AGC2 AdaptiveDigitalGainApplier and AdaptiveAgc code improvements
This CL was written in preparation for the next CL in the chain and it contains the following changes: - SignalWithLevels -> AdaptiveDigitalGainApplier::FrameInfo - Frame view removed from AdaptiveDigitalGainApplier::FrameInfo - AdaptiveDigitalGainApplier::Process now gets side info as const& to avoid unnecessary copies - AdaptiveAgc::Process: `last_audio_level` renamed to `limiter_envelope` to better reflect what that actually is - Missing class/method docstrings added Tested: bit-exactness verified with audioproc_f Bug: webrtc:7494 Change-Id: Ie25dcd389d6eed74ea9a65f0720eeb8f20f0096b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186040 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32251}
This commit is contained in:
Alessio Bazzica
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commit
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@ -16,6 +16,18 @@
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace {
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void DumpDebugData(const AdaptiveDigitalGainApplier::FrameInfo& info,
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ApmDataDumper& dumper) {
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dumper.DumpRaw("agc2_vad_probability", info.vad_result.speech_probability);
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dumper.DumpRaw("agc2_vad_rms_dbfs", info.vad_result.rms_dbfs);
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dumper.DumpRaw("agc2_vad_peak_dbfs", info.vad_result.peak_dbfs);
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dumper.DumpRaw("agc2_noise_estimate_dbfs", info.input_noise_level_dbfs);
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dumper.DumpRaw("agc2_last_limiter_audio_level", info.limiter_envelope_dbfs);
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}
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} // namespace
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AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper)
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: speech_level_estimator_(apm_data_dumper),
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@ -40,37 +52,17 @@ AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper,
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AdaptiveAgc::~AdaptiveAgc() = default;
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void AdaptiveAgc::Process(AudioFrameView<float> float_frame,
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float last_audio_level) {
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auto signal_with_levels = SignalWithLevels(float_frame);
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signal_with_levels.vad_result = vad_.AnalyzeFrame(float_frame);
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apm_data_dumper_->DumpRaw("agc2_vad_probability",
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signal_with_levels.vad_result.speech_probability);
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apm_data_dumper_->DumpRaw("agc2_vad_rms_dbfs",
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signal_with_levels.vad_result.rms_dbfs);
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apm_data_dumper_->DumpRaw("agc2_vad_peak_dbfs",
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signal_with_levels.vad_result.peak_dbfs);
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speech_level_estimator_.Update(signal_with_levels.vad_result);
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signal_with_levels.input_level_dbfs = speech_level_estimator_.level_dbfs();
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signal_with_levels.input_noise_level_dbfs =
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noise_level_estimator_.Analyze(float_frame);
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apm_data_dumper_->DumpRaw("agc2_noise_estimate_dbfs",
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signal_with_levels.input_noise_level_dbfs);
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signal_with_levels.limiter_audio_level_dbfs =
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last_audio_level > 0 ? FloatS16ToDbfs(last_audio_level) : -90.f;
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apm_data_dumper_->DumpRaw("agc2_last_limiter_audio_level",
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signal_with_levels.limiter_audio_level_dbfs);
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signal_with_levels.estimate_is_confident =
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speech_level_estimator_.IsConfident();
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// The gain applier applies the gain.
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gain_applier_.Process(signal_with_levels);
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void AdaptiveAgc::Process(AudioFrameView<float> frame, float limiter_envelope) {
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AdaptiveDigitalGainApplier::FrameInfo info;
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info.vad_result = vad_.AnalyzeFrame(frame);
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speech_level_estimator_.Update(info.vad_result);
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info.input_level_dbfs = speech_level_estimator_.level_dbfs();
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info.input_noise_level_dbfs = noise_level_estimator_.Analyze(frame);
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info.limiter_envelope_dbfs =
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limiter_envelope > 0 ? FloatS16ToDbfs(limiter_envelope) : -90.f;
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info.estimate_is_confident = speech_level_estimator_.IsConfident();
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DumpDebugData(info, *apm_data_dumper_);
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gain_applier_.Process(info, frame);
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}
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void AdaptiveAgc::Reset() {
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@ -21,6 +21,8 @@
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namespace webrtc {
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class ApmDataDumper;
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// Adaptive digital gain controller.
