Add support for GCM cipher suites from RFC 7714.

webrtc/pc/channel.h

Index: webrtc/pc/channel.h
diff --git a/webrtc/pc/channel.h b/webrtc/pc/channel.h
index 4518301d3b65e9e6fdf89586d258b519e42a4677..5379afe49682e8dc0c8276a480c0c3cce4bbce6e 100644
--- a/webrtc/pc/channel.h
+++ b/webrtc/pc/channel.h
@@ -158,6 +158,8 @@ class BaseChannel
 
   SrtpFilter* srtp_filter() { return &srtp_filter_; }
 
+  bool SetCryptoOptions(const rtc::CryptoOptions& crypto_options);
+
  protected:
   virtual MediaChannel* media_channel() const { return media_channel_; }
   // Sets the |transport_channel_| (and |rtcp_transport_channel_|, if |rtcp_| is
@@ -283,7 +285,8 @@ class BaseChannel
 
   // Handled in derived classes
   // Get the SRTP crypto suites to use for RTP media
-  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const = 0;
+  virtual void GetSrtpCryptoSuites(const rtc::CryptoOptions& crypto_options,
+      std::vector<int>* crypto_suites) const = 0;
   virtual void OnConnectionMonitorUpdate(ConnectionMonitor* monitor,
       const std::vector<ConnectionInfo>& infos) = 0;
 
@@ -321,6 +324,7 @@ class BaseChannel
   bool has_received_packet_;
   bool dtls_keyed_;
   bool secure_required_;
+  rtc::CryptoOptions crypto_options_;

[mattdr, 2016/05/06 22:34:13]

Should we make this available as a public or protected getter, and then avoid
having to plumb crypto_options through the GetSrtpCryptoSuites functions?

[joachim, 2016/05/09 23:21:40]

Done.

   int rtp_abs_sendtime_extn_id_;
 };
 
@@ -411,7 +415,8 @@ class VoiceChannel : public BaseChannel {
   bool GetStats_w(VoiceMediaInfo* stats);
 
   virtual void OnMessage(rtc::Message* pmsg);
-  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
+  void GetSrtpCryptoSuites(const rtc::CryptoOptions& crypto_options,
+      std::vector<int>* crypto_suites) const override;
   virtual void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
   virtual void OnMediaMonitorUpdate(
@@ -482,7 +487,8 @@ class VideoChannel : public BaseChannel {
   bool SetRtpParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
 
   virtual void OnMessage(rtc::Message* pmsg);
-  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
+  void GetSrtpCryptoSuites(const rtc::CryptoOptions& crypto_options,
+      std::vector<int>* crypto_suites) const override;
   virtual void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
   virtual void OnMediaMonitorUpdate(
@@ -592,7 +598,8 @@ class DataChannel : public BaseChannel {
   virtual bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
 
   virtual void OnMessage(rtc::Message* pmsg);
-  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
+  void GetSrtpCryptoSuites(const rtc::CryptoOptions& crypto_options,
+      std::vector<int>* crypto_suites) const override;
   virtual void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
   virtual void OnMediaMonitorUpdate(