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feature: remove libao dependency and use QAudioSink(QAudioOutput) to play the pcm audio format

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Xiao YiFang 2022-11-05 13:44:10 +08:00
parent e3d904f8b8
commit 85aad0f80c
5 changed files with 257 additions and 156 deletions
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187
audiooutput.cpp Normal file
View file

@ -0,0 +1,187 @@
#include "audiooutput.h"
#include <QAudioFormat>
#include <QDebug>
#include <QtConcurrent/qtconcurrentrun.h>
#include <QFuture>
#include <QWaitCondition>
#include <QCoreApplication>
#include <QThreadPool>
#if QT_VERSION < QT_VERSION_CHECK( 6, 0, 0 )
#include <QAudioOutput>
#else
#include <QAudioSink>
#endif
#include <QtGlobal>
#include <QBuffer>
static QAudioFormat format( int sampleRate, int channelCount )
{
QAudioFormat out;
out.setSampleRate( sampleRate );
out.setChannelCount( 2 );
#if QT_VERSION < QT_VERSION_CHECK( 6, 0, 0 )
out.setByteOrder( QAudioFormat::LittleEndian );
out.setCodec( QLatin1String( "audio/pcm" ) );
#endif
#if QT_VERSION < QT_VERSION_CHECK( 6, 0, 0 )
out.setSampleSize( 16 );
out.setSampleType( QAudioFormat::SignedInt );
#else
out.setSampleFormat( QAudioFormat::Int16 );
#endif
return out;
}
class AudioOutputPrivate: public QIODevice
{
public:
AudioOutputPrivate()
{
open( QIODevice::ReadOnly );
threadPool.setMaxThreadCount( 1 );
}
QFuture< void > audioPlayFuture;
#if QT_VERSION < QT_VERSION_CHECK( 6, 0, 0 )
using AudioOutput = QAudioOutput;
#else
using AudioOutput = QAudioSink;
#endif
AudioOutput * audioOutput = nullptr;
QByteArray buffer;
qint64 offset = 0;
bool quit = 0;
QMutex mutex;
QWaitCondition cond;
QThreadPool threadPool;
int sampleRate = 0;
int channels = 0;
void setAudioFormat( int _sampleRate, int _channels )
{
sampleRate = _sampleRate;
channels = _channels;
}
qint64 readData( char * data, qint64 len ) override
{
if( !len )
return 0;
QMutexLocker locker( &mutex );
qint64 bytesWritten = 0;
while( len && !quit )
{
if( buffer.isEmpty() )
{
// Wait for more frames
if( bytesWritten == 0 )
cond.wait( &mutex );
if( buffer.isEmpty() )
break;
}
auto sampleData = buffer.data();
const int toWrite = qMin( (qint64) buffer.size(), len );
memcpy( &data[bytesWritten], sampleData, toWrite );
buffer.remove( 0, toWrite );
bytesWritten += toWrite;
// data += toWrite;
len -= toWrite;
}
return bytesWritten;
}
qint64 writeData( const char *, qint64 ) override { return 0; }
qint64 size() const override { return buffer.size(); }
qint64 bytesAvailable() const override { return buffer.size(); }
bool isSequential() const override { return true; }
bool atEnd() const override { return buffer.isEmpty(); }
void init( const QAudioFormat & fmt )
{
if( !audioOutput || ( fmt.isValid() && audioOutput->format() != fmt )
|| audioOutput->state() == QAudio::StoppedState )
{
if( audioOutput )
audioOutput->deleteLater();
audioOutput = new AudioOutput( fmt );
QObject::connect( audioOutput, &AudioOutput::stateChanged, audioOutput, [ & ]( QAudio::State state ) {
switch( state )
{
case QAudio::StoppedState:
if( audioOutput->error() != QAudio::NoError )
qWarning() << "QAudioOutput stopped:" << audioOutput->error();
break;
default:
break;
}
} );
audioOutput->start( this );
}
// audioOutput->setVolume(volume);
}
void doPlayAudio()
{
while( !quit )
{
QMutexLocker locker( &mutex );
cond.wait( &mutex, 10 );
auto fmt = sampleRate == 0 ? QAudioFormat() : format( sampleRate, channels );
locker.unlock();
if( fmt.isValid() )
init( fmt );
QCoreApplication::processEvents();
}
if( audioOutput )
{
audioOutput->stop();
audioOutput->deleteLater();
}
audioOutput = nullptr;
}
};
AudioOutput::AudioOutput( QObject * parent ): QObject( parent ), d_ptr( new AudioOutputPrivate )
{
#if QT_VERSION < QT_VERSION_CHECK( 6, 0, 0 )
d_ptr->audioPlayFuture = QtConcurrent::run( &d_ptr->threadPool, d_ptr.data(), &AudioOutputPrivate::doPlayAudio );
#else
d_ptr->audioPlayFuture = QtConcurrent::run( &d_ptr->threadPool, &AudioOutputPrivate::doPlayAudio, d_ptr.data() );
#endif
}
void AudioOutput::setAudioFormat( int sampleRate, int channels ) { d_ptr->setAudioFormat( sampleRate, channels ); }
AudioOutput::~AudioOutput()
{
Q_D( AudioOutput );
d->quit = true;
d->cond.wakeAll();
d->audioPlayFuture.waitForFinished();
}
bool AudioOutput::play( const uint8_t * data, qint64 len )
{
Q_D( AudioOutput );
if( d->quit )
return false;
QMutexLocker locker( &d->mutex );
auto cuint = const_cast< uint8_t * >( data );
auto cptr = reinterpret_cast< char * >( cuint );
d->buffer.append( cptr, len );
d->cond.wakeAll();
return true;
}

