Added basic framework for AEC3 in the audio processing module

webrtc/modules/audio_processing/aec3/echo_canceller3.cc

+/*
+ *  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/aec3/echo_canceller3.h"
+
+#include "webrtc/base/atomicops.h"
+#include "webrtc/system_wrappers/include/logging.h"
+
+namespace webrtc {
+
+int EchoCanceller3::instance_count_ = 0;
+
+EchoCanceller3::EchoCanceller3(int sample_rate_hz, bool use_anti_hum_filter) {
+  int band_sample_rate_hz = (sample_rate_hz == 8000 ? sample_rate_hz : 16000);
+  frame_length_ = rtc::CheckedDivExact(band_sample_rate_hz, 100);
+
+  LOG(LS_INFO) << "AEC3 created : "
+               << "{ instance_count: " << instance_count_ << "}";
+  instance_count_ = rtc::AtomicOps::Increment(&instance_count_);
+}
+
+EchoCanceller3::~EchoCanceller3() = default;
+
+bool EchoCanceller3::AnalyzeRender(AudioBuffer* render) {
+  RTC_DCHECK_EQ(1u, render->num_channels());
+  RTC_DCHECK_EQ(frame_length_, render->num_frames_per_band());
+  return true;
+}
+
+void EchoCanceller3::AnalyzeCapture(AudioBuffer* capture) {}
+
+void EchoCanceller3::ProcessCapture(AudioBuffer* capture,
+                                    bool known_echo_path_change) {
+  RTC_DCHECK_EQ(1u, capture->num_channels());
+  RTC_DCHECK_EQ(frame_length_, capture->num_frames_per_band());
+}
+
+std::string EchoCanceller3::ToString(
+    const AudioProcessing::Config::EchoCanceller3& config) {
+  std::stringstream ss;
+  ss << "{"
+     << "enabled: " << (config.enabled ? "true" : "false") << "}";
+  return ss.str();
+}
+
+bool EchoCanceller3::Validate(
+    const AudioProcessing::Config::EchoCanceller3& config) {
+  return true;
+}
+
+}  // namespace webrtc