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/*
* 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/aec3/echo_canceller3.h"
#include <algorithm>
#include <utility>
#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/atomic_ops.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
enum class EchoCanceller3ApiCall { kCapture, kRender };
bool DetectSaturation(rtc::ArrayView<const float> y) {
for (auto y_k : y) {
if (y_k >= 32700.0f || y_k <= -32700.0f) {
return true;
}
}
return false;
}
// Method for adjusting config parameter dependencies..
EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
EchoCanceller3Config adjusted_cfg = config;
if (field_trial::IsEnabled("WebRTC-Aec3ShortHeadroomKillSwitch")) {
// Two blocks headroom.
adjusted_cfg.delay.delay_headroom_samples = kBlockSize * 2;
}
return adjusted_cfg;
}
void FillSubFrameView(AudioBuffer* frame,
size_t sub_frame_index,
std::vector<rtc::ArrayView<float>>* sub_frame_view) {
RTC_DCHECK_GE(1, sub_frame_index);
RTC_DCHECK_LE(0, sub_frame_index);
RTC_DCHECK_EQ(frame->num_bands(), sub_frame_view->size());
for (size_t k = 0; k < sub_frame_view->size(); ++k) {
(*sub_frame_view)[k] = rtc::ArrayView<float>(
&frame->split_bands_f(0)[k][sub_frame_index * kSubFrameLength],
kSubFrameLength);
}
}
void FillSubFrameView(std::vector<std::vector<float>>* frame,
size_t sub_frame_index,
std::vector<rtc::ArrayView<float>>* sub_frame_view) {
RTC_DCHECK_GE(1, sub_frame_index);
RTC_DCHECK_EQ(frame->size(), sub_frame_view->size());
for (size_t k = 0; k < frame->size(); ++k) {
(*sub_frame_view)[k] = rtc::ArrayView<float>(
&(*frame)[k][sub_frame_index * kSubFrameLength], kSubFrameLength);
}
}
void ProcessCaptureFrameContent(
AudioBuffer* capture,
bool level_change,
bool saturated_microphone_signal,
size_t sub_frame_index,
FrameBlocker* capture_blocker,
BlockFramer* output_framer,
BlockProcessor* block_processor,
std::vector<std::vector<float>>* block,
std::vector<rtc::ArrayView<float>>* sub_frame_view) {
FillSubFrameView(capture, sub_frame_index, sub_frame_view);
capture_blocker->InsertSubFrameAndExtractBlock(*sub_frame_view, block);
block_processor->ProcessCapture(level_change, saturated_microphone_signal,
block);
output_framer->InsertBlockAndExtractSubFrame(*block, sub_frame_view);
}
void ProcessRemainingCaptureFrameContent(
bool level_change,
bool saturated_microphone_signal,
FrameBlocker* capture_blocker,
BlockFramer* output_framer,
BlockProcessor* block_processor,
std::vector<std::vector<float>>* block) {
if (!capture_blocker->IsBlockAvailable()) {
return;
}
capture_blocker->ExtractBlock(block);
block_processor->ProcessCapture(level_change, saturated_microphone_signal,
block);
output_framer->InsertBlock(*block);
}
void BufferRenderFrameContent(
std::vector<std::vector<float>>* render_frame,
size_t sub_frame_index,
FrameBlocker* render_blocker,
BlockProcessor* block_processor,
std::vector<std::vector<float>>* block,
std::vector<rtc::ArrayView<float>>* sub_frame_view) {
FillSubFrameView(render_frame, sub_frame_index, sub_frame_view);
render_blocker->InsertSubFrameAndExtractBlock(*sub_frame_view, block);
block_processor->BufferRender(*block);
}
void BufferRemainingRenderFrameContent(FrameBlocker* render_blocker,
BlockProcessor* block_processor,
std::vector<std::vector<float>>* block) {
if (!render_blocker->IsBlockAvailable()) {
return;
}
render_blocker->ExtractBlock(block);
block_processor->BufferRender(*block);
}
void CopyBufferIntoFrame(AudioBuffer* buffer,
size_t num_bands,
size_t frame_length,
std::vector<std::vector<float>>* frame) {
RTC_DCHECK_EQ(num_bands, frame->size());
RTC_DCHECK_EQ(frame_length, (*frame)[0].size());
for (size_t k = 0; k < num_bands; ++k) {
rtc::ArrayView<float> buffer_view(&buffer->split_bands_f(0)[k][0],
frame_length);
std::copy(buffer_view.begin(), buffer_view.end(), (*frame)[k].begin());
}
}
// [B,A] = butter(2,100/4000,'high')
const CascadedBiQuadFilter::BiQuadCoefficients
kHighPassFilterCoefficients_8kHz = {{0.94598f, -1.89195f, 0.94598f},
{-1.88903f, 0.89487f}};
const int kNumberOfHighPassBiQuads_8kHz = 1;
// [B,A] = butter(2,100/8000,'high')
