Bump to WebRTC M120 release
Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone. We're continuing to carry iSAC even though it's gone upstream, but maybe we'll want to drop that soon.
This commit is contained in:
Arun Raghavan
@ -11,11 +11,15 @@
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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 <atomic>
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#include <memory>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "modules/audio_processing/agc/agc.h"
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#include "modules/audio_processing/agc2/clipping_predictor.h"
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#include "modules/audio_processing/audio_buffer.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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#include "rtc_base/gtest_prod_util.h"
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@ -24,89 +28,168 @@ namespace webrtc {
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class MonoAgc;
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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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// Adaptive Gain Controller (AGC) that controls the input volume and a digital
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// gain. The input volume controller recommends what volume to use, handles
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// volume changes and clipping. In particular, it handles changes triggered by
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// the user (e.g., volume set to zero by a HW mute button). The digital
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// controller chooses and applies the digital compression gain.
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// This class is not thread-safe.
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// TODO(bugs.webrtc.org/7494): Use applied/recommended input volume naming
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// convention.
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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 num_capture_channels,
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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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int sample_rate_hz);
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// Ctor. `num_capture_channels` specifies the number of channels for the audio
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// passed to `AnalyzePreProcess()` and `Process()`. Clamps
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// `analog_config.startup_min_level` in the [12, 255] range.
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AgcManagerDirect(
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int num_capture_channels,
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const AudioProcessing::Config::GainController1::AnalogGainController&
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analog_config);
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~AgcManagerDirect();
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AgcManagerDirect(const AgcManagerDirect&) = delete;
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AgcManagerDirect& operator=(const AgcManagerDirect&) = delete;
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void Initialize();
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void SetupDigitalGainControl(GainControl* gain_control) const;
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void AnalyzePreProcess(const AudioBuffer* audio);
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void Process(const AudioBuffer* audio);
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// Configures `gain_control` to work as a fixed digital controller so that the
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// adaptive part is only handled by this gain controller. Must be called if
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// `gain_control` is also used to avoid the side-effects of running two AGCs.
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void SetupDigitalGainControl(GainControl& gain_control) const;
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// Sets the applied input volume.
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void set_stream_analog_level(int level);
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// TODO(bugs.webrtc.org/7494): Add argument for the applied input volume and
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// remove `set_stream_analog_level()`.
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// Analyzes `audio` before `Process()` is called so that the analysis can be
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// performed before external digital processing operations take place (e.g.,
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// echo cancellation). The analysis consists of input clipping detection and
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// prediction (if enabled). Must be called after `set_stream_analog_level()`.
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void AnalyzePreProcess(const AudioBuffer& audio_buffer);
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// Processes `audio_buffer`. Chooses a digital compression gain and the new
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// input volume to recommend. Must be called after `AnalyzePreProcess()`. If
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// `speech_probability` (range [0.0f, 1.0f]) and `speech_level_dbfs` (range
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// [-90.f, 30.0f]) are given, uses them to override the estimated RMS error.
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// TODO(webrtc:7494): This signature is needed for testing purposes, unify
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// the signatures when the clean-up is done.
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void Process(const AudioBuffer& audio_buffer,
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absl::optional<float> speech_probability,
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absl::optional<float> speech_level_dbfs);
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// Processes `audio_buffer`. Chooses a digital compression gain and the new
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// input volume to recommend. Must be called after `AnalyzePreProcess()`.
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void Process(const AudioBuffer& audio_buffer);
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// TODO(bugs.webrtc.org/7494): Return recommended input volume and remove
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// `recommended_analog_level()`.
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// Returns the recommended input volume. If the input volume contoller is
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// disabled, returns the input volume set via the latest
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// `set_stream_analog_level()` call. Must be called after
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// `AnalyzePreProcess()` and `Process()`.
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int recommended_analog_level() const { return recommended_input_volume_; }
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// Call when the capture stream output has been flagged to be used/not-used.
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// If unused, the manager disregards all incoming audio.
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void HandleCaptureOutputUsedChange(bool capture_output_used);
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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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float voice_probability() const;
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int stream_analog_level() const { return stream_analog_level_; }
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void set_stream_analog_level(int level);
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int num_channels() const { return num_capture_channels_; }
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int sample_rate_hz() const { return sample_rate_hz_; }
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// If available, returns a new compression gain for the digital gain control.
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// If available, returns the latest digital compression gain that has been
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// chosen.
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absl::optional<int> GetDigitalComressionGain();
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// Returns true if clipping prediction is enabled.
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bool clipping_predictor_enabled() const { return !!clipping_predictor_; }
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// Returns true if clipping prediction is used to adjust the input volume.
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bool use_clipping_predictor_step() const {
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return use_clipping_predictor_step_;
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}
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private:
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friend class AgcManagerDirectTest;
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friend class AgcManagerDirectTestHelper;
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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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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectTest, DisableDigitalDisablesDigital);
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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectTest,
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AgcMinMicLevelExperimentDefault);
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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectTest,
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AgcMinMicLevelExperimentDisabled);
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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectTest,
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AgcMinMicLevelExperimentOutOfRangeAbove);
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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectTest,
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AgcMinMicLevelExperimentOutOfRangeBelow);
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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectTest,
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AgcMinMicLevelExperimentEnabled50);
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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectTest,
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AgcMinMicLevelExperimentEnabledAboveStartupLevel);
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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectParametrizedTest,
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ClippingParametersVerified);
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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectParametrizedTest,
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DisableClippingPredictorDoesNotLowerVolume);
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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectParametrizedTest,
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UsedClippingPredictionsProduceLowerAnalogLevels);
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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectParametrizedTest,
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UnusedClippingPredictionsProduceEqualAnalogLevels);
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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectParametrizedTest,
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EmptyRmsErrorOverrideHasNoEffect);
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FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectParametrizedTest,
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NonEmptyRmsErrorOverrideHasEffect);
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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 sample_rate_hz);
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void AnalyzePreProcess(const float* const* audio, size_t samples_per_channel);
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// Ctor that creates a single channel AGC and by injecting `agc`.
