Refactor tcp_ecn_check_ce and __tcp_ecn_check_ce to accept struct sock*
instead of tcp_sock* to clean up type casts. This is a pure refactor
patch.
Signed-off-by: Yousuk Seung <ysseung@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This is additional to the
commit ea1627c20c ("tcp: minor optimizations around tcp_hdr() usage").
At this point, skb->data is same with tcp_hdr() as tcp header has not
been pulled yet. So use the less expensive one to get the tcp header.
Remove the third parameter of tcp_rcv_established() and put it into
the function body.
Furthermore, the local variables are listed as a reverse christmas tree :)
Cc: Eric Dumazet <edumazet@google.com>
Signed-off-by: Yafang Shao <laoar.shao@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
ECN signals currently forces TCP to enter quickack mode for
up to 16 (TCP_MAX_QUICKACKS) following incoming packets.
We believe this is not needed, and only sending one immediate ack
for the current packet should be enough.
This should reduce the extra load noticed in DCTCP environments,
after congestion events.
This is part 2 of our effort to reduce pure ACK packets.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
We want to add finer control of the number of ACK packets sent after
ECN events.
This patch is not changing current behavior, it only enables following
change.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This per netns sysctl allows for TCP SACK compression fine-tuning.
This limits number of SACK that can be compressed.
Using 0 disables SACK compression.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This per netns sysctl allows for TCP SACK compression fine-tuning.
Its default value is 1,000,000, or 1 ms to meet TSO autosizing period.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
When TCP receives an out-of-order packet, it immediately sends
a SACK packet, generating network load but also forcing the
receiver to send 1-MSS pathological packets, increasing its
RTX queue length/depth, and thus processing time.
Wifi networks suffer from this aggressive behavior, but generally
speaking, all these SACK packets add fuel to the fire when networks
are under congestion.
This patch adds a high resolution timer and tp->compressed_ack counter.
Instead of sending a SACK, we program this timer with a small delay,
based on RTT and capped to 1 ms :
delay = min ( 5 % of RTT, 1 ms)
If subsequent SACKs need to be sent while the timer has not yet
expired, we simply increment tp->compressed_ack.
When timer expires, a SACK is sent with the latest information.
Whenever an ACK is sent (if data is sent, or if in-order
data is received) timer is canceled.
Note that tcp_sack_new_ofo_skb() is able to force a SACK to be sent
if the sack blocks need to be shuffled, even if the timer has not
expired.
A new SNMP counter is added in the following patch.
Two other patches add sysctls to allow changing the 1,000,000 and 44
values that this commit hard-coded.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Toke Høiland-Jørgensen <toke@toke.dk>
Signed-off-by: David S. Miller <davem@davemloft.net>
As explained in commit 9f9843a751 ("tcp: properly handle stretch
acks in slow start"), TCP stacks have to consider how many packets
are acknowledged in one single ACK, because of GRO, but also
because of ACK compression or losses.
We plan to add SACK compression in the following patch, we
must therefore not call tcp_enter_quickack_mode()
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
An RTO event indicates the head has not been acked for a long time
after its last (re)transmission. But the other packets are not
necessarily lost if they have been only sent recently (for example
due to application limit). This patch would prohibit marking packets
sent within an RTT to be lost on RTO event, using similar logic in
TCP RACK detection.
Normally the head (SND.UNA) would be marked lost since RTO should
fire strictly after the head was sent. An exception is when the
most recent RACK RTT measurement is larger than the (previous)
RTO. To address this exception the head is always marked lost.
Congestion control interaction: since we may not mark every packet
lost, the congestion window may be more than 1 (inflight plus 1).
But only one packet will be retransmitted after RTO, since
tcp_retransmit_timer() calls tcp_retransmit_skb(...,segs=1). The
connection still performs slow start from one packet (with Cubic
congestion control).
