diff --git a/CMakeLists.txt b/CMakeLists.txt
index 45586a0daa8a4f81f7227ed340ffda7ea5e0fb75..11c0dd07f65d1eb6d1c1b82aed2b1fdc69f5ac52 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -122,17 +122,6 @@ if(MSVC)
################################################################################
if("${CMAKE_VS_PLATFORM_NAME}" STREQUAL "x64")
target_include_directories(${PROJECT_NAME} PUBLIC
- "$<$<CONFIG:ReleaseLib_win32>:"
- "${CMAKE_CURRENT_SOURCE_DIR}/contrib/msvc/include/upnp;"
- "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/ffmpeg/Build/win32/x64/include;"
- "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/sndfile/src;"
- "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/openssl/include;"
- "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/asio/asio/include;"
- "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/restinio/dev;"
- "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/fmt/include;"
- "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/http_parser;"
- "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/natpmp"
- ">"
"${CMAKE_CURRENT_SOURCE_DIR}/.;"
"${CMAKE_CURRENT_SOURCE_DIR}/src;"
"${CMAKE_CURRENT_SOURCE_DIR}/src/client;"
@@ -152,6 +141,15 @@ if(MSVC)
"${CMAKE_CURRENT_SOURCE_DIR}/compat/msvc;"
"${CMAKE_CURRENT_SOURCE_DIR}/contrib/msvc;"
"${CMAKE_CURRENT_SOURCE_DIR}/contrib/msvc/include;"
+ "${CMAKE_CURRENT_SOURCE_DIR}/contrib/msvc/include/upnp;"
+ "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/ffmpeg/Build/win32/x64/include;"
+ "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/sndfile/src;"
+ "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/openssl/include;"
+ "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/asio/asio/include;"
+ "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/restinio/dev;"
+ "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/fmt/include;"
+ "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/http_parser;"
+ "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/natpmp"
"${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/msgpack-c/include;"
"${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/opendht/include;"
"${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/libarchive/libarchive;"
@@ -164,6 +162,7 @@ if(MSVC)
"${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/pjproject/third_party;"
"${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/pjproject/pjmedia/include"
"${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/speexdsp/include;"
+ "${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/webrtc-audio-processing"
)
endif()
@@ -190,6 +189,9 @@ if(MSVC)
"NOMINMAX;"
"HAVE_CONFIG_H;"
"WIN32_LEAN_AND_MEAN;"
+ "WEBRTC_WIN;"
+ "WEBRTC_AUDIO_PROCESSING_ONLY_BUILD;"
+ "WEBRTC_NS_FLOAT;"
)
endif()
@@ -274,6 +276,7 @@ if(MSVC)
${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/openssl/libcrypto.lib
${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/openssl/libssl.lib
${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/speexdsp/lib/libspeexdsp.lib
+ ${CMAKE_CURRENT_SOURCE_DIR}/contrib/build/webrtc-audio-processing/build/Release/webrtc-audio-processing.lib
/ignore:4006
"
)
diff --git a/compat/msvc/config.h b/compat/msvc/config.h
index a45cd4de226a7b38ca1b7990b15e90a890e24c5e..ed4b174ae857cf006ffef694aabd40714625660f 100644
--- a/compat/msvc/config.h
+++ b/compat/msvc/config.h
@@ -13,7 +13,7 @@ systems. This function is required for `alloca.c' support on those systems.
#define HAVE_ALLOCA 1
/* Define to 1 if you have <alloca.h> and it should be used (not on Ultrix).
-*/
+ */
#define HAVE_ALLOCA_H 1
/* Define if you have alsa */
@@ -100,6 +100,9 @@ systems. This function is required for `alloca.c' support on those systems.
