blob: 9b400ff11194eae525bc030219deb9f33e0ce843 [file] [log] [blame]
/*
* Copyright (c) 2017 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/aec3/render_delay_controller.h"
#include <algorithm>
#include <memory>
#include <string>
#include <vector>
#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/aec3/echo_path_delay_estimator.h"
#include "modules/audio_processing/aec3/render_delay_controller_metrics.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/atomicops.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
namespace {
class RenderDelayControllerImpl final : public RenderDelayController {
public:
RenderDelayControllerImpl(
const AudioProcessing::Config::EchoCanceller3& config,
int sample_rate_hz);
~RenderDelayControllerImpl() override;
void Reset() override;
void SetDelay(size_t render_delay) override;
size_t GetDelay(const DownsampledRenderBuffer& render_buffer,
rtc::ArrayView<const float> capture) override;
rtc::Optional<size_t> AlignmentHeadroomSamples() const override {
return headroom_samples_;
}
private:
static int instance_count_;
std::unique_ptr<ApmDataDumper> data_dumper_;
size_t delay_ = kMinEchoPathDelayBlocks;
EchoPathDelayEstimator delay_estimator_;
size_t blocks_since_last_delay_estimate_ = 300000;
int echo_path_delay_samples_ = kMinEchoPathDelayBlocks * kBlockSize;
size_t align_call_counter_ = 0;
rtc::Optional<size_t> headroom_samples_;
std::vector<float> capture_delay_buffer_;
int capture_delay_buffer_index_ = 0;
RenderDelayControllerMetrics metrics_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderDelayControllerImpl);
};
size_t ComputeNewBufferDelay(size_t current_delay,
size_t echo_path_delay_samples) {
// The below division is not exact and the truncation is intended.
const int echo_path_delay_blocks = echo_path_delay_samples / kBlockSize;
constexpr int kDelayHeadroomBlocks = 1;
// Compute the buffer delay increase required to achieve the desired latency.
size_t new_delay = std::max(echo_path_delay_blocks - kDelayHeadroomBlocks, 0);
// Add hysteresis.
if (new_delay == current_delay + 1) {
new_delay = current_delay;
}
return new_delay;
}
int RenderDelayControllerImpl::instance_count_ = 0;
RenderDelayControllerImpl::RenderDelayControllerImpl(
const AudioProcessing::Config::EchoCanceller3& config,
int sample_rate_hz)
: data_dumper_(
new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
delay_estimator_(data_dumper_.get(), config),
capture_delay_buffer_(kBlockSize * (kMaxApiCallsJitterBlocks + 2), 0.f) {
RTC_DCHECK(ValidFullBandRate(sample_rate_hz));
}
RenderDelayControllerImpl::~RenderDelayControllerImpl() = default;
void RenderDelayControllerImpl::Reset() {
delay_ = kMinEchoPathDelayBlocks;
blocks_since_last_delay_estimate_ = 300000;
echo_path_delay_samples_ = delay_ * kBlockSize;
align_call_counter_ = 0;
headroom_samples_ = rtc::Optional<size_t>();
std::fill(capture_delay_buffer_.begin(), capture_delay_buffer_.end(), 0.f);
delay_estimator_.Reset();
}
void RenderDelayControllerImpl::SetDelay(size_t render_delay) {
if (delay_ != render_delay) {
// If a the delay set does not match the actual delay, reset the delay
// controller.
Reset();
delay_ = render_delay;
}
}
size_t RenderDelayControllerImpl::GetDelay(
const DownsampledRenderBuffer& render_buffer,
rtc::ArrayView<const float> capture) {
RTC_DCHECK_EQ(kBlockSize, capture.size());
++align_call_counter_;
// Estimate the delay with a delayed capture signal in order to catch
// noncausal delays.
RTC_DCHECK_LT(capture_delay_buffer_index_ + kBlockSize - 1,
capture_delay_buffer_.size());
const rtc::Optional<size_t> echo_path_delay_samples_shifted =
delay_estimator_.EstimateDelay(
render_buffer,
rtc::ArrayView<const float>(
&capture_delay_buffer_[capture_delay_buffer_index_], kBlockSize));
std::copy(capture.begin(), capture.end(),
capture_delay_buffer_.begin() + capture_delay_buffer_index_);
capture_delay_buffer_index_ =
(capture_delay_buffer_index_ + kBlockSize) % capture_delay_buffer_.size();
if (echo_path_delay_samples_shifted) {
blocks_since_last_delay_estimate_ = 0;
// Correct for the capture signal delay.
const int echo_path_delay_samples_corrected =
static_cast<int>(*echo_path_delay_samples_shifted) -
static_cast<int>(capture_delay_buffer_.size());
echo_path_delay_samples_ = std::max(0, echo_path_delay_samples_corrected);
// Compute and set new render delay buffer delay.
const size_t new_delay =
ComputeNewBufferDelay(delay_, echo_path_delay_samples_);
if (align_call_counter_ > kNumBlocksPerSecond) {
delay_ = new_delay;
// Update render delay buffer headroom.
if (echo_path_delay_samples_corrected >= 0) {
const int headroom = echo_path_delay_samples_ - delay_ * kBlockSize;
RTC_DCHECK_LE(0, headroom);
headroom_samples_ = rtc::Optional<size_t>(headroom);
} else {
headroom_samples_ = rtc::Optional<size_t>();
}
}
metrics_.Update(rtc::Optional<size_t>(echo_path_delay_samples_), delay_);
} else {
metrics_.Update(rtc::Optional<size_t>(), delay_);
}
data_dumper_->DumpRaw("aec3_render_delay_controller_delay", 1,
&echo_path_delay_samples_);
data_dumper_->DumpRaw("aec3_render_delay_controller_buffer_delay", delay_);
return delay_;
}
} // namespace
RenderDelayController* RenderDelayController::Create(
const AudioProcessing::Config::EchoCanceller3& config,
int sample_rate_hz) {
return new RenderDelayControllerImpl(config, sample_rate_hz);
}
} // namespace webrtc