mirror of
https://github.com/xiaoyifang/goldendict-ng.git
synced 2024-12-05 00:24:06 +00:00
0683afa2b0
[autofix.ci] apply automated fixes
407 lines
11 KiB
C++
407 lines
11 KiB
C++
#ifdef MAKE_FFMPEG_PLAYER
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#include "audiooutput.hh"
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#include "ffmpegaudio.hh"
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#include <errno.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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#include <libavutil/avutil.h>
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#include "libswresample/swresample.h"
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}
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#include <QString>
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#include <QDataStream>
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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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using std::vector;
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namespace Ffmpeg {
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QMutex DecoderThread::deviceMutex_;
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static inline QString avErrorString( int errnum )
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{
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char buf[ 64 ];
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av_strerror( errnum, buf, 64 );
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return QString::fromLatin1( buf );
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}
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AudioService & AudioService::instance()
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{
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static AudioService a;
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return a;
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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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}
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void AudioService::playMemory( const char * ptr, int size )
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{
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emit cancelPlaying( false );
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QByteArray audioData( ptr, size );
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thread = std::make_shared< DecoderThread >( audioData, this );
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connect( this, &AudioService::cancelPlaying, thread.get(), [ this ]( bool waitFinished ) {
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thread->cancel( waitFinished );
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} );
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thread->start();
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}
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void AudioService::stop()
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{
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emit cancelPlaying( false );
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}
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DecoderContext::DecoderContext( QByteArray const & audioData, QAtomicInt & isCancelled ):
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isCancelled_( isCancelled ),
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audioData_( audioData ),
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audioDataStream_( audioData_ ),
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formatContext_( nullptr ),
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codec_( nullptr ),
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codecContext_( nullptr ),
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avioContext_( nullptr ),
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audioStream_( nullptr ),
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audioOutput( new AudioOutput ),
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avformatOpened_( false ),
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swr_( nullptr )
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{
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}
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DecoderContext::~DecoderContext()
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{
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closeOutputDevice();
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closeCodec();
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}
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static int readAudioData( void * opaque, unsigned char * buffer, int bufferSize )
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{
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QDataStream * pStream = (QDataStream *)opaque;
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// This function is passed as the read_packet callback into avio_alloc_context().
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// The documentation for this callback parameter states:
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// For stream protocols, must never return 0 but rather a proper AVERROR code.
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if ( pStream->atEnd() )
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return AVERROR_EOF;
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const int bytesRead = pStream->readRawData( (char *)buffer, bufferSize );
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// QDataStream::readRawData() returns 0 at EOF => return AVERROR_EOF in this case.
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// An error is unlikely here, so just print a warning and return AVERROR_EOF too.
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if ( bytesRead < 0 )
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gdWarning( "readAudioData: error while reading raw data." );
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return bytesRead > 0 ? bytesRead : AVERROR_EOF;
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}
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bool DecoderContext::openCodec( QString & errorString )
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{
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formatContext_ = avformat_alloc_context();
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if ( !formatContext_ ) {
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errorString = "avformat_alloc_context() failed.";
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return false;
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}
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unsigned char * avioBuffer = (unsigned char *)av_malloc( kBufferSize + AV_INPUT_BUFFER_PADDING_SIZE );
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if ( !avioBuffer ) {
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errorString = "av_malloc() failed.";
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return false;
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}
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// Don't free buffer allocated here (if succeeded), it will be cleaned up automatically.
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avioContext_ = avio_alloc_context( avioBuffer, kBufferSize, 0, &audioDataStream_, readAudioData, nullptr, nullptr );
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if ( !avioContext_ ) {
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av_free( avioBuffer );
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errorString = "avio_alloc_context() failed.";
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return false;
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}
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avioContext_->seekable = 0;
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avioContext_->write_flag = 0;
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// If pb not set, avformat_open_input() simply crash.