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// `agc` will be owned by this class; hence, do not delete it.
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AgcManagerDirect(
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const AudioProcessing::Config::GainController1::AnalogGainController&
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analog_config,
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Agc* agc);
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void AggregateChannelLevels();
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const bool analog_controller_enabled_;
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const absl::optional<int> min_mic_level_override_;
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std::unique_ptr<ApmDataDumper> data_dumper_;
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static int instance_counter_;
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const bool use_min_channel_level_;
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const int sample_rate_hz_;
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static std::atomic<int> instance_counter_;
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const int num_capture_channels_;
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const bool disable_digital_adaptive_;
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int frames_since_clipped_;
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int stream_analog_level_ = 0;
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bool capture_muted_;
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// TODO(bugs.webrtc.org/7494): Create a separate member for the applied input
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// volume.
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// TODO(bugs.webrtc.org/7494): Once
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// `AudioProcessingImpl::recommended_stream_analog_level()` becomes a trivial
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// getter, leave uninitialized.
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// Recommended input volume. After `set_stream_analog_level()` is called it
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// holds the observed input volume. Possibly updated by `AnalyzePreProcess()`
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// and `Process()`; after these calls, holds the recommended input volume.
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int recommended_input_volume_ = 0;
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bool capture_output_used_;
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int channel_controlling_gain_ = 0;
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const int clipped_level_step_;
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const float clipped_ratio_threshold_;
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const int clipped_wait_frames_;
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std::vector<std::unique_ptr<MonoAgc>> channel_agcs_;
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std::vector<absl::optional<int>> new_compressions_to_set_;
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const std::unique_ptr<ClippingPredictor> clipping_predictor_;
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const bool use_clipping_predictor_step_;
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float clipping_rate_log_;
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int clipping_rate_log_counter_;
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};
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// TODO(bugs.webrtc.org/7494): Use applied/recommended input volume naming
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// convention.
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class MonoAgc {
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public:
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MonoAgc(ApmDataDumper* data_dumper,
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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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int min_mic_level);
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~MonoAgc();
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@ -114,16 +197,27 @@ class MonoAgc {
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MonoAgc& operator=(const MonoAgc&) = delete;
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void Initialize();
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void SetCaptureMuted(bool muted);
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void HandleCaptureOutputUsedChange(bool capture_output_used);
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void HandleClipping();
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// Sets the current input volume.
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void set_stream_analog_level(int level) { recommended_input_volume_ = level; }
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void Process(const int16_t* audio,
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size_t samples_per_channel,
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int sample_rate_hz);
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// Lowers the recommended input volume in response to clipping based on the
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// suggested reduction `clipped_level_step`. Must be called after
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// `set_stream_analog_level()`.
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void HandleClipping(int clipped_level_step);
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// Analyzes `audio`, requests the RMS error from AGC, updates the recommended
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// input volume based on the estimated speech level and, if enabled, updates
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// the (digital) compression gain to be applied by `agc_`. Must be called
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// after `HandleClipping()`. If `rms_error_override` has a value, RMS error
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// from AGC is overridden by it.
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void Process(rtc::ArrayView<const int16_t> audio,
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absl::optional<int> rms_error_override);
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// Returns the recommended input volume. Must be called after `Process()`.
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int recommended_analog_level() const { return recommended_input_volume_; }
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void set_stream_analog_level(int level) { stream_analog_level_ = level; }
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int stream_analog_level() const { return stream_analog_level_; }
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float voice_probability() const { return agc_->voice_probability(); }
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void ActivateLogging() { log_to_histograms_ = true; }
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absl::optional<int> new_compression() const {
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@ -133,20 +227,19 @@ class MonoAgc {
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// Only used for testing.
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void set_agc(Agc* agc) { agc_.reset(agc); }
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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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private:
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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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// Sets a new input volume, after first checking that it hasn't been updated
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// 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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// Set the maximum input volume the AGC is allowed to apply. Also updates the
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// maximum compression gain to compensate. The volume 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 UpdateGain(int rms_error_db);
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void UpdateCompressor();
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const int min_mic_level_;
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@ -158,15 +251,26 @@ class MonoAgc {
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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_ = false;
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bool capture_output_used_ = true;
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bool check_volume_on_next_process_ = true;
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bool startup_ = true;
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int startup_min_level_;
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int calls_since_last_gain_log_ = 0;
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int stream_analog_level_ = 0;
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// TODO(bugs.webrtc.org/7494): Create a separate member for the applied
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// input volume.
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// Recommended input volume. After `set_stream_analog_level()` is
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// called, it holds the observed applied input volume. Possibly updated by
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// `HandleClipping()` and `Process()`; after these calls, holds the
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// recommended input volume.
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int recommended_input_volume_ = 0;
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absl::optional<int> new_compression_to_set_;
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bool log_to_histograms_ = false;
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const int clipped_level_min_;
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// Frames since the last `UpdateGain()` call.
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int frames_since_update_gain_ = 0;
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// Set to true for the first frame after startup and reset, otherwise false.
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bool is_first_frame_ = true;
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};
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} // namespace webrtc
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