Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.

webrtc/pc/channel.cc

 /*
  *  Copyright 2004 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 25 matching lines @@
                        std::unique_ptr<webrtc::AudioSinkInterface>* sink) {
   channel->SetRawAudioSink(ssrc, std::move(*sink));
   return true;
 }
 
 struct SendPacketMessageData : public rtc::MessageData {
   rtc::CopyOnWriteBuffer packet;
   rtc::PacketOptions options;
 };
 
-#if defined(ENABLE_EXTERNAL_AUTH)
 // Returns the named header extension if found among all extensions,
 // nullptr otherwise.
 const webrtc::RtpExtension* FindHeaderExtension(
     const std::vector<webrtc::RtpExtension>& extensions,
     const std::string& uri) {
   for (const auto& extension : extensions) {
     if (extension.uri == uri)
       return &extension;
   }
   return nullptr;
 }
-#endif
 
 }  // namespace
 
 enum {
   MSG_EARLYMEDIATIMEOUT = 1,
   MSG_SEND_RTP_PACKET,
   MSG_SEND_RTCP_PACKET,
   MSG_CHANNEL_ERROR,
   MSG_READYTOSENDDATA,
   MSG_DATARECEIVED,
@@ skipping 660 matching lines @@
     TRACE_EVENT0("webrtc", "SRTP Encode");
     bool res;
     uint8_t* data = packet->data();
     int len = static_cast<int>(packet->size());
     if (!rtcp) {
     // If ENABLE_EXTERNAL_AUTH flag is on then packet authentication is not done
     // inside libsrtp for a RTP packet. A external HMAC module will be writing
     // a fake HMAC value. This is ONLY done for a RTP packet.
     // Socket layer will update rtp sendtime extension header if present in
     // packet with current time before updating the HMAC.
-#if !defined(ENABLE_EXTERNAL_AUTH)
-      res = srtp_filter_.ProtectRtp(
-          data, len, static_cast<int>(packet->capacity()), &len);
-#else
-      updated_options.packet_time_params.rtp_sendtime_extension_id =
-          rtp_abs_sendtime_extn_id_;
-      res = srtp_filter_.ProtectRtp(
-          data, len, static_cast<int>(packet->capacity()), &len,
-          &updated_options.packet_time_params.srtp_packet_index);
-      // If protection succeeds, let's get auth params from srtp.
-      if (res) {
-        uint8_t* auth_key = NULL;
-        int key_len;
-        res = srtp_filter_.GetRtpAuthParams(
-            &auth_key, &key_len,
-            &updated_options.packet_time_params.srtp_auth_tag_len);
+      if (!srtp_filter_.AllowExternalAuth()) {
+        res = srtp_filter_.ProtectRtp(
+            data, len, static_cast<int>(packet->capacity()), &len);
+      } else {
+        updated_options.packet_time_params.rtp_sendtime_extension_id =
+            rtp_abs_sendtime_extn_id_;
+        res = srtp_filter_.ProtectRtp(
+            data, len, static_cast<int>(packet->capacity()), &len,
+            &updated_options.packet_time_params.srtp_packet_index);
+        // If protection succeeds, let's get auth params from srtp.
         if (res) {
-          updated_options.packet_time_params.srtp_auth_key.resize(key_len);
-          updated_options.packet_time_params.srtp_auth_key.assign(
-              auth_key, auth_key + key_len);
+          uint8_t* auth_key = NULL;
+          int key_len;
+          res = srtp_filter_.GetRtpAuthParams(
+              &auth_key, &key_len,
+              &updated_options.packet_time_params.srtp_auth_tag_len);
+          if (res) {
+            updated_options.packet_time_params.srtp_auth_key.resize(key_len);
+            updated_options.packet_time_params.srtp_auth_key.assign(
+                auth_key, auth_key + key_len);
+          }
         }
       }
-#endif
       if (!res) {
         int seq_num = -1;
         uint32_t ssrc = 0;
         GetRtpSeqNum(data, len, &seq_num);
         GetRtpSsrc(data, len, &ssrc);
         LOG(LS_ERROR) << "Failed to protect " << content_name_
                       << " RTP packet: size=" << len
                       << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
         return false;
       }
@@ skipping 681 matching lines @@
   }
   remote_streams_ = streams;
   return ret;
 }
 
 void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension_w(
     const std::vector<webrtc::RtpExtension>& extensions) {
 // Absolute Send Time extension id is used only with external auth,
 // so do not bother searching for it and making asyncronious call to set
 // something that is not used.
-#if defined(ENABLE_EXTERNAL_AUTH)

[Taylor Brandstetter, 2017/02/28 22:53:40]

Replace #ifdef with "if (external_auth_enabled_)"?

   const webrtc::RtpExtension* send_time_extension =
       FindHeaderExtension(extensions, webrtc::RtpExtension::kAbsSendTimeUri);
   int rtp_abs_sendtime_extn_id =
       send_time_extension ? send_time_extension->id : -1;
   invoker_.AsyncInvoke<void>(
       RTC_FROM_HERE, network_thread_,
       Bind(&BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n, this,
            rtp_abs_sendtime_extn_id));
-#endif
 }
 
 void BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n(
     int rtp_abs_sendtime_extn_id) {
   rtp_abs_sendtime_extn_id_ = rtp_abs_sendtime_extn_id;
 }
 
 void BaseChannel::OnMessage(rtc::Message *pmsg) {
   TRACE_EVENT0("webrtc", "BaseChannel::OnMessage");
   switch (pmsg->message_id) {
@@ skipping 974 matching lines @@
   signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA,
                            new DataChannelReadyToSendMessageData(writable));
 }
 
 void RtpDataChannel::GetSrtpCryptoSuites_n(
     std::vector<int>* crypto_suites) const {
   GetSupportedDataCryptoSuites(crypto_options(), crypto_suites);
 }
 
 }  // namespace cricket