直接贴代码，说明都写在注释中了。
从下面代码中可以看出iOS平台的VPIO自己本身已经支持AEC、AGC和NS所以不使用WebRTC的软件算法。
在iOS平台可以通过ios_force_software_aec_HACK 强制开启软件回声消除:echo_cancellation/extended_filter_aec，
但是AGC和NS目前没有选项可以设置。

如果安卓平台内置了AEC、AGC和NS也是不使用WebRTC软件算法而使用平台内置算法。

src/media/engine/webrtcvoiceengine.cc

bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::ApplyOptions: "
                   << options_in.ToString();
  AudioOptions options = options_in;  // The options are modified below.

  // Set and adjust echo canceller options.
  // kEcConference is AEC with high suppression.
  webrtc::EcModes ec_mode = webrtc::kEcConference;

#if defined(WEBRTC_IOS)
  // wbt: 在ios上强制软件回声消除
  if (options.ios_force_software_aec_HACK &&
      *options.ios_force_software_aec_HACK) {
    // EC may be forced on for a device known to have non-functioning platform
    // AEC.
    options.echo_cancellation = true;
    options.extended_filter_aec = true;
    RTC_LOG(LS_WARNING)
        << "Force software AEC on iOS. May conflict with platform AEC.";
  } else {// wbt: 默认使用VPIO内置的回声消除
    // On iOS, VPIO provides built-in EC.
    options.echo_cancellation = false;
    options.extended_filter_aec = false;
    RTC_LOG(LS_INFO) << "Always disable AEC on iOS. Use built-in instead.";
  }
#elif defined(WEBRTC_ANDROID)
  ec_mode = webrtc::kEcAecm;
  options.extended_filter_aec = false;
#endif

  // wbt :除iOS平台外：如果启用了估计延时不确定性Delay Agnostic设置则自动开启aec
  // Delay Agnostic AEC automatically turns on EC if not set except on iOS
  // where the feature is not supported.
  bool use_delay_agnostic_aec = false;
#if !defined(WEBRTC_IOS)
  if (options.delay_agnostic_aec) {
    use_delay_agnostic_aec = *options.delay_agnostic_aec;
    if (use_delay_agnostic_aec) {
      options.echo_cancellation = true;
      options.extended_filter_aec = true;
      ec_mode = webrtc::kEcConference;
    }
  }
#endif

// Set and adjust noise suppressor options.
#if defined(WEBRTC_IOS)
  // On iOS, VPIO provides built-in NS.
  // wbt: iOS平台使用VIPIO内置的降噪。
  // 关闭键盘声音检测（因为触摸设备没有键盘)，并且关闭实验性降噪。
  options.noise_suppression = false;
  options.typing_detection = false;
  options.experimental_ns = false;
  RTC_LOG(LS_INFO) << "Always disable NS on iOS. Use built-in instead.";
#elif defined(WEBRTC_ANDROID)
  // wbt: Android平台
  // 关闭键盘声音检测（因为触摸设备没有键盘)，并且关闭实验性降噪。
  options.typing_detection = false;
  options.experimental_ns = false;
#endif

// Set and adjust gain control options.
#if defined(WEBRTC_IOS)
  // On iOS, VPIO provides built-in AGC.
  // wbt: iOS平台使用VIPIO内置的AGC增益。并且关闭试验性AGC.
  options.auto_gain_control = false;
  options.experimental_agc = false;
  RTC_LOG(LS_INFO) << "Always disable AGC on iOS. Use built-in instead.";
#elif defined(WEBRTC_ANDROID)
  // wbt: Android平台关闭试验性AGC.
  options.experimental_agc = false;
#endif

#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
  // wbt: 手机平台
  // 如果设置了"WebRTC-Audio-MinimizeResamplingOnMobile"则关闭AGC。
  // 然后如果开启降噪且未开启回声消除则关闭高通滤波器。