/* Define if you have libspeexdsp */
#define HAVE_SPEEXDSP 1
+/* Define if you have webrtc-audio-processing */
+#define HAVE_WEBRTC_AP 1
+
/* Define to 1 if stdbool.h conforms to C99. */
#define HAVE_STDBOOL_H 1
diff --git a/src/media/audio/audiolayer.cpp b/src/media/audio/audiolayer.cpp
index 927db4896d6171e5f73dff7e24483adf90a410d4..2fa6cc4d4288c6122f9f5956f0aed293e3c65bc8 100644
--- a/src/media/audio/audiolayer.cpp
+++ b/src/media/audio/audiolayer.cpp
@@ -27,7 +27,13 @@
#include "audio/resampler.h"
#include "tonecontrol.h"
#include "client/ring_signal.h"
+
+// aec
+#if HAVE_WEBRTC_AP
+#include "echo-cancel/webrtc_echo_canceller.h"
+#else
#include "echo-cancel/null_echo_canceller.h"
+#endif
#include <ctime>
#include <algorithm>
@@ -118,12 +124,32 @@ AudioLayer::setHasNativeAEC(bool hasEAC)
void
AudioLayer::checkAEC()
{
+ std::lock_guard<std::mutex> lk(ecMutex_);
bool shouldSoftAEC = not hasNativeAEC_ and playbackStarted_ and recordStarted_;
-
if (not echoCanceller_ and shouldSoftAEC) {
- JAMI_WARN("Starting AEC");
- echoCanceller_.reset(new NullEchoCanceller(audioFormat_, audioFormat_.sample_rate / 100));
- } else if (echoCanceller_ and not shouldSoftAEC) {
+ auto nb_channels = std::min(audioFormat_.nb_channels, audioInputFormat_.nb_channels);
+ auto sample_rate = std::min(audioFormat_.sample_rate, audioInputFormat_.sample_rate);
+ if (sample_rate % 16000u != 0)
+ sample_rate = 16000u * ((sample_rate / 16000u) + 1u);
+ sample_rate = std::clamp(sample_rate, 16000u, 96000u);
+ AudioFormat format {sample_rate, nb_channels};
+ auto frame_size = sample_rate / 100u;
+ JAMI_WARN("Input {%d Hz, %d channels}",
+ audioInputFormat_.sample_rate,
+ audioInputFormat_.nb_channels);
+ JAMI_WARN("Output {%d Hz, %d channels}", audioFormat_.sample_rate, audioFormat_.nb_channels);
+ JAMI_WARN("Starting AEC {%d Hz, %d channels, %d samples/frame}",
+ sample_rate,
+ nb_channels,
+ frame_size);
+
+#if HAVE_WEBRTC_AP
+ echoCanceller_.reset(new WebRTCEchoCanceller(format, frame_size));
+#else
+ echoCanceller_.reset(new NullEchoCanceller(format, frame_size));
+#endif
+ } else if (echoCanceller_ and not shouldSoftAEC and not playbackStarted_
+ and not recordStarted_) {
JAMI_WARN("Stopping AEC");
echoCanceller_.reset();
}
@@ -209,11 +235,18 @@ AudioLayer::getToPlay(AudioFormat format, size_t writableSamples)
} else if (auto buf = bufferPool.getData(RingBufferPool::DEFAULT_ID)) {
resampled = resampler_->resample(std::move(buf), format);
} else {
+ if (echoCanceller_) {
+ auto silence = std::make_shared<AudioFrame>(format, writableSamples);
+ libav_utils::fillWithSilence(silence->pointer());
+ std::lock_guard<std::mutex> lk(ecMutex_);
+ echoCanceller_->putPlayback(silence);