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// TODO(crbug.com/webrtc/7494): Unify with `AdaptiveDigitalGainApplier`.
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class AdaptiveAgc {
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public:
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explicit AdaptiveAgc(ApmDataDumper* apm_data_dumper);
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@ -28,7 +30,10 @@ class AdaptiveAgc {
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const AudioProcessing::Config::GainController2& config);
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~AdaptiveAgc();
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void Process(AudioFrameView<float> float_frame, float last_audio_level);
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// Analyzes `frame` and applies a digital adaptive gain to it. Takes into
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// account the envelope measured by the limiter.
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// TODO(crbug.com/webrtc/7494): Make the class depend on the limiter.
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void Process(AudioFrameView<float> frame, float limiter_envelope);
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void Reset();
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private:
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@ -85,43 +85,40 @@ float ComputeGainChangeThisFrameDb(float target_gain_db,
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}
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} // namespace
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SignalWithLevels::SignalWithLevels(AudioFrameView<float> float_frame)
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: float_frame(float_frame) {}
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SignalWithLevels::SignalWithLevels(const SignalWithLevels&) = default;
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AdaptiveDigitalGainApplier::AdaptiveDigitalGainApplier(
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ApmDataDumper* apm_data_dumper)
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: gain_applier_(false, DbToRatio(last_gain_db_)),
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apm_data_dumper_(apm_data_dumper) {}
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: apm_data_dumper_(apm_data_dumper),
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gain_applier_(
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/*hard_clip_samples=*/false,
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/*initial_gain_factor=*/DbToRatio(kInitialAdaptiveDigitalGainDb)),
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calls_since_last_gain_log_(0),
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gain_increase_allowed_(true),
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last_gain_db_(kInitialAdaptiveDigitalGainDb) {}
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void AdaptiveDigitalGainApplier::Process(SignalWithLevels signal_with_levels) {
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void AdaptiveDigitalGainApplier::Process(const FrameInfo& info,
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AudioFrameView<float> frame) {
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RTC_DCHECK_GE(info.input_level_dbfs, -150.f);
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RTC_DCHECK_GE(frame.num_channels(), 1);
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RTC_DCHECK_GE(frame.samples_per_channel(), 1);
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// Log every second.
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calls_since_last_gain_log_++;
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if (calls_since_last_gain_log_ == 100) {
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calls_since_last_gain_log_ = 0;
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RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.DigitalGainApplied",
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last_gain_db_, 0, kMaxGainDb, kMaxGainDb + 1);
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RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.EstimatedNoiseLevel",
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-signal_with_levels.input_noise_level_dbfs, 0,
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100, 101);
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-info.input_noise_level_dbfs, 0, 100, 101);
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}
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signal_with_levels.input_level_dbfs =
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std::min(signal_with_levels.input_level_dbfs, 0.f);
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RTC_DCHECK_GE(signal_with_levels.input_level_dbfs, -150.f);
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RTC_DCHECK_GE(signal_with_levels.float_frame.num_channels(), 1);
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RTC_DCHECK_GE(signal_with_levels.float_frame.samples_per_channel(), 1);
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const float target_gain_db = LimitGainByLowConfidence(
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LimitGainByNoise(ComputeGainDb(signal_with_levels.input_level_dbfs),
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signal_with_levels.input_noise_level_dbfs,
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apm_data_dumper_),
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last_gain_db_, signal_with_levels.limiter_audio_level_dbfs,
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signal_with_levels.estimate_is_confident);
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LimitGainByNoise(ComputeGainDb(std::min(info.input_level_dbfs, 0.f)),
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info.input_noise_level_dbfs, apm_data_dumper_),
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last_gain_db_, info.limiter_envelope_dbfs, info.estimate_is_confident);
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// Forbid increasing the gain when there is no speech.
