns: TRUE if the data points to a valid RTP packet.
*/
gboolean
gst_rtp_buffer_validate_data (guint8 * data, guint len)
{
guint8 padding;
guint8 csrc_count;
guint header_len;
guint8 version;
g_return_val_if_fail (data != NULL, FALSE);
header_len = GST_RTP_HEADER_LEN;
if (G_UNLIKELY (len < header_len))
goto wrong_length;
/* check version */
version = (data[0] & 0xc0);
if (G_UNLIKELY (version != (GST_RTP_VERSION << 6)))
goto wrong_version;
/* calc header length with csrc */
csrc_count = (data[0] & 0x0f);
header_len += csrc_count * sizeof (guint32);
/* calc extension length when present. */
if (data[0] & 0x10) {
guint8 *extpos;
guint16 extlen;
/* this points to the extenstion bits and header length */
extpos = &data[header_len];
/* skip the header and check that we have enough space */
header_len += 4;
if (G_UNLIKELY (len < header_len))
goto wrong_length;
/* skip id */
extpos += 2;
/* read length as the number of 32 bits words */
extlen = GST_READ_UINT16_BE (extpos);
header_len += extlen * sizeof (guint32);
}
/* check for padding */
if (data[0] & 0x20)
padding = data[len - 1];
else
padding = 0;
/* check if padding and header not bigger than packet length */
if (G_UNLIKELY (len < padding + header_len))
goto wrong_padding;
return TRUE;
/* ERRORS */
wrong_length:
{
GST_DEBUG ("len < header_len check failed (%d < %d)", len, header_len);
return FALSE;
}
wrong_version:
{
GST_DEBUG ("version check failed (%d != %d)", version, GST_RTP_VERSION);
return FALSE;
}
wrong_padding:
{
GST_DEBUG ("padding check failed (%d - %d < %d)", len, header_len, padding);
return FALSE;
}
}
/**
* gst_rtp_buffer_validate:
* @buffer: the buffer to validate
*
* Check if the data pointed to by @buffer is a valid RTP packet using
* gst_rtp_buffer_validate_data().
* Use this function to validate a packet before using the other functions in
* this module.
*
* Returns: TRUE if @buffer is a valid RTP packet.
*/
gboolean
gst_rtp_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_rtp_buffer_validate_data (data, len);
}
/**
* gst_rtp_buffer_set_packet_len:
* @buffer: the buffer
* @len: the new packet length
*
* Set the total @buffer size to @len. The data in the buffer will be made
* larger if needed. Any padding will be removed from the packet.
*/
void
gst_rtp_buffer_set_packet_len (GstBuffer * buffer, guint len)
{
guint oldlen;
oldlen = GST_BUFFER_SIZE (buffer);
if (oldlen < len) {
guint8 *newdata;
newdata = g_realloc (GST_BUFFER_MALLOCDATA (buffer), len);
GST_BUFFER_MALLOCDATA (buffer) = newdata;
GST_BUFFER_DATA (buffer) = newdata;
}
GST_BUFFER_SIZE (buffer) = len;
/* remove any padding */
GST_RTP_HEADER_PADDING (buffer) = FALSE;
}
/**
* gst_rtp_buffer_get_packet_len:
* @buffer: the buffer
*
* Return the total length of the packet in @buffer.
*
* Returns: The total length of the packet in @buffer.
*/
guint
gst_rtp_buffer_get_packet_len (GstBuffer * buffer)
{
return GST_BUFFER_SIZE (buffer);
}
/**
* gst_rtp_buffer_get_header_len:
* @buffer: the buffer
*
* Return the total length of the header in @buffer. This include the length of
* the fixed header, the CSRC list and the extension header.
*
* Returns: The total length of the header in @buffer.
*/
guint
gst_rtp_buffer_get_header_len (GstBuffer * buffer)
{
guint len;
len = GST_RTP_HEADER_LEN + GST_RTP_HEADER_CSRC_SIZE (buffer);
if (GST_RTP_HEADER_EXTENSION (buffer))
len += GST_READ_UINT16_BE (GST_BUFFER_DATA (buffer) + len + 2) * 4 + 4;
return len;
}
/**
* gst_rtp_buffer_get_version:
* @buffer: the buffer
*
* Get the version number of the RTP packet in @buffer.
