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..d920889ca53b8537c33f161dde2579515d645ad2
--- /dev/null
+++ b/webrtc/modules/audio_processing/agc2/gain_controller2.cc
@@ -0,0 +1,66 @@
+/*
+ *  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 {
+
+namespace {
+
+constexpr float kGain = 0.9f;
+
+}  // namespace
+
+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_(),
+      gain_(kGain) {
+  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("gain_", 1, &gain_);
+}
+
+GainController2::~GainController2() = default;
+
+void GainController2::Process(AudioBuffer* audio) {
+  RTC_DCHECK_LT(0, audio->num_channels());

[peah-webrtc, 2017/05/18 12:27:05]

Since you anyway store sample_rate_hz_, I'd suggest that you DCHECK on the rate
being the same in the process call as when AGC2 was created.

[peah-webrtc, 2017/05/18 12:27:05]

Is this DCHECK necessary? It will fail for 0 number of channels, but the current
code actually works with that as well.

[AleBzk, 2017/05/19 13:15:41]

I couldn't find any sample_rate property in AudioBuffer.
I therefore added a DCHECK in APM against
AudioProcessingImpl::proc_sample_rate_hz() (see audio_processing_impl.cc line
1321.

[peah-webrtc, 2017/05/22 08:16:00]

Ah, sorry, I forgot about it not being present.

Please remove the check in line 1321 above. The problem is that you add an api
call in AGC just for that DCHECK which make the API more complex and clutters
the code. Furthermore, I think it can be discussed whether it really increases
the code security. 
My aim with the DCHECK suggestion was that since you anyway had stored the
sample rate, you basically got that DCHECK for free.

+  for (size_t k = 0; k < audio->num_channels(); ++k) {
+    auto channel_view = rtc::ArrayView<float>(
+        audio->channels_f()[k], audio->num_frames());
+    digital_gain_applier_.Process(gain_, channel_view);
+  }
+}
+
+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