Revert of Disabled two iOS tests due to bot breakage. Affected tests are

webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc

 /*
  *  Copyright (c) 2015 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 580 matching lines @@
 // Failing when running on real iOS devices: bugs.webrtc.org/6889.
 TEST_F(AudioDeviceTest, DISABLED_StartStopPlayout) {
   StartPlayout();
   StopPlayout();
   StartPlayout();
   StopPlayout();
 }
 
 // Tests that recording can be initiated, started and stopped. No audio callback
 // is registered in this test.
-// Disabled due to failure: bugs.webrtc.org/7056
-TEST_F(AudioDeviceTest, DISABLED_StartStopRecording) {
+TEST_F(AudioDeviceTest, StartStopRecording) {
   StartRecording();
   StopRecording();
   StartRecording();
   StopRecording();
 }
 
 // Verify that calling StopPlayout() will leave us in an uninitialized state
 // which will require a new call to InitPlayout(). This test does not call
 // StartPlayout() while being uninitialized since doing so will hit a
 // RTC_DCHECK.
@@ skipping 114 matching lines @@
   StartPlayout();
   StartRecording();
   test_is_done_->Wait(kTestTimeOutInMilliseconds);
   StopRecording();
   StopPlayout();
 }
 
 // Start playout and read audio from an external PCM file when the audio layer
 // asks for data to play out. Real audio is played out in this test but it does
 // not contain any explicit verification that the audio quality is perfect.
-// Disabled due to failure: bugs.webrtc.org/7056
-TEST_F(AudioDeviceTest, DISABLED_RunPlayoutWithFileAsSource) {
+TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
   // TODO(henrika): extend test when mono output is supported.
   EXPECT_EQ(1, playout_channels());
   NiceMock<MockAudioTransportIOS> mock(kPlayout);
   const int num_callbacks = kFilePlayTimeInSec * kNumCallbacksPerSecond;
   std::string file_name = GetFileName(playout_sample_rate());
   std::unique_ptr<FileAudioStream> file_audio_stream(
       new FileAudioStream(num_callbacks, file_name, playout_sample_rate()));
   mock.HandleCallbacks(test_is_done_.get(), file_audio_stream.get(),
                        num_callbacks);
   // SetMaxPlayoutVolume();
@@ skipping 69 matching lines @@
   StopPlayout();
   StopRecording();
   // Verify that the correct number of transmitted impulses are detected.
   EXPECT_EQ(latency_audio_stream->num_latency_values(),
             static_cast<size_t>(
                 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
   latency_audio_stream->PrintResults();
 }
 
 }  // namespace webrtc