mirror of
https://github.com/qTox/qTox.git
synced 2024-03-22 14:00:36 +08:00
refactor: remove dependency on libfilteraudio
The audio filtering/echo compensation didn't ever work reliably, so just remove it.
This commit is contained in:
Anthony Bilinski
parent
715794f664
commit
4f7056385f
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@ -24,7 +24,6 @@ cmake_minimum_required(VERSION 2.8.11)
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set(CMAKE_MODULE_PATH ${CMAKE_CURRENT_SOURCE_DIR}/cmake)
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option(PLATFORM_EXTENSIONS "Enable platform specific extensions, requires extra dependencies" ON)
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option(USE_FILTERAUDIO "Enable the echo canceling backend" ON)
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option(UPDATE_CHECK "Enable automatic update check" ON)
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option(USE_CCACHE "Use ccache when available" ON)
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option(SPELL_CHECK "Enable spell cheching support" ON)
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@ -623,19 +622,6 @@ if(AVFOUNDATION_FOUND)
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src/platform/camera/avfoundation.h)
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endif()
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if(${USE_FILTERAUDIO})
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search_dependency(FILTERAUDIO LIBRARY filteraudio HEADER filter_audio.h OPTIONAL)
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if(${FILTERAUDIO_FOUND})
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set(${PROJECT_NAME}_SOURCES ${${PROJECT_NAME}_SOURCES}
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src/audio/backend/openal2.cpp
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src/audio/backend/openal2.h)
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add_definitions(-DUSE_FILTERAUDIO=1)
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message(STATUS "using filteraudio")
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else()
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message(STATUS "not using filteraudio, libfilteraudio not found")
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endif()
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endif()
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if(${UPDATE_CHECK})
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add_definitions(-DUPDATE_CHECK_ENABLED=1)
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set(${PROJECT_NAME}_SOURCES ${${PROJECT_NAME}_SOURCES}
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@ -46,7 +46,6 @@
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| [qrencode] | >= 3.0.3 | |
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| [sqlcipher] | >= 3.2.0 | |
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| [pkg-config] | >= 0.28 | |
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| [filteraudio] | >= 0.0.1 | optional dependency |
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| [snorenotify] | >= 0.7.0 | optional dependency |
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## Optional dependencies
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@ -812,6 +811,5 @@ Switches:
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[Qt]: https://www.qt.io/
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[sqlcipher]: https://www.zetetic.net/sqlcipher/
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[toxcore]: https://github.com/TokTok/c-toxcore/
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[filteraudio]: https://github.com/irungentoo/filter_audio
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[sonnet]: https://github.com/KDE/sonnet
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[snorenotify]: https://techbase.kde.org/Projects/Snorenotify
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@ -41,7 +41,6 @@ QT_VER=($(ls ${QT_DIR} | sed -n -e 's/^\([0-9]*\.([0-9]*\.([0-9]*\).*/\1/' -e '1
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QT_DIR_VER="${QT_DIR}/${QT_VER[1]}"
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TOXCORE_DIR="${MAIN_DIR}/toxcore" # Change to Git location
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FILTERAUIO_DIR="${MAIN_DIR}/filter_audio" # Change to Git location
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LIB_INSTALL_PREFIX="${QTOX_DIR}/libs"
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@ -208,12 +207,6 @@ install() {
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kill $DOT_PID
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fi
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fcho "Cloning filter_audio ... "
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git clone --branch v0.0.1 --depth=1 https://github.com/irungentoo/filter_audio "$FILTERAUIO_DIR"
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cd "$FILTERAUIO_DIR"
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fcho "Installing filter_audio ... "
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make install PREFIX="$LIB_INSTALL_PREFIX"
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QT_VER=($(ls ${QT_DIR} | sed -n -e 's/^\([0-9]*\.([0-9]*\.([0-9]*\).*/\1/' -e '1p;$p'))
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QT_DIR_VER="${QT_DIR}/${QT_VER[1]}"
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@ -22,9 +22,6 @@
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#include "src/audio/audio.h"
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#include "src/audio/iaudiosettings.h"
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#include "src/audio/backend/openal.h"
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#ifdef USE_FILTERAUDIO
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#include "src/audio/backend/openal2.h"
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#endif
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/**
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* @brief Select the audio backend
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@ -33,14 +30,5 @@
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*/
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std::unique_ptr<IAudioControl> Audio::makeAudio(IAudioSettings& settings)
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{
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#ifdef USE_FILTERAUDIO
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const bool Backend2 = settings.getEnableBackend2();
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if (Backend2) {
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return std::unique_ptr<IAudioControl>(new OpenAL2());
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} else
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#endif
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{
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return std::unique_ptr<IAudioControl>(new OpenAL());
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}
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return std::unique_ptr<IAudioControl>(new OpenAL());
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}
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@ -1,357 +0,0 @@
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/*
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Copyright © 2017-2019 by The qTox Project Contributors
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This file is part of qTox, a Qt-based graphical interface for Tox.