This commit was tested in an A/B test with Google web servers,
and showed a reduction of 2% in (spurious) retransmits post
timeout (SlowStartRetrans), and correspondingly reduced DSACKs
(DSACKIgnoredOld) by 7%.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Reviewed-by: Soheil Hassas Yeganeh <soheil@google.com>
Reviewed-by: Priyaranjan Jha <priyarjha@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Create and export a new helper tcp_rack_skb_timeout and move tcp_is_rack
to prepare the final RTO change.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Reviewed-by: Soheil Hassas Yeganeh <soheil@google.com>
Reviewed-by: Priyaranjan Jha <priyarjha@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Previously when TCP times out, it first updates cwnd and ssthresh,
marks packets lost, and then updates congestion state again. This
was fine because everything not yet delivered is marked lost,
so the inflight is always 0 and cwnd can be safely set to 1 to
retransmit one packet on timeout.
But the inflight may not always be 0 on timeout if TCP changes to
mark packets lost based on packet sent time. Therefore we must
first mark the packet lost, then set the cwnd based on the
(updated) inflight.
This is not a pure refactor. Congestion control may potentially
break if it uses (not yet updated) inflight to compute ssthresh.
Fortunately all existing congestion control modules does not do that.
Also it changes the inflight when CA_LOSS_EVENT is called, and only
westwood processes such an event but does not use inflight.
This change has two other minor side benefits:
1) consistent with Fast Recovery s.t. the inflight is updated
first before tcp_enter_recovery flips state to CA_Recovery.
2) avoid intertwining loss marking with state update, making the
code more readable.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Reviewed-by: Soheil Hassas Yeganeh <soheil@google.com>
Reviewed-by: Priyaranjan Jha <priyarjha@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Refactor using a new helper, tcp_timeout_mark_loss(), that marks packets
lost upon RTO.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Reviewed-by: Soheil Hassas Yeganeh <soheil@google.com>
Reviewed-by: Priyaranjan Jha <priyarjha@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
The previous approach for the lost and retransmit bits was to
wipe the slate clean: zero all the lost and retransmit bits,
correspondingly zero the lost_out and retrans_out counters, and
then add back the lost bits (and correspondingly increment lost_out).
The new approach is to treat this very much like marking packets
lost in fast recovery. We don’t wipe the slate clean. We just say
that for all packets that were not yet marked sacked or lost, we now
mark them as lost in exactly the same way we do for fast recovery.
This fixes the lost retransmit accounting at RTO time and greatly
simplifies the RTO code by sharing much of the logic with Fast
Recovery.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Reviewed-by: Soheil Hassas Yeganeh <soheil@google.com>
Reviewed-by: Priyaranjan Jha <priyarjha@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This is a rewrite of NewReno loss recovery implementation that is
simpler and standalone for readability and better performance by
using less states.
Note that NewReno refers to RFC6582 as a modification to the fast
recovery algorithm. It is used only if the connection does not
support SACK in Linux. It should not to be confused with the Reno
(AIMD) congestion control.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Reviewed-by: Soheil Hassas Yeganeh <soheil@google.com>
Reviewed-by: Priyaranjan Jha <priyarjha@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch disables RFC6675 loss detection and make sysctl
net.ipv4.tcp_recovery = 1 controls a binary choice between RACK
(1) or RFC6675 (0).
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Reviewed-by: Soheil Hassas Yeganeh <soheil@google.com>
Reviewed-by: Priyaranjan Jha <priyarjha@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Called when a TCP segment is acknowledged.
Could be used by application protocols who hold additional
metadata associated with the stream data.
This is required by TLS device offload to release
metadata associated with acknowledged TLS records.
Signed-off-by: Ilya Lesokhin <ilyal@mellanox.com>
Signed-off-by: Boris Pismenny <borisp@mellanox.com>
Signed-off-by: Aviad Yehezkel <aviadye@mellanox.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This reverts commit <c6849a3ac17e> ("net: init sk_cookie for inet socket")
Per discussion with Eric, when update sock_net(sk)->cookie_gen, the
whole cache cache line will be invalidated, as this cache line is shared
with all cpus, that may cause great performace hit.
Bellow is the data form Eric.
"Performance is reduced from ~5 Mpps to ~3.8 Mpps with 16 RX queues on
my host" when running synflood test.
