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git://git.kernel.org/pub/scm/linux/kernel/git/dhowells/linux-fs
David Howells says:
====================
rxrpc: Preparation for slow-start algorithm [ver #2]
Here are some patches that prepare for improvements in ACK generation and
for the implementation of the slow-start part of the protocol:
(1) Stop storing the protocol header in the Tx socket buffers, but rather
generate it on the fly. This potentially saves a little space and
makes it easier to alter the header just before transmission (the
flags may get altered and the serial number has to be changed).
(2) Mask off the Tx buffer annotations and add a flag to record which ones
have already been resent.
(3) Track RTT on a per-peer basis for use in future changes. Tracepoints
are added to log this.
(4) Send PING ACKs in response to incoming calls to elicit a PING-RESPONSE
ACK from which RTT data can be calculated. The response also carries
other useful information.
(5) Expedite PING-RESPONSE ACK generation from sendmsg. If we're actively
using sendmsg, this allows us, under some circumstances, to avoid
having to rely on the background work item to run to generate this
ACK.
This requires ktime_sub_ms() to be added.
(6) Set the REQUEST-ACK flag on some DATA packets to elicit ACK-REQUESTED
ACKs from which RTT data can be calculated.
(7) Limit the use of pings and ACK requests for RTT determination.
Changes:
(V2) Don't use the C division operator for 64-bit division. One instance
should use do_div() and the other should be using nsecs_to_jiffies().
The last two patches got transposed, leading to an undefined symbol
in one of them.
Reported-by: kbuild test robot <lkp@intel.com>
====================
Signed-off-by: David S. Miller <davem@davemloft.net>
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Add a ktime_sub_ms() to go with ktime_add_ms() and co. for use in AF_RXRPC
RTT determination.
Signed-off-by: David Howells <dhowells@redhat.com>
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Add a function to track the average RTT for a peer. Sources of RTT data
will be added in subsequent patches.
The RTT data will be useful in the future for determining resend timeouts
and for handling the slow-start part of the Rx protocol.
Also add a pair of tracepoints, one to log transmissions to elicit a
response for RTT purposes and one to log responses that contribute RTT
data.
Signed-off-by: David Howells <dhowells@redhat.com>
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To something more meaningful these days, specially because this is
working on packet headers or lengths and which are not tied to any CPU
arch but to the protocol itself.
So, WORD_TRUNC becomes SCTP_TRUNC4 and WORD_ROUND becomes SCTP_PAD4.
Reported-by: David Laight <David.Laight@ACULAB.COM>
Reported-by: David Miller <davem@davemloft.net>
Signed-off-by: Marcelo Ricardo Leitner <marcelo.leitner@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Drivers must be ready to accept NULL from ptp_clock_register() if the
PTP clock subsystem is configured out.
This patch documents that and ensures that all drivers cope well
with a NULL return.
Signed-off-by: Nicolas Pitre <nico@linaro.org>
Reviewed-by: Eugenia Emantayev <eugenia@mellanox.com>
Acked-by: Richard Cochran <richardcochran@gmail.com>
Acked-by: Edward Cree <ecree@solarflare.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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TCA_VLAN_ACT_MODIFY allows one to change an existing tag.
It accepts same attributes as TCA_VLAN_ACT_PUSH (protocol, id,
priority).
If packet is vlan tagged, then the tag gets overwritten according to
user specified attributes.
For example, this allows user to replace a tag's vid while preserving
its priority bits (as opposed to "action vlan pop pipe action vlan push").
Signed-off-by: Shmulik Ladkani <shmulik.ladkani@gmail.com>
Acked-by: Jamal Hadi Salim <jhs@mojatatu.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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This exports the functionality of extracting the tag from the payload,
without moving next vlan tag into hw accel tag.
Signed-off-by: Shmulik Ladkani <shmulik.ladkani@gmail.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Call into offloaded filters to update stats.
Signed-off-by: Jakub Kicinski <jakub.kicinski@netronome.com>
Acked-by: Daniel Borkmann <daniel@iogearbox.net>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Advanced JIT compilers and translators may want to use
eBPF verifier as a base for parsers or to perform custom
checks and validations.
