6df09f6f6a
The functions replace some existing code and will be used in the the new AutomaticGainController. Bug: webrtc:7949 Change-Id: I9a32132d4a4699a507b8548a2eac10972a2f3fd6 Reviewed-on: https://webrtc-review.googlesource.com/53141 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22045}
77 lines
2.5 KiB
C++
77 lines
2.5 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/agc2/gain_controller2.h"
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#include <cmath>
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#include "common_audio/include/audio_util.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/atomicops.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/numerics/safe_minmax.h"
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namespace webrtc {
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int GainController2::instance_count_ = 0;
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GainController2::GainController2()
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: data_dumper_(
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new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
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sample_rate_hz_(AudioProcessing::kSampleRate48kHz),
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fixed_gain_(1.f) {}
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GainController2::~GainController2() = default;
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void GainController2::Initialize(int sample_rate_hz) {
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RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
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sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
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sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
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sample_rate_hz == AudioProcessing::kSampleRate48kHz);
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sample_rate_hz_ = sample_rate_hz;
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data_dumper_->InitiateNewSetOfRecordings();
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data_dumper_->DumpRaw("sample_rate_hz", sample_rate_hz_);
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data_dumper_->DumpRaw("fixed_gain_linear", fixed_gain_);
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}
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void GainController2::Process(AudioBuffer* audio) {
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if (fixed_gain_ == 1.f)
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return;
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for (size_t k = 0; k < audio->num_channels(); ++k) {
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for (size_t j = 0; j < audio->num_frames(); ++j) {
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audio->channels_f()[k][j] = rtc::SafeClamp(
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fixed_gain_ * audio->channels_f()[k][j], -32768.f, 32767.f);
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}
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}
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}
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void GainController2::ApplyConfig(
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const AudioProcessing::Config::GainController2& config) {
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RTC_DCHECK(Validate(config));
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fixed_gain_ = DbToRatio(config.fixed_gain_db);
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}
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bool GainController2::Validate(
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const AudioProcessing::Config::GainController2& config) {
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return config.fixed_gain_db >= 0.f;
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}
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std::string GainController2::ToString(
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const AudioProcessing::Config::GainController2& config) {
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std::stringstream ss;
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ss << "{enabled: " << (config.enabled ? "true" : "false") << ", "
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<< "fixed_gain_dB: " << config.fixed_gain_db << "}";
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return ss.str();
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}
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} // namespace webrtc
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