Remove iOS ifdefs in src/content.

content/renderer/media/media_stream_audio_processor_options.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 "content/renderer/media/media_stream_audio_processor_options.h"
 
 #include <stddef.h>
 #include <utility>
 
 #include "base/files/file_path.h"
@@ skipping 37 matching lines @@
 namespace {
 
 // Constant constraint keys which enables default audio constraints on
 // mediastreams with audio.
 struct {
   const char* key;
   bool value;
 } const kDefaultAudioConstraints[] = {
   { MediaAudioConstraints::kEchoCancellation, true },
   { MediaAudioConstraints::kGoogEchoCancellation, true },
-#if defined(OS_ANDROID) || defined(OS_IOS)
+#if defined(OS_ANDROID)
   { MediaAudioConstraints::kGoogExperimentalEchoCancellation, false },
 #else
   // Enable the extended filter mode AEC on all non-mobile platforms.
   { MediaAudioConstraints::kGoogExperimentalEchoCancellation, true },
 #endif
   { MediaAudioConstraints::kGoogAutoGainControl, true },
   { MediaAudioConstraints::kGoogExperimentalAutoGainControl, true },
   { MediaAudioConstraints::kGoogNoiseSuppression, true },
   { MediaAudioConstraints::kGoogHighpassFilter, true },
   { MediaAudioConstraints::kGoogTypingNoiseDetection, true },
@@ skipping 72 matching lines @@
     *field = rtc::Optional<bool>(value);
   }
 }
 
 }  // namespace
 
 // TODO(xians): Remove this method after the APM in WebRtc is deprecated.
 void MediaAudioConstraints::ApplyFixedAudioConstraints(
     cricket::AudioOptions* options) {
   SetIfNotSet(&options->echo_cancellation, true);
-#if defined(OS_ANDROID) || defined(OS_IOS)
+#if defined(OS_ANDROID)
   SetIfNotSet(&options->extended_filter_aec, false);
 #else
   // Enable the extended filter mode AEC on all non-mobile platforms.
   SetIfNotSet(&options->extended_filter_aec, true);
 #endif
   SetIfNotSet(&options->auto_gain_control, true);
   SetIfNotSet(&options->experimental_agc, true);
   SetIfNotSet(&options->noise_suppression, true);
   SetIfNotSet(&options->highpass_filter, true);
   SetIfNotSet(&options->typing_detection, true);
@@ skipping 180 matching lines @@
     // Map |fraction_poor_delays| to an Echo Cancellation quality and log in UMA
     // histogram. See DelayBasedEchoQuality for information on histogram
     // buckets.
     UMA_HISTOGRAM_ENUMERATION("WebRTC.AecDelayBasedQuality",
                               EchoDelayFrequencyToQuality(fraction_poor_delays),
                               DELAY_BASED_ECHO_QUALITY_MAX);
   }
 }
 
 void EnableEchoCancellation(AudioProcessing* audio_processing) {
-#if defined(OS_ANDROID) || defined(OS_IOS)
+#if defined(OS_ANDROID)
   // Mobile devices are using AECM.
   CHECK_EQ(0, audio_processing->echo_control_mobile()->set_routing_mode(
                   webrtc::EchoControlMobile::kSpeakerphone));
   CHECK_EQ(0, audio_processing->echo_control_mobile()->Enable(true));
   return;
 #endif
   int err = audio_processing->echo_cancellation()->set_suppression_level(
       webrtc::EchoCancellation::kHighSuppression);
 
   // Enable the metrics for AEC.
@@ skipping 38 matching lines @@
   if (audio_processing->StartDebugRecording(stream))
     DLOG(ERROR) << "Fail to start AEC debug recording";
 }
 
 void StopEchoCancellationDump(AudioProcessing* audio_processing) {
   if (audio_processing->StopDebugRecording())
     DLOG(ERROR) << "Fail to stop AEC debug recording";
 }
 
 void EnableAutomaticGainControl(AudioProcessing* audio_processing) {
-#if defined(OS_ANDROID) || defined(OS_IOS)
+#if defined(OS_ANDROID)
   const webrtc::GainControl::Mode mode = webrtc::GainControl::kFixedDigital;
 #else
   const webrtc::GainControl::Mode mode = webrtc::GainControl::kAdaptiveAnalog;
 #endif
   int err = audio_processing->gain_control()->set_mode(mode);
   err |= audio_processing->gain_control()->Enable(true);
   CHECK_EQ(err, 0);
 }
 
 void GetAecStats(webrtc::EchoCancellation* echo_cancellation,
@@ skipping 45 matching lines @@
 
   // Give preference to the audio constraint over the device-supplied mic
   // positions. This is mainly for testing purposes.
   return WebrtcPointsFromMediaPoints(
       constraints_geometry.empty()
           ? input_params.mic_positions
           : media::ParsePointsFromString(constraints_geometry));
 }
 
 }  // namespace content