<feed xmlns='http://www.w3.org/2005/Atom'>
<title>linux-toradex.git/include/linux/tcp.h, branch v4.6-rc7</title>
<subtitle>Linux kernel for Apalis and Colibri modules</subtitle>
<link rel='alternate' type='text/html' href='https://git.toradex.cn/cgit/linux-toradex.git/'/>
<entry>
<title>tcp: Add RFC4898 tcpEStatsPerfDataSegsOut/In</title>
<updated>2016-03-14T18:55:26+00:00</updated>
<author>
<name>Martin KaFai Lau</name>
<email>kafai@fb.com</email>
</author>
<published>2016-03-14T17:52:15+00:00</published>
<link rel='alternate' type='text/html' href='https://git.toradex.cn/cgit/linux-toradex.git/commit/?id=a44d6eacdaf56f74fad699af7f4925a5f5ac0e7f'/>
<id>a44d6eacdaf56f74fad699af7f4925a5f5ac0e7f</id>
<content type='text'>
Per RFC4898, they count segments sent/received
containing a positive length data segment (that includes
retransmission segments carrying data).  Unlike
tcpi_segs_out/in, tcpi_data_segs_out/in excludes segments
carrying no data (e.g. pure ack).

The patch also updates the segs_in in tcp_fastopen_add_skb()
so that segs_in &gt;= data_segs_in property is kept.

Together with retransmission data, tcpi_data_segs_out
gives a better signal on the rxmit rate.

v6: Rebase on the latest net-next

v5: Eric pointed out that checking skb-&gt;len is still needed in
tcp_fastopen_add_skb() because skb can carry a FIN without data.
Hence, instead of open coding segs_in and data_segs_in, tcp_segs_in()
helper is used.  Comment is added to the fastopen case to explain why
segs_in has to be reset and tcp_segs_in() has to be called before
__skb_pull().

v4: Add comment to the changes in tcp_fastopen_add_skb()
and also add remark on this case in the commit message.

v3: Add const modifier to the skb parameter in tcp_segs_in()

v2: Rework based on recent fix by Eric:
commit a9d99ce28ed3 ("tcp: fix tcpi_segs_in after connection establishment")

Signed-off-by: Martin KaFai Lau &lt;kafai@fb.com&gt;
Cc: Chris Rapier &lt;rapier@psc.edu&gt;
Cc: Eric Dumazet &lt;edumazet@google.com&gt;
Cc: Marcelo Ricardo Leitner &lt;mleitner@redhat.com&gt;
Cc: Neal Cardwell &lt;ncardwell@google.com&gt;
Cc: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
Per RFC4898, they count segments sent/received
containing a positive length data segment (that includes
retransmission segments carrying data).  Unlike
tcpi_segs_out/in, tcpi_data_segs_out/in excludes segments
carrying no data (e.g. pure ack).

The patch also updates the segs_in in tcp_fastopen_add_skb()
so that segs_in &gt;= data_segs_in property is kept.

Together with retransmission data, tcpi_data_segs_out
gives a better signal on the rxmit rate.

v6: Rebase on the latest net-next

v5: Eric pointed out that checking skb-&gt;len is still needed in
tcp_fastopen_add_skb() because skb can carry a FIN without data.
Hence, instead of open coding segs_in and data_segs_in, tcp_segs_in()
helper is used.  Comment is added to the fastopen case to explain why
segs_in has to be reset and tcp_segs_in() has to be called before
__skb_pull().

v4: Add comment to the changes in tcp_fastopen_add_skb()
and also add remark on this case in the commit message.

v3: Add const modifier to the skb parameter in tcp_segs_in()

v2: Rework based on recent fix by Eric:
commit a9d99ce28ed3 ("tcp: fix tcpi_segs_in after connection establishment")