Have to revert it to prevent from cache line false sharing.
Signed-off-by: Yafang Shao <laoar.shao@gmail.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
With sk_cookie we can identify a socket, that is very helpful for
traceing and statistic, i.e. tcp tracepiont and ebpf.
So we'd better init it by default for inet socket.
When using it, we just need call atomic64_read(&sk->sk_cookie).
Signed-off-by: Yafang Shao <laoar.shao@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
tcp_rcv_space_adjust is called every time data is copied to user space,
introducing a tcp tracepoint for which could show us when the packet is
copied to user.
When a tcp packet arrives, tcp_rcv_established() will be called and with
the existed tracepoint tcp_probe we could get the time when this packet
arrives.
Then this packet will be copied to user, and tcp_rcv_space_adjust will
be called and with this new introduced tracepoint we could get the time
when this packet is copied to user.
With these two tracepoints, we could figure out whether the user program
processes this packet immediately or there's latency.
Hence in the printk message, sk_cookie is printed as a key to relate
tcp_rcv_space_adjust with tcp_probe.
Maybe we could export sockfd in this new tracepoint as well, then we
could relate this new tracepoint with epoll/read/recv* tracepoints, and
finally that could show us the whole lifespan of this packet. But we
could also implement that with pid as these functions are executed in
process context.
Signed-off-by: Yafang Shao <laoar.shao@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Export data delivered and delivered with CE marks to
1) SNMP TCPDelivered and TCPDeliveredCE
2) getsockopt(TCP_INFO)
3) Timestamping API SOF_TIMESTAMPING_OPT_STATS
Note that for SCM_TSTAMP_ACK, the delivery info in
SOF_TIMESTAMPING_OPT_STATS is reported before the info
was fully updated on the ACK.
These stats help application monitor TCP delivery and ECN status
on per host, per connection, even per message level.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Reviewed-by: Neal Cardwell <ncardwell@google.com>
Reviewed-by: Soheil Hassas Yeganeh <soheil@google.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Introduce a new delivered_ce stat in tcp socket to estimate
number of packets being marked with CE bits. The estimation is
done via ACKs with ECE bit. Depending on the actual receiver
behavior, the estimation could have biases.
Since the TCP sender can't really see the CE bit in the data path,
so the sender is technically counting packets marked delivered with
the "ECE / ECN-Echo" flag set.
With RFC3168 ECN, because the ECE bit is sticky, this count can
drastically overestimate the nummber of CE-marked data packets
With DCTCP-style ECN this should be reasonably precise unless there
is loss in the ACK path, in which case it's not precise.
With AccECN proposal this can be made still more precise, even in
the case some degree of ACK loss.
However this is sender's best estimate of CE information.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Reviewed-by: Neal Cardwell <ncardwell@google.com>
Reviewed-by: Soheil Hassas Yeganeh <soheil@google.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Add new helper tcp_newly_delivered() to prepare the ECN accounting change.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Reviewed-by: Neal Cardwell <ncardwell@google.com>
Reviewed-by: Soheil Hassas Yeganeh <soheil@google.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
the tcp_sock:delivered has inconsistent accounting for SYN and FIN.
1. it counts pure FIN
2. it counts pure SYN
3. it counts SYN-data twice
4. it does not count SYN-ACK
For congestion control perspective it does not matter much as C.C. only
cares about the difference not the aboslute value. But the next patch
would export this field to user-space so it's better to report the absolute
value w/o these caveats.
This patch counts SYN, SYN-ACK, or SYN-data delivery once always in
the "delivered" field.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Reviewed-by: Neal Cardwell <ncardwell@google.com>
Reviewed-by: Soheil Hassas Yeganeh <soheil@google.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
SO_RCVLOWAT is properly handled in tcp_poll(), so that POLLIN is only
generated when enough bytes are available in receive queue, after
David change (commit c7004482e8 "tcp: Respect SO_RCVLOWAT in tcp_poll().")
But TCP still calls sk->sk_data_ready() for each chunk added in receive
queue, meaning thread is awaken, and goes back to sleep shortly after.