25
audiooutput.h Normal file
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@ -0,0 +1,25 @@
#ifndef AUDIOOUTPUT_H
#define AUDIOOUTPUT_H
#include <QObject>
#include <QScopedPointer>
class AudioOutputPrivate;
class AudioOutput: public QObject
{
public:
AudioOutput( QObject * parent = nullptr );
~AudioOutput();
bool play( const uint8_t * data, qint64 len );
void setAudioFormat( int sampleRate, int channels );
protected:
QScopedPointer< AudioOutputPrivate > d_ptr;
private:
Q_DISABLE_COPY( AudioOutput )
Q_DECLARE_PRIVATE( AudioOutput )
};
#endif // AUDIOOUTPUT_H

2
config.hh
View file

@ -224,7 +224,7 @@ public:
private:
#ifdef MAKE_FFMPEG_PLAYER
static InternalPlayerBackend ffmpeg()
{ return InternalPlayerBackend( "FFmpeg+libao" ); }
{ return InternalPlayerBackend( "FFmpeg" ); }
#endif
#ifdef MAKE_QTMULTIMEDIA_PLAYER

197
ffmpegaudio.cc
View file

@ -1,20 +1,11 @@
#ifdef MAKE_FFMPEG_PLAYER
#include "audiooutput.h"
#include "ffmpegaudio.hh"
#include <math.h>
#include <errno.h>
#ifndef INT64_C
#define INT64_C(c) (c ## LL)
#endif
#ifndef UINT64_C
#define UINT64_C(c) (c ## ULL)
#endif
#include <ao/ao.h>
extern "C" {
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
@ -27,7 +18,11 @@ extern "C" {
#include <QDebug>
#include <vector>
#if( QT_VERSION >= QT_VERSION_CHECK( 6, 2, 0 ) )
#include <QMediaDevices>
#include <QAudioDevice>
#endif
#include "gddebug.hh"
#include "utils.hh"
@ -53,13 +48,13 @@ AudioService & AudioService::instance()
AudioService::AudioService()
{
ao_initialize();
// ao_initialize();
}
AudioService::~AudioService()
{
emit cancelPlaying( true );
ao_shutdown();
// ao_shutdown();
}
void AudioService::playMemory( const char * ptr, int size )
@ -100,7 +95,8 @@ struct DecoderContext
AVCodecContext * codecContext_;
AVIOContext * avioContext_;
AVStream * audioStream_;
ao_device * aoDevice_;
// ao_device * aoDevice_;
AudioOutput * audioOutput;
bool avformatOpened_;
SwrContext *swr_;
@ -113,7 +109,7 @@ struct DecoderContext
bool openOutputDevice( QString & errorString );
void closeOutputDevice();
bool play( QString & errorString );
bool normalizeAudio( AVFrame * frame, vector<char> & samples );
bool normalizeAudio( AVFrame * frame, vector<uint8_t> & samples );
void playFrame( AVFrame * frame );
};
@ -126,7 +122,7 @@ DecoderContext::DecoderContext( QByteArray const & audioData, QAtomicInt & isCan
codecContext_( NULL ),
avioContext_( NULL ),
audioStream_( NULL ),
aoDevice_( NULL ),
audioOutput( new AudioOutput ),
avformatOpened_( false ),
swr_( NULL )
{
@ -243,15 +239,15 @@ bool DecoderContext::openCodec( QString & errorString )
gdDebug( "Codec open: %s: channels: %d, rate: %d, format: %s\n", codec_->long_name,
