From 992a96f68e4cadc69ff85ffe85500010e761e2bd Mon Sep 17 00:00:00 2001 From: Gustaf Ullberg Date: Tue, 8 Dec 2020 13:03:55 +0100 Subject: [PATCH] AEC3: Prevent diverging coarse filter from influencing the refined filter MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit After the refined filter has been determined to perform better than the coarse filter, and the coefficients of the coarse filters are overwritten by the ones from the refined filter, at least 100 ms have to pass before the adaptation of the refined filter is allowed to speed up due to good coarse filter performance. This change solves the vicious circle described in webrtc:12265, where the coarse and refined filters can diverge over time. This feature can be disabled remotely via a kill-switch. When disabled the AEC output is bit-exact to before the change. Bug: webrtc:12265,chromium:1155477 Change-Id: Iacd6e325e987dd8a475bb3e8163fee714c65b20a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196501 Reviewed-by: Per Åhgren Commit-Queue: Gustaf Ullberg Cr-Commit-Position: refs/heads/master@{#32801} --- api/audio/echo_canceller3_config.cc | 1 + api/audio/echo_canceller3_config.h | 1 + api/audio/echo_canceller3_config_json.cc | 4 ++++ modules/audio_processing/aec3/aec_state.cc | 5 +++++ .../aec3/refined_filter_update_gain.cc | 3 ++- .../aec3/refined_filter_update_gain.h | 1 + .../refined_filter_update_gain_unittest.cc | 5 +++-- modules/audio_processing/aec3/subtractor.cc | 22 ++++++++++++++++++- modules/audio_processing/aec3/subtractor.h | 2 ++ .../output_data_float.pb.sha1 | 2 +- .../output_data_float_avx2.pb.sha1 | 2 +- 11 files changed, 42 insertions(+), 6 deletions(-) diff --git a/api/audio/echo_canceller3_config.cc b/api/audio/echo_canceller3_config.cc index aeb809efa9..5f1923e90f 100644 --- a/api/audio/echo_canceller3_config.cc +++ b/api/audio/echo_canceller3_config.cc @@ -153,6 +153,7 @@ bool EchoCanceller3Config::Validate(EchoCanceller3Config* config) { res = res & Limit(&c->filter.config_change_duration_blocks, 0, 100000); res = res & Limit(&c->filter.initial_state_seconds, 0.f, 100.f); + res = res & Limit(&c->filter.coarse_reset_hangover_blocks, 0, 2500); res = res & Limit(&c->erle.min, 1.f, 100000.f); res = res & Limit(&c->erle.max_l, 1.f, 100000.f); diff --git a/api/audio/echo_canceller3_config.h b/api/audio/echo_canceller3_config.h index 3ed11ff8b3..55281af93b 100644 --- a/api/audio/echo_canceller3_config.h +++ b/api/audio/echo_canceller3_config.h @@ -86,6 +86,7 @@ struct RTC_EXPORT EchoCanceller3Config { size_t config_change_duration_blocks = 250; float initial_state_seconds = 2.5f; + int coarse_reset_hangover_blocks = 25; bool conservative_initial_phase = false; bool enable_coarse_filter_output_usage = true; bool use_linear_filter = true; diff --git a/api/audio/echo_canceller3_config_json.cc b/api/audio/echo_canceller3_config_json.cc index 907b472714..9d10da9949 100644 --- a/api/audio/echo_canceller3_config_json.cc +++ b/api/audio/echo_canceller3_config_json.cc @@ -223,6 +223,8 @@ void Aec3ConfigFromJsonString(absl::string_view json_string, &cfg.filter.config_change_duration_blocks); ReadParam(section, "initial_state_seconds", &cfg.filter.initial_state_seconds); + ReadParam(section, "coarse_reset_hangover_blocks", + &cfg.filter.coarse_reset_hangover_blocks); ReadParam(section, "conservative_initial_phase", &cfg.filter.conservative_initial_phase); ReadParam(section, "enable_coarse_filter_output_usage", @@ -502,6 +504,8 @@ std::string Aec3ConfigToJsonString(const EchoCanceller3Config& config) { << config.filter.config_change_duration_blocks << ","; ost << "\"initial_state_seconds\": " << config.filter.initial_state_seconds << ","; + ost << "\"coarse_reset_hangover_blocks\": " + << config.filter.coarse_reset_hangover_blocks << ","; ost << "\"conservative_initial_phase\": " << (config.filter.conservative_initial_phase ? "true" : "false") << ","; ost << "\"enable_coarse_filter_output_usage\": " diff --git a/modules/audio_processing/aec3/aec_state.cc b/modules/audio_processing/aec3/aec_state.cc index c7361093f3..5b31e3cb9f 100644 --- a/modules/audio_processing/aec3/aec_state.cc +++ b/modules/audio_processing/aec3/aec_state.cc @@ -322,6 +322,11 @@ void AecState::Update( external_delay ? 