Add ability for external users to invoke the verifier
and provide callbacks to be invoked for every intruction
checked. For now only add most basic callback for
per-instruction pre-interpretation checks is added. More
advanced users may also like to have per-instruction post
callback and state comparison callback.
Signed-off-by: Jakub Kicinski <jakub.kicinski@netronome.com>
Acked-by: Alexei Starovoitov <ast@kernel.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Move verifier's internal structures to a header file and
prefix their names with bpf_ to avoid potential namespace
conflicts. Those structures will soon be used by external
analyzers.
Signed-off-by: Jakub Kicinski <jakub.kicinski@netronome.com>
Acked-by: Alexei Starovoitov <ast@kernel.org>
Acked-by: Daniel Borkmann <daniel@iogearbox.net>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Add cls_bpf support for the TCA_CLS_FLAGS_SKIP_HW flag.
Unlike U32 and flower cls_bpf already has some netlink
flags defined. Create a new attribute to be able to use
the same flag values as the above.
Unlike U32 and flower reject unknown flags.
Signed-off-by: Jakub Kicinski <jakub.kicinski@netronome.com>
Acked-by: Daniel Borkmann <daniel@iogearbox.net>
Signed-off-by: David S. Miller <davem@davemloft.net>
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This patch adds hardware offload capability to cls_bpf classifier,
similar to what have been done with U32 and flower.
Signed-off-by: Jakub Kicinski <jakub.kicinski@netronome.com>
Acked-by: Daniel Borkmann <daniel@iogearbox.net>
Signed-off-by: David S. Miller <davem@davemloft.net>
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This commit implements a new TCP congestion control algorithm: BBR
(Bottleneck Bandwidth and RTT). A detailed description of BBR will be
published in ACM Queue, Vol. 14 No. 5, September-October 2016, as
"BBR: Congestion-Based Congestion Control".
BBR has significantly increased throughput and reduced latency for
connections on Google's internal backbone networks and google.com and
YouTube Web servers.
BBR requires only changes on the sender side, not in the network or
the receiver side. Thus it can be incrementally deployed on today's
Internet, or in datacenters.
The Internet has predominantly used loss-based congestion control
(largely Reno or CUBIC) since the 1980s, relying on packet loss as the
signal to slow down. While this worked well for many years, loss-based
congestion control is unfortunately out-dated in today's networks. On
today's Internet, loss-based congestion control causes the infamous
bufferbloat problem, often causing seconds of needless queuing delay,
since it fills the bloated buffers in many last-mile links. On today's
high-speed long-haul links using commodity switches with shallow
buffers, loss-based congestion control has abysmal throughput because
it over-reacts to losses caused by transient traffic bursts.
In 1981 Kleinrock and Gale showed that the optimal operating point for
a network maximizes delivered bandwidth while minimizing delay and
loss, not only for single connections but for the network as a
whole. Finding that optimal operating point has been elusive, since
any single network measurement is ambiguous: network measurements are
the result of both bandwidth and propagation delay, and those two
cannot be measured simultaneously.
While it is impossible to disambiguate any single bandwidth or RTT
measurement, a connection's behavior over time tells a clearer
story. BBR uses a measurement strategy designed to resolve this
ambiguity. It combines these measurements with a robust servo loop
using recent control systems advances to implement a distributed
congestion control algorithm that reacts to actual congestion, not
packet loss or transient queue delay, and is designed to converge with
high probability to a point near the optimal operating point.
In a nutshell, BBR creates an explicit model of the network pipe by
sequentially probing the bottleneck bandwidth and RTT. On the arrival
of each ACK, BBR derives the current delivery rate of the last round
trip, and feeds it through a windowed max-filter to estimate the
bottleneck bandwidth. Conversely it uses a windowed min-filter to
estimate the round trip propagation delay. The max-filtered bandwidth
and min-filtered RTT estimates form BBR's model of the network pipe.
Using its model, BBR sets control parameters to govern sending
behavior. The primary control is the pacing rate: BBR applies a gain
multiplier to transmit faster or slower than the observed bottleneck
bandwidth. The conventional congestion window (cwnd) is now the
secondary control; the cwnd is set to a small multiple of the
estimated BDP (bandwidth-delay product) in order to allow full
utilization and bandwidth probing while bounding the potential amount
of queue at the bottleneck.