codecContext_->channels, codecContext_->sample_rate, av_get_sample_fmt_name( codecContext_->sample_fmt ) );
if ( codecContext_->sample_fmt == AV_SAMPLE_FMT_S32 ||
codecContext_->sample_fmt == AV_SAMPLE_FMT_S32P ||
codecContext_->sample_fmt == AV_SAMPLE_FMT_FLT ||
codecContext_->sample_fmt == AV_SAMPLE_FMT_FLTP ||
codecContext_->sample_fmt == AV_SAMPLE_FMT_DBL ||
codecContext_->sample_fmt == AV_SAMPLE_FMT_DBLP )
// if ( codecContext_->sample_fmt == AV_SAMPLE_FMT_S32 ||
// codecContext_->sample_fmt == AV_SAMPLE_FMT_S32P ||
// codecContext_->sample_fmt == AV_SAMPLE_FMT_FLT ||
// codecContext_->sample_fmt == AV_SAMPLE_FMT_FLTP ||
// codecContext_->sample_fmt == AV_SAMPLE_FMT_DBL ||
// codecContext_->sample_fmt == AV_SAMPLE_FMT_DBLP )
{
swr_ = swr_alloc_set_opts( NULL,
codecContext_->channel_layout,
av_get_default_channel_layout(2),
AV_SAMPLE_FMT_S16,
codecContext_->sample_rate,
codecContext_->channel_layout,
@ -317,75 +313,25 @@ void DecoderContext::closeCodec()
bool DecoderContext::openOutputDevice( QString & errorString )
{
// Prepare for audio output
int aoDriverId = ao_default_driver_id();
ao_info * aoDrvInfo = ao_driver_info( aoDriverId );
if ( aoDriverId < 0 || !aoDrvInfo )
{
errorString = QObject::tr( "Cannot find usable audio output device." );
return false;
}
ao_sample_format aoSampleFormat;
memset (&aoSampleFormat, 0, sizeof(aoSampleFormat) );
aoSampleFormat.channels = codecContext_->channels;
aoSampleFormat.rate = codecContext_->sample_rate;
aoSampleFormat.byte_format = AO_FMT_NATIVE;
aoSampleFormat.matrix = 0;
aoSampleFormat.bits = qMin( 16, av_get_bytes_per_sample( codecContext_->sample_fmt ) << 3 );
if ( aoSampleFormat.bits == 0 )
{
errorString = QObject::tr( "Unsupported sample format." );
return false;
}
gdDebug( "ao_open_live(): %s: channels: %d, rate: %d, bits: %d\n",
aoDrvInfo->name, aoSampleFormat.channels, aoSampleFormat.rate, aoSampleFormat.bits );
aoDevice_ = ao_open_live( aoDriverId, &aoSampleFormat, NULL );
if ( !aoDevice_ )
{
errorString = QObject::tr( "ao_open_live() failed: " );
switch ( errno )
{
case AO_ENODRIVER:
errorString += QObject::tr( "No driver." );
break;
case AO_ENOTLIVE:
errorString += QObject::tr( "This driver is not a live output device." );
break;
case AO_EBADOPTION:
errorString += QObject::tr( "A valid option key has an invalid value." );
break;
case AO_EOPENDEVICE:
errorString += QObject::tr( "Cannot open the device: %1, channels: %2, rate: %3, bits: %4." )
.arg( aoDrvInfo->short_name )
.arg( aoSampleFormat.channels )
.arg( aoSampleFormat.rate )
.arg( aoSampleFormat.bits );
break;
default:
errorString += QObject::tr( "Unknown error." );
