json.hpp中的->size()
时间: 2023-11-23 12:07:59 浏览: 160
在 `json.hpp` 中,`json` 类是一个完全的 C++ JSON 解析器和生成器。`json` 对象表示 JSON 值,它可以是一个数组、对象、字符串、数字、布尔值或空值。
`->` 是 C++ 中指针访问成员的运算符。在 `json.hpp` 中,`json` 对象可以被看做一个指向数组或对象的指针。所以 `->size()` 表示获取指向的数组或对象的大小。当 `json` 对象不是数组或对象时,`->size()` 将会抛出异常。
相关问题
C++怎么使用#include <nlohmann/json.hpp>创建TCP通讯并传输和接收json数据
使用C++创建TCP通讯可以使用socket库,而传输和接收json数据可以使用nlohmann/json库。下面是基本的代码示例:
发送端:
```cpp
#include <iostream>
#include <string>
#include <nlohmann/json.hpp>
#include <sys/socket.h>
#include <arpa/inet.h>
#include <unistd.h>
using json = nlohmann::json;
int main() {
// 创建socket
int sock = socket(AF_INET, SOCK_STREAM, IPPROTO_TCP);
if (sock == -1) {
std::cerr << "socket creation failed" << std::endl;
return 1;
}
// 设置服务器地址和端口号
struct sockaddr_in server_addr;
server_addr.sin_family = AF_INET;
server_addr.sin_addr.s_addr = inet_addr("127.0.0.1");
server_addr.sin_port = htons(12345);
// 连接服务器
if (connect(sock, (struct sockaddr*)&server_addr, sizeof(server_addr)) == -1) {
std::cerr << "connection failed" << std::endl;
return 1;
}
// 构造json数据
json j;
j["name"] = "Alice";
j["age"] = 20;
std::string json_str = j.dump();
// 发送数据
if (send(sock, json_str.c_str(), json_str.size(), 0) == -1) {
std::cerr << "send failed" << std::endl;
return 1;
}
// 关闭socket
close(sock);
return 0;
}
```
接收端:
```cpp
#include <iostream>
#include <string>
#include <nlohmann/json.hpp>
#include <sys/socket.h>
#include <arpa/inet.h>
#include <unistd.h>
using json = nlohmann::json;
int main() {
// 创建socket
int sock = socket(AF_INET, SOCK_STREAM, IPPROTO_TCP);
if (sock == -1) {
std::cerr << "socket creation failed" << std::endl;
return 1;
}
// 设置本机地址和端口号
struct sockaddr_in my_addr;
my_addr.sin_family = AF_INET;
my_addr.sin_addr.s_addr = INADDR_ANY;
my_addr.sin_port = htons(12345);
// 绑定socket
if (bind(sock, (struct sockaddr*)&my_addr, sizeof(my_addr)) == -1) {
std::cerr << "bind failed" << std::endl;
return 1;
}
// 监听socket
if (listen(sock, SOMAXCONN) == -1) {
std::cerr << "listen failed" << std::endl;
return 1;
}
// 接收连接
struct sockaddr_in client_addr;
socklen_t client_addr_len = sizeof(client_addr);
int client_sock = accept(sock, (struct sockaddr*)&client_addr, &client_addr_len);
if (client_sock == -1) {
std::cerr << "accept failed" << std::endl;
return 1;
}
// 接收数据
char buffer[1024];
int length = recv(client_sock, buffer, sizeof(buffer), 0);
if (length == -1) {
std::cerr << "receive failed" << std::endl;
return 1;
}
// 解析json数据
std::string json_str(buffer, length);
json j = json::parse(json_str);
// 输出数据
std::cout << "name: " << j["name"].get<std::string>() << std::endl;
std::cout << "age: " << j["age"].get<int>() << std::endl;
// 关闭socket
close(client_sock);
close(sock);
return 0;
}
```
上面的代码示例中,发送端使用nlohmann/json库构造json数据,并使用socket库发送数据;接收端使用socket库接收数据,并使用nlohmann/json库解析json数据。
mediasoup-demo-v3中添加flexfec需要RtpStreamSend.hpp如何修改
要在mediasoup-demo-v3中添加FlexFEC,需要修改RtpStreamSend.hpp文件中的代码。具体的步骤如下:
1. 打开mediasoup-demo-v3源代码目录中的`worker/src/RtpStreamSend.hpp`文件。
2. 在该文件中找到如下代码:
```
class RtpStreamSend :
public RTC::VideoSinkInterface<webrtc::VideoFrame>,
public RTC::AudioSinkInterface<webrtc::AudioFrame>,
public webrtc::Transport, public RtcpProcessor::Listener
{
public:
RtpStreamSend(
const RTC::scoped_refptr<webrtc::PeerConnectionFactoryInterface>& peerConnectionFactory,
const RTC::scoped_refptr<webrtc::AudioTrackInterface>& audioTrack,
const RTC::scoped_refptr<webrtc::VideoTrackInterface>& videoTrack,
const RTC::scoped_refptr<webrtc::AudioEncoderFactory>& audioEncoderFactory,
const RTC::scoped_refptr<webrtc::VideoEncoderFactory>& videoEncoderFactory,
const RTC::scoped_refptr<webrtc::AudioDecoderFactory>& audioDecoderFactory,
const RTC::scoped_refptr<webrtc::VideoDecoderFactory>& videoDecoderFactory,
const RTC::scoped_refptr<webrtc::AudioProcessing>& audioProcessing,
const RTC::scoped_refptr<webrtc::TaskQueueFactory>& taskQueueFactory,
const RTC::scoped_refptr<webrtc::Call>& call,
std::function<void(const uint8_t*, size_t)> sendVideoCallback,
std::function<void(const uint8_t*, size_t)> sendAudioCallback,
std::function<void(const uint8_t*, size_t)> sendRtpCallback,
std::function<void(const uint8_t*, size_t)> sendRtcpCallback,
std::function<void(const uint8_t*, size_t)> sendSrtpCallback,
std::function<void(const uint8_t*, size_t)> sendSrtcpCallback,
const Json::Value& peerCapabilities,
const Json::Value& appData);
~RtpStreamSend();
// Implementations of webrtc::Transport.
