forked from Minki/linux
4929c9428a
Congestion control algorithms, which access the rate sample through the tcp_cong_control function, only have access to the maximum of the send and receive interval, for cases where the acknowledgment rate may be inaccurate due to ACK compression or decimation. Algorithms may want to use send rates and receive rates as separate signals. Signed-off-by: Deepti Raghavan <deeptir@mit.edu> Acked-by: Neal Cardwell <ncardwell@google.com> Signed-off-by: Eric Dumazet <edumazet@google.com> Signed-off-by: David S. Miller <davem@davemloft.net>
197 lines
7.7 KiB
C
197 lines
7.7 KiB
C
#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)
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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 = tcp_stamp_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 = 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 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 = tp->tcp_mstamp;
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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 || 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 = tcp_stamp_us_delta(tp->tcp_mstamp,
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rs->prior_mstamp); /* ack phase */
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rs->interval_us = max(snd_us, ack_us);
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/* Record both segment send and ack receive intervals */
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rs->snd_interval_us = snd_us;
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rs->rcv_interval_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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EXPORT_SYMBOL_GPL(tcp_rate_check_app_limited);
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