Signed-off-by: Martin KaFai Lau &lt;kafai@fb.com&gt;
Cc: Chris Rapier &lt;rapier@psc.edu&gt;
Cc: Eric Dumazet &lt;edumazet@google.com&gt;
Cc: Marcelo Ricardo Leitner &lt;mleitner@redhat.com&gt;
Cc: Neal Cardwell &lt;ncardwell@google.com&gt;
Cc: Yuchung Cheng &lt;ycheng@google.com&gt;
Acked-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: __tcp_hdrlen() helper</title>
<updated>2016-02-11T08:54:14+00:00</updated>
<author>
<name>Craig Gallek</name>
<email>kraig@google.com</email>
</author>
<published>2016-02-10T16:50:37+00:00</published>
<link rel='alternate' type='text/html' href='https://git.toradex.cn/cgit/linux-toradex.git/commit/?id=d9b3fca27385eafe61c3ca6feab6cb1e7dc77482'/>
<id>d9b3fca27385eafe61c3ca6feab6cb1e7dc77482</id>
<content type='text'>
tcp_hdrlen is wasteful if you already have a pointer to struct tcphdr.
This splits the size calculation into a helper function that can be
used if a struct tcphdr is already available.

Signed-off-by: Craig Gallek &lt;kraig@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
tcp_hdrlen is wasteful if you already have a pointer to struct tcphdr.
This splits the size calculation into a helper function that can be
used if a struct tcphdr is already available.

Signed-off-by: Craig Gallek &lt;kraig@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: new delivery accounting</title>
<updated>2016-02-07T19:09:51+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2016-02-02T18:33:06+00:00</published>
<link rel='alternate' type='text/html' href='https://git.toradex.cn/cgit/linux-toradex.git/commit/?id=ddf1af6fa00e772fdb67a7d22cb83fac2b8968a8'/>
<id>ddf1af6fa00e772fdb67a7d22cb83fac2b8968a8</id>
<content type='text'>
This patch changes the accounting of how many packets are
newly acked or sacked when the sender receives an ACK.

The current approach basically computes

   newly_acked_sacked = (prior_packets - prior_sacked) -
                        (tp-&gt;packets_out - tp-&gt;sacked_out)

   where prior_packets and prior_sacked out are snapshot
   at the beginning of the ACK processing.

The new approach tracks the delivery information via a new
TCP state variable "delivered" which monotically increases
as new packets are delivered in order or out-of-order.

The reason for this change is that the current approach is
brittle that produces negative or inaccurate estimate.

   1) For non-SACK connections, an ACK that advances the SND.UNA
   could reset the DUPACK counters (tp-&gt;sacked_out) in
   tcp_process_loss() or tcp_fastretrans_alert(). This inflates
   the inflight suddenly and causes under-estimate or even
   negative estimate. Here is a real example:

                   before   after (processing ACK)
   packets_out     75       73
   sacked_out      23        0
   ca state        Loss     Open

   The old approach computes (75-23) - (73 - 0) = -21 delivered
   while the new approach computes 1 delivered since it
   considers the 2nd-24th packets are delivered OOO.

   2) MSS change would re-count packets_out and sacked_out so
   the estimate is in-accurate and can even become negative.
   E.g., the inflight is doubled when MSS is halved.

   3) Spurious retransmission signaled by DSACK is not accounted

The new approach is simpler and more robust. For SACK connections,
tp-&gt;delivered increments as packets are being acked or sacked in
SACK and ACK processing.

For non-sack connections, it's done in tcp_remove_reno_sacks() and
tcp_add_reno_sack(). When an ACK advances the SND.UNA, tp-&gt;delivered
is incremented by the number of packets ACKed (less the current
number of DUPACKs received plus one packet hole).  Upon receiving
a DUPACK, tp-&gt;delivered is incremented assuming one out-of-order
packet is delivered.

Upon receiving a DSACK, tp-&gt;delivered is incremtened assuming one
retransmission is delivered in tcp_sacktag_write_queue().

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: Eric Dumazet &lt;ncardwell@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
This patch changes the accounting of how many packets are
newly acked or sacked when the sender receives an ACK.

The current approach basically computes

   newly_acked_sacked = (prior_packets - prior_sacked) -
                        (tp-&gt;packets_out - tp-&gt;sacked_out)

   where prior_packets and prior_sacked out are snapshot
   at the beginning of the ACK processing.