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qTox is libre software: you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation, either version 3 of the License, or
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(at your option) any later version.
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qTox is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with qTox. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include "openal2.h"
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#include "src/persistence/settings.h"
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#include <QDebug>
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#include <QFile>
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#include <QMutexLocker>
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#include <QPointer>
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#include <QThread>
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#include <QWaitCondition>
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#include <QtMath>
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#include <cassert>
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extern "C"
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{
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#include <filter_audio.h>
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}
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/**
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* @class OpenAL
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* @brief Provides the OpenAL audio backend
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*
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* The following graphic describes the audio rendering pipeline with echo canceling
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*
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* | alProxyContext | | alOutContext |
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* peerSources[] | | |
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* \ | | |
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* -> alProxyDev -> filter_audio -> alProxySource -> alOutDev -> Soundcard
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* /
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* alMainSource
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*
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* Without echo cancelling the pipeline is simplified through alProxyDev = alOutDev
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* and alProxyContext = alOutContext
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*
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* | alProxyContext |
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* peerSources[] |
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* \ |
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* -> alOutDev -> Soundcard
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* /
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* alMainSource
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*
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* To keep all functions in writing to the correct context, all functions changing
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* the context MUST exit with alProxyContext as active context and MUST not be
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* interrupted. For this to work, all functions of the base class modifying the
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* context have to be overriden.
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*/
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static const unsigned int PROXY_BUFFER_COUNT = 4;
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#define GET_PROC_ADDR(dev, name) name = reinterpret_cast<LP##name>(alcGetProcAddress(dev, #name))
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typedef LPALCLOOPBACKOPENDEVICESOFT LPalcLoopbackOpenDeviceSOFT;
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typedef LPALCISRENDERFORMATSUPPORTEDSOFT LPalcIsRenderFormatSupportedSOFT;
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typedef LPALGETSOURCEDVSOFT LPalGetSourcedvSOFT;
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typedef LPALCRENDERSAMPLESSOFT LPalcRenderSamplesSOFT;
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OpenAL2::OpenAL2()
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: alProxyDev{nullptr}
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, alProxyContext{nullptr}
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, alProxySource{0}
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, alProxyBuffer{0}
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{}
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bool OpenAL2::initInput(const QString& deviceName)
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{
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return OpenAL::initInput(deviceName, 1);
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}
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/**
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* @brief Loads the OpenAL extension methods needed for echo cancellation
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* @param dev OpenAL device used for the output
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* @return True when functions successfully loaded, false otherwise
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*/
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bool OpenAL2::loadOpenALExtensions(ALCdevice* dev)
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{
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// load OpenAL extension functions
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GET_PROC_ADDR(dev, alcLoopbackOpenDeviceSOFT);
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checkAlcError(dev);
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if (!alcLoopbackOpenDeviceSOFT) {
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qDebug() << "Failed to load alcLoopbackOpenDeviceSOFT function!";
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return false;
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}
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GET_PROC_ADDR(dev, alcIsRenderFormatSupportedSOFT);
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checkAlcError(dev);
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if (!alcIsRenderFormatSupportedSOFT) {
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qDebug() << "Failed to load alcIsRenderFormatSupportedSOFT function!";
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return false;
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}
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GET_PROC_ADDR(dev, alGetSourcedvSOFT);
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checkAlcError(dev);
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if (!alGetSourcedvSOFT) {
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qDebug() << "Failed to load alGetSourcedvSOFT function!";
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return false;
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}
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GET_PROC_ADDR(dev, alcRenderSamplesSOFT);
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checkAlcError(dev);
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if (!alcRenderSamplesSOFT) {
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qDebug() << "Failed to load alcRenderSamplesSOFT function!";
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return false;
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}
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return true;
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}
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/**
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* @brief Initializes the output with echo cancelling enabled
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* @return true on success, false otherwise
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* Creates a loopback device and a proxy source on the main output device.
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* If this function returns true only the proxy source should be used for
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* audio output.
