tg2sip/webrtc_dsp/modules/audio_processing/aec3/echo_canceller3.cc

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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
/*
* 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/atomicops.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;
}
bool UseShortDelayEstimatorWindow() {
return field_trial::IsEnabled("WebRTC-Aec3UseShortDelayEstimatorWindow");
}
bool EnableReverbBasedOnRender() {
return !field_trial::IsEnabled("WebRTC-Aec3ReverbBasedOnRenderKillSwitch");
}
bool EnableReverbModelling() {
return !field_trial::IsEnabled("WebRTC-Aec3ReverbModellingKillSwitch");
}
bool EnableUnityInitialRampupGain() {
return field_trial::IsEnabled("WebRTC-Aec3EnableUnityInitialRampupGain");
}
bool EnableUnityNonZeroRampupGain() {
return field_trial::IsEnabled("WebRTC-Aec3EnableUnityNonZeroRampupGain");
}
bool EnableLongReverb() {
return field_trial::IsEnabled("WebRTC-Aec3ShortReverbKillSwitch");
}
bool EnableNewFilterParams() {
return !field_trial::IsEnabled("WebRTC-Aec3NewFilterParamsKillSwitch");
}
bool EnableLegacyDominantNearend() {
return field_trial::IsEnabled("WebRTC-Aec3EnableLegacyDominantNearend");
}
bool UseLegacyNormalSuppressorTuning() {
return field_trial::IsEnabled("WebRTC-Aec3UseLegacyNormalSuppressorTuning");
}
bool ActivateStationarityProperties() {
return field_trial::IsEnabled("WebRTC-Aec3UseStationarityProperties");
}
bool ActivateStationarityPropertiesAtInit() {
return field_trial::IsEnabled("WebRTC-Aec3UseStationarityPropertiesAtInit");
}
bool EnableNewRenderBuffering() {
return !field_trial::IsEnabled("WebRTC-Aec3NewRenderBufferingKillSwitch");
}
bool UseEarlyDelayDetection() {
return !field_trial::IsEnabled("WebRTC-Aec3EarlyDelayDetectionKillSwitch");
}
// Method for adjusting config parameter dependencies..
EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
EchoCanceller3Config adjusted_cfg = config;
const EchoCanceller3Config default_cfg;
if (!EnableReverbModelling()) {
adjusted_cfg.ep_strength.default_len = 0.f;
}
if (UseShortDelayEstimatorWindow()) {
adjusted_cfg.delay.num_filters =
std::min(adjusted_cfg.delay.num_filters, static_cast<size_t>(5));
}
bool use_new_render_buffering =
EnableNewRenderBuffering() && config.buffering.use_new_render_buffering;
// Old render buffering needs one more filter to cover the same delay.
if (!use_new_render_buffering) {
adjusted_cfg.delay.num_filters += 1;
}
if (EnableReverbBasedOnRender() == false) {
adjusted_cfg.ep_strength.reverb_based_on_render = false;
}
if (!EnableNewFilterParams()) {
adjusted_cfg.filter.main.leakage_diverged = 0.01f;
adjusted_cfg.filter.main.error_floor = 0.1f;
adjusted_cfg.filter.main.error_ceil = 1E10f;
adjusted_cfg.filter.main_initial.error_ceil = 1E10f;
}
if (EnableUnityInitialRampupGain() &&
adjusted_cfg.echo_removal_control.gain_rampup.initial_gain ==
default_cfg.echo_removal_control.gain_rampup.initial_gain) {
adjusted_cfg.echo_removal_control.gain_rampup.initial_gain = 1.f;
}
if (EnableUnityNonZeroRampupGain() &&
adjusted_cfg.echo_removal_control.gain_rampup.first_non_zero_gain ==
default_cfg.echo_removal_control.gain_rampup.first_non_zero_gain) {
adjusted_cfg.echo_removal_control.gain_rampup.first_non_zero_gain = 1.f;
}
if (EnableLongReverb()) {
adjusted_cfg.ep_strength.default_len = 0.88f;
}
if (EnableLegacyDominantNearend()) {
adjusted_cfg.suppressor.nearend_tuning =
EchoCanceller3Config::Suppressor::Tuning(
EchoCanceller3Config::Suppressor::MaskingThresholds(.2f, .3f, .3f),
EchoCanceller3Config::Suppressor::MaskingThresholds(.07f, .1f, .3f),
2.0f, 0.25f);
}
if (UseLegacyNormalSuppressorTuning()) {
adjusted_cfg.suppressor.normal_tuning =
EchoCanceller3Config::Suppressor::Tuning(
EchoCanceller3Config::Suppressor::MaskingThresholds(.2f, .3f, .3f),
EchoCanceller3Config::Suppressor::MaskingThresholds(.07f, .1f, .3f),
2.0f, 0.25f);
adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold = 10.f;
adjusted_cfg.suppressor.dominant_nearend_detection.snr_threshold = 10.f;
adjusted_cfg.suppressor.dominant_nearend_detection.hold_duration = 25;
}
if (ActivateStationarityProperties()) {
adjusted_cfg.echo_audibility.use_stationary_properties = true;
}
if (ActivateStationarityPropertiesAtInit()) {
adjusted_cfg.echo_audibility.use_stationarity_properties_at_init = true;
}
if (!UseEarlyDelayDetection()) {
adjusted_cfg.delay.delay_selection_thresholds = {25, 25};
}
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>(
EnableNewRenderBuffering() &&
config.buffering.use_new_render_buffering
? BlockProcessor::Create2(AdjustConfig(config),
sample_rate_hz)
: 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));
// 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) {
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