Commit b814bdce authored by Eric Dumazet's avatar Eric Dumazet Committed by Jakub Kicinski
Browse files

tcp: move tcp_rate_gen to tcp_input.c



This function is called from one caller only, in TCP fast path.

Move it to tcp_input.c so that compiler can inline it.

$ scripts/bloat-o-meter -t vmlinux.old vmlinux.new
add/remove: 0/2 grow/shrink: 1/0 up/down: 226/-300 (-74)
Function                                     old     new   delta
tcp_ack                                     5405    5631    +226
__pfx_tcp_rate_gen                            16       -     -16
tcp_rate_gen                                 284       -    -284
Total: Before=22566536, After=22566462, chg -0.00%

Signed-off-by: default avatarEric Dumazet <edumazet@google.com>
Reviewed-by: default avatarNeal Cardwell <ncardwell@google.com>
Link: https://patch.msgid.link/20260121095923.3134639-2-edumazet@google.com


Signed-off-by: default avatarJakub Kicinski <kuba@kernel.org>
parent 3b87882b
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+0 −2
Original line number Diff line number Diff line
@@ -1356,8 +1356,6 @@ static inline void tcp_ca_event(struct sock *sk, const enum tcp_ca_event event)
void tcp_set_ca_state(struct sock *sk, const u8 ca_state);

/* From tcp_rate.c */
void tcp_rate_gen(struct sock *sk, u32 delivered, u32 lost,
		  bool is_sack_reneg, struct rate_sample *rs);
void tcp_rate_check_app_limited(struct sock *sk);

static inline bool tcp_skb_sent_after(u64 t1, u64 t2, u32 seq1, u32 seq2)
+110 −0
Original line number Diff line number Diff line
@@ -1637,6 +1637,116 @@ static u8 tcp_sacktag_one(struct sock *sk,
	return sacked;
}

/* The bandwidth estimator estimates the rate at which the network
 * can currently deliver outbound data packets for this flow. At a high
 * level, it operates by taking a delivery rate sample for each ACK.
 *
 * A rate sample records the rate at which the network delivered packets
 * for this flow, calculated over the time interval between the transmission
 * of a data packet and the acknowledgment of that packet.
 *
 * Specifically, over the interval between each transmit and corresponding ACK,
 * the estimator generates a delivery rate sample. Typically it uses the rate
 * at which packets were acknowledged. However, the approach of using only the
 * acknowledgment rate faces a challenge under the prevalent ACK decimation or
 * compression: packets can temporarily appear to be delivered much quicker
 * than the bottleneck rate. Since it is physically impossible to do that in a
 * sustained fashion, when the estimator notices that the ACK rate is faster
 * than the transmit rate, it uses the latter:
 *
 *    send_rate = #pkts_delivered/(last_snd_time - first_snd_time)
 *    ack_rate  = #pkts_delivered/(last_ack_time - first_ack_time)
 *    bw = min(send_rate, ack_rate)
 *
 * Notice the estimator essentially estimates the goodput, not always the
 * network bottleneck link rate when the sending or receiving is limited by
 * other factors like applications or receiver window limits.  The estimator
 * deliberately avoids using the inter-packet spacing approach because that
 * approach requires a large number of samples and sophisticated filtering.
 *
 * TCP flows can often be application-limited in request/response workloads.
 * The estimator marks a bandwidth sample as application-limited if there
 * was some moment during the sampled window of packets when there was no data
 * ready to send in the write queue.
 */

/* Update the connection delivery information and generate a rate sample. */
static void tcp_rate_gen(struct sock *sk, u32 delivered, u32 lost,
			 bool is_sack_reneg, struct rate_sample *rs)
{
	struct tcp_sock *tp = tcp_sk(sk);
	u32 snd_us, ack_us;

	/* Clear app limited if bubble is acked and gone. */
	if (tp->app_limited && after(tp->delivered, tp->app_limited))
		tp->app_limited = 0;

