/* 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);
/* 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;
/* 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);
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