Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.

webrtc/pc/srtpfilter.h

 /*
  *  Copyright 2009 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 96 matching lines @@
   // If an HMAC is used, this will decrease the packet size.
   bool UnprotectRtp(void* data, int in_len, int* out_len);
   bool UnprotectRtcp(void* data, int in_len, int* out_len);
 
   // Returns rtp auth params from srtp context.
   bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len);
 
   // Returns srtp overhead for rtp packets.
   bool GetSrtpOverhead(int* srtp_overhead) const;
 
+#if defined(ENABLE_EXTERNAL_AUTH)
+  // A SRTP filter supports external creation of the auth tag if a non-GCM
+  // cipher is used. This method is only valid after the RTP params have
+  // been set.
+  bool IsExternalAuthActive() const;
+#endif
+
   // Update the silent threshold (in ms) for signaling errors.
   void set_signal_silent_time(int signal_silent_time_in_ms);
 
   bool ResetParams();
 
   sigslot::repeater3<uint32_t, Mode, Error> SignalSrtpError;
 
  protected:
   bool ExpectOffer(ContentSource source);
   bool StoreParams(const std::vector<CryptoParams>& params,
@@ skipping 72 matching lines @@
   // Decrypts/verifies an invidiual RTP/RTCP packet.
   // If an HMAC is used, this will decrease the packet size.
   bool UnprotectRtp(void* data, int in_len, int* out_len);
   bool UnprotectRtcp(void* data, int in_len, int* out_len);
 
   // Helper method to get authentication params.
   bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len);
 
   int GetSrtpOverhead() const;
 
+#if defined(ENABLE_EXTERNAL_AUTH)
+  // A SRTP session supports external creation of the auth tag if a non-GCM
+  // cipher is used. This method is only valid after the RTP params have
+  // been set.
+  bool IsExternalAuthActive() const;
+#endif
+
   // Update the silent threshold (in ms) for signaling errors.
   void set_signal_silent_time(int signal_silent_time_in_ms);
 
   // Calls srtp_shutdown if it's initialized.
   static void Terminate();
 
   sigslot::repeater3<uint32_t, SrtpFilter::Mode, SrtpFilter::Error>
       SignalSrtpError;
 
  private:
   bool SetKey(int type, int cs, const uint8_t* key, size_t len);
     // Returns send stream current packet index from srtp db.
   bool GetSendStreamPacketIndex(void* data, int in_len, int64_t* index);
 
   static bool Init();
   void HandleEvent(const srtp_event_data_t* ev);
   static void HandleEventThunk(srtp_event_data_t* ev);
 
   rtc::ThreadChecker thread_checker_;
-  srtp_ctx_t_* session_;
-  int rtp_auth_tag_len_;
-  int rtcp_auth_tag_len_;
+  srtp_ctx_t_* session_ = nullptr;
+  int rtp_auth_tag_len_ = 0;
+  int rtcp_auth_tag_len_ = 0;
   std::unique_ptr<SrtpStat> srtp_stat_;
   static bool inited_;
   static rtc::GlobalLockPod lock_;
-  int last_send_seq_num_;
+  int last_send_seq_num_ = -1;
+#if defined(ENABLE_EXTERNAL_AUTH)
+  bool external_auth_active_ = false;
+#endif
   RTC_DISALLOW_COPY_AND_ASSIGN(SrtpSession);
 };
 
 // Class that collects failures of SRTP.
 class SrtpStat {
  public:
   SrtpStat();
 
   // Report RTP protection results to the handler.
   void AddProtectRtpResult(uint32_t ssrc, int result);
@@ skipping 58 matching lines @@
   std::map<FailureKey, FailureStat> failures_;
   // Threshold in ms to silent the signaling errors.
   int signal_silent_time_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(SrtpStat);
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_SRTPFILTER_H_