rtsp協議之.c++實現，rtsp，rtp協議框架，模擬多路音視頻h264，265，aac，數據幀傳輸，接收（二）

rtsp協議之.c++實現，rtsp，rtp協議框架，模擬多路音視頻h264，265，aac，數據幀傳輸，接收（二）

1、RTSP 服務器核心：處理 RTSP 會話管理、請求解析和響應生成

2、媒體源抽象：定義了通用媒體源接口，支持不同類型的媒體流
具體媒體源實現：
H264MediaSource：模擬 H.264 視頻流
H265MediaSource：模擬 H.265 視頻流
AACMediaSource：模擬 AAC 音頻流

3、RTP 包生成：根據不同媒體類型生成相應的 RTP 包
使用時，只需創建 RTSP 服務器實例，添加多個媒體源，然后啟動服務器即可。服務器會模擬多個相機的碼流，可以通過 RTSP 客戶端訪問這些模擬的媒體流。

在給出例子前，先解釋一些概念。

1、會話id
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// 偽代碼示例（需在實際代碼中實現）
std::string RTSPServer::generateSessionId() {// 生成唯一會話ID（例如隨機數+時間戳）static std::atomic<uint32_t> counter(0);return std::to_string(time(nullptr)) + "_" + std::to_string(counter++);
}void RTSPServer::handleSetupRequest(const RTSPRequest& request, RTSPResponse& response) {// 為客戶端分配會話IDstd::string sessionId = generateSessionId();response.setHeader("Session", sessionId);// 保存會話信息（如端口、媒體源等）SessionInfo session;session.id = sessionId;session.clientPort = parseClientPort(request.getHeader("Transport"));// ... 其他會話信息// 存儲會話（通常用std::map）m_sessions[sessionId] = session;
}

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2、
在視頻流處理中，區分關鍵幀（I 幀）、預測幀（P 幀）和分片（Fragmentation）是保證視頻正確解碼和高效傳輸的關鍵。以下從原理、協議和實際應用角度詳細解釋：
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在你的 RTSP 模擬器中，可通過以下方式模擬幀類型和分片：

class MediaSource {
public:enum FrameType {FRAME_I,FRAME_P,FRAME_B};// 生成模擬幀（需在子類實現）virtual bool getNextFrame(uint8_t*& data, size_t& size, FrameType& type) = 0;
};class H264MediaSource : public MediaSource {
public:bool getNextFrame(uint8_t*& data, size_t& size, FrameType& type) override {// 模擬：每30幀生成一個I幀，其余為P幀static int frameCounter = 0;if (frameCounter++ % 30 == 0) {type = FRAME_I;generateIFrame(data, size);} else {type = FRAME_P;generatePFrame(data, size);}return true;}
};

分片處理

class RTPGenerator {
public:void generateRtpPackets(const uint8_t* nalData, size_t nalSize, FrameType frameType, std::vector<RtpPacket>& packets) {const int MAX_RTP_PAYLOAD = 1400;if (nalSize <= MAX_RTP_PAYLOAD) {// 小NAL單元直接封裝packets.emplace_back(nalData, nalSize, frameType, false);} else {// 分片封裝const uint8_t* payload = nalData + 1;  // 跳過NAL頭size_t payloadSize = nalSize - 1;size_t offset = 0;// 第一個分片packets.emplace_back(createFirstFragment(nalData, payload, offset, MAX_RTP_PAYLOAD));offset += MAX_RTP_PAYLOAD - 2;  // 減去FU頭大小// 中間分片while (offset + MAX_RTP_PAYLOAD - 2 < payloadSize) {packets.emplace_back(createMiddleFragment(nalData, payload, offset, MAX_RTP_PAYLOAD));offset += MAX_RTP_PAYLOAD - 2;}// 最后一個分片packets.emplace_back(createLastFragment(nalData, payload, offset, payloadSize - offset));}}
};

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組合后就是如下
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或者

// H.264 NAL單元分片為RTP包
void fragmentNalUnit(const uint8_t* nalData, size_t nalSize, std::vector<RtpPacket>& packets) {const int MAX_PAYLOAD_SIZE = 1400;  // MTU - RTP頭(12字節) - FU頭(2字節)const uint8_t FU_INDICATOR = 0x7C;  // 固定為0x7C (Type=28)// 提取原始NAL類型（低5位）uint8_t originalNalType = nalData[0] & 0x1F;// 計算需要的分片數size_t numFragments = (nalSize - 1 + MAX_PAYLOAD_SIZE - 1) / MAX_PAYLOAD_SIZE;size_t offset = 1;  // 跳過原始NAL頭(1字節)for (size_t i = 0; i < numFragments; i++) {size_t fragmentSize = std::min(nalSize - offset, MAX_PAYLOAD_SIZE);bool isFirst = (i == 0);bool isLast = (i == numFragments - 1);// 創建FU headeruint8_t fuHeader = originalNalType;if (isFirst) fuHeader |= 0x80;  // 設置起始位if (isLast) fuHeader |= 0x40;   // 設置結束位// 創建RTP包RtpPacket packet;packet.payloadSize = 2 + fragmentSize;  // FU指示符(1) + FU頭(1) + 數據packet.payload[0] = FU_INDICATOR;packet.payload[1] = fuHeader;memcpy(packet.payload + 2, nalData + offset, fragmentSize);packets.push_back(packet);offset += fragmentSize;}
}

接收端重組邏輯
接收端需要根據 FU-A 頭信息重組原始 NAL 單元
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或者

// 重組H.264 FU-A分片
bool reassembleFuA(const uint8_t* fragment, size_t fragSize, bool isFirst, bool isLast, std::vector<uint8_t>& nalBuffer) {if (isFirst) {// 開始新的NAL單元nalBuffer.clear();// 從FU indicator中提取NRIuint8_t nri = (fragment[0] & 0x60);// 從FU header中提取原始NAL類型uint8_t originalType = fragment[1] & 0x1F;// 重建原始NAL頭uint8_t originalNalHeader = nri | originalType;nalBuffer.push_back(originalNalHeader);}// 添加分片數據（跳過FU indicator和FU header）nalBuffer.insert(nalBuffer.end(), fragment + 2, fragment + fragSize);// 返回是否完成重組return isLast;
}

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在 H.264/H.265 視頻流中，判斷一幀是 I 幀、P 幀還是 B 幀需要分析 NAL 單元的類型和內容。以下是具體的判斷方法：

  1. 通過 NAL 單元類型判斷（最直接方法）
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// 判斷H.264 NAL單元是否為I幀
bool isH264IFrame(const uint8_t* nalData, size_t nalSize) {if (nalSize < 1) return false;// 提取NAL頭部的Type字段（低5位）uint8_t nalType = nalData[0] & 0x1F;// 如果是IDR圖像（Type=5），直接判定為I幀if (nalType == 5) return true;// 對于Type=1（非IDR圖像），需要進一步分析slice_typeif (nalType == 1 && nalSize >= 5) {// 定位到slice_header中的slice_type字段// 注意：實際解析需要考慮NAL單元的RBSP解碼和EBSP去封裝uint8_t sliceType = parseSliceType(nalData + 1, nalSize - 1);return (sliceType == 2 || sliceType == 7);  // slice_type=2或7表示I幀}return false;
}// 判斷H.264 NAL單元是否為B幀
bool isH264BFrame(const uint8_t* nalData, size_t nalSize) {if (nalSize < 1) return false;uint8_t nalType = nalData[0] & 0x1F;// 對于Type=1（非IDR圖像），分析slice_typeif (nalType == 1 && nalSize >= 5) {uint8_t sliceType = parseSliceType(nalData + 1, nalSize - 1);return (sliceType == 1 || sliceType == 6);  // slice_type=1或6表示B幀}return false;
}// 解析slice_type字段（簡化版）
uint8_t parseSliceType(const uint8_t* data, size_t size) {// 實際解析需要考慮：// 1. RBSP解碼（移除防止競爭字節0x03）// 2. EBSP去封裝// 3. 按位解析slice_header結構// 此處為簡化示例，實際代碼需根據H.264標準實現return data[0] & 0x07;  // 假設slice_type在第一個字節的低3位
}

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// 處理RTP包回調
void handleRtpPacket(const uint8_t* rtpData, size_t rtpSize) {// 解析RTP頭部uint16_t seqNum = (rtpData[2] << 8) | rtpData[3];uint32_t timestamp = (rtpData[4] << 24) | (rtpData[5] << 16) | (rtpData[6] << 8) | rtpData[7];// 提取RTP負載const uint8_t* payload = rtpData + 12;  // RTP頭部默認12字節size_t payloadSize = rtpSize - 12;// 判斷是否為FU-A分片if (payloadSize >= 2 && (payload[0] & 0x1F) == 28) {  // FU-A類型uint8_t fuHeader = payload[1];bool isStart = (fuHeader & 0x80) != 0;bool isEnd = (fuHeader & 0x40) != 0;if (isStart) {// 開始新的NAL單元重組nalBuffer.clear();// 重建原始NAL頭部uint8_t originalNalHeader = (payload[0] & 0xE0) | (fuHeader & 0x1F);nalBuffer.push_back(originalNalHeader);}// 添加分片數據（跳過FU indicator和FU header）nalBuffer.insert(nalBuffer.end(), payload + 2, payload + payloadSize);if (isEnd) {// NAL單元重組完成，判斷幀類型if (isH264IFrame(nalBuffer.data(), nalBuffer.size())) {printf("收到I幀，時間戳: %u\n", timestamp);} else if (isH264BFrame(nalBuffer.data(), nalBuffer.size())) {printf("收到B幀，時間戳: %u\n", timestamp);} else {printf("收到P幀，時間戳: %u\n", timestamp);}}} else {// 非分片NAL單元，直接判斷if (isH264IFrame(payload, payloadSize)) {printf("收到完整I幀，時間戳: %u\n", timestamp);}}
}