  // Turn off the gain control if specified by the field trial.
  // The purpose of the field trial is to reduce the amount of resampling
  // performed inside the audio processing module on mobile platforms by
  // whenever possible turning off the fixed AGC mode and the high-pass filter.
  // (https://bugs.chromium.org/p/webrtc/issues/detail?id=6181).
  if (webrtc::field_trial::IsEnabled(
          "WebRTC-Audio-MinimizeResamplingOnMobile")) {
    options.auto_gain_control = false;
    RTC_LOG(LS_INFO) << "Disable AGC according to field trial.";
    if (!(options.noise_suppression.value_or(false) ||
          options.echo_cancellation.value_or(false))) {
      // If possible, turn off the high-pass filter.
      RTC_LOG(LS_INFO)
          << "Disable high-pass filter in response to field trial.";
      options.highpass_filter = false;
    }
  }
#endif

  if (options.echo_cancellation) {
    // Check if platform supports built-in EC. Currently only supported on
    // Android and in combination with Java based audio layer.
    // TODO(henrika): investigate possibility to support built-in EC also
    // in combination with Open SL ES audio.
    // wbt: 目前只有android支持内置aec
    // 如果支持内置aec而且开启回声消除echo_cancellation和未开启use_delay_agnostic_aec
    // 且内置EnableBuiltInAEC时关闭软件回声消除从而使用平台内置回声消除。
    const bool built_in_aec = adm()->BuiltInAECIsAvailable();//除了Android平台为true，其他平台默认为false
    if (built_in_aec) {
      // Built-in EC exists on this device and use_delay_agnostic_aec is not
      // overriding it. Enable/Disable it according to the echo_cancellation
      // audio option.
      const bool enable_built_in_aec =
          *options.echo_cancellation && !use_delay_agnostic_aec;
      if (adm()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&
          enable_built_in_aec) {
        // Disable internal software EC if built-in EC is enabled,
        // i.e., replace the software EC with the built-in EC.
        options.echo_cancellation = false;
        RTC_LOG(LS_INFO)
            << "Disabling EC since built-in EC will be used instead";
      }
    }
    // 这里的ec_mode在上面会根据情况设置为默认会议积极的aec还是手机型aec。
    webrtc::apm_helpers::SetEcStatus(apm(), *options.echo_cancellation,
                                     ec_mode);
  }
  // 判断是否支持平台内置agc,如果支持则关闭软件agc
  if (options.auto_gain_control) {
    bool built_in_agc_avaliable = adm()->BuiltInAGCIsAvailable();
    if (built_in_agc_avaliable) {
      if (adm()->EnableBuiltInAGC(*options.auto_gain_control) == 0 &&
          *options.auto_gain_control) {
        // Disable internal software AGC if built-in AGC is enabled,
        // i.e., replace the software AGC with the built-in AGC.
        options.auto_gain_control = false;
        RTC_LOG(LS_INFO)
            << "Disabling AGC since built-in AGC will be used instead";
      }
    }
    webrtc::apm_helpers::SetAgcStatus(apm(), *options.auto_gain_control);
  }

  //agc参数设置
  if (options.tx_agc_target_dbov || options.tx_agc_digital_compression_gain ||
      options.tx_agc_limiter) {
    // Override default_agc_config_. Generally, an unset option means "leave
    // the VoE bits alone" in this function, so we want whatever is set to be
    // stored as the new "default". If we didn't, then setting e.g.
    // tx_agc_target_dbov would reset digital compression gain and limiter
    // settings.
    default_agc_config_.targetLeveldBOv = options.tx_agc_target_dbov.value_or(
        default_agc_config_.targetLeveldBOv);
    default_agc_config_.digitalCompressionGaindB =
        options.tx_agc_digital_compression_gain.value_or(
            default_agc_config_.digitalCompressionGaindB);
    default_agc_config_.limiterEnable =
        options.tx_agc_limiter.value_or(default_agc_config_.limiterEnable);
    webrtc::apm_helpers::SetAgcConfig(apm(), default_agc_config_);
  }

