mirror of
https://github.com/xiaoyifang/goldendict-ng.git
synced 2024-11-23 20:14:05 +00:00
feature: remove libao dependency and use QAudioSink(QAudioOutput) to play the pcm audio format
This commit is contained in:
Xiao YiFang
parent
e3d904f8b8
commit
85aad0f80c
187
audiooutput.cpp
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187
audiooutput.cpp
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@ -0,0 +1,187 @@
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#include "audiooutput.h"
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#include <QAudioFormat>
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#include <QDebug>
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#include <QtConcurrent/qtconcurrentrun.h>
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#include <QFuture>
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#include <QWaitCondition>
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#include <QCoreApplication>
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#include <QThreadPool>
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#if QT_VERSION < QT_VERSION_CHECK( 6, 0, 0 )
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#include <QAudioOutput>
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#else
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#include <QAudioSink>
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#endif
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#include <QtGlobal>
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#include <QBuffer>
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static QAudioFormat format( int sampleRate, int channelCount )
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{
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QAudioFormat out;
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out.setSampleRate( sampleRate );
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out.setChannelCount( 2 );
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#if QT_VERSION < QT_VERSION_CHECK( 6, 0, 0 )
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out.setByteOrder( QAudioFormat::LittleEndian );
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out.setCodec( QLatin1String( "audio/pcm" ) );
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#endif
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#if QT_VERSION < QT_VERSION_CHECK( 6, 0, 0 )
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out.setSampleSize( 16 );
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out.setSampleType( QAudioFormat::SignedInt );
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#else
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out.setSampleFormat( QAudioFormat::Int16 );
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#endif
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return out;
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}
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class AudioOutputPrivate: public QIODevice
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{
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public:
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AudioOutputPrivate()
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{
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open( QIODevice::ReadOnly );
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threadPool.setMaxThreadCount( 1 );
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}
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QFuture< void > audioPlayFuture;
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#if QT_VERSION < QT_VERSION_CHECK( 6, 0, 0 )
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using AudioOutput = QAudioOutput;
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#else
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using AudioOutput = QAudioSink;
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#endif
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AudioOutput * audioOutput = nullptr;
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QByteArray buffer;
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qint64 offset = 0;
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bool quit = 0;
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QMutex mutex;
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QWaitCondition cond;
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QThreadPool threadPool;
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int sampleRate = 0;
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int channels = 0;
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void setAudioFormat( int _sampleRate, int _channels )
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{
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sampleRate = _sampleRate;
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channels = _channels;
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}
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qint64 readData( char * data, qint64 len ) override
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{
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if( !len )
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return 0;
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QMutexLocker locker( &mutex );
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qint64 bytesWritten = 0;
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while( len && !quit )
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{
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if( buffer.isEmpty() )
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{
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// Wait for more frames
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if( bytesWritten == 0 )
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cond.wait( &mutex );
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if( buffer.isEmpty() )
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break;
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}
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auto sampleData = buffer.data();
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const int toWrite = qMin( (qint64) buffer.size(), len );
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memcpy( &data[bytesWritten], sampleData, toWrite );
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buffer.remove( 0, toWrite );
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bytesWritten += toWrite;
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// data += toWrite;
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len -= toWrite;
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}
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return bytesWritten;
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}
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qint64 writeData( const char *, qint64 ) override { return 0; }
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qint64 size() const override { return buffer.size(); }
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qint64 bytesAvailable() const override { return buffer.size(); }
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bool isSequential() const override { return true; }
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bool atEnd() const override { return buffer.isEmpty(); }
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void init( const QAudioFormat & fmt )
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{
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if( !audioOutput || ( fmt.isValid() && audioOutput->format() != fmt )
