Ensures that modules_unittests runs on iOS

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  Copyright (c) 2011 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 419 matching lines @@
                                     size_t* payload_len) {
   rtp_info->header.sequenceNumber = frame_index;
   rtp_info->header.timestamp = timestamp;
   rtp_info->header.ssrc = 0x1234;  // Just an arbitrary SSRC.
   rtp_info->header.payloadType = 98;  // WB CNG.
   rtp_info->header.markerBit = 0;
   payload[0] = 64;  // Noise level -64 dBov, quite arbitrarily chosen.
   *payload_len = 1;  // Only noise level, no spectral parameters.
 }
 
-TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestBitExactness)) {
+// TODO(henrika): add support for IOS for all tests in this file.
+// See https://code.google.com/p/webrtc/issues/detail?id=4752 for details.
+TEST_F(NetEqDecodingTest,
+       DISABLED_ON_IOS(DISABLED_ON_ANDROID(TestBitExactness))) {
   const std::string input_rtp_file = webrtc::test::ProjectRootPath() +
       "resources/audio_coding/neteq_universal_new.rtp";
   // Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm
   // are identical. The latter could have been removed, but if clients still
   // have a copy of the file, the test will fail.
   const std::string input_ref_file =
       webrtc::test::ResourcePath("audio_coding/neteq4_universal_ref", "pcm");
 #if defined(_MSC_VER) && (_MSC_VER >= 1700)
   // For Visual Studio 2012 and later, we will have to use the generic reference
   // file, rather than the windows-specific one.
@@ skipping 594 matching lines @@
   NetEqBgnTestFade() : NetEqBgnTest() {
     config_.background_noise_mode = NetEq::kBgnFade;
   }
 
   void TestCondition(double sum_squared_noise, bool should_be_faded) {
     if (should_be_faded)
       EXPECT_EQ(0, sum_squared_noise);
   }
 };
 
-TEST_F(NetEqBgnTestOn, RunTest) {
+TEST_F(NetEqBgnTestOn, DISABLED_ON_IOS(RunTest)) {
   CheckBgn(8000);
   CheckBgn(16000);
   CheckBgn(32000);
 }
 
-TEST_F(NetEqBgnTestOff, RunTest) {
+TEST_F(NetEqBgnTestOff, DISABLED_ON_IOS(RunTest)) {
   CheckBgn(8000);
   CheckBgn(16000);
   CheckBgn(32000);
 }
 
-TEST_F(NetEqBgnTestFade, RunTest) {
+TEST_F(NetEqBgnTestFade, DISABLED_ON_IOS(RunTest)) {
   CheckBgn(8000);
   CheckBgn(16000);
   CheckBgn(32000);
 }
 
 TEST_F(NetEqDecodingTest, SyncPacketInsert) {
   WebRtcRTPHeader rtp_info;
   uint32_t receive_timestamp = 0;
   // For the readability use the following payloads instead of the defaults of
   // this test.
@@ skipping 453 matching lines @@
     // Pull audio once.
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
     ASSERT_EQ(kBlockSize16kHz, out_len);
   }
   // Verify speech output.
   EXPECT_EQ(kOutputNormal, type);
 }
 
 }  // namespace webrtc