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rtsp_server.h

#ifndef RTSP_SERVER_H
#define RTSP_SERVER_H#include <string>
#include <vector>
#include <map>
#include <thread>
#include <mutex>
#include <memory>
#include <cstdint>
#include <functional>
#include <unordered_map>
#include <queue>
#include <condition_variable>class RTSPSession;
class MediaSource;class RTSPServer {
public:RTSPServer(int port = 8554);~RTSPServer();bool start();void stop();void addMediaSource(const std::string& streamName, std::shared_ptr<MediaSource> source);void removeMediaSource(const std::string& streamName);private:void handleClient(int clientSocket);void handleRTSPRequest(int clientSocket, const std::string& request);std::shared_ptr<RTSPSession> createSession(const std::string& streamName);int m_serverSocket;bool m_running;std::thread m_serverThread;std::mutex m_sessionsMutex;std::map<std::string, std::shared_ptr<RTSPSession>> m_sessions;std::unordered_map<std::string, std::shared_ptr<MediaSource>> m_mediaSources;int m_nextSessionId;
};class RTSPSession {
public:RTSPSession(int sessionId, std::shared_ptr<MediaSource> mediaSource);~RTSPSession();int getSessionId() const;std::string getSessionDescription() const;void setup(int clientRtpPort, int clientRtcpPort);void play();void pause();void teardown();private:void sendRTPData();int m_sessionId;std::shared_ptr<MediaSource> m_mediaSource;int m_clientRtpPort;int m_clientRtcpPort;int m_rtpSocket;bool m_playing;std::thread m_senderThread;bool m_senderRunning;
};class MediaSource {
public:virtual ~MediaSource() = default;virtual std::string getMimeType() const = 0;virtual std::string getMediaDescription() const = 0;virtual std::string getAttribute() const = 0;virtual void getNextFrame(std::vector<uint8_t>& frame, uint32_t& timestamp, bool& marker) = 0;virtual uint32_t getClockRate() const = 0;
};#endif // RTSP_SERVER_H

rtsp_server.cpp

#include "rtsp_server.h"
#include <iostream>
#include <sstream>
#include <cstring>
#include <sys/socket.h>
#include <arpa/inet.h>
#include <unistd.h>
#include <netdb.h>
#include <ctime>
#include <random>RTSPServer::RTSPServer(int port) : m_serverSocket(-1), m_running(false), m_nextSessionId(1000) {// 創建套接字m_serverSocket = socket(AF_INET, SOCK_STREAM, 0);if (m_serverSocket == -1) {std::cerr << "Failed to create socket" << std::endl;return;}// 設置套接字選項int opt = 1;if (setsockopt(m_serverSocket, SOL_SOCKET, SO_REUSEADDR, &opt, sizeof(opt)) == -1) {std::cerr << "Failed to set socket options" << std::endl;close(m_serverSocket);m_serverSocket = -1;return;}// 綁定套接字sockaddr_in serverAddr;memset(&serverAddr, 0, sizeof(serverAddr));serverAddr.sin_family = AF_INET;serverAddr.sin_addr.s_addr = INADDR_ANY;serverAddr.sin_port = htons(port);if (bind(m_serverSocket, (sockaddr*)&serverAddr, sizeof(serverAddr)) == -1) {std::cerr << "Failed to bind socket" << std::endl;close(m_serverSocket);m_serverSocket = -1;return;}// 監聽連接if (listen(m_serverSocket, 5) == -1) {std::cerr << "Failed to listen on socket" << std::endl;close(m_serverSocket);m_serverSocket = -1;return;}
}RTSPServer::~RTSPServer() {stop();if (m_serverSocket != -1) {close(m_serverSocket);}
}bool RTSPServer::start() {if (m_running) return true;if (m_serverSocket == -1) return false;m_running = true;m_serverThread = std::thread([this]() {while (m_running) {sockaddr_in clientAddr;socklen_t clientAddrLen = sizeof(clientAddr);int clientSocket = accept(m_serverSocket, (sockaddr*)&clientAddr, &clientAddrLen);if (clientSocket == -1) {if (m_running) {std::cerr << "Failed to accept client connection" << std::endl;}continue;}// 處理客戶端連接std::thread([this, clientSocket]() {handleClient(clientSocket);close(clientSocket);}).detach();}});return true;
}void RTSPServer::stop() {if (!m_running) return;m_running = false;if (m_serverSocket != -1) {// 關閉服務器套接字以中斷accept調用close(m_serverSocket);m_serverSocket = -1;}if (m_serverThread.joinable()) {m_serverThread.join();}// 清理所有會話std::lock_guard<std::mutex> lock(m_sessionsMutex);for (auto& session : m_sessions) {session.second->teardown();}m_sessions.clear();
}void RTSPServer::addMediaSource(const std::string& streamName, std::shared_ptr<MediaSource> source) {std::lock_guard<std::mutex> lock(m_sessionsMutex);m_mediaSources[streamName] = source;
}void RTSPServer::removeMediaSource(const std::string& streamName) {std::lock_guard<std::mutex> lock(m_sessionsMutex);m_mediaSources.erase(streamName);
}void RTSPServer::handleClient(int clientSocket) {char buffer[4096];memset(buffer, 0, sizeof(buffer));// 讀取RTSP請求ssize_t bytesRead = recv(clientSocket, buffer, sizeof(buffer) - 1, 0);if (bytesRead <= 0) {return;}std::string request(buffer, bytesRead);handleRTSPRequest(clientSocket, request);
}void RTSPServer::handleRTSPRequest(int clientSocket, const std::string& request) {std::istringstream iss(request);std::string method, url, version;iss >> method >> url >> version;std::stringstream response;if (method == "OPTIONS") {response << "RTSP/1.0 200 OK\r\n";response << "CSeq: " << 1 << "\r\n";response << "Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE\r\n";response << "\r\n";} else if (method == "DESCRIBE") {// 解析URL獲取流名稱size_t pos = url.find_last_of('/');std::string streamName = (pos != std::string::npos) ? url.substr(pos + 1) : url;std::lock_guard<std::mutex> lock(m_sessionsMutex);auto it = m_mediaSources.find(streamName);if (it == m_mediaSources.end()) {response << "RTSP/1.0 404 Not Found\r\n";response << "CSeq: " << 1 << "\r\n";response << "\r\n";} else {std::shared_ptr<MediaSource> mediaSource = it->second;response << "RTSP/1.0 200 OK\r\n";response << "CSeq: " << 1 << "\r\n";response << "Content-Type: application/sdp\r\n";response << "Content-Length: ";std::string sdp = "v=0\r\n";sdp += "o=- " + std::to_string(time(nullptr)) + " 1 IN IP4 127.0.0.1\r\n";sdp += "s=Media Session\r\n";sdp += "i=" + streamName + "\r\n";sdp += "c=IN IP4 0.0.0.0\r\n";sdp += "t=0 0\r\n";sdp += "m=video 0 RTP/AVP " + std::to_string(96) + "\r\n";sdp += "a=rtpmap:" + std::to_string(96) + " " + mediaSource->getMimeType() + "/" + std::to_string(mediaSource->getClockRate()) + "\r\n";sdp += "a=control:trackID=0\r\n";sdp += mediaSource->getAttribute() + "\r\n";response << sdp.length() << "\r\n\r\n";response << sdp;}}else if (method == "SETUP") {// 解析URL獲取流名稱size_t pos = url.find_last_of('/');std::string streamName = (pos != std::string::npos) ? url.substr(pos + 1) : url;// 解析客戶端端口pos = request.find("client_port=");int clientRtpPort = 0, clientRtcpPort = 0;if (pos != std::string::npos) {pos += 12;size_t endPos = request.find('-', pos);if (endPos != std::string::npos) {clientRtpPort = std::stoi(request.substr(pos, endPos - pos));pos = endPos + 1;endPos = request.find('\r', pos);if (endPos != std::string::npos) {clientRtcpPort = std::stoi(request.substr(pos, endPos - pos));}}}std::lock_guard<std::mutex> lock(m_sessionsMutex);auto it = m_mediaSources.find(streamName);if (it == m_mediaSources.end()) {response << "RTSP/1.0 404 Not Found\r\n";response << "CSeq: " << 1 << "\r\n";response << "\r\n";} else {// 創建會話std::shared_ptr<RTSPSession> session = createSession(streamName);session->setup(clientRtpPort, clientRtcpPort);response << "RTSP/1.0 200 OK\r\n";response << "CSeq: " << 1 << "\r\n";response << "Session: " << session->getSessionId() << ";timeout=60\r\n";response << "Transport: RTP/AVP/UDP;unicast;client_port=" << clientRtpPort << "-" << clientRtcpPort << ";server_port=" << (clientRtpPort + 1) << "-" << (clientRtcpPort + 1) << "\r\n";response << "\r\n";}}else if (method == "PLAY") {// 解析會話IDsize_t pos = request.find("Session: ");int sessionId = 0;if (pos != std::string::npos) {pos += 9;size_t endPos = request.find('\r', pos);if (endPos != std::string::npos) {sessionId = std::stoi(request.substr(pos, endPos - pos));}}std::lock_guard<std::mutex> lock(m_sessionsMutex);auto it = m_sessions.find(std::to_string(sessionId));if (it == m_sessions.end()) {response << "RTSP/1.0 454 Session Not Found\r\n";response << "CSeq: " << 1 << "\r\n";response << "\r\n";} else {it->second->play();response << "RTSP/1.0 200 OK\r\n";response << "CSeq: " << 1 << "\r\n";response << "Session: " << sessionId << "\r\n";response << "Range: npt=0.000-\r\n";response << "\r\n";}}else if (method == "PAUSE") {// 解析會話IDsize_t pos = request.find("Session: ");int sessionId = 0;if (pos != std::string::npos) {pos += 9;size_t endPos = request.find('\r', pos);if (endPos != std::string::npos) {sessionId = std::stoi(request.substr(pos, endPos - pos));}}std::lock_guard<std::mutex> lock(m_sessionsMutex);auto it = m_sessions.find(std::to_string(sessionId));if (it == m_sessions.end()) {response << "RTSP/1.0 454 Session Not Found\r\n";response << "CSeq: " << 1 << "\r\n";response << "\r\n";} else {it->second->pause();response << "RTSP/1.0 200 OK\r\n";response << "CSeq: " << 1 << "\r\n";response << "Session: " << sessionId << "\r\n";response << "\r\n";}}else if (method == "TEARDOWN") {// 解析會話IDsize_t pos = request.find("Session: ");int sessionId = 0;if (pos != std::string::npos) {pos += 9;size_t endPos = request.find('\r', pos);if (endPos != std::string::npos) {sessionId = std::stoi(request.substr(pos, endPos - pos));}}std::lock_guard<std::mutex> lock(m_sessionsMutex);auto it = m_sessions.find(std::to_string(sessionId));if (it == m_sessions.end()) {response << "RTSP/1.0 454 Session Not Found\r\n";response << "CSeq: " << 1 << "\r\n";response << "\r\n";} else {it->second->teardown();m_sessions.erase(std::to_string(sessionId));response << "RTSP/1.0 200 OK\r\n";response << "CSeq: " << 1 << "\r\n";response << "Session: " << sessionId << "\r\n";response << "\r\n";}}else {response << "RTSP/1.0 501 Not Implemented\r\n";response << "CSeq: " << 1 << "\r\n";response << "\r\n";}// 發送響應std::string responseStr = response.str();send(clientSocket, responseStr.c_str(), responseStr.length(), 0);
}std::shared_ptr<RTSPSession> RTSPServer::createSession(const std::string& streamName) {auto it = m_mediaSources.find(streamName);if (it == m_mediaSources.end()) {return nullptr;}int sessionId = m_nextSessionId++;std::shared_ptr<RTSPSession> session = std::make_shared<RTSPSession>(sessionId, it->second);m_sessions[std::to_string(sessionId)] = session;return session;
}RTSPSession::RTSPSession(int sessionId, std::shared_ptr<MediaSource> mediaSource): m_sessionId(sessionId), m_mediaSource(mediaSource), m_clientRtpPort(0), m_clientRtcpPort(0), m_rtpSocket(-1), m_playing(false), m_senderRunning(false) {// 創建RTP套接字m_rtpSocket = socket(AF_INET, SOCK_DGRAM, 0);if (m_rtpSocket == -1) {std::cerr << "Failed to create RTP socket" << std::endl;}
}RTSPSession::~RTSPSession() {teardown();if (m_rtpSocket != -1) {close(m_rtpSocket);}
}int RTSPSession::getSessionId() const {return m_sessionId;
}std::string RTSPSession::getSessionDescription() const {return m_mediaSource->getMediaDescription();
}void RTSPSession::setup(int clientRtpPort, int clientRtcpPort) {m_clientRtpPort = clientRtpPort;m_clientRtcpPort = clientRtcpPort;
}void RTSPSession::play() {if (m_playing) return;m_playing = true;m_senderRunning = true;m_senderThread = std::thread([this]() {sendRTPData();});
}void RTSPSession::pause() {m_playing = false;if (m_senderThread.joinable()) {m_senderThread.join();}
}void RTSPSession::teardown() {m_playing = false;m_senderRunning = false;if (m_senderThread.joinable()) {m_senderThread.join();}
}void RTSPSession::sendRTPData() {if (m_clientRtpPort == 0 || m_rtpSocket == -1) return;// 設置目標地址sockaddr_in destAddr;memset(&destAddr, 0, sizeof(destAddr));destAddr.sin_family = AF_INET;destAddr.sin_addr.s_addr = inet_addr("127.0.0.1");destAddr.sin_port = htons(m_clientRtpPort);// RTP包計數器和時間戳uint16_t sequenceNumber = 0;uint32_t timestamp = 0;uint32_t ssrc = rand();// 幀率控制uint32_t clockRate = m_mediaSource->getClockRate();uint32_t frameDuration = clockRate / 25; // 25fpswhile (m_senderRunning) {if (!m_playing) {std::this_thread::sleep_for(std::chrono::milliseconds(10));continue;}// 獲取下一幀數據std::vector<uint8_t> frame;bool marker = false;m_mediaSource->getNextFrame(frame, timestamp, marker);if (frame.empty()) {std::this_thread::sleep_for(std::chrono::milliseconds(40));continue;}// 構建RTP頭部const int RTP_HEADER_SIZE = 12;uint8_t rtpHeader[RTP_HEADER_SIZE];// 版本(2)、填充(0)、擴展(0)、CSRC計數(0)rtpHeader[0] = (2 << 6) | (0 << 5) | (0 << 4) | 0;// 標記位和負載類型(96 for H.264)rtpHeader[1] = (marker ? 0x80 : 0) | 96;// 序列號rtpHeader[2] = (sequenceNumber >> 8) & 0xFF;rtpHeader[3] = sequenceNumber & 0xFF;// 時間戳rtpHeader[4] = (timestamp >> 24) & 0xFF;rtpHeader[5] = (timestamp >> 16) & 0xFF;rtpHeader[6] = (timestamp >> 8) & 0xFF;rtpHeader[7] = timestamp & 0xFF;// SSRCrtpHeader[8] = (ssrc >> 24) & 0xFF;rtpHeader[9] = (ssrc >> 16) & 0xFF;rtpHeader[10] = (ssrc >> 8) & 0xFF;rtpHeader[11] = ssrc & 0xFF;// 發送RTP包std::vector<uint8_t> rtpPacket;rtpPacket.insert(rtpPacket.end(), rtpHeader, rtpHeader + RTP_HEADER_SIZE);rtpPacket.insert(rtpPacket.end(), frame.begin(), frame.end());sendto(m_rtpSocket, rtpPacket.data(), rtpPacket.size(), 0, (sockaddr*)&destAddr, sizeof(destAddr));// 更新序列號和時間戳sequenceNumber++;timestamp += frameDuration;// 控制幀率std::this_thread::sleep_for(std::chrono::milliseconds(40)); // 25fps}
}

