Ensures that modules_unittests runs on iOS

webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc

 /*
  *  Copyright (c) 2014 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 577 matching lines @@
   rtc::scoped_ptr<ThreadWrapper> pull_audio_thread_;
   const rtc::scoped_ptr<EventWrapper> test_complete_;
   int send_count_;
   int insert_packet_count_;
   int pull_audio_count_ GUARDED_BY(crit_sect_);
   const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<SimulatedClock> fake_clock_;
 };
 
-TEST_F(AudioCodingModuleMtTestOldApi, DoTest) {
+TEST_F(AudioCodingModuleMtTestOldApi, DISABLED_ON_IOS(DoTest)) {
   EXPECT_EQ(kEventSignaled, RunTest());
 }
 
 // This is a multi-threaded ACM test using iSAC. The test encodes audio
 // from a PCM file. The most recent encoded frame is used as input to the
 // receiving part. Depending on timing, it may happen that the same RTP packet
 // is inserted into the receiver multiple times, but this is a valid use-case,
 // and simplifies the test code a lot.
 class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
  protected:
@@ skipping 79 matching lines @@
       }
     }
     return false;
   }
 
   int last_packet_number_;
   std::vector<uint8_t> last_payload_vec_;
   test::AudioLoop audio_loop_;
 };
 
-TEST_F(AcmIsacMtTestOldApi, DoTest) {
+TEST_F(AcmIsacMtTestOldApi, DISABLED_ON_IOS(DoTest)) {
   EXPECT_EQ(kEventSignaled, RunTest());
 }
 
+// Disabling all of these tests on iOS until file support has been added.
+// See https://code.google.com/p/webrtc/issues/detail?id=4752 for details.
+#if !defined(WEBRTC_IOS)
+
 class AcmReceiverBitExactnessOldApi : public ::testing::Test {
  public:
   static std::string PlatformChecksum(std::string win64,
                                       std::string android,
                                       std::string others) {
 #if defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS)
     return win64;
 #elif defined(WEBRTC_ANDROID)
     return android;
 #else
@@ skipping 805 matching lines @@
   Run(32000, 16000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle16KhzTo8Khz) {
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
+
+#endif
+
 }  // namespace webrtc