Add empty residual echo detector.

webrtc/modules/audio_processing/residual_echo_detector.cc

+/*

[hlundin-webrtc, 2016/10/13 11:45:20]

The order of the methods in this file does not match the .h file.

[ivoc, 2016/10/13 13:46:15]

Fixed now.

+ *  Copyright (c) 2016 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/residual_echo_detector.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/echo_detector/echo_detector.h"
+
+namespace webrtc {
+
+ResidualEchoDetector::ResidualEchoDetector(rtc::CriticalSection* crit_render,
+                                           rtc::CriticalSection* crit_capture)
+    : crit_render_(crit_render), crit_capture_(crit_capture) {
+  RTC_DCHECK(crit_render);
+  RTC_DCHECK(crit_capture);
+}
+
+ResidualEchoDetector::~ResidualEchoDetector() {}
+
+void ResidualEchoDetector::AnalyzeRenderAudio(const AudioBuffer* audio) {
+  rtc::CritScope cs_render(crit_render_);
+
+  RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+
+  render_queue_buffer_.clear();
+
+  // Buffer the samples in the render queue.
+  render_queue_buffer_.insert(render_queue_buffer_.end(),
+                              audio->split_bands_const_f(0)[kBand0To8kHz],
+                              (audio->split_bands_const_f(0)[kBand0To8kHz] +
+                               audio->num_frames_per_band()));
+
+  // Insert the samples into the queue.
+  if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
+    // The data queue is full and needs to be emptied.
+    ReadQueuedRenderData();
+
+    // Retry the insert (should always work).
+    RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);

[hlundin-webrtc, 2016/10/13 11:45:20]

The argument of DCHECK is only evaluated in debug mode. You must change this to
store the result and then DCHECK on that. 

Also, no need to DCHECK_EQ(x, true) -- simply DCHECK(x) suffices.

[ivoc, 2016/10/13 13:46:16]

Good point, since I copied this from the AEC, it should probably be fixed there
as well (I can make a follow-up CL).

[hlundin-webrtc, 2016/10/14 06:59:18]

Oh. Yes, please do!

[ivoc, 2016/10/14 09:53:18]

Seems like Per beat me to it :)

+  }
+}
+
+float ResidualEchoDetector::get_echo_likelihood() {
+  return 0.0f;
+}
+
+// Read chunks of data that were received and queued on the render side from
+// a queue. All the data chunks are buffered into the farend signal of the AEC.
+void ResidualEchoDetector::ReadQueuedRenderData() {
+  rtc::CritScope cs_capture(crit_capture_);
+
+  while (render_signal_queue_->Remove(&capture_queue_buffer_)) {

[hlundin-webrtc, 2016/10/13 11:45:20]

Is capture_queue_buffer_ only a container that conveys the audio from
render_signal_queue_ to detector_? If so, does it have to be a class member?
Perhaps performance considerations says it does.

[ivoc, 2016/10/13 13:46:16]

I think this is indeed done for performance reasons (copied from AEC again).

[hlundin-webrtc, 2016/10/14 06:59:18]

Acknowledged.

+    const size_t num_frames_per_band = capture_queue_buffer_.size();
+    detector_->BufferFarend(capture_queue_buffer_.data(), num_frames_per_band);
+  }
+}
+
+void ResidualEchoDetector::AnalyzeCaptureAudio(AudioBuffer* audio) {
+  rtc::CritScope cs_capture(crit_capture_);
+
+  RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+
+  detector_->Process(audio->split_bands_const_f(0)[kBand0To8kHz],
+                     audio->num_frames_per_band());
+}
+
+void ResidualEchoDetector::Initialize(int sample_rate_hz) {
+  rtc::CritScope cs_render(crit_render_);
+  rtc::CritScope cs_capture(crit_capture_);
+
+  RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
+             sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
+             sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
+             sample_rate_hz == AudioProcessing::kSampleRate48kHz);
+
+  detector_->Initialize(sample_rate_hz);
+}
+
+}  // namespace webrtc