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-rw-r--r--net/ipv4/tcp_rate.c186
1 files changed, 186 insertions, 0 deletions
diff --git a/net/ipv4/tcp_rate.c b/net/ipv4/tcp_rate.c
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+++ b/net/ipv4/tcp_rate.c
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+#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) {
+ tp->first_tx_mstamp = skb->skb_mstamp;
+ tp->delivered_mstamp = skb->skb_mstamp;
+ }
+
+ 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.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 highest prior_delivered count.
+ */
+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);
+
+ if (!scb->tx.delivered_mstamp.v64)
+ return;
+
+ if (!rs->prior_delivered ||
+ after(scb->tx.delivered, rs->prior_delivered)) {
+ 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;
+
+ /* Find the duration of the "send phase" of this window: */
+ rs->interval_us = skb_mstamp_us_delta(
+ &skb->skb_mstamp,
+ &scb->tx.first_tx_mstamp);
+
+ /* Record send time of most recently ACKed packet: */
+ tp->first_tx_mstamp = skb->skb_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.v64 = 0;
+}
+
+/* Update the connection delivery information and generate a rate sample. */
+void tcp_rate_gen(struct sock *sk, u32 delivered, u32 lost,
+ struct skb_mstamp *now, 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 = *now;
+
+ rs->acked_sacked = delivered; /* freshly ACKed or SACKed */
+ rs->losses = lost; /* freshly marked lost */
+ /* Return an invalid sample if no timing information is available. */
+ if (!rs->prior_mstamp.v64) {
+ rs->delivered = -1;
+ rs->interval_us = -1;
+ return;
+ }
+ rs->delivered = tp->delivered - rs->prior_delivered;
+
+ /* 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 = skb_mstamp_us_delta(now, &rs->prior_mstamp); /* ack phase */
+ rs->interval_us = max(snd_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) < tp->snd_cwnd &&
+ /* All lost packets have been retransmitted. */
+ tp->lost_out <= tp->retrans_out)
+ tp->app_limited =
+ (tp->delivered + tcp_packets_in_flight(tp)) ? : 1;
+}