When a BBR connection starts, it enters STARTUP mode and applies a
high gain to perform an exponential search to quickly probe the
bottleneck bandwidth (doubling its sending rate each round trip, like
slow start). However, instead of continuing until it fills up the
buffer (i.e. a loss), or until delay or ACK spacing reaches some
threshold (like Hystart), it uses its model of the pipe to estimate
when that pipe is full: it estimates the pipe is full when it notices
the estimated bandwidth has stopped growing. At that point it exits
STARTUP and enters DRAIN mode, where it reduces its pacing rate to
drain the queue it estimates it has created.
Then BBR enters steady state. In steady state, PROBE_BW mode cycles
between first pacing faster to probe for more bandwidth, then pacing
slower to drain any queue that created if no more bandwidth was
available, and then cruising at the estimated bandwidth to utilize the
pipe without creating excess queue. Occasionally, on an as-needed
basis, it sends significantly slower to probe for RTT (PROBE_RTT
mode).
BBR has been fully deployed on Google's wide-area backbone networks
and we're experimenting with BBR on Google.com and YouTube on a global
scale. Replacing CUBIC with BBR has resulted in significant
improvements in network latency and application (RPC, browser, and
video) metrics. For more details please refer to our upcoming ACM
Queue publication.
Example performance results, to illustrate the difference between BBR
and CUBIC:
Resilience to random loss (e.g. from shallow buffers):
Consider a netperf TCP_STREAM test lasting 30 secs on an emulated
path with a 10Gbps bottleneck, 100ms RTT, and 1% packet loss
rate. CUBIC gets 3.27 Mbps, and BBR gets 9150 Mbps (2798x higher).
Low latency with the bloated buffers common in today's last-mile links:
Consider a netperf TCP_STREAM test lasting 120 secs on an emulated
path with a 10Mbps bottleneck, 40ms RTT, and 1000-packet bottleneck
buffer. Both fully utilize the bottleneck bandwidth, but BBR
achieves this with a median RTT 25x lower (43 ms instead of 1.09
secs).
Our long-term goal is to improve the congestion control algorithms
used on the Internet. We are hopeful that BBR can help advance the
efforts toward this goal, and motivate the community to do further
research.
Test results, performance evaluations, feedback, and BBR-related
discussions are very welcome in the public e-mail list for BBR:
https://groups.google.com/forum/#!forum/bbr-dev
NOTE: BBR *must* be used with the fq qdisc ("man tc-fq") with pacing
enabled, since pacing is integral to the BBR design and
implementation. BBR without pacing would not function properly, and
may incur unnecessary high packet loss rates.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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The TCP CUBIC module already uses 64 bytes.
The upcoming TCP BBR module uses 88 bytes.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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This commit introduces an optional new "omnipotent" hook,
cong_control(), for congestion control modules. The cong_control()
function is called at the end of processing an ACK (i.e., after
updating sequence numbers, the SACK scoreboard, and loss
detection). At that moment we have precise delivery rate information
the congestion control module can use to control the sending behavior
(using cwnd, TSO skb size, and pacing rate) in any CA state.
This function can also be used by a congestion control that prefers
not to use the default cwnd reduction approach (i.e., the PRR
algorithm) during CA_Recovery to control the cwnd and sending rate
during loss recovery.
We take advantage of the fact that recent changes defer the
retransmission or transmission of new data (e.g. by F-RTO) in recovery
until the new tcp_cong_control() function is run.
With this commit, we only run tcp_update_pacing_rate() if the
congestion control is not using this new API. New congestion controls
which use the new API do not want the TCP stack to run the default
pacing rate calculation and overwrite whatever pacing rate they have
chosen at initialization time.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Currently the TCP send buffer expands to twice cwnd, in order to allow
limited transmits in the CA_Recovery state. This assumes that cwnd
does not increase in the CA_Recovery.
For some congestion control algorithms, like the upcoming BBR module,
if the losses in recovery do not indicate congestion then we may
continue to raise cwnd multiplicatively in recovery. In such cases the
current multiplier will falsely limit the sending rate, much as if it
were limited by the application.