Tested:
tcp_mmap test program, receiving 32768 MB of data with SO_RCVLOWAT set to 512KB
-> Should get ~2 wakeups (c-switches) per MB, regardless of how many
(tiny or big) packets were received.
High speed (mostly full size GRO packets)
received 32768 MB (100 % mmap'ed) in 8.03112 s, 34.2266 Gbit,
cpu usage user:0.037 sys:1.404, 43.9758 usec per MB, 65497 c-switches
received 32768 MB (99.9954 % mmap'ed) in 7.98453 s, 34.4263 Gbit,
cpu usage user:0.03 sys:1.422, 44.3115 usec per MB, 65485 c-switches
Low speed (sender is ratelimited and sends 1-MSS at a time, so GRO is not helping)
received 22474.5 MB (100 % mmap'ed) in 6015.35 s, 0.0313414 Gbit,
cpu usage user:0.05 sys:1.586, 72.7952 usec per MB, 44950 c-switches
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
We should not delay acks if there are not enough bytes
in receive queue to satisfy SO_RCVLOWAT.
Since [E]POLLIN event is not going to be generated, there is little
hope for a delayed ack to be useful.
In fact, delaying ACK prevents sender from completing
the transfer.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Minor conflicts in drivers/net/ethernet/mellanox/mlx5/core/en_rep.c,
we had some overlapping changes:
1) In 'net' MLX5E_PARAMS_LOG_{SQ,RQ}_SIZE -->
MLX5E_REP_PARAMS_LOG_{SQ,RQ}_SIZE
2) In 'net-next' params->log_rq_size is renamed to be
params->log_rq_mtu_frames.
3) In 'net-next' params->hard_mtu is added.
Signed-off-by: David S. Miller <davem@davemloft.net>
Currently, the SMC experimental TCP option in a SYN packet is lost on
the server side when SYN Cookies are active. However, the corresponding
SYNACK sent back to the client contains the SMC option. This causes an
inconsistent view of the SMC capabilities on the client and server.
This patch disables the SMC option in the SYNACK when SYN Cookies are
active to avoid this issue.
Fixes: 60e2a77807 ("tcp: TCP experimental option for SMC")
Signed-off-by: Hans Wippel <hwippel@linux.vnet.ibm.com>
Signed-off-by: Ursula Braun <ubraun@linux.vnet.ibm.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
All of the conflicts were cases of overlapping changes.
In net/core/devlink.c, we have to make care that the
resouce size_params have become a struct member rather
than a pointer to such an object.
Signed-off-by: David S. Miller <davem@davemloft.net>
When the connection is reset, there is no point in
keeping the packets on the write queue until the connection
is closed.
RFC 793 (page 70) and RFC 793-bis (page 64) both suggest
purging the write queue upon RST:
https://tools.ietf.org/html/draft-ietf-tcpm-rfc793bis-07
Moreover, this is essential for a correct MSG_ZEROCOPY
implementation, because userspace cannot call close(fd)
before receiving zerocopy signals even when the connection
is reset.
Fixes: f214f915e7 ("tcp: enable MSG_ZEROCOPY")
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This reverts commit 89fe18e44f.
While the patch could detect more spurious timeouts, it could cause
poor TCP performance on broken middle-boxes that modifies TCP packets
(e.g. receive window, SACK options). Since the performance gain is
much smaller compared to the potential loss. The best solution is
to fully revert the change.
Fixes: 89fe18e44f ("tcp: extend F-RTO to catch more spurious timeouts")
Reported-by: Teodor Milkov <tm@del.bg>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This reverts commit cc663f4d4c. While fixing
some broken middle-boxes that modifies receive window fields, it does not
address middle-boxes that strip off SACK options. The best solution is
to fully revert this patch and the root F-RTO enhancement.
Fixes: cc663f4d4c ("tcp: restrict F-RTO to work-around broken middle-boxes")
Reported-by: Teodor Milkov <tm@del.bg>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
After previous commit, sk_can_gso() is always true.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
배석진 reported that in some situations, packets for a given 5-tuple
end up being processed by different CPUS.
