Adding audio RepetitionDetector in AudioProcessingModule.

webrtc/modules/audio_processing/repetition_detector.cc

+/*
+ *  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/repetition_detector.h"
+
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+static const RepetitionDetector::Pattern kRepetitionPatterns[] = {
+    // {id_, look_back_, length_}
+    {0, 10, 10},
+    {1, 100, 10}
+};
+static const size_t kMaxFrames = 480;  // 10ms * 48kHz
+}
+
+RepetitionDetector::State::State(int id, int look_back_ms, int min_length_ms)
+    : id_(id),
+      look_back_ms_(look_back_ms),
+      min_length_ms_(min_length_ms) {
+  Reset();
+}
+
+void RepetitionDetector::State::Increment(bool zero) {
+  if (0 == count_frames_ && zero) {
+    all_zero_ = true;
+  }
+  ++count_frames_;
+  if (!zero) {
+    all_zero_ = false;
+  }
+}
+
+bool RepetitionDetector::State::HasValidReport(int sample_rate_hz) const {
+  return (!all_zero_ && count_frames_ >=
+      rtc::checked_cast<size_t>(min_length_ms_ * sample_rate_hz / 1000));
+}
+
+void RepetitionDetector::State::Reset() {
+  count_frames_ = 0;
+  all_zero_ = true;
+  reported_ = false;
+}
+
+RepetitionDetector::RepetitionDetector()
+    : max_look_back_ms_(0),
+      sample_rate_hz_(0),
+      buffer_size_frames_(0),
+      buffer_end_index_(0),
+      max_frames_(kMaxFrames) {
+  RegisterRepetitionPatterns(kRepetitionPatterns,
+                             sizeof(kRepetitionPatterns) / sizeof (Pattern));
+}
+
+void RepetitionDetector::RegisterRepetitionPatterns(const Pattern* patterns,
+                                                    size_t num_patterns) {
+  Pattern pattern;
+  for (size_t idx = 0; idx < num_patterns; idx++) {
+    pattern = patterns[idx];
+    states_.push_back(new State(pattern.id_, pattern.look_back_ms_,
+                                pattern.min_length_ms_));
+    if (pattern.look_back_ms_ > max_look_back_ms_) {
+      max_look_back_ms_ = pattern.look_back_ms_;
+    }
+  }
+}
+
+void RepetitionDetector::Reset(size_t num_channels, int sample_rate_hz) {
+  num_channels_ = num_channels;
+  sample_rate_hz_ = sample_rate_hz;
+  int sample_1k = max_look_back_ms_ * sample_rate_hz_;
+  buffer_size_frames_ = (sample_1k + 999) / 1000 + max_frames_;

[hlundin-webrtc, 2015/09/15 09:26:38]

Now just comment that what you implemented is in fact rounding up. Courtesy to
future you. :)

[minyue-webrtc, 2015/09/17 13:45:22]

Done.

+  audio_buffer_.resize(buffer_size_frames_ * num_channels_);
+  for (auto state : states_) {
+    state->Reset();
+  }
+}
+
+void RepetitionDetector::AddFramesToBuffer(const float* data,
+                                           size_t num_frames) {
+  DCHECK_LE(num_frames, buffer_size_frames_);
+  size_t margin = buffer_size_frames_ - buffer_end_index_;

[hlundin-webrtc, 2015/09/15 09:26:38]

const

[minyue-webrtc, 2015/09/17 13:45:22]

Done.

+  const auto it = audio_buffer_.begin() + buffer_end_index_ * num_channels_;
+  if (num_frames <= margin) {
+    std::copy(data, data + num_frames * num_channels_, it);
+    buffer_end_index_ += num_frames;
+  } else {
+    std::copy(data, data + margin * num_channels_, it);
+    std::copy(data + margin * num_channels_, data + num_frames * num_channels_,
+              audio_buffer_.begin());
+    buffer_end_index_ = num_frames - margin;
+  }
+}
+
+bool RepetitionDetector::Equal(const float* frame,
+                               int look_back_frames) const {
+  const size_t look_back_index =
+      (buffer_end_index_ + buffer_size_frames_ - look_back_frames) %
+      buffer_size_frames_;
+  auto it = audio_buffer_.begin() + look_back_index * num_channels_;
+  for (size_t cdx = 0; cdx < num_channels_; ++cdx, ++frame, ++it) {
+    if (*frame != *it) {
+      return false;
+    }
+  }
+  return true;
+}
+
+bool RepetitionDetector::IsZero(const float* frame) const {
+  for (size_t cdx = 0; cdx < num_channels_; ++cdx, ++frame) {
+    if (*frame != 0) {
+      return false;
+    }
+  }
+  return true;
+}
+
+void RepetitionDetector::Detect(const float* data, size_t num_frames,
+                                size_t num_channels, int sample_rate_hz) {
+  DCHECK_GT(states_.size(), 0ul);
+  if (num_channels != num_channels_ || sample_rate_hz != sample_rate_hz_) {
+    Reset(num_channels, sample_rate_hz);
+  }
+
+  while (num_frames > max_frames_) {
+    Detect(data, max_frames_, num_channels, sample_rate_hz);
+    data += max_frames_ * num_channels;
+    num_frames -= max_frames_;
+  }
+
+  if (num_frames == 0)
+    return;
+
+  AddFramesToBuffer(data, num_frames);
+
+  for (size_t idx = num_frames; idx > 0; --idx, data += num_channels) {
+    for (auto state : states_) {
+      const size_t look_back_frames =
+          rtc::CheckedDivExact(state->look_back_ms() * sample_rate_hz_, 1000);
+      // Equal(data, offset) checks if |data| equals the audio frame located
+      // |offset| frames from the end of buffer. Now a full frame has been
+      // inserted to the buffer, and thus |offset| should compensate for it.
+      if (Equal(data, look_back_frames + idx)) {
+        if (!state->reported()) {
+          state->Increment(IsZero(data));
+          if (state->HasValidReport(sample_rate_hz)) {
+            ReportRepetition(state->id());
+            state->set_reported(true);
+          }
+        }
+      } else {
+        state->Reset();
+      }
+    }
+  }
+}
+
+}  // namespace webrtc