Begin switch from webrtc-audio-processing to libwebrtc.
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49bf3d44dc
commit
54451640d7
3 changed files with 36 additions and 37 deletions
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@ -7,7 +7,6 @@ list(APPEND CMAKE_MODULE_PATH ${CMAKE_CURRENT_SOURCE_DIR}/cmake)
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find_package(Threads REQUIRED)
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find_package(Qt5 COMPONENTS Core Widgets Multimedia REQUIRED)
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find_package(WebRTC REQUIRED)
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if (MSVC)
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set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} /wd4267")
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@ -1,4 +1,7 @@
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find_package(LibWebRTC REQUIRED)
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include(${LIBWEBRTC_USE_FILE})
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set(TARGET_NAME speex_webrtc_test)
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set(CMAKE_AUTOMOC ON)
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@ -17,7 +20,7 @@ endif()
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target_link_libraries(${TARGET_NAME}
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speexdsp
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WebRTC
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${LIBWEBRTC_LIBRARIES}
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Qt5::Core
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Qt5::Widgets
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Qt5::Multimedia
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@ -1,7 +1,7 @@
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#include "WebRTCDSP.h"
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#include <webrtc/api/audio/audio_frame.h>
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#include <webrtc/modules/audio_processing/include/audio_processing.h>
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#include <webrtc/modules/interface/module_common_types.h>
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#include <QLoggingCategory>
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@ -9,6 +9,8 @@ namespace SpeexWebRTCTest {
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namespace {
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using NoiseSuppressionLevel = webrtc::AudioProcessing::Config::NoiseSuppression::Level;
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Q_LOGGING_CATEGORY(WebRTC, "webrtc")
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void convert(const QAudioBuffer& from, webrtc::AudioFrame& to)
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@ -16,8 +18,7 @@ void convert(const QAudioBuffer& from, webrtc::AudioFrame& to)
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to.num_channels_ = from.format().channelCount();
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to.sample_rate_hz_ = from.format().sampleRate();
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to.samples_per_channel_ = from.frameCount();
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to.interleaved_ = (from.format().channelCount() > 1);
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memcpy(to.data_, from.constData<char>(), from.byteCount());
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memcpy(to.mutable_data(), from.constData<char>(), from.byteCount());
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}
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void convert(const webrtc::AudioFrame& from, QAudioBuffer& to)
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@ -30,7 +31,7 @@ void convert(const webrtc::AudioFrame& from, QAudioBuffer& to)
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format.setByteOrder(QAudioFormat::LittleEndian);
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format.setSampleType(QAudioFormat::SignedInt);
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QByteArray data(reinterpret_cast<const char*>(from.data_),
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QByteArray data(reinterpret_cast<const char*>(from.data()),
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from.samples_per_channel_ * from.num_channels_ * sizeof(std::int16_t));
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to = QAudioBuffer(data, format);
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}
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@ -40,33 +41,29 @@ void convert(const webrtc::AudioFrame& from, QAudioBuffer& to)
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WebRTCDSP::WebRTCDSP(const QAudioFormat& mainFormat, const QAudioFormat& auxFormat)
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: AudioEffect(mainFormat, auxFormat)
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{
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apm_ = webrtc::AudioProcessing::Create();
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apm_ = webrtc::AudioProcessingBuilder().Create();
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if (!apm_)
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throw std::runtime_error("failed to create webrtc::AudioProcessing instance");
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webrtc::Config config;
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config.Set<webrtc::DelayAgnostic>(new webrtc::DelayAgnostic(true));
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config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true));
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apm_->SetExtraOptions(config);
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webrtc::AudioProcessing::Config config;
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apm_->voice_detection()->Enable(true);
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apm_->voice_detection()->set_frame_size_ms(300);
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apm_->voice_detection()->set_likelihood(webrtc::VoiceDetection::kModerateLikelihood);
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config.voice_detection.enabled = true;
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apm_->noise_suppression()->Enable(false);
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apm_->noise_suppression()->set_level(webrtc::NoiseSuppression::kLow);
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config.noise_suppression.enabled = false;
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config.noise_suppression.level = NoiseSuppressionLevel::kLow;
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apm_->echo_cancellation()->Enable(false);
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apm_->echo_cancellation()->set_suppression_level(webrtc::EchoCancellation::kLowSuppression);
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apm_->set_stream_delay_ms(100);
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config.echo_canceller.enabled = false;
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config.echo_canceller.mobile_mode = false;
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config.residual_echo_detector.enabled = false;
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apm_->gain_control()->Enable(false);
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apm_->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
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apm_->gain_control()->enable_limiter(true);
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apm_->gain_control()->set_compression_gain_db(0);
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apm_->gain_control()->set_target_level_dbfs(0);
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config.gain_controller1.enabled = false;
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config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kAdaptiveDigital;
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config.gain_controller1.enable_limiter = true;
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config.gain_controller1.compression_gain_db = 0;
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config.gain_controller1.target_level_dbfs = 0;
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// apm_->Initialize();
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apm_->ApplyConfig(config);
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}
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WebRTCDSP::~WebRTCDSP()
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@ -132,7 +129,7 @@ void WebRTCDSP::processFrame(QAudioBuffer& mainBuffer, const QAudioBuffer& auxBu
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int error;
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if (apm_->echo_cancellation()->is_enabled())
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if (apm_->GetConfig().echo_canceller.enabled)
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{
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error = apm_->ProcessReverseStream(&auxFrame);
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if (error != 0)
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@ -153,30 +150,30 @@ void WebRTCDSP::processFrame(QAudioBuffer& mainBuffer, const QAudioBuffer& auxBu
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convert(mainFrame, mainBuffer);
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setVoiceActive(apm_->voice_detection()->stream_has_voice());
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setVoiceActive(*apm_->GetStatistics().voice_detected);
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}
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void WebRTCDSP::setParameter(const QString& param, QVariant value)
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{
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auto config = apm_->GetConfig();
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if (param == "noise_reduction_enabled")
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apm_->noise_suppression()->Enable(value.toBool());
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config.noise_suppression.enabled = value.toBool();
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else if (param == "noise_reduction_suppression_level")
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apm_->noise_suppression()->set_level(static_cast<webrtc::NoiseSuppression::Level>(
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webrtc::NoiseSuppression::kLow + value.toUInt()));
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config.noise_suppression.level =
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static_cast<NoiseSuppressionLevel>(NoiseSuppressionLevel::kLow + value.toUInt());
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else if (param == "echo_cancellation_enabled")
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apm_->echo_cancellation()->Enable(value.toBool());
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config.echo_canceller.enabled = value.toBool();
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else if (param == "echo_cancellation_suppression_level")
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apm_->echo_cancellation()->set_suppression_level(
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static_cast<webrtc::EchoCancellation::SuppressionLevel>(
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webrtc::EchoCancellation::kLowSuppression + value.toUInt()));
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return; // TODO ???
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else if (param == "gain_control_enabled")
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apm_->gain_control()->Enable(value.toBool());
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config.gain_controller1.enabled = value.toBool();
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else if (param == "gain_control_target_level")
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apm_->gain_control()->set_target_level_dbfs(value.toInt());
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config.gain_controller1.target_level_dbfs = value.toInt();
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else if (param == "gain_control_max_gain")
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apm_->gain_control()->set_compression_gain_db(value.toInt());
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config.gain_controller1.compression_gain_db = value.toInt();
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else
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throw std::invalid_argument("Invalid param");
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apm_->ApplyConfig(config);
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}
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unsigned int WebRTCDSP::requiredFrameSizeMs() const
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