Update audio_processing module

Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1

Update notes:

 * Pull in third party license file

 * Replace .gypi files with BUILD.gn to keep track of what changes
   upstream

 * Bunch of new filse pulled in as dependencies

 * Won't build yet due to changes needed on top of these

webrtc/modules/audio_processing/agc/agc.cc

@@ -0,0 +1,101 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+#include "webrtc/modules/audio_processing/agc/agc.h"
+
+#include <cmath>
+#include <cstdlib>
+
+#include <algorithm>
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_processing/agc/histogram.h"
+#include "webrtc/modules/audio_processing/agc/utility.h"
+#include "webrtc/modules/interface/module_common_types.h"
+
+namespace webrtc {
+namespace {
+
+const int kDefaultLevelDbfs = -18;
+const int kNumAnalysisFrames = 100;
+const double kActivityThreshold = 0.3;
+
+}  // namespace
+
+Agc::Agc()
+    : target_level_loudness_(Dbfs2Loudness(kDefaultLevelDbfs)),
+      target_level_dbfs_(kDefaultLevelDbfs),
+      histogram_(Histogram::Create(kNumAnalysisFrames)),
+      inactive_histogram_(Histogram::Create()) {
+  }
+
+Agc::~Agc() {}
+
+float Agc::AnalyzePreproc(const int16_t* audio, size_t length) {
+  assert(length > 0);
+  size_t num_clipped = 0;
+  for (size_t i = 0; i < length; ++i) {
+    if (audio[i] == 32767 || audio[i] == -32768)
+      ++num_clipped;
+  }
+  return 1.0f * num_clipped / length;
+}
+
+int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
+  vad_.ProcessChunk(audio, length, sample_rate_hz);
+  const std::vector<double>& rms = vad_.chunkwise_rms();
+  const std::vector<double>& probabilities =
+      vad_.chunkwise_voice_probabilities();
+  RTC_DCHECK_EQ(rms.size(), probabilities.size());
+  for (size_t i = 0; i < rms.size(); ++i) {
+    histogram_->Update(rms[i], probabilities[i]);
+  }
+  return 0;
+}
+
+bool Agc::GetRmsErrorDb(int* error) {
+  if (!error) {
+    assert(false);
+    return false;
+  }
+
+  if (histogram_->num_updates() < kNumAnalysisFrames) {
+    // We haven't yet received enough frames.
+    return false;
+  }
+
+  if (histogram_->AudioContent() < kNumAnalysisFrames * kActivityThreshold) {
+    // We are likely in an inactive segment.
+    return false;
+  }
+
+  double loudness = Linear2Loudness(histogram_->CurrentRms());
+  *error = std::floor(Loudness2Db(target_level_loudness_ - loudness) + 0.5);
+  histogram_->Reset();
+  return true;
+}
+
+void Agc::Reset() {
+  histogram_->Reset();
+}
+
+int Agc::set_target_level_dbfs(int level) {
+  // TODO(turajs): just some arbitrary sanity check. We can come up with better
+  // limits. The upper limit should be chosen such that the risk of clipping is
+  // low. The lower limit should not result in a too quiet signal.
+  if (level >= 0 || level <= -100)
+    return -1;
+  target_level_dbfs_ = level;
+  target_level_loudness_ = Dbfs2Loudness(level);
+  return 0;
+}
+
+}  // namespace webrtc