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*/
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bool OpenAL2::initOutputEchoCancel()
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{
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// check for the needed extensions for echo cancelation
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if (alcIsExtensionPresent(alOutDev, "ALC_SOFT_loopback") != AL_TRUE) {
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qDebug() << "Device doesn't support loopback";
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return false;
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}
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if (alIsExtensionPresent("AL_SOFT_source_latency") != AL_TRUE) {
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qDebug() << "Device doesn't support source latency";
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return false;
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}
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if (!loadOpenALExtensions(alOutDev)) {
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qDebug() << "Couldn't load needed OpenAL extensions";
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return false;
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}
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// source for proxy output
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alGenSources(1, &alProxySource);
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checkAlError();
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// configuration for the loopback device
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ALCint attrs[] = {ALC_FORMAT_CHANNELS_SOFT,
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ALC_MONO_SOFT,
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ALC_FORMAT_TYPE_SOFT,
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ALC_SHORT_SOFT,
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ALC_FREQUENCY,
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IAudioControl::AUDIO_SAMPLE_RATE,
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0}; // End of List
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alProxyDev = alcLoopbackOpenDeviceSOFT(nullptr);
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checkAlcError(alProxyDev);
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if (!alProxyDev) {
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qDebug() << "Couldn't create proxy device";
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alDeleteSources(1, &alProxySource); // cleanup source
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return false;
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}
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if (!alcIsRenderFormatSupportedSOFT(alProxyDev, attrs[5], attrs[1], attrs[3])) {
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qDebug() << "Unsupported format for loopback";
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alcCloseDevice(alProxyDev); // cleanup loopback dev
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alDeleteSources(1, &alProxySource); // cleanup source
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return false;
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}
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alProxyContext = alcCreateContext(alProxyDev, attrs);
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checkAlcError(alProxyDev);
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if (!alProxyContext) {
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qDebug() << "Couldn't create proxy context";
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alcCloseDevice(alProxyDev); // cleanup loopback dev
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alDeleteSources(1, &alProxySource); // cleanup source
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return false;
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}
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if (!alcMakeContextCurrent(alProxyContext)) {
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qDebug() << "Cannot activate proxy context";
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alcDestroyContext(alProxyContext);
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alcCloseDevice(alProxyDev); // cleanup loopback dev
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alDeleteSources(1, &alProxySource); // cleanup source
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return false;
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}
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qDebug() << "Echo cancelation enabled";
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return true;
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}
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/**
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* @brief Open an audio output device
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*/
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bool OpenAL2::initOutput(const QString& deviceName)
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{
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assert(sinks.empty());
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outputInitialized = false;
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if (!Settings::getInstance().getAudioOutDevEnabled()) {
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return false;
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}
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qDebug() << "Opening audio output" << deviceName;
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assert(!alOutDev);
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const QByteArray qDevName = deviceName.toUtf8();
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const ALchar* tmpDevName = qDevName.isEmpty() ? nullptr : qDevName.constData();
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alOutDev = alcOpenDevice(tmpDevName);
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if (!alOutDev) {
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qWarning() << "Cannot open output audio device" << deviceName;
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return false;
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}
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qDebug() << "Opened audio output" << deviceName;
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alOutContext = alcCreateContext(alOutDev, nullptr);
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checkAlcError(alOutDev);
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if (!alcMakeContextCurrent(alOutContext)) {
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qWarning() << "Cannot create output audio context";
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return false;
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}
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// try to init echo cancellation
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echoCancelSupported = initOutputEchoCancel();
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if (!echoCancelSupported) {
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// fallback to normal, no proxy device needed
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qDebug() << "Echo cancellation disabled";
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alProxyDev = alOutDev;
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alProxyContext = alOutContext;
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}
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// init master volume
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alListenerf(AL_GAIN, Settings::getInstance().getOutVolume() * 0.01f);
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checkAlError();
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// ensure alProxyContext is active
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alcMakeContextCurrent(alProxyContext);
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outputInitialized = true;
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return true;
