blob: 4265500ab568bf56fc3c11870be75ddf715d9c9c [file] [log] [blame]
alessiob3ec96df2017-05-22 06:57:06 -07001/*
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#include "modules/audio_processing/agc2/gain_controller2.h"
alessiob3ec96df2017-05-22 06:57:06 -070012
Alessio Bazzica270f7b52017-10-13 11:05:17 +020013#include <cmath>
14
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020015#include "modules/audio_processing/audio_buffer.h"
16#include "modules/audio_processing/logging/apm_data_dumper.h"
17#include "rtc_base/atomicops.h"
18#include "rtc_base/checks.h"
Alessio Bazzica270f7b52017-10-13 11:05:17 +020019#include "rtc_base/safe_minmax.h"
alessiob3ec96df2017-05-22 06:57:06 -070020
21namespace webrtc {
22
alessiob3ec96df2017-05-22 06:57:06 -070023int GainController2::instance_count_ = 0;
24
Alessio Bazzica270f7b52017-10-13 11:05:17 +020025GainController2::GainController2()
26 : data_dumper_(
27 new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
28 sample_rate_hz_(AudioProcessing::kSampleRate48kHz),
29 fixed_gain_(1.f) {}
alessiob3ec96df2017-05-22 06:57:06 -070030
31GainController2::~GainController2() = default;
32
Alessio Bazzica270f7b52017-10-13 11:05:17 +020033void GainController2::Initialize(int sample_rate_hz) {
34 RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
35 sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
36 sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
37 sample_rate_hz == AudioProcessing::kSampleRate48kHz);
38 sample_rate_hz_ = sample_rate_hz;
39 data_dumper_->InitiateNewSetOfRecordings();
40 data_dumper_->DumpRaw("sample_rate_hz", sample_rate_hz_);
41 data_dumper_->DumpRaw("fixed_gain_linear", fixed_gain_);
42}
43
alessiob3ec96df2017-05-22 06:57:06 -070044void GainController2::Process(AudioBuffer* audio) {
Alessio Bazzica270f7b52017-10-13 11:05:17 +020045 if (fixed_gain_ == 1.f)
46 return;
47
alessiob3ec96df2017-05-22 06:57:06 -070048 for (size_t k = 0; k < audio->num_channels(); ++k) {
Alessio Bazzica270f7b52017-10-13 11:05:17 +020049 for (size_t j = 0; j < audio->num_frames(); ++j) {
50 audio->channels_f()[k][j] = rtc::SafeClamp(
51 fixed_gain_ * audio->channels_f()[k][j], -32768.f, 32767.f);
52 }
alessiob3ec96df2017-05-22 06:57:06 -070053 }
54}
55
Alessio Bazzica270f7b52017-10-13 11:05:17 +020056void GainController2::ApplyConfig(
57 const AudioProcessing::Config::GainController2& config) {
58 RTC_DCHECK(Validate(config));
59 fixed_gain_ = std::pow(10.f, config.fixed_gain_db / 20.f);
60}
61
alessiob3ec96df2017-05-22 06:57:06 -070062bool GainController2::Validate(
63 const AudioProcessing::Config::GainController2& config) {
Alessio Bazzica270f7b52017-10-13 11:05:17 +020064 return config.fixed_gain_db >= 0.f;
alessiob3ec96df2017-05-22 06:57:06 -070065}
66
67std::string GainController2::ToString(
68 const AudioProcessing::Config::GainController2& config) {
69 std::stringstream ss;
Alessio Bazzica270f7b52017-10-13 11:05:17 +020070 ss << "{enabled: " << (config.enabled ? "true" : "false") << ", "
71 << "fixed_gain_dB: " << config.fixed_gain_db << "}";
alessiob3ec96df2017-05-22 06:57:06 -070072 return ss.str();
73}
74
75} // namespace webrtc