This commit adds an optional congestion control callback to use a
different multiplier to expand the TCP send buffer. For congestion
control modules that do not specificy this callback, TCP continues to
use the previous default of 2.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Acked-by: Stephen Hemminger <stephen@networkplumber.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
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To allow congestion control modules to use the default TSO auto-sizing
algorithm as one of the ingredients in their own decision about TSO sizing:
1) Export tcp_tso_autosize() so that CC modules can use it.
2) Change tcp_tso_autosize() to allow callers to specify a minimum
number of segments per TSO skb, in case the congestion control
module has a different notion of the best floor for TSO skbs for
the connection right now. For very low-rate paths or policed
connections it can be appropriate to use smaller TSO skbs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Add the tso_segs_goal() function in tcp_congestion_ops to allow the
congestion control module to specify the number of segments that
should be in a TSO skb sent by tcp_write_xmit() and
tcp_xmit_retransmit_queue(). The congestion control module can either
request a particular number of segments in TSO skb that we transmit,
or return 0 if it doesn't care.
This allows the upcoming BBR congestion control module to select small
TSO skb sizes if the module detects that the bottleneck bandwidth is
very low, or that the connection is policed to a low rate.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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This commit export two new fields in struct tcp_info:
tcpi_delivery_rate: The most recent goodput, as measured by
tcp_rate_gen(). If the socket is limited by the sending
application (e.g., no data to send), it reports the highest
measurement instead of the most recent. The unit is bytes per
second (like other rate fields in tcp_info).
tcpi_delivery_rate_app_limited: A boolean indicating if the goodput
was measured when the socket's throughput was limited by the
sending application.
This delivery rate information can be useful for applications that
want to know the current throughput the TCP connection is seeing,
e.g. adaptive bitrate video streaming. It can also be very useful for
debugging or troubleshooting.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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This commit adds code to track whether the delivery rate represented
by each rate_sample was limited by the application.
Upon each transmit, we store in the is_app_limited field in the skb a
boolean bit indicating whether there is a known "bubble in the pipe":
a point in the rate sample interval where the sender was
application-limited, and did not transmit even though the cwnd and
pacing rate allowed it.
This logic marks the flow app-limited on a write if *all* of the
following are true:
1) There is less than 1 MSS of unsent data in the write queue
available to transmit.
2) There is no packet in the sender's queues (e.g. in fq or the NIC
tx queue).
3) The connection is not limited by cwnd.
4) There are no lost packets to retransmit.
The tcp_rate_check_app_limited() code in tcp_rate.c determines whether
the connection is application-limited at the moment. If the flow is
application-limited, it sets the tp->app_limited field. If the flow is
application-limited then that means there is effectively a "bubble" of
silence in the pipe now, and this silence will be reflected in a lower
bandwidth sample for any rate samples from now until we get an ACK
indicating this bubble has exited the pipe: specifically, until we get
an ACK for the next packet we transmit.
When we send every skb we record in scb->tx.is_app_limited whether the
resulting rate sample will be application-limited.
The code in tcp_rate_gen() checks to see when it is safe to mark all
known application-limited bubbles of silence as having exited the
pipe. It does this by checking to see when the delivered count moves
past the tp->app_limited marker. At this point it zeroes the
tp->app_limited marker, as all known bubbles are out of the pipe.
We make room for the tx.is_app_limited bit in the skb by borrowing a
bit from the in_flight field used by NV to record the number of bytes
in flight. The receive window in the TCP header is 16 bits, and the
max receive window scaling shift factor is 14 (RFC 1323). So the max
receive window offered by the TCP protocol is 2^(16+14) = 2^30. So we
only need 30 bits for the tx.in_flight used by NV.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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This patch generates data delivery rate (throughput) samples on a
per-ACK basis. These rate samples can be used by congestion control
modules, and specifically will be used by TCP BBR in later patches in
this series.
Key state:
tp->delivered: Tracks the total number of data packets (original or not)
delivered so far. This is an already-existing field.
tp->delivered_mstamp: the last time tp->delivered was updated.