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}
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/**
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* @brief Close active audio output device
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*/
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void OpenAL2::cleanupOutput()
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{
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OpenAL::cleanupOutput();
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if (echoCancelSupported) {
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alcMakeContextCurrent(alOutContext);
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alSourceStop(alProxySource);
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ALint processed = 0;
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ALuint bufids[PROXY_BUFFER_COUNT];
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alGetSourcei(alProxySource, AL_BUFFERS_PROCESSED, &processed);
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alSourceUnqueueBuffers(alProxySource, processed, bufids);
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alDeleteBuffers(processed, bufids);
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alcMakeContextCurrent(nullptr);
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alcDestroyContext(alOutContext);
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alOutContext = nullptr;
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alcCloseDevice(alOutDev);
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alOutDev = nullptr;
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} else {
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alOutContext = nullptr;
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alOutDev = nullptr;
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}
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}
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/**
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* @brief Handle audio output
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*/
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void OpenAL2::doOutput()
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{
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if (!echoCancelSupported) {
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kill_filter_audio(filterer);
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filterer = nullptr;
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}
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alcMakeContextCurrent(alOutContext);
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ALuint bufids[PROXY_BUFFER_COUNT];
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ALint processed = 0, queued = 0;
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alGetSourcei(alProxySource, AL_BUFFERS_PROCESSED, &processed);
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alGetSourcei(alProxySource, AL_BUFFERS_QUEUED, &queued);
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if (processed > 0) {
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// unqueue all processed buffers
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alSourceUnqueueBuffers(alProxySource, processed, bufids);
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// delete all but the first buffer, reuse first for new data
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alDeleteBuffers(processed - 1, bufids + 1);
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} else if (queued < static_cast<ALint>(PROXY_BUFFER_COUNT)) {
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// create new buffer until the maximum is reached
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alGenBuffers(1, bufids);
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} else {
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alcMakeContextCurrent(alProxyContext);
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return;
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}
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|
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ALdouble latency[2] = {0};
|
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if (echoCancelSupported) {
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alGetSourcedvSOFT(alProxySource, AL_SEC_OFFSET_LATENCY_SOFT, latency);
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}
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checkAlError();
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ALshort outBuf[AUDIO_FRAME_SAMPLE_COUNT_PER_CHANNEL] = {0};
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if (echoCancelSupported) {
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alcMakeContextCurrent(alProxyContext);
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alcRenderSamplesSOFT(alProxyDev, outBuf, AUDIO_FRAME_SAMPLE_COUNT_PER_CHANNEL);
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checkAlcError(alProxyDev);
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alcMakeContextCurrent(alOutContext);
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}
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|
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alBufferData(bufids[0], AL_FORMAT_MONO16, outBuf, AUDIO_FRAME_SAMPLE_COUNT_PER_CHANNEL * 2,
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AUDIO_SAMPLE_RATE);
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|
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alSourceQueueBuffers(alProxySource, 1, bufids);
|
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|
||||
// initialize echo canceler if supported
|
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if (echoCancelSupported && !filterer) {
|
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filterer = new_filter_audio(AUDIO_SAMPLE_RATE);
|
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int16_t filterLatency = latency[1] * 1000 * 2 + AUDIO_FRAME_DURATION;
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qDebug() << "Setting filter delay to: " << filterLatency << "ms";
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set_echo_delay_ms(filterer, filterLatency);
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enable_disable_filters(filterer, 1, 1, 1, 0);
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}
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// do echo cancel
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pass_audio_output(filterer, outBuf, AUDIO_FRAME_SAMPLE_COUNT_PER_CHANNEL);
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ALint state;
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alGetSourcei(alProxySource, AL_SOURCE_STATE, &state);
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if (state != AL_PLAYING) {
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qDebug() << "Proxy source underflow detected";
|
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alSourcePlay(alProxySource);
|
||||
}
|
||||
alcMakeContextCurrent(alProxyContext);
|
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}
|
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|
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void OpenAL2::captureSamples(ALCdevice* device, int16_t* buffer, ALCsizei samples)
|
||||
{
|
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alcCaptureSamples(device, buffer, samples);
|
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if (echoCancelSupported && filterer) {
|
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filter_audio(filterer, buffer, samples);
|
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}
|
||||
}
|
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|
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@ -1,83 +0,0 @@
|
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/*
|
||||
Copyright © 2014-2019 by The qTox Project Contributors
|
||||
|
||||
This file is part of qTox, a Qt-based graphical interface for Tox.
|
||||
|
||||
qTox is libre 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.
|
||||
|
||||
qTox 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 qTox. If not, see <http://www.gnu.org/licenses/>.
