UltraCoder/speex-webrtc-test
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Begin switch from webrtc-audio-processing to libwebrtc.

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This commit is contained in:
Kirill Kirilenko 2020-11-10 17:44:42 +03:00
parent 49bf3d44dc
commit 54451640d7
3 changed files with 36 additions and 37 deletions
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1
CMakeLists.txt
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@ -7,7 +7,6 @@ list(APPEND CMAKE_MODULE_PATH ${CMAKE_CURRENT_SOURCE_DIR}/cmake)
find_package(Threads REQUIRED) find_package(Threads REQUIRED)
find_package(Qt5 COMPONENTS Core Widgets Multimedia REQUIRED) find_package(Qt5 COMPONENTS Core Widgets Multimedia REQUIRED)
find_package(WebRTC REQUIRED)
if (MSVC) if (MSVC)
set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} /wd4267") set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} /wd4267")

5
src/CMakeLists.txt
View file

@ -1,4 +1,7 @@
find_package(LibWebRTC REQUIRED)
include(${LIBWEBRTC_USE_FILE})
set(TARGET_NAME speex_webrtc_test) set(TARGET_NAME speex_webrtc_test)
set(CMAKE_AUTOMOC ON) set(CMAKE_AUTOMOC ON)
@ -17,7 +20,7 @@ endif()
target_link_libraries(${TARGET_NAME} target_link_libraries(${TARGET_NAME}
speexdsp speexdsp
WebRTC ${LIBWEBRTC_LIBRARIES}
Qt5::Core Qt5::Core
Qt5::Widgets Qt5::Widgets
Qt5::Multimedia Qt5::Multimedia