模擬生產frame視頻幀
media_sources.h

#ifndef MEDIA_SOURCES_H
#define MEDIA_SOURCES_H#include "rtsp_server.h"
#include <random>class H264MediaSource : public MediaSource {
public:H264MediaSource(int width = 640, int height = 480);std::string getMimeType() const override;std::string getMediaDescription() const override;std::string getAttribute() const override;void getNextFrame(std::vector<uint8_t>& frame, uint32_t& timestamp, bool& marker) override;uint32_t getClockRate() const override;private:int m_width;int m_height;uint32_t m_currentTimestamp;std::mt19937 m_rng;
};class H265MediaSource : public MediaSource {
public:H265MediaSource(int width = 640, int height = 480);std::string getMimeType() const override;std::string getMediaDescription() const override;std::string getAttribute() const override;void getNextFrame(std::vector<uint8_t>& frame, uint32_t& timestamp, bool& marker) override;uint32_t getClockRate() const override;private:int m_width;int m_height;uint32_t m_currentTimestamp;std::mt19937 m_rng;
};class AACMediaSource : public MediaSource {
public:AACMediaSource(int sampleRate = 44100, int channels = 2);std::string getMimeType() const override;std::string getMediaDescription() const override;std::string getAttribute() const override;void getNextFrame(std::vector<uint8_t>& frame, uint32_t& timestamp, bool& marker) override;uint32_t getClockRate() const override;private:int m_sampleRate;int m_channels;uint32_t m_currentTimestamp;std::mt19937 m_rng;
};#endif // MEDIA_SOURCES_H

media_sources.cpp

#include "media_sources.h"
#include <random>H264MediaSource::H264MediaSource(int width, int height): m_width(width), m_height(height), m_currentTimestamp(0) {std::random_device rd;m_rng = std::mt19937(rd());
}std::string H264MediaSource::getMimeType() const {return "H264";
}std::string H264MediaSource::getMediaDescription() const {return "H.264 Video Stream";
}std::string H264MediaSource::getAttribute() const {return "a=fmtp:96 packetization-mode=1;profile-level-id=42E01F;sprop-parameter-sets=Z0LAH9kA8AoAAAMAAgAAAwDAAAAwDxYuS,aM48gA==";
}void H264MediaSource::getNextFrame(std::vector<uint8_t>& frame, uint32_t& timestamp, bool& marker) {static bool isKeyFrame = true;static int frameCount = 0;// 每30幀生成一個關鍵幀if (frameCount++ % 30 == 0) {isKeyFrame = true;} else {isKeyFrame = false;}// 模擬H.264幀size_t frameSize = isKeyFrame ? 10000 + (m_rng() % 5000) : 2000 + (m_rng() % 3000);frame.resize(frameSize);// 模擬H.264 NAL單元頭if (isKeyFrame) {// IDR幀frame[0] = 0x00;frame[1] = 0x00;frame[2] = 0x00;frame[3] = 0x01;frame[4] = 0x65; // IDR NAL單元類型} else {// P幀frame[0] = 0x00;frame[1] = 0x00;frame[2] = 0x00;frame[3] = 0x01;frame[4] = 0x41; // P NAL單元類型}// 填充隨機數據for (size_t i = 5; i < frameSize; ++i) {frame[i] = m_rng() % 256;}timestamp = m_currentTimestamp;m_currentTimestamp += 90000 / 25; // 25fps，90kHz時鐘marker = true; // 每幀都設置標記位
}uint32_t H264MediaSource::getClockRate() const {return 90000; // 90kHz
}H265MediaSource::H265MediaSource(int width, int height): m_width(width), m_height(height), m_currentTimestamp(0) {std::random_device rd;m_rng = std::mt19937(rd());
}std::string H265MediaSource::getMimeType() const {return "H265";
}std::string H265MediaSource::getMediaDescription() const {return "H.265 Video Stream";
}std::string H265MediaSource::getAttribute() const {return "a=fmtp:96 profile-id=1;level-id=110;packetization-mode=1";
}void H265MediaSource::getNextFrame(std::vector<uint8_t>& frame, uint32_t& timestamp, bool& marker) {static bool isKeyFrame = true;static int frameCount = 0;// 每30幀生成一個關鍵幀if (frameCount++ % 30 == 0) {isKeyFrame = true;} else {isKeyFrame = false;}// 模擬H.265幀size_t frameSize = isKeyFrame ? 12000 + (m_rng() % 6000) : 2500 + (m_rng() % 3500);frame.resize(frameSize);// 模擬H.265 NAL單元頭if (isKeyFrame) {// IDR幀frame[0] = 0x00;frame[1] = 0x00;frame[2] = 0x00;frame[3] = 0x01;frame[4] = 0x40; // IDR_W_RADL NAL單元類型frame[5] = 0x01;} else {// P幀frame[0] = 0x00;frame[1] = 0x00;frame[2] = 0x00;frame[3] = 0x01;frame[4] = 0x02; // TRAIL_R NAL單元類型frame[5] = 0x01;}// 填充隨機數據for (size_t i = 6; i < frameSize; ++i) {frame[i] = m_rng() % 256;}timestamp = m_currentTimestamp;m_currentTimestamp += 90000 / 25; // 25fps，90kHz時鐘marker = true; // 每幀都設置標記位
}uint32_t H265MediaSource::getClockRate() const {return 90000; // 90kHz
}AACMediaSource::AACMediaSource(int sampleRate, int channels): m_sampleRate(sampleRate), m_channels(channels), m_currentTimestamp(0) {std::random_device rd;m_rng = std::mt19937(rd());
}std::string AACMediaSource::getMimeType() const {return "MPEG4-GENERIC";
}std::string AACMediaSource::getMediaDescription() const {return "AAC Audio Stream";
}std::string AACMediaSource::getAttribute() const {return "a=fmtp:96 streamtype=5;profile-level-id=15;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=1408";
}void AACMediaSource::getNextFrame(std::vector<uint8_t>& frame, uint32_t& timestamp, bool& marker) {// 模擬AAC幀size_t frameSize = 1024 + (m_rng() % 512); // 1024-1536字節frame.resize(frameSize);// 填充隨機數據for (size_t i = 0; i < frameSize; ++i) {frame[i] = m_rng() % 256;}timestamp = m_currentTimestamp;m_currentTimestamp += 1024; // 每幀1024個樣本marker = true;
}uint32_t AACMediaSource::getClockRate() const {return m_sampleRate;
}