|
||||
*/
|
||||
|
||||
|
||||
#ifndef OPENAL2_H
|
||||
#define OPENAL2_H
|
||||
|
||||
#include "src/audio/iaudiocontrol.h"
|
||||
#include "src/audio/backend/openal.h"
|
||||
|
||||
#include <atomic>
|
||||
#include <cmath>
|
||||
|
||||
#include <QMutex>
|
||||
#include <QObject>
|
||||
#include <QTimer>
|
||||
|
||||
#include <cassert>
|
||||
|
||||
#include <AL/al.h>
|
||||
#include <AL/alc.h>
|
||||
#include <AL/alext.h>
|
||||
|
||||
extern "C"
|
||||
{
|
||||
#include <filter_audio.h>
|
||||
}
|
||||
|
||||
// needed because Ubuntu 14.04 lacks the AL_SOFT_source_latency extension
|
||||
#ifndef AL_SOFT_source_latency
|
||||
#define AL_SAMPLE_OFFSET_LATENCY_SOFT 0x1200
|
||||
#define AL_SEC_OFFSET_LATENCY_SOFT 0x1201
|
||||
extern "C" typedef void(AL_APIENTRY* LPALGETSOURCEDVSOFT)(ALuint, ALenum, const ALdouble*);
|
||||
#endif
|
||||
|
||||
class OpenAL2 : public OpenAL
|
||||
{
|
||||
Q_OBJECT
|
||||
|
||||
public:
|
||||
OpenAL2();
|
||||
|
||||
protected:
|
||||
bool initInput(const QString& deviceName) override;
|
||||
bool initOutput(const QString& outDevDescr) override;
|
||||
void cleanupOutput() override;
|
||||
void playMono16SoundCleanup();
|
||||
void doOutput() override;
|
||||
bool loadOpenALExtensions(ALCdevice* dev);
|
||||
bool initOutputEchoCancel();
|
||||
void captureSamples(ALCdevice* device, int16_t* buffer, ALCsizei samples) override;
|
||||
|
||||
private:
|
||||
ALCdevice* alProxyDev;
|
||||
ALCcontext* alProxyContext;
|
||||
ALuint alProxySource;
|
||||
ALuint alProxyBuffer;
|
||||
bool echoCancelSupported = false;
|
||||
|
||||
Filter_Audio* filterer = nullptr;
|
||||
LPALCLOOPBACKOPENDEVICESOFT alcLoopbackOpenDeviceSOFT = nullptr;
|
||||
LPALCISRENDERFORMATSUPPORTEDSOFT alcIsRenderFormatSupportedSOFT = nullptr;
|
||||
LPALGETSOURCEDVSOFT alGetSourcedvSOFT = nullptr;
|
||||
LPALCRENDERSAMPLESSOFT alcRenderSamplesSOFT = nullptr;
|
||||
};
|
||||
|
||||
#endif // OPENAL2_H
|
||||
|
|
@ -57,9 +57,6 @@ public:
|
|||
virtual bool getEnableTestSound() const = 0;
|
||||
virtual void setEnableTestSound(bool newValue) = 0;
|
||||
|
||||
virtual bool getEnableBackend2() const = 0;
|
||||
virtual void setEnableBackend2(bool enabled) = 0;
|
||||
|
||||
DECLARE_SIGNAL(inDevChanged, const QString& device);
|
||||
DECLARE_SIGNAL(audioInDevEnabledChanged, bool enabled);
|
||||
|
||||
|
|
@ -71,7 +68,6 @@ public:
|
|||
DECLARE_SIGNAL(outVolumeChanged, int volume);
|
||||
DECLARE_SIGNAL(audioBitrateChanged, int bitrate);
|
||||
DECLARE_SIGNAL(enableTestSoundChanged, bool newValue);
|
||||
DECLARE_SIGNAL(enableBackend2Changed, bool enabled);
|
||||
};
|
||||
|
||||
#endif // I_AUDIO_SETTINGS_H
|
||||
|
|
|
|||
|
|
@ -243,10 +243,6 @@ void Settings::loadGlobal()
|
|||
outVolume = s.value("outVolume", 100).toInt();
|
||||
enableTestSound = s.value("enableTestSound", true).toBool();
|
||||
audioBitrate = s.value("audioBitrate", 64).toInt();
|
||||
enableBackend2 = false;
|
||||
#ifdef USE_FILTERAUDIO
|
||||
enableBackend2 = s.value("enableBackend2", false).toBool();
|
||||
#endif
|
||||
}
|
||||
s.endGroup();
|
||||
|
||||
|
|