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|| audioOutput->state() == QAudio::StoppedState )
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{
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if( audioOutput )
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audioOutput->deleteLater();
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audioOutput = new AudioOutput( fmt );
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QObject::connect( audioOutput, &AudioOutput::stateChanged, audioOutput, [ & ]( QAudio::State state ) {
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switch( state )
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{
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case QAudio::StoppedState:
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if( audioOutput->error() != QAudio::NoError )
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qWarning() << "QAudioOutput stopped:" << audioOutput->error();
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break;
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default:
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break;
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}
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} );
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audioOutput->start( this );
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}
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// audioOutput->setVolume(volume);
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}
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void doPlayAudio()
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{
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while( !quit )
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{
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QMutexLocker locker( &mutex );
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cond.wait( &mutex, 10 );
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auto fmt = sampleRate == 0 ? QAudioFormat() : format( sampleRate, channels );
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locker.unlock();
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if( fmt.isValid() )
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init( fmt );
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QCoreApplication::processEvents();
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}
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if( audioOutput )
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{
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audioOutput->stop();
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audioOutput->deleteLater();
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}
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audioOutput = nullptr;
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}
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};
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AudioOutput::AudioOutput( QObject * parent ): QObject( parent ), d_ptr( new AudioOutputPrivate )
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{
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#if QT_VERSION < QT_VERSION_CHECK( 6, 0, 0 )
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d_ptr->audioPlayFuture = QtConcurrent::run( &d_ptr->threadPool, d_ptr.data(), &AudioOutputPrivate::doPlayAudio );
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#else
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d_ptr->audioPlayFuture = QtConcurrent::run( &d_ptr->threadPool, &AudioOutputPrivate::doPlayAudio, d_ptr.data() );
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#endif
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}
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void AudioOutput::setAudioFormat( int sampleRate, int channels ) { d_ptr->setAudioFormat( sampleRate, channels ); }
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AudioOutput::~AudioOutput()
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{
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Q_D( AudioOutput );
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d->quit = true;
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d->cond.wakeAll();
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d->audioPlayFuture.waitForFinished();
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}
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bool AudioOutput::play( const uint8_t * data, qint64 len )
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{
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Q_D( AudioOutput );
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if( d->quit )
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return false;
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QMutexLocker locker( &d->mutex );
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auto cuint = const_cast< uint8_t * >( data );
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auto cptr = reinterpret_cast< char * >( cuint );
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d->buffer.append( cptr, len );
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d->cond.wakeAll();
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return true;
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}
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25
audiooutput.h
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25
audiooutput.h
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@ -0,0 +1,25 @@
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#ifndef AUDIOOUTPUT_H
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#define AUDIOOUTPUT_H
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#include <QObject>
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#include <QScopedPointer>
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class AudioOutputPrivate;
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class AudioOutput: public QObject
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{
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public:
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AudioOutput( QObject * parent = nullptr );
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~AudioOutput();
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bool play( const uint8_t * data, qint64 len );
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void setAudioFormat( int sampleRate, int channels );
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protected:
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QScopedPointer< AudioOutputPrivate > d_ptr;
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private:
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Q_DISABLE_COPY( AudioOutput )
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Q_DECLARE_PRIVATE( AudioOutput )
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};
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#endif // AUDIOOUTPUT_H
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@ -224,7 +224,7 @@ public:
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private:
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#ifdef MAKE_FFMPEG_PLAYER
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static InternalPlayerBackend ffmpeg()
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{ return InternalPlayerBackend( "FFmpeg+libao" ); }
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{ return InternalPlayerBackend( "FFmpeg" ); }
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#endif
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#ifdef MAKE_QTMULTIMEDIA_PLAYER
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197