+ }
break;
}
if (resampled) {
if (echoCanceller_) {
+ std::lock_guard<std::mutex> lk(ecMutex_);
echoCanceller_->putPlayback(resampled);
}
playbackQueue_->enqueue(std::move(resampled));
@@ -228,9 +261,11 @@ void
AudioLayer::putRecorded(std::shared_ptr<AudioFrame>&& frame)
{
if (echoCanceller_) {
+ std::lock_guard<std::mutex> lk(ecMutex_);
echoCanceller_->putRecorded(std::move(frame));
- while (auto rec = echoCanceller_->getProcessed())
+ while (auto rec = echoCanceller_->getProcessed()) {
mainRingBuffer_->put(std::move(rec));
+ }
} else {
mainRingBuffer_->put(std::move(frame));
}
diff --git a/src/media/audio/audiolayer.h b/src/media/audio/audiolayer.h
index 679b35610c5b284ae38440dfaac64bd411582d49..79ad74d36b24c13277d3fec0c38693f356b56b47 100644
--- a/src/media/audio/audiolayer.h
+++ b/src/media/audio/audiolayer.h
@@ -289,6 +289,7 @@ protected:
*/
std::unique_ptr<Resampler> resampler_;
+ std::mutex ecMutex_ {};
std::unique_ptr<EchoCanceller> echoCanceller_;
private:
diff --git a/src/media/audio/echo-cancel/CMakeLists.txt b/src/media/audio/echo-cancel/CMakeLists.txt
index fec1af7d8f9cf8e5df463c637ab74cb28cb07e42..93e8440fbaa4ed620880cdca7b4098f86360bd35 100644
--- a/src/media/audio/echo-cancel/CMakeLists.txt
+++ b/src/media/audio/echo-cancel/CMakeLists.txt
@@ -7,6 +7,8 @@ list (APPEND Source_Files__media__audio__echo_cancel
"${CMAKE_CURRENT_SOURCE_DIR}/null_echo_canceller.cpp"
"${CMAKE_CURRENT_SOURCE_DIR}/speex_echo_canceller.h"
"${CMAKE_CURRENT_SOURCE_DIR}/speex_echo_canceller.cpp"
+ "${CMAKE_CURRENT_SOURCE_DIR}/webrtc_echo_canceller.h"
+ "${CMAKE_CURRENT_SOURCE_DIR}/webrtc_echo_canceller.cpp"
)
set (Source_Files__media__audio__echo_cancel ${Source_Files__media__audio__echo_cancel} PARENT_SCOPE)
\ No newline at end of file
diff --git a/src/media/audio/echo-cancel/Makefile.am b/src/media/audio/echo-cancel/Makefile.am
index 2222ce86a5758d60d9e3fb6007a6dbd679e65010..a7eab76fe0f13c97380b952edc9a23dc6c717d4b 100644
--- a/src/media/audio/echo-cancel/Makefile.am
+++ b/src/media/audio/echo-cancel/Makefile.am
@@ -10,6 +10,11 @@ EC_SRC += speex_echo_canceller.cpp
EC_HDR += speex_echo_canceller.h
endif
+if HAVE_WEBRTC_AP
+EC_SRC += webrtc_echo_canceller.cpp
+EC_HDR += webrtc_echo_canceller.h
+endif
+
libecho_cancel_la_SOURCES = \
$(EC_SRC)
diff --git a/src/media/audio/echo-cancel/echo_canceller.h b/src/media/audio/echo-cancel/echo_canceller.h
index c159957f4866c0224189d798eaf8764ba1ddb1b7..b925f2fa994361df7ef389cf9a546907a2acb552 100644
--- a/src/media/audio/echo-cancel/echo_canceller.h
+++ b/src/media/audio/echo-cancel/echo_canceller.h
@@ -22,9 +22,13 @@
#include "noncopyable.h"
#include "audio/audio_frame_resizer.h"
