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Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.cc | 103 |
1 files changed, 103 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.cc b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.cc new file mode 100644 index 0000000000..f9e276d3a8 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.cc @@ -0,0 +1,103 @@ +/* + * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/audio_processing/agc2/gain_applier.h" + +#include "api/array_view.h" +#include "modules/audio_processing/agc2/agc2_common.h" +#include "rtc_base/numerics/safe_minmax.h" + +namespace webrtc { +namespace { + +// Returns true when the gain factor is so close to 1 that it would +// not affect int16 samples. +bool GainCloseToOne(float gain_factor) { + return 1.f - 1.f / kMaxFloatS16Value <= gain_factor && + gain_factor <= 1.f + 1.f / kMaxFloatS16Value; +} + +void ClipSignal(AudioFrameView<float> signal) { + for (int k = 0; k < signal.num_channels(); ++k) { + rtc::ArrayView<float> channel_view = signal.channel(k); + for (auto& sample : channel_view) { + sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value); + } + } +} + +void ApplyGainWithRamping(float last_gain_linear, + float gain_at_end_of_frame_linear, + float inverse_samples_per_channel, + AudioFrameView<float> float_frame) { + // Do not modify the signal. + if (last_gain_linear == gain_at_end_of_frame_linear && + GainCloseToOne(gain_at_end_of_frame_linear)) { + return; + } + + // Gain is constant and different from 1. + if (last_gain_linear == gain_at_end_of_frame_linear) { + for (int k = 0; k < float_frame.num_channels(); ++k) { + rtc::ArrayView<float> channel_view = float_frame.channel(k); + for (auto& sample : channel_view) { + sample *= gain_at_end_of_frame_linear; + } + } + return; + } + + // The gain changes. We have to change slowly to avoid discontinuities. + const float increment = (gain_at_end_of_frame_linear - last_gain_linear) * + inverse_samples_per_channel; + float gain = last_gain_linear; + for (int i = 0; i < float_frame.samples_per_channel(); ++i) { + for (int ch = 0; ch < float_frame.num_channels(); ++ch) { + float_frame.channel(ch)[i] *= gain; + } + gain += increment; + } +} + +} // namespace + +GainApplier::GainApplier(bool hard_clip_samples, float initial_gain_factor) + : hard_clip_samples_(hard_clip_samples), + last_gain_factor_(initial_gain_factor), + current_gain_factor_(initial_gain_factor) {} + +void GainApplier::ApplyGain(AudioFrameView<float> signal) { + if (static_cast<int>(signal.samples_per_channel()) != samples_per_channel_) { + Initialize(signal.samples_per_channel()); + } + + ApplyGainWithRamping(last_gain_factor_, current_gain_factor_, + inverse_samples_per_channel_, signal); + + last_gain_factor_ = current_gain_factor_; + + if (hard_clip_samples_) { + ClipSignal(signal); + } +} + +// TODO(bugs.webrtc.org/7494): Remove once switched to gains in dB. +void GainApplier::SetGainFactor(float gain_factor) { + RTC_DCHECK_GT(gain_factor, 0.f); + current_gain_factor_ = gain_factor; +} + +void GainApplier::Initialize(int samples_per_channel) { + RTC_DCHECK_GT(samples_per_channel, 0); + samples_per_channel_ = static_cast<int>(samples_per_channel); + inverse_samples_per_channel_ = 1.f / samples_per_channel_; +} + +} // namespace webrtc |