goldendict-ng/src/ffmpegaudio.cc
YiFang Xiao 0683afa2b0 opt: add Ctrl+Shift+S to stop the current playing sound
[autofix.ci] apply automated fixes
2023-08-25 08:44:21 +08:00

407 lines
11 KiB
C++

#ifdef MAKE_FFMPEG_PLAYER
#include "audiooutput.hh"
#include "ffmpegaudio.hh"
#include <errno.h>
extern "C" {
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/avutil.h>
#include "libswresample/swresample.h"
}
#include <QString>
#include <QDataStream>
#include <vector>
#if ( QT_VERSION >= QT_VERSION_CHECK( 6, 2, 0 ) )
#include <QMediaDevices>
#include <QAudioDevice>
#endif
#include "gddebug.hh"
#include "utils.hh"
using std::vector;
namespace Ffmpeg {
QMutex DecoderThread::deviceMutex_;
static inline QString avErrorString( int errnum )
{
char buf[ 64 ];
av_strerror( errnum, buf, 64 );
return QString::fromLatin1( buf );
}
AudioService & AudioService::instance()
{
static AudioService a;
return a;
}
AudioService::~AudioService()
{
emit cancelPlaying( true );
}
void AudioService::playMemory( const char * ptr, int size )
{
emit cancelPlaying( false );
QByteArray audioData( ptr, size );
thread = std::make_shared< DecoderThread >( audioData, this );
connect( this, &AudioService::cancelPlaying, thread.get(), [ this ]( bool waitFinished ) {
thread->cancel( waitFinished );
} );
thread->start();
}
void AudioService::stop()
{
emit cancelPlaying( false );
}
DecoderContext::DecoderContext( QByteArray const & audioData, QAtomicInt & isCancelled ):
isCancelled_( isCancelled ),
audioData_( audioData ),
audioDataStream_( audioData_ ),
formatContext_( nullptr ),
codec_( nullptr ),
codecContext_( nullptr ),
avioContext_( nullptr ),
audioStream_( nullptr ),
audioOutput( new AudioOutput ),
avformatOpened_( false ),
swr_( nullptr )
{
}
DecoderContext::~DecoderContext()
{
closeOutputDevice();
closeCodec();
}
static int readAudioData( void * opaque, unsigned char * buffer, int bufferSize )
{
QDataStream * pStream = (QDataStream *)opaque;
// This function is passed as the read_packet callback into avio_alloc_context().
// The documentation for this callback parameter states:
// For stream protocols, must never return 0 but rather a proper AVERROR code.
if ( pStream->atEnd() )
return AVERROR_EOF;
const int bytesRead = pStream->readRawData( (char *)buffer, bufferSize );
// QDataStream::readRawData() returns 0 at EOF => return AVERROR_EOF in this case.
// An error is unlikely here, so just print a warning and return AVERROR_EOF too.
if ( bytesRead < 0 )
gdWarning( "readAudioData: error while reading raw data." );
return bytesRead > 0 ? bytesRead : AVERROR_EOF;
}
bool DecoderContext::openCodec( QString & errorString )
{
formatContext_ = avformat_alloc_context();
if ( !formatContext_ ) {
errorString = "avformat_alloc_context() failed.";
return false;
}
unsigned char * avioBuffer = (unsigned char *)av_malloc( kBufferSize + AV_INPUT_BUFFER_PADDING_SIZE );
if ( !avioBuffer ) {
errorString = "av_malloc() failed.";
return false;
}
// Don't free buffer allocated here (if succeeded), it will be cleaned up automatically.
avioContext_ = avio_alloc_context( avioBuffer, kBufferSize, 0, &audioDataStream_, readAudioData, nullptr, nullptr );
if ( !avioContext_ ) {
av_free( avioBuffer );
errorString = "avio_alloc_context() failed.";
return false;
}
avioContext_->seekable = 0;
avioContext_->write_flag = 0;
// If pb not set, avformat_open_input() simply crash.
formatContext_->pb = avioContext_;
formatContext_->flags |= AVFMT_FLAG_CUSTOM_IO;
int ret = 0;
avformatOpened_ = true;
ret = avformat_open_input( &formatContext_, nullptr, nullptr, nullptr );
if ( ret < 0 ) {
errorString = QString( "avformat_open_input() failed: %1." ).arg( avErrorString( ret ) );
return false;
}
ret = avformat_find_stream_info( formatContext_, nullptr );
if ( ret < 0 ) {
errorString = QString( "avformat_find_stream_info() failed: %1." ).arg( avErrorString( ret ) );
return false;
}
// Find audio stream, use the first audio stream if available
for ( unsigned i = 0; i < formatContext_->nb_streams; i++ ) {
if ( formatContext_->streams[ i ]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO ) {
audioStream_ = formatContext_->streams[ i ];
break;
}
}
if ( !audioStream_ ) {
errorString = QString( "Could not find audio stream." );
return false;
}
codec_ = avcodec_find_decoder( audioStream_->codecpar->codec_id );
if ( !codec_ ) {
errorString = QString( "Codec [id: %1] not found." ).arg( audioStream_->codecpar->codec_id );
return false;
}
codecContext_ = avcodec_alloc_context3( codec_ );
if ( !codecContext_ ) {
errorString = QString( "avcodec_alloc_context3() failed." );
return false;
}
avcodec_parameters_to_context( codecContext_, audioStream_->codecpar );
ret = avcodec_open2( codecContext_, codec_, nullptr );
if ( ret < 0 ) {
errorString = QString( "avcodec_open2() failed: %1." ).arg( avErrorString( ret ) );
return false;
}
