初步添加rtsp转webrtc相关功能

webrtc/srtp_session.h

@@ -0,0 +1,54 @@
+#ifndef MS_RTC_SRTP_SESSION_HPP
+#define MS_RTC_SRTP_SESSION_HPP
+
+#include "rtc_dtls_transport.h"
+#include "utils.h"
+#include <srtp2/srtp.h>
+#include <vector>
+
+namespace RTC {
+
+class DepLibSRTP {
+public:
+    static void ClassInit();
+    static void ClassDestroy();
+    static bool IsError(srtp_err_status_t code) { return (code != srtp_err_status_ok); }
+    static const char *GetErrorString(srtp_err_status_t code) {
+        // This throws out_of_range if the given index is not in the vector.
+        return DepLibSRTP::errors.at(code);
+    }
+
+private:
+    static std::vector<const char *> errors;
+};
+
+class SrtpSession {
+public:
+public:
+    enum class Type { INBOUND = 1, OUTBOUND };
+
+public:
+    static void ClassInit();
+
+private:
+    static void OnSrtpEvent(srtp_event_data_t *data);
+
+public:
+    SrtpSession(Type type, CryptoSuite cryptoSuite, uint8_t *key, size_t keyLen);
+    ~SrtpSession();
+
+public:
+    bool EncryptRtp(const uint8_t **data, size_t *len);
+    bool DecryptSrtp(uint8_t *data, size_t *len);
+    bool EncryptRtcp(const uint8_t **data, size_t *len);
+    bool DecryptSrtcp(uint8_t *data, size_t *len);
+    void RemoveStream(uint32_t ssrc) { srtp_remove_stream(this->session, uint32_t{htonl(ssrc)}); }
+
+private:
+    bool is_init = false;
+    // Allocated by this.
+    srtp_t session{nullptr};
+};
+}// namespace RTC
+
+#endif