main.cpp

#include "rtsp_server.h"
#include "media_sources.h"
#include <iostream>
#include <memory>int main() {// 創建RTSP服務器，監聽8554端口RTSPServer server(8554);// 創建并添加多路媒體源auto camera1_h264 = std::make_shared<H264MediaSource>(1280, 720);auto camera2_h264 = std::make_shared<H264MediaSource>(800, 600);auto camera3_h265 = std::make_shared<H265MediaSource>(1920, 1080);auto audio_aac = std::make_shared<AACMediaSource>(44100, 2);server.addMediaSource("camera1", camera1_h264);server.addMediaSource("camera2", camera2_h264);server.addMediaSource("camera3", camera3_h265);server.addMediaSource("audio", audio_aac);// 啟動服務器if (server.start()) {std::cout << "RTSP Server started on port 8554" << std::endl;std::cout << "Available streams:" << std::endl;std::cout << "rtsp://localhost:8554/camera1 (H.264 1280x720)" << std::endl;std::cout << "rtsp://localhost:8554/camera2 (H.264 800x600)" << std::endl;std::cout << "rtsp://localhost:8554/camera3 (H.265 1920x1080)" << std::endl;std::cout << "rtsp://localhost:8554/audio (AAC 44.1kHz stereo)" << std::endl;// 保持服務器運行std::cout << "Press Enter to stop the server..." << std::endl;std::cin.get();} else {std::cerr << "Failed to start RTSP server" << std::endl;}// 停止服務器server.stop();return 0;
}

rtsp_client.h

#ifndef RTSP_CLIENT_H
#define RTSP_CLIENT_H#include <string>
#include <memory>
#include <thread>
#include <mutex>
#include <vector>
#include <functional>
#include <cstdint>// RTP包結構
struct RTPPacket {uint8_t version;      // 版本號uint8_t padding;      // 填充標志uint8_t extension;    // 擴展標志uint8_t cc;           // CSRC計數器uint8_t marker;       // 標記位uint8_t payloadType;  // 負載類型uint16_t sequence;    // 序列號uint32_t timestamp;   // 時間戳uint32_t ssrc;        // 同步源標識符std::vector<uint8_t> payload; // 負載數據
};// RTSP客戶端類
class RTSPClient {
public:// 幀數據回調函數類型using FrameCallback = std::function<void(const uint8_t*, size_t, uint32_t, bool)>;RTSPClient();~RTSPClient();// 連接到RTSP服務器bool connect(const std::string& rtspUrl);// 斷開連接void disconnect();// 設置幀數據回調void setFrameCallback(const FrameCallback& callback);// 開始播放bool play();// 暫停播放bool pause();// 獲取媒體信息std::string getMediaInfo() const;private:// 狀態枚舉enum class State {INIT,READY,PLAYING,PAUSED,TEARDOWN};// 解析RTSP URLbool parseUrl(const std::string& rtspUrl);// 發送RTSP請求std::string sendRequest(const std::string& request);// 接收RTP/RTCP數據void receiveRTPData();// 解析SDP信息bool parseSDP(const std::string& sdp);// 解析RTP包bool parseRTPPacket(const std::vector<uint8_t>& data, RTPPacket& packet);// 處理視頻幀void processVideoFrame(const RTPPacket& packet);// 處理音頻幀void processAudioFrame(const RTPPacket& packet);// 成員變量std::string m_serverAddress;int m_serverPort;std::string m_streamPath;int m_rtspSocket;std::string m_sessionId;State m_state;int m_cseq;int m_rtpPort;int m_rtcpPort;std::thread m_receiveThread;bool m_running;mutable std::mutex m_mutex;FrameCallback m_frameCallback;std::string m_mediaInfo;std::string m_codec;int m_clockRate;
};#endif // RTSP_CLIENT_H

