0e3198e434
This CL refactors the analog AGC functionality. In particular it: -Breaks then tight dependency between the analog AGC and the digital AGC implementation. -Removes the complicated callback interface for reporting the analog level and replaces it with an int. Bug: webrtc:10859 Change-Id: I3572d60ab98edebbcffa25af64cc74c66f9868fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159039 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29838}
126 lines
4.1 KiB
C++
126 lines
4.1 KiB
C++
/*
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* Copyright (c) 2013 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_AGC_AGC_MANAGER_DIRECT_H_
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#define MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
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#include <memory>
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#include "absl/types/optional.h"
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#include "modules/audio_processing/agc/agc.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/constructor_magic.h"
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#include "rtc_base/gtest_prod_util.h"
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namespace webrtc {
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class AudioFrame;
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class GainControl;
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// Direct interface to use AGC to set volume and compression values.
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// AudioProcessing uses this interface directly to integrate the callback-less
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// AGC.
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//
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// This class is not thread-safe.
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class AgcManagerDirect final {
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public:
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// AgcManagerDirect will configure GainControl internally. The user is
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// responsible for processing the audio using it after the call to Process.
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// The operating range of startup_min_level is [12, 255] and any input value
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// outside that range will be clamped.
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AgcManagerDirect(int startup_min_level,
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int clipped_level_min,
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bool use_agc2_level_estimation,
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bool disable_digital_adaptive);
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~AgcManagerDirect();
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void Initialize();
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void ConfigureGainControl(GainControl* gain_control) const;
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void AnalyzePreProcess(const float* const* audio,
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int num_channels,
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size_t samples_per_channel);
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void Process(const float* audio,
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size_t length,
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int sample_rate_hz,
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GainControl* gain_control);
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// Call when the capture stream has been muted/unmuted. This causes the
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// manager to disregard all incoming audio; chances are good it's background
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// noise to which we'd like to avoid adapting.
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void SetCaptureMuted(bool muted);
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bool capture_muted() { return capture_muted_; }
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float voice_probability();
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int stream_analog_level() const { return stream_analog_level_; }
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void set_stream_analog_level(int level) { stream_analog_level_ = level; }
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private:
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friend class AgcManagerDirectTest;
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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectStandaloneTest,
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DisableDigitalDisablesDigital);
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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectStandaloneTest,
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AgcMinMicLevelExperiment);
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// Dependency injection for testing. Don't delete |agc| as the memory is owned
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// by the manager.
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AgcManagerDirect(Agc* agc,
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int startup_min_level,
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int clipped_level_min);
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int min_mic_level() const { return min_mic_level_; }
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int startup_min_level() const { return startup_min_level_; }
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// Sets a new microphone level, after first checking that it hasn't been
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// updated by the user, in which case no action is taken.
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void SetLevel(int new_level);
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// Set the maximum level the AGC is allowed to apply. Also updates the
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// maximum compression gain to compensate. The level must be at least
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// |kClippedLevelMin|.
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void SetMaxLevel(int level);
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int CheckVolumeAndReset();
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void UpdateGain();
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void UpdateCompressor();
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std::unique_ptr<ApmDataDumper> data_dumper_;
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static int instance_counter_;
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std::unique_ptr<Agc> agc_;
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int frames_since_clipped_;
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int level_;
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int max_level_;
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int max_compression_gain_;
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int target_compression_;
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int compression_;
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float compression_accumulator_;
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bool capture_muted_;
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bool check_volume_on_next_process_;
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bool startup_;
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const int min_mic_level_;
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const bool disable_digital_adaptive_;
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int startup_min_level_;
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const int clipped_level_min_;
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int calls_since_last_gain_log_ = 0;
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int stream_analog_level_ = 0;
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absl::optional<int> new_compression_to_set_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AgcManagerDirect);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
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