const CascadedBiQuadFilter::BiQuadCoefficients
kHighPassFilterCoefficients_16kHz = {{0.97261f, -1.94523f, 0.97261f},
{-1.94448f, 0.94598f}};
const int kNumberOfHighPassBiQuads_16kHz = 1;
} // namespace
class EchoCanceller3::RenderWriter {
public:
RenderWriter(ApmDataDumper* data_dumper,
SwapQueue<std::vector<std::vector<float>>,
Aec3RenderQueueItemVerifier>* render_transfer_queue,
std::unique_ptr<CascadedBiQuadFilter> render_highpass_filter,
int sample_rate_hz,
int frame_length,
int num_bands);
~RenderWriter();
void Insert(AudioBuffer* input);
private:
ApmDataDumper* data_dumper_;
const int sample_rate_hz_;
const size_t frame_length_;
const int num_bands_;
std::unique_ptr<CascadedBiQuadFilter> render_highpass_filter_;
std::vector<std::vector<float>> render_queue_input_frame_;
SwapQueue<std::vector<std::vector<float>>, Aec3RenderQueueItemVerifier>*
render_transfer_queue_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderWriter);
};
EchoCanceller3::RenderWriter::RenderWriter(
ApmDataDumper* data_dumper,
SwapQueue<std::vector<std::vector<float>>, Aec3RenderQueueItemVerifier>*
render_transfer_queue,
std::unique_ptr<CascadedBiQuadFilter> render_highpass_filter,
int sample_rate_hz,
int frame_length,
int num_bands)
: data_dumper_(data_dumper),
sample_rate_hz_(sample_rate_hz),
frame_length_(frame_length),
num_bands_(num_bands),
render_highpass_filter_(std::move(render_highpass_filter)),
render_queue_input_frame_(num_bands_,
std::vector<float>(frame_length_, 0.f)),
render_transfer_queue_(render_transfer_queue) {
RTC_DCHECK(data_dumper);
}
EchoCanceller3::RenderWriter::~RenderWriter() = default;
void EchoCanceller3::RenderWriter::Insert(AudioBuffer* input) {
RTC_DCHECK_EQ(1, input->num_channels());
RTC_DCHECK_EQ(frame_length_, input->num_frames_per_band());
RTC_DCHECK_EQ(num_bands_, input->num_bands());
// TODO(bugs.webrtc.org/8759) Temporary work-around.
if (num_bands_ != static_cast<int>(input->num_bands()))
return;
data_dumper_->DumpWav("aec3_render_input", frame_length_,
&input->split_bands_f(0)[0][0],
LowestBandRate(sample_rate_hz_), 1);
CopyBufferIntoFrame(input, num_bands_, frame_length_,
&render_queue_input_frame_);
if (render_highpass_filter_) {
render_highpass_filter_->Process(render_queue_input_frame_[0]);
}
static_cast<void>(render_transfer_queue_->Insert(&render_queue_input_frame_));
}
int EchoCanceller3::instance_count_ = 0;
EchoCanceller3::EchoCanceller3(const EchoCanceller3Config& config,
int sample_rate_hz,
bool use_highpass_filter)
: EchoCanceller3(
AdjustConfig(config),
sample_rate_hz,
use_highpass_filter,
std::unique_ptr<BlockProcessor>(
BlockProcessor::Create(AdjustConfig(config), sample_rate_hz))) {}
EchoCanceller3::EchoCanceller3(const EchoCanceller3Config& config,
int sample_rate_hz,
bool use_highpass_filter,
std::unique_ptr<BlockProcessor> block_processor)
: data_dumper_(
new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
config_(config),
sample_rate_hz_(sample_rate_hz),
num_bands_(NumBandsForRate(sample_rate_hz_)),
frame_length_(rtc::CheckedDivExact(LowestBandRate(sample_rate_hz_), 100)),
output_framer_(num_bands_),
capture_blocker_(num_bands_),
render_blocker_(num_bands_),
render_transfer_queue_(
kRenderTransferQueueSizeFrames,
std::vector<std::vector<float>>(
num_bands_,
std::vector<float>(frame_length_, 0.f)),
Aec3RenderQueueItemVerifier(num_bands_, frame_length_)),
block_processor_(std::move(block_processor)),
render_queue_output_frame_(num_bands_,
std::vector<float>(frame_length_, 0.f)),
block_(num_bands_, std::vector<float>(kBlockSize, 0.f)),
sub_frame_view_(num_bands_),
block_delay_buffer_(num_bands_,
frame_length_,
config_.delay.fixed_capture_delay_samples) {
RTC_DCHECK(ValidFullBandRate(sample_rate_hz_));
std::unique_ptr<CascadedBiQuadFilter> render_highpass_filter;
if (use_highpass_filter) {
render_highpass_filter.reset(new CascadedBiQuadFilter(
sample_rate_hz_ == 8000 ? kHighPassFilterCoefficients_8kHz
: kHighPassFilterCoefficients_16kHz,
sample_rate_hz_ == 8000 ? kNumberOfHighPassBiQuads_8kHz
: kNumberOfHighPassBiQuads_16kHz));
capture_highpass_filter_.reset(new CascadedBiQuadFilter(