The new approach tracks the delivery information via a new
TCP state variable "delivered" which monotically increases
as new packets are delivered in order or out-of-order.

The reason for this change is that the current approach is
brittle that produces negative or inaccurate estimate.

   1) For non-SACK connections, an ACK that advances the SND.UNA
   could reset the DUPACK counters (tp-&gt;sacked_out) in
   tcp_process_loss() or tcp_fastretrans_alert(). This inflates
   the inflight suddenly and causes under-estimate or even
   negative estimate. Here is a real example:

                   before   after (processing ACK)
   packets_out     75       73
   sacked_out      23        0
   ca state        Loss     Open

   The old approach computes (75-23) - (73 - 0) = -21 delivered
   while the new approach computes 1 delivered since it
   considers the 2nd-24th packets are delivered OOO.

   2) MSS change would re-count packets_out and sacked_out so
   the estimate is in-accurate and can even become negative.
   E.g., the inflight is doubled when MSS is halved.

   3) Spurious retransmission signaled by DSACK is not accounted

The new approach is simpler and more robust. For SACK connections,
tp-&gt;delivered increments as packets are being acked or sacked in
SACK and ACK processing.

For non-sack connections, it's done in tcp_remove_reno_sacks() and
tcp_add_reno_sack(). When an ACK advances the SND.UNA, tp-&gt;delivered
is incremented by the number of packets ACKed (less the current
number of DUPACKs received plus one packet hole).  Upon receiving
a DUPACK, tp-&gt;delivered is incremented assuming one out-of-order
packet is delivered.

Upon receiving a DSACK, tp-&gt;delivered is incremtened assuming one
retransmission is delivered in tcp_sacktag_write_queue().

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: Eric Dumazet &lt;ncardwell@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: fix req-&gt;saved_syn race</title>
<updated>2015-11-05T19:36:09+00:00</updated>
<author>
<name>Eric Dumazet</name>
<email>edumazet@google.com</email>
</author>
<published>2015-11-05T19:07:13+00:00</published>
<link rel='alternate' type='text/html' href='https://git.toradex.cn/cgit/linux-toradex.git/commit/?id=805c4bc05705fb2b71ec970960b456eee9900953'/>
<id>805c4bc05705fb2b71ec970960b456eee9900953</id>
<content type='text'>
For the reasons explained in commit ce1050089c96 ("tcp/dccp: fix
ireq-&gt;pktopts race"), we need to make sure we do not access
req-&gt;saved_syn unless we own the request sock.

This fixes races for listeners using TCP_SAVE_SYN option.

Fixes: e994b2f0fb92 ("tcp: do not lock listener to process SYN packets")
Fixes: 079096f103fa ("tcp/dccp: install syn_recv requests into ehash table")
Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Reported-by: Ying Cai &lt;ycai@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
For the reasons explained in commit ce1050089c96 ("tcp/dccp: fix
ireq-&gt;pktopts race"), we need to make sure we do not access
req-&gt;saved_syn unless we own the request sock.

This fixes races for listeners using TCP_SAVE_SYN option.

Fixes: e994b2f0fb92 ("tcp: do not lock listener to process SYN packets")
Fixes: 079096f103fa ("tcp/dccp: install syn_recv requests into ehash table")
Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Reported-by: Ying Cai &lt;ycai@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: fastopen: limit max_qlen</title>
<updated>2015-10-22T13:22:13+00:00</updated>
<author>
<name>Eric Dumazet</name>
<email>edumazet@google.com</email>
</author>
<published>2015-10-20T20:17:40+00:00</published>
<link rel='alternate' type='text/html' href='https://git.toradex.cn/cgit/linux-toradex.git/commit/?id=dbf650b67bb4db1b95807d2aafe2d7cfafd458da'/>
<id>dbf650b67bb4db1b95807d2aafe2d7cfafd458da</id>
<content type='text'>
Allowing an application to set whatever limit for
the list of recently RST fastopen sessions [1] is not wise,
as it open ways to deplete kernel memory.

Cap the user provided limit by somaxconn sysctl,
like listen() backlog.

