Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 110 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_processing/gain_controller2.h b/third_party/libwebrtc/modules/audio_processing/gain_controller2.h
new file mode 100644
index 0000000000..43b5828d35
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/gain_controller2.h
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+/*
+ *  Copyright (c) 2017 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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
+#define MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
+
+#include <atomic>
+#include <memory>
+#include <string>
+
+#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/gain_applier.h"
+#include "modules/audio_processing/agc2/input_volume_controller.h"
+#include "modules/audio_processing/agc2/limiter.h"
+#include "modules/audio_processing/agc2/noise_level_estimator.h"
+#include "modules/audio_processing/agc2/saturation_protector.h"
+#include "modules/audio_processing/agc2/speech_level_estimator.h"
+#include "modules/audio_processing/agc2/vad_wrapper.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+class AudioBuffer;
+
+// Gain Controller 2 aims to automatically adjust levels by acting on the
+// microphone gain and/or applying digital gain.
+class GainController2 {
+ public:
+  // Ctor. If `use_internal_vad` is true, an internal voice activity
+  // detector is used for digital adaptive gain.
+  GainController2(
+      const AudioProcessing::Config::GainController2& config,
+      const InputVolumeController::Config& input_volume_controller_config,
+      int sample_rate_hz,
+      int num_channels,
+      bool use_internal_vad);
+  GainController2(const GainController2&) = delete;
+  GainController2& operator=(const GainController2&) = delete;
+  ~GainController2();
+
+  // Sets the fixed digital gain.
+  void SetFixedGainDb(float gain_db);
+
+  // Updates the input volume controller about whether the capture output is
+  // used or not.
+  void SetCaptureOutputUsed(bool capture_output_used);
+
+  // Analyzes `audio_buffer` before `Process()` is called so that the analysis
+  // can be performed before digital processing operations take place (e.g.,
+  // echo cancellation). The analysis consists of input clipping detection and
+  // prediction (if enabled). The value of `applied_input_volume` is limited to
+  // [0, 255].
+  void Analyze(int applied_input_volume, const AudioBuffer& audio_buffer);
+
+  // Updates the recommended input volume, applies the adaptive digital and the
+  // fixed digital gains and runs a limiter on `audio`.
+  // When the internal VAD is not used, `speech_probability` should be specified
+  // and in the [0, 1] range. Otherwise ignores `speech_probability` and
+  // computes the speech probability via `vad_`.
+  // Handles input volume changes; if the caller cannot determine whether an
+  // input volume change occurred, set `input_volume_changed` to false.
+  void Process(absl::optional<float> speech_probability,
+               bool input_volume_changed,
+               AudioBuffer* audio);
+
+  static bool Validate(const AudioProcessing::Config::GainController2& config);
+
+  AvailableCpuFeatures GetCpuFeatures() const { return cpu_features_; }
+
+  absl::optional<int> recommended_input_volume() const {
+    return recommended_input_volume_;
+  }
+
+ private:
+  static std::atomic<int> instance_count_;
+  const AvailableCpuFeatures cpu_features_;
+  ApmDataDumper data_dumper_;
+
+  GainApplier fixed_gain_applier_;
+  std::unique_ptr<NoiseLevelEstimator> noise_level_estimator_;
+  std::unique_ptr<VoiceActivityDetectorWrapper> vad_;
+  std::unique_ptr<SpeechLevelEstimator> speech_level_estimator_;
+  std::unique_ptr<InputVolumeController> input_volume_controller_;
+  // TODO(bugs.webrtc.org/7494): Rename to `CrestFactorEstimator`.
+  std::unique_ptr<SaturationProtector> saturation_protector_;
+  std::unique_ptr<AdaptiveDigitalGainController> adaptive_digital_controller_;
+  Limiter limiter_;
+
+  int calls_since_last_limiter_log_;
+
+  // TODO(bugs.webrtc.org/7494): Remove intermediate storing at this level once
+  // APM refactoring is completed.
+  // Recommended input volume from `InputVolumecontroller`. Non-empty after
+  // `Process()` if input volume controller is enabled and
+  // `InputVolumeController::Process()` has returned a non-empty value.
+  absl::optional<int> recommended_input_volume_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_