This involves RPS, and fragmentation.
배석진 is seeing packet drops when a SYN_RECV request socket is
moved into ESTABLISH state. Other states are protected by socket lock.
This is caused by a CPU losing the race, and simply not caring enough.
Since this seems to occur frequently, we can do better and perform
a second lookup.
Note that all needed memory barriers are already in the existing code,
thanks to the spin_lock()/spin_unlock() pair in inet_ehash_insert()
and reqsk_put(). The second lookup must find the new socket,
unless it has already been accepted and closed by another cpu.
Note that the fragmentation could be avoided in the first place by
use of a correct TCP MSS option in the SYN{ACK} packet, but this
does not mean we can not be more robust.
Many thanks to 배석진 for a very detailed analysis.
Reported-by: 배석진 <soukjin.bae@samsung.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This is the mindless scripted replacement of kernel use of POLL*
variables as described by Al, done by this script:
for V in IN OUT PRI ERR RDNORM RDBAND WRNORM WRBAND HUP RDHUP NVAL MSG; do
L=`git grep -l -w POLL$V | grep -v '^t' | grep -v /um/ | grep -v '^sa' | grep -v '/poll.h$'|grep -v '^D'`
for f in $L; do sed -i "-es/^\([^\"]*\)\(\<POLL$V\>\)/\\1E\\2/" $f; done
done
with de-mangling cleanups yet to come.
NOTE! On almost all architectures, the EPOLL* constants have the same
values as the POLL* constants do. But they keyword here is "almost".
For various bad reasons they aren't the same, and epoll() doesn't
actually work quite correctly in some cases due to this on Sparc et al.
The next patch from Al will sort out the final differences, and we
should be all done.
Scripted-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
A persistent connection may send tiny amount of data (e.g. health-check)
for a long period of time. BBR's windowed min RTT filter may only see
RTT samples from delayed ACKs causing BBR to grossly over-estimate
the path delay depending how much the ACK was delayed at the receiver.
This patch skips RTT samples that are likely coming from delayed ACKs. Note
that it is possible the sender never obtains a valid measure to set the
min RTT. In this case BBR will continue to set cwnd to initial window
which seems fine because the connection is thin stream.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Priyaranjan Jha <priyarjha@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch avoids having TCP sender or congestion control
overestimate the min RTT by orders of magnitude. This happens when
all the samples in the windowed filter are one-packet transfer
like small request and health-check like chit-chat, which is farily
common for applications using persistent connections. This patch
tries to conservatively labels and skip RTT samples obtained from
this type of workload.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This adds an event to trace TCP stat variables with
slightly intrusive trace-event. This uses ftrace/perf
event log buffer to trace those state, no needs to
prepare own ring-buffer, nor custom user apps.
User can use ftrace to trace this event as below;
# cd /sys/kernel/debug/tracing
# echo 1 > events/tcp/tcp_probe/enable
(run workloads)
# cat trace
Signed-off-by: Masami Hiramatsu <mhiramat@kernel.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
Three sets of overlapping changes, two in the packet scheduler
and one in the meson-gxl PHY driver.
Signed-off-by: David S. Miller <davem@davemloft.net>
When ms timestamp is used, current logic uses 1us in
tcp_rcv_rtt_update() when the real rcv_rtt is within 1 - 999us.
This could cause rcv_rtt underestimation.
Fix it by always using a min value of 1ms if ms timestamp is used.
Fixes: 645f4c6f2e ("tcp: switch rcv_rtt_est and rcvq_space to high resolution timestamps")
Signed-off-by: Wei Wang <weiwan@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Back in linux-3.13 (commit b0983d3c9b ("tcp: fix dynamic right sizing"))
I addressed the pressing issues we had with receiver autotuning.
But DRS suffers from extra latencies caused by rcv_rtt_est.rtt_us
drifts. One common problem happens during slow start, since the
apparent RTT measured by the receiver can be inflated by ~50%,
at the end of one packet train.
Also, a single drop can delay read() calls by one RTT, meaning
tcp_rcv_space_adjust() can be called one RTT too late.
