tcp_input.c

/*
 * INET		An implementation of the TCP/IP protocol suite for the LINUX
 *		operating system.  INET is implemented using the  BSD Socket
 *		interface as the means of communication with the user level.
 *
 *		Implementation of the Transmission Control Protocol(TCP).
 *
 * Authors:	Ross Biro
 *		Fred N. van Kempen, <waltje@uWalt.NL.Mugnet.ORG>
 *		Mark Evans, <evansmp@uhura.aston.ac.uk>
 *		Corey Minyard <wf-rch!minyard@relay.EU.net>
 *		Florian La Roche, <flla@stud.uni-sb.de>
 *		Charles Hedrick, <hedrick@klinzhai.rutgers.edu>
 *		Linus Torvalds, <torvalds@cs.helsinki.fi>
 *		Alan Cox, <gw4pts@gw4pts.ampr.org>
 *		Matthew Dillon, <dillon@apollo.west.oic.com>
 *		Arnt Gulbrandsen, <agulbra@nvg.unit.no>
 *		Jorge Cwik, <jorge@laser.satlink.net>
 */

/*
 * Changes:
 *		Pedro Roque	:	Fast Retransmit/Recovery.
 *					Two receive queues.
 *					Retransmit queue handled by TCP.
 *					Better retransmit timer handling.
 *					New congestion avoidance.
 *					Header prediction.
 *					Variable renaming.
 *
 *		Eric		:	Fast Retransmit.
 *		Randy Scott	:	MSS option defines.
 *		Eric Schenk	:	Fixes to slow start algorithm.
 *		Eric Schenk	:	Yet another double ACK bug.
 *		Eric Schenk	:	Delayed ACK bug fixes.
 *		Eric Schenk	:	Floyd style fast retrans war avoidance.
 *		David S. Miller	:	Don't allow zero congestion window.
 *		Eric Schenk	:	Fix retransmitter so that it sends
 *					next packet on ack of previous packet.
 *		Andi Kleen	:	Moved open_request checking here
 *					and process RSTs for open_requests.
 *		Andi Kleen	:	Better prune_queue, and other fixes.
 *		Andrey Savochkin:	Fix RTT measurements in the presence of
 *					timestamps.
 *		Andrey Savochkin:	Check sequence numbers correctly when
 *					removing SACKs due to in sequence incoming
 *					data segments.
 *		Andi Kleen:		Make sure we never ack data there is not
 *					enough room for. Also make this condition
 *					a fatal error if it might still happen.
 *		Andi Kleen:		Add tcp_measure_rcv_mss to make
 *					connections with MSS<min(MTU,ann. MSS)
 *					work without delayed acks.
 *		Andi Kleen:		Process packets with PSH set in the
 *					fast path.
 *		J Hadi Salim:		ECN support
 *	 	Andrei Gurtov,
 *		Pasi Sarolahti,
 *		Panu Kuhlberg:		Experimental audit of TCP (re)transmission
 *					engine. Lots of bugs are found.
 *		Pasi Sarolahti:		F-RTO for dealing with spurious RTOs
 */

#define pr_fmt(fmt) "TCP: " fmt

#include <linux/mm.h>
#include <linux/slab.h>
#include <linux/module.h>
#include <linux/sysctl.h>
#include <linux/kernel.h>
#include <net/dst.h>
#include <net/tcp.h>
#include <net/inet_common.h>
#include <linux/ipsec.h>
#include <asm/unaligned.h>
#include <net/netdma.h>

int sysctl_tcp_timestamps __read_mostly = 1;
int sysctl_tcp_window_scaling __read_mostly = 1;
int sysctl_tcp_sack __read_mostly = 1;
int sysctl_tcp_fack __read_mostly = 1;
int sysctl_tcp_reordering __read_mostly = TCP_FASTRETRANS_THRESH;
EXPORT_SYMBOL(sysctl_tcp_reordering);
int sysctl_tcp_dsack __read_mostly = 1;
int sysctl_tcp_app_win __read_mostly = 31;
int sysctl_tcp_adv_win_scale __read_mostly = 1;
EXPORT_SYMBOL(sysctl_tcp_adv_win_scale);

/* rfc5961 challenge ack rate limiting */
int sysctl_tcp_challenge_ack_limit = 100;

int sysctl_tcp_stdurg __read_mostly;
int sysctl_tcp_rfc1337 __read_mostly;
int sysctl_tcp_max_orphans __read_mostly = NR_FILE;
int sysctl_tcp_frto __read_mostly = 2;

int sysctl_tcp_thin_dupack __read_mostly;

int sysctl_tcp_moderate_rcvbuf __read_mostly = 1;
int sysctl_tcp_early_retrans __read_mostly = 3;

#define FLAG_DATA		0x01 /* Incoming frame contained data.		*/
#define FLAG_WIN_UPDATE		0x02 /* Incoming ACK was a window update.	*/
#define FLAG_DATA_ACKED		0x04 /* This ACK acknowledged new data.		*/
#define FLAG_RETRANS_DATA_ACKED	0x08 /* "" "" some of which was retransmitted.	*/
#define FLAG_SYN_ACKED		0x10 /* This ACK acknowledged SYN.		*/
#define FLAG_DATA_SACKED	0x20 /* New SACK.				*/
#define FLAG_ECE		0x40 /* ECE in this ACK				*/
#define FLAG_SLOWPATH		0x100 /* Do not skip RFC checks for window update.*/
#define FLAG_ORIG_SACK_ACKED	0x200 /* Never retransmitted data are (s)acked	*/
#define FLAG_SND_UNA_ADVANCED	0x400 /* Snd_una was changed (!= FLAG_DATA_ACKED) */
#define FLAG_DSACKING_ACK	0x800 /* SACK blocks contained D-SACK info */
#define FLAG_SACK_RENEGING	0x2000 /* snd_una advanced to a sacked seq */
#define FLAG_UPDATE_TS_RECENT	0x4000 /* tcp_replace_ts_recent() */