67
src/WebRTCDSP.cpp
View file

@ -1,7 +1,7 @@
#include "WebRTCDSP.h" #include "WebRTCDSP.h"
#include <webrtc/api/audio/audio_frame.h>
#include <webrtc/modules/audio_processing/include/audio_processing.h> #include <webrtc/modules/audio_processing/include/audio_processing.h>
#include <webrtc/modules/interface/module_common_types.h>
#include <QLoggingCategory> #include <QLoggingCategory>
@ -9,6 +9,8 @@ namespace SpeexWebRTCTest {
namespace { namespace {
using NoiseSuppressionLevel = webrtc::AudioProcessing::Config::NoiseSuppression::Level;
Q_LOGGING_CATEGORY(WebRTC, "webrtc") Q_LOGGING_CATEGORY(WebRTC, "webrtc")
void convert(const QAudioBuffer& from, webrtc::AudioFrame& to) void convert(const QAudioBuffer& from, webrtc::AudioFrame& to)
@ -16,8 +18,7 @@ void convert(const QAudioBuffer& from, webrtc::AudioFrame& to)
to.num_channels_ = from.format().channelCount(); to.num_channels_ = from.format().channelCount();
to.sample_rate_hz_ = from.format().sampleRate(); to.sample_rate_hz_ = from.format().sampleRate();
to.samples_per_channel_ = from.frameCount(); to.samples_per_channel_ = from.frameCount();
to.interleaved_ = (from.format().channelCount() > 1); memcpy(to.mutable_data(), from.constData<char>(), from.byteCount());
memcpy(to.data_, from.constData<char>(), from.byteCount());
} }
void convert(const webrtc::AudioFrame& from, QAudioBuffer& to) void convert(const webrtc::AudioFrame& from, QAudioBuffer& to)
@ -30,7 +31,7 @@ void convert(const webrtc::AudioFrame& from, QAudioBuffer& to)
format.setByteOrder(QAudioFormat::LittleEndian); format.setByteOrder(QAudioFormat::LittleEndian);
format.setSampleType(QAudioFormat::SignedInt); format.setSampleType(QAudioFormat::SignedInt);
QByteArray data(reinterpret_cast<const char*>(from.data_), QByteArray data(reinterpret_cast<const char*>(from.data()),
from.samples_per_channel_ * from.num_channels_ * sizeof(std::int16_t)); from.samples_per_channel_ * from.num_channels_ * sizeof(std::int16_t));
to = QAudioBuffer(data, format); to = QAudioBuffer(data, format);
} }
@ -40,33 +41,29 @@ void convert(const webrtc::AudioFrame& from, QAudioBuffer& to)
WebRTCDSP::WebRTCDSP(const QAudioFormat& mainFormat, const QAudioFormat& auxFormat) WebRTCDSP::WebRTCDSP(const QAudioFormat& mainFormat, const QAudioFormat& auxFormat)
: AudioEffect(mainFormat, auxFormat) : AudioEffect(mainFormat, auxFormat)
{ {
apm_ = webrtc::AudioProcessing::Create(); apm_ = webrtc::AudioProcessingBuilder().Create();
if (!apm_) if (!apm_)
throw std::runtime_error("failed to create webrtc::AudioProcessing instance"); throw std::runtime_error("failed to create webrtc::AudioProcessing instance");
webrtc::Config config; webrtc::AudioProcessing::Config config;
config.Set<webrtc::DelayAgnostic>(new webrtc::DelayAgnostic(true));
config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true));
apm_->SetExtraOptions(config);
apm_->voice_detection()->Enable(true); config.voice_detection.enabled = true;
apm_->voice_detection()->set_frame_size_ms(300);
apm_->voice_detection()->set_likelihood(webrtc::VoiceDetection::kModerateLikelihood);
apm_->noise_suppression()->Enable(false); config.noise_suppression.enabled = false;
apm_->noise_suppression()->set_level(webrtc::NoiseSuppression::kLow); config.noise_suppression.level = NoiseSuppressionLevel::kLow;
apm_->echo_cancellation()->Enable(false); config.echo_canceller.enabled = false;
apm_->echo_cancellation()->set_suppression_level(webrtc::EchoCancellation::kLowSuppression); config.echo_canceller.mobile_mode = false;
apm_->set_stream_delay_ms(100); config.residual_echo_detector.enabled = false;
apm_->gain_control()->Enable(false); config.gain_controller1.enabled = false;
apm_->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital); config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kAdaptiveDigital;
apm_->gain_control()->enable_limiter(true); config.gain_controller1.enable_limiter = true;
apm_->gain_control()->set_compression_gain_db(0); config.gain_controller1.compression_gain_db = 0;
apm_->gain_control()->set_target_level_dbfs(0); config.gain_controller1.target_level_dbfs = 0;
// apm_->Initialize(); apm_->ApplyConfig(config);
} }
WebRTCDSP::~WebRTCDSP() WebRTCDSP::~WebRTCDSP()
@ -132,7 +129,7 @@ void WebRTCDSP::processFrame(QAudioBuffer& mainBuffer, const QAudioBuffer& auxBu
int error; int error;
if (apm_->echo_cancellation()->is_enabled()) if (apm_->GetConfig().echo_canceller.enabled)
{ {
error = apm_->ProcessReverseStream(&auxFrame); error = apm_->ProcessReverseStream(&auxFrame);
if (error != 0) if (error != 0)
@ -153,30 +150,30 @@ void WebRTCDSP::processFrame(QAudioBuffer& mainBuffer, const QAudioBuffer& auxBu
convert(mainFrame, mainBuffer); convert(mainFrame, mainBuffer);
setVoiceActive(apm_->voice_detection()->stream_has_voice()); setVoiceActive(*apm_->GetStatistics().voice_detected);
} }
void WebRTCDSP::setParameter(const QString& param, QVariant value) void WebRTCDSP::setParameter(const QString& param, QVariant value)
{ {
auto config = apm_->GetConfig();
if (param == "noise_reduction_enabled") if (param == "noise_reduction_enabled")
apm_->noise_suppression()->Enable(value.toBool()); config.noise_suppression.enabled = value.toBool();
else if (param == "noise_reduction_suppression_level") else if (param == "noise_reduction_suppression_level")
apm_->noise_suppression()->set_level(static_cast<webrtc::NoiseSuppression::Level>( config.noise_suppression.level =
webrtc::NoiseSuppression::kLow + value.toUInt())); static_cast<NoiseSuppressionLevel>(NoiseSuppressionLevel::kLow + value.toUInt());
else if (param == "echo_cancellation_enabled") else if (param == "echo_cancellation_enabled")
apm_->echo_cancellation()->Enable(value.toBool()); config.echo_canceller.enabled = value.toBool();
else if (param == "echo_cancellation_suppression_level") else if (param == "echo_cancellation_suppression_level")
apm_->echo_cancellation()->set_suppression_level( return; // TODO ???
static_cast<webrtc::EchoCancellation::SuppressionLevel>(
webrtc::EchoCancellation::kLowSuppression + value.toUInt()));
else if (param == "gain_control_enabled") else if (param == "gain_control_enabled")
apm_->gain_control()->Enable(value.toBool()); config.gain_controller1.enabled = value.toBool();
else if (param == "gain_control_target_level") else if (param == "gain_control_target_level")
apm_->gain_control()->set_target_level_dbfs(value.toInt()); config.gain_controller1.target_level_dbfs = value.toInt();
else if (param == "gain_control_max_gain") else if (param == "gain_control_max_gain")
apm_->gain_control()->set_compression_gain_db(value.toInt()); config.gain_controller1.compression_gain_db = value.toInt();
else else
throw std::invalid_argument("Invalid param"); throw std::invalid_argument("Invalid param");
apm_->ApplyConfig(config);
} }
unsigned int WebRTCDSP::requiredFrameSizeMs() const unsigned int WebRTCDSP::requiredFrameSizeMs() const