[1] https://tools.ietf.org/html/rfc7413#section-5.1

Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
Allowing an application to set whatever limit for
the list of recently RST fastopen sessions [1] is not wise,
as it open ways to deplete kernel memory.

Cap the user provided limit by somaxconn sysctl,
like listen() backlog.

[1] https://tools.ietf.org/html/rfc7413#section-5.1

Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: track the packet timings in RACK</title>
<updated>2015-10-21T14:00:48+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2015-10-17T04:57:46+00:00</published>
<link rel='alternate' type='text/html' href='https://git.toradex.cn/cgit/linux-toradex.git/commit/?id=659a8ad56f490279f0efee43a62ffa1ac914a4e0'/>
<id>659a8ad56f490279f0efee43a62ffa1ac914a4e0</id>
<content type='text'>
This patch is the first half of the RACK loss recovery.

RACK loss recovery uses the notion of time instead
of packet sequence (FACK) or counts (dupthresh). It's inspired by the
previous FACK heuristic in tcp_mark_lost_retrans(): when a limited
transmit (new data packet) is sacked, then current retransmitted
sequence below the newly sacked sequence must been lost,
since at least one round trip time has elapsed.

But it has several limitations:
1) can't detect tail drops since it depends on limited transmit
2) is disabled upon reordering (assumes no reordering)
3) only enabled in fast recovery ut not timeout recovery

RACK (Recently ACK) addresses these limitations with the notion
of time instead: a packet P1 is lost if a later packet P2 is s/acked,
as at least one round trip has passed.

Since RACK cares about the time sequence instead of the data sequence
of packets, it can detect tail drops when later retransmission is
s/acked while FACK or dupthresh can't. For reordering RACK uses a
dynamically adjusted reordering window ("reo_wnd") to reduce false
positives on ever (small) degree of reordering.

This patch implements tcp_advanced_rack() which tracks the
most recent transmission time among the packets that have been
delivered (ACKed or SACKed) in tp-&gt;rack.mstamp. This timestamp
is the key to determine which packet has been lost.

Consider an example that the sender sends six packets:
T1: P1 (lost)
T2: P2
T3: P3
T4: P4
T100: sack of P2. rack.mstamp = T2
T101: retransmit P1
T102: sack of P2,P3,P4. rack.mstamp = T4
T205: ACK of P4 since the hole is repaired. rack.mstamp = T101

We need to be careful about spurious retransmission because it may
falsely advance tp-&gt;rack.mstamp by an RTT or an RTO, causing RACK
to falsely mark all packets lost, just like a spurious timeout.

We identify spurious retransmission by the ACK's TS echo value.
If TS option is not applicable but the retransmission is acknowledged
less than min-RTT ago, it is likely to be spurious. We refrain from
using the transmission time of these spurious retransmissions.

The second half is implemented in the next patch that marks packet
lost using RACK timestamp.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
This patch is the first half of the RACK loss recovery.

RACK loss recovery uses the notion of time instead
of packet sequence (FACK) or counts (dupthresh). It's inspired by the
previous FACK heuristic in tcp_mark_lost_retrans(): when a limited
transmit (new data packet) is sacked, then current retransmitted
sequence below the newly sacked sequence must been lost,
since at least one round trip time has elapsed.

But it has several limitations:
1) can't detect tail drops since it depends on limited transmit
2) is disabled upon reordering (assumes no reordering)
3) only enabled in fast recovery ut not timeout recovery

RACK (Recently ACK) addresses these limitations with the notion
of time instead: a packet P1 is lost if a later packet P2 is s/acked,
as at least one round trip has passed.

Since RACK cares about the time sequence instead of the data sequence
of packets, it can detect tail drops when later retransmission is
s/acked while FACK or dupthresh can't. For reordering RACK uses a
dynamically adjusted reordering window ("reo_wnd") to reduce false
positives on ever (small) degree of reordering.

This patch implements tcp_advanced_rack() which tracks the
most recent transmission time among the packets that have been
delivered (ACKed or SACKed) in tp-&gt;rack.mstamp. This timestamp
is the key to determine which packet has been lost.