+#include "audio/resampler.h"
#include "audio/audiobuffer.h"
#include "libav_deps.h"
+#include <atomic>
+#include <memory>
+
namespace jami {
class EchoCanceller
@@ -36,19 +40,26 @@ public:
EchoCanceller(AudioFormat format, unsigned frameSize)
: playbackQueue_(format, frameSize)
, recordQueue_(format, frameSize)
- , sampleRate_(format.sample_rate)
+ , resampler_(new Resampler)
+ , format_(format)
, frameSize_(frameSize)
{}
virtual ~EchoCanceller() = default;
virtual void putRecorded(std::shared_ptr<AudioFrame>&& buf)
{
- recordQueue_.enqueue(std::move(buf));
+ recordStarted_ = true;
+ if (!playbackStarted_)
+ return;
+ enqueue(recordQueue_, std::move(buf));
};
virtual void putPlayback(const std::shared_ptr<AudioFrame>& buf)
{
- auto c = buf;
- playbackQueue_.enqueue(std::move(c));
+ playbackStarted_ = true;
+ if (!recordStarted_)
+ return;
+ auto copy = buf;
+ enqueue(playbackQueue_, std::move(copy));
};
virtual std::shared_ptr<AudioFrame> getProcessed() = 0;
virtual void done() = 0;
@@ -56,8 +67,21 @@ public:
protected:
AudioFrameResizer playbackQueue_;
AudioFrameResizer recordQueue_;
- unsigned sampleRate_;
+ std::unique_ptr<Resampler> resampler_;
+ std::atomic_bool playbackStarted_;
+ std::atomic_bool recordStarted_;
+ AudioFormat format_;
unsigned frameSize_;
+
+private:
+ void enqueue(AudioFrameResizer& frameResizer, std::shared_ptr<AudioFrame>&& buf)
+ {
+ if (buf->getFormat() != format_) {
+ auto resampled = resampler_->resample(std::move(buf), format_);
+ frameResizer.enqueue(std::move(resampled));
+ } else
+ frameResizer.enqueue(std::move(buf));
+ };
};
} // namespace jami
diff --git a/src/media/audio/echo-cancel/speex_echo_canceller.cpp b/src/media/audio/echo-cancel/speex_echo_canceller.cpp
index aace775c12372fdb7776f59def00cfbf0fe2eb7c..463292720d759eeddc5de2d383efed13903228e8 100644
--- a/src/media/audio/echo-cancel/speex_echo_canceller.cpp
+++ b/src/media/audio/echo-cancel/speex_echo_canceller.cpp
@@ -50,7 +50,7 @@ SpeexEchoCanceller::SpeexEchoCanceller(AudioFormat format, unsigned frameSize)
: EchoCanceller(format, frameSize)
, pimpl_(std::make_unique<SpeexEchoStateImpl>(format, frameSize))
{
- speex_echo_ctl(pimpl_->state.get(), SPEEX_ECHO_SET_SAMPLING_RATE, &sampleRate_);
+ speex_echo_ctl(pimpl_->state.get(), SPEEX_ECHO_SET_SAMPLING_RATE, &format_.sample_rate);
}
void
diff --git a/src/media/audio/echo-cancel/webrtc_echo_canceller.cpp b/src/media/audio/echo-cancel/webrtc_echo_canceller.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..8b20507edce1b3520851113c8339eb31da8a921d
--- /dev/null
+++ b/src/media/audio/echo-cancel/webrtc_echo_canceller.cpp
@@ -0,0 +1,170 @@
+/*
+ * Copyright (C) 2021 Savoir-faire Linux Inc.