break;
}
// only check device when qt version is greater than 6.2
#if (QT_VERSION >= QT_VERSION_CHECK(6,2,0))
QAudioDevice m_outputDevice = QMediaDevices::defaultAudioOutput();
if(m_outputDevice.isNull()){
errorString += QObject::tr( "Can not found default audio output device" );
return false;
}
#endif
audioOutput->setAudioFormat( codecContext_->sample_rate, codecContext_->channels );
return true;
}
void DecoderContext::closeOutputDevice()
{
// ao_close() is synchronous, it will wait until all audio streams flushed
if ( aoDevice_ )
{
ao_close( aoDevice_ );
aoDevice_ = NULL;
}
// if(audioOutput){
// delete audioOutput;
// audioOutput = 0;
// }
}
bool DecoderContext::play( QString & errorString )
@ -440,79 +386,17 @@ bool DecoderContext::play( QString & errorString )
return true;
}
bool DecoderContext::normalizeAudio( AVFrame * frame, vector<char> & samples )
bool DecoderContext::normalizeAudio( AVFrame * frame, vector<uint8_t > & samples )
{
int lineSize = 0;
int dataSize = av_samples_get_buffer_size( &lineSize, codecContext_->channels,
frame->nb_samples, codecContext_->sample_fmt, 1 );
// int dataSize = av_samples_get_buffer_size( &lineSize, codecContext_->channels,
// frame->nb_samples, codecContext_->sample_fmt, 1 );
int dataSize = frame->nb_samples * 2 * 2;
samples.resize( dataSize );
uint8_t *data[2] = { 0 };
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
// Portions from: https://code.google.com/p/lavfilters/source/browse/decoder/LAVAudio/LAVAudio.cpp
// But this one use 8, 16, 32 bits integer, respectively.
switch ( codecContext_->sample_fmt )
{
case AV_SAMPLE_FMT_U8:
case AV_SAMPLE_FMT_S16:
{
samples.resize( dataSize );
memcpy( &samples.front(), frame->data[0], lineSize );
}
break;
// Planar
case AV_SAMPLE_FMT_U8P:
{
samples.resize( dataSize );
uint8_t * out = ( uint8_t * )&samples.front();
for ( int i = 0; i < frame->nb_samples; i++ )
{
for ( int ch = 0; ch < codecContext_->channels; ch++ )
{
*out++ = ( ( uint8_t * )frame->extended_data[ch] )[i];
}
}
}
break;
case AV_SAMPLE_FMT_S16P:
{
samples.resize( dataSize );
int16_t * out = ( int16_t * )&samples.front();
for ( int i = 0; i < frame->nb_samples; i++ )
{
for ( int ch = 0; ch < codecContext_->channels; ch++ )
{
*out++ = ( ( int16_t * )frame->extended_data[ch] )[i];
}
}
}
break;
case AV_SAMPLE_FMT_S32:
/* Pass through */
case AV_SAMPLE_FMT_S32P:
/* Pass through */
case AV_SAMPLE_FMT_FLT:
/* Pass through */
case AV_SAMPLE_FMT_FLTP:
/* Pass through */
{
samples.resize( dataSize / 2 );
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 ) :

2
goldendict.pro
View file

@ -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 \