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formatContext_->pb = avioContext_;
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formatContext_->flags |= AVFMT_FLAG_CUSTOM_IO;
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int ret = 0;
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avformatOpened_ = true;
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ret = avformat_open_input( &formatContext_, nullptr, nullptr, nullptr );
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if ( ret < 0 ) {
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errorString = QString( "avformat_open_input() failed: %1." ).arg( avErrorString( ret ) );
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return false;
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}
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ret = avformat_find_stream_info( formatContext_, nullptr );
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if ( ret < 0 ) {
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errorString = QString( "avformat_find_stream_info() failed: %1." ).arg( avErrorString( ret ) );
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return false;
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}
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// Find audio stream, use the first audio stream if available
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for ( unsigned i = 0; i < formatContext_->nb_streams; i++ ) {
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if ( formatContext_->streams[ i ]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO ) {
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audioStream_ = formatContext_->streams[ i ];
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break;
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}
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}
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if ( !audioStream_ ) {
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errorString = QString( "Could not find audio stream." );
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return false;
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}
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codec_ = avcodec_find_decoder( audioStream_->codecpar->codec_id );
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if ( !codec_ ) {
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errorString = QString( "Codec [id: %1] not found." ).arg( audioStream_->codecpar->codec_id );
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return false;
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}
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codecContext_ = avcodec_alloc_context3( codec_ );
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if ( !codecContext_ ) {
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errorString = QString( "avcodec_alloc_context3() failed." );
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return false;
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}
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avcodec_parameters_to_context( codecContext_, audioStream_->codecpar );
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ret = avcodec_open2( codecContext_, codec_, nullptr );
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if ( ret < 0 ) {
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errorString = QString( "avcodec_open2() failed: %1." ).arg( avErrorString( ret ) );
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return false;
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}
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gdDebug( "Codec open: %s: channels: %d, rate: %d, format: %s\n",
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codec_->long_name,
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codecContext_->channels,
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codecContext_->sample_rate,
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av_get_sample_fmt_name( codecContext_->sample_fmt ) );
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auto layout = codecContext_->channel_layout;
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if ( !layout ) {
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layout = av_get_default_channel_layout( codecContext_->channels );
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codecContext_->channel_layout = layout;
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}
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swr_ = swr_alloc_set_opts( nullptr,
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layout,
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AV_SAMPLE_FMT_S16,
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44100,
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layout,
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codecContext_->sample_fmt,
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codecContext_->sample_rate,
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0,
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nullptr );
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if ( !swr_ || swr_init( swr_ ) < 0 ) {
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av_log( nullptr, AV_LOG_ERROR, "Cannot create sample rate converter \n" );
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swr_free( &swr_ );
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return false;
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}
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return true;
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}
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void DecoderContext::closeCodec()
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{
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if ( swr_ ) {
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swr_free( &swr_ );
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}
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if ( !formatContext_ ) {
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if ( avioContext_ ) {
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av_free( avioContext_->buffer );
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avioContext_ = nullptr;
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}
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return;
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}
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// avformat_open_input() is not called, just free the buffer associated with
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// the AVIOContext, and the AVFormatContext
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if ( !avformatOpened_ ) {
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if ( formatContext_ ) {
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avformat_free_context( formatContext_ );
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formatContext_ = nullptr;
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}
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if ( avioContext_ ) {
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av_free( avioContext_->buffer );
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avioContext_ = nullptr;
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}
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return;
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}
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avformatOpened_ = false;
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// Closing a codec context without prior avcodec_open2() will result in
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// a crash in ffmpeg
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if ( audioStream_ && codecContext_ && codec_ ) {
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audioStream_->discard = AVDISCARD_ALL;
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avcodec_close( codecContext_ );
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avcodec_free_context( &codecContext_ );
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}
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avformat_close_input( &formatContext_ );
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av_free( avioContext_->buffer );
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}
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bool DecoderContext::openOutputDevice( QString & errorString )
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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 += QString( "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( 44100, codecContext_->channels );
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return true;
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}
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void DecoderContext::closeOutputDevice() {}
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bool DecoderContext::play( QString & errorString )