+ *
+ * Author: Andreas Traczyk <andreas.traczyk@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "webrtc_echo_canceller.h"
+
+#include <webrtc/modules/audio_processing/include/audio_processing.h>
+
+namespace jami {
+
+WebRTCEchoCanceller::WebRTCEchoCanceller(AudioFormat format, unsigned frameSize)
+ : EchoCanceller(format, frameSize)
+ , pimpl_(std::make_unique<WebRTCAPMImpl>(format, frameSize))
+ , fRecordBuffer_(format.nb_channels, std::vector<float>(frameSize_, 0))
+ , fPlaybackBuffer_(format.nb_channels, std::vector<float>(frameSize_, 0))
+ , iRecordBuffer_(frameSize_, format)
+ , iPlaybackBuffer_(frameSize_, format)
+{}
+
+struct WebRTCEchoCanceller::WebRTCAPMImpl
+{
+ using APMPtr = std::unique_ptr<webrtc::AudioProcessing>;
+ APMPtr apm;
+ webrtc::StreamConfig streamConfig;
+
+ WebRTCAPMImpl(AudioFormat format, unsigned frameSize)
+ : streamConfig(format.sample_rate, format.nb_channels)
+ {
+ webrtc::ProcessingConfig pconfig;
+ webrtc::Config config;
+
+ config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true));
+ config.Set<webrtc::DelayAgnostic>(new webrtc::DelayAgnostic(true));
+
+ apm.reset(webrtc::AudioProcessing::Create(config));
+
+ pconfig = {
+ streamConfig, /* input stream */
+ streamConfig, /* output stream */
+ streamConfig, /* reverse input stream */
+ streamConfig, /* reverse output stream */
+ };
+
+ if (apm->Initialize(pconfig) != webrtc::AudioProcessing::kNoError) {
+ JAMI_ERR("[webrtc-ap] Error initialising audio processing module");
+ }
+
+ // aec
+ apm->echo_cancellation()->set_suppression_level(
+ webrtc::EchoCancellation::SuppressionLevel::kModerateSuppression);
+ apm->echo_cancellation()->enable_drift_compensation(true);
+ apm->echo_cancellation()->Enable(true);
+
+ // hpf
+ apm->high_pass_filter()->Enable(true);
+
+ // ns
+ apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
+ apm->noise_suppression()->Enable(true);
+
+ // agc
+ apm->gain_control()->set_analog_level_limits(0, 255);
+ apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog);
+ apm->gain_control()->Enable(true);
+ }
+};
+
+void
+WebRTCEchoCanceller::putRecorded(std::shared_ptr<AudioFrame>&& buf)
+{
+ EchoCanceller::putRecorded(std::move(buf));
+}
+
+void
+WebRTCEchoCanceller::putPlayback(const std::shared_ptr<AudioFrame>& buf)
+{
+ EchoCanceller::putPlayback(buf);
+}
+
+std::shared_ptr<AudioFrame>
+WebRTCEchoCanceller::getProcessed()
+{
+ while (recordQueue_.samples() > recordQueue_.frameSize() * 10) {
+ JAMI_DBG("record overflow %d / %d", recordQueue_.samples(), frameSize_);
+ recordQueue_.dequeue();
+ }
+ while (playbackQueue_.samples() > playbackQueue_.frameSize() * 10) {
+ JAMI_DBG("playback overflow %d / %d", playbackQueue_.samples(), frameSize_);
+ playbackQueue_.dequeue();
+ }
+ if (recordQueue_.samples() < recordQueue_.frameSize()
+ || playbackQueue_.samples() < playbackQueue_.frameSize()) {
+ // If there are not enough samples in either queue, we can't
+ // process anything.
+ // JAMI_DBG("underrun p:%d / r:%d", playbackQueue_.samples(), recordQueue_.samples());
+ return {};
+ }
+
+ int driftSamples = playbackQueue_.samples() - recordQueue_.samples();
+
+ auto playback = playbackQueue_.dequeue();
+ auto record = recordQueue_.dequeue();
+ if (!playback || !record)
+ return {};
+
+ auto processed = std::make_shared<AudioFrame>(format_, frameSize_);
+
+ webrtc::StreamConfig& sc = pimpl_->streamConfig;
+
+ // analyze deinterleaved float playback data
+ iPlaybackBuffer_.deinterleave((const AudioSample*) playback->pointer()->data[0],
+ frameSize_,
+ format_.nb_channels);