By replacing the tri-modal heuristic with a continuous function,
we can offset the effects of not growing 'at the optimal time'.
The curve of the function matches prior behavior if the space
increased by 25% and 50% exactly.
Cost of added multiply/divide is small, considering a TCP flow
typically would run this part of the code few times in its life.
I tested this patch with 100 ms RTT / 1% loss link, 100 runs
of (netperf -l 5), and got an average throughput of 4600 Mbit
instead of 1700 Mbit.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Wei Wang <weiwan@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
When using large tcp_rmem[2] values (I did tests with 500 MB),
I noticed overflows while computing rcvwin.
Lets fix this before the following patch.
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Wei Wang <weiwan@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
While rcvbuf is properly clamped by tcp_rmem[2], rcvwin
is left to a potentially too big value.
It has no serious effect, since :
1) tcp_grow_window() has very strict checks.
2) window_clamp can be mangled by user space to any value anyway.
tcp_init_buffer_space() and companions use tcp_full_space(),
we use tcp_win_from_space() to avoid reloading sk->sk_rcvbuf
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Wei Wang <weiwan@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
When sender detects spurious retransmission, all packets
marked lost are remarked to be in-flight. However some may
be considered lost based on its timestamps in RACK. This patch
forces RACK to re-evaluate, which may be skipped previously if
the ACK does not advance RACK timestamp.
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Reviewed-by: Neal Cardwell <ncardwell@google.com>
Reviewed-by: Priyaranjan Jha <priyarjha@google.com>
Reviewed-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Mark tcp_sock during a SACK reneging event and invalidate rate samples
while marked. Such rate samples may overestimate bw by including packets
that were SACKed before reneging.
< ack 6001 win 10000 sack 7001:38001
< ack 7001 win 0 sack 8001:38001 // Reneg detected
> seq 7001:8001 // RTO, SACK cleared.
< ack 38001 win 10000
In above example the rate sample taken after the last ack will count
7001-38001 as delivered while the actual delivery rate likely could
be much lower i.e. 7001-8001.
This patch adds a new field tcp_sock.sack_reneg and marks it when we
declare SACK reneging and entering TCP_CA_Loss, and unmarks it after
the last rate sample was taken before moving back to TCP_CA_Open. This
patch also invalidates rate samples taken while tcp_sock.is_sack_reneg
is set.
Fixes: b9f64820fb ("tcp: track data delivery rate for a TCP connection")
Signed-off-by: Yousuk Seung <ysseung@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Eric Dumazet <edumazet@google.com>
Acked-by: Priyaranjan Jha <priyarjha@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
When I switched rcv_rtt_est to high resolution timestamps, I forgot
that tp->tcp_mstamp needed to be refreshed in tcp_rcv_space_adjust()
Using an old timestamp leads to autotuning lags.
Fixes: 645f4c6f2e ("tcp: switch rcv_rtt_est and rcvq_space to high resolution timestamps")
Signed-off-by: Eric Dumazet <edumazet@google.com>
Cc: Wei Wang <weiwan@google.com>
Cc: Neal Cardwell <ncardwell@google.com>
Cc: Yuchung Cheng <ycheng@google.com>
Acked-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Fix the TLP scheduling logic so that when scheduling a TLP probe, we
ensure that the estimated time at which an RTO would fire accounts for
the fact that ACKs indicating forward progress should push back RTO
times.
After the following fix:
df92c8394e ("tcp: fix xmit timer to only be reset if data ACKed/SACKed")
we had an unintentional behavior change in the following kind of
scenario: suppose the RTT variance has been very low recently. Then
suppose we send out a flight of N packets and our RTT is 100ms:
t=0: send a flight of N packets
t=100ms: receive an ACK for N-1 packets
The response before df92c8394e that was:
-> schedule a TLP for now + RTO_interval
The response after df92c8394e is:
-> schedule a TLP for t=0 + RTO_interval
Since RTO_interval = srtt + RTT_variance, this means that we have
scheduled a TLP timer at a point in the future that only accounts for
RTT_variance. If the RTT_variance term is small, this means that the
timer fires soon.
