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libtgvoip/webrtc_dsp/modules/audio_processing/residual_echo_detector.cc
Grishka 5caaaafa42 Updated WebRTC APM
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
2018-11-23 04:02:53 +03:00

216 lines
8.7 KiB
C++

/*
* 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 "absl/types/optional.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/atomicops.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 {
int ResidualEchoDetector::instance_count_ = 0;
ResidualEchoDetector::ResidualEchoDetector()
: data_dumper_(
new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
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 absl::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;
}
void EchoDetector::PackRenderAudioBuffer(AudioBuffer* audio,
std::vector<float>* packed_buffer) {
packed_buffer->clear();
packed_buffer->insert(packed_buffer->end(), audio->channels_f()[0],
audio->channels_f()[0] + audio->num_frames());
}
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