Implement InputVolumeController
Implement InputVolumeController and RecommendedInputVolumeEstimator based on the copy of agc classes AgcManagerDirect and MonoAgc. Copies of the original files created in https://webrtc-review.googlesource.com/c/src/+/278624. Bug: webrtc:7494 Change-Id: I74acee57b0db5cc8a6b666be9ba619c6c98a1773 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278625 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Hanna Silen <silen@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38533}
This commit is contained in:
Hanna Silen
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WebRTC LUCI CQ
WebRTC LUCI CQ
parent
adbcbf73fa
commit
9f06ef1cc3
@ -17,35 +17,55 @@
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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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namespace webrtc {
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class RecommendedInputVolumeEstimator;
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class GainControl;
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class MonoInputVolumeController;
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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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// Input volume controller that controls the input volume. The input volume
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// controller recommends what volume to use, handles volume changes and
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// clipping. In particular, it handles changes triggered by the user (e.g.,
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// volume set to zero by a HW mute button). The digital controller chooses and
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// applies the digital compression gain. 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 InputVolumeController final {
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public:
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// Config for the constructor.
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struct Config {
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bool enabled = false;
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// TODO(bugs.webrtc.org/1275566): Describe `startup_min_volume`.
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int startup_min_volume = 0;
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// Lowest analog microphone level that will be applied in response to
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// clipping.
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int clipped_level_min = 70;
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// If true, an adaptive digital gain is applied.
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bool digital_adaptive_follows = true;
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// Amount the microphone level is lowered with every clipping event.
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// Limited to (0, 255].
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int clipped_level_step = 15;
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// Proportion of clipped samples required to declare a clipping event.
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// Limited to (0.f, 1.f).
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float clipped_ratio_threshold = 0.1f;
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// Time in frames to wait after a clipping event before checking again.
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// Limited to values higher than 0.
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int clipped_wait_frames = 300;
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// Enables clipping prediction functionality.
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bool enable_clipping_predictor = false;
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// Minimum and maximum digital gain used before input volume is
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// adjusted.
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int max_digital_gain_db = 30;
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int min_digital_gain_db = 0;
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};
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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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InputVolumeController(
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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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// `config.startup_min_level` in the [12, 255] range.
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InputVolumeController(int num_capture_channels, const Config& config);
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~InputVolumeController();
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InputVolumeController(const InputVolumeController&) = delete;
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@ -53,11 +73,6 @@ class InputVolumeController final {
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void Initialize();
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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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@ -69,20 +84,13 @@ class InputVolumeController final {
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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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// Chooses a digital compression gain and the new input volume to recommend.
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// Must be called after `AnalyzePreProcess()`. `speech_probability`
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// (range [0.0f, 1.0f]) and `speech_level_dbfs` (range [-90.f, 30.0f]) are
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// used to compute the RMS error.
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void Process(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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@ -138,25 +146,17 @@ class InputVolumeController final {
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UnusedClippingPredictionsProduceEqualAnalogLevels);
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FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerParametrizedTest,
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EmptyRmsErrorOverrideHasNoEffect);
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FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerParametrizedTest,
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NonEmptyRmsErrorOverrideHasEffect);
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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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InputVolumeController(
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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 std::atomic<int> instance_counter_;
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const bool use_min_channel_level_;
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const int num_capture_channels_;
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// TODO(webrtc:7494): Replace with `digital_adaptive_follows_`.
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const bool disable_digital_adaptive_;
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int frames_since_clipped_;
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@ -178,8 +178,7 @@ class InputVolumeController final {
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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<RecommendedInputVolumeEstimator>> channel_agcs_;
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std::vector<absl::optional<int>> new_compressions_to_set_;
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std::vector<std::unique_ptr<MonoInputVolumeController>> channel_controllers_;
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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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@ -189,18 +188,18 @@ class InputVolumeController final {
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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 RecommendedInputVolumeEstimator {
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class MonoInputVolumeController {
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public:
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RecommendedInputVolumeEstimator(ApmDataDumper* data_dumper,
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int startup_min_level,
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int clipped_level_min,
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bool disable_digital_adaptive,
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int min_mic_level);
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~RecommendedInputVolumeEstimator();
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RecommendedInputVolumeEstimator(const RecommendedInputVolumeEstimator&) =
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MonoInputVolumeController(int startup_min_level,
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int clipped_level_min,
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bool disable_digital_adaptive,
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int min_mic_level,
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int max_digital_gain_db,
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int min_digital_gain_db);
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~MonoInputVolumeController();
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MonoInputVolumeController(const MonoInputVolumeController&) = delete;
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MonoInputVolumeController& operator=(const MonoInputVolumeController&) =
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delete;
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RecommendedInputVolumeEstimator& operator=(
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const RecommendedInputVolumeEstimator&) = delete;
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void Initialize();
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void HandleCaptureOutputUsedChange(bool capture_output_used);
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@ -213,25 +212,16 @@ class RecommendedInputVolumeEstimator {
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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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// Updates the recommended input volume based on the estimated speech level
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// RMS error. Must be called after `HandleClipping()`.
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void Process(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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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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return new_compression_to_set_;
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}
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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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@ -240,29 +230,27 @@ class RecommendedInputVolumeEstimator {
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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 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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// Set the maximum input volume the input volume controller is allowed to
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// apply. The volume must be at least `kClippedLevelMin`.
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void SetMaxLevel(int level);
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int CheckVolumeAndReset();
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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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// TODO(webrtc:7494): Replace with `digital_adaptive_follows_`.
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const bool disable_digital_adaptive_;
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std::unique_ptr<Agc> agc_;
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const int max_digital_gain_db_;
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const int min_digital_gain_db_;
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int level_ = 0;
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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_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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// TODO(bugs.webrtc.org/7494): Create a separate member for the applied
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// input volume.
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@ -272,13 +260,12 @@ class RecommendedInputVolumeEstimator {
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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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