@ -714,7 +710,6 @@ void Settings::saveGlobal()
|
|||
s.setValue("outVolume", outVolume);
|
||||
s.setValue("enableTestSound", enableTestSound);
|
||||
s.setValue("audioBitrate", audioBitrate);
|
||||
s.setValue("enableBackend2", enableBackend2);
|
||||
}
|
||||
s.endGroup();
|
||||
|
||||
|
|
@ -2012,22 +2007,6 @@ void Settings::setAudioBitrate(int bitrate)
|
|||
}
|
||||
}
|
||||
|
||||
bool Settings::getEnableBackend2() const
|
||||
{
|
||||
QMutexLocker locker{&bigLock};
|
||||
return enableBackend2;
|
||||
}
|
||||
|
||||
void Settings::setEnableBackend2(bool enabled)
|
||||
{
|
||||
QMutexLocker locker{&bigLock};
|
||||
|
||||
if (enabled != enableBackend2) {
|
||||
enableBackend2 = enabled;
|
||||
emit enableBackend2Changed(enabled);
|
||||
}
|
||||
}
|
||||
|
||||
QRect Settings::getScreenRegion() const
|
||||
{
|
||||
QMutexLocker locker(&bigLock);
|
||||
|
|
|
|||
|
|
@ -118,8 +118,6 @@ class Settings : public QObject,
|
|||
audioOutDevEnabledChanged FINAL)
|
||||
Q_PROPERTY(int outVolume READ getOutVolume WRITE setOutVolume NOTIFY outVolumeChanged FINAL)
|
||||
Q_PROPERTY(int audioBitrate READ getAudioBitrate WRITE setAudioBitrate NOTIFY audioBitrateChanged FINAL)
|
||||
Q_PROPERTY(bool enableBackend2 READ getEnableBackend2 WRITE setEnableBackend2 NOTIFY
|
||||
enableBackend2Changed FINAL)
|
||||
|
||||
// Video
|
||||
Q_PROPERTY(QString videoDev READ getVideoDev WRITE setVideoDev NOTIFY videoDevChanged FINAL)
|
||||
|
|
@ -371,9 +369,6 @@ public:
|
|||
bool getEnableTestSound() const override;
|
||||
void setEnableTestSound(bool newValue) override;
|
||||
|
||||
bool getEnableBackend2() const override;
|
||||
void setEnableBackend2(bool enabled) override;
|
||||
|
||||
SIGNAL_IMPL(Settings, inDevChanged, const QString& device)
|
||||
SIGNAL_IMPL(Settings, audioInDevEnabledChanged, bool enabled)
|
||||
|
||||
|
|
@ -385,7 +380,6 @@ public:
|
|||
SIGNAL_IMPL(Settings, outVolumeChanged, int volume)
|
||||
SIGNAL_IMPL(Settings, audioBitrateChanged, int bitrate)
|
||||
SIGNAL_IMPL(Settings, enableTestSoundChanged, bool newValue)
|
||||
SIGNAL_IMPL(Settings, enableBackend2Changed, bool enabled)
|
||||
|
||||
QString getVideoDev() const override;
|
||||
void setVideoDev(const QString& deviceSpecifier) override;
|
||||
|
|
@ -673,7 +667,6 @@ private:
|
|||
int outVolume;
|
||||
int audioBitrate;
|
||||
bool enableTestSound;
|
||||
bool enableBackend2;
|
||||
|
||||
// Video
|
||||
QString videoDev;
|
||||
|
|
|
|||
|
|
@ -62,11 +62,6 @@ AVForm::AVForm(IAudioControl& audio, CoreAV* coreAV, CameraSource& camera,
|
|||
cbEnableTestSound->setChecked(audioSettings->getEnableTestSound());
|
||||
cbEnableTestSound->setToolTip(tr("Play a test sound while changing the output volume."));
|
||||
|
||||
#ifndef USE_FILTERAUDIO
|
||||
cbEnableBackend2->setVisible(false);
|
||||
#endif
|
||||
cbEnableBackend2->setChecked(audioSettings->getEnableBackend2());
|
||||
|
||||
connect(rescanButton, &QPushButton::clicked, this, &AVForm::rescanDevices);
|
||||
|
||||
playbackSlider->setTracking(false);
|
||||
|
|
@ -179,11 +174,6 @@ void AVForm::setVolume(float value)
|
|||
