| /* |
| * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "modules/audio_processing/residual_echo_detector.h" |
| |
| #include <algorithm> |
| #include <numeric> |
| #include <optional> |
| |
| #include "modules/audio_processing/logging/apm_data_dumper.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "system_wrappers/include/metrics.h" |
| |
| namespace { |
| |
| float Power(rtc::ArrayView<const float> input) { |
| if (input.empty()) { |
| return 0.f; |
| } |
| return std::inner_product(input.begin(), input.end(), input.begin(), 0.f) / |
| input.size(); |
| } |
| |
| constexpr size_t kLookbackFrames = 650; |
| // TODO(ivoc): Verify the size of this buffer. |
| constexpr size_t kRenderBufferSize = 30; |
| constexpr float kAlpha = 0.001f; |
| // 10 seconds of data, updated every 10 ms. |
| constexpr size_t kAggregationBufferSize = 10 * 100; |
| |
| } // namespace |
| |
| namespace webrtc { |
| |
| std::atomic<int> ResidualEchoDetector::instance_count_(0); |
| |
| ResidualEchoDetector::ResidualEchoDetector() |
| : data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)), |
| render_buffer_(kRenderBufferSize), |
| render_power_(kLookbackFrames), |
| render_power_mean_(kLookbackFrames), |
| render_power_std_dev_(kLookbackFrames), |
| covariances_(kLookbackFrames), |
| recent_likelihood_max_(kAggregationBufferSize) {} |
| |
| ResidualEchoDetector::~ResidualEchoDetector() = default; |
| |
| void ResidualEchoDetector::AnalyzeRenderAudio( |
| rtc::ArrayView<const float> render_audio) { |
| // Dump debug data assuming 48 kHz sample rate (if this assumption is not |
| // valid the dumped audio will need to be converted offline accordingly). |
| data_dumper_->DumpWav("ed_render", render_audio.size(), render_audio.data(), |
| 48000, 1); |
| |
| if (render_buffer_.Size() == 0) { |
| frames_since_zero_buffer_size_ = 0; |
| } else if (frames_since_zero_buffer_size_ >= kRenderBufferSize) { |
| // This can happen in a few cases: at the start of a call, due to a glitch |
| // or due to clock drift. The excess capture value will be ignored. |
| // TODO(ivoc): Include how often this happens in APM stats. |
| render_buffer_.Pop(); |
| frames_since_zero_buffer_size_ = 0; |
| } |
| ++frames_since_zero_buffer_size_; |
| float power = Power(render_audio); |
| render_buffer_.Push(power); |
| } |
| |
| void ResidualEchoDetector::AnalyzeCaptureAudio( |
| rtc::ArrayView<const float> capture_audio) { |
| // Dump debug data assuming 48 kHz sample rate (if this assumption is not |
| // valid the dumped audio will need to be converted offline accordingly). |
| data_dumper_->DumpWav("ed_capture", capture_audio.size(), |
| capture_audio.data(), 48000, 1); |
| |
| if (first_process_call_) { |
| // On the first process call (so the start of a call), we must flush the |
| // render buffer, otherwise the render data will be delayed. |
| render_buffer_.Clear(); |
| first_process_call_ = false; |
| } |
| |
| // Get the next render value. |
| const std::optional<float> buffered_render_power = render_buffer_.Pop(); |
| if (!buffered_render_power) { |
| // This can happen in a few cases: at the start of a call, due to a glitch |
| // or due to clock drift. The excess capture value will be ignored. |
| // TODO(ivoc): Include how often this happens in APM stats. |
| return; |
| } |
| // Update the render statistics, and store the statistics in circular buffers. |
| render_statistics_.Update(*buffered_render_power); |
| RTC_DCHECK_LT(next_insertion_index_, kLookbackFrames); |
| render_power_[next_insertion_index_] = *buffered_render_power; |
| render_power_mean_[next_insertion_index_] = render_statistics_.mean(); |
| render_power_std_dev_[next_insertion_index_] = |
| render_statistics_.std_deviation(); |
| |
| // Get the next capture value, update capture statistics and add the relevant |
| // values to the buffers. |
| const float capture_power = Power(capture_audio); |