  // 如果支持平台内置降噪则关闭软件降噪
  if (options.noise_suppression) {
    if (adm()->BuiltInNSIsAvailable()) {
      bool builtin_ns = *options.noise_suppression;
      if (adm()->EnableBuiltInNS(builtin_ns) == 0 && builtin_ns) {
        // Disable internal software NS if built-in NS is enabled,
        // i.e., replace the software NS with the built-in NS.
        options.noise_suppression = false;
        RTC_LOG(LS_INFO)
            << "Disabling NS since built-in NS will be used instead";
      }
    }
    webrtc::apm_helpers::SetNsStatus(apm(), *options.noise_suppression);
  }

  // 立体声通道交换
  if (options.stereo_swapping) {
    RTC_LOG(LS_INFO) << "Stereo swapping enabled? " << *options.stereo_swapping;
    audio_state()->SetStereoChannelSwapping(*options.stereo_swapping);
  }

  // jitter buffer最大包数设置
  if (options.audio_jitter_buffer_max_packets) {
    RTC_LOG(LS_INFO) << "NetEq capacity is "
                     << *options.audio_jitter_buffer_max_packets;
    audio_jitter_buffer_max_packets_ =
        std::max(20, *options.audio_jitter_buffer_max_packets);
  }
  // jitter buffer加速设置
  if (options.audio_jitter_buffer_fast_accelerate) {
    RTC_LOG(LS_INFO) << "NetEq fast mode? "
                     << *options.audio_jitter_buffer_fast_accelerate;
    audio_jitter_buffer_fast_accelerate_ =
        *options.audio_jitter_buffer_fast_accelerate;
  }
  // jitter buffer最小延迟（毫秒）
  if (options.audio_jitter_buffer_min_delay_ms) {
    RTC_LOG(LS_INFO) << "NetEq minimum delay is "
                     << *options.audio_jitter_buffer_min_delay_ms;
    audio_jitter_buffer_min_delay_ms_ =
        *options.audio_jitter_buffer_min_delay_ms;
  }

  // 键盘声音检测
  if (options.typing_detection) {
    RTC_LOG(LS_INFO) << "Typing detection is enabled? "
                     << *options.typing_detection;
    webrtc::apm_helpers::SetTypingDetectionStatus(apm(),
                                                  *options.typing_detection);
  }

  webrtc::Config config;

  if (options.delay_agnostic_aec)
    delay_agnostic_aec_ = options.delay_agnostic_aec;
  if (delay_agnostic_aec_) {
    RTC_LOG(LS_INFO) << "Delay agnostic aec is enabled? "
                     << *delay_agnostic_aec_;
    config.Set<webrtc::DelayAgnostic>(
        new webrtc::DelayAgnostic(*delay_agnostic_aec_));
  }

  if (options.extended_filter_aec) {
    extended_filter_aec_ = options.extended_filter_aec;
  }
  if (extended_filter_aec_) {
    RTC_LOG(LS_INFO) << "Extended filter aec is enabled? "
                     << *extended_filter_aec_;
    config.Set<webrtc::ExtendedFilter>(
        new webrtc::ExtendedFilter(*extended_filter_aec_));
  }

  if (options.experimental_ns) {
    experimental_ns_ = options.experimental_ns;
  }
  if (experimental_ns_) {
    RTC_LOG(LS_INFO) << "Experimental ns is enabled? " << *experimental_ns_;
    config.Set<webrtc::ExperimentalNs>(
        new webrtc::ExperimentalNs(*experimental_ns_));
  }

  webrtc::AudioProcessing::Config apm_config = apm()->GetConfig();

  if (options.highpass_filter) {
    apm_config.high_pass_filter.enabled = *options.highpass_filter;
  }

  // 残留回声检测
  if (options.residual_echo_detector) {
    apm_config.residual_echo_detector.enabled = *options.residual_echo_detector;
  }

  apm()->SetExtraOptions(config);
  apm()->ApplyConfig(apm_config);
  return true;
}

虽然没做android，但是顺便看了一下android上使用平台内置硬件AEC，NS的代码，代码在：
src/sdk/android/src/java/org/webrtc/audio/WebRtcAudioEffects.java
在java中使用AudioEffect检测是否支持硬件AEC和NS，使用AcousticEchoCanceler和NoiseSuppressor处理AEC和NS。
而AGC没有平台硬件支持，直接使用WebRTC中的算法。