#define FLAG_ACKED		(FLAG_DATA_ACKED|FLAG_SYN_ACKED)
#define FLAG_NOT_DUP		(FLAG_DATA|FLAG_WIN_UPDATE|FLAG_ACKED)
#define FLAG_CA_ALERT		(FLAG_DATA_SACKED|FLAG_ECE)
#define FLAG_FORWARD_PROGRESS	(FLAG_ACKED|FLAG_DATA_SACKED)

#define TCP_REMNANT (TCP_FLAG_FIN|TCP_FLAG_URG|TCP_FLAG_SYN|TCP_FLAG_PSH)
#define TCP_HP_BITS (~(TCP_RESERVED_BITS|TCP_FLAG_PSH))

/* Adapt the MSS value used to make delayed ack decision to the
 * real world.
 */
static void tcp_measure_rcv_mss(struct sock *sk, const struct sk_buff *skb)
{
	struct inet_connection_sock *icsk = inet_csk(sk);
	const unsigned int lss = icsk->icsk_ack.last_seg_size;
	unsigned int len;

	icsk->icsk_ack.last_seg_size = 0;

	/* skb->len may jitter because of SACKs, even if peer
	 * sends good full-sized frames.
	 */
	len = skb_shinfo(skb)->gso_size ? : skb->len;
	if (len >= icsk->icsk_ack.rcv_mss) {
		icsk->icsk_ack.rcv_mss = len;
	} else {
		/* Otherwise, we make more careful check taking into account,
		 * that SACKs block is variable.
		 *
		 * "len" is invariant segment length, including TCP header.
		 */
		len += skb->data - skb_transport_header(skb);
		if (len >= TCP_MSS_DEFAULT + sizeof(struct tcphdr) ||
		    /* If PSH is not set, packet should be
		     * full sized, provided peer TCP is not badly broken.
		     * This observation (if it is correct 8)) allows
		     * to handle super-low mtu links fairly.
		     */
		    (len >= TCP_MIN_MSS + sizeof(struct tcphdr) &&
		     !(tcp_flag_word(tcp_hdr(skb)) & TCP_REMNANT))) {
			/* Subtract also invariant (if peer is RFC compliant),
			 * tcp header plus fixed timestamp option length.
			 * Resulting "len" is MSS free of SACK jitter.
			 */
			len -= tcp_sk(sk)->tcp_header_len;
			icsk->icsk_ack.last_seg_size = len;
			if (len == lss) {
				icsk->icsk_ack.rcv_mss = len;
				return;
			}
		}
		if (icsk->icsk_ack.pending & ICSK_ACK_PUSHED)
			icsk->icsk_ack.pending |= ICSK_ACK_PUSHED2;
		icsk->icsk_ack.pending |= ICSK_ACK_PUSHED;
	}
}

static void tcp_incr_quickack(struct sock *sk)
{
	struct inet_connection_sock *icsk = inet_csk(sk);
	unsigned int quickacks = tcp_sk(sk)->rcv_wnd / (2 * icsk->icsk_ack.rcv_mss);

	if (quickacks == 0)
		quickacks = 2;
	if (quickacks > icsk->icsk_ack.quick)
		icsk->icsk_ack.quick = min(quickacks, TCP_MAX_QUICKACKS);
}

static void tcp_enter_quickack_mode(struct sock *sk)
{
	struct inet_connection_sock *icsk = inet_csk(sk);
	tcp_incr_quickack(sk);
	icsk->icsk_ack.pingpong = 0;
	icsk->icsk_ack.ato = TCP_ATO_MIN;
}

/* Send ACKs quickly, if "quick" count is not exhausted
 * and the session is not interactive.
 */

static inline bool tcp_in_quickack_mode(const struct sock *sk)
{
	const struct inet_connection_sock *icsk = inet_csk(sk);

	return icsk->icsk_ack.quick && !icsk->icsk_ack.pingpong;
}

static inline void TCP_ECN_queue_cwr(struct tcp_sock *tp)
{
	if (tp->ecn_flags & TCP_ECN_OK)
		tp->ecn_flags |= TCP_ECN_QUEUE_CWR;
}

static inline void TCP_ECN_accept_cwr(struct tcp_sock *tp, const struct sk_buff *skb)
{
	if (tcp_hdr(skb)->cwr)
		tp->ecn_flags &= ~TCP_ECN_DEMAND_CWR;
}

static inline void TCP_ECN_withdraw_cwr(struct tcp_sock *tp)
{
	tp->ecn_flags &= ~TCP_ECN_DEMAND_CWR;
}

static inline void TCP_ECN_check_ce(struct tcp_sock *tp, const struct sk_buff *skb)
{
	if (!(tp->ecn_flags & TCP_ECN_OK))
		return;