ffmpegaudio.cc
197
ffmpegaudio.cc
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@ -1,20 +1,11 @@
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#ifdef MAKE_FFMPEG_PLAYER
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#include "audiooutput.h"
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#include "ffmpegaudio.hh"
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#include <math.h>
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#include <errno.h>
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#ifndef INT64_C
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#define INT64_C(c) (c ## LL)
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#endif
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#ifndef UINT64_C
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#define UINT64_C(c) (c ## ULL)
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#endif
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#include <ao/ao.h>
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extern "C" {
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#include <libavcodec/avcodec.h>
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#include <libavformat/avformat.h>
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@ -27,7 +18,11 @@ extern "C" {
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#include <QDebug>
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#include <vector>
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#if( QT_VERSION >= QT_VERSION_CHECK( 6, 2, 0 ) )
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#include <QMediaDevices>
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#include <QAudioDevice>
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#endif
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#include "gddebug.hh"
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#include "utils.hh"
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@ -53,13 +48,13 @@ AudioService & AudioService::instance()
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AudioService::AudioService()
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{
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ao_initialize();
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// ao_initialize();
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}
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AudioService::~AudioService()
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{
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emit cancelPlaying( true );
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ao_shutdown();
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// ao_shutdown();
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}
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void AudioService::playMemory( const char * ptr, int size )
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@ -100,7 +95,8 @@ struct DecoderContext
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AVCodecContext * codecContext_;
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AVIOContext * avioContext_;
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AVStream * audioStream_;
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ao_device * aoDevice_;
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// ao_device * aoDevice_;
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AudioOutput * audioOutput;
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bool avformatOpened_;
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SwrContext *swr_;
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@ -113,7 +109,7 @@ struct DecoderContext
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bool openOutputDevice( QString & errorString );
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void closeOutputDevice();
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bool play( QString & errorString );
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bool normalizeAudio( AVFrame * frame, vector<char> & samples );
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bool normalizeAudio( AVFrame * frame, vector<uint8_t> & samples );
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void playFrame( AVFrame * frame );
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};
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@ -126,7 +122,7 @@ DecoderContext::DecoderContext( QByteArray const & audioData, QAtomicInt & isCan
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codecContext_( NULL ),
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avioContext_( NULL ),
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audioStream_( NULL ),
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aoDevice_( NULL ),
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audioOutput( new AudioOutput ),
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avformatOpened_( false ),
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swr_( NULL )
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{
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@ -243,15 +239,15 @@ bool DecoderContext::openCodec( QString & errorString )
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gdDebug( "Codec open: %s: channels: %d, rate: %d, format: %s\n", codec_->long_name,
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codecContext_->channels, codecContext_->sample_rate, av_get_sample_fmt_name( codecContext_->sample_fmt ) );
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if ( codecContext_->sample_fmt == AV_SAMPLE_FMT_S32 ||
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codecContext_->sample_fmt == AV_SAMPLE_FMT_S32P ||
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codecContext_->sample_fmt == AV_SAMPLE_FMT_FLT ||
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codecContext_->sample_fmt == AV_SAMPLE_FMT_FLTP ||
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codecContext_->sample_fmt == AV_SAMPLE_FMT_DBL ||
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codecContext_->sample_fmt == AV_SAMPLE_FMT_DBLP )
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// if ( codecContext_->sample_fmt == AV_SAMPLE_FMT_S32 ||
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// codecContext_->sample_fmt == AV_SAMPLE_FMT_S32P ||
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// codecContext_->sample_fmt == AV_SAMPLE_FMT_FLT ||
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// codecContext_->sample_fmt == AV_SAMPLE_FMT_FLTP ||
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// codecContext_->sample_fmt == AV_SAMPLE_FMT_DBL ||
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// codecContext_->sample_fmt == AV_SAMPLE_FMT_DBLP )
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{
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swr_ = swr_alloc_set_opts( NULL,
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codecContext_->channel_layout,
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av_get_default_channel_layout(2),
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AV_SAMPLE_FMT_S16,
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codecContext_->sample_rate,
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codecContext_->channel_layout,
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@ -317,75 +313,25 @@ void DecoderContext::closeCodec()
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bool DecoderContext::openOutputDevice( QString & errorString )
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{
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// Prepare for audio output
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int aoDriverId = ao_default_driver_id();