bool SendRtp(const uint8_t* data, size_t len, const webrtc::PacketOptions& options) override;
bool SendRtcp(const uint8_t* data, size_t len) override;
private:
// ...
};
```
3. 在该代码中添加FlexFEC相关的成员变量和方法,如下所示:
```
class RtpStreamSend :
public RTC::VideoSinkInterface<webrtc::VideoFrame>,
public RTC::AudioSinkInterface<webrtc::AudioFrame>,
public webrtc::Transport, public RtcpProcessor::Listener
{
public:
RtpStreamSend(
// ...
);
~RtpStreamSend();
// Implementations of webrtc::Transport.
bool SendRtp(const uint8_t* data, size_t len, const webrtc::PacketOptions& options) override;
bool SendRtcp(const uint8_t* data, size_t len) override;
// FlexFEC related members and methods
private:
std::unique_ptr<webrtc::FlexfecSender> _flexfecSender;
std::map<uint32_t, std::unique_ptr<webrtc::RtpPacketToSend>> _flexfecPackets;
void handleFlexfecPackets(std::map<uint32_t, std::unique_ptr<webrtc::RtpPacketToSend>>& packets);
void sendFlexfecPacket(std::unique_ptr<webrtc::RtpPacketToSend> packet);
};
```
4. 在该代码中实现FlexFEC相关的方法,如下所示:
```
void RtpStreamSend::handleFlexfecPackets(std::map<uint32_t, std::unique_ptr<webrtc::RtpPacketToSend>>& packets)
{
if (_flexfecSender)
{
std::vector<webrtc::RtpSequenceNumberMap::SequenceNumberWithTimestamp> fec_packets;
for (auto& kv : packets)
{
std::unique_ptr<webrtc::RtpPacketToSend>& packet = kv.second;
const webrtc::RtpPacket& rtp_packet = packet->packet;
// Add packet to FlexFEC
if (_flexfecSender->AddRtpPacket(rtp_packet) == webrtc::FlexfecSender::kSuccess)
{
fec_packets.push_back(
webrtc::RtpSequenceNumberMap::SequenceNumberWithTimestamp{
rtp_packet.SequenceNumber(), rtp_packet.Timestamp()});
}
}
if (!fec_packets.empty())
{
// Generate FEC packet
webrtc::RtpPacketToSend fec_packet(_flexfecSender->BuildFlexfecPacket(fec_packets));
if (fec_packet.size() > 0)
{
sendFlexfecPacket(std::make_unique<webrtc::RtpPacketToSend>(std::move(fec_packet)));
}
}
}
}
void RtpStreamSend::sendFlexfecPacket(std::unique_ptr<webrtc::RtpPacketToSend> packet)
{
const uint8_t* data = packet->data();
size_t length = packet->size();
// Send FlexFEC packet via transport
if (_sendRtpCallback)
{
_sendRtpCallback(data, length);
}
}
bool RtpStreamSend::SendRtp(const uint8_t* data, size_t len, const webrtc::PacketOptions& options)
{
// Create RTP packet
webrtc::RtpPacketToSend packet(&_rtpConfig);
if (!packet.Parse(data, len))
{
RTC_LOG(LS_ERROR) << "Failed to parse RTP packet.";
return false;
}
// Check if packet is FEC
if (packet.PayloadType() == _flexfecPayloadType)
{
// Add packet to FlexFEC buffer
uint32_t ssrc = packet.Ssrc();
_flexfecPackets[ssrc] = std::make_unique<webrtc::RtpPacketToSend>(packet);
handleFlexfecPackets(_flexfecPackets);
_flexfecPackets.erase(ssrc);
}
else
{
// Send RTP packet via transport
if (_sendRtpCallback)
{
_sendRtpCallback(data, len);
}
}
return true;
}
bool RtpStreamSend::SendRtcp(const uint8_t* data, size_t len)
{
// ...
}
```
5. 保存文件并重新编译mediasoup-demo-v3。
注意:上述代码只是添加了FlexFEC的基本实现,具体的实现方式可能根据实际情况有所不同。此外,代码中的`_flexfecPayloadType`需要根据实际情况设置为正确的FlexFEC负载类型。
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