CF 2 B. The least round way

本文探讨了利用动态规划方法解决特定路径问题的策略,特别关注于通过两次递归过程找到2和5的幂次方路径,并在路径中包含0时,通过比较路径长度来确定最优解。此外,文章提供了详细的算法实现步骤和代码解析。

题目:The least round way

思路:两次dp分别求2,5出现的幂的一条路径,如果这条路径上有0,那么最后的结果肯定是1(1个0),不然就是只出现2或5的情况;不然就是2和5出现的最小的那种情况。


#include <cstdio>
#include <cstring>
#include <algorithm>
#include <cmath>
#include <iostream>
#define maxn 1010
#define inf 0xfffffff
using namespace std;
int tmp[maxn][maxn],five[maxn][maxn],two[maxn][maxn];
int dp2[maxn][maxn],dp5[maxn][maxn];
char dir2[maxn][maxn],dir5[maxn][maxn];
int n;
int get(int p,int n)
{
    if(n==0)
        return 1;
    int cnt=0;
    while(n%p==0)
    {
        n/=p;
        cnt++;
    }
    return cnt;
}
int zero_x,zero_y;
bool flag;
void print(char dir[][maxn],int x,int y)
{
    //cout<<x<<":"<<y<<endl;
    if(x==1&&y==1)
        return ;
    if(dir[x][y]=='D')
        print(dir,x-1,y);
    else
        print(dir,x,y-1);
    printf("%c",dir[x][y]);
}
int main()
{
    scanf("%d",&n);
    flag=true;
    for(int i=1;i<=n;i++)
        for(int j=1;j<=n;j++)
        {
            scanf("%d",&tmp[i][j]);
            if(tmp[i][j])
            {
                two[i][j]=get(2,tmp[i][j]);
                five[i][j]=get(5,tmp[i][j]);
            }
            else
            {
                flag=false;
                zero_x=i;
                zero_y=j;
            }
        }
    memset(dp2,0,sizeof(dp2));
    memset(dp5,0,sizeof(dp5));
    for(int i=1;i<=n;i++)
        for(int j=1;j<=n;j++)
        {
            if(tmp[i][j]==0)
            {
                dp2[i][j]=inf;
            }
            else if(i==1&&j==1)
            {
                dir2[i][j]=' ';
                dp2[i][j]=two[i][j];
            }
            else if(i==1)
            {
                dp2[i][j]=dp2[i][j-1]+two[i][j];
                dir2[i][j]='R';
            }
            else if(j==1)
            {
                dp2[i][j]=dp2[i-1][j]+two[i][j];
                dir2[i][j]='D';
            }
            else
            {
                if(dp2[i-1][j]<dp2[i][j-1])
                    dir2[i][j]='D';
                else
                    dir2[i][j]='R';
                dp2[i][j]=min(dp2[i-1][j],dp2[i][j-1])+two[i][j];
            }
        }
    for(int i=1;i<=n;i++)
        for(int j=1;j<=n;j++)
        {
            if(tmp[i][j]==0)
            {
                dp5[i][j]=inf;
            }
            else if(i==1&&j==1)
            {
                dir5[i][j]=' ';
                dp5[i][j]=five[i][j];
            }
            else if(j==1)
            {
                dir5[i][j]='D';
                dp5[i][j]=dp5[i-1][j]+five[i][j];
            }
            else if(i==1)
            {
                dir5[i][j]='R';
                dp5[i][j]=dp5[i][j-1]+five[i][j];
            }
            else
            {
                if(dp5[i-1][j]<dp5[i][j-1])
                    dir5[i][j]='D';
                else
                    dir5[i][j]='R';
                dp5[i][j]=min(dp5[i-1][j],dp5[i][j-1])+five[i][j];
            }
        }
    if(!flag)
    {
        if(dp2[n][n]==0)
        {
            printf("0\n");
            print(dir2,n,n);
        }
        else if(dp5[n][n]==0)
        {
            printf("0\n");
            print(dir5,n,n);
        }
        else
        {
            printf("1\n");
            for(int i=2;i<=zero_x;i++)
                printf("D");
            for(int j=2;j<=zero_y;j++)
                printf("R");
            for(int i=zero_x+1;i<=n;i++)
                printf("D");
            for(int j=zero_y+1;j<=n;j++)
                printf("R");
        }
    }
    else
    {
        if(dp2[n][n]>dp5[n][n])
        {
            printf("%d\n",dp5[n][n]);
            print(dir5,n,n);
        }
        else
        {
            printf("%d\n",dp2[n][n]);
            print(dir2,n,n);
        }
    }
    printf("\n");
    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
评论
成就一亿技术人!
拼手气红包6.0元
还能输入1000个字符
 
红包 添加红包
表情包 插入表情
 条评论被折叠 查看
添加红包

请填写红包祝福语或标题

红包个数最小为10个

红包金额最低5元

当前余额3.43前往充值 >
需支付:10.00
成就一亿技术人!
领取后你会自动成为博主和红包主的粉丝 规则
hope_wisdom
发出的红包
实付
使用余额支付
点击重新获取
扫码支付
钱包余额 0

抵扣说明:

1.余额是钱包充值的虚拟货币,按照1:1的比例进行支付金额的抵扣。
2.余额无法直接购买下载,可以购买VIP、付费专栏及课程。

余额充值