	switch (TCP_SKB_CB(skb)->ip_dsfield & INET_ECN_MASK) {
	case INET_ECN_NOT_ECT:
		/* Funny extension: if ECT is not set on a segment,
		 * and we already seen ECT on a previous segment,
		 * it is probably a retransmit.
		 */
		if (tp->ecn_flags & TCP_ECN_SEEN)
			tcp_enter_quickack_mode((struct sock *)tp);
		break;
	case INET_ECN_CE:
		if (!(tp->ecn_flags & TCP_ECN_DEMAND_CWR)) {
			/* Better not delay acks, sender can have a very low cwnd */
			tcp_enter_quickack_mode((struct sock *)tp);
			tp->ecn_flags |= TCP_ECN_DEMAND_CWR;
		}
		/* fallinto */
	default:
		tp->ecn_flags |= TCP_ECN_SEEN;
	}
}

static inline void TCP_ECN_rcv_synack(struct tcp_sock *tp, const struct tcphdr *th)
{
	if ((tp->ecn_flags & TCP_ECN_OK) && (!th->ece || th->cwr))
		tp->ecn_flags &= ~TCP_ECN_OK;
}

static inline void TCP_ECN_rcv_syn(struct tcp_sock *tp, const struct tcphdr *th)
{
	if ((tp->ecn_flags & TCP_ECN_OK) && (!th->ece || !th->cwr))
		tp->ecn_flags &= ~TCP_ECN_OK;
}

static bool TCP_ECN_rcv_ecn_echo(const struct tcp_sock *tp, const struct tcphdr *th)
{
	if (th->ece && !th->syn && (tp->ecn_flags & TCP_ECN_OK))
		return true;
	return false;
}

/* Buffer size and advertised window tuning.
 *
 * 1. Tuning sk->sk_sndbuf, when connection enters established state.
 */

static void tcp_sndbuf_expand(struct sock *sk)
{
	const struct tcp_sock *tp = tcp_sk(sk);
	int sndmem, per_mss;
	u32 nr_segs;

	/* Worst case is non GSO/TSO : each frame consumes one skb
	 * and skb->head is kmalloced using power of two area of memory
	 */
	per_mss = max_t(u32, tp->rx_opt.mss_clamp, tp->mss_cache) +
		  MAX_TCP_HEADER +
		  SKB_DATA_ALIGN(sizeof(struct skb_shared_info));

	per_mss = roundup_pow_of_two(per_mss) +
		  SKB_DATA_ALIGN(sizeof(struct sk_buff));

	nr_segs = max_t(u32, TCP_INIT_CWND, tp->snd_cwnd);
	nr_segs = max_t(u32, nr_segs, tp->reordering + 1);

	/* Fast Recovery (RFC 5681 3.2) :
	 * Cubic needs 1.7 factor, rounded to 2 to include
	 * extra cushion (application might react slowly to POLLOUT)
	 */
	sndmem = 2 * nr_segs * per_mss;

	if (sk->sk_sndbuf < sndmem)
		sk->sk_sndbuf = min(sndmem, sysctl_tcp_wmem[2]);
}

/* 2. Tuning advertised window (window_clamp, rcv_ssthresh)
 *
 * All tcp_full_space() is split to two parts: "network" buffer, allocated
 * forward and advertised in receiver window (tp->rcv_wnd) and
 * "application buffer", required to isolate scheduling/application
 * latencies from network.
 * window_clamp is maximal advertised window. It can be less than
 * tcp_full_space(), in this case tcp_full_space() - window_clamp
 * is reserved for "application" buffer. The less window_clamp is
 * the smoother our behaviour from viewpoint of network, but the lower
 * throughput and the higher sensitivity of the connection to losses. 8)
 *
 * rcv_ssthresh is more strict window_clamp used at "slow start"
 * phase to predict further behaviour of this connection.
 * It is used for two goals:
 * - to enforce header prediction at sender, even when application
 *   requires some significant "application buffer". It is check #1.
 * - to prevent pruning of receive queue because of misprediction
 *   of receiver window. Check #2.
 *
 * The scheme does not work when sender sends good segments opening
 * window and then starts to feed us spaghetti. But it should work
 * in common situations. Otherwise, we have to rely on queue collapsing.
 */

/* Slow part of check#2. */
static int __tcp_grow_window(const struct sock *sk, const struct sk_buff *skb)
{
	struct tcp_sock *tp = tcp_sk(sk);
	/* Optimize this! */
	int truesize = tcp_win_from_space(skb->truesize) >> 1;
	int window = tcp_win_from_space(sysctl_tcp_rmem[2]) >> 1;

	while (tp->rcv_ssthresh <= window) {
		if (truesize <= skb->len)
			return 2 * inet_csk(sk)->icsk_ack.rcv_mss;

		truesize >>= 1;
		window >>= 1;
	}
	return 0;
}

static void tcp_grow_window(struct sock *sk, const struct sk_buff *skb)
{
	struct tcp_sock *tp = tcp_sk(sk);

	/* Check #1 */
	if (tp->rcv_ssthresh < tp->window_clamp &&
	    (int)tp->rcv_ssthresh < tcp_space(sk) &&
	    !sk_under_memory_pressure(sk)) {
		int incr;

		/* Check #2. Increase window, if skb with such overhead
		 * will fit to rcvbuf in future.
		 */
		if (tcp_win_from_space(skb->truesize) <= skb->len)
			incr = 2 * tp->advmss;
		else
			incr = __tcp_grow_window(sk, skb);

		if (incr) {
			incr = max_t(int, incr, 2 * skb->len);
			tp->rcv_ssthresh = min(tp->rcv_ssthresh + incr,
					       tp->window_clamp);
			inet_csk(sk)->icsk_ack.quick |= 1;
		}
	}
}

