Ada and Queue

本文介绍了一个关于Ada the Ladybug的编程挑战,她使用队列来管理任务,并且能够从队列的前后两端取出任务、反转队列以及将任务加入到队列的不同位置。文章通过一个具体的例子展示了如何使用双端队列(deque)来解决这个问题。

Ada and Queue

 

Ada the Ladybug has many things to do. She puts them into her queue. Anyway she is very indecisive, so sometime she uses the top, sometime the back and sometime she decides to reverses it.

Input

The first line consists of 1 ≤ Q ≤ 106, number of queries. Each of them contains one of following commands

back - Print number from back and then erase it

front - Print number from front and then erase it

reverse - Reverses all elements in queue

push_back N - Add element N to back

toFront N - Put element N to front

All numbers will be 0 ≤ N ≤ 100

Output

For each back/front query print appropriate number.

If you would get this type of query and the queue would be empty, print "No job for Ada?" instead.

Example Input

15
toFront 93
front
back
reverse
back
reverse
toFront 80
push_back 53
push_back 50
front
front
reverse
push_back 66
reverse
front

Example Output

93
No job for Ada?
No job for Ada?
80
53
66
【分析】用deque模拟。

反转后,输入也要反转。

#include <iostream>
#include <cstdio>
#include <stack>
#include <cstring>
using namespace std;
const int maxn = 1e4 + 10;
typedef long long LL;
#define cl(a,b) memset(a,b,sizeof a);
int main()
{
    int n;
    while(~scanf("%d",&n)){
        deque<int> q;
        char s[100];
        int x;
        int flag=1;
        while(n--){
            scanf("%s",s);
            if(s[0] == 'b'){
                if(q.empty())
                    puts("No job for Ada?");
                else{
                    if(flag){
                        printf("%d\n",q.back());
                        q.pop_back();
                    }
                    else{
                        printf("%d\n",q.front());
                        q.pop_front();
                    }
                }
            }
            else
            if(s[0] == 'f'){
                if(q.empty())
                    puts("No job for Ada?");
                else{
                    if(!flag){
                        printf("%d\n",q.back());
                        q.pop_back();
                    }
                    else{
                        printf("%d\n",q.front());
                        q.pop_front();
                    }
                }
            }
            else
            if(s[0] == 'r'){
                flag = 1-flag;
            }
            else
            if(s[0] == 'p'){
                scanf("%d",&x);
                if(flag)
                    q.push_back(x);
                else
                    q.push_front(x);
            }
            else
            if(s[0] == 't'){
                scanf("%d",&x);
                if(flag)
                    q.push_front(x);
                else
                    q.push_back(x);
            }
        }
    }
    return 0;
}