Algorithm:
A rate sample is calculated as (d1 - d0)/(t1 - t0) on a per-ACK basis:
d1: the current tp->delivered after processing the ACK
t1: the current time after processing the ACK
d0: the prior tp->delivered when the acked skb was transmitted
t0: the prior tp->delivered_mstamp when the acked skb was transmitted
When an skb is transmitted, we snapshot d0 and t0 in its control
block in tcp_rate_skb_sent().
When an ACK arrives, it may SACK and ACK some skbs. For each SACKed
or ACKed skb, tcp_rate_skb_delivered() updates the rate_sample struct
to reflect the latest (d0, t0).
Finally, tcp_rate_gen() generates a rate sample by storing
(d1 - d0) in rs->delivered and (t1 - t0) in rs->interval_us.
One caveat: if an skb was sent with no packets in flight, then
tp->delivered_mstamp may be either invalid (if the connection is
starting) or outdated (if the connection was idle). In that case,
we'll re-stamp tp->delivered_mstamp.
At first glance it seems t0 should always be the time when an skb was
transmitted, but actually this could over-estimate the rate due to
phase mismatch between transmit and ACK events. To track the delivery
rate, we ensure that if packets are in flight then t0 and and t1 are
times at which packets were marked delivered.
If the initial and final RTTs are different then one may be corrupted
by some sort of noise. The noise we see most often is sending gaps
caused by delayed, compressed, or stretched acks. This either affects
both RTTs equally or artificially reduces the final RTT. We approach
this by recording the info we need to compute the initial RTT
(duration of the "send phase" of the window) when we recorded the
associated inflight. Then, for a filter to avoid bandwidth
overestimates, we generalize the per-sample bandwidth computation
from:
bw = delivered / ack_phase_rtt
to the following:
bw = delivered / max(send_phase_rtt, ack_phase_rtt)
In large-scale experiments, this filtering approach incorporating
send_phase_rtt is effective at avoiding bandwidth overestimates due to
ACK compression or stretched ACKs.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Count the number of packets that a TCP connection marks lost.
Congestion control modules can use this loss rate information for more
intelligent decisions about how fast to send.
Specifically, this is used in TCP BBR policer detection. BBR uses a
high packet loss rate as one signal in its policer detection and
policer bandwidth estimation algorithm.
The BBR policer detection algorithm cannot simply track retransmits,
because a retransmit can be (and often is) an indicator of packets
lost long, long ago. This is particularly true in a long CA_Loss
period that repairs the initial massive losses when a policer kicks
in.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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This commit adds to the fq module a low_rate_threshold parameter to
insert a delay after all packets if the socket requests a pacing rate
below the threshold.
This helps achieve more precise control of the sending rate with
low-rate paths, especially policers. The basic issue is that if a
congestion control module detects a policer at a certain rate, it may
want fq to be able to shape to that policed rate. That way the sender
can avoid policer drops by having the packets arrive at the policer at
or just under the policed rate.
The default threshold of 550Kbps was chosen analytically so that for
policers or links at 500Kbps or 512Kbps fq would very likely invoke
this mechanism, even if the pacing rate was briefly slightly above the
available bandwidth. This value was then empirically validated with
two years of production testing on YouTube video servers.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Refactor the TCP min_rtt code to reuse the new win_minmax library in
lib/win_minmax.c to simplify the TCP code.
This is a pure refactor: the functionality is exactly the same. We
just moved the windowed min code to make TCP easier to read and
maintain, and to allow other parts of the kernel to use the windowed
min/max filter code.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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This commit introduces a generic library to estimate either the min or
max value of a time-varying variable over a recent time window. This
is code originally from Kathleen Nichols. The current form of the code
is from Van Jacobson.
A single struct minmax_sample will track the estimated windowed-max
value of the series if you call minmax_running_max() or the estimated
windowed-min value of the series if you call minmax_running_min().
Nearly equivalent code is already in place for minimum RTT estimation
in the TCP stack. This commit extracts that code and generalizes it to
handle both min and max. Moving the code here reduces the footprint
and complexity of the TCP code base and makes the filter generally
available for other parts of the codebase, including an upcoming TCP
congestion control module.
This library works well for time series where the measurements are
smoothly increasing or decreasing.