/* 3. Tuning rcvbuf, when connection enters established state. */
static void tcp_fixup_rcvbuf(struct sock *sk)
{
	u32 mss = tcp_sk(sk)->advmss;
	int rcvmem;

	rcvmem = 2 * SKB_TRUESIZE(mss + MAX_TCP_HEADER) *
		 tcp_default_init_rwnd(mss);

	/* Dynamic Right Sizing (DRS) has 2 to 3 RTT latency
	 * Allow enough cushion so that sender is not limited by our window
	 */
	if (sysctl_tcp_moderate_rcvbuf)
		rcvmem <<= 2;

	if (sk->sk_rcvbuf < rcvmem)
		sk->sk_rcvbuf = min(rcvmem, sysctl_tcp_rmem[2]);
}

/* 4. Try to fixup all. It is made immediately after connection enters
 *    established state.
 */
void tcp_init_buffer_space(struct sock *sk)
{
	struct tcp_sock *tp = tcp_sk(sk);
	int maxwin;

	if (!(sk->sk_userlocks & SOCK_RCVBUF_LOCK))
		tcp_fixup_rcvbuf(sk);
	if (!(sk->sk_userlocks & SOCK_SNDBUF_LOCK))
		tcp_sndbuf_expand(sk);

	tp->rcvq_space.space = tp->rcv_wnd;
	tp->rcvq_space.time = tcp_time_stamp;
	tp->rcvq_space.seq = tp->copied_seq;

	maxwin = tcp_full_space(sk);

	if (tp->window_clamp >= maxwin) {
		tp->window_clamp = maxwin;

		if (sysctl_tcp_app_win && maxwin > 4 * tp->advmss)
			tp->window_clamp = max(maxwin -
					       (maxwin >> sysctl_tcp_app_win),
					       4 * tp->advmss);
	}

	/* Force reservation of one segment. */
	if (sysctl_tcp_app_win &&
	    tp->window_clamp > 2 * tp->advmss &&
	    tp->window_clamp + tp->advmss > maxwin)
		tp->window_clamp = max(2 * tp->advmss, maxwin - tp->advmss);

	tp->rcv_ssthresh = min(tp->rcv_ssthresh, tp->window_clamp);
	tp->snd_cwnd_stamp = tcp_time_stamp;
}

/* 5. Recalculate window clamp after socket hit its memory bounds. */
static void tcp_clamp_window(struct sock *sk)
{
	struct tcp_sock *tp = tcp_sk(sk);
	struct inet_connection_sock *icsk = inet_csk(sk);

	icsk->icsk_ack.quick = 0;

	if (sk->sk_rcvbuf < sysctl_tcp_rmem[2] &&
	    !(sk->sk_userlocks & SOCK_RCVBUF_LOCK) &&
	    !sk_under_memory_pressure(sk) &&
	    sk_memory_allocated(sk) < sk_prot_mem_limits(sk, 0)) {
		sk->sk_rcvbuf = min(atomic_read(&sk->sk_rmem_alloc),
				    sysctl_tcp_rmem[2]);
	}
	if (atomic_read(&sk->sk_rmem_alloc) > sk->sk_rcvbuf)
		tp->rcv_ssthresh = min(tp->window_clamp, 2U * tp->advmss);
}

/* Initialize RCV_MSS value.
 * RCV_MSS is an our guess about MSS used by the peer.
 * We haven't any direct information about the MSS.
 * It's better to underestimate the RCV_MSS rather than overestimate.
 * Overestimations make us ACKing less frequently than needed.
 * Underestimations are more easy to detect and fix by tcp_measure_rcv_mss().
 */
void tcp_initialize_rcv_mss(struct sock *sk)
{
	const struct tcp_sock *tp = tcp_sk(sk);
	unsigned int hint = min_t(unsigned int, tp->advmss, tp->mss_cache);

	hint = min(hint, tp->rcv_wnd / 2);
	hint = min(hint, TCP_MSS_DEFAULT);
	hint = max(hint, TCP_MIN_MSS);

	inet_csk(sk)->icsk_ack.rcv_mss = hint;
}
EXPORT_SYMBOL(tcp_initialize_rcv_mss);

/* Receiver "autotuning" code.
 *
 * The algorithm for RTT estimation w/o timestamps is based on
 * Dynamic Right-Sizing (DRS) by Wu Feng and Mike Fisk of LANL.
 * <http://public.lanl.gov/radiant/pubs.html#DRS>
 *
 * More detail on this code can be found at
 * <http://staff.psc.edu/jheffner/>,
 * though this reference is out of date.  A new paper
 * is pending.
 */
static void tcp_rcv_rtt_update(struct tcp_sock *tp, u32 sample, int win_dep)
{
	u32 new_sample = tp->rcv_rtt_est.rtt;
	long m = sample;

	if (m == 0)
		m = 1;

	if (new_sample != 0) {
		/* If we sample in larger samples in the non-timestamp
		 * case, we could grossly overestimate the RTT especially
		 * with chatty applications or bulk transfer apps which
		 * are stalled on filesystem I/O.
		 *
		 * Also, since we are only going for a minimum in the
		 * non-timestamp case, we do not smooth things out
		 * else with timestamps disabled convergence takes too
		 * long.
		 */
		if (!win_dep) {
			m -= (new_sample >> 3);
			new_sample += m;
		} else {
			m <<= 3;
			if (m < new_sample)
				new_sample = m;
		}
	} else {
		/* No previous measure. */
		new_sample = m << 3;
	}

