AGC2-fixed-digital: Level Estimator

This CL adds the Level Estimator of the new gain controller. The Level
Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub
frames. We take the maximal sample values in every sub frame. We then
apply attack/decay smoothing. This is the final level estimate.

The results will be used with InterpolatedGainCurve (see this CL
https://webrtc-review.googlesource.com/c/src/+/51920). For every level
estimate value, we look up a gain with
InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to
the signal.

Bug: webrtc:7949
Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1
Reviewed-on: https://webrtc-review.googlesource.com/52381
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22054}

modules/audio_processing/include/audio_frame_view.h

@@ -27,6 +27,14 @@ class AudioFrameView {
         num_channels_(num_channels),
         channel_size_(channel_size) {}
 
+  // Implicit cast to allow converting Frame<float> to
+  // Frame<const float>.
+  template <class U>
+  AudioFrameView(AudioFrameView<U> other)
+      : audio_samples_(other.data()),
+        num_channels_(other.num_channels()),
+        channel_size_(other.samples_per_channel()) {}
+
   AudioFrameView() = delete;
 
   size_t num_channels() const { return num_channels_; }
@@ -45,6 +53,8 @@ class AudioFrameView {
     return rtc::ArrayView<const T>(audio_samples_[idx], channel_size_);
   }
 
+  T* const* data() { return audio_samples_; }
+
  private:
   T* const* audio_samples_;
   size_t num_channels_;