| /* |
| * 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 "modules/audio_processing/agc2/gain_controller2.h" |
| |
| #include <cmath> |
| |
| #include "modules/audio_processing/audio_buffer.h" |
| #include "modules/audio_processing/logging/apm_data_dumper.h" |
| #include "rtc_base/atomicops.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/safe_minmax.h" |
| |
| namespace webrtc { |
| |
| int GainController2::instance_count_ = 0; |
| |
| GainController2::GainController2() |
| : data_dumper_( |
| new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))), |
| sample_rate_hz_(AudioProcessing::kSampleRate48kHz), |
| fixed_gain_(1.f) {} |
| |
| GainController2::~GainController2() = default; |
| |
| void GainController2::Initialize(int sample_rate_hz) { |
| RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz || |
| sample_rate_hz == AudioProcessing::kSampleRate16kHz || |
| sample_rate_hz == AudioProcessing::kSampleRate32kHz || |
| sample_rate_hz == AudioProcessing::kSampleRate48kHz); |
| sample_rate_hz_ = sample_rate_hz; |
| data_dumper_->InitiateNewSetOfRecordings(); |
| data_dumper_->DumpRaw("sample_rate_hz", sample_rate_hz_); |
| data_dumper_->DumpRaw("fixed_gain_linear", fixed_gain_); |
| } |
| |
| void GainController2::Process(AudioBuffer* audio) { |
| if (fixed_gain_ == 1.f) |
| return; |
| |
| for (size_t k = 0; k < audio->num_channels(); ++k) { |
| for (size_t j = 0; j < audio->num_frames(); ++j) { |
| audio->channels_f()[k][j] = rtc::SafeClamp( |
| fixed_gain_ * audio->channels_f()[k][j], -32768.f, 32767.f); |
| } |
| } |
| } |
| |
| void GainController2::ApplyConfig( |
| const AudioProcessing::Config::GainController2& config) { |
| RTC_DCHECK(Validate(config)); |
| fixed_gain_ = std::pow(10.f, config.fixed_gain_db / 20.f); |
| } |
| |
| bool GainController2::Validate( |
| const AudioProcessing::Config::GainController2& config) { |
| return config.fixed_gain_db >= 0.f; |
| } |
| |
| std::string GainController2::ToString( |
| const AudioProcessing::Config::GainController2& config) { |
| std::stringstream ss; |
| ss << "{enabled: " << (config.enabled ? "true" : "false") << ", " |
| << "fixed_gain_dB: " << config.fixed_gain_db << "}"; |
| return ss.str(); |
| } |
| |
| } // namespace webrtc |