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{
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AVFrame * frame = av_frame_alloc();
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if ( !frame ) {
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errorString = QString( "avcodec_alloc_frame() failed." );
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return false;
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}
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AVPacket * packet = av_packet_alloc();
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while ( !Utils::AtomicInt::loadAcquire( isCancelled_ ) && av_read_frame( formatContext_, packet ) >= 0 ) {
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if ( packet->stream_index == audioStream_->index ) {
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int ret = avcodec_send_packet( codecContext_, packet );
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/* read all the output frames (in general there may be any number of them) */
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while ( ret >= 0 ) {
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ret = avcodec_receive_frame( codecContext_, frame );
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if ( Utils::AtomicInt::loadAcquire( isCancelled_ ) || ret < 0 )
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break;
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playFrame( frame );
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}
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}
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av_packet_unref( packet );
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}
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/* flush the decoder */
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packet->data = nullptr;
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packet->size = 0;
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int ret = avcodec_send_packet( codecContext_, packet );
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while ( ret >= 0 ) {
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ret = avcodec_receive_frame( codecContext_, frame );
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if ( Utils::AtomicInt::loadAcquire( isCancelled_ ) || ret < 0 )
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break;
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playFrame( frame );
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}
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av_frame_free( &frame );
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av_packet_free( &packet );
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return true;
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}
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void DecoderContext::stop()
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{
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if ( audioOutput ) {
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audioOutput->stop();
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audioOutput->deleteLater();
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audioOutput = nullptr;
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}
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}
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bool DecoderContext::normalizeAudio( AVFrame * frame, vector< uint8_t > & samples )
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{
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auto dst_freq = 44100;
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auto dst_channels = codecContext_->channels;
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int out_count = (int64_t)frame->nb_samples * dst_freq / frame->sample_rate + 256;
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int out_size = av_samples_get_buffer_size( nullptr, dst_channels, out_count, AV_SAMPLE_FMT_S16, 1 );
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samples.resize( out_size );
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uint8_t * data[ 2 ] = { nullptr };
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data[ 0 ] = &samples.front();
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auto out_nb_samples = swr_convert( swr_, data, out_count, (const uint8_t **)frame->extended_data, frame->nb_samples );
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if ( out_nb_samples < 0 ) {
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av_log( nullptr, AV_LOG_ERROR, "converte fail \n" );
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return false;
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}
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else {
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// qDebug( "out_count:%d, out_nb_samples:%d, frame->nb_samples:%d \n", out_count, out_nb_samples, frame->nb_samples );
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}
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int actual_size = av_samples_get_buffer_size( nullptr, dst_channels, out_nb_samples, AV_SAMPLE_FMT_S16, 1 );
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samples.resize( actual_size );
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return true;
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}
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void DecoderContext::playFrame( AVFrame * frame )
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{
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if ( !frame )
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return;
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vector< uint8_t > samples;
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if ( normalizeAudio( frame, samples ) ) {
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audioOutput->play( &samples.front(), samples.size() );
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}
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}
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DecoderThread::DecoderThread( QByteArray const & audioData, QObject * parent ):
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QThread( parent ),
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isCancelled_( 0 ),
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audioData_( audioData ),
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d( audioData_, isCancelled_ )
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{
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}
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DecoderThread::~DecoderThread()
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{
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isCancelled_.ref();
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d.stop();
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}
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void DecoderThread::run()
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{
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QString errorString;
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if ( !d.openCodec( errorString ) ) {
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emit error( errorString );
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return;
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}
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while ( !deviceMutex_.tryLock( 100 ) ) {
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if ( Utils::AtomicInt::loadAcquire( isCancelled_ ) )
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return;
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}
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if ( !d.openOutputDevice( errorString ) )
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emit error( errorString );
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else if ( !d.play( errorString ) )
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emit error( errorString );
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d.closeOutputDevice();
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deviceMutex_.unlock();
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}
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void DecoderThread::cancel( bool waitUntilFinished )
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{
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isCancelled_.ref();
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d.stop();
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if ( waitUntilFinished )
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this->wait();
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}
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} // namespace Ffmpeg
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#endif // MAKE_FFMPEG_PLAYER
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