92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
66 lines
2.0 KiB
C++
66 lines
2.0 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 "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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namespace webrtc {
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namespace {
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constexpr float kGain = 0.5f;
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} // namespace
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int GainController2::instance_count_ = 0;
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GainController2::GainController2(int sample_rate_hz)
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: sample_rate_hz_(sample_rate_hz),
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data_dumper_(new ApmDataDumper(
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rtc::AtomicOps::Increment(&instance_count_))),
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digital_gain_applier_(),
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gain_(kGain) {
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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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data_dumper_->InitiateNewSetOfRecordings();
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data_dumper_->DumpRaw("gain_", 1, &gain_);
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}
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GainController2::~GainController2() = default;
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void GainController2::Process(AudioBuffer* audio) {
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for (size_t k = 0; k < audio->num_channels(); ++k) {
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auto channel_view = rtc::ArrayView<float>(
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audio->channels_f()[k], audio->num_frames());
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digital_gain_applier_.Process(gain_, channel_view);
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}
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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 true;
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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 << "{"
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<< "enabled: " << (config.enabled ? "true" : "false") << "}";
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return ss.str();
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}
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} // namespace webrtc
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