rtsp_client.cpp

#include "rtsp_client.h"
#include <iostream>
#include <sstream>
#include <cstring>
#include <sys/socket.h>
#include <arpa/inet.h>
#include <netinet/in.h>
#include <unistd.h>
#include <regex>RTSPClient::RTSPClient(): m_serverPort(554), m_rtspSocket(-1), m_state(State::INIT), m_cseq(0), m_rtpPort(0), m_rtcpPort(0), m_running(false), m_clockRate(90000) {
}RTSPClient::~RTSPClient() {disconnect();
}bool RTSPClient::connect(const std::string& rtspUrl) {std::lock_guard<std::mutex> lock(m_mutex);if (m_state != State::INIT) {std::cerr << "RTSPClient: Already connected or in progress" << std::endl;return false;}if (!parseUrl(rtspUrl)) {std::cerr << "RTSPClient: Failed to parse URL: " << rtspUrl << std::endl;return false;}// 創建RTSP套接字m_rtspSocket = socket(AF_INET, SOCK_STREAM, 0);if (m_rtspSocket < 0) {std::cerr << "RTSPClient: Failed to create socket" << std::endl;return false;}// 設置服務器地址sockaddr_in serverAddr;memset(&serverAddr, 0, sizeof(serverAddr));serverAddr.sin_family = AF_INET;serverAddr.sin_port = htons(m_serverPort);// 解析服務器地址if (inet_pton(AF_INET, m_serverAddress.c_str(), &serverAddr.sin_addr) <= 0) {std::cerr << "RTSPClient: Invalid address/ Address not supported" << std::endl;close(m_rtspSocket);m_rtspSocket = -1;return false;}// 連接服務器if (connect(m_rtspSocket, (struct sockaddr*)&serverAddr, sizeof(serverAddr)) < 0) {std::cerr << "RTSPClient: Connection failed" << std::endl;close(m_rtspSocket);m_rtspSocket = -1;return false;}// 發送OPTIONS請求std::string request = "OPTIONS " + rtspUrl + " RTSP/1.0\r\n";request += "CSeq: " + std::to_string(++m_cseq) + "\r\n";request += "User-Agent: RTSPClient\r\n\r\n";std::string response = sendRequest(request);if (response.empty()) {std::cerr << "RTSPClient: No response to OPTIONS request" << std::endl;disconnect();return false;}// 發送DESCRIBE請求request = "DESCRIBE " + rtspUrl + " RTSP/1.0\r\n";request += "CSeq: " + std::to_string(++m_cseq) + "\r\n";request += "Accept: application/sdp\r\n";request += "User-Agent: RTSPClient\r\n\r\n";response = sendRequest(request);if (response.empty()) {std::cerr << "RTSPClient: No response to DESCRIBE request" << std::endl;disconnect();return false;}// 解析SDP信息if (!parseSDP(response)) {std::cerr << "RTSPClient: Failed to parse SDP" << std::endl;disconnect();return false;}m_state = State::READY;return true;
}void RTSPClient::disconnect() {std::lock_guard<std::mutex> lock(m_mutex);if (m_state == State::TEARDOWN) {return;}// 發送TEARDOWN請求if (m_state != State::INIT && m_rtspSocket != -1) {std::string request = "TEARDOWN " + m_streamPath + " RTSP/1.0\r\n";request += "CSeq: " + std::to_string(++m_cseq) + "\r\n";if (!m_sessionId.empty()) {request += "Session: " + m_sessionId + "\r\n";}request += "User-Agent: RTSPClient\r\n\r\n";sendRequest(request);}m_state = State::TEARDOWN;// 停止接收線程m_running = false;if (m_receiveThread.joinable()) {m_receiveThread.join();}// 關閉套接字if (m_rtspSocket != -1) {close(m_rtspSocket);m_rtspSocket = -1;}
}bool RTSPClient::play() {std::lock_guard<std::mutex> lock(m_mutex);if (m_state != State::READY && m_state != State::PAUSED) {std::cerr << "RTSPClient: Cannot play in current state" << std::endl;return false;}// 發送SETUP請求std::string request = "SETUP " + m_streamPath + "/track1 RTSP/1.0\r\n";request += "CSeq: " + std::to_string(++m_cseq) + "\r\n";if (!m_sessionId.empty()) {request += "Session: " + m_sessionId + "\r\n";}request += "Transport: RTP/AVP;unicast;client_port=" + std::to_string(m_rtpPort) + "-" + std::to_string(m_rtcpPort) + "\r\n";request += "User-Agent: RTSPClient\r\n\r\n";std::string response = sendRequest(request);if (response.empty()) {std::cerr << "RTSPClient: No response to SETUP request" << std::endl;return false;}// 提取會話IDstd::regex sessionRegex("Session: (.*?)\r\n");std::smatch sessionMatch;if (std::regex_search(response, sessionMatch, sessionRegex) && sessionMatch.size() > 1) {m_sessionId = sessionMatch[1].str();// 去除可能的超時參數size_t semicolonPos = m_sessionId.find(';');if (semicolonPos != std::string::npos) {m_sessionId = m_sessionId.substr(0, semicolonPos);}} else {std::cerr << "RTSPClient: Session ID not found in SETUP response" << std::endl;return false;}// 提取服務器端口std::regex serverPortRegex("server_port=(\\d+)-(\\d+)");std::smatch portMatch;if (std::regex_search(response, portMatch, serverPortRegex) && portMatch.size() > 2) {// 這里可以獲取服務器的RTP和RTCP端口，但實際使用中我們通常只需要客戶端端口}// 發送PLAY請求request = "PLAY " + m_streamPath + " RTSP/1.0\r\n";request += "CSeq: " + std::to_string(++m_cseq) + "\r\n";request += "Session: " + m_sessionId + "\r\n";request += "User-Agent: RTSPClient\r\n\r\n";response = sendRequest(request);if (response.empty()) {std::cerr << "RTSPClient: No response to PLAY request" << std::endl;return false;}// 啟動接收線程m_running = true;m_receiveThread = std::thread(&RTSPClient::receiveRTPData, this);m_state = State::PLAYING;return true;
}bool RTSPClient::pause() {std::lock_guard<std::mutex> lock(m_mutex);if (m_state != State::PLAYING) {std::cerr << "RTSPClient: Cannot pause in current state" << std::endl;return false;}// 發送PAUSE請求std::string request = "PAUSE " + m_streamPath + " RTSP/1.0\r\n";request += "CSeq: " + std::to_string(++m_cseq) + "\r\n";request += "Session: " + m_sessionId + "\r\n";request += "User-Agent: RTSPClient\r\n\r\n";std::string response = sendRequest(request);if (response.empty()) {std::cerr << "RTSPClient: No response to PAUSE request" << std::endl;return false;}m_state = State::PAUSED;return true;
}std::string RTSPClient::getMediaInfo() const {std::lock_guard<std::mutex> lock(m_mutex);return m_mediaInfo;
}void RTSPClient::setFrameCallback(const FrameCallback& callback) {std::lock_guard<std::mutex> lock(m_mutex);m_frameCallback = callback;
}bool RTSPClient::parseUrl(const std::string& rtspUrl) {// 解析RTSP URL: rtsp://server:port/pathstd::regex urlRegex("rtsp://([^:/]+)(?::(\\d+))?(/.*)");std::smatch match;if (std::regex_match(rtspUrl, match, urlRegex) && match.size() >= 3) {m_serverAddress = match[1].str();if (match.length(2) > 0) {m_serverPort = std::stoi(match[2].str());}m_streamPath = "rtsp://" + m_serverAddress + (match.length(2) > 0 ? (":" + match[2].str()) : "") + match[3].str();return true;}return false;
}std::string RTSPClient::sendRequest(const std::string& request) {if (m_rtspSocket == -1) {return "";}// 發送請求ssize_t sent = send(m_rtspSocket, request.c_str(), request.length(), 0);if (sent < 0) {std::cerr << "RTSPClient: Error sending request" << std::endl;return "";}// 接收響應char buffer[4096];std::string response;while (true) {memset(buffer, 0, sizeof(buffer));ssize_t recvSize = recv(m_rtspSocket, buffer, sizeof(buffer) - 1, 0);if (recvSize <= 0) {break;}response.append(buffer, recvSize);// 檢查是否接收到完整的響應if (response.find("\r\n\r\n") != std::string::npos) {break;}}return response;
}bool RTSPClient::parseSDP(const std::string& sdp) {// 查找SDP部分size_t sdpStart = sdp.find("\r\n\r\n");if (sdpStart == std::string::npos) {return false;}std::string sdpContent = sdp.substr(sdpStart + 4);// 解析媒體信息std::regex mediaRegex("m=([^ ]+) ([^ ]+) ([^ ]+) ([^\r\n]+)");std::sregex_iterator currentMatch(sdpContent.begin(), sdpContent.end(), mediaRegex);std::sregex_iterator end;bool foundVideo = false;bool foundAudio = false;while (currentMatch != end) {std::smatch match = *currentMatch;std::string mediaType = match[1].str();std::string port = match[2].str();std::string proto = match[3].str();std::string fmt = match[4].str();if (mediaType == "video" && !foundVideo) {foundVideo = true;m_mediaInfo += "Video: " + fmt + "\n";// 查找編碼信息std::regex codecRegex("a=rtpmap:(\\d+) (H264|H265|MP4V-ES)(?:/([^/\\r\\n]+))?");std::smatch codecMatch;if (std::regex_search(sdpContent, codecMatch, codecRegex) && codecMatch.size() >= 3) {m_codec = codecMatch[2].str();if (codecMatch.size() > 3 && codecMatch.length(3) > 0) {m_clockRate = std::stoi(codecMatch[3].str());}}} else if (mediaType == "audio" && !foundAudio) {foundAudio = true;m_mediaInfo += "Audio: " + fmt + "\n";// 查找編碼信息std::regex codecRegex("a=rtpmap:(\\d+) (MPEG4-GENERIC|PCMA|PCMU)(?:/([^/\\r\\n]+))?");std::smatch codecMatch;if (std::regex_search(sdpContent, codecMatch, codecRegex) && codecMatch.size() >= 3) {if (codecMatch[2].str() == "MPEG4-GENERIC") {m_codec = "AAC";} else {m_codec = codecMatch[2].str();}if (codecMatch.size() > 3 && codecMatch.length(3) > 0) {m_clockRate = std::stoi(codecMatch[3].str());}}}++currentMatch;}// 為RTP和RTCP選擇隨機端口m_rtpPort = 10000 + (rand() % 5000) * 2;m_rtcpPort = m_rtpPort + 1;return true;
}void RTSPClient::receiveRTPData() {// 創建RTP套接字int rtpSocket = socket(AF_INET, SOCK_DGRAM, 0);if (rtpSocket < 0) {std::cerr << "RTSPClient: Failed to create RTP socket" << std::endl;return;}// 綁定RTP端口sockaddr_in rtpAddr;memset(&rtpAddr, 0, sizeof(rtpAddr));rtpAddr.sin_family = AF_INET;rtpAddr.sin_addr.s_addr = INADDR_ANY;rtpAddr.sin_port = htons(m_rtpPort);if (bind(rtpSocket, (struct sockaddr*)&rtpAddr, sizeof(rtpAddr)) < 0) {std::cerr << "RTSPClient: Failed to bind RTP socket" << std::endl;close(rtpSocket);return;}// 設置非阻塞模式// int flags = fcntl(rtpSocket, F_GETFL, 0);// fcntl(rtpSocket, F_SETFL, flags | O_NONBLOCK);char buffer[16000];sockaddr_in senderAddr;socklen_t senderAddrLen = sizeof(senderAddr);std::vector<uint8_t> frameData;uint32_t lastTimestamp = 0;bool isKeyFrame = false;while (m_running) {memset(buffer, 0, sizeof(buffer));ssize_t recvSize = recvfrom(rtpSocket, buffer, sizeof(buffer), 0, (struct sockaddr*)&senderAddr, &senderAddrLen);if (recvSize > 0) {std::vector<uint8_t> packetData(buffer, buffer + recvSize);RTPPacket packet;if (parseRTPPacket(packetData, packet)) {// 處理視頻幀if (m_codec == "H264" || m_codec == "H265") {processVideoFrame(packet);} // 處理音頻幀else if (m_codec == "AAC") {processAudioFrame(packet);}}} else {// 超時或錯誤，短暫休眠std::this_thread::sleep_for(std::chrono::milliseconds(10));}}close(rtpSocket);
}bool RTSPClient::parseRTPPacket(const std::vector<uint8_t>& data, RTPPacket& packet) {if (data.size() < 12) {return false;}// 解析RTP頭部packet.version = (data[0] >> 6) & 0x03;packet.padding = (data[0] >> 5) & 0x01;packet.extension = (data[0] >> 4) & 0x01;packet.cc = data[0] & 0x0F;packet.marker = (data[1] >> 7) & 0x01;packet.payloadType = data[1] & 0x7F;packet.sequence = (data[2] << 8) | data[3];packet.timestamp = (data[4] << 24) | (data[5] << 16) | (data[6] << 8) | data[7];packet.ssrc = (data[8] << 24) | (data[9] << 16) | (data[10] << 8) | data[11];// 提取負載數據size_t headerSize = 12 + (packet.cc * 4);if (packet.extension && (data.size() > headerSize + 4)) {uint16_t extLength = (data[headerSize + 2] << 8) | data[headerSize + 3];headerSize += 4 + (extLength * 4);}if (data.size() > headerSize) {packet.payload.assign(data.begin() + headerSize, data.end());return true;}return false;
}void RTSPClient::processVideoFrame(const RTPPacket& packet) {std::lock_guard<std::mutex> lock(m_mutex);if (!m_frameCallback) {return;}if (m_codec == "H264" && packet.payload.size() > 0) {// H.264 NAL單元類型uint8_t nalType = packet.payload[0] & 0x1F;// 判斷是否為關鍵幀bool isKeyFrame = (nalType == 7) || (nalType == 5);// 調用回調函數m_frameCallback(packet.payload.data(), packet.payload.size(), packet.timestamp, isKeyFrame);} else if (m_codec == "H265" && packet.payload.size() > 0) {// H.265 NAL單元類型uint8_t nalType = (packet.payload[0] >> 1) & 0x3F;// 判斷是否為關鍵幀bool isKeyFrame = (nalType >= 16 && nalType <= 23);// 調用回調函數m_frameCallback(packet.payload.data(), packet.payload.size(), packet.timestamp, isKeyFrame);}
}void RTSPClient::processAudioFrame(const RTPPacket& packet) {std::lock_guard<std::mutex> lock(m_mutex);if (!m_frameCallback) {return;}// 對于AAC，每個RTP包通常包含一個完整的ADTS幀m_frameCallback(packet.payload.data(), packet.payload.size(), packet.timestamp, false);
}

client_example.cpp

#include "rtsp_client.h"
#include <iostream>
#include <fstream>
#include <atomic>
#include <thread>// 簡單的幀處理器
class FrameProcessor {
public:FrameProcessor(const std::string& outputPath) : m_outputPath(outputPath), m_frameCount(0) {}void handleFrame(const uint8_t* data, size_t size, uint32_t timestamp, bool isKeyFrame) {std::cout << "Frame received: size=" << size << ", timestamp=" << timestamp << ", keyframe=" << (isKeyFrame ? "YES" : "NO") << std::endl;// 保存幀數據到文件if (m_outputPath.empty()) {return;}std::stringstream ss;ss << m_outputPath << "/frame_" << m_frameCount++ << ".h264";std::ofstream frameFile(ss.str(), std::ios::binary);if (frameFile.is_open()) {frameFile.write(reinterpret_cast<const char*>(data), size);frameFile.close();}}private:std::string m_outputPath;std::atomic<int> m_frameCount;
};int main(int argc, char* argv[]) {if (argc < 2) {std::cout << "Usage: " << argv[0] << " <rtsp_url> [output_dir]" << std::endl;return 1;}std::string rtspUrl = argv[1];std::string outputPath = (argc > 2) ? argv[2] : "";// 創建幀處理器FrameProcessor processor(outputPath);// 創建RTSP客戶端RTSPClient client;// 設置幀回調client.setFrameCallback([&processor](const uint8_t* data, size_t size, uint32_t timestamp, bool isKeyFrame) {processor.handleFrame(data, size, timestamp, isKeyFrame);});// 連接到RTSP服務器if (!client.connect(rtspUrl)) {std::cerr << "Failed to connect to RTSP server" << std::endl;return 1;}// 打印媒體信息std::cout << "Media information:" << std::endl;std::cout << client.getMediaInfo() << std::endl;// 開始播放if (!client.play()) {std::cerr << "Failed to start playing" << std::endl;client.disconnect();return 1;}std::cout << "Playing stream from: " << rtspUrl << std::endl;std::cout << "Press Enter to stop..." << std::endl;// 等待用戶輸入std::cin.get();// 停止播放client.disconnect();std::cout << "RTSP client stopped" << std::endl;return 0;
}