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gain_increase_allowed_ = signal_with_levels.vad_result.speech_probability >
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kVadConfidenceThreshold;
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gain_increase_allowed_ =
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info.vad_result.speech_probability > kVadConfidenceThreshold;
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const float gain_change_this_frame_db = ComputeGainChangeThisFrameDb(
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target_gain_db, last_gain_db_, gain_increase_allowed_);
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@ -137,7 +134,7 @@ void AdaptiveDigitalGainApplier::Process(SignalWithLevels signal_with_levels) {
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gain_applier_.SetGainFactor(
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DbToRatio(last_gain_db_ + gain_change_this_frame_db));
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}
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gain_applier_.ApplyGain(signal_with_levels.float_frame);
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gain_applier_.ApplyGain(frame);
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// Remember that the gain has changed for the next iteration.
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last_gain_db_ = last_gain_db_ + gain_change_this_frame_db;
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@ -11,7 +11,6 @@
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#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
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#include "modules/audio_processing/agc2/agc2_common.h"
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#include "modules/audio_processing/agc2/gain_applier.h"
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#include "modules/audio_processing/agc2/vad_with_level.h"
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#include "modules/audio_processing/include/audio_frame_view.h"
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@ -20,36 +19,38 @@ namespace webrtc {
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class ApmDataDumper;
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struct SignalWithLevels {
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SignalWithLevels(AudioFrameView<float> float_frame);
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SignalWithLevels(const SignalWithLevels&);
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float input_level_dbfs = -1.f;
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float input_noise_level_dbfs = -1.f;
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VadLevelAnalyzer::Result vad_result;
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float limiter_audio_level_dbfs = -1.f;
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bool estimate_is_confident = false;
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AudioFrameView<float> float_frame;
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};
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// Part of the adaptive digital controller that applies a digital adaptive gain.
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// The gain is updated towards a target. The logic decides when gain updates are
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// allowed, it controls the adaptation speed and caps the target based on the
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// estimated noise level and the speech level estimate confidence.
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class AdaptiveDigitalGainApplier {
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public:
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// Information about a frame to process.
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struct FrameInfo {
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float input_level_dbfs; // Estimated speech plus noise level.
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float input_noise_level_dbfs; // Estimated noise level.
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VadLevelAnalyzer::Result vad_result;
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float limiter_envelope_dbfs; // Envelope level from the limiter.
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bool estimate_is_confident;
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};
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explicit AdaptiveDigitalGainApplier(ApmDataDumper* apm_data_dumper);
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// Decide what gain to apply.
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void Process(SignalWithLevels signal_with_levels);
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AdaptiveDigitalGainApplier(const AdaptiveDigitalGainApplier&) = delete;
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AdaptiveDigitalGainApplier& operator=(const AdaptiveDigitalGainApplier&) =
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delete;
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// Analyzes `info`, updates the digital gain and applies it to `frame`.
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void Process(const FrameInfo& info, AudioFrameView<float> frame);
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private:
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float last_gain_db_ = kInitialAdaptiveDigitalGainDb;
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ApmDataDumper* const apm_data_dumper_;
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GainApplier gain_applier_;
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int calls_since_last_gain_log_ = 0;
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// For some combinations of noise and speech probability, increasing
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// the level is not allowed. Since we may get VAD results in bursts,
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// we keep track of this variable until the next VAD results come
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// in.
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bool gain_increase_allowed_ = true;
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ApmDataDumper* apm_data_dumper_ = nullptr;
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int calls_since_last_gain_log_;
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bool gain_increase_allowed_;
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float last_gain_db_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
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@ -20,6 +20,7 @@
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namespace webrtc {
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namespace {
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// Constants used in place of estimated noise levels.
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constexpr float kNoNoiseDbfs = -90.f;
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constexpr float kWithNoiseDbfs = -20.f;
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@ -36,31 +37,25 @@ float RunOnConstantLevel(int num_iterations,
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for (int i = 0; i < num_iterations; ++i) {
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VectorFloatFrame fake_audio(1, 1, 1.f);
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SignalWithLevels signal_with_levels(fake_audio.float_frame_view());
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signal_with_levels.input_level_dbfs = input_level_dbfs;
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signal_with_levels.input_noise_level_dbfs = kNoNoiseDbfs;
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signal_with_levels.vad_result = vad_level;
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signal_with_levels.limiter_audio_level_dbfs = -2.f;
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signal_with_levels.estimate_is_confident = true;
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gain_applier->Process(signal_with_levels);
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AdaptiveDigitalGainApplier::FrameInfo info;
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info.input_level_dbfs = input_level_dbfs;
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info.input_noise_level_dbfs = kNoNoiseDbfs;
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info.vad_result = vad_level;
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info.limiter_envelope_dbfs = -2.f;
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info.estimate_is_confident = true;
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gain_applier->Process(info, fake_audio.float_frame_view());
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gain_linear = fake_audio.float_frame_view().channel(0)[0];
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}
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return gain_linear;
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}
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// Returns 'SignalWithLevels' for typical GainApplier behavior. Voice on, no
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// noise, low limiter, confident level.