	/* TODO: there are multiple places throughout tcp_ack() to get
	 * current time. Refactor the code using a new "tcp_acktag_state"
	 * to carry current time, flags, stats like "tcp_sacktag_state".
	 */
	if (delivered)
		tp->delivered_mstamp = tp->tcp_mstamp;

	rs->acked_sacked = delivered;	/* freshly ACKed or SACKed */
	rs->losses = lost;		/* freshly marked lost */
	/* Return an invalid sample if no timing information is available or
	 * in recovery from loss with SACK reneging. Rate samples taken during
	 * a SACK reneging event may overestimate bw by including packets that
	 * were SACKed before the reneg.
	 */
	if (!rs->prior_mstamp || is_sack_reneg) {
		rs->delivered = -1;
		rs->interval_us = -1;
		return;
	}
	rs->delivered   = tp->delivered - rs->prior_delivered;

	rs->delivered_ce = tp->delivered_ce - rs->prior_delivered_ce;
	/* delivered_ce occupies less than 32 bits in the skb control block */
	rs->delivered_ce &= TCPCB_DELIVERED_CE_MASK;

	/* Model sending data and receiving ACKs as separate pipeline phases
	 * for a window. Usually the ACK phase is longer, but with ACK
	 * compression the send phase can be longer. To be safe we use the
	 * longer phase.
	 */
	snd_us = rs->interval_us;				/* send phase */
	ack_us = tcp_stamp_us_delta(tp->tcp_mstamp,
				    rs->prior_mstamp); /* ack phase */
	rs->interval_us = max(snd_us, ack_us);

	/* Record both segment send and ack receive intervals */
	rs->snd_interval_us = snd_us;
	rs->rcv_interval_us = ack_us;

	/* Normally we expect interval_us >= min-rtt.
	 * Note that rate may still be over-estimated when a spuriously
	 * retransmistted skb was first (s)acked because "interval_us"
	 * is under-estimated (up to an RTT). However continuously
	 * measuring the delivery rate during loss recovery is crucial
	 * for connections suffer heavy or prolonged losses.
	 */
	if (unlikely(rs->interval_us < tcp_min_rtt(tp))) {
		if (!rs->is_retrans)
			pr_debug("tcp rate: %ld %d %u %u %u\n",
				 rs->interval_us, rs->delivered,
				 inet_csk(sk)->icsk_ca_state,
				 tp->rx_opt.sack_ok, tcp_min_rtt(tp));
		rs->interval_us = -1;
		return;
	}

	/* Record the last non-app-limited or the highest app-limited bw */
	if (!rs->is_app_limited ||
	    ((u64)rs->delivered * tp->rate_interval_us >=
	     (u64)tp->rate_delivered * rs->interval_us)) {
		tp->rate_delivered = rs->delivered;
		tp->rate_interval_us = rs->interval_us;
		tp->rate_app_limited = rs->is_app_limited;
	}
}

/* When an skb is sacked or acked, we fill in the rate sample with the (prior)
 * delivery information when the skb was last transmitted.
 *
+0 −110
Original line number Diff line number Diff line
// SPDX-License-Identifier: GPL-2.0-only
#include <net/tcp.h>

/* The bandwidth estimator estimates the rate at which the network
 * can currently deliver outbound data packets for this flow. At a high
 * level, it operates by taking a delivery rate sample for each ACK.
 *
 * A rate sample records the rate at which the network delivered packets
 * for this flow, calculated over the time interval between the transmission
 * of a data packet and the acknowledgment of that packet.
 *
 * Specifically, over the interval between each transmit and corresponding ACK,
 * the estimator generates a delivery rate sample. Typically it uses the rate
 * at which packets were acknowledged. However, the approach of using only the
 * acknowledgment rate faces a challenge under the prevalent ACK decimation or
 * compression: packets can temporarily appear to be delivered much quicker
 * than the bottleneck rate. Since it is physically impossible to do that in a
 * sustained fashion, when the estimator notices that the ACK rate is faster
 * than the transmit rate, it uses the latter:
 *
 *    send_rate = #pkts_delivered/(last_snd_time - first_snd_time)
 *    ack_rate  = #pkts_delivered/(last_ack_time - first_ack_time)
 *    bw = min(send_rate, ack_rate)
 *
 * Notice the estimator essentially estimates the goodput, not always the
 * network bottleneck link rate when the sending or receiving is limited by
 * other factors like applications or receiver window limits.  The estimator
 * deliberately avoids using the inter-packet spacing approach because that
 * approach requires a large number of samples and sophisticated filtering.
 *
 * TCP flows can often be application-limited in request/response workloads.
 * The estimator marks a bandwidth sample as application-limited if there
 * was some moment during the sampled window of packets when there was no data
 * ready to send in the write queue.
 */