	if (tp->rcv_rtt_est.rtt != new_sample)
		tp->rcv_rtt_est.rtt = new_sample;
}

static inline void tcp_rcv_rtt_measure(struct tcp_sock *tp)
{
	if (tp->rcv_rtt_est.time == 0)
		goto new_measure;
	if (before(tp->rcv_nxt, tp->rcv_rtt_est.seq))
		return;
	tcp_rcv_rtt_update(tp, tcp_time_stamp - tp->rcv_rtt_est.time, 1);

new_measure:
	tp->rcv_rtt_est.seq = tp->rcv_nxt + tp->rcv_wnd;
	tp->rcv_rtt_est.time = tcp_time_stamp;
}

static inline void tcp_rcv_rtt_measure_ts(struct sock *sk,
					  const struct sk_buff *skb)
{
	struct tcp_sock *tp = tcp_sk(sk);
	if (tp->rx_opt.rcv_tsecr &&
	    (TCP_SKB_CB(skb)->end_seq -
	     TCP_SKB_CB(skb)->seq >= inet_csk(sk)->icsk_ack.rcv_mss))
		tcp_rcv_rtt_update(tp, tcp_time_stamp - tp->rx_opt.rcv_tsecr, 0);
}

/*
 * This function should be called every time data is copied to user space.
 * It calculates the appropriate TCP receive buffer space.
 */
void tcp_rcv_space_adjust(struct sock *sk)
{
	struct tcp_sock *tp = tcp_sk(sk);
	int time;
	int copied;

	time = tcp_time_stamp - tp->rcvq_space.time;
	if (time < (tp->rcv_rtt_est.rtt >> 3) || tp->rcv_rtt_est.rtt == 0)
		return;

	/* Number of bytes copied to user in last RTT */
	copied = tp->copied_seq - tp->rcvq_space.seq;
	if (copied <= tp->rcvq_space.space)
		goto new_measure;

	/* A bit of theory :
	 * copied = bytes received in previous RTT, our base window
	 * To cope with packet losses, we need a 2x factor
	 * To cope with slow start, and sender growing its cwin by 100 %
	 * every RTT, we need a 4x factor, because the ACK we are sending
	 * now is for the next RTT, not the current one :
	 * <prev RTT . ><current RTT .. ><next RTT .... >
	 */

	if (sysctl_tcp_moderate_rcvbuf &&
	    !(sk->sk_userlocks & SOCK_RCVBUF_LOCK)) {
		int rcvwin, rcvmem, rcvbuf;

		/* minimal window to cope with packet losses, assuming
		 * steady state. Add some cushion because of small variations.
		 */
		rcvwin = (copied << 1) + 16 * tp->advmss;

		/* If rate increased by 25%,
		 *	assume slow start, rcvwin = 3 * copied
		 * If rate increased by 50%,
		 *	assume sender can use 2x growth, rcvwin = 4 * copied
		 */
		if (copied >=
		    tp->rcvq_space.space + (tp->rcvq_space.space >> 2)) {
			if (copied >=
			    tp->rcvq_space.space + (tp->rcvq_space.space >> 1))
				rcvwin <<= 1;
			else
				rcvwin += (rcvwin >> 1);
		}

		rcvmem = SKB_TRUESIZE(tp->advmss + MAX_TCP_HEADER);
		while (tcp_win_from_space(rcvmem) < tp->advmss)
			rcvmem += 128;

		rcvbuf = min(rcvwin / tp->advmss * rcvmem, sysctl_tcp_rmem[2]);
		if (rcvbuf > sk->sk_rcvbuf) {
			sk->sk_rcvbuf = rcvbuf;

			/* Make the window clamp follow along.  */
			tp->window_clamp = rcvwin;
		}
	}
	tp->rcvq_space.space = copied;

new_measure:
	tp->rcvq_space.seq = tp->copied_seq;
	tp->rcvq_space.time = tcp_time_stamp;
}

/* There is something which you must keep in mind when you analyze the
 * behavior of the tp->ato delayed ack timeout interval.  When a
 * connection starts up, we want to ack as quickly as possible.  The
 * problem is that "good" TCP's do slow start at the beginning of data
 * transmission.  The means that until we send the first few ACK's the
 * sender will sit on his end and only queue most of his data, because
 * he can only send snd_cwnd unacked packets at any given time.  For
 * each ACK we send, he increments snd_cwnd and transmits more of his
 * queue.  -DaveM
 */
static void tcp_event_data_recv(struct sock *sk, struct sk_buff *skb)
{
	struct tcp_sock *tp = tcp_sk(sk);
	struct inet_connection_sock *icsk = inet_csk(sk);
	u32 now;

	inet_csk_schedule_ack(sk);

	tcp_measure_rcv_mss(sk, skb);

	tcp_rcv_rtt_measure(tp);

	now = tcp_time_stamp;

	if (!icsk->icsk_ack.ato) {
		/* The _first_ data packet received, initialize
		 * delayed ACK engine.
		 */
		tcp_incr_quickack(sk);
		icsk->icsk_ack.ato = TCP_ATO_MIN;
	} else {
		int m = now - icsk->icsk_ack.lrcvtime;

