Pull the Voice Activity Detector out from the AGC

webrtc/modules/audio_processing/vad/voice_activity_detector.cc

Index: webrtc/modules/audio_processing/vad/voice_activity_detector.cc
diff --git a/webrtc/modules/audio_processing/vad/voice_activity_detector.cc b/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
new file mode 100644
index 0000000000000000000000000000000000000000..b39ed66b897eb84bfdb4ce8a61281b554729a99d
--- /dev/null
+++ b/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2015 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/vad/voice_activity_detector.h"
+
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+namespace {
+
+const int kMaxLength = 320;
+const int kNumChannels = 1;
+
+const double kDefaultVoiceValue = 1.0;
+const double kNeutralProbability = 0.5;
+const double kLowProbability = 0.01;
+
+}  // namespace
+
+VoiceActivityDetector::VoiceActivityDetector()
+    : last_voice_probability_(kDefaultVoiceValue),
+      // Initialize to the most common resampling situation.
+      resampler_(kMaxLength, kLength10Ms, kNumChannels),
+      standalone_vad_(StandaloneVad::Create()) {
+}
+
+// Because ISAC has a different chunk length, it updates
+// |chunkwise_voice_probabilities_| and |chunkwise_rms_| when there is new data.
+// Otherwise it clears them.
+void VoiceActivityDetector::ProcessChunk(const int16_t* audio,
+                                         int length,
+                                         int sample_rate_hz) {
+  DCHECK_EQ(length, sample_rate_hz / 100);
+  DCHECK_LE(length, kMaxLength);
+  // Resample to the required rate.
+  const int16_t* resampled_ptr = audio;
+  if (sample_rate_hz != kSampleRateHz) {
+    CHECK_EQ(
+        resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels),
+        0);
+    resampler_.Push(audio, length, resampled_, kLength10Ms, length);
+    resampled_ptr = resampled_;
+  }
+  DCHECK_EQ(length, kLength10Ms);
+
+  // Each chunk needs to be passed into |standalone_vad_|, because internally it
+  // buffers the audio and processes it all at once when GetActivity() is
+  // called.
+  CHECK_EQ(standalone_vad_->AddAudio(resampled_ptr, length), 0);

[aluebs-webrtc, 2015/06/29 23:34:32]

The bug that ASAN caught was that I was passing in |audio| instead of
|resampled_ptr|. I guess the test cases for which I tested bit exactness didn't
exercise the resampler at all.

[Andrew MacDonald, 2015/06/29 23:38:23]

Thanks goodness for ASAN.

[aluebs-webrtc, 2015/06/29 23:40:56]

Certainly :)

+
+  audio_processing_.ExtractFeatures(resampled_ptr, length, &features_);
+
+  chunkwise_voice_probabilities_.resize(features_.num_frames);
+  chunkwise_rms_.resize(features_.num_frames);
+  std::copy(features_.rms, features_.rms + chunkwise_rms_.size(),
+            chunkwise_rms_.begin());
+  if (features_.num_frames > 0) {
+    if (features_.silence) {
+      // The other features are invalid, so set the voice probabilities to an
+      // arbitrary low value.
+      std::fill(chunkwise_voice_probabilities_.begin(),
+                chunkwise_voice_probabilities_.end(), kLowProbability);
+    } else {
+      std::fill(chunkwise_voice_probabilities_.begin(),
+                chunkwise_voice_probabilities_.end(), kNeutralProbability);
+      CHECK_GE(
+          standalone_vad_->GetActivity(&chunkwise_voice_probabilities_[0],
+                                       chunkwise_voice_probabilities_.size()),
+          0);
+      CHECK_GE(pitch_based_vad_.VoicingProbability(
+                   features_, &chunkwise_voice_probabilities_[0]),
+               0);
+    }
+    last_voice_probability_ = chunkwise_voice_probabilities_.back();
+  }
+}
+
+}  // namespace webrtc