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SignalWithLevels TestSignalWithLevel(AudioFrameView<float> float_frame) {
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SignalWithLevels result(float_frame);
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result.input_level_dbfs = -1;
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result.input_noise_level_dbfs = kNoNoiseDbfs;
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result.vad_result = kVadSpeech;
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result.estimate_is_confident = true;
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result.limiter_audio_level_dbfs = -2.f;
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return result;
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}
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} // namespace
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// Voice on, no noise, low limiter, confident level.
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constexpr AdaptiveDigitalGainApplier::FrameInfo kFrameInfo{
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/*input_level_dbfs=*/-1.f,
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/*input_noise_level_dbfs=*/kNoNoiseDbfs,
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/*vad_result=*/kVadSpeech,
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/*limiter_envelope_dbfs=*/-2.f,
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/*estimate_is_confident=*/true};
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TEST(AutomaticGainController2AdaptiveGainApplier, GainApplierShouldNotCrash) {
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ApmDataDumper apm_data_dumper(0);
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@ -68,9 +63,9 @@ TEST(AutomaticGainController2AdaptiveGainApplier, GainApplierShouldNotCrash) {
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// Make one call with reasonable audio level values and settings.
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VectorFloatFrame fake_audio(2, 480, 10000.f);
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auto signal_with_level = TestSignalWithLevel(fake_audio.float_frame_view());
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signal_with_level.input_level_dbfs = -5.0;
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gain_applier.Process(signal_with_level);
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AdaptiveDigitalGainApplier::FrameInfo info = kFrameInfo;
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info.input_level_dbfs = -5.0;
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gain_applier.Process(kFrameInfo, fake_audio.float_frame_view());
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}
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// Check that the output is -kHeadroom dBFS.
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@ -120,9 +115,9 @@ TEST(AutomaticGainController2AdaptiveGainApplier, GainDoesNotChangeFast) {
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for (int i = 0; i < kNumFramesToAdapt; ++i) {
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SCOPED_TRACE(i);
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VectorFloatFrame fake_audio(1, 1, 1.f);
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auto signal_with_level = TestSignalWithLevel(fake_audio.float_frame_view());
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signal_with_level.input_level_dbfs = initial_level_dbfs;
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gain_applier.Process(signal_with_level);
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AdaptiveDigitalGainApplier::FrameInfo info = kFrameInfo;
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info.input_level_dbfs = initial_level_dbfs;
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gain_applier.Process(info, fake_audio.float_frame_view());
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float current_gain_linear = fake_audio.float_frame_view().channel(0)[0];
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EXPECT_LE(std::abs(current_gain_linear - last_gain_linear),
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kMaxChangePerFrameLinear);
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@ -133,9 +128,9 @@ TEST(AutomaticGainController2AdaptiveGainApplier, GainDoesNotChangeFast) {
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for (int i = 0; i < kNumFramesToAdapt; ++i) {
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SCOPED_TRACE(i);
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VectorFloatFrame fake_audio(1, 1, 1.f);
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auto signal_with_level = TestSignalWithLevel(fake_audio.float_frame_view());
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signal_with_level.input_level_dbfs = 0.f;
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gain_applier.Process(signal_with_level);
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AdaptiveDigitalGainApplier::FrameInfo info = kFrameInfo;
