Reduce the number of ACK-Requests we set on DATA packets that we're sending
to reduce network traffic.  We set the flag on odd-numbered DATA packets to
start off the RTT cache until we have at least three entries in it and then
probe once per second thereafter to keep it topped up.

This could be made tunable in future.

Note that from this point, the RXRPC_REQUEST_ACK flag is set on DATA
packets as we transmit them and not stored statically in the sk_buff.
Signed-off-by: NDavid Howells <dhowells@redhat.com>

In addition to sending a PING ACK to gain RTT data, we can set the
RXRPC_REQUEST_ACK flag on a DATA packet and get a REQUESTED-ACK ACK.  The
ACK packet contains the serial number of the packet it is in response to,
so we can look through the Tx buffer for a matching DATA packet.

This requires that the data packets be stamped with the time of
transmission as a ktime rather than having the resend_at time in jiffies.

This further requires the resend code to do the resend determination in
ktimes and convert to jiffies to set the timer.
Signed-off-by: NDavid Howells <dhowells@redhat.com>

Add a ktime_sub_ms() to go with ktime_add_ms() and co. for use in AF_RXRPC
RTT determination.
Signed-off-by: NDavid Howells <dhowells@redhat.com>

Expedite the transmission of a response to a PING ACK by sending it from
sendmsg if one is pending.  We're most likely to see a PING ACK during the
client call Tx phase as the other side may use it to determine a number of
parameters, such as the client's receive window size, the RTT and whether
the client is doing slow start (similar to RFC5681).

If we don't expedite it, it's left to the background processing thread to
transmit.
Signed-off-by: NDavid Howells <dhowells@redhat.com>

Send a PING ACK packet to the peer when we get a new incoming call from a
peer we don't have a record for.  The PING RESPONSE ACK packet will tell us
the following about the peer:

 (1) its receive window size

 (2) its MTU sizes

 (3) its support for jumbo DATA packets

 (4) if it supports slow start (similar to RFC 5681)

 (5) an estimate of the RTT

This is necessary because the peer won't normally send us an ACK until it
gets to the Rx phase and we send it a packet, but we would like to know
some of this information before we start sending packets.

A pair of tracepoints are added so that RTT determination can be observed.
Signed-off-by: NDavid Howells <dhowells@redhat.com>

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: NDavid Howells <dhowells@redhat.com>

Add a Tx-phase annotation for packet buffers to indicate that a buffer has
already been retransmitted.  This will be used by future congestion
management.  Re-retransmissions of a packet don't affect the congestion
window managment in the same way as initial retransmissions.
Signed-off-by: NDavid Howells <dhowells@redhat.com>

Don't store the rxrpc protocol header in sk_buffs on the transmit queue,
but rather generate it on the fly and pass it to kernel_sendmsg() as a
separate iov.  This reduces the amount of storage required.

Note that the security header is still stored in the sk_buff as it may get
encrypted along with the data (and doesn't change with each transmission).
Signed-off-by: NDavid Howells <dhowells@redhat.com>

Jiri Pirko says:

====================
mlxsw: Replace Hw related const with resource query results

Nogah says:

