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InputVolumeController: Add configurable speech probability aggregation

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Make speech probability threshold configurable by replacing
kSpeechProbabilitySilenceThreshold with speech_probability_threshold in
InputVolumeController::Config.

Make the processing more robust against outliers in speech probability
estimaton by computing an aggregate speech activity over a speech
segment. In MonoInputVolumeController::Process(), use the passed
non-empty speech probabilities to compute the speech activity over the
speech segment and only allow updates for segments with a high enough
ratio of speech frames. Pass RMS error and speech probability for every
frame in Process(): If rms_error_dbfs is empty, volume updates are not
allowed; if speech_probability is empty, the frame counts as a non-
speech frame.

Remove startup_min_volume from the config since it's no longer used
after https://webrtc-review.googlesource.com/c/src/+/282821.

Bug: webrtc:7494
Change-Id: I0ab81b03371496315348f552133aa9909bd36f26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283523
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38685}
This commit is contained in:
Hanna Silen
2022-11-18 19:36:34 +01:00
committed by WebRTC LUCI CQ
parent f45f823541
commit bf28277774
3 changed files with 232 additions and 80 deletions
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43
modules/audio_processing/agc2/input_volume_controller.h
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@ -36,16 +36,13 @@ class InputVolumeController final {
// Config for the constructor.
struct Config {
bool enabled = false;
// TODO(bugs.webrtc.org/1275566): Describe `startup_min_volume`.
int startup_min_volume = 0;
// Lowest analog microphone level that will be applied in response to
// clipping.
// Lowest input volume level that will be applied in response to clipping.
int clipped_level_min = 70;
// Amount the microphone level is lowered with every clipping event.
// Limited to (0, 255].
// Amount input volume level is lowered with every clipping event. Limited
// to (0, 255].
int clipped_level_step = 15;
// Proportion of clipped samples required to declare a clipping event.
// Limited to (0.f, 1.f).
// Limited to (0.0f, 1.0f).
float clipped_ratio_threshold = 0.1f;
// Time in frames to wait after a clipping event before checking again.
// Limited to values higher than 0.
@ -65,6 +62,12 @@ class InputVolumeController final {
int target_range_min_dbfs = -48;
// Number of wait frames between the recommended input volume updates.
int update_input_volume_wait_frames = 100;
// Speech probability threshold: speech probabilities below the threshold
// are considered silence. Limited to [0.0f, 1.0f].
float speech_probability_threshold = 0.7f;
// Minimum speech frame ratio for volume updates to be allowed. Limited to
// [0.0f, 1.0f].
float speech_ratio_threshold = 0.9f;
};
// Ctor. `num_capture_channels` specifies the number of channels for the audio
@ -90,6 +93,7 @@ class InputVolumeController final {
// prediction (if enabled). Must be called after `set_stream_analog_level()`.
void AnalyzePreProcess(const AudioBuffer& audio_buffer);
// TODO(bugs.webrtc.org/7494): Rename, audio not passed to the method anymore.
// Adjusts the recommended input volume upwards/downwards based on
// `speech_level_dbfs`. Must be called after `AnalyzePreProcess()`. The value
// of `speech_probability` is expected to be in the range [0.0f, 1.0f] and
@ -185,7 +189,9 @@ class MonoInputVolumeController {
public:
MonoInputVolumeController(int clipped_level_min,
int min_mic_level,
int update_input_volume_wait_frames);
int update_input_volume_wait_frames,
float speech_probability_threshold,
float speech_ratio_threshold);
~MonoInputVolumeController();
MonoInputVolumeController(const MonoInputVolumeController&) = delete;
MonoInputVolumeController& operator=(const MonoInputVolumeController&) =
@ -202,10 +208,13 @@ class MonoInputVolumeController {
// `set_stream_analog_level()`.
void HandleClipping(int clipped_level_step);
// Adjusts the recommended input volume upwards/downwards depending on whether
// `rms_error_dbfs` is positive or negative. Must be called after
// `HandleClipping()`.
void Process(absl::optional<int> rms_error_dbfs);
// TODO(bugs.webrtc.org/7494): Rename, audio not passed to the method anymore.
// Adjusts the recommended input volume upwards/downwards depending on
// whether `rms_error_dbfs` is positive or negative. Updates are only allowed
// for active speech segments and when `rms_error_dbfs` is not empty. Must be
// called after `HandleClipping()`.
void Process(absl::optional<int> rms_error_dbfs,
absl::optional<float> speech_probability);
// Returns the recommended input volume. Must be called after `Process()`.
int recommended_analog_level() const { return recommended_input_volume_; }
@ -254,10 +263,18 @@ class MonoInputVolumeController {
const int clipped_level_min_;
// Number of frames waited between the calls to `UpdateInputVolume()`.
// Counters for frames and speech frames since the last update in the
// recommended input volume.
const int update_input_volume_wait_frames_;
int frames_since_update_input_volume_ = 0;
int speech_frames_since_update_input_volume_ = 0;
bool is_first_frame_ = true;
// Speech probability threshold for a frame to be considered speech (instead
// of silence). Limited to [0.0f, 1.0f].
const float speech_probability_threshold_;
// Minimum ratio of speech frames. Limited to [0.0f, 1.0f].
const float speech_ratio_threshold_;
};
} // namespace webrtc