191 lines
7.6 KiB
C
191 lines
7.6 KiB
C
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#include <net/tcp.h>
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/* The bandwidth estimator estimates the rate at which the network
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* can currently deliver outbound data packets for this flow. At a high
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* level, it operates by taking a delivery rate sample for each ACK.
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*
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* A rate sample records the rate at which the network delivered packets
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* for this flow, calculated over the time interval between the transmission
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* of a data packet and the acknowledgment of that packet.
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*
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* Specifically, over the interval between each transmit and corresponding ACK,
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* the estimator generates a delivery rate sample. Typically it uses the rate
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* at which packets were acknowledged. However, the approach of using only the
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* acknowledgment rate faces a challenge under the prevalent ACK decimation or
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* compression: packets can temporarily appear to be delivered much quicker
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* than the bottleneck rate. Since it is physically impossible to do that in a
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* sustained fashion, when the estimator notices that the ACK rate is faster
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* than the transmit rate, it uses the latter:
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*
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* send_rate = #pkts_delivered/(last_snd_time - first_snd_time)
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* ack_rate = #pkts_delivered/(last_ack_time - first_ack_time)
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* bw = min(send_rate, ack_rate)
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*
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* Notice the estimator essentially estimates the goodput, not always the
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* network bottleneck link rate when the sending or receiving is limited by
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* other factors like applications or receiver window limits. The estimator
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* deliberately avoids using the inter-packet spacing approach because that
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* approach requires a large number of samples and sophisticated filtering.
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*
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* TCP flows can often be application-limited in request/response workloads.
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* The estimator marks a bandwidth sample as application-limited if there
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* was some moment during the sampled window of packets when there was no data
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* ready to send in the write queue.
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*/
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/* Snapshot the current delivery information in the skb, to generate
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* a rate sample later when the skb is (s)acked in tcp_rate_skb_delivered().
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*/
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void tcp_rate_skb_sent(struct sock *sk, struct sk_buff *skb)
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{
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struct tcp_sock *tp = tcp_sk(sk);
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/* In general we need to start delivery rate samples from the
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* time we received the most recent ACK, to ensure we include
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* the full time the network needs to deliver all in-flight
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* packets. If there are no packets in flight yet, then we
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* know that any ACKs after now indicate that the network was
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* able to deliver those packets completely in the sampling
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* interval between now and the next ACK.
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*
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* Note that we use packets_out instead of tcp_packets_in_flight(tp)
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* because the latter is a guess based on RTO and loss-marking
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* heuristics. We don't want spurious RTOs or loss markings to cause
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* a spuriously small time interval, causing a spuriously high
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* bandwidth estimate.
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*/
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if (!tp->packets_out) {
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tp->first_tx_mstamp = skb->skb_mstamp;
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tp->delivered_mstamp = skb->skb_mstamp;
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}
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TCP_SKB_CB(skb)->tx.first_tx_mstamp = tp->first_tx_mstamp;
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TCP_SKB_CB(skb)->tx.delivered_mstamp = tp->delivered_mstamp;
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TCP_SKB_CB(skb)->tx.delivered = tp->delivered;
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TCP_SKB_CB(skb)->tx.is_app_limited = tp->app_limited ? 1 : 0;
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}
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/* When an skb is sacked or acked, we fill in the rate sample with the (prior)
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* delivery information when the skb was last transmitted.
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*
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* If an ACK (s)acks multiple skbs (e.g., stretched-acks), this function is
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* called multiple times. We favor the information from the most recently
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* sent skb, i.e., the skb with the highest prior_delivered count.
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*/
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void tcp_rate_skb_delivered(struct sock *sk, struct sk_buff *skb,
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struct rate_sample *rs)
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{
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struct tcp_sock *tp = tcp_sk(sk);
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struct tcp_skb_cb *scb = TCP_SKB_CB(skb);
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if (!scb->tx.delivered_mstamp.v64)
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return;
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if (!rs->prior_delivered ||
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after(scb->tx.delivered, rs->prior_delivered)) {
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rs->prior_delivered = scb->tx.delivered;
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rs->prior_mstamp = scb->tx.delivered_mstamp;
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rs->is_app_limited = scb->tx.is_app_limited;
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rs->is_retrans = scb->sacked & TCPCB_RETRANS;
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/* Find the duration of the "send phase" of this window: */
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rs->interval_us = skb_mstamp_us_delta(
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&skb->skb_mstamp,
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&scb->tx.first_tx_mstamp);
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/* Record send time of most recently ACKed packet: */
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tp->first_tx_mstamp = skb->skb_mstamp;
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}
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/* Mark off the skb delivered once it's sacked to avoid being
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* used again when it's cumulatively acked. For acked packets
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* we don't need to reset since it'll be freed soon.
