Disabling test on iOS64 debug bot

webrtc/ortc/ortcfactory_integrationtest.cc

 /*
  *  Copyright 2017 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 463 matching lines @@
 TEST_F(OrtcFactoryIntegrationTest,
        BasicTwoWayAudioVideoRtpSendersAndReceivers) {
   auto udp_transports = CreateAndConnectUdpTransportPair();
   auto rtp_transports =
       CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
   bool expect_success = true;
   BasicTwoWayRtpSendersAndReceiversTest(std::move(rtp_transports),
                                         expect_success);
 }
 
+#if !(defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_64_BITS) && !defined(NDEBUG))
 TEST_F(OrtcFactoryIntegrationTest,
        BasicTwoWayAudioVideoSrtpSendersAndReceivers) {
   auto udp_transports = CreateAndConnectUdpTransportPair();
   auto srtp_transports = CreateSrtpTransportPairAndSetKeys(
       MakeRtcpMuxParameters(), udp_transports);
   bool expect_success = true;
   BasicTwoWayRtpSendersAndReceiversTest(std::move(srtp_transports),
                                         expect_success);
 }
+#endif
 
 // Tests that the packets cannot be decoded if the keys are mismatched.
 TEST_F(OrtcFactoryIntegrationTest, SrtpSendersAndReceiversWithMismatchingKeys) {
   auto udp_transports = CreateAndConnectUdpTransportPair();
   auto srtp_transports = CreateSrtpTransportPairAndSetMismatchingKeys(
       MakeRtcpMuxParameters(), udp_transports);
   bool expect_success = false;
   BasicTwoWayRtpSendersAndReceiversTest(std::move(srtp_transports),
                                         expect_success);
   // No frames are expected to be decoded.
@@ skipping 177 matching lines @@
 // BaseChannel model relies on there being a single VoiceChannel and
 // VideoChannel, and these only support a single set of codecs/etc. per
 // send/receive direction.
 
 // TODO(deadbeef): End-to-end test for simulcast, once that's supported by this
 // API.
 
 #endif  // if !defined(THREAD_SANITIZER)
 
 }  // namespace webrtc