/* Update the connection delivery information and generate a rate sample. */
void tcp_rate_gen(struct sock *sk, u32 delivered, u32 lost,
		  bool is_sack_reneg, struct rate_sample *rs)
{
	struct tcp_sock *tp = tcp_sk(sk);
	u32 snd_us, ack_us;

	/* Clear app limited if bubble is acked and gone. */
	if (tp->app_limited && after(tp->delivered, tp->app_limited))
		tp->app_limited = 0;

	/* TODO: there are multiple places throughout tcp_ack() to get
	 * current time. Refactor the code using a new "tcp_acktag_state"
	 * to carry current time, flags, stats like "tcp_sacktag_state".
	 */
	if (delivered)
		tp->delivered_mstamp = tp->tcp_mstamp;

	rs->acked_sacked = delivered;	/* freshly ACKed or SACKed */
	rs->losses = lost;		/* freshly marked lost */
	/* Return an invalid sample if no timing information is available or
	 * in recovery from loss with SACK reneging. Rate samples taken during
	 * a SACK reneging event may overestimate bw by including packets that
	 * were SACKed before the reneg.
	 */
	if (!rs->prior_mstamp || is_sack_reneg) {
		rs->delivered = -1;
		rs->interval_us = -1;
		return;
	}
	rs->delivered   = tp->delivered - rs->prior_delivered;

	rs->delivered_ce = tp->delivered_ce - rs->prior_delivered_ce;
	/* delivered_ce occupies less than 32 bits in the skb control block */
	rs->delivered_ce &= TCPCB_DELIVERED_CE_MASK;

	/* Model sending data and receiving ACKs as separate pipeline phases
	 * for a window. Usually the ACK phase is longer, but with ACK
	 * compression the send phase can be longer. To be safe we use the
	 * longer phase.
	 */
	snd_us = rs->interval_us;				/* send phase */
	ack_us = tcp_stamp_us_delta(tp->tcp_mstamp,
				    rs->prior_mstamp); /* ack phase */
	rs->interval_us = max(snd_us, ack_us);

	/* Record both segment send and ack receive intervals */
	rs->snd_interval_us = snd_us;
	rs->rcv_interval_us = ack_us;

	/* Normally we expect interval_us >= min-rtt.
	 * Note that rate may still be over-estimated when a spuriously
	 * retransmistted skb was first (s)acked because "interval_us"
	 * is under-estimated (up to an RTT). However continuously
	 * measuring the delivery rate during loss recovery is crucial
	 * for connections suffer heavy or prolonged losses.
	 */
	if (unlikely(rs->interval_us < tcp_min_rtt(tp))) {
		if (!rs->is_retrans)
			pr_debug("tcp rate: %ld %d %u %u %u\n",
				 rs->interval_us, rs->delivered,
				 inet_csk(sk)->icsk_ca_state,
				 tp->rx_opt.sack_ok, tcp_min_rtt(tp));
		rs->interval_us = -1;
		return;
	}

	/* Record the last non-app-limited or the highest app-limited bw */
	if (!rs->is_app_limited ||
	    ((u64)rs->delivered * tp->rate_interval_us >=
	     (u64)tp->rate_delivered * rs->interval_us)) {
		tp->rate_delivered = rs->delivered;
		tp->rate_interval_us = rs->interval_us;
		tp->rate_app_limited = rs->is_app_limited;
	}
}

/* If a gap is detected between sends, mark the socket application-limited. */
void tcp_rate_check_app_limited(struct sock *sk)
{