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*/
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if (scb->sacked & TCPCB_SACKED_ACKED)
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scb->tx.delivered_mstamp.v64 = 0;
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}
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/* Update the connection delivery information and generate a rate sample. */
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void tcp_rate_gen(struct sock *sk, u32 delivered, u32 lost,
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bool is_sack_reneg, struct skb_mstamp *now, struct rate_sample *rs)
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{
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struct tcp_sock *tp = tcp_sk(sk);
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u32 snd_us, ack_us;
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/* Clear app limited if bubble is acked and gone. */
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if (tp->app_limited && after(tp->delivered, tp->app_limited))
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tp->app_limited = 0;
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/* TODO: there are multiple places throughout tcp_ack() to get
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* current time. Refactor the code using a new "tcp_acktag_state"
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* to carry current time, flags, stats like "tcp_sacktag_state".
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*/
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if (delivered)
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tp->delivered_mstamp = *now;
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rs->acked_sacked = delivered; /* freshly ACKed or SACKed */
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rs->losses = lost; /* freshly marked lost */
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/* Return an invalid sample if no timing information is available or
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* in recovery from loss with SACK reneging. Rate samples taken during
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* a SACK reneging event may overestimate bw by including packets that
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* were SACKed before the reneg.
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*/
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if (!rs->prior_mstamp.v64 || is_sack_reneg) {
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rs->delivered = -1;
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rs->interval_us = -1;
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return;
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}
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rs->delivered = tp->delivered - rs->prior_delivered;
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/* Model sending data and receiving ACKs as separate pipeline phases
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* for a window. Usually the ACK phase is longer, but with ACK
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* compression the send phase can be longer. To be safe we use the
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* longer phase.
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*/
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snd_us = rs->interval_us; /* send phase */
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ack_us = skb_mstamp_us_delta(now, &rs->prior_mstamp); /* ack phase */
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rs->interval_us = max(snd_us, ack_us);
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/* Normally we expect interval_us >= min-rtt.
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* Note that rate may still be over-estimated when a spuriously
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* retransmistted skb was first (s)acked because "interval_us"
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* is under-estimated (up to an RTT). However continuously
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* measuring the delivery rate during loss recovery is crucial
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* for connections suffer heavy or prolonged losses.
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*/
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if (unlikely(rs->interval_us < tcp_min_rtt(tp))) {
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if (!rs->is_retrans)
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pr_debug("tcp rate: %ld %d %u %u %u\n",
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rs->interval_us, rs->delivered,
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inet_csk(sk)->icsk_ca_state,
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tp->rx_opt.sack_ok, tcp_min_rtt(tp));
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rs->interval_us = -1;
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return;
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}
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/* Record the last non-app-limited or the highest app-limited bw */
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if (!rs->is_app_limited ||
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((u64)rs->delivered * tp->rate_interval_us >=
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(u64)tp->rate_delivered * rs->interval_us)) {
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tp->rate_delivered = rs->delivered;
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tp->rate_interval_us = rs->interval_us;
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tp->rate_app_limited = rs->is_app_limited;
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}
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}
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/* If a gap is detected between sends, mark the socket application-limited. */
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void tcp_rate_check_app_limited(struct sock *sk)
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{
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struct tcp_sock *tp = tcp_sk(sk);
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if (/* We have less than one packet to send. */
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tp->write_seq - tp->snd_nxt < tp->mss_cache &&
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/* Nothing in sending host's qdisc queues or NIC tx queue. */
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sk_wmem_alloc_get(sk) < SKB_TRUESIZE(1) &&
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/* We are not limited by CWND. */
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tcp_packets_in_flight(tp) < tp->snd_cwnd &&
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/* All lost packets have been retransmitted. */
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tp->lost_out <= tp->retrans_out)
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tp->app_limited =
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(tp->delivered + tcp_packets_in_flight(tp)) ? : 1;
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}
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