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ao_info * aoDrvInfo = ao_driver_info( aoDriverId );
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if ( aoDriverId < 0 || !aoDrvInfo )
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{
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errorString = QObject::tr( "Cannot find usable audio output device." );
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return false;
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}
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ao_sample_format aoSampleFormat;
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memset (&aoSampleFormat, 0, sizeof(aoSampleFormat) );
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aoSampleFormat.channels = codecContext_->channels;
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aoSampleFormat.rate = codecContext_->sample_rate;
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aoSampleFormat.byte_format = AO_FMT_NATIVE;
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aoSampleFormat.matrix = 0;
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aoSampleFormat.bits = qMin( 16, av_get_bytes_per_sample( codecContext_->sample_fmt ) << 3 );
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if ( aoSampleFormat.bits == 0 )
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{
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errorString = QObject::tr( "Unsupported sample format." );
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return false;
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}
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gdDebug( "ao_open_live(): %s: channels: %d, rate: %d, bits: %d\n",
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aoDrvInfo->name, aoSampleFormat.channels, aoSampleFormat.rate, aoSampleFormat.bits );
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aoDevice_ = ao_open_live( aoDriverId, &aoSampleFormat, NULL );
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if ( !aoDevice_ )
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{
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errorString = QObject::tr( "ao_open_live() failed: " );
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switch ( errno )
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{
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case AO_ENODRIVER:
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errorString += QObject::tr( "No driver." );
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break;
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case AO_ENOTLIVE:
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errorString += QObject::tr( "This driver is not a live output device." );
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break;
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case AO_EBADOPTION:
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errorString += QObject::tr( "A valid option key has an invalid value." );
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break;
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case AO_EOPENDEVICE:
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errorString += QObject::tr( "Cannot open the device: %1, channels: %2, rate: %3, bits: %4." )
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.arg( aoDrvInfo->short_name )
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.arg( aoSampleFormat.channels )
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.arg( aoSampleFormat.rate )
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.arg( aoSampleFormat.bits );
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break;
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default:
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errorString += QObject::tr( "Unknown error." );
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break;
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}
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// only check device when qt version is greater than 6.2
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#if (QT_VERSION >= QT_VERSION_CHECK(6,2,0))
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QAudioDevice m_outputDevice = QMediaDevices::defaultAudioOutput();
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if(m_outputDevice.isNull()){
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errorString += QObject::tr( "Can not found default audio output device" );
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return false;
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}
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#endif
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audioOutput->setAudioFormat( codecContext_->sample_rate, codecContext_->channels );
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return true;
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}
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void DecoderContext::closeOutputDevice()
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{
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// ao_close() is synchronous, it will wait until all audio streams flushed
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if ( aoDevice_ )
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{
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ao_close( aoDevice_ );
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aoDevice_ = NULL;
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}
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// if(audioOutput){
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// delete audioOutput;
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// audioOutput = 0;
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// }
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}
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bool DecoderContext::play( QString & errorString )
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@ -440,79 +386,17 @@ bool DecoderContext::play( QString & errorString )
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return true;
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}
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bool DecoderContext::normalizeAudio( AVFrame * frame, vector<char> & samples )
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bool DecoderContext::normalizeAudio( AVFrame * frame, vector<uint8_t > & samples )
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{
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int lineSize = 0;
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int dataSize = av_samples_get_buffer_size( &lineSize, codecContext_->channels,
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frame->nb_samples, codecContext_->sample_fmt, 1 );
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// int dataSize = av_samples_get_buffer_size( &lineSize, codecContext_->channels,
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// frame->nb_samples, codecContext_->sample_fmt, 1 );
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int dataSize = frame->nb_samples * 2 * 2;
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samples.resize( dataSize );
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uint8_t *data[2] = { 0 };
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data[0] = &samples.front(); //输出格式为AV_SAMPLE_FMT_S16(packet类型),所以转换后的LR两通道都存在data[0]中
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// Portions from: https://code.google.com/p/lavfilters/source/browse/decoder/LAVAudio/LAVAudio.cpp
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// But this one use 8, 16, 32 bits integer, respectively.