volumeDisplay->setValue(getStepsFromValue(value, audio.minOutputVolume(), audio.maxOutputVolume()));
|
||||
}
|
||||
|
||||
void AVForm::on_cbEnableBackend2_stateChanged()
|
||||
{
|
||||
audioSettings->setEnableBackend2(cbEnableBackend2->isChecked());
|
||||
}
|
||||
|
||||
void AVForm::on_videoModescomboBox_currentIndexChanged(int index)
|
||||
{
|
||||
assert(0 <= index && index < videoModes.size());
|
||||
|
|
|
|||
|
|
@ -84,8 +84,6 @@ private slots:
|
|||
void rescanDevices();
|
||||
void setVolume(float value);
|
||||
|
||||
void on_cbEnableBackend2_stateChanged();
|
||||
|
||||
protected:
|
||||
void updateVideoModes(int curIndex);
|
||||
|
||||
|
|
|
|||
|
|
@ -124,12 +124,12 @@
|
|||
</item>
|
||||
<item row="5" column="1" colspan="2">
|
||||
<widget class="QProgressBar" name="volumeDisplay">
|
||||
<property name="orientation">
|
||||
<enum>Qt::Horizontal</enum>
|
||||
</property>
|
||||
<property name="textVisible">
|
||||
<bool>false</bool>
|
||||
</property>
|
||||
<property name="orientation">
|
||||
<enum>Qt::Horizontal</enum>
|
||||
</property>
|
||||
</widget>
|
||||
</item>
|
||||
<item row="0" column="0">
|
||||
|
|
@ -159,19 +159,6 @@
|
|||
</property>
|
||||
</widget>
|
||||
</item>
|
||||
<item row="7" column="0" colspan="3">
|
||||
<widget class="QCheckBox" name="cbEnableBackend2">
|
||||
<property name="enabled">
|
||||
<bool>true</bool>
|
||||
</property>
|
||||
<property name="toolTip">
|
||||
<string>Enables the experimental audio backend with echo cancelling support, needs qTox restart to take effect.</string>
|
||||
</property>
|
||||
<property name="text">
|
||||
<string>Enable experimental audio backend</string>
|
||||
</property>
|
||||
</widget>
|
||||
</item>
|
||||
</layout>
|
||||
</widget>
|
||||
</item>
|
||||
|
|
|
|||
|
|
@ -95,7 +95,6 @@ workspace
|
|||
│ ├── dep-cache
|
||||
│ │ ├── libexif
|
||||
│ │ ├── libffmpeg
|
||||
│ │ ├── libfilteraudio
|
||||
│ │ ├── libopenal
|
||||
│ │ ├── libopenssl
|
||||
│ │ ├── libopus
|
||||
|
|
@ -114,7 +113,6 @@ workspace
|
|||
├── dep-cache
|
||||
│ ├── libexif
|
||||
│ ├── libffmpeg
|
||||
│ ├── libfilteraudio
|
||||
│ ├── libopenal
|
||||
│ ├── libopenssl
|
||||
│ ├── libopus
|
||||
|
|
|
|||
|
|
@ -611,136 +611,6 @@ else
|
|||
fi
|
||||
|
||||
|
||||
# Filteraudio
|
||||
|
||||
FILTERAUDIO_PREFIX_DIR="$DEP_DIR/libfilteraudio"
|
||||
FILTERAUDIO_VERSION=ada2f4fdc04940cdeee47caffe43add4fa017096
|
||||
FILTERAUDIO_HASH="cf481e87c860aaf28b50d125195d84556f36d0ebb529d7ebdac39f8cc497256a"
|
||||
if [ ! -f "$FILTERAUDIO_PREFIX_DIR/done" ]
|
||||
then
|
||||
rm -rf "$FILTERAUDIO_PREFIX_DIR"
|
||||
mkdir -p "$FILTERAUDIO_PREFIX_DIR"
|
||||
|
||||
git clone https://github.com/irungentoo/filter_audio filter_audio
|
||||
cd filter*
|
||||
git checkout $FILTERAUDIO_VERSION
|
||||
check_sha256_git "$FILTERAUDIO_HASH"
|
||||
|
||||
$ARCH-w64-mingw32-gcc -O2 -g0 -c \
|
||||
aec/aec_core.c \
|
||||
aec/aec_core_sse2.c \
|
||||
aec/aec_rdft.c \
|
||||
aec/aec_rdft_sse2.c \
|
||||
aec/aec_resampler.c \
|
||||
aec/echo_cancellation.c \
|
||||
agc/analog_agc.c \
|
||||
agc/digital_agc.c \
