diff options
Diffstat (limited to 'net/ipv4/tcp_rate.c')
| -rw-r--r-- | net/ipv4/tcp_rate.c | 186 |
1 files changed, 186 insertions, 0 deletions
diff --git a/net/ipv4/tcp_rate.c b/net/ipv4/tcp_rate.c new file mode 100644 index 000000000000..9be1581a5a08 --- /dev/null +++ b/net/ipv4/tcp_rate.c @@ -0,0 +1,186 @@ +#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; +} |