Add detection for repeated audio in capturing.

content/renderer/media/audio_repetition_detector.cc

+// Copyright 2015 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "content/renderer/media/audio_repetition_detector.h"
+
+#include "base/logging.h"
+#include "base/macros.h"
+
+namespace content {
+
+AudioRepetitionDetector::AudioRepetitionDetector(int min_length_ms,
+                                                 size_t max_frames,
+                                                 const int* look_back_times,
+                                                 size_t num_look_back)
+    : max_look_back_ms_(0),
+      min_length_ms_(min_length_ms),
+      sample_rate_(0),
+      buffer_size_frames_(0),
+      buffer_end_index_(0),
+      max_frames_(max_frames) {
+  DCHECK(main_thread_checker_.CalledOnValidThread());
+  processing_thread_checker_.DetachFromThread();
+  for (size_t i = 0; i < num_look_back; ++i) {
+    RegisterLookbackTime(look_back_times[i]);
+  }
+}
+
+AudioRepetitionDetector::~AudioRepetitionDetector() {
+  DCHECK(main_thread_checker_.CalledOnValidThread());
+}
+
+void AudioRepetitionDetector::Detect(const float* data, size_t num_frames,
+                                     size_t num_channels, int sample_rate) {
+  DCHECK(processing_thread_checker_.CalledOnValidThread());
+  DCHECK(!states_.empty());
+
+  if (num_channels != num_channels_ || sample_rate != sample_rate_)
+    Reset(num_channels, sample_rate);
+
+  // The maximum number of frames |audio_buffer_| can take in is |max_frames_|.
+  // Therefore, input data with larger frames needs be divided into chunks.
+  const size_t chunk_size = max_frames_ * num_channels;
+  while (num_frames > max_frames_) {
+    Detect(data, max_frames_, num_channels, sample_rate);
+    data += chunk_size;
+    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 (State* state : states_) {
+      // Look back position depends on the sample rate. It is rounded down to
+      // the closest integer.
+      const size_t look_back_frames =
+          state->look_back_ms() * sample_rate_ / 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, num_channels));
+          if (HasValidReport(state)) {
+            ReportRepetition(state->look_back_ms());
+            state->set_reported(true);
+          }
+        }
+      } else {
+        state->Reset();
+      }
+    }
+  }
+}
+
+AudioRepetitionDetector::State::State(int look_back_ms)
+    : look_back_ms_(look_back_ms) {
+  Reset();
+}
+
+void AudioRepetitionDetector::State::Increment(bool zero) {
+  if (0 == count_frames_ && zero) {
+    // If a repetition starts with zeros, we enter the all zero mode until
+    // a non zero is found later. The point is that the beginning zeros should
+    // be counted in the length of the repetition as long as the repetition does
+    // not comprise only zeros.
+    all_zero_ = true;
+  }
+  ++count_frames_;
+  if (!zero)
+    all_zero_ = false;
+}
+
+void AudioRepetitionDetector::State::Reset() {
+  count_frames_ = 0;
+  all_zero_ = true;
+  reported_ = false;
+}
+
+void AudioRepetitionDetector::RegisterLookbackTime(int look_back_ms) {
+  DCHECK(main_thread_checker_.CalledOnValidThread());

[Henrik Grunell, 2015/10/20 08:50:16]

|states_| is accessed to different threads.

[minyue, 2015/10/23 12:05:23]

That is true. But I have no better idea than limiting main_thread to call ctor
and dtor only. Do you have better suggestion?

[Henrik Grunell, 2015/10/23 13:49:02]

You have a few options: call this function on the processing thread, inject the
lookback times in the ctor, use a lock.

[minyue, 2015/10/26 10:41:46]

This is already a private member that is called from ctor, so it is actually
"inject the lookback times in the ctor". Do you mean we even need to merge this
helper function into ctor?

[Henrik Grunell, 2015/10/28 14:09:23]

Ah, I see. Then it's fine. Actually, since the ctor only does this and this
function may only be called from the ctor, move it to the ctor.

+
+  // States are added in the order of their look back times.
+  auto it = states_.begin();
+  for (; it != states_.end(); ++it) {
+    const int it_look_back = (*it)->look_back_ms();
+    if (it_look_back == look_back_ms)
+      return;
+    if (it_look_back < look_back_ms)
+      break;
+  }
+  states_.insert(it, new State(look_back_ms));
+  if (look_back_ms > max_look_back_ms_) {
+    max_look_back_ms_ = look_back_ms;
+  }
+}
+
+void AudioRepetitionDetector::Reset(size_t num_channels, int sample_rate) {
+  DCHECK(processing_thread_checker_.CalledOnValidThread());
+  num_channels_ = num_channels;
+  sample_rate_ = sample_rate;
+
+  // |(xxx + 999) / 1000| is an arithmetic way to round up |xxx / 1000|.
+  buffer_size_frames_ =
+      (max_look_back_ms_ * sample_rate_ + 999) / 1000 + max_frames_;
+
+  audio_buffer_.resize(buffer_size_frames_ * num_channels_);
+  for (State* state : states_)
+    state->Reset();
+}
+
+void AudioRepetitionDetector::AddFramesToBuffer(const float* data,
+                                                size_t num_frames) {
+  DCHECK(processing_thread_checker_.CalledOnValidThread());
+  DCHECK_LE(num_frames, buffer_size_frames_);
+  const size_t margin = buffer_size_frames_ - buffer_end_index_;
+  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 AudioRepetitionDetector::Equal(const float* frame,
+                                    int look_back_frames) const {
+  DCHECK(processing_thread_checker_.CalledOnValidThread());
+  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 channel = 0; channel < num_channels_; ++channel, ++frame, ++it) {
+    if (*frame != *it)
+      return false;
+  }
+  return true;
+}
+
+bool AudioRepetitionDetector::IsZero(const float* frame,
+                                     size_t num_channels) const {
+  for (size_t channel = 0; channel < num_channels; ++channel, ++frame) {
+    if (*frame != 0)
+      return false;
+  }
+  return true;
+}
+
+bool AudioRepetitionDetector::HasValidReport(const State* state) const {
+  return (!state->all_zero() && state->count_frames() >=
+      static_cast<size_t>(min_length_ms_ * sample_rate_ / 1000));
+}
+
+}  // namespace content