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The `recommended_stream_analog_level()` getter is used to retrieve both the applied and the recommended input volume. This behavior is error-prone since the caller must know what is returned based on the point in the code (namely, before/after the AGC has changed the last applied input volume into a recommended level). This CL is a first step to make clarity on which input volume is handled in different parts of APM. Next in the pipeline: make `recommended_stream_analog_level()` a trivial getter that always returns the recommended level. Main changes: - When `recommended_stream_analog_level()` is called but `set_stream_analog_level()` is not called, APM logs an error and returns a fall-back volume (which should not be applied since, when `set_stream_analog_level()` is not called, no external input volume is expected to be present - When APM is used without calling the `*_stream_analog_level()` methods (e.g., when the caller does not provide any input volume), the recorded AEC dumps won't store `Stream::applied_input_level` Other changes: - Removed `AudioProcessingImpl::capture_::prev_analog_mic_level` - Removed redundant code in `GainController2` around detecting input volume changes (already done by APM) - Adapted the `audioproc_f` and `unpack_aecdump` tools - Data dumps clean-up: the applied and the recommended input volumes are now recorded in an AGC implementation agnostic way Bug: webrtc:7494, b/241923537 Change-Id: I3cb4a731fd9f3dc19bf6ac679b7ed8c969ea283b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271544 Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38054}
77 lines
2.9 KiB
C++
77 lines
2.9 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
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#define MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
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#include <atomic>
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#include <memory>
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#include <string>
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#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h"
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#include "modules/audio_processing/agc2/cpu_features.h"
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#include "modules/audio_processing/agc2/gain_applier.h"
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#include "modules/audio_processing/agc2/limiter.h"
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#include "modules/audio_processing/agc2/vad_wrapper.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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namespace webrtc {
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class AudioBuffer;
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// Gain Controller 2 aims to automatically adjust levels by acting on the
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// microphone gain and/or applying digital gain.
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class GainController2 {
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public:
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// Ctor. If `use_internal_vad` is true, an internal voice activity
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// detector is used for digital adaptive gain.
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GainController2(const AudioProcessing::Config::GainController2& config,
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int sample_rate_hz,
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int num_channels,
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bool use_internal_vad);
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GainController2(const GainController2&) = delete;
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GainController2& operator=(const GainController2&) = delete;
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~GainController2();
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// Detects and handles changes of sample rate and/or number of channels.
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void Initialize(int sample_rate_hz, int num_channels);
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// Sets the fixed digital gain.
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void SetFixedGainDb(float gain_db);
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// Applies fixed and adaptive digital gains to `audio` and runs a limiter.
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// If the internal VAD is used, `speech_probability` is ignored. Otherwise
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// `speech_probability` is used for digital adaptive gain if it's available
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// (limited to values [0.0, 1.0]). Handles input volume changes; if the caller
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// cannot determine whether an input volume change occurred, set
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// `input_volume_changed` to false.
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void Process(absl::optional<float> speech_probability,
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bool input_volume_changed,
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AudioBuffer* audio);
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static bool Validate(const AudioProcessing::Config::GainController2& config);
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AvailableCpuFeatures GetCpuFeatures() const { return cpu_features_; }
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private:
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static std::atomic<int> instance_count_;
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const AvailableCpuFeatures cpu_features_;
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ApmDataDumper data_dumper_;
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GainApplier fixed_gain_applier_;
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std::unique_ptr<VoiceActivityDetectorWrapper> vad_;
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std::unique_ptr<AdaptiveDigitalGainController> adaptive_digital_controller_;
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Limiter limiter_;
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int calls_since_last_limiter_log_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
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