2、完整一點的判斷幀類型，分包，解包

#include <iostream>
#include <string>
#include <vector>
#include <map>
#include <memory>
#include <cstring>
#include <thread>
#include <atomic>
#include <random>
#include <ctime>// RTSP協議常量
const int RTSP_PORT = 8554;
const int RTP_PORT_BASE = 5000;
const int RTCP_PORT_BASE = 5001;
const int MAX_PACKET_SIZE = 1400;  // MTU - IP頭 - UDP頭// NAL單元類型
enum NalUnitType {NAL_UNKNOWN = 0,NAL_SLICE = 1,NAL_IDR_SLICE = 5,NAL_SEI = 6,NAL_SPS = 7,NAL_PPS = 8,NAL_AUD = 9
};// 幀類型
enum FrameType {FRAME_I = 0,FRAME_P = 1,FRAME_B = 2
};// RTP包結構
struct RTPPacket {uint8_t version;      // 版本號(2 bits)uint8_t padding;      // 填充標志(1 bit)uint8_t extension;    // 擴展標志(1 bit)uint8_t cc;           // CSRC計數器(4 bits)uint8_t marker;       // 標記位(1 bit)uint8_t payloadType;  // 負載類型(7 bits)uint16_t sequence;    // 序列號uint32_t timestamp;   // 時間戳uint32_t ssrc;        // 同步源標識符std::vector<uint8_t> payload;  // 負載數據RTPPacket() : version(2), padding(0), extension(0), cc(0), marker(0), payloadType(96), sequence(0), timestamp(0), ssrc(0) {}
};// 會話信息
struct Session {std::string sessionId;std::string streamName;int clientRtpPort;int clientRtcpPort;int serverRtpPort;int serverRtcpPort;std::atomic<bool> isPlaying;std::thread streamThread;Session() : clientRtpPort(0), clientRtcpPort(0), serverRtpPort(0), serverRtcpPort(0), isPlaying(false) {}
};// 媒體源接口
class MediaSource {
public:virtual ~MediaSource() {}virtual std::string getMimeType() = 0;virtual std::string getSdpDescription() = 0;virtual bool getNextFrame(std::vector<uint8_t>& frameData, FrameType& frameType) = 0;
};// H.264媒體源實現
class H264MediaSource : public MediaSource {
private:int width;int height;int fps;uint32_t frameCount;std::mt19937 randomGenerator;std::uniform_int_distribution<int> frameTypeDist;public:H264MediaSource(int w, int h, int f = 30) : width(w), height(h), fps(f), frameCount(0) {randomGenerator.seed(std::time(nullptr));frameTypeDist = std::uniform_int_distribution<int>(0, 9);  // 0-9的隨機數}std::string getMimeType() override {return "video/H264";}std::string getSdpDescription() override {std::string sdp = "m=video 0 RTP/AVP 96\r\n""a=rtpmap:96 H264/90000\r\n""a=fmtp:96 packetization-mode=1;profile-level-id=42001F;sprop-parameter-sets=Z0IAKeKQDwBE/LwEBAAMAAgAAAwAymA0A,aM48gA==\r\n""a=control:streamid=0\r\n";return sdp;}bool getNextFrame(std::vector<uint8_t>& frameData, FrameType& frameType) override {// 模擬生成H.264幀frameCount++;// 每30幀生成一個I幀，其余為P幀(90%)或B幀(10%)if (frameCount % 30 == 0) {frameType = FRAME_I;generateIFrame(frameData);} else {int randVal = frameTypeDist(randomGenerator);if (randVal < 9) {frameType = FRAME_P;generatePFrame(frameData);} else {frameType = FRAME_B;generateBFrame(frameData);}}return true;}private:void generateIFrame(std::vector<uint8_t>& frameData) {// 模擬生成I幀 (SPS + PPS + IDR)// SPS (Sequence Parameter Set)std::vector<uint8_t> sps = {0x67, 0x42, 0x00, 0x1F, 0xE9, 0x01, 0x40, 0x16, 0xE3, 0x88, 0x80, 0x00, 0x00, 0x03, 0x00, 0x40, 0x00, 0x00, 0x0C, 0x8C, 0x3C, 0x60};// PPS (Picture Parameter Set)std::vector<uint8_t> pps = {0x68, 0xCE, 0x38, 0x80};// IDR Slice (隨機數據模擬)std::vector<uint8_t> idr(1024 * 4);  // 4KB的IDR圖像idr[0] = 0x65;  // NAL單元類型: IDR Slicefor (size_t i = 1; i < idr.size(); i++) {idr[i] = rand() % 256;}// 組合成完整的I幀frameData.insert(frameData.end(), sps.begin(), sps.end());frameData.insert(frameData.end(), pps.begin(), pps.end());frameData.insert(frameData.end(), idr.begin(), idr.end());}void generatePFrame(std::vector<uint8_t>& frameData) {// 模擬生成P幀frameData.resize(1024 * 2);  // 2KB的P幀frameData[0] = 0x41;  // NAL單元類型: 非IDR Slicefor (size_t i = 1; i < frameData.size(); i++) {frameData[i] = rand() % 256;}}void generateBFrame(std::vector<uint8_t>& frameData) {// 模擬生成B幀frameData.resize(1024);  // 1KB的B幀frameData[0] = 0x41;  // NAL單元類型: 非IDR Slice (與P幀相同)for (size_t i = 1; i < frameData.size(); i++) {frameData[i] = rand() % 256;}}
};// AAC媒體源實現
class AACMediaSource : public MediaSource {
private:int sampleRate;int channels;public:AACMediaSource(int sr = 44100, int ch = 2) : sampleRate(sr), channels(ch) {}std::string getMimeType() override {return "audio/AAC";}std::string getSdpDescription() override {std::string sdp = "m=audio 0 RTP/AVP 97\r\n""a=rtpmap:97 MPEG4-GENERIC/" + std::to_string(sampleRate) + "/" + std::to_string(channels) + "\r\n""a=fmtp:97 profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=1210\r\n""a=control:streamid=1\r\n";return sdp;}bool getNextFrame(std::vector<uint8_t>& frameData, FrameType& frameType) override {// 模擬生成AAC音頻幀frameData.resize(1024);  // 1KB的AAC幀for (size_t i = 0; i < frameData.size(); i++) {frameData[i] = rand() % 256;}return true;}
};// RTSP服務器
class RTSPServer {
private:int serverSocket;std::map<std::string, std::shared_ptr<MediaSource>> mediaSources;std::map<std::string, Session> sessions;std::atomic<bool> running;std::thread acceptThread;public:RTSPServer(int port = RTSP_PORT) : serverSocket(-1), running(false) {// 創建TCP套接字serverSocket = socket(AF_INET, SOCK_STREAM, 0);if (serverSocket < 0) {std::cerr << "Failed to create socket" << std::endl;return;}// 設置套接字選項int opt = 1;if (setsockopt(serverSocket, SOL_SOCKET, SO_REUSEADDR, &opt, sizeof(opt)) < 0) {std::cerr << "Failed to set socket options" << std::endl;close(serverSocket);serverSocket = -1;return;}// 綁定地址sockaddr_in serverAddr;memset(&serverAddr, 0, sizeof(serverAddr));serverAddr.sin_family = AF_INET;serverAddr.sin_addr.s_addr = INADDR_ANY;serverAddr.sin_port = htons(port);if (bind(serverSocket, (struct sockaddr*)&serverAddr, sizeof(serverAddr)) < 0) {std::cerr << "Failed to bind socket" << std::endl;close(serverSocket);serverSocket = -1;return;}// 監聽連接if (listen(serverSocket, 5) < 0) {std::cerr << "Failed to listen on socket" << std::endl;close(serverSocket);serverSocket = -1;return;}}~RTSPServer() {stop();if (serverSocket >= 0) {close(serverSocket);}}void addMediaSource(const std::string& name, std::shared_ptr<MediaSource> source) {mediaSources[name] = source;}bool start() {if (serverSocket < 0 || running) {return false;}running = true;acceptThread = std::thread(&RTSPServer::acceptConnections, this);return true;}void stop() {running = false;if (acceptThread.joinable()) {acceptThread.join();}// 停止所有會話for (auto& pair : sessions) {pair.second.isPlaying = false;if (pair.second.streamThread.joinable()) {pair.second.streamThread.join();}}sessions.clear();}private:void acceptConnections() {while (running) {sockaddr_in clientAddr;socklen_t clientAddrLen = sizeof(clientAddr);int clientSocket = accept(serverSocket, (struct sockaddr*)&clientAddr, &clientAddrLen);if (clientSocket < 0) {if (running) {std::cerr << "Failed to accept client connection" << std::endl;}continue;}// 為客戶端創建處理線程std::thread([this, clientSocket]() {handleClient(clientSocket);close(clientSocket);}).detach();}}void handleClient(int clientSocket) {char buffer[4096];std::string sessionId;while (running) {memset(buffer, 0, sizeof(buffer));int bytesRead = recv(clientSocket, buffer, sizeof(buffer) - 1, 0);if (bytesRead <= 0) {break;  // 客戶端斷開連接}std::string request(buffer, bytesRead);std::string response = processRequest(request, sessionId);send(clientSocket, response.c_str(), response.size(), 0);}}std::string processRequest(const std::string& request, std::string& sessionId) {// 解析RTSP請求方法size_t methodEnd = request.find(' ');if (methodEnd == std::string::npos) {return "RTSP/1.0 400 Bad Request\r\n\r\n";}std::string method = request.substr(0, methodEnd);size_t uriEnd = request.find(' ', methodEnd + 1);if (uriEnd == std::string::npos) {return "RTSP/1.0 400 Bad Request\r\n\r\n";}std::string uri = request.substr(methodEnd + 1, uriEnd - methodEnd - 1);size_t seqStart = request.find("CSeq: ") + 6;size_t seqEnd = request.find("\r\n", seqStart);std::string cseq = request.substr(seqStart, seqEnd - seqStart);// 處理不同的RTSP方法if (method == "OPTIONS") {return "RTSP/1.0 200 OK\r\n""CSeq: " + cseq + "\r\n""Public: OPTIONS, DESCRIBE, SETUP, PLAY, PAUSE, TEARDOWN\r\n\r\n";}else if (method == "DESCRIBE") {// 提取流名稱size_t streamStart = uri.find_last_of('/') + 1;std::string streamName = uri.substr(streamStart);auto it = mediaSources.find(streamName);if (it == mediaSources.end()) {return "RTSP/1.0 404 Not Found\r\n""CSeq: " + cseq + "\r\n\r\n";}std::string sdp = it->second->getSdpDescription();return "RTSP/1.0 200 OK\r\n""CSeq: " + cseq + "\r\n""Content-Base: " + uri + "/\r\n""Content-Type: application/sdp\r\n""Content-Length: " + std::to_string(sdp.size()) + "\r\n\r\n"+ sdp;}else if (method == "SETUP") {// 提取流名稱size_t streamStart = uri.find_last_of('/') + 1;std::string streamName = uri.substr(streamStart);auto it = mediaSources.find(streamName);if (it == mediaSources.end()) {return "RTSP/1.0 404 Not Found\r\n""CSeq: " + cseq + "\r\n\r\n";}// 提取客戶端端口size_t transportStart = request.find("Transport: ") + 11;size_t transportEnd = request.find("\r\n", transportStart);std::string transport = request.substr(transportStart, transportEnd - transportStart);size_t clientPortStart = transport.find("client_port=") + 12;size_t clientPortEnd = transport.find("-", clientPortStart);int clientRtpPort = std::stoi(transport.substr(clientPortStart, clientPortEnd - clientPortStart));int clientRtcpPort = std::stoi(transport.substr(clientPortEnd + 1));// 生成會話IDsessionId = generateSessionId();// 創建會話Session session;session.sessionId = sessionId;session.streamName = streamName;session.clientRtpPort = clientRtpPort;session.clientRtcpPort = clientRtcpPort;session.serverRtpPort = RTP_PORT_BASE + (sessions.size() * 2);session.serverRtcpPort = RTCP_PORT_BASE + (sessions.size() * 2);session.isPlaying = false;sessions[sessionId] = session;return "RTSP/1.0 200 OK\r\n""CSeq: " + cseq + "\r\n""Session: " + sessionId + "\r\n""Transport: RTP/AVP;unicast;client_port=" + std::to_string(clientRtpPort) + "-" + std::to_string(clientRtcpPort) + ";server_port=" + std::to_string(session.serverRtpPort) + "-" + std::to_string(session.serverRtcpPort) + "\r\n\r\n";}else if (method == "PLAY") {if (sessionId.empty()) {return "RTSP/1.0 454 Session Not Found\r\n""CSeq: " + cseq + "\r\n\r\n";}auto it = sessions.find(sessionId);if (it == sessions.end()) {return "RTSP/1.0 454 Session Not Found\r\n""CSeq: " + cseq + "\r\n\r\n";}// 啟動流發送線程if (!it->second.isPlaying) {it->second.isPlaying = true;it->second.streamThread = std::thread(&RTSPServer::sendMediaStream, this, sessionId);}return "RTSP/1.0 200 OK\r\n""CSeq: " + cseq + "\r\n""Session: " + sessionId + "\r\n""RTP-Info: url=" + uri + "\r\n\r\n";}else if (method == "PAUSE") {if (sessionId.empty()) {return "RTSP/1.0 454 Session Not Found\r\n""CSeq: " + cseq + "\r\n\r\n";}auto it = sessions.find(sessionId);if (it == sessions.end()) {return "RTSP/1.0 454 Session Not Found\r\n""CSeq: " + cseq + "\r\n\r\n";}it->second.isPlaying = false;if (it->second.streamThread.joinable()) {it->second.streamThread.join();}return "RTSP/1.0 200 OK\r\n""CSeq: " + cseq + "\r\n""Session: " + sessionId + "\r\n\r\n";}else if (method == "TEARDOWN") {if (sessionId.empty()) {return "RTSP/1.0 454 Session Not Found\r\n""CSeq: " + cseq + "\r\n\r\n";}auto it = sessions.find(sessionId);if (it == sessions.end()) {return "RTSP/1.0 454 Session Not Found\r\n""CSeq: " + cseq + "\r\n\r\n";}it->second.isPlaying = false;if (it->second.streamThread.joinable()) {it->second.streamThread.join();}sessions.erase(sessionId);sessionId.clear();return "RTSP/1.0 200 OK\r\n""CSeq: " + cseq + "\r\n""Session: " + sessionId + "\r\n\r\n";}else {return "RTSP/1.0 501 Not Implemented\r\n""CSeq: " + cseq + "\r\n\r\n";}}std::string generateSessionId() {static std::atomic<uint32_t> sessionCounter(0);return std::to_string(time(nullptr)) + "_" + std::to_string(sessionCounter++);}void sendMediaStream(const std::string& sessionId) {auto it = sessions.find(sessionId);if (it == sessions.end()) {return;}Session& session = it->second;auto mediaSourceIt = mediaSources.find(session.streamName);if (mediaSourceIt == mediaSources.end()) {return;}std::shared_ptr<MediaSource> mediaSource = mediaSourceIt->second;// 創建RTP套接字int rtpSocket = socket(AF_INET, SOCK_DGRAM, 0);if (rtpSocket < 0) {std::cerr << "Failed to create RTP socket" << std::endl;return;}// 設置目標地址sockaddr_in clientAddr;memset(&clientAddr, 0, sizeof(clientAddr));clientAddr.sin_family = AF_INET;clientAddr.sin_addr.s_addr = INADDR_ANY;  // 注意：實際應用中應使用客戶端IPclientAddr.sin_port = htons(session.clientRtpPort);// 初始化RTP參數RTPPacket rtpPacket;rtpPacket.ssrc = rand();rtpPacket.sequence = rand() % 65536;rtpPacket.timestamp = rand() % 4294967296;// 幀率控制const int fps = 30;const int frameDuration = 1000 / fps;  // 毫秒uint32_t frameCount = 0;while (session.isPlaying) {std::vector<uint8_t> frameData;FrameType frameType;// 獲取下一幀if (!mediaSource->getNextFrame(frameData, frameType)) {break;}// 分片并發送if (mediaSource->getMimeType() == "video/H264") {sendH264Frame(rtpSocket, clientAddr, frameData, frameType, rtpPacket);} else {// 其他媒體類型的處理sendRtpPacket(rtpSocket, clientAddr, frameData, rtpPacket);}// 更新時間戳和序列號rtpPacket.sequence = (rtpPacket.sequence + 1) % 65536;rtpPacket.timestamp += 90000 / fps;  // 90kHz時鐘frameCount++;// 控制幀率std::this_thread::sleep_for(std::chrono::milliseconds(frameDuration));}close(rtpSocket);}void sendH264Frame(int socket, sockaddr_in& addr, const std::vector<uint8_t>& frameData, FrameType frameType, RTPPacket& rtpPacket) {// 查找幀中的各個NAL單元size_t offset = 0;while (offset < frameData.size()) {// 查找NAL單元起始碼 (0x00000001或0x000001)size_t nalStart = offset;if (frameData.size() - offset >= 4 && frameData[offset] == 0x00 && frameData[offset+1] == 0x00 && frameData[offset+2] == 0x00 && frameData[offset+3] == 0x01) {nalStart += 4;offset += 4;} else if (frameData.size() - offset >= 3 && frameData[offset] == 0x00 && frameData[offset+1] == 0x00 && frameData[offset+2] == 0x01) {nalStart += 3;offset += 3;} else {// 沒有找到起始碼，可能是第一個NALnalStart = offset;}// 查找下一個NAL單元的起始位置size_t nextNalStart = frameData.size();for (size_t i = offset; i < frameData.size() - 3; i++) {if ((frameData[i] == 0x00 && frameData[i+1] == 0x00 && frameData[i+2] == 0x00 && frameData[i+3] == 0x01) ||(frameData[i] == 0x00 && frameData[i+1] == 0x00 && frameData[i+2] == 0x01)) {nextNalStart = i;break;}}size_t nalSize = nextNalStart - nalStart;if (nalSize <= 0) {break;}// 處理NAL單元const uint8_t* nalData = &frameData[nalStart];uint8_t nalType = nalData[0] & 0x1F;// 根據NAL單元類型設置RTP標記位bool isKeyFrame = (nalType == NAL_IDR_SLICE || nalType == NAL_SPS || nalType == NAL_PPS);rtpPacket.marker = isKeyFrame ? 1 : 0;// 分片處理if (nalSize > MAX_PACKET_SIZE) {fragmentAndSendNAL(socket, addr, nalData, nalSize, rtpPacket);} else {// 不需要分片，直接發送rtpPacket.payload.assign(nalData, nalData + nalSize);sendRtpPacket(socket, addr, rtpPacket);rtpPacket.sequence = (rtpPacket.sequence + 1) % 65536;}offset = nextNalStart;}}void fragmentAndSendNAL(int socket, sockaddr_in& addr, const uint8_t* nalData, size_t nalSize, RTPPacket& rtpPacket) {uint8_t originalNalType = nalData[0] & 0x1F;uint8_t fuIndicator = 0x7C;  // FU-A indicator (Type=28)size_t payloadSize = MAX_PACKET_SIZE - 2;  // 減去FU indicator和FU header的大小size_t offset = 1;  // 跳過原始NAL頭部while (offset < nalSize) {size_t fragmentSize = std::min(payloadSize, nalSize - offset);bool isFirst = (offset == 1);bool isLast = (offset + fragmentSize >= nalSize);// 創建FU headeruint8_t fuHeader = originalNalType;if (isFirst) fuHeader |= 0x80;  // 設置起始位if (isLast) fuHeader |= 0x40;   // 設置結束位// 構建RTP負載rtpPacket.payload.resize(2 + fragmentSize);rtpPacket.payload[0] = fuIndicator;rtpPacket.payload[1] = fuHeader;memcpy(&rtpPacket.payload[2], nalData + offset, fragmentSize);// 設置標記位（僅最后一個分片設置）rtpPacket.marker = isLast ? 1 : 0;// 發送RTP包sendRtpPacket(socket, addr, rtpPacket);// 更新序列號rtpPacket.sequence = (rtpPacket.sequence + 1) % 65536;offset += fragmentSize;}}void sendRtpPacket(int socket, sockaddr_in& addr, const std::vector<uint8_t>& payload, RTPPacket& rtpPacket) {// 構建RTP頭部uint8_t rtpHeader[12];rtpHeader[0] = (rtpPacket.version << 6) | (rtpPacket.padding << 5) | (rtpPacket.extension << 4) | rtpPacket.cc;rtpHeader[1] = (rtpPacket.marker << 7) | rtpPacket.payloadType;rtpHeader[2] = (rtpPacket.sequence >> 8) & 0xFF;rtpHeader[3] = rtpPacket.sequence & 0xFF;rtpHeader[4] = (rtpPacket.timestamp >> 24) & 0xFF;rtpHeader[5] = (rtpPacket.timestamp >> 16) & 0xFF;rtpHeader[6] = (rtpPacket.timestamp >> 8) & 0xFF;rtpHeader[7] = rtpPacket.timestamp & 0xFF;rtpHeader[8] = (rtpPacket.ssrc >> 24) & 0xFF;rtpHeader[9] = (rtpPacket.ssrc >> 16) & 0xFF;rtpHeader[10] = (rtpPacket.ssrc >> 8) & 0xFF;rtpHeader[11] = rtpPacket.ssrc & 0xFF;// 發送RTP包std::vector<uint8_t> packet;packet.insert(packet.end(), rtpHeader, rtpHeader + 12);packet.insert(packet.end(), payload.begin(), payload.end());sendto(socket, packet.data(), packet.size(), 0, (struct sockaddr*)&addr, sizeof(addr));}void sendRtpPacket(int socket, sockaddr_in& addr, RTPPacket& rtpPacket) {sendRtpPacket(socket, addr, rtpPacket.payload, rtpPacket);}