sample_rate_hz_ == 8000 ? kHighPassFilterCoefficients_8kHz
: kHighPassFilterCoefficients_16kHz,
sample_rate_hz_ == 8000 ? kNumberOfHighPassBiQuads_8kHz
: kNumberOfHighPassBiQuads_16kHz));
}
render_writer_.reset(
new RenderWriter(data_dumper_.get(), &render_transfer_queue_,
std::move(render_highpass_filter), sample_rate_hz_,
frame_length_, num_bands_));
RTC_DCHECK_EQ(num_bands_, std::max(sample_rate_hz_, 16000) / 16000);
RTC_DCHECK_GE(kMaxNumBands, num_bands_);
}
EchoCanceller3::~EchoCanceller3() = default;
void EchoCanceller3::AnalyzeRender(AudioBuffer* render) {
RTC_DCHECK_RUNS_SERIALIZED(&render_race_checker_);
RTC_DCHECK(render);
data_dumper_->DumpRaw("aec3_call_order",
static_cast<int>(EchoCanceller3ApiCall::kRender));
return render_writer_->Insert(render);
}
void EchoCanceller3::AnalyzeCapture(AudioBuffer* capture) {
RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
RTC_DCHECK(capture);
data_dumper_->DumpWav("aec3_capture_analyze_input", capture->num_frames(),
capture->channels_f()[0], sample_rate_hz_, 1);
saturated_microphone_signal_ = false;
for (size_t k = 0; k < capture->num_channels(); ++k) {
saturated_microphone_signal_ |=
DetectSaturation(rtc::ArrayView<const float>(capture->channels_f()[k],
capture->num_frames()));
if (saturated_microphone_signal_) {
break;
}
}
}
void EchoCanceller3::ProcessCapture(AudioBuffer* capture, bool level_change) {
RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
RTC_DCHECK(capture);
RTC_DCHECK_EQ(1u, capture->num_channels());
RTC_DCHECK_EQ(num_bands_, capture->num_bands());
RTC_DCHECK_EQ(frame_length_, capture->num_frames_per_band());
data_dumper_->DumpRaw("aec3_call_order",
static_cast<int>(EchoCanceller3ApiCall::kCapture));
// Report capture call in the metrics and periodically update API call
// metrics.
api_call_metrics_.ReportCaptureCall();
// Optionally delay the capture signal.
if (config_.delay.fixed_capture_delay_samples > 0) {
block_delay_buffer_.DelaySignal(capture);
}
rtc::ArrayView<float> capture_lower_band =
rtc::ArrayView<float>(&capture->split_bands_f(0)[0][0], frame_length_);
data_dumper_->DumpWav("aec3_capture_input", capture_lower_band,
LowestBandRate(sample_rate_hz_), 1);
EmptyRenderQueue();
if (capture_highpass_filter_) {
capture_highpass_filter_->Process(capture_lower_band);
}
ProcessCaptureFrameContent(
capture, level_change, saturated_microphone_signal_, 0, &capture_blocker_,
&output_framer_, block_processor_.get(), &block_, &sub_frame_view_);
if (sample_rate_hz_ != 8000) {
ProcessCaptureFrameContent(
capture, level_change, saturated_microphone_signal_, 1,
&capture_blocker_, &output_framer_, block_processor_.get(), &block_,
&sub_frame_view_);
}
ProcessRemainingCaptureFrameContent(
level_change, saturated_microphone_signal_, &capture_blocker_,
&output_framer_, block_processor_.get(), &block_);
data_dumper_->DumpWav("aec3_capture_output", frame_length_,
&capture->split_bands_f(0)[0][0],
LowestBandRate(sample_rate_hz_), 1);
}
EchoControl::Metrics EchoCanceller3::GetMetrics() const {
RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
Metrics metrics;
block_processor_->GetMetrics(&metrics);
return metrics;
}
void EchoCanceller3::SetAudioBufferDelay(size_t delay_ms) {
RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
block_processor_->SetAudioBufferDelay(delay_ms);
}
void EchoCanceller3::EmptyRenderQueue() {
RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
bool frame_to_buffer =
render_transfer_queue_.Remove(&render_queue_output_frame_);
while (frame_to_buffer) {
// Report render call in the metrics.
api_call_metrics_.ReportRenderCall();
BufferRenderFrameContent(&render_queue_output_frame_, 0, &render_blocker_,
block_processor_.get(), &block_, &sub_frame_view_);
if (sample_rate_hz_ != 8000) {
BufferRenderFrameContent(&render_queue_output_frame_, 1, &render_blocker_,
block_processor_.get(), &block_,
&sub_frame_view_);
}
BufferRemainingRenderFrameContent(&render_blocker_, block_processor_.get(),
&block_);
frame_to_buffer =
render_transfer_queue_.Remove(&render_queue_output_frame_);
}
}
} // namespace webrtc