1 : 0); data_dumper_->DumpRaw("aec3_filter_tail_freq_resp_est", GetReverbFrequencyResponse()); + data_dumper_->DumpRaw("aec3_subtractor_y2", subtractor_output[0].y2); + data_dumper_->DumpRaw("aec3_subtractor_e2_coarse", + subtractor_output[0].e2_coarse); + data_dumper_->DumpRaw("aec3_subtractor_e2_refined", + subtractor_output[0].e2_refined); } AecState::InitialState::InitialState(const EchoCanceller3Config& config) diff --git a/modules/audio_processing/aec3/refined_filter_update_gain.cc b/modules/audio_processing/aec3/refined_filter_update_gain.cc index 138329ad38..db5203d542 100644 --- a/modules/audio_processing/aec3/refined_filter_update_gain.cc +++ b/modules/audio_processing/aec3/refined_filter_update_gain.cc @@ -73,6 +73,7 @@ void RefinedFilterUpdateGain::Compute( rtc::ArrayView erl, size_t size_partitions, bool saturated_capture_signal, + bool disallow_leakage_diverged, FftData* gain_fft) { RTC_DCHECK(gain_fft); // Introducing shorter notation to improve readability. @@ -125,7 +126,7 @@ void RefinedFilterUpdateGain::Compute( // H_error = H_error + factor * erl. for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) { - if (E2_coarse[k] >= E2_refined[k]) { + if (E2_refined[k] <= E2_coarse[k] || disallow_leakage_diverged) { H_error_[k] += current_config_.leakage_converged * erl[k]; } else { H_error_[k] += current_config_.leakage_diverged * erl[k]; diff --git a/modules/audio_processing/aec3/refined_filter_update_gain.h b/modules/audio_processing/aec3/refined_filter_update_gain.h index 5730979567..ae4fe84df5 100644 --- a/modules/audio_processing/aec3/refined_filter_update_gain.h +++ b/modules/audio_processing/aec3/refined_filter_update_gain.h @@ -51,6 +51,7 @@ class RefinedFilterUpdateGain { rtc::ArrayView erl, size_t size_partitions, bool saturated_capture_signal, + bool disallow_leakage_diverged, FftData* gain_fft); // Sets a new config. diff --git a/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc b/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc index 2393fddd6f..6fce858f16 100644 --- a/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc +++ b/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc @@ -196,7 +196,8 @@ void RunFilterUpdateTest(int num_blocks_to_process, std::array erl; ComputeErl(optimization, H2[0], erl); refined_gain.Compute(render_power, render_signal_analyzer, output[0], erl, - refined_filter.SizePartitions(), saturation, &G); + refined_filter.SizePartitions(), saturation, false, + &G); refined_filter.Adapt(*render_delay_buffer->GetRenderBuffer(), G, &h[0]); // Update the delay. @@ -247,7 +248,7 @@ TEST(RefinedFilterUpdateGainDeathTest, NullDataOutputGain) { erl.fill(0.f); EXPECT_DEATH( gain.Compute(render_power, analyzer, output, erl, - config.filter.refined.length_blocks, false, nullptr), + config.filter.refined.length_blocks, false, false, nullptr), ""); } diff --git a/modules/audio_processing/aec3/subtractor.cc b/modules/audio_processing/aec3/subtractor.cc index d15229934f..d10e4ffc52 100644 --- a/modules/audio_processing/aec3/subtractor.cc +++ b/modules/audio_processing/aec3/subtractor.cc @@ -19,11 +19,17 @@ #include "modules/audio_processing/logging/apm_data_dumper.h" #include "rtc_base/checks.h" #include "rtc_base/numerics/safe_minmax.h" +#include "system_wrappers/include/field_trial.h" namespace webrtc { namespace { +bool UseCoarseFilterResetHangover() { + return !field_trial::IsEnabled( + "WebRTC-Aec3CoarseFilterResetHangoverKillSwitch"); +} + void PredictionError(const Aec3Fft& fft, const