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info.input_level_dbfs = 0.f;
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gain_applier.Process(info, fake_audio.float_frame_view());
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float current_gain_linear = fake_audio.float_frame_view().channel(0)[0];
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EXPECT_LE(std::abs(current_gain_linear - last_gain_linear),
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kMaxChangePerFrameLinear);
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@ -151,9 +146,9 @@ TEST(AutomaticGainController2AdaptiveGainApplier, GainIsRampedInAFrame) {
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constexpr int num_samples = 480;
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VectorFloatFrame fake_audio(1, num_samples, 1.f);
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auto signal_with_level = TestSignalWithLevel(fake_audio.float_frame_view());
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signal_with_level.input_level_dbfs = initial_level_dbfs;
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gain_applier.Process(signal_with_level);
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AdaptiveDigitalGainApplier::FrameInfo info = kFrameInfo;
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info.input_level_dbfs = initial_level_dbfs;
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gain_applier.Process(info, fake_audio.float_frame_view());
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float maximal_difference = 0.f;
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float current_value = 1.f * DbToRatio(kInitialAdaptiveDigitalGainDb);
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for (const auto& x : fake_audio.float_frame_view().channel(0)) {
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@ -182,10 +177,10 @@ TEST(AutomaticGainController2AdaptiveGainApplier, NoiseLimitsGain) {
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for (int i = 0; i < num_initial_frames + num_frames; ++i) {
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VectorFloatFrame fake_audio(1, num_samples, 1.f);
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auto signal_with_level = TestSignalWithLevel(fake_audio.float_frame_view());
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signal_with_level.input_level_dbfs = initial_level_dbfs;
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signal_with_level.input_noise_level_dbfs = kWithNoiseDbfs;
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gain_applier.Process(signal_with_level);
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AdaptiveDigitalGainApplier::FrameInfo info = kFrameInfo;
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info.input_level_dbfs = initial_level_dbfs;
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info.input_noise_level_dbfs = kWithNoiseDbfs;
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gain_applier.Process(info, fake_audio.float_frame_view());
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// Wait so that the adaptive gain applier has time to lower the gain.
|
||||
if (i > num_initial_frames) {
|
||||
@ -204,9 +199,9 @@ TEST(AutomaticGainController2GainApplier, CanHandlePositiveSpeechLevels) {
|
||||
|
||||
// Make one call with positive audio level values and settings.
|
||||
VectorFloatFrame fake_audio(2, 480, 10000.f);
|
||||
auto signal_with_level = TestSignalWithLevel(fake_audio.float_frame_view());
|
||||
signal_with_level.input_level_dbfs = 5.0f;
|
||||
gain_applier.Process(signal_with_level);
|
||||
AdaptiveDigitalGainApplier::FrameInfo info = kFrameInfo;
|
||||
info.input_level_dbfs = 5.f;
|
||||
gain_applier.Process(info, fake_audio.float_frame_view());
|
||||
}
|
||||
|
||||
TEST(AutomaticGainController2GainApplier, AudioLevelLimitsGain) {
|
||||
@ -223,11 +218,11 @@ TEST(AutomaticGainController2GainApplier, AudioLevelLimitsGain) {
|
||||
|
||||
for (int i = 0; i < num_initial_frames + num_frames; ++i) {
|
||||
VectorFloatFrame fake_audio(1, num_samples, 1.f);
|
||||
auto signal_with_level = TestSignalWithLevel(fake_audio.float_frame_view());
|
||||
signal_with_level.input_level_dbfs = initial_level_dbfs;
|
||||
signal_with_level.limiter_audio_level_dbfs = 1.f;
|
||||
signal_with_level.estimate_is_confident = false;
|
||||
gain_applier.Process(signal_with_level);
|
||||
AdaptiveDigitalGainApplier::FrameInfo info = kFrameInfo;
|
||||
info.input_level_dbfs = initial_level_dbfs;
|
||||
info.limiter_envelope_dbfs = 1.f;
|
||||
info.estimate_is_confident = false;
|
||||
gain_applier.Process(info, fake_audio.float_frame_view());
|
||||
|
||||
// Wait so that the adaptive gain applier has time to lower the gain.
|
||||
if (i > num_initial_frames) {
|
||||
@ -239,4 +234,6 @@ TEST(AutomaticGainController2GainApplier, AudioLevelLimitsGain) {
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
} // namespace
|
||||
} // namespace webrtc
|
||||
|
||||
Reference in New Issue
Block a user