| capture_statistics_.Update(capture_power); |
| const float capture_mean = capture_statistics_.mean(); |
| const float capture_std_deviation = capture_statistics_.std_deviation(); |
| |
| // Update the covariance values and determine the new echo likelihood. |
| echo_likelihood_ = 0.f; |
| size_t read_index = next_insertion_index_; |
| |
| int best_delay = -1; |
| for (size_t delay = 0; delay < covariances_.size(); ++delay) { |
| RTC_DCHECK_LT(read_index, render_power_.size()); |
| covariances_[delay].Update(capture_power, capture_mean, |
| capture_std_deviation, render_power_[read_index], |
| render_power_mean_[read_index], |
| render_power_std_dev_[read_index]); |
| read_index = read_index > 0 ? read_index - 1 : kLookbackFrames - 1; |
| |
| if (covariances_[delay].normalized_cross_correlation() > echo_likelihood_) { |
| echo_likelihood_ = covariances_[delay].normalized_cross_correlation(); |
| best_delay = static_cast<int>(delay); |
| } |
| } |
| // This is a temporary log message to help find the underlying cause for echo |
| // likelihoods > 1.0. |
| // TODO(ivoc): Remove once the issue is resolved. |
| if (echo_likelihood_ > 1.1f) { |
| // Make sure we don't spam the log. |
| if (log_counter_ < 5 && best_delay != -1) { |
| size_t read_index = kLookbackFrames + next_insertion_index_ - best_delay; |
| if (read_index >= kLookbackFrames) { |
| read_index -= kLookbackFrames; |
| } |
| RTC_DCHECK_LT(read_index, render_power_.size()); |
| RTC_LOG_F(LS_ERROR) << "Echo detector internal state: {" |
| "Echo likelihood: " |
| << echo_likelihood_ << ", Best Delay: " << best_delay |
| << ", Covariance: " |
| << covariances_[best_delay].covariance() |
| << ", Last capture power: " << capture_power |
| << ", Capture mean: " << capture_mean |
| << ", Capture_standard deviation: " |
| << capture_std_deviation << ", Last render power: " |
| << render_power_[read_index] |
| << ", Render mean: " << render_power_mean_[read_index] |
| << ", Render standard deviation: " |
| << render_power_std_dev_[read_index] |
| << ", Reliability: " << reliability_ << "}"; |
| log_counter_++; |
| } |
| } |
| RTC_DCHECK_LT(echo_likelihood_, 1.1f); |
| |
| reliability_ = (1.0f - kAlpha) * reliability_ + kAlpha * 1.0f; |
| echo_likelihood_ *= reliability_; |
| // This is a temporary fix to prevent echo likelihood values > 1.0. |
| // TODO(ivoc): Find the root cause of this issue and fix it. |
| echo_likelihood_ = std::min(echo_likelihood_, 1.0f); |
| int echo_percentage = static_cast<int>(echo_likelihood_ * 100); |
| RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ResidualEchoDetector.EchoLikelihood", |
| echo_percentage, 0, 100, 100 /* number of bins */); |
| |
| // Update the buffer of recent likelihood values. |
| recent_likelihood_max_.Update(echo_likelihood_); |
| |
| // Update the next insertion index. |
| next_insertion_index_ = next_insertion_index_ < (kLookbackFrames - 1) |
| ? next_insertion_index_ + 1 |
| : 0; |
| } |
| |
| void ResidualEchoDetector::Initialize(int /*capture_sample_rate_hz*/, |
| int /*num_capture_channels*/, |
| int /*render_sample_rate_hz*/, |
| int /*num_render_channels*/) { |
| render_buffer_.Clear(); |
| std::fill(render_power_.begin(), render_power_.end(), 0.f); |
| std::fill(render_power_mean_.begin(), render_power_mean_.end(), 0.f); |
| std::fill(render_power_std_dev_.begin(), render_power_std_dev_.end(), 0.f); |
| render_statistics_.Clear(); |
| capture_statistics_.Clear(); |
| recent_likelihood_max_.Clear(); |
| for (auto& cov : covariances_) { |
| cov.Clear(); |
| } |
| echo_likelihood_ = 0.f; |
| next_insertion_index_ = 0; |
| reliability_ = 0.f; |
| } |
| |
| EchoDetector::Metrics ResidualEchoDetector::GetMetrics() const { |
| EchoDetector::Metrics metrics; |
| metrics.echo_likelihood = echo_likelihood_; |
| metrics.echo_likelihood_recent_max = recent_likelihood_max_.max(); |
| return metrics; |
| } |
| } // namespace webrtc |