Repo created

TMessagesProj/jni/voip/webrtc/modules/audio_processing/agc/agc.cc

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+/*
+ *  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 "modules/audio_processing/agc/agc.h"
+
+#include <cmath>
+#include <cstdlib>
+#include <vector>
+
+#include "modules/audio_processing/agc/loudness_histogram.h"
+#include "modules/audio_processing/agc/utility.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kDefaultLevelDbfs = -18;
+constexpr int kNumAnalysisFrames = 100;
+constexpr double kActivityThreshold = 0.3;
+constexpr int kNum10msFramesInOneSecond = 100;
+constexpr int kMaxSampleRateHz = 384000;
+
+}  // namespace
+
+Agc::Agc()
+    : target_level_loudness_(Dbfs2Loudness(kDefaultLevelDbfs)),
+      target_level_dbfs_(kDefaultLevelDbfs),
+      histogram_(LoudnessHistogram::Create(kNumAnalysisFrames)),
+      inactive_histogram_(LoudnessHistogram::Create()) {}
+
+Agc::~Agc() = default;
+
+void Agc::Process(rtc::ArrayView<const int16_t> audio) {
+  const int sample_rate_hz = audio.size() * kNum10msFramesInOneSecond;
+  RTC_DCHECK_LE(sample_rate_hz, kMaxSampleRateHz);
+  vad_.ProcessChunk(audio.data(), audio.size(), 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]);
+  }
+}
+
+bool Agc::GetRmsErrorDb(int* error) {
+  if (!error) {
+    RTC_DCHECK_NOTREACHED();
+    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;
+}
+
+int Agc::target_level_dbfs() const {
+  return target_level_dbfs_;
+}
+
+float Agc::voice_probability() const {
+  return vad_.last_voice_probability();
+}
+
+}  // namespace webrtc