Signed-off-by: Van Jacobson <vanj@google.com>
Signed-off-by: Neal Cardwell <ncardwell@google.com>
Signed-off-by: Yuchung Cheng <ycheng@google.com>
Signed-off-by: Nandita Dukkipati <nanditad@google.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: Soheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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This work implements direct packet access for helpers and direct packet
write in a similar fashion as already available for XDP types via commits
4acf6c0b84c9 ("bpf: enable direct packet data write for xdp progs") and
6841de8b0d03 ("bpf: allow helpers access the packet directly"), and as a
complementary feature to the already available direct packet read for tc
(cls/act) programs.
For enabling this, we need to introduce two helpers, bpf_skb_pull_data()
and bpf_csum_update(). The first is generally needed for both, read and
write, because they would otherwise only be limited to the current linear
skb head. Usually, when the data_end test fails, programs just bail out,
or, in the direct read case, use bpf_skb_load_bytes() as an alternative
to overcome this limitation. If such data sits in non-linear parts, we
can just pull them in once with the new helper, retest and eventually
access them.
At the same time, this also makes sure the skb is uncloned, which is, of
course, a necessary condition for direct write. As this needs to be an
invariant for the write part only, the verifier detects writes and adds
a prologue that is calling bpf_skb_pull_data() to effectively unclone the
skb from the very beginning in case it is indeed cloned. The heuristic
makes use of a similar trick that was done in 233577a22089 ("net: filter:
constify detection of pkt_type_offset"). This comes at zero cost for other
programs that do not use the direct write feature. Should a program use
this feature only sparsely and has read access for the most parts with,
for example, drop return codes, then such write action can be delegated
to a tail called program for mitigating this cost of potential uncloning
to a late point in time where it would have been paid similarly with the
bpf_skb_store_bytes() as well. Advantage of direct write is that the
writes are inlined whereas the helper cannot make any length assumptions
and thus needs to generate a call to memcpy() also for small sizes, as well
as cost of helper call itself with sanity checks are avoided. Plus, when
direct read is already used, we don't need to cache or perform rechecks
on the data boundaries (due to verifier invalidating previous checks for
helpers that change skb->data), so more complex programs using rewrites
can benefit from switching to direct read plus write.
For direct packet access to helpers, we save the otherwise needed copy into
a temp struct sitting on stack memory when use-case allows. Both facilities
are enabled via may_access_direct_pkt_data() in verifier. For now, we limit
this to map helpers and csum_diff, and can successively enable other helpers
where we find it makes sense. Helpers that definitely cannot be allowed for
this are those part of bpf_helper_changes_skb_data() since they can change
underlying data, and those that write into memory as this could happen for
packet typed args when still cloned. bpf_csum_update() helper accommodates
for the fact that we need to fixup checksum_complete when using direct write
instead of bpf_skb_store_bytes(), meaning the programs can use available
helpers like bpf_csum_diff(), and implement csum_add(), csum_sub(),
csum_block_add(), csum_block_sub() equivalents in eBPF together with the
new helper. A usage example will be provided for iproute2's examples/bpf/
directory.
Signed-off-by: Daniel Borkmann <daniel@iogearbox.net>
Acked-by: Alexei Starovoitov <ast@kernel.org>
Signed-off-by: David S. Miller <davem@davemloft.net>
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git://git.kernel.org/pub/scm/linux/kernel/git/bluetooth/bluetooth-next
Johan Hedberg says:
====================
pull request: bluetooth-next 2016-09-19
Here's the main bluetooth-next pull request for the 4.9 kernel.
- Added new messages for monitor sockets for better mgmt tracing
- Added local name and appearance support in scan response
- Added new Qualcomm WCNSS SMD based HCI driver
- Minor fixes & cleanup to 802.15.4 code
- New USB ID to btusb driver
- Added Marvell support to HCI UART driver
- Add combined LED trigger for controller power
- Other minor fixes here and there
Please let me know if there are any issues pulling. Thanks.
====================
Signed-off-by: David S. Miller <davem@davemloft.net>
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This patch enables prepending appearance value to scan response data.
It also adds support for setting appearance value through mgmt command.
If currently advertised instance has apperance flag set it is expired
immediately.