*
* Returns: The version of @buffer.
*/
guint8
gst_rtp_buffer_get_version (GstBuffer * buffer)
{
return GST_RTP_HEADER_VERSION (buffer);
}
/**
* gst_rtp_buffer_set_version:
* @buffer: the buffer
* @version: the new version
*
* Set the version of the RTP packet in @buffer to @version.
*/
void
gst_rtp_buffer_set_version (GstBuffer * buffer, guint8 version)
{
g_return_if_fail (version < 0x04);
GST_RTP_HEADER_VERSION (buffer) = version;
}
/**
* gst_rtp_buffer_get_padding:
* @buffer: the buffer
*
* Check if the padding bit is set on the RTP packet in @buffer.
*
* Returns: TRUE if @buffer has the padding bit set.
*/
gboolean
gst_rtp_buffer_get_padding (GstBuffer * buffer)
{
return GST_RTP_HEADER_PADDING (buffer);
}
/**
* gst_rtp_buffer_set_padding:
* @buffer: the buffer
* @padding: the new padding
*
* Set the padding bit on the RTP packet in @buffer to @padding.
*/
void
gst_rtp_buffer_set_padding (GstBuffer * buffer, gboolean padding)
{
GST_RTP_HEADER_PADDING (buffer) = padding;
}
/**
* gst_rtp_buffer_pad_to:
* @buffer: the buffer
* @len: the new amount of padding
*
* Set the amount of padding in the RTP packet in @buffer to
* @len. If @len is 0, the padding is removed.
*
* NOTE: This function does not work correctly.
*/
void
gst_rtp_buffer_pad_to (GstBuffer * buffer, guint len)
{
if (len > 0)
GST_RTP_HEADER_PADDING (buffer) = TRUE;
else
GST_RTP_HEADER_PADDING (buffer) = FALSE;
/* FIXME, set the padding byte at the end of the payload data */
}
/**
* gst_rtp_buffer_get_extension:
* @buffer: the buffer
*
* Check if the extension bit is set on the RTP packet in @buffer.
*
* Returns: TRUE if @buffer has the extension bit set.
*/
gboolean
gst_rtp_buffer_get_extension (GstBuffer * buffer)
{
return GST_RTP_HEADER_EXTENSION (buffer);
}
/**
* gst_rtp_buffer_set_extension:
* @buffer: the buffer
* @extension: the new extension
*
* Set the extension bit on the RTP packet in @buffer to @extension.
*/
void
gst_rtp_buffer_set_extension (GstBuffer * buffer, gboolean extension)
{
GST_RTP_HEADER_EXTENSION (buffer) = extension;
}
/**
* gst_rtp_buffer_get_extension_data:
* @buffer: the buffer
* @bits: location for result bits
* @data: location for data
* @wordlen: location for length of @data in 32 bits words
*
* Get the extension data. @bits will contain the extension 16 bits of custom
* data. @data will point to the data in the extension and @wordlen will contain
* the length of @data in 32 bits words.
*
* If @buffer did not contain an extension, this function will return %FALSE
* with @bits, @data and @wordlen unchanged.
*
* Returns: TRUE if @buffer had the extension bit set.
*
* Since: 0.10.15
*/
gboolean
gst_rtp_buffer_get_extension_data (GstBuffer * buffer, guint16 * bits,
gpointer * data, guint * wordlen)
{
guint len;
guint8 *pdata;
if (!GST_RTP_HEADER_EXTENSION (buffer))
return FALSE;
/* move to the extension */
len = GST_RTP_HEADER_LEN + GST_RTP_HEADER_CSRC_SIZE (buffer);
pdata = GST_BUFFER_DATA (buffer) + len;
if (bits)
*bits = GST_READ_UINT16_BE (pdata);
if (wordlen)
*wordlen = GST_READ_UINT16_BE (pdata + 2);
if (data)
*data = pdata + 4;
return TRUE;
}
/**
* gst_rtp_buffer_set_extension_data:
* @buffer: the buffer
* @bits: the bits specific for the extension
* @length: the length that counts the number of 32-bit words in
* the extension, excluding the extension header ( therefore zero is a valid length)
*
* Set the extension bit of the rtp buffer and fill in the @bits and @length of the
* extension header. It will refuse to set the extension data if the buffer is not
* large enough.