};// RTSP客戶端
class RTSPClient {
private:int rtspSocket;std::string sessionId;int cseq;std::string serverAddress;int serverPort;int rtpPort;int rtcpPort;std::atomic<bool> isPlaying;std::thread receiveThread;std::function<void(const uint8_t*, size_t, uint32_t, bool)> frameCallback;public:RTSPClient() : rtspSocket(-1), cseq(0), serverPort(RTSP_PORT), rtpPort(0), rtcpPort(0), isPlaying(false) {}~RTSPClient() {disconnect();}bool connect(const std::string& url) {// 解析URLsize_t protocolEnd = url.find("://");if (protocolEnd == std::string::npos) {std::cerr << "Invalid RTSP URL" << std::endl;return false;}std::string protocol = url.substr(0, protocolEnd);if (protocol != "rtsp") {std::cerr << "Unsupported protocol: " << protocol << std::endl;return false;}size_t hostStart = protocolEnd + 3;size_t hostEnd = url.find(":", hostStart);size_t pathStart = url.find("/", hostStart);if (hostEnd != std::string::npos && pathStart != std::string::npos && hostEnd < pathStart) {// 包含端口號serverAddress = url.substr(hostStart, hostEnd - hostStart);serverPort = std::stoi(url.substr(hostEnd + 1, pathStart - hostEnd - 1));} else if (pathStart != std::string::npos) {// 不包含端口號，使用默認端口serverAddress = url.substr(hostStart, pathStart - hostStart);} else {std::cerr << "Invalid RTSP URL" << std::endl;return false;}std::string path = url.substr(pathStart);// 創建RTSP套接字rtspSocket = socket(AF_INET, SOCK_STREAM, 0);if (rtspSocket < 0) {std::cerr << "Failed to create RTSP socket" << std::endl;return false;}// 連接服務器sockaddr_in serverAddr;memset(&serverAddr, 0, sizeof(serverAddr));serverAddr.sin_family = AF_INET;serverAddr.sin_port = htons(serverPort);inet_pton(AF_INET, serverAddress.c_str(), &serverAddr.sin_addr);if (connect(rtspSocket, (struct sockaddr*)&serverAddr, sizeof(serverAddr)) < 0) {std::cerr << "Failed to connect to RTSP server" << std::endl;close(rtspSocket);rtspSocket = -1;return false;}// 發送OPTIONS請求std::string response = sendRequest("OPTIONS " + url + " RTSP/1.0\r\n\r\n");if (!parseResponseStatus(response, 200)) {std::cerr << "OPTIONS request failed" << std::endl;disconnect();return false;}// 發送DESCRIBE請求response = sendRequest("DESCRIBE " + url + " RTSP/1.0\r\n""Accept: application/sdp\r\n\r\n");if (!parseResponseStatus(response, 200)) {std::cerr << "DESCRIBE request failed" << std::endl;disconnect();return false;}// 解析SDP信息std::string sdp = parseSDP(response);if (sdp.empty()) {std::cerr << "Failed to parse SDP" << std::endl;disconnect();return false;}// 隨機選擇RTP/RTCP端口rtpPort = 10000 + (rand() % 5000) * 2;rtcpPort = rtpPort + 1;// 發送SETUP請求response = sendRequest("SETUP " + url + " RTSP/1.0\r\n""Transport: RTP/AVP;unicast;client_port=" + std::to_string(rtpPort) + "-" + std::to_string(rtcpPort) + "\r\n\r\n");if (!parseResponseStatus(response, 200)) {std::cerr << "SETUP request failed" << std::endl;disconnect();return false;}// 解析會話IDsessionId = parseSessionId(response);if (sessionId.empty()) {std::cerr << "Failed to get session ID" << std::endl;disconnect();return false;}return true;}void disconnect() {if (isPlaying) {pause();}if (rtspSocket >= 0) {// 發送TEARDOWN請求if (!sessionId.empty()) {sendRequest("TEARDOWN * RTSP/1.0\r\n""Session: " + sessionId + "\r\n\r\n");}close(rtspSocket);rtspSocket = -1;sessionId.clear();}if (receiveThread.joinable()) {receiveThread.join();}}bool play() {if (!isPlaying && rtspSocket >= 0 && !sessionId.empty()) {std::string response = sendRequest("PLAY * RTSP/1.0\r\n""Session: " + sessionId + "\r\n\r\n");if (!parseResponseStatus(response, 200)) {std::cerr << "PLAY request failed" << std::endl;return false;}isPlaying = true;receiveThread = std::thread(&RTSPClient::receiveRtpPackets, this);return true;}return false;}bool pause() {if (isPlaying && rtspSocket >= 0 && !sessionId.empty()) {isPlaying = false;if (receiveThread.joinable()) {receiveThread.join();}std::string response = sendRequest("PAUSE * RTSP/1.0\r\n""Session: " + sessionId + "\r\n\r\n");return parseResponseStatus(response, 200);}return false;}void setFrameCallback(const std::function<void(const uint8_t*, size_t, uint32_t, bool)>& callback) {frameCallback = callback;}std::string getMediaInfo() {// 這里應該返回從SDP解析的媒體信息return "Media Info: RTSP Stream";}private:std::string sendRequest(const std::string& request) {if (rtspSocket < 0) {return "";}// 增加CSeqcseq++;std::string requestWithCSeq = request;size_t crlfPos = request.find("\r\n");if (crlfPos != std::string::npos) {requestWithCSeq.insert(crlfPos + 2, "CSeq: " + std::to_string(cseq) + "\r\n");}// 發送請求send(rtspSocket, requestWithCSeq.c_str(), requestWithCSeq.size(), 0);// 接收響應char buffer[4096];memset(buffer, 0, sizeof(buffer));int bytesRead = recv(rtspSocket, buffer, sizeof(buffer) - 1, 0);return std::string(buffer, bytesRead);}bool parseResponseStatus(const std::string& response, int expectedStatus) {size_t statusStart = response.find("RTSP/1.0 ") + 9;size_t statusEnd = response.find(" ", statusStart);if (statusStart == std::string::npos || statusEnd == std::string::npos) {return false;}int statusCode = std::stoi(response.substr(statusStart, statusEnd - statusStart));return (statusCode == expectedStatus);}std::string parseSDP(const std::string& response) {size_t sdpStart = response.find("\r\n\r\n");if (sdpStart == std::string::npos) {return "";}return response.substr(sdpStart + 4);}std::string parseSessionId(const std::string& response) {size_t sessionStart = response.find("Session: ");if (sessionStart == std::string::npos) {return "";}sessionStart += 9;size_t sessionEnd = response.find("\r\n", sessionStart);if (sessionEnd == std::string::npos) {return "";}std::string sessionLine = response.substr(sessionStart, sessionEnd - sessionStart);size_t idEnd = sessionLine.find(";");if (idEnd != std::string::npos) {return sessionLine.substr(0, idEnd);}return sessionLine;}void receiveRtpPackets() {// 創建RTP套接字int rtpSocket = socket(AF_INET, SOCK_DGRAM, 0);if (rtpSocket < 0) {std::cerr << "Failed to create RTP socket" << std::endl;return;}// 綁定端口sockaddr_in localAddr;memset(&localAddr, 0, sizeof(localAddr));localAddr.sin_family = AF_INET;localAddr.sin_addr.s_addr = INADDR_ANY;localAddr.sin_port = htons(rtpPort);if (bind(rtpSocket, (struct sockaddr*)&localAddr, sizeof(localAddr)) < 0) {std::cerr << "Failed to bind RTP socket" << std::endl;close(rtpSocket);return;}// 創建緩沖區uint8_t buffer[65536];std::vector<uint8_t> nalBuffer;uint32_t currentTimestamp = 0;bool isKeyFrame = false;// 接收RTP包while (isPlaying) {sockaddr_in senderAddr;socklen_t senderAddrLen = sizeof(senderAddr);int bytesRead = recvfrom(rtpSocket, buffer, sizeof(buffer), 0, (struct sockaddr*)&senderAddr, &senderAddrLen);if (bytesRead <= 0) {continue;}// 解析RTP頭部uint8_t version = (buffer[0] >> 6) & 0x03;uint8_t padding = (buffer[0] >> 5) & 0x01;uint8_t extension = (buffer[0] >> 4) & 0x01;uint8_t cc = buffer[0] & 0x0F;uint8_t marker = (buffer[1] >> 7) & 0x01;uint8_t payloadType = buffer[1] & 0x7F;uint16_t sequence = (buffer[2] << 8) | buffer[3];uint32_t timestamp = (buffer[4] << 24) | (buffer[5] << 16) | (buffer[6] << 8) | buffer[7];uint32_t ssrc = (buffer[8] << 24) | (buffer[9] << 16) | (buffer[10] << 8) | buffer[11];// 提取RTP負載size_t headerSize = 12 + (cc * 4);if (extension) {// 處理擴展頭部headerSize += 4 + ((buffer[12] << 8) | buffer[13]) * 4;}const uint8_t* payload = buffer + headerSize;size_t payloadSize = bytesRead - headerSize;// 檢查是否有幀回調if (frameCallback && payloadSize > 0) {// 檢查NAL單元類型if (payloadSize >= 1) {uint8_t nalType = payload[0] & 0x1F;// 判斷是否為關鍵幀isKeyFrame = (nalType == NAL_IDR_SLICE || nalType == NAL_SPS || nalType == NAL_PPS);// 處理FU-A分片if (nalType == 28 && payloadSize >= 2) {  // FU-Auint8_t fuHeader = payload[1];bool isStart = (fuHeader & 0x80) != 0;bool isEnd = (fuHeader & 0x40) != 0;if (isStart) {// 開始新的NAL單元nalBuffer.clear();// 重建原始NAL頭部uint8_t originalNalHeader = (payload[0] & 0xE0) | (fuHeader & 0x1F);nalBuffer.push_back(originalNalHeader);// 更新時間戳currentTimestamp = timestamp;}// 添加分片數據nalBuffer.insert(nalBuffer.end(), payload + 2, payload + payloadSize);// 如果是結束分片且有回調，則傳遞完整的NAL單元if (isEnd && !nalBuffer.empty()) {frameCallback(nalBuffer.data(), nalBuffer.size(), currentTimestamp, isKeyFrame);}} else {// 非分片NAL單元，直接傳遞frameCallback(payload, payloadSize, timestamp, isKeyFrame);}}}}close(rtpSocket);}
};

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