210 lines
8.2 KiB
C
210 lines
8.2 KiB
C
// 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.
|
|
*/
|
|
|
|
/* Snapshot the current delivery information in the skb, to generate
|
|
* a rate sample later when the skb is (s)acked in tcp_rate_skb_delivered().
|
|
*/
|
|
void tcp_rate_skb_sent(struct sock *sk, struct sk_buff *skb)
|
|
{
|
|
struct tcp_sock *tp = tcp_sk(sk);
|
|
|
|
/* In general we need to start delivery rate samples from the
|
|
* time we received the most recent ACK, to ensure we include
|
|
* the full time the network needs to deliver all in-flight
|
|
* packets. If there are no packets in flight yet, then we
|
|
* know that any ACKs after now indicate that the network was
|
|
* able to deliver those packets completely in the sampling
|
|
* interval between now and the next ACK.
|
|
*
|
|
* Note that we use packets_out instead of tcp_packets_in_flight(tp)
|
|
* because the latter is a guess based on RTO and loss-marking
|
|
* heuristics. We don't want spurious RTOs or loss markings to cause
|
|
* a spuriously small time interval, causing a spuriously high
|
|
* bandwidth estimate.
|
|
*/
|
|
if (!tp->packets_out) {
|
|
u64 tstamp_us = tcp_skb_timestamp_us(skb);
|
|
|
|
tp->first_tx_mstamp = tstamp_us;
|
|
tp->delivered_mstamp = tstamp_us;
|
|
}
|
|
|
|
TCP_SKB_CB(skb)->tx.first_tx_mstamp = tp->first_tx_mstamp;
|
|
TCP_SKB_CB(skb)->tx.delivered_mstamp = tp->delivered_mstamp;
|
|
TCP_SKB_CB(skb)->tx.delivered = tp->delivered;
|
|
TCP_SKB_CB(skb)->tx.delivered_ce = tp->delivered_ce;
|
|
TCP_SKB_CB(skb)->tx.is_app_limited = tp->app_limited ? 1 : 0;
|
|
}
|
|
|
|
/* When an skb is sacked or acked, we fill in the rate sample with the (prior)
|
|
* delivery information when the skb was last transmitted.
|
|
*
|
|
* If an ACK (s)acks multiple skbs (e.g., stretched-acks), this function is
|
|
* called multiple times. We favor the information from the most recently
|
|
* sent skb, i.e., the skb with the most recently sent time and the highest
|
|
* sequence.
|
|
*/
|
|
void tcp_rate_skb_delivered(struct sock *sk, struct sk_buff *skb,
|
|
struct rate_sample *rs)
|
|
{
|
|
struct tcp_sock *tp = tcp_sk(sk);
|
|
struct tcp_skb_cb *scb = TCP_SKB_CB(skb);
|
|
u64 tx_tstamp;
|
|
|
|
if (!scb->tx.delivered_mstamp)
|
|
return;
|
|
|
|
tx_tstamp = tcp_skb_timestamp_us(skb);
|
|
if (!rs->prior_delivered ||
|
|
tcp_skb_sent_after(tx_tstamp, tp->first_tx_mstamp,
|
|
scb->end_seq, rs->last_end_seq)) {
|
|
rs->prior_delivered_ce = scb->tx.delivered_ce;
|
|
rs->prior_delivered = scb->tx.delivered;
|
|
rs->prior_mstamp = scb->tx.delivered_mstamp;
|
|
rs->is_app_limited = scb->tx.is_app_limited;
|
|
rs->is_retrans = scb->sacked & TCPCB_RETRANS;
|
|
rs->last_end_seq = scb->end_seq;
|
|
|
|
/* Record send time of most recently ACKed packet: */
|
|
tp->first_tx_mstamp = tx_tstamp;
|
|
/* Find the duration of the "send phase" of this window: */
|
|
rs->interval_us = tcp_stamp_us_delta(tp->first_tx_mstamp,
|
|
scb->tx.first_tx_mstamp);
|
|
|
|
}
|
|
/* Mark off the skb delivered once it's sacked to avoid being
|
|
* used again when it's cumulatively acked. For acked packets
|
|
* we don't need to reset since it'll be freed soon.
|
|
*/
|
|
if (scb->sacked & TCPCB_SACKED_ACKED)
|
|
scb->tx.delivered_mstamp = 0;
|
|
}
|
|
|
|
/* 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)
|
|
{
|
|
struct tcp_sock *tp = tcp_sk(sk);
|
|
|
|
if (/* We have less than one packet to send. */
|
|
tp->write_seq - tp->snd_nxt < tp->mss_cache &&
|
|
/* Nothing in sending host's qdisc queues or NIC tx queue. */
|
|
sk_wmem_alloc_get(sk) < SKB_TRUESIZE(1) &&
|
|
/* We are not limited by CWND. */
|
|
tcp_packets_in_flight(tp) < tcp_snd_cwnd(tp) &&
|
|
/* All lost packets have been retransmitted. */
|
|
tp->lost_out <= tp->retrans_out)
|
|
tp->app_limited =
|
|
(tp->delivered + tcp_packets_in_flight(tp)) ? : 1;
|
|
}
|
|
EXPORT_SYMBOL_GPL(tcp_rate_check_app_limited);
|