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 ) );
auto layout = codecContext_->channel_layout;
if ( !layout ) {
layout = av_get_default_channel_layout( codecContext_->channels );
codecContext_->channel_layout = layout;
}
swr_ = swr_alloc_set_opts( nullptr,
layout,
AV_SAMPLE_FMT_S16,
44100,
layout,
codecContext_->sample_fmt,
codecContext_->sample_rate,
0,
nullptr );
if ( !swr_ || swr_init( swr_ ) < 0 ) {
av_log( nullptr, AV_LOG_ERROR, "Cannot create sample rate converter \n" );
swr_free( &swr_ );
return false;
}
return true;
}
void DecoderContext::closeCodec()
{
if ( swr_ ) {
swr_free( &swr_ );
}
if ( !formatContext_ ) {
if ( avioContext_ ) {
av_free( avioContext_->buffer );
avioContext_ = nullptr;
}
return;
}
// avformat_open_input() is not called, just free the buffer associated with
// the AVIOContext, and the AVFormatContext
if ( !avformatOpened_ ) {
if ( formatContext_ ) {
avformat_free_context( formatContext_ );
formatContext_ = nullptr;
}
if ( avioContext_ ) {
av_free( avioContext_->buffer );
avioContext_ = nullptr;
}
return;
}
avformatOpened_ = false;
// Closing a codec context without prior avcodec_open2() will result in
// a crash in ffmpeg
if ( audioStream_ && codecContext_ && codec_ ) {
audioStream_->discard = AVDISCARD_ALL;
avcodec_close( codecContext_ );
avcodec_free_context( &codecContext_ );
}
avformat_close_input( &formatContext_ );
av_free( avioContext_->buffer );
}
bool DecoderContext::openOutputDevice( QString & errorString )
{
// 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 += QString( "Can not found default audio output device" );
return false;
}
#endif
audioOutput->setAudioFormat( 44100, codecContext_->channels );
return true;
}
void DecoderContext::closeOutputDevice() {}
bool DecoderContext::play( QString & errorString )
{
AVFrame * frame = av_frame_alloc();
if ( !frame ) {
errorString = QString( "avcodec_alloc_frame() failed." );
return false;
}
AVPacket * packet = av_packet_alloc();
while ( !Utils::AtomicInt::loadAcquire( isCancelled_ ) && av_read_frame( formatContext_, packet ) >= 0 ) {
if ( packet->stream_index == audioStream_->index ) {
int ret = avcodec_send_packet( codecContext_, packet );
/* read all the output frames (in general there may be any number of them) */
while ( ret >= 0 ) {
ret = avcodec_receive_frame( codecContext_, frame );
if ( Utils::AtomicInt::loadAcquire( isCancelled_ ) || ret < 0 )
break;
playFrame( frame );
}
}
av_packet_unref( packet );
}
/* flush the decoder */
packet->data = nullptr;
packet->size = 0;
int ret = avcodec_send_packet( codecContext_, packet );
while ( ret >= 0 ) {
ret = avcodec_receive_frame( codecContext_, frame );
if ( Utils::AtomicInt::loadAcquire( isCancelled_ ) || ret < 0 )
break;
playFrame( frame );
}
av_frame_free( &frame );
av_packet_free( &packet );
return true;
}
void DecoderContext::stop()
{
if ( audioOutput ) {
audioOutput->stop();
audioOutput->deleteLater();
audioOutput = nullptr;
}
}
bool DecoderContext::normalizeAudio( AVFrame * frame, vector< uint8_t > & samples )
{
auto dst_freq = 44100;
auto dst_channels = codecContext_->channels;
int out_count = (int64_t)frame->nb_samples * dst_freq / frame->sample_rate + 256;
int out_size = av_samples_get_buffer_size( nullptr, dst_channels, out_count, AV_SAMPLE_FMT_S16, 1 );
samples.resize( out_size );
uint8_t * data[ 2 ] = { nullptr };
data[ 0 ] = &samples.front();
auto out_nb_samples = swr_convert( swr_, data, out_count, (const uint8_t **)frame->extended_data, frame->nb_samples );
if ( out_nb_samples < 0 ) {
av_log( nullptr, AV_LOG_ERROR, "converte fail \n" );
return false;
}
else {
// qDebug( "out_count:%d, out_nb_samples:%d, frame->nb_samples:%d \n", out_count, out_nb_samples, frame->nb_samples );
}
int actual_size = av_samples_get_buffer_size( nullptr, dst_channels, out_nb_samples, AV_SAMPLE_FMT_S16, 1 );
samples.resize( actual_size );
return true;
}
void DecoderContext::playFrame( AVFrame * frame )
{
if ( !frame )
return;
vector< uint8_t > samples;
if ( normalizeAudio( frame, samples ) ) {
audioOutput->play( &samples.front(), samples.size() );
}
}
DecoderThread::DecoderThread( QByteArray const & audioData, QObject * parent ):
QThread( parent ),
isCancelled_( 0 ),
audioData_( audioData ),
d( audioData_, isCancelled_ )
{
}
DecoderThread::~DecoderThread()
{
isCancelled_.ref();
d.stop();
}
void DecoderThread::run()
{
QString errorString;
if ( !d.openCodec( errorString ) ) {
emit error( errorString );
return;
}
while ( !deviceMutex_.tryLock( 100 ) ) {
if ( Utils::AtomicInt::loadAcquire( isCancelled_ ) )
return;
}
if ( !d.openOutputDevice( errorString ) )
emit error( errorString );
else if ( !d.play( errorString ) )
emit error( errorString );
d.closeOutputDevice();
deviceMutex_.unlock();
}
void DecoderThread::cancel( bool waitUntilFinished )
{
isCancelled_.ref();
d.stop();
if ( waitUntilFinished )
this->wait();
}
} // namespace Ffmpeg
#endif // MAKE_FFMPEG_PLAYER