Signed-off-by: Michał Narajowski <michal.narajowski@codecoup.pl>
Signed-off-by: Szymon Janc <szymon.janc@codecoup.pl>
Signed-off-by: Marcel Holtmann <marcel@holtmann.org>
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While the subsystem version information are purely informational,
increase the minor number due to the addition of user channel and
management control monitoring suppport. It is helpful for debugging
purposes to see the version numbers change.
Signed-off-by: Marcel Holtmann <marcel@holtmann.org>
Signed-off-by: Johan Hedberg <johan.hedberg@intel.com>
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This command is used to retrieve the current state and basic
information of a controller. It is typically used right after
getting the response to the Read Controller Index List command
or an Index Added event (or its extended counterparts).
When any of the values in the EIR_Data field changes, the event
Extended Controller Information Changed will be used to inform
clients about the updated information.
Signed-off-by: Marcel Holtmann <marcel@holtmann.org>
Signed-off-by: Michał Narajowski <michal.narajowski@codecoup.pl>
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Instead of keeping a version string around, use version and revision
numbers and then stringify them for use as module parameter.
Signed-off-by: Marcel Holtmann <marcel@holtmann.org>
Signed-off-by: Johan Hedberg <johan.hedberg@intel.com>
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Instead of hiding everything behind a general managment events flag,
introduce indivdual flags that allow fine control over which events are
send to a given management channel.
Signed-off-by: Marcel Holtmann <marcel@holtmann.org>
Signed-off-by: Johan Hedberg <johan.hedberg@intel.com>
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This adds support for tracing all management commands and events via the
monitor interface.
Signed-off-by: Marcel Holtmann <marcel@holtmann.org>
Signed-off-by: Johan Hedberg <johan.hedberg@intel.com>
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This sends new notifications to the monitor support whenever a
management channel has been opened or closed. This allows tracing of
control channels really easily.
Signed-off-by: Marcel Holtmann <marcel@holtmann.org>
Signed-off-by: Johan Hedberg <johan.hedberg@intel.com>
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The mgmt version information will be also needed for the control
changell tracing feature. This provides a helper to pack them.
Signed-off-by: Marcel Holtmann <marcel@holtmann.org>
Signed-off-by: Johan Hedberg <johan.hedberg@intel.com>
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To further allow unique identification and tracking of control socket,
store cookie and comm information when binding the socket.
Signed-off-by: Marcel Holtmann <marcel@holtmann.org>
Signed-off-by: Johan Hedberg <johan.hedberg@intel.com>
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Commit 5177a83827cd ("Bluetooth: Add debugfs fields for hardware and
firmware info") introduced hci_set_hw_info() and hci_set_fw_info().
These functions use kvasprintf_const() but are not marked with a
__printf attribute. Adding such an attribute helps detecting issues
related to printf-formatting at build time.
Signed-off-by: Nicolas Iooss <nicolas.iooss_linux@m4x.org>
Signed-off-by: Johan Hedberg <johan.hedberg@intel.com>
Signed-off-by: Marcel Holtmann <marcel@holtmann.org>
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This patch assigns the next free HCI device identifier to Bluetooth
devices based on the Qualcomm Shared Memory channels.
Signed-off-by: Bjorn Andersson <bjorn.andersson@sonymobile.com>
Signed-off-by: Bjorn Andersson <bjorn.andersson@linaro.org>
Signed-off-by: Marcel Holtmann <marcel@holtmann.org>
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The led_trigger field in hci_dev should be conditional based on if
CONFIG_BT_LEDS is set or not.
Signed-off-by: Marcel Holtmann <marcel@holtmann.org>
Signed-off-by: Johan Hedberg <johan.hedberg@intel.com>
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The insecure_elasticity setting is an ugly wart brought out by
users who need to insert duplicate objects (that is, distinct
objects with identical keys) into the same table.
In fact, those users have a much bigger problem. Once those
duplicate objects are inserted, they don't have an interface to
find them (unless you count the walker interface which walks
over the entire table).
Some users have resorted to doing a manual walk over the hash
table which is of course broken because they don't handle the
potential existence of multiple hash tables. The result is that
they will break sporadically when they encounter a hash table
resize/rehash.
This patch provides a way out for those users, at the expense
of an extra pointer per object. Essentially each object is now
a list of objects carrying the same key. The hash table will
only see the lists so nothing changes as far as rhashtable is
concerned.