Many of the ASIC's properties can be read from the HW with resources query.
This patchset adds new resources to the resource query and implement
using them, instead of the constants that we currently use.
Those resources are lag, kvd and router related.
====================
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Replace max rif const with using the result from resource query.
Signed-off-by: NNogah Frankel <nogahf@mellanox.com>
Reviewed-by: NIdo Schimmel <idosch@mellanox.com>
Signed-off-by: NJiri Pirko <jiri@mellanox.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Add the max number of rif (router interfaces) to resource query.
Signed-off-by: NNogah Frankel <nogahf@mellanox.com>
Reviewed-by: NIdo Schimmel <idosch@mellanox.com>
Signed-off-by: NJiri Pirko <jiri@mellanox.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Add max system ports, max regions and max vlan groups to resource query.
Signed-off-by: NNogah Frankel <nogahf@mellanox.com>
Reviewed-by: NIdo Schimmel <idosch@mellanox.com>
Signed-off-by: NJiri Pirko <jiri@mellanox.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Replace max virtual routers const with the result from
the resource query.
Signed-off-by: NNogah Frankel <nogahf@mellanox.com>
Reviewed-by: NIdo Schimmel <idosch@mellanox.com>
Signed-off-by: NJiri Pirko <jiri@mellanox.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Add the max number of virtual routers to resource query.
Signed-off-by: NNogah Frankel <nogahf@mellanox.com>
Reviewed-by: NIdo Schimmel <idosch@mellanox.com>
Signed-off-by: NJiri Pirko <jiri@mellanox.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Use resources from resource query to determine values for
the profile configuration.
Add KVD determined section sizes to the resources struct.
Change the profile struct and value to match this changes.
Signed-off-by: NNogah Frankel <nogahf@mellanox.com>
Reviewed-by: NIdo Schimmel <idosch@mellanox.com>
Signed-off-by: NJiri Pirko <jiri@mellanox.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Add KVD size, and minimum sizes for the single and double
sections resources to resources query.
Signed-off-by: NNogah Frankel <nogahf@mellanox.com>
Reviewed-by: NIdo Schimmel <idosch@mellanox.com>
Signed-off-by: NJiri Pirko <jiri@mellanox.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Use max lag and max ports in lag resources as the result of resource query
instead of using const to save them.
Signed-off-by: NNogah Frankel <nogahf@mellanox.com>
Reviewed-by: NIdo Schimmel <idosch@mellanox.com>
Signed-off-by: NJiri Pirko <jiri@mellanox.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Add max lag and max ports in lag resources to resources query.
Signed-off-by: NNogah Frankel <nogahf@mellanox.com>
Reviewed-by: NIdo Schimmel <idosch@mellanox.com>
Signed-off-by: NJiri Pirko <jiri@mellanox.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The offloads stats functions are local to this file, make them static.

Fixes: fc1bbb0f ('mlxsw: spectrum: Implement offload stats ndo [..]')
Signed-off-by: NOr Gerlitz <ogerlitz@mellanox.com>
Acked-by: NJiri Pirko <jiri@mellanox.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Neal Cardwell says:

====================
tcp: BBR congestion control algorithm

This patch series implements a new TCP congestion control algorithm:
BBR (Bottleneck Bandwidth and RTT). A paper with a detailed
description of BBR will be published in ACM Queue, September-October
2016, as "BBR: Congestion-Based Congestion Control". BBR is widely
deployed in production at Google.

The patch series starts with a set of supporting infrastructure
changes, including a few that extend the congestion control
framework. The last patch adds BBR as a TCP congestion control
module. Please see individual patches for the details.

- v3 -> v4:
 - Updated tcp_bbr.c in "tcp_bbr: add BBR congestion control"
   to use const to qualify all the constant parameters.
   Thanks to Stephen Hemminger.
 - In "tcp_bbr: add BBR congestion control", remove the bbr_rate_kbps()
   function, which had a 64-bit divide that would be problematic on some
   architectures, and just use bbr_rate_bytes_per_sec() directly.
   Thanks to Kenneth Klette Jonassen for suggesting this.
 - In "tcp: switch back to proper tcp_skb_cb size check in tcp_init()",
   switched from sizeof(skb->cb) to FIELD_SIZEOF.
   Thanks to Lance Richardson for suggesting this.
 - Updated "tcp_bbr: add BBR congestion control" commit message with
   performance data, more details about deployment at Google, and
   another reminder to use fq with BBR.
 - Updated tcp_bbr.c in "tcp_bbr: add BBR congestion control"
   to use MODULE_LICENSE("Dual BSD/GPL").

- v2 -> v3: fix another issue caught by build bots:
 - adjust rate_sample struct initialization syntax to allow gcc-4.4 to compile
   the "tcp: track data delivery rate for a TCP connection" patch; also
   adjusted some similar syntax in "tcp_bbr: add BBR congestion control"

