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| 1 /* |
| 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 |
| 11 #include "webrtc/modules/audio_processing/agc2/gain_controller2.h" |
| 12 |
| 13 #include "webrtc/base/atomicops.h" |
| 14 #include "webrtc/base/checks.h" |
| 15 #include "webrtc/modules/audio_processing/audio_buffer.h" |
| 16 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" |
| 17 |
| 18 namespace webrtc { |
| 19 |
| 20 namespace { |
| 21 |
| 22 constexpr float kGain = 0.5f; |
| 23 |
| 24 } // namespace |
| 25 |
| 26 int GainController2::instance_count_ = 0; |
| 27 |
| 28 GainController2::GainController2(int sample_rate_hz) |
| 29 : sample_rate_hz_(sample_rate_hz), |
| 30 data_dumper_(new ApmDataDumper( |
| 31 rtc::AtomicOps::Increment(&instance_count_))), |
| 32 digital_gain_applier_(), |
| 33 gain_(kGain) { |
| 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 data_dumper_->InitiateNewSetOfRecordings(); |
| 39 data_dumper_->DumpRaw("gain_", 1, &gain_); |
| 40 } |
| 41 |
| 42 GainController2::~GainController2() = default; |
| 43 |
| 44 void GainController2::Process(AudioBuffer* audio) { |
| 45 for (size_t k = 0; k < audio->num_channels(); ++k) { |
| 46 auto channel_view = rtc::ArrayView<float>( |
| 47 audio->channels_f()[k], audio->num_frames()); |
| 48 digital_gain_applier_.Process(gain_, channel_view); |
| 49 } |
| 50 } |
| 51 |
| 52 bool GainController2::Validate( |
| 53 const AudioProcessing::Config::GainController2& config) { |
| 54 return true; |
| 55 } |
| 56 |
| 57 std::string GainController2::ToString( |
| 58 const AudioProcessing::Config::GainController2& config) { |
| 59 std::stringstream ss; |
| 60 ss << "{" |
| 61 << "enabled: " << (config.enabled ? "true" : "false") << "}"; |
| 62 return ss.str(); |
| 63 } |
| 64 |
| 65 } // namespace webrtc |
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