Adding upstream version 4.19.249.upstream/4.19.249

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 196 insertions, 0 deletions

diff --git a/net/ipv4/tcp_rate.c b/net/ipv4/tcp_rate.c
new file mode 100644
index 000000000..4dff40dad
--- /dev/null
+++ b/net/ipv4/tcp_rate.c
@@ -0,0 +1,196 @@
+#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)
+		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      = tcp_stamp_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 = 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;
+
+	/* 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) < 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;
+}
+EXPORT_SYMBOL_GPL(tcp_rate_check_app_limited);