<!-- go/cmark --> <!--* freshness: {owner: 'sprang' reviewed: '2021-04-12'} *--> # Paced Sending The paced sender, often referred to as just the "pacer", is a part of the WebRTC RTP stack used primarily to smooth the flow of packets sent onto the network. ## Background Consider a video stream at 5Mbps and 30fps. This would in an ideal world result in each frame being ~21kB large and packetized into 18 RTP packets. While the average bitrate over say a one second sliding window would be a correct 5Mbps, on a shorter time scale it can be seen as a burst of 167Mbps every 33ms, each followed by a 32ms silent period. Further, it is quite common that video encoders overshoot the target frame size in case of sudden movement especially dealing with screensharing. Frames being 10x or even 100x larger than the ideal size is an all too real scenario. These packet bursts can cause several issues, such as congesting networks and causing buffer bloat or even packet loss. Most sessions have more than one media stream, e.g. a video and an audio track. If you put a frame on the wire in one go, and those packets take 100ms to reach the other side - that means you have now blocked any audio packets from reaching the remote end in time as well. The paced sender solves this by having a buffer in which media is queued, and then using a _leaky bucket_ algorithm to pace them onto the network. The buffer contains separate fifo streams for all media tracks so that e.g. audio can be prioritized over video - and equal prio streams can be sent in a round-robin fashion to avoid any one stream blocking others. Since the pacer is in control of the bitrate sent on the wire, it is also used to generate padding in cases where a minimum send rate is required - and to generate packet trains if bitrate probing is used. ## Life of a Packet The typical path for media packets when using the paced sender looks something like this: 1. `RTPSenderVideo` or `RTPSenderAudio` packetizes media into RTP packets. 2. The packets are sent to the [RTPSender] class for transmission. 3. The pacer is called via [RtpPacketSender] interface to enqueue the packet batch. 4. The packets are put into a queue within the pacer awaiting opportune moments to send them. 5. At a calculated time, the pacer calls the `PacingController::PacketSender()` callback method, normally implemented by the [PacketRouter] class. 6. The router forwards the packet to the correct RTP module based on the packet's SSRC, and in which the `RTPSenderEgress` class makes final time stamping, potentially records it for retransmissions etc. 7. The packet is sent to the low-level `Transport` interface, after which it is now out of scope. Asynchronously to this, the estimated available send bandwidth is determined - and the target send rate is set on the `RtpPacketPacer` via the `void SetPacingRates(DataRate pacing_rate, DataRate padding_rate)` method. ## Packet Prioritization The pacer prioritized packets based on two criteria: * Packet type, with most to least prioritized: 1. Audio 2. Retransmissions 3. Video and FEC 4. Padding * Enqueue order The enqueue order is enforced on a per stream (SSRC) basis. Given equal priority, the [RoundRobinPacketQueue] alternates between media streams to ensure no stream needlessly blocks others. ## Implementations The main class to use is called [TaskQueuePacedSender]. It uses a task queue to manage thread safety and schedule delayed tasks, but delegates most of the actual work to the `PacingController` class. This way, it's possible to develop a custom pacer with different scheduling mechanism - but ratain the same pacing logic. ## The Packet Router An adjacent component called [PacketRouter] is used to route packets coming out of the pacer and into the correct RTP module. It has the following functions: * The `SendPacket` method looks up an RTP module with an SSRC corresponding to the packet for further routing to the network. * If send-side bandwidth estimation is used, it populates the transport-wide sequence number extension. * Generate padding. Modules supporting payload-based padding are prioritized, with the last module to have sent media always being the first choice. * Returns any generated FEC after having sent media. * Forwards REMB and/or TransportFeedback messages to suitable RTP modules. At present the FEC is generated on a per SSRC basis, so is always returned from an RTP module after sending media. Hopefully one day we will support covering multiple streams with a single FlexFEC stream - and the packet router is the likely place for that FEC generator to live. It may even be used for FEC padding as an alternative to RTX. ## The API The section outlines the classes and methods relevant to a few different use cases of the pacer. ### Packet sending For sending packets, use `RtpPacketSender::EnqueuePackets(std::vector<std::unique_ptr<RtpPacketToSend>> packets)` The pacer takes a `PacingController::PacketSender` as constructor argument, this callback is used when it's time to actually send packets. ### Send rates To control the send rate, use `void SetPacingRates(DataRate pacing_rate, DataRate padding_rate)` If the packet queue becomes empty and the send rate drops below `padding_rate`, the pacer will request padding packets from the `PacketRouter`. In order to completely suspend/resume sending data (e.g. due to network availability), use the `Pause()` and `Resume()` methods. The specified pacing rate may be overriden in some cases, e.g. due to extreme encoder overshoot. Use `void SetQueueTimeLimit(TimeDelta limit)` to specify the longest time you want packets to spend waiting in the pacer queue (pausing excluded). The actual send rate may then be increased past the pacing_rate to try to make the _average_ queue time less than that requested limit. The rationale for this is that if the send queue is say longer than three seconds, it's better to risk packet loss and then try to recover using a key-frame rather than cause severe delays. ### Bandwidth estimation If the bandwidth estimator supports bandwidth probing, it may request a cluster of packets to be sent at a specified rate in order to gauge if this causes increased delay/loss on the network. Use the `void CreateProbeCluster(...)` method - packets sent via this `PacketRouter` will be marked with the corresponding cluster_id in the attached `PacedPacketInfo` struct. If congestion window pushback is used, the state can be updated using `SetCongestionWindow()` and `UpdateOutstandingData()`. A few more methods control how we pace: * `SetAccountForAudioPackets()` determines if audio packets count into bandwidth consumed. * `SetIncludeOverhead()` determines if the entire RTP packet size counts into bandwidth used (otherwise just media payload). * `SetTransportOverhead()` sets an additional data size consumed per packet, representing e.g. UDP/IP headers. ### Stats Several methods are used to gather statistics in pacer state: * `OldestPacketWaitTime()` time since the oldest packet in the queue was added. * `QueueSizeData()` total bytes currently in the queue. * `FirstSentPacketTime()` absolute time the first packet was sent. * `ExpectedQueueTime()` total bytes in the queue divided by the send rate. [RTPSender]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.h;drc=77ee8542dd35d5143b5788ddf47fb7cdb96eb08e [RtpPacketSender]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/rtp_packet_sender.h;drc=ea55b0872f14faab23a4e5dbcb6956369c8ed5dc [RtpPacketPacer]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/pacing/rtp_packet_pacer.h;drc=e7bc3a347760023dd4840cf6ebdd1e6c8592f4d7 [PacketRouter]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/pacing/packet_router.h;drc=3d2210876e31d0bb5c7de88b27fd02ceb1f4e03e [TaskQueuePacedSender]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/pacing/task_queue_paced_sender.h;drc=5051693ada61bc7b78855c6fb3fa87a0394fa813 [RoundRobinPacketQueue]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/pacing/round_robin_packet_queue.h;drc=b571ff48f8fe07678da5a854cd6c3f5dde02855f 翻译
最新发布
12-03
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