Consider an example that the sender sends six packets:
T1: P1 (lost)
T2: P2
T3: P3
T4: P4
T100: sack of P2. rack.mstamp = T2
T101: retransmit P1
T102: sack of P2,P3,P4. rack.mstamp = T4
T205: ACK of P4 since the hole is repaired. rack.mstamp = T101

We need to be careful about spurious retransmission because it may
falsely advance tp-&gt;rack.mstamp by an RTT or an RTO, causing RACK
to falsely mark all packets lost, just like a spurious timeout.

We identify spurious retransmission by the ACK's TS echo value.
If TS option is not applicable but the retransmission is acknowledged
less than min-RTT ago, it is likely to be spurious. We refrain from
using the transmission time of these spurious retransmissions.

The second half is implemented in the next patch that marks packet
lost using RACK timestamp.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: remove tcp_mark_lost_retrans()</title>
<updated>2015-10-21T14:00:44+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2015-10-17T04:57:43+00:00</published>
<link rel='alternate' type='text/html' href='https://git.toradex.cn/cgit/linux-toradex.git/commit/?id=af82f4e84866ecd360a53f770d6217637116e6c1'/>
<id>af82f4e84866ecd360a53f770d6217637116e6c1</id>
<content type='text'>
Remove the existing lost retransmit detection because RACK subsumes
it completely. This also stops the overloading the ack_seq field of
the skb control block.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
Remove the existing lost retransmit detection because RACK subsumes
it completely. This also stops the overloading the ack_seq field of
the skb control block.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: track min RTT using windowed min-filter</title>
<updated>2015-10-21T14:00:43+00:00</updated>
<author>
<name>Yuchung Cheng</name>
<email>ycheng@google.com</email>
</author>
<published>2015-10-17T04:57:42+00:00</published>
<link rel='alternate' type='text/html' href='https://git.toradex.cn/cgit/linux-toradex.git/commit/?id=f672258391b42a5c7cc2732c9c063e56a85c8dbe'/>
<id>f672258391b42a5c7cc2732c9c063e56a85c8dbe</id>
<content type='text'>
Kathleen Nichols' algorithm for tracking the minimum RTT of a
data stream over some measurement window. It uses constant space
and constant time per update. Yet it almost always delivers
the same minimum as an implementation that has to keep all
the data in the window. The measurement window is tunable via
sysctl.net.ipv4.tcp_min_rtt_wlen with a default value of 5 minutes.

The algorithm keeps track of the best, 2nd best &amp; 3rd best min
values, maintaining an invariant that the measurement time of
the n'th best &gt;= n-1'th best. It also makes sure that the three
values are widely separated in the time window since that bounds
the worse case error when that data is monotonically increasing
over the window.

Upon getting a new min, we can forget everything earlier because
it has no value - the new min is less than everything else in the
window by definition and it's the most recent. So we restart fresh
on every new min and overwrites the 2nd &amp; 3rd choices. The same
property holds for the 2nd &amp; 3rd best.

Therefore we have to maintain two invariants to maximize the
information in the samples, one on values (1st.v &lt;= 2nd.v &lt;=
3rd.v) and the other on times (now-win &lt;=1st.t &lt;= 2nd.t &lt;= 3rd.t &lt;=
now). These invariants determine the structure of the code

The RTT input to the windowed filter is the minimum RTT measured
from ACK or SACK, or as the last resort from TCP timestamps.

The accessor tcp_min_rtt() returns the minimum RTT seen in the
window. ~0U indicates it is not available. The minimum is 1usec
even if the true RTT is below that.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
Kathleen Nichols' algorithm for tracking the minimum RTT of a
data stream over some measurement window. It uses constant space
and constant time per update. Yet it almost always delivers
the same minimum as an implementation that has to keep all
the data in the window. The measurement window is tunable via
sysctl.net.ipv4.tcp_min_rtt_wlen with a default value of 5 minutes.

The algorithm keeps track of the best, 2nd best &amp; 3rd best min
values, maintaining an invariant that the measurement time of
the n'th best &gt;= n-1'th best. It also makes sure that the three
values are widely separated in the time window since that bounds
the worse case error when that data is monotonically increasing
over the window.