*
* Returns: True if done.
*
* Since : 0.10.18
*/
gboolean
gst_rtp_buffer_set_extension_data (GstBuffer * buffer, guint16 bits,
guint16 length)
{
guint32 min_size = 0;
guint8 *data;
/* check if the buffer is big enough to hold the extension */
min_size =
GST_RTP_HEADER_LEN + GST_RTP_HEADER_CSRC_SIZE (buffer) + 4 +
length * sizeof (guint32);
if (G_UNLIKELY (min_size > GST_BUFFER_SIZE (buffer)))
goto too_small;
/* now we can set the extension bit */
gst_rtp_buffer_set_extension (buffer, TRUE);
data = GST_BUFFER_DATA (buffer) + GST_RTP_HEADER_LEN +
GST_RTP_HEADER_CSRC_SIZE (buffer);
GST_WRITE_UINT16_BE (data, bits);
GST_WRITE_UINT16_BE (data + 2, length);
return TRUE;
/* ERRORS */
too_small:
{
g_warning
("rtp buffer too small: need more than %d bytes but only have %d bytes",
min_size, GST_BUFFER_SIZE (buffer));
return FALSE;
}
}
/**
* gst_rtp_buffer_get_ssrc:
* @buffer: the buffer
*
* Get the SSRC of the RTP packet in @buffer.
*
* Returns: the SSRC of @buffer in host order.
*/
guint32
gst_rtp_buffer_get_ssrc (GstBuffer * buffer)
{
return g_ntohl (GST_RTP_HEADER_SSRC (buffer));
}
/**
* gst_rtp_buffer_set_ssrc:
* @buffer: the buffer
* @ssrc: the new SSRC
*
* Set the SSRC on the RTP packet in @buffer to @ssrc.
*/
void
gst_rtp_buffer_set_ssrc (GstBuffer * buffer, guint32 ssrc)
{
GST_RTP_HEADER_SSRC (buffer) = g_htonl (ssrc);
}
/**
* gst_rtp_buffer_get_csrc_count:
* @buffer: the buffer
*
* Get the CSRC count of the RTP packet in @buffer.
*
* Returns: the CSRC count of @buffer.
*/
guint8
gst_rtp_buffer_get_csrc_count (GstBuffer * buffer)
{
return GST_RTP_HEADER_CSRC_COUNT (buffer);
}
/**
* gst_rtp_buffer_get_csrc:
* @buffer: the buffer
* @idx: the index of the CSRC to get
*
* Get the CSRC at index @idx in @buffer.
*
* Returns: the CSRC at index @idx in host order.
*/
guint32
gst_rtp_buffer_get_csrc (GstBuffer * buffer, guint8 idx)
{
g_return_val_if_fail (idx < GST_RTP_HEADER_CSRC_COUNT (buffer), 0);
return GST_READ_UINT32_BE (GST_RTP_HEADER_CSRC_LIST_OFFSET (buffer, idx));
}
/**
* gst_rtp_buffer_set_csrc:
* @buffer: the buffer
* @idx: the CSRC index to set
* @csrc: the CSRC in host order to set at @idx
*
* Modify the CSRC at index @idx in @buffer to @csrc.
*/
void
gst_rtp_buffer_set_csrc (GstBuffer * buffer, guint8 idx, guint32 csrc)
{
g_return_if_fail (idx < GST_RTP_HEADER_CSRC_COUNT (buffer));
GST_WRITE_UINT32_BE (GST_RTP_HEADER_CSRC_LIST_OFFSET (buffer, idx), csrc);
}
/**
* gst_rtp_buffer_get_marker:
* @buffer: the buffer
*
* Check if the marker bit is set on the RTP packet in @buffer.
*
* Returns: TRUE if @buffer has the marker bit set.
