chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.cc

/*
 *  Copyright (c) 2012 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/gain_control_impl.h"

#include <cstdint>

#include "absl/types/optional.h"
#include "api/audio/audio_processing.h"
#include "modules/audio_processing/agc/legacy/gain_control.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/field_trial.h"

namespace webrtc {

Handle;

namespace {
int16_t MapSetting(GainControl::Mode mode) { … }

// Applies the sub-frame `gains` to all the bands in `out` and clamps the output
// in the signed 16 bit range.
void ApplyDigitalGain(const int32_t gains[11],
                      size_t num_bands,
                      float* const* out) { … }

}  // namespace

struct GainControlImpl::MonoAgcState { … };

int GainControlImpl::instance_counter_ = …;

GainControlImpl::GainControlImpl()
    : … { … }

GainControlImpl::~GainControlImpl() = default;

void GainControlImpl::ProcessRenderAudio(
    rtc::ArrayView<const int16_t> packed_render_audio) { … }

void GainControlImpl::PackRenderAudioBuffer(
    const AudioBuffer& audio,
    std::vector<int16_t>* packed_buffer) { … }

int GainControlImpl::AnalyzeCaptureAudio(const AudioBuffer& audio) { … }

int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio,
                                         bool stream_has_echo) { … }

// TODO(ajm): ensure this is called under kAdaptiveAnalog.
int GainControlImpl::set_stream_analog_level(int level) { … }

int GainControlImpl::stream_analog_level() const { … }

int GainControlImpl::set_mode(Mode mode) { … }

int GainControlImpl::set_analog_level_limits(int minimum, int maximum) { … }

int GainControlImpl::set_target_level_dbfs(int level) { … }

int GainControlImpl::set_compression_gain_db(int gain) { … }

int GainControlImpl::enable_limiter(bool enable) { … }

void GainControlImpl::Initialize(size_t num_proc_channels, int sample_rate_hz) { … }

int GainControlImpl::Configure() { … }
}  // namespace webrtc