		if (m <= TCP_ATO_MIN / 2) {
			/* The fastest case is the first. */
			icsk->icsk_ack.ato = (icsk->icsk_ack.ato >> 1) + TCP_ATO_MIN / 2;
		} else if (m < icsk->icsk_ack.ato) {
			icsk->icsk_ack.ato = (icsk->icsk_ack.ato >> 1) + m;
			if (icsk->icsk_ack.ato > icsk->icsk_rto)
				icsk->icsk_ack.ato = icsk->icsk_rto;
		} else if (m > icsk->icsk_rto) {
			/* Too long gap. Apparently sender failed to
			 * restart window, so that we send ACKs quickly.
			 */
			tcp_incr_quickack(sk);
			sk_mem_reclaim(sk);
		}
	}
	icsk->icsk_ack.lrcvtime = now;

	TCP_ECN_check_ce(tp, skb);

	if (skb->len >= 128)
		tcp_grow_window(sk, skb);
}

/* Called to compute a smoothed rtt estimate. The data fed to this
 * routine either comes from timestamps, or from segments that were
 * known _not_ to have been retransmitted [see Karn/Partridge
 * Proceedings SIGCOMM 87]. The algorithm is from the SIGCOMM 88
 * piece by Van Jacobson.
 * NOTE: the next three routines used to be one big routine.
 * To save cycles in the RFC 1323 implementation it was better to break
 * it up into three procedures. -- erics
 */
static void tcp_rtt_estimator(struct sock *sk, long mrtt_us)
{
	struct tcp_sock *tp = tcp_sk(sk);
	long m = mrtt_us; /* RTT */
	u32 srtt = tp->srtt_us;

	/*	The following amusing code comes from Jacobson's
	 *	article in SIGCOMM '88.  Note that rtt and mdev
	 *	are scaled versions of rtt and mean deviation.
	 *	This is designed to be as fast as possible
	 *	m stands for "measurement".
	 *
	 *	On a 1990 paper the rto value is changed to:
	 *	RTO = rtt + 4 * mdev
	 *
	 * Funny. This algorithm seems to be very broken.
	 * These formulae increase RTO, when it should be decreased, increase
	 * too slowly, when it should be increased quickly, decrease too quickly
	 * etc. I guess in BSD RTO takes ONE value, so that it is absolutely
	 * does not matter how to _calculate_ it. Seems, it was trap
	 * that VJ failed to avoid. 8)
	 */
	if (srtt != 0) {
		m -= (srtt >> 3);	/* m is now error in rtt est */
		srtt += m;		/* rtt = 7/8 rtt + 1/8 new */
		if (m < 0) {
			m = -m;		/* m is now abs(error) */
			m -= (tp->mdev_us >> 2);   /* similar update on mdev */
			/* This is similar to one of Eifel findings.
			 * Eifel blocks mdev updates when rtt decreases.
			 * This solution is a bit different: we use finer gain
			 * for mdev in this case (alpha*beta).
			 * Like Eifel it also prevents growth of rto,
			 * but also it limits too fast rto decreases,
			 * happening in pure Eifel.
			 */
			if (m > 0)
				m >>= 3;
		} else {
			m -= (tp->mdev_us >> 2);   /* similar update on mdev */
		}
		tp->mdev_us += m;		/* mdev = 3/4 mdev + 1/4 new */
		if (tp->mdev_us > tp->mdev_max_us) {
			tp->mdev_max_us = tp->mdev_us;
			if (tp->mdev_max_us > tp->rttvar_us)
				tp->rttvar_us = tp->mdev_max_us;
		}
		if (after(tp->snd_una, tp->rtt_seq)) {
			if (tp->mdev_max_us < tp->rttvar_us)
				tp->rttvar_us -= (tp->rttvar_us - tp->mdev_max_us) >> 2;
			tp->rtt_seq = tp->snd_nxt;
			tp->mdev_max_us = tcp_rto_min_us(sk);
		}
	} else {
		/* no previous measure. */
		srtt = m << 3;		/* take the measured time to be rtt */
		tp->mdev_us = m << 1;	/* make sure rto = 3*rtt */
		tp->rttvar_us = max(tp->mdev_us, tcp_rto_min_us(sk));
		tp->mdev_max_us = tp->rttvar_us;
		tp->rtt_seq = tp->snd_nxt;
	}
	tp->srtt_us = max(1U, srtt);
}

/* Set the sk_pacing_rate to allow proper sizing of TSO packets.
 * Note: TCP stack does not yet implement pacing.
 * FQ packet scheduler can be used to implement cheap but effective
 * TCP pacing, to smooth the burst on large writes when packets
 * in flight is significantly lower than cwnd (or rwin)
 */
static void tcp_update_pacing_rate(struct sock *sk)
{
	const struct tcp_sock *tp = tcp_sk(sk);
	u64 rate;

	/* set sk_pacing_rate to 200 % of current rate (mss * cwnd / srtt) */
	rate = (u64)tp->mss_cache * 2 * (USEC_PER_SEC << 3);

	rate *= max(tp->snd_cwnd, tp->packets_out);

	if (likely(tp->srtt_us))
		do_div(rate, tp->srtt_us);

	/* ACCESS_ONCE() is needed because sch_fq fetches sk_pacing_rate
	 * without any lock. We want to make sure compiler wont store
	 * intermediate values in this location.
	 */
	ACCESS_ONCE(sk->sk_pacing_rate) = min_t(u64, rate,
						sk->sk_max_pacing_rate);
}