|
||||
ns/ns_core.c \
|
||||
ns/nsx_core_c.c \
|
||||
ns/nsx_core.c \
|
||||
ns/noise_suppression_x.c \
|
||||
ns/noise_suppression.c \
|
||||
other/get_scaling_square.c \
|
||||
other/resample_by_2.c \
|
||||
other/spl_sqrt.c \
|
||||
other/delay_estimator.c \
|
||||
other/complex_bit_reverse.c \
|
||||
other/dot_product_with_scale.c \
|
||||
other/cross_correlation.c \
|
||||
other/min_max_operations.c \
|
||||
other/resample_48khz.c \
|
||||
other/high_pass_filter.c \
|
||||
other/energy.c \
|
||||
other/randomization_functions.c \
|
||||
other/speex_resampler.c \
|
||||
other/copy_set_operations.c \
|
||||
other/downsample_fast.c \
|
||||
other/complex_fft.c \
|
||||
other/vector_scaling_operations.c \
|
||||
other/resample_by_2_internal.c \
|
||||
other/delay_estimator_wrapper.c \
|
||||
other/real_fft.c \
|
||||
other/spl_sqrt_floor.c \
|
||||
other/resample_fractional.c \
|
||||
other/ring_buffer.c \
|
||||
other/splitting_filter.c \
|
||||
other/fft4g.c \
|
||||
other/division_operations.c \
|
||||
other/spl_init.c \
|
||||
other/float_util.c \
|
||||
zam/filters.c \
|
||||
vad/vad_sp.c \
|
||||
vad/vad_core.c \
|
||||
vad/webrtc_vad.c \
|
||||
vad/vad_gmm.c \
|
||||
vad/vad_filterbank.c \
|
||||
filter_audio.c \
|
||||
-lpthread \
|
||||
-lm
|
||||
|
||||
ar rcs libfilteraudio.a \
|
||||
aec_core.o \
|
||||
aec_core_sse2.o \
|
||||
aec_rdft.o \
|
||||
aec_rdft_sse2.o \
|
||||
aec_resampler.o \
|
||||
echo_cancellation.o \
|
||||
analog_agc.o \
|
||||
digital_agc.o \
|
||||
ns_core.o \
|
||||
nsx_core_c.o \
|
||||
nsx_core.o \
|
||||
noise_suppression_x.o \
|
||||
noise_suppression.o \
|
||||
get_scaling_square.o \
|
||||
resample_by_2.o \
|
||||
spl_sqrt.o \
|
||||
delay_estimator.o \
|
||||
complex_bit_reverse.o \
|
||||
dot_product_with_scale.o \
|
||||
cross_correlation.o \
|
||||
min_max_operations.o \
|
||||
resample_48khz.o \
|
||||
high_pass_filter.o \
|
||||
energy.o \
|
||||
randomization_functions.o \
|
||||
speex_resampler.o \
|
||||
copy_set_operations.o \
|
||||
downsample_fast.o \
|
||||
complex_fft.o \
|
||||
vector_scaling_operations.o \
|
||||
resample_by_2_internal.o \
|
||||
delay_estimator_wrapper.o \
|
||||
real_fft.o \
|
||||
spl_sqrt_floor.o \
|
||||
resample_fractional.o \
|
||||
ring_buffer.o \
|
||||
splitting_filter.o \
|
||||
fft4g.o \
|
||||
division_operations.o \
|
||||
spl_init.o \
|
||||
float_util.o \
|
||||
filters.o \
|
||||
vad_sp.o \
|
||||
vad_core.o \
|
||||
webrtc_vad.o \
|
||||
vad_gmm.o \
|
||||
vad_filterbank.o \
|
||||
filter_audio.o
|
||||
|
||||
mkdir $FILTERAUDIO_PREFIX_DIR/include
|
||||
mkdir $FILTERAUDIO_PREFIX_DIR/lib
|
||||
cp filter_audio.h $FILTERAUDIO_PREFIX_DIR/include
|
||||
cp libfilteraudio.a $FILTERAUDIO_PREFIX_DIR/lib
|
||||
echo -n $FILTERAUDIO_VERSION > $FILTERAUDIO_PREFIX_DIR/done
|
||||
|
||||
cd ..
|
||||
rm -rf ./filter*
|
||||
else
|
||||
echo "Using cached build of Filteraudio `cat $FILTERAUDIO_PREFIX_DIR/done`"
|
||||
fi
|
||||
|
||||
|
||||
# QREncode
|
||||
|
||||
QRENCODE_PREFIX_DIR="$DEP_DIR/libqrencode"
|
||||
|
|
|
|||
Loading…
Reference in New Issue
Block a user