AGC2 as a new APM sub-module operating with hard-coded gain.

webrtc/modules/audio_processing/agc2/gain_controller2.cc

Index: webrtc/modules/audio_processing/agc2/gain_controller2.cc
diff --git a/webrtc/modules/audio_processing/agc2/gain_controller2.cc b/webrtc/modules/audio_processing/agc2/gain_controller2.cc
new file mode 100644
index 0000000000000000000000000000000000000000..f0b4f1000e323d117fc21ac4f7a7e8d144805257
--- /dev/null
+++ b/webrtc/modules/audio_processing/agc2/gain_controller2.cc
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2017 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/agc2/gain_controller2.h"
+
+#include "webrtc/base/atomicops.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+int GainController2::instance_count_ = 0;
+
+GainController2::GainController2(int sample_rate_hz)
+    : sample_rate_hz_(sample_rate_hz),
+      data_dumper_(new ApmDataDumper(
+        rtc::AtomicOps::Increment(&instance_count_))),
+      digital_gain_applier_(),
+      hard_coded_digital_gain_(0.9f) {
+  RTC_DCHECK(sample_rate_hz_ == AudioProcessing::kSampleRate8kHz ||
+             sample_rate_hz_ == AudioProcessing::kSampleRate16kHz ||
+             sample_rate_hz_ == AudioProcessing::kSampleRate32kHz ||
+             sample_rate_hz_ == AudioProcessing::kSampleRate48kHz);
+  data_dumper_->InitiateNewSetOfRecordings();
+  data_dumper_->DumpRaw("hard_coded_digital_gain_", 1,
+                        &hard_coded_digital_gain_);
+}
+
+GainController2::~GainController2() = default;
+
+void GainController2::Process(AudioBuffer* audio) {
+  RTC_DCHECK_LT(0, audio->num_channels());
+  digital_gain_applier_.Process(hard_coded_digital_gain_, audio);

[peah-webrtc, 2017/05/18 10:51:19]

As commented on elsewhere. AudioBuffer is handy to use, but we'll need to change
it in the future so I'd prefer to create ArrayViews from the AudioBuffer vectors
and operate on those instead.

[AleBzk, 2017/05/18 12:06:22]

Done.

+}
+
+bool GainController2::Validate(
+    const AudioProcessing::Config::GainController2& config) {
+  return true;
+}
+
+std::string GainController2::ToString(
+    const AudioProcessing::Config::GainController2& config) {
+  std::stringstream ss;
+  ss << "{"
+     << "enabled: " << (config.enabled ? "true" : "false") << "}";
+  return ss.str();
+}
+
+}  // namespace webrtc