Moved the GainControlForNewAGC class to a separate file.

webrtc/modules/audio_processing/gain_control_for_experimental_agc.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/gain_control_for_experimental_agc.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+GainControlForExperimentalAgc::GainControlForExperimentalAgc(
+    GainControl* gain_control,
+    rtc::CriticalSection* crit_capture)
+    : real_gain_control_(gain_control),
+      volume_(0),
+      crit_capture_(crit_capture) {}
+
+int GainControlForExperimentalAgc::Enable(bool enable) {
+  return real_gain_control_->Enable(enable);
+}
+
+bool GainControlForExperimentalAgc::is_enabled() const {
+  return real_gain_control_->is_enabled();
+}
+
+int GainControlForExperimentalAgc::set_stream_analog_level(int level) {
+  rtc::CritScope cs_capture(crit_capture_);
+  volume_ = level;
+  return AudioProcessing::kNoError;
+}
+
+int GainControlForExperimentalAgc::stream_analog_level() {
+  rtc::CritScope cs_capture(crit_capture_);
+  return volume_;
+}
+
+int GainControlForExperimentalAgc::set_mode(Mode mode) {
+  return AudioProcessing::kNoError;
+}
+
+GainControl::Mode GainControlForExperimentalAgc::mode() const {
+  return GainControl::kAdaptiveAnalog;
+}
+
+int GainControlForExperimentalAgc::set_target_level_dbfs(int level) {
+  return AudioProcessing::kNoError;
+}
+
+int GainControlForExperimentalAgc::target_level_dbfs() const {
+  return real_gain_control_->target_level_dbfs();
+}
+
+int GainControlForExperimentalAgc::set_compression_gain_db(int gain) {
+  return AudioProcessing::kNoError;
+}
+
+int GainControlForExperimentalAgc::compression_gain_db() const {
+  return real_gain_control_->compression_gain_db();
+}
+
+int GainControlForExperimentalAgc::enable_limiter(bool enable) {
+  return AudioProcessing::kNoError;
+}
+
+bool GainControlForExperimentalAgc::is_limiter_enabled() const {
+  return real_gain_control_->is_limiter_enabled();
+}
+
+int GainControlForExperimentalAgc::set_analog_level_limits(int minimum,
+                                                           int maximum) {
+  return AudioProcessing::kNoError;
+}
+
+int GainControlForExperimentalAgc::analog_level_minimum() const {
+  return real_gain_control_->analog_level_minimum();
+}
+
+int GainControlForExperimentalAgc::analog_level_maximum() const {
+  return real_gain_control_->analog_level_maximum();
+}
+
+bool GainControlForExperimentalAgc::stream_is_saturated() const {
+  return real_gain_control_->stream_is_saturated();
+}
+
+void GainControlForExperimentalAgc::SetMicVolume(int volume) {
+  rtc::CritScope cs_capture(crit_capture_);
+  volume_ = volume;
+}
+
+int GainControlForExperimentalAgc::GetMicVolume() {
+  rtc::CritScope cs_capture(crit_capture_);
+  return volume_;
+}
+
+}  // namespace webrtc