Remove iOS ifdefs in src/content.

content/renderer/media/media_stream_audio_processor_unittest.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include <stddef.h>
 #include <stdint.h>
 
 #include <vector>
 
 #include "base/files/file_path.h"
@@ skipping 104 matching lines @@
     const base::TimeDelta output_buffer_duration =
         expected_output_buffer_size * base::TimeDelta::FromSeconds(1) /
             expected_output_sample_rate;
     for (int i = 0; i < kNumberOfPacketsForTest; ++i) {
       data_bus->FromInterleaved(data_ptr, data_bus->frames(), 2);
       audio_processor->PushCaptureData(*data_bus, input_capture_delay);
 
       // |audio_processor| does nothing when the audio processing is off in
       // the processor.
       webrtc::AudioProcessing* ap = audio_processor->audio_processing_.get();
-#if defined(OS_ANDROID) || defined(OS_IOS)
+#if defined(OS_ANDROID)
       const bool is_aec_enabled = ap && ap->echo_control_mobile()->is_enabled();
       // AEC should be turned off for mobiles.
       DCHECK(!ap || !ap->echo_cancellation()->is_enabled());
 #else
       const bool is_aec_enabled = ap && ap->echo_cancellation()->is_enabled();
 #endif
       if (is_aec_enabled) {
         if (params.channels() > kMaxNumberOfPlayoutDataChannels) {
           for (int i = 0; i < kMaxNumberOfPlayoutDataChannels; ++i) {
             data_bus_playout->SetChannelData(
@@ skipping 26 matching lines @@
   }
 
   void VerifyDefaultComponents(MediaStreamAudioProcessor* audio_processor) {
     webrtc::AudioProcessing* audio_processing =
         audio_processor->audio_processing_.get();
 #if defined(OS_ANDROID)
     EXPECT_TRUE(audio_processing->echo_control_mobile()->is_enabled());
     EXPECT_TRUE(audio_processing->echo_control_mobile()->routing_mode() ==
         webrtc::EchoControlMobile::kSpeakerphone);
     EXPECT_FALSE(audio_processing->echo_cancellation()->is_enabled());
-#elif !defined(OS_IOS)
+#else
     EXPECT_TRUE(audio_processing->echo_cancellation()->is_enabled());
     EXPECT_TRUE(audio_processing->echo_cancellation()->suppression_level() ==
         webrtc::EchoCancellation::kHighSuppression);
     EXPECT_TRUE(audio_processing->echo_cancellation()->are_metrics_enabled());
     EXPECT_TRUE(
         audio_processing->echo_cancellation()->is_delay_logging_enabled());
 #endif
 
     EXPECT_TRUE(audio_processing->noise_suppression()->is_enabled());
     EXPECT_TRUE(audio_processing->noise_suppression()->level() ==
         webrtc::NoiseSuppression::kHigh);
     EXPECT_TRUE(audio_processing->high_pass_filter()->is_enabled());
     EXPECT_TRUE(audio_processing->gain_control()->is_enabled());
-#if defined(OS_ANDROID) || defined(OS_IOS)
+#if defined(OS_ANDROID)
     EXPECT_TRUE(audio_processing->gain_control()->mode() ==
         webrtc::GainControl::kFixedDigital);
     EXPECT_FALSE(audio_processing->voice_detection()->is_enabled());
 #else
     EXPECT_TRUE(audio_processing->gain_control()->mode() ==
         webrtc::GainControl::kAdaptiveAnalog);
     EXPECT_TRUE(audio_processing->voice_detection()->is_enabled());
     EXPECT_TRUE(audio_processing->voice_detection()->likelihood() ==
         webrtc::VoiceDetection::kVeryLowLikelihood);
 #endif
@@ skipping 353 matching lines @@
   ProcessDataAndVerifyFormat(audio_processor.get(),
                              kAudioProcessingSampleRate,
                              kAudioProcessingNumberOfChannel,
                              kAudioProcessingSampleRate / 100);
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
   // |audio_processor|.
   audio_processor = NULL;
 }
 
 }  // namespace content