/* Calculate rto without backoff.  This is the second half of Van Jacobson's
 * routine referred to above.
 */
static void tcp_set_rto(struct sock *sk)
{
	const struct tcp_sock *tp = tcp_sk(sk);
	/* Old crap is replaced with new one. 8)
	 *
	 * More seriously:
	 * 1. If rtt variance happened to be less 50msec, it is hallucination.
	 *    It cannot be less due to utterly erratic ACK generation made
	 *    at least by solaris and freebsd. "Erratic ACKs" has _nothing_
	 *    to do with delayed acks, because at cwnd>2 true delack timeout
	 *    is invisible. Actually, Linux-2.4 also generates erratic
	 *    ACKs in some circumstances.
	 */
	inet_csk(sk)->icsk_rto = __tcp_set_rto(tp);

	/* 2. Fixups made earlier cannot be right.
	 *    If we do not estimate RTO correctly without them,
	 *    all the algo is pure shit and should be replaced
	 *    with correct one. It is exactly, which we pretend to do.
	 */

	/* NOTE: clamping at TCP_RTO_MIN is not required, current algo
	 * guarantees that rto is higher.
	 */
	tcp_bound_rto(sk);
}

__u32 tcp_init_cwnd(const struct tcp_sock *tp, const struct dst_entry *dst)
{
	__u32 cwnd = (dst ? dst_metric(dst, RTAX_INITCWND) : 0);

	if (!cwnd)
		cwnd = TCP_INIT_CWND;
	return min_t(__u32, cwnd, tp->snd_cwnd_clamp);
}

/*
 * Packet counting of FACK is based on in-order assumptions, therefore TCP
 * disables it when reordering is detected
 */
void tcp_disable_fack(struct tcp_sock *tp)
{
	/* RFC3517 uses different metric in lost marker => reset on change */
	if (tcp_is_fack(tp))
		tp->lost_skb_hint = NULL;
	tp->rx_opt.sack_ok &= ~TCP_FACK_ENABLED;
}

/* Take a notice that peer is sending D-SACKs */
static void tcp_dsack_seen(struct tcp_sock *tp)
{
	tp->rx_opt.sack_ok |= TCP_DSACK_SEEN;
}

static void tcp_update_reordering(struct sock *sk, const int metric,
				  const int ts)
{
	struct tcp_sock *tp = tcp_sk(sk);
	if (metric > tp->reordering) {
		int mib_idx;

		tp->reordering = min(TCP_MAX_REORDERING, metric);

		/* This exciting event is worth to be remembered. 8) */
		if (ts)
			mib_idx = LINUX_MIB_TCPTSREORDER;
		else if (tcp_is_reno(tp))
			mib_idx = LINUX_MIB_TCPRENOREORDER;
		else if (tcp_is_fack(tp))
			mib_idx = LINUX_MIB_TCPFACKREORDER;
		else
			mib_idx = LINUX_MIB_TCPSACKREORDER;

		NET_INC_STATS_BH(sock_net(sk), mib_idx);
#if FASTRETRANS_DEBUG > 1
		pr_debug("Disorder%d %d %u f%u s%u rr%d\n",
			 tp->rx_opt.sack_ok, inet_csk(sk)->icsk_ca_state,
			 tp->reordering,
			 tp->fackets_out,
			 tp->sacked_out,
			 tp->undo_marker ? tp->undo_retrans : 0);
#endif
		tcp_disable_fack(tp);
	}

	if (metric > 0)
		tcp_disable_early_retrans(tp);
}

/* This must be called before lost_out is incremented */
static void tcp_verify_retransmit_hint(struct tcp_sock *tp, struct sk_buff *skb)
{
	if ((tp->retransmit_skb_hint == NULL) ||
	    before(TCP_SKB_CB(skb)->seq,
		   TCP_SKB_CB(tp->retransmit_skb_hint)->seq))
		tp->retransmit_skb_hint = skb;

	if (!tp->lost_out ||
	    after(TCP_SKB_CB(skb)->end_seq, tp->retransmit_high))
		tp->retransmit_high = TCP_SKB_CB(skb)->end_seq;
}

static void tcp_skb_mark_lost(struct tcp_sock *tp, struct sk_buff *skb)
{
	if (!(TCP_SKB_CB(skb)->sacked & (TCPCB_LOST|TCPCB_SACKED_ACKED))) {
		tcp_verify_retransmit_hint(tp, skb);

		tp->lost_out += tcp_skb_pcount(skb);
		TCP_SKB_CB(skb)->sacked |= TCPCB_LOST;
	}
}

static void tcp_skb_mark_lost_uncond_verify(struct tcp_sock *tp,
					    struct sk_buff *skb)
{
	tcp_verify_retransmit_hint(tp, skb);

	if (!(TCP_SKB_CB(skb)->sacked & (TCPCB_LOST|TCPCB_SACKED_ACKED))) {
		tp->lost_out += tcp_skb_pcount(skb);
		TCP_SKB_CB(skb)->sacked |= TCPCB_LOST;
	}
}

