mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-01-11 09:55:36 +00:00
Code Issues 1 Projects Releases Wiki Activity

gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing.

Browse source 
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
(gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
(gst_rtcp_packet_bye_set_reason):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Fix RB block parsing and writing.
Add support for constructing BYE packets.
This commit is contained in:
Wim Taymans 2007-04-27 15:01:40 +00:00
parent 9e873a3c83
commit f23356bd8f
3 changed files with 133 additions and 34 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

11
ChangeLog
View file

@ -1,3 +1,14 @@
2007-04-27 Wim Taymans <wim@fluendo.com>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
(gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
(gst_rtcp_packet_bye_set_reason):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Fix RB block parsing and writing.
Add support for constructing BYE packets.
2007-04-25 Tim-Philipp Müller <tim at centricular dot net>
* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),

145
gst-libs/gst/rtp/gstrtcpbuffer.c
View file

@ -749,10 +749,10 @@ gst_rtcp_packet_get_rb (GstRTCPPacket * packet, guint nth, guint32 * ssrc,
if (packet->type == GST_RTCP_TYPE_RR)
data += 4;
else
data += 36;
data += 24;
/* move to requested index */
data += (nth * 36);
data += (nth * 24);
if (ssrc)
*ssrc = GST_READ_UINT32_BE (data);
@ -822,24 +822,23 @@ gst_rtcp_packet_add_rb (GstRTCPPacket * packet, guint32 ssrc,
/* skip header */
offset = packet->offset + 4;
if (packet->type == GST_RTCP_TYPE_RR)
offset += 4;
offset += 8;
else
offset += 36;
offset += 28;
/* move to current index */
offset += (packet->count * 36);
offset += (packet->count * 24);
if (offset >= size)
/* we need 24 free bytes now */
if (offset + 24 >= size)
goto no_space;
data += packet->offset;
/* increment packet count and length */
packet->count++;
data[0]++;
data[packet->offset]++;
packet->length += 6;
data[2] = (packet->length) >> 8;
data[3] = (packet->length) & 0xff;
data[packet->offset + 2] = (packet->length) >> 8;
data[packet->offset + 3] = (packet->length) & 0xff;
/* move to new report block offset */
data += offset;
@ -1202,12 +1201,10 @@ gst_rtcp_packet_sdes_add_item (GstRTCPPacket * packet, guint32 ssrc)
/* ERRORS */
no_space:
{
g_print ("no space\n");
return FALSE;
}
no_next:
{
g_print ("no next\n");
packet->count--;
return FALSE;
}
@ -1270,7 +1267,6 @@ gst_rtcp_packet_sdes_add_entry (GstRTCPPacket * packet, GstRTCPSDESType type,
/* ERRORS */
no_space:
{
g_print ("no space\n");
return FALSE;
}
}
@ -1347,16 +1343,53 @@ gst_rtcp_packet_bye_get_nth_ssrc (GstRTCPPacket * packet, guint nth)
*
* Add @ssrc to the BYE @packet.
*
* Note: Not implemented.
* 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.
*/
void
gboolean
gst_rtcp_packet_bye_add_ssrc (GstRTCPPacket * packet, guint32 ssrc)
{
g_return_if_fail (packet != NULL);
g_return_if_fail (packet->type == GST_RTCP_TYPE_BYE);
g_return_if_fail (GST_IS_BUFFER (packet->buffer));
guint8 *data;
guint size, offset;
g_warning ("not implemented");
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;
}
}
/**
@ -1367,30 +1400,37 @@ gst_rtcp_packet_bye_add_ssrc (GstRTCPPacket * packet, guint32 ssrc)
*
* Adds @len SSRCs in @ssrc to BYE @packet.
*
* Note: Not implemented.
* 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.
*/
void
gboolean
gst_rtcp_packet_bye_add_ssrcs (GstRTCPPacket * packet, guint32 * ssrc,
guint len)
{
g_return_if_fail (packet != NULL);
g_return_if_fail (packet->type == GST_RTCP_TYPE_BYE);
g_return_if_fail (GST_IS_BUFFER (packet->buffer));
guint i;
gboolean res;
g_warning ("not implemented");
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 sc;
guint offset;
/* get amount of sources */
sc = packet->count;
/* get amount of sources plus header */
offset = 1 + packet->count;
offset = 1 + sc;
/* check that we don't go past the packet length */
if (offset > packet->length)
return 0;
@ -1489,11 +1529,52 @@ gst_rtcp_packet_bye_get_reason (GstRTCPPacket * packet)
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);
g_warning ("not implemented");
if (reason == NULL)
return TRUE;
return FALSE;
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;
}
}

11
gst-libs/gst/rtp/gstrtcpbuffer.h
View file

@ -105,6 +105,13 @@ typedef enum
*/
#define GST_RTCP_MAX_SDES_ITEM_COUNT 31
/**
* GST_RTCP_MAX_BYE_SSRC_COUNT:
*
* The maximum amount of SSRCs in a BYE packet.
*/
#define GST_RTCP_MAX_BYE_SSRC_COUNT 31
/**
* GST_RTCP_VALID_MASK:
*
@ -216,8 +223,8 @@ gboolean gst_rtcp_packet_sdes_add_entry (GstRTCPPacket *packet, Gs
/* bye packet */
guint gst_rtcp_packet_bye_get_ssrc_count (GstRTCPPacket *packet);
guint32 gst_rtcp_packet_bye_get_nth_ssrc (GstRTCPPacket *packet, guint nth);
void gst_rtcp_packet_bye_add_ssrc (GstRTCPPacket *packet, guint32 ssrc);
void gst_rtcp_packet_bye_add_ssrcs (GstRTCPPacket *packet, guint32 *ssrc, guint len);
gboolean gst_rtcp_packet_bye_add_ssrc (GstRTCPPacket *packet, guint32 ssrc);
gboolean gst_rtcp_packet_bye_add_ssrcs (GstRTCPPacket *packet, guint32 *ssrc, guint len);
guint8 gst_rtcp_packet_bye_get_reason_len (GstRTCPPacket *packet);
gchar* gst_rtcp_packet_bye_get_reason (GstRTCPPacket *packet);
gboolean gst_rtcp_packet_bye_set_reason (GstRTCPPacket *packet, const gchar *reason);