*/
gboolean
gst_rtp_buffer_get_marker (GstBuffer * buffer)
{
return GST_RTP_HEADER_MARKER (buffer);
}
/**
* gst_rtp_buffer_set_marker:
* @buffer: the buffer
* @marker: the new marker
*
* Set the marker bit on the RTP packet in @buffer to @marker.
*/
void
gst_rtp_buffer_set_marker (GstBuffer * buffer, gboolean marker)
{
GST_RTP_HEADER_MARKER (buffer) = marker;
}
/**
* gst_rtp_buffer_get_payload_type:
* @buffer: the buffer
*
* Get the payload type of the RTP packet in @buffer.
*
* Returns: The payload type.
*/
guint8
gst_rtp_buffer_get_payload_type (GstBuffer * buffer)
{
return GST_RTP_HEADER_PAYLOAD_TYPE (buffer);
}
/**
* gst_rtp_buffer_set_payload_type:
* @buffer: the buffer
* @payload_type: the new type
*
* Set the payload type of the RTP packet in @buffer to @payload_type.
*/
void
gst_rtp_buffer_set_payload_type (GstBuffer * buffer, guint8 payload_type)
{
g_return_if_fail (payload_type < 0x80);
GST_RTP_HEADER_PAYLOAD_TYPE (buffer) = payload_type;
}
/**
* gst_rtp_buffer_get_seq:
* @buffer: the buffer
*
* Get the sequence number of the RTP packet in @buffer.
*
* Returns: The sequence number in host order.
*/
guint16
gst_rtp_buffer_get_seq (GstBuffer * buffer)
{
return g_ntohs (GST_RTP_HEADER_SEQ (buffer));
}
/**
* gst_rtp_buffer_set_seq:
* @buffer: the buffer
* @seq: the new sequence number
*
* Set the sequence number of the RTP packet in @buffer to @seq.
*/
void
gst_rtp_buffer_set_seq (GstBuffer * buffer, guint16 seq)
{
GST_RTP_HEADER_SEQ (buffer) = g_htons (seq);
}
/**
* gst_rtp_buffer_get_timestamp:
* @buffer: the buffer
*
* Get the timestamp of the RTP packet in @buffer.
*
* Returns: The timestamp in host order.
*/
guint32
gst_rtp_buffer_get_timestamp (GstBuffer * buffer)
{
return g_ntohl (GST_RTP_HEADER_TIMESTAMP (buffer));
}
/**
* gst_rtp_buffer_set_timestamp:
* @buffer: the buffer
* @timestamp: the new timestamp
*
* Set the timestamp of the RTP packet in @buffer to @timestamp.
*/
void
gst_rtp_buffer_set_timestamp (GstBuffer * buffer, guint32 timestamp)
{
GST_RTP_HEADER_TIMESTAMP (buffer) = g_htonl (timestamp);
}
/**
* gst_rtp_buffer_get_payload_subbuffer:
* @buffer: the buffer
* @offset: the offset in the payload
* @len: the length in the payload
*
* Create a subbuffer of the payload of the RTP packet in @buffer. @offset bytes
* are skipped in the payload and the subbuffer will be of size @len.
* If @len is -1 the total payload starting from @offset if subbuffered.
*
* Returns: A new buffer with the specified data of the payload.
*
* Since: 0.10.10
*/
GstBuffer *
gst_rtp_buffer_get_payload_subbuffer (GstBuffer * buffer, guint offset,
guint len)
{
guint poffset, plen;
plen = gst_rtp_buffer_get_payload_len (buffer);
/* we can't go past the length */
if (G_UNLIKELY (offset >= plen))
goto wrong_offset;
/* apply offset */
poffset = gst_rtp_buffer_get_header_len (buffer) + offset;
plen -= offset;
/* see if we need to shrink the buffer based on @len */
if (len != -1 && len < plen)
plen = len;
return gst_buffer_create_sub (buffer, poffset, plen);
/* ERRORS */
wrong_offset:
{
g_warning ("offset=%u should be less then plen=%u", offset, plen);
return NULL;
}
}
/**
* gst_rtp_buffer_get_payload_buffer:
* @buffer: the buffer
*
* Create a buffer of the payload of the RTP packet in @buffer. This function
* will internally create a subbuffer of @buffer so that a memcpy can be
* avoided.