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switch ( codecContext_->sample_fmt )
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{
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case AV_SAMPLE_FMT_U8:
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case AV_SAMPLE_FMT_S16:
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{
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samples.resize( dataSize );
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memcpy( &samples.front(), frame->data[0], lineSize );
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}
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break;
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// Planar
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case AV_SAMPLE_FMT_U8P:
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{
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samples.resize( dataSize );
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uint8_t * out = ( uint8_t * )&samples.front();
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for ( int i = 0; i < frame->nb_samples; i++ )
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{
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for ( int ch = 0; ch < codecContext_->channels; ch++ )
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{
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*out++ = ( ( uint8_t * )frame->extended_data[ch] )[i];
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}
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}
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}
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break;
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case AV_SAMPLE_FMT_S16P:
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{
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samples.resize( dataSize );
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int16_t * out = ( int16_t * )&samples.front();
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for ( int i = 0; i < frame->nb_samples; i++ )
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{
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for ( int ch = 0; ch < codecContext_->channels; ch++ )
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{
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*out++ = ( ( int16_t * )frame->extended_data[ch] )[i];
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}
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}
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}
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break;
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case AV_SAMPLE_FMT_S32:
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/* Pass through */
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case AV_SAMPLE_FMT_S32P:
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/* Pass through */
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case AV_SAMPLE_FMT_FLT:
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/* Pass through */
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case AV_SAMPLE_FMT_FLTP:
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/* Pass through */
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{
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samples.resize( dataSize / 2 );
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|
||||
uint8_t *out = ( uint8_t * )&samples.front();
|
||||
swr_convert( swr_, &out, frame->nb_samples, (const uint8_t**)frame->extended_data, frame->nb_samples );
|
||||
}
|
||||
break;
|
||||
case AV_SAMPLE_FMT_DBL:
|
||||
case AV_SAMPLE_FMT_DBLP:
|
||||
{
|
||||
samples.resize( dataSize / 4 );
|
||||
|
||||
uint8_t *out = ( uint8_t * )&samples.front();
|
||||
swr_convert( swr_, &out, frame->nb_samples, (const uint8_t**)frame->extended_data, frame->nb_samples );
|
||||
}
|
||||
break;
|
||||
default:
|
||||
return false;
|
||||
}
|
||||
swr_convert( swr_, data, frame->nb_samples, (const uint8_t**)frame->data, frame->nb_samples );
|
||||
|
||||
return true;
|
||||
}
|
||||
|
|
@ -522,9 +406,12 @@ void DecoderContext::playFrame( AVFrame * frame )
|
|||
if ( !frame )
|
||||
return;
|
||||
|
||||
vector<char> samples;
|
||||
vector<uint8_t> samples;
|
||||
if ( normalizeAudio( frame, samples ) )
|
||||
ao_play( aoDevice_, &samples.front(), samples.size() );
|
||||
{
|
||||
// ao_play( aoDevice_, &samples.front(), samples.size() );
|
||||
audioOutput->play(&samples.front(), samples.size());
|
||||
}
|
||||
}
|
||||
|
||||
DecoderThread::DecoderThread( QByteArray const & audioData, QObject * parent ) :
|
||||
|
|
|
|||
|
|
@ -263,6 +263,7 @@ HEADERS += folding.hh \
|
|||
ankiconnector.h \
|
||||
article_inspect.h \
|
||||
articlewebpage.h \
|
||||
audiooutput.h \
|
||||
base/globalregex.hh \
|
||||
globalbroadcaster.h \
|
||||
headwordsmodel.h \
|
||||
|
|
@ -407,6 +408,7 @@ SOURCES += folding.cc \
|
|||
ankiconnector.cpp \
|
||||
article_inspect.cpp \
|
||||
articlewebpage.cpp \
|
||||
audiooutput.cpp \
|
||||
base/globalregex.cc \
|
||||
globalbroadcaster.cpp \
|
||||
headwordsmodel.cpp \
|
||||
|
|
|
|||
Loading…
Reference in a new issue