- v1 -> v2: fix issues caught by build bots:
 - fix "tcp: export data delivery rate" to use rate64 instead of rate,
   so there is a 64-bit numerator for the do_div call
 - fix conflicting definitions for minmax caused by
   "tcp: use windowed min filter library for TCP min_rtt estimation"
   with a new commit:
   tcp: cdg: rename struct minmax in tcp_cdg.c to avoid a naming conflict
 - fix warning about the use of __packed in
   "tcp: track data delivery rate for a TCP connection",
   which involves the addition of a new commit:
   tcp: switch back to proper tcp_skb_cb size check in tcp_init()
====================
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The TCP CUBIC module already uses 64 bytes.
The upcoming TCP BBR module uses 88 bytes.
Signed-off-by: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Acked-by: NStephen Hemminger <stephen@networkplumber.org>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Export tcp_mss_to_mtu(), so that congestion control modules can use
this to help calculate a pacing rate.
Signed-off-by: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Revert to the tcp_skb_cb size check that tcp_init() had before commit
b4772ef8 ("net: use common macro for assering skb->cb[] available
size in protocol families"). As related commit 744d5a3e ("net:
move skb->dropcount to skb->cb[]") explains, the
sock_skb_cb_check_size() mechanism was added to ensure that there is
space for dropcount, "for protocol families using it". But TCP is not
a protocol using dropcount, so tcp_init() doesn't need to provision
space for dropcount in the skb->cb[], and thus we can revert to the
older form of the tcp_skb_cb size check. Doing so allows TCP to use 4
more bytes of the skb->cb[] space.

Fixes: b4772ef8 ("net: use common macro for assering skb->cb[] available size in protocol families")
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

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: NVan Jacobson <vanj@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NNandita Dukkipati <nanditad@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

The upcoming change "lib/win_minmax: windowed min or max estimator"
introduces a struct called minmax, which is then included in
include/linux/tcp.h in the upcoming change "tcp: use windowed min
filter library for TCP min_rtt estimation". This would create a
compilation error for tcp_cdg.c, which defines its own minmax
struct. To avoid this naming conflict (and potentially others in the
future), this commit renames the version used in tcp_cdg.c to
cdg_minmax.
Signed-off-by: NSoheil Hassas Yeganeh <soheil@google.com>
Signed-off-by: NNeal Cardwell <ncardwell@google.com>
Signed-off-by: NYuchung Cheng <ycheng@google.com>
Signed-off-by: NEric Dumazet <edumazet@google.com>
Cc: Kenneth Klette Jonassen <kennetkl@ifi.uio.no>
Acked-by: NKenneth Klette Jonassen <kennetkl@ifi.uio.no>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

data_src is unchanged inside the loop, so this patch moves
the assignment to outside the loop to avoid unnecessarily
assignment
Signed-off-by: NSean Wang <sean.wang@mediatek.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

An address can be loaded in the ATU with multiple ports, for instance
when adding multiple ports to a Multicast group with "bridge mdb".

The current code doesn't allow that. Add an helper to get a single entry
from the ATU, then set or clear the requested port, before loading the
entry back in the ATU.

Note that the required _mv88e6xxx_atu_getnext function is defined below
mv88e6xxx_port_db_load_purge, so forward-declare it for the moment. The
ATU code will be isolated in future patches.

Fixes: 83dabd1f ("net: dsa: mv88e6xxx: make switchdev DB ops generic")
Signed-off-by: NVivien Didelot <vivien.didelot@savoirfairelinux.com>
Reviewed-by: NAndrew Lunn <andrew@lunn.ch>
Reviewed-by: NAndrew Lunn <andrew@lunn.ch>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

With the batch changes that translated transient actions into
a temporary list lost in the translation was the fact that
tcf_action_destroy() will eventually delete the action from
the permanent location if the refcount is zero.

Example of what broke:
...add a gact action to drop
sudo $TC actions add action drop index 10
...now retrieve it, looks good
sudo $TC actions get action gact index 10
...retrieve it again and find it is gone!
sudo $TC actions get action gact index 10

Fixes: 22dc13c8 ("net_sched: convert tcf_exts from list to pointer array"),
Fixes: 824a7e88 ("net_sched: remove an unnecessary list_del()")
Fixes: f07fed82 ("net_sched: remove the leftover cleanup_a()")
Acked-by: NCong Wang <xiyou.wangcong@gmail.com>
Signed-off-by: NJamal Hadi Salim <jhs@mojatatu.com>
Signed-off-by: NDavid S. Miller <davem@davemloft.net>

Daniel Borkmann says:

====================
BPF direct packet access improvements

This set adds write support to the currently available read support
for {cls,act}_bpf programs. First one is a fix for affected commit
sitting in net-next and prerequisite for the second one, last patch
adds a number of test cases against the verifier. For details, please
see individual patches.
====================
Signed-off-by: NDavid S. Miller <davem@davemloft.net>