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366a50c4ef
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate. The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes. Bug: webrtc:8982 Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357 Reviewed-on: https://webrtc-review.googlesource.com/98460 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24605}
79 lines
2.7 KiB
C++
79 lines
2.7 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/gain_controller2.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/include/audio_frame_view.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/strings/string_builder.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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fixed_gain_controller_(data_dumper_.get()),
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adaptive_agc_(data_dumper_.get()) {}
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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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fixed_gain_controller_.SetSampleRate(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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}
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void GainController2::Process(AudioBuffer* audio) {
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AudioFrameView<float> float_frame(audio->channels_f(), audio->num_channels(),
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audio->num_frames());
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if (adaptive_digital_mode_) {
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adaptive_agc_.Process(float_frame);
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}
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fixed_gain_controller_.Process(float_frame);
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}
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void GainController2::NotifyAnalogLevel(int level) {
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if (analog_level_ != level && adaptive_digital_mode_) {
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adaptive_agc_.Reset();
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}
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analog_level_ = level;
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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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config_ = config;
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fixed_gain_controller_.SetGain(config_.fixed_gain_db);
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adaptive_digital_mode_ = config_.adaptive_digital_mode;
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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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rtc::StringBuilder 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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