/* This procedure tags the retransmission queue when SACKs arrive.
 *
 * We have three tag bits: SACKED(S), RETRANS(R) and LOST(L).
 * Packets in queue with these bits set are counted in variables
 * sacked_out, retrans_out and lost_out, correspondingly.
 *
 * Valid combinations are:
 * Tag  InFlight	Description
 * 0	1		- orig segment is in flight.
 * S	0		- nothing flies, orig reached receiver.
 * L	0		- nothing flies, orig lost by net.
 * R	2		- both orig and retransmit are in flight.
 * L|R	1		- orig is lost, retransmit is in flight.
 * S|R  1		- orig reached receiver, retrans is still in flight.
 * (L|S|R is logically valid, it could occur when L|R is sacked,
 *  but it is equivalent to plain S and code short-curcuits it to S.
 *  L|S is logically invalid, it would mean -1 packet in flight 8))
 *
 * These 6 states form finite state machine, controlled by the following events:
 * 1. New ACK (+SACK) arrives. (tcp_sacktag_write_queue())
 * 2. Retransmission. (tcp_retransmit_skb(), tcp_xmit_retransmit_queue())
 * 3. Loss detection event of two flavors:
 *	A. Scoreboard estimator decided the packet is lost.
 *	   A'. Reno "three dupacks" marks head of queue lost.
 *	   A''. Its FACK modification, head until snd.fack is lost.
 *	B. SACK arrives sacking SND.NXT at the moment, when the
 *	   segment was retransmitted.
 * 4. D-SACK added new rule: D-SACK changes any tag to S.
 *
 * It is pleasant to note, that state diagram turns out to be commutative,
 * so that we are allowed not to be bothered by order of our actions,
 * when multiple events arrive simultaneously. (see the function below).
 *
 * Reordering detection.
 * --------------------
 * Reordering metric is maximal distance, which a packet can be displaced
 * in packet stream. With SACKs we can estimate it:
 *
 * 1. SACK fills old hole and the corresponding segment was not
 *    ever retransmitted -> reordering. Alas, we cannot use it
 *    when segment was retransmitted.
 * 2. The last flaw is solved with D-SACK. D-SACK arrives
 *    for retransmitted and already SACKed segment -> reordering..
 * Both of these heuristics are not used in Loss state, when we cannot
 * account for retransmits accurately.
 *
 * SACK block validation.
 * ----------------------
 *
 * SACK block range validation checks that the received SACK block fits to
 * the expected sequence limits, i.e., it is between SND.UNA and SND.NXT.
 * Note that SND.UNA is not included to the range though being valid because
 * it means that the receiver is rather inconsistent with itself reporting
 * SACK reneging when it should advance SND.UNA. Such SACK block this is
 * perfectly valid, however, in light of RFC2018 which explicitly states
 * that "SACK block MUST reflect the newest segment.  Even if the newest
 * segment is going to be discarded ...", not that it looks very clever
 * in case of head skb. Due to potentional receiver driven attacks, we
 * choose to avoid immediate execution of a walk in write queue due to
 * reneging and defer head skb's loss recovery to standard loss recovery
 * procedure that will eventually trigger (nothing forbids us doing this).
 *
 * Implements also blockage to start_seq wrap-around. Problem lies in the
 * fact that though start_seq (s) is before end_seq (i.e., not reversed),
 * there's no guarantee that it will be before snd_nxt (n). The problem
 * happens when start_seq resides between end_seq wrap (e_w) and snd_nxt
 * wrap (s_w):
 *
 *         <- outs wnd ->                          <- wrapzone ->
 *         u     e      n                         u_w   e_w  s n_w
 *         |     |      |                          |     |   |  |
 * |<------------+------+----- TCP seqno space --------------+---------->|
 * ...-- <2^31 ->|                                           |<--------...
 * ...---- >2^31 ------>|                                    |<--------...
 *
 * Current code wouldn't be vulnerable but it's better still to discard such
 * crazy SACK blocks. Doing this check for start_seq alone closes somewhat
 * similar case (end_seq after snd_nxt wrap) as earlier reversed check in
 * snd_nxt wrap -> snd_una region will then become "well defined", i.e.,
 * equal to the ideal case (infinite seqno space without wrap caused issues).
 *
 * With D-SACK the lower bound is extended to cover sequence space below
 * SND.UNA down to undo_marker, which is the last point of interest. Yet
 * again, D-SACK block must not to go across snd_una (for the same reason as
 * for the normal SACK blocks, explained above). But there all simplicity
 * ends, TCP might receive valid D-SACKs below that. As long as they reside
 * fully below undo_marker they do not affect behavior in anyway and can
 * therefore be safely ignored. In rare cases (which are more or less
 * theoretical ones), the D-SACK will nicely cross that boundary due to skb
 * fragmentation and packet reordering past skb's retransmission. To consider
 * them correctly, the acceptable range must be extended even more though
 * the exact amount is rather hard to quantify. However, tp->max_window can
 * be used as an exaggerated estimate.
 */
static bool tcp_is_sackblock_valid(struct tcp_sock *tp, bool is_dsack,
				   u32 start_seq, u32 end_seq)
{
	/* Too far in future, or reversed (interpretation is ambiguous) */
	if (after(end_seq, tp->snd_nxt) || !before(start_seq, end_seq))
		return false;

	/* Nasty start_seq wrap-around check (see comments above) */
	if (!before(start_seq, tp->snd_nxt))
		return false;

	/* In outstanding window? ...This is valid exit for D-SACKs too.
	 * start_seq == snd_una is non-sensical (see comments above)
	 */
	if (after(start_seq, tp->snd_una))
		return true;

	if (!is_dsack || !tp->undo_marker)
		return false;