To use this new interface, you need to insert a struct rhlist_head
into your objects instead of struct rhash_head. While the hash
table is unchanged, for type-safety you'll need to use struct
rhltable instead of struct rhashtable. All the existing interfaces
have been duplicated for rhlist, including the hash table walker.
One missing feature is nulls marking because AFAIK the only potential
user of it does not need duplicate objects. Should anyone need
this it shouldn't be too hard to add.
Signed-off-by: Herbert Xu <herbert@gondor.apana.org.au>
Acked-by: Thomas Graf <tgraf@suug.ch>
Signed-off-by: David S. Miller <davem@davemloft.net>
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Sample use case of how this is encoded:
user space via tuntap (or a connected VM/Machine/container)
encodes the tcindex TLV.
Sample use case of decoding:
IFE action decodes it and the skb->tc_index is then used to classify.
So something like this for encoded ICMP packets:
.. first decode then reclassify... skb->tcindex will be set
sudo $TC filter add dev $ETH parent ffff: prio 2 protocol 0xbeef \
u32 match u32 0 0 flowid 1:1 \
action ife decode reclassify
...next match the decode icmp packet...
sudo $TC filter add dev $ETH parent ffff: prio 4 protocol ip \
u32 match ip protocol 1 0xff flowid 1:1 \
action continue
... last classify it using the tcindex classifier and do someaction..
sudo $TC filter add dev $ETH parent ffff: prio 5 protocol ip \
handle 0x11 tcindex classid 1:1 \
action blah..
Signed-off-by: Jamal Hadi Salim <jhs@mojatatu.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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encoder and checker for 16 bits metadata
Signed-off-by: Jamal Hadi Salim <jhs@mojatatu.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
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git://git.kernel.org/pub/scm/linux/kernel/git/dhowells/linux-fs
David Howells says:
====================
rxrpc: Tracepoint addition and improvement
Here is a set of patches that add some more tracepoints and improve a couple
of existing ones. New additions include:
(1) Connection refcount tracking.
(2) Client connection state machine tracking.
(3) Tx and Rx packet lifecycle.
(4) ACK reception and transmission.
(5) recvmsg processing.
Updates include:
(1) Print the symbolic packet name in the Rx packet tracepoint.
(2) Additional call refcount trace events.
(3) Improvements to sk_buff tracking with AF_RXRPC.
In addition:
(1) Config option to inject packet loss during both transmission and
reception.
(2) Removal of some printks.
This series needs to be applied on top of the previously posted fixes.
====================
Signed-off-by: David S. Miller <davem@davemloft.net>
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Improve sk_buff tracing within AF_RXRPC by the following means:
(1) Use an enum to note the event type rather than plain integers and use
an array of event names rather than a big multi ?: list.
(2) Distinguish Rx from Tx packets and account them separately. This
requires the call phase to be tracked so that we know what we might
find in rxtx_buffer[].
(3) Add a parameter to rxrpc_{new,see,get,free}_skb() to indicate the
event type.
(4) A pair of 'rotate' events are added to indicate packets that are about
to be rotated out of the Rx and Tx windows.
(5) A pair of 'lost' events are added, along with rxrpc_lose_skb() for
packet loss injection recording.
Signed-off-by: David Howells <dhowells@redhat.com>
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Add a tracepoint to follow what recvmsg does within AF_RXRPC.
Signed-off-by: David Howells <dhowells@redhat.com>
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Add a tracepoint to follow the life of packets that get added to a call's
receive buffer.
Signed-off-by: David Howells <dhowells@redhat.com>
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Add a tracepoint to log information about ACK transmission.
Signed-off-by: David Howels <dhowells@redhat.com>
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Add a tracepoint to log information from received ACK packets.
Signed-off-by: David Howells <dhowells@redhat.com>
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Add a tracepoint to follow the insertion of a packet into the transmit
buffer, its transmission and its rotation out of the buffer.
Signed-off-by: David Howells <dhowells@redhat.com>
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Add a pair of tracepoints, one to track rxrpc_connection struct ref
counting and the other to track the client connection cache state.
Signed-off-by: David Howells <dhowells@redhat.com>
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