Before df92c8394e this would not happen, because in that code, when
we receive an ACK for a prefix of flight, we did:
1) Near the top of tcp_ack(), switch from TLP timer to RTO
at write_queue_head->paket_tx_time + RTO_interval:
if (icsk->icsk_pending == ICSK_TIME_LOSS_PROBE)
tcp_rearm_rto(sk);
2) In tcp_clean_rtx_queue(), update the RTO to now + RTO_interval:
if (flag & FLAG_ACKED) {
tcp_rearm_rto(sk);
3) In tcp_ack() after tcp_fastretrans_alert() switch from RTO
to TLP at now + RTO_interval:
if (icsk->icsk_pending == ICSK_TIME_RETRANS)
tcp_schedule_loss_probe(sk);
In df92c8394e we removed that 3-phase dance, and instead directly
set the TLP timer once: we set the TLP timer in cases like this to
write_queue_head->packet_tx_time + RTO_interval. So if the RTT
variance is small, then this means that this is setting the TLP timer
to fire quite soon. This means if the ACK for the tail of the flight
takes longer than an RTT to arrive (often due to delayed ACKs), then
the TLP timer fires too quickly.
Fixes: df92c8394e ("tcp: fix xmit timer to only be reset if data ACKed/SACKed")
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Acked-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Merge updates from Andrew Morton:
- a few misc bits
- ocfs2 updates
- almost all of MM
* emailed patches from Andrew Morton <akpm@linux-foundation.org>: (131 commits)
memory hotplug: fix comments when adding section
mm: make alloc_node_mem_map a void call if we don't have CONFIG_FLAT_NODE_MEM_MAP
mm: simplify nodemask printing
mm,oom_reaper: remove pointless kthread_run() error check
mm/page_ext.c: check if page_ext is not prepared
writeback: remove unused function parameter
mm: do not rely on preempt_count in print_vma_addr
mm, sparse: do not swamp log with huge vmemmap allocation failures
mm/hmm: remove redundant variable align_end
mm/list_lru.c: mark expected switch fall-through
mm/shmem.c: mark expected switch fall-through
mm/page_alloc.c: broken deferred calculation
mm: don't warn about allocations which stall for too long
fs: fuse: account fuse_inode slab memory as reclaimable
mm, page_alloc: fix potential false positive in __zone_watermark_ok
mm: mlock: remove lru_add_drain_all()
mm, sysctl: make NUMA stats configurable
shmem: convert shmem_init_inodecache() to void
Unify migrate_pages and move_pages access checks
mm, pagevec: rename pagevec drained field
...
Patch series "kmemcheck: kill kmemcheck", v2.
As discussed at LSF/MM, kill kmemcheck.
KASan is a replacement that is able to work without the limitation of
kmemcheck (single CPU, slow). KASan is already upstream.
We are also not aware of any users of kmemcheck (or users who don't
consider KASan as a suitable replacement).
The only objection was that since KASAN wasn't supported by all GCC
versions provided by distros at that time we should hold off for 2
years, and try again.
Now that 2 years have passed, and all distros provide gcc that supports
KASAN, kill kmemcheck again for the very same reasons.
This patch (of 4):
Remove kmemcheck annotations, and calls to kmemcheck from the kernel.
[alexander.levin@verizon.com: correctly remove kmemcheck call from dma_map_sg_attrs]
Link: http://lkml.kernel.org/r/20171012192151.26531-1-alexander.levin@verizon.com
Link: http://lkml.kernel.org/r/20171007030159.22241-2-alexander.levin@verizon.com
Signed-off-by: Sasha Levin <alexander.levin@verizon.com>
Cc: Alexander Potapenko <glider@google.com>
Cc: Eric W. Biederman <ebiederm@xmission.com>
Cc: Michal Hocko <mhocko@kernel.org>
Cc: Pekka Enberg <penberg@kernel.org>
Cc: Steven Rostedt <rostedt@goodmis.org>
Cc: Tim Hansen <devtimhansen@gmail.com>
Cc: Vegard Nossum <vegardno@ifi.uio.no>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