+ std::vector<float*> playData {format_.nb_channels};
+ for (auto c = 0; c < format_.nb_channels; ++c) {
+ playData[c] = fPlaybackBuffer_[c].data();
+ iPlaybackBuffer_.channelToFloat(playData[c], c);
+ }
+ if (pimpl_->apm->ProcessReverseStream(playData.data(), sc, sc, playData.data())
+ != webrtc::AudioProcessing::kNoError)
+ JAMI_ERR("[webrtc-ap] ProcessReverseStream failed");
+
+ // process deinterleaved float recorded data
+ iRecordBuffer_.deinterleave((const AudioSample*) record->pointer()->data[0],
+ frameSize_,
+ format_.nb_channels);
+ std::vector<float*> recData {format_.nb_channels};
+ for (auto c = 0; c < format_.nb_channels; ++c) {
+ recData[c] = fRecordBuffer_[c].data();
+ iRecordBuffer_.channelToFloat(recData[c], c);
+ }
+ // TODO: implement this correctly (it MUST be called prior to ProcessStream)
+ // delay = (t_render - t_analyze) + (t_process - t_capture)
+ pimpl_->apm->set_stream_delay_ms(0);
+ pimpl_->apm->gain_control()->set_stream_analog_level(analogLevel_);
+ pimpl_->apm->echo_cancellation()->set_stream_drift_samples(driftSamples);
+ if (pimpl_->apm->ProcessStream(recData.data(), sc, sc, recData.data())
+ != webrtc::AudioProcessing::kNoError)
+ JAMI_ERR("[webrtc-ap] ProcessStream failed");
+ analogLevel_ = pimpl_->apm->gain_control()->stream_analog_level();
+
+ // return interleaved s16 data
+ iRecordBuffer_.convertFloatPlanarToSigned16((uint8_t**) recData.data(),
+ frameSize_,
+ format_.nb_channels);
+ iRecordBuffer_.interleave((AudioSample*) processed->pointer()->data[0]);
+ return processed;
+}
+
+void
+WebRTCEchoCanceller::done()
+{}
+
+} // namespace jami
diff --git a/src/media/audio/echo-cancel/webrtc_echo_canceller.h b/src/media/audio/echo-cancel/webrtc_echo_canceller.h
new file mode 100644
index 0000000000000000000000000000000000000000..02e7eece35522c0af1489c51fda10506774ed002
--- /dev/null
+++ b/src/media/audio/echo-cancel/webrtc_echo_canceller.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (C) 2021 Savoir-faire Linux Inc.
+ *
+ * Author: Andreas Traczyk <andreas.traczyk@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#pragma once
+
+#include "audio/echo-cancel/echo_canceller.h"
+#include "audio/audio_frame_resizer.h"
+
+#include <memory>
+
+namespace jami {
+
+class WebRTCEchoCanceller final : public EchoCanceller
+{
+public:
+ WebRTCEchoCanceller(AudioFormat format, unsigned frameSize);
+ ~WebRTCEchoCanceller() = default;
+
+ // Inherited via EchoCanceller
+ void putRecorded(std::shared_ptr<AudioFrame>&& buf) override;
+ void putPlayback(const std::shared_ptr<AudioFrame>& buf) override;
+ std::shared_ptr<AudioFrame> getProcessed() override;
+ void done() override;
+
+private:
+ struct WebRTCAPMImpl;
+ std::unique_ptr<WebRTCAPMImpl> pimpl_;
+
+ using fChannelBuffer = std::vector<std::vector<float>>;
+ fChannelBuffer fRecordBuffer_;
+ fChannelBuffer fPlaybackBuffer_;
+ AudioBuffer iRecordBuffer_;
+ AudioBuffer iPlaybackBuffer_;
+ int analogLevel_ {0};
+};
+} // namespace jami
diff --git a/src/media/audio/portaudio/portaudiolayer.cpp b/src/media/audio/portaudio/portaudiolayer.cpp
index 3a6aca9fb8c5611f5254e764635abf88108449bb..f473c11875c9981251c18ce595c53b9a81ba25f3 100644
--- a/src/media/audio/portaudio/portaudiolayer.cpp
+++ b/src/media/audio/portaudio/portaudiolayer.cpp
@@ -91,6 +91,8 @@ PortAudioLayer::PortAudioLayer(const AudioPreference& pref)
: AudioLayer {pref}
, pimpl_ {new PortAudioLayerImpl(*this, pref)}
{
+ setHasNativeAEC(false);
+
auto numDevices = Pa_GetDeviceCount();
if (numDevices < 0) {
JAMI_ERR("Pa_CountDevices returned 0x%x", numDevices);