| /* |
| * Copyright (c) 2012 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. |
| */ |
| |
| #ifndef WEBRTC_TOOLS_AGC_FAKE_AGC_H_ |
| #define WEBRTC_TOOLS_AGC_FAKE_AGC_H_ |
| |
| #include "webrtc/modules/audio_processing/agc/agc.h" |
| |
| namespace webrtc { |
| |
| class FakeAgc : public Agc { |
| public: |
| FakeAgc() |
| : counter_(0), |
| volume_(kMaxVolume / 2) { |
| } |
| |
| virtual int Process(const AudioFrame& audio_frame) { |
| const int kUpdateIntervalFrames = 10; |
| const int kMaxVolume = 255; |
| if (counter_ % kUpdateIntervalFrames == 0) { |
| volume_ = (++volume_) % kMaxVolume; |
| } |
| counter_++; |
| return 0; |
| } |
| |
| virtual int FakeAgc::MicVolume() { |
| return volume_; |
| } |
| |
| private: |
| int counter_; |
| int volume_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_TOOLS_AGC_FAKE_AGC_H_ |