Upon getting a new min, we can forget everything earlier because
it has no value - the new min is less than everything else in the
window by definition and it's the most recent. So we restart fresh
on every new min and overwrites the 2nd &amp; 3rd choices. The same
property holds for the 2nd &amp; 3rd best.

Therefore we have to maintain two invariants to maximize the
information in the samples, one on values (1st.v &lt;= 2nd.v &lt;=
3rd.v) and the other on times (now-win &lt;=1st.t &lt;= 2nd.t &lt;= 3rd.t &lt;=
now). These invariants determine the structure of the code

The RTT input to the windowed filter is the minimum RTT measured
from ACK or SACK, or as the last resort from TCP timestamps.

The accessor tcp_min_rtt() returns the minimum RTT seen in the
window. ~0U indicates it is not available. The minimum is 1usec
even if the true RTT is below that.

Signed-off-by: Yuchung Cheng &lt;ycheng@google.com&gt;
Signed-off-by: Neal Cardwell &lt;ncardwell@google.com&gt;
Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: shrink tcp_timewait_sock by 8 bytes</title>
<updated>2015-10-13T02:28:24+00:00</updated>
<author>
<name>Eric Dumazet</name>
<email>edumazet@google.com</email>
</author>
<published>2015-10-09T02:33:24+00:00</published>
<link rel='alternate' type='text/html' href='https://git.toradex.cn/cgit/linux-toradex.git/commit/?id=d475f090bf1c0dc2999e98bbf2e7cb2243358849'/>
<id>d475f090bf1c0dc2999e98bbf2e7cb2243358849</id>
<content type='text'>
Reducing tcp_timewait_sock from 280 bytes to 272 bytes
allows SLAB to pack 15 objects per page instead of 14 (on x86)

Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
Reducing tcp_timewait_sock from 280 bytes to 272 bytes
allows SLAB to pack 15 objects per page instead of 14 (on x86)

Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
<entry>
<title>tcp: prepare fastopen code for upcoming listener changes</title>
<updated>2015-09-29T23:53:10+00:00</updated>
<author>
<name>Eric Dumazet</name>
<email>edumazet@google.com</email>
</author>
<published>2015-09-29T14:42:52+00:00</published>
<link rel='alternate' type='text/html' href='https://git.toradex.cn/cgit/linux-toradex.git/commit/?id=0536fcc039a8926ec12ec587f41a83f7acafeb82'/>
<id>0536fcc039a8926ec12ec587f41a83f7acafeb82</id>
<content type='text'>
While auditing TCP stack for upcoming 'lockless' listener changes,
I found I had to change fastopen_init_queue() to properly init the object
before publishing it.

Otherwise an other cpu could try to lock the spinlock before it gets
properly initialized.

Instead of adding appropriate barriers, just remove dynamic memory
allocations :
- Structure is 28 bytes on 64bit arches. Using additional 8 bytes
  for holding a pointer seems overkill.
- Two listeners can share same cache line and performance would suffer.

If we really want to save few bytes, we would instead dynamically allocate
whole struct request_sock_queue in the future.

Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</content>
<content type='xhtml'>
<div xmlns='http://www.w3.org/1999/xhtml'>
<pre>
While auditing TCP stack for upcoming 'lockless' listener changes,
I found I had to change fastopen_init_queue() to properly init the object
before publishing it.

Otherwise an other cpu could try to lock the spinlock before it gets
properly initialized.

Instead of adding appropriate barriers, just remove dynamic memory
allocations :
- Structure is 28 bytes on 64bit arches. Using additional 8 bytes
  for holding a pointer seems overkill.
- Two listeners can share same cache line and performance would suffer.

If we really want to save few bytes, we would instead dynamically allocate
whole struct request_sock_queue in the future.

Signed-off-by: Eric Dumazet &lt;edumazet@google.com&gt;
Signed-off-by: David S. Miller &lt;davem@davemloft.net&gt;
</pre>
</div>
</content>
</entry>
</feed>