*
* Returns: A new buffer with the data of the payload.
*/
GstBuffer *
gst_rtp_buffer_get_payload_buffer (GstBuffer * buffer)
{
return gst_rtp_buffer_get_payload_subbuffer (buffer, 0, -1);
}
/**
* gst_rtp_buffer_get_payload_len:
* @buffer: the buffer
*
* Get the length of the payload of the RTP packet in @buffer.
*
* Returns: The length of the payload in @buffer.
*/
guint
gst_rtp_buffer_get_payload_len (GstBuffer * buffer)
{
guint len, size;
size = GST_BUFFER_SIZE (buffer);
len = size - gst_rtp_buffer_get_header_len (buffer);
if (GST_RTP_HEADER_PADDING (buffer))
len -= GST_BUFFER_DATA (buffer)[size - 1];
return len;
}
/**
* gst_rtp_buffer_get_payload:
* @buffer: the buffer
*
* Get a pointer to the payload data in @buffer. This pointer is valid as long
* as a reference to @buffer is held.
*
* Returns: A pointer to the payload data in @buffer.
*/
gpointer
gst_rtp_buffer_get_payload (GstBuffer * buffer)
{
return GST_BUFFER_DATA (buffer) + gst_rtp_buffer_get_header_len (buffer);
}
/**
* gst_rtp_buffer_default_clock_rate:
* @payload_type: the static payload type
*
* Get the default clock-rate for the static payload type @payload_type.
*
* Returns: the default clock rate or -1 if the payload type is not static or
* the clock-rate is undefined.
*
* Since: 0.10.13
*/
guint32
gst_rtp_buffer_default_clock_rate (guint8 payload_type)
{
const GstRTPPayloadInfo *info;
guint32 res;
info = gst_rtp_payload_info_for_pt (payload_type);
if (!info)
return -1;
res = info->clock_rate;
/* 0 means unknown so we have to return -1 from this function */
if (res == 0)
res = -1;
return res;
}
/**
* gst_rtp_buffer_compare_seqnum:
* @seqnum1: a sequence number
* @seqnum2: a sequence number
*
* Compare two sequence numbers, taking care of wraparounds. This function
* returns the difference between @seqnum1 and @seqnum2.
*
* Returns: a negative value if @seqnum1 is bigger than @seqnum2, 0 if they
* are equal or a positive value if @seqnum1 is smaller than @segnum2.
*
* Since: 0.10.15
*/
gint
gst_rtp_buffer_compare_seqnum (guint16 seqnum1, guint16 seqnum2)
{
return (gint16) (seqnum2 - seqnum1);
}
/**
* gst_rtp_buffer_ext_timestamp:
* @exttimestamp: a previous extended timestamp
* @timestamp: a new timestamp
*
* Update the @exttimestamp field with @timestamp. For the first call of the
* method, @exttimestamp should point to a location with a value of -1.
*
* This function makes sure that the returned value is a constantly increasing
* value even in the case where there is a timestamp wraparound.
*
* Returns: The extended timestamp of @timestamp.
*
* Since: 0.10.15
*/
guint64
gst_rtp_buffer_ext_timestamp (guint64 * exttimestamp, guint32 timestamp)
{
guint64 result, diff, ext;
g_return_val_if_fail (exttimestamp != NULL, -1);
ext = *exttimestamp;
if (ext == -1) {
result = timestamp;
} else {
/* pick wraparound counter from previous timestamp and add to new timestamp */
result = timestamp + (ext & ~(G_GINT64_CONSTANT (0xffffffff)));
/* check for timestamp wraparound */
if (result < ext)
diff = ext - result;
else
diff = result - ext;
if (diff > G_MAXINT32) {
/* timestamp went backwards more than allowed, we wrap around and get
* updated extended timestamp. */
result += (G_GINT64_CONSTANT (1) << 32);
}
}
*exttimestamp = result;
return result;
}