FftData& S, rtc::ArrayView y, @@ -66,12 +72,14 @@ Subtractor::Subtractor(const EchoCanceller3Config& config, optimization_(optimization), config_(config), num_capture_channels_(num_capture_channels), + use_coarse_filter_reset_hangover_(UseCoarseFilterResetHangover()), refined_filters_(num_capture_channels_), coarse_filter_(num_capture_channels_), refined_gains_(num_capture_channels_), coarse_gains_(num_capture_channels_), filter_misadjustment_estimators_(num_capture_channels_), poor_coarse_filter_counters_(num_capture_channels_, 0), + coarse_filter_reset_hangover_(num_capture_channels_, 0), refined_frequency_responses_( num_capture_channels_, std::vector>( @@ -228,11 +236,19 @@ void Subtractor::Process(const RenderBuffer& render_buffer, // Update the refined filter. if (!refined_filters_adjusted) { + // Do not allow the performance of the coarse filter to affect the + // adaptation speed of the refined filter just after the coarse filter has + // been reset. + const bool disallow_leakage_diverged = + coarse_filter_reset_hangover_[ch] > 0 && + use_coarse_filter_reset_hangover_; + std::array erl; ComputeErl(optimization_, refined_frequency_responses_[ch], erl); refined_gains_[ch]->Compute(X2_refined, render_signal_analyzer, output, erl, refined_filters_[ch]->SizePartitions(), - aec_state.SaturatedCapture(), &G); + aec_state.SaturatedCapture(), + disallow_leakage_diverged, &G); } else { G.re.fill(0.f); G.im.fill(0.f); @@ -256,6 +272,8 @@ void Subtractor::Process(const RenderBuffer& render_buffer, coarse_gains_[ch]->Compute(X2_coarse, render_signal_analyzer, E_coarse, coarse_filter_[ch]->SizePartitions(), aec_state.SaturatedCapture(), &G); + coarse_filter_reset_hangover_[ch] = + std::max(coarse_filter_reset_hangover_[ch] - 1, 0); } else { poor_coarse_filter_counters_[ch] = 0; coarse_filter_[ch]->SetFilter(refined_filters_[ch]->SizePartitions(), @@ -263,6 +281,8 @@ void Subtractor::Process(const RenderBuffer& render_buffer, coarse_gains_[ch]->Compute(X2_coarse, render_signal_analyzer, E_refined, coarse_filter_[ch]->SizePartitions(), aec_state.SaturatedCapture(), &G); + coarse_filter_reset_hangover_[ch] = + config_.filter.coarse_reset_hangover_blocks; } coarse_filter_[ch]->Adapt(render_buffer, G); diff --git a/modules/audio_processing/aec3/subtractor.h b/modules/audio_processing/aec3/subtractor.h index 42ca3729ca..560f6568eb 100644 --- a/modules/audio_processing/aec3/subtractor.h +++ b/modules/audio_processing/aec3/subtractor.h @@ -120,6 +120,7 @@ class Subtractor { const Aec3Optimization optimization_; const EchoCanceller3Config config_; const size_t num_capture_channels_; + const bool use_coarse_filter_reset_hangover_; std::vector> refined_filters_; std::vector> coarse_filter_; @@ -127,6 +128,7 @@ class Subtractor { std::vector> coarse_gains_; std::vector filter_misadjustment_estimators_; std::vector poor_coarse_filter_counters_; + std::vector coarse_filter_reset_hangover_; std::vector>> refined_frequency_responses_; std::vector> refined_impulse_responses_; diff --git a/resources/audio_processing/output_data_float.pb.sha1 b/resources/audio_processing/output_data_float.pb.sha1 index 6c3ab91815..c895b963f4 100644 --- a/resources/audio_processing/output_data_float.pb.sha1 +++ b/resources/audio_processing/output_data_float.pb.sha1 @@ -1 +1 @@ -ed1172c80a1a001a8aa7ac0680a99018cbb7d278 \ No newline at end of file +365a02046fdb30357d649e73766d2f6eb2b33677 \ No newline at end of file diff --git a/resources/audio_processing/output_data_float_avx2.pb.sha1 b/resources/audio_processing/output_data_float_avx2.pb.sha1 index 2d4ad0c141..12ec621bba 100644 --- a/resources/audio_processing/output_data_float_avx2.pb.sha1 +++ b/resources/audio_processing/output_data_float_avx2.pb.sha1 @@ -1 +1 @@ -a1dd718a6882bf8033a934e5beec73086cc91240 \ No newline at end of file +847035cbe0bc7ee0620c32fa5ac857cc5b2c7ec4 \ No newline at end of file