fixed OSS and SDL

git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@428 0778d3d1-df1d-0410-868b-ea421aaaa00d

ChangeLog

@@ -1,3 +1,16 @@
+2006-11-15	Javier Serrano Polo <jasp00/at/terra/dot/es>
+
+	* include/audio_alsa.h:
+	* src/audio/audio_alsa.cpp:
+	* src/audio/audio_device.cpp:
+	* src/audio/audio_file_wave.cpp:
+	convert-endian-based conversion instead of little-endian-based,
+	fixes OSS and SDL
+
+	* include/audio_sdl.h:
+	* src/audio/audio_sdl.cpp:
+	fill the whole audio buffer
+
 2006-11-11	Javier Serrano Polo <jasp00/at/terra/dot/es>
 
 	* src/core/piano_roll.cpp:

configure.in

@@ -2,8 +2,8 @@
 # Process this file with autoconf to produce a configure script.
 
 AC_PREREQ(2.50)
-AC_INIT(lmms, 0.2.1-svn20061111, lmms-devel/at/lists/dot/sf/dot/net)
-AM_INIT_AUTOMAKE(lmms, 0.2.1-svn20061111)
+AC_INIT(lmms, 0.2.1-svn20061115, lmms-devel/at/lists/dot/sf/dot/net)
+AM_INIT_AUTOMAKE(lmms, 0.2.1-svn20061115)
 
 AM_CONFIG_HEADER(config.h)
 

include/audio_alsa.h

@@ -99,7 +99,7 @@ private:
 	snd_pcm_hw_params_t * m_hwParams;
 	snd_pcm_sw_params_t * m_swParams;
 
-	bool m_littleEndian;
+	bool m_convertEndian;
 	volatile bool m_quit;
 
 } ;

include/audio_sdl.h

@@ -80,10 +80,14 @@ private:
 	virtual void stopProcessing( void );
 
 	static void sdlAudioCallback( void * _udata, Uint8 * _buf, int _len );
+	void sdlAudioCallback( Uint8 * _buf, int _len );
 
 	SDL_AudioSpec m_audioHandle;
 
 	surroundSampleFrame * m_outBuf;
+	Uint8 * m_convertedBuf;
+	int m_convertedBuf_pos;
+	int m_convertedBuf_size;
 
 	bool m_convertEndian;
 

src/audio/audio_alsa.cpp

@@ -65,7 +65,7 @@ audioALSA::audioALSA( const sample_rate_t _sample_rate, bool & _success_ful,
 	m_handle( NULL ),
 	m_hwParams( NULL ),
 	m_swParams( NULL ),
-	m_littleEndian( isLittleEndian() ),
+	m_convertEndian( FALSE ),
 	m_quit( FALSE )
 {
 	_success_ful = FALSE;
@@ -228,7 +228,7 @@ void audioALSA::run( void )
 		const f_cnt_t frames = getNextBuffer( temp );
 
 		convertToS16( temp, frames, getMixer()->masterGain(), outbuf,
-					m_littleEndian );
+					m_convertEndian );
 
 		f_cnt_t frame = 0;
 		int_sample_t * ptr = outbuf;
@@ -298,7 +298,11 @@ int audioALSA::setHWParams( const sample_rate_t _sample_rate,
 					"playback: %s\n", snd_strerror( err ) );
 			return( err );
 		}
-		m_littleEndian = FALSE;
+		m_convertEndian = isLittleEndian();
+	}
+	else
+	{
+		m_convertEndian = !isLittleEndian();
 	}
 
 	// set the count of channels

src/audio/audio_device.cpp

@@ -297,23 +297,9 @@ Uint32 FASTCALL audioDevice::convertToS16( const surroundSampleFrame * _ab,
 						const fpab_t _frames,
 						const float _master_gain,
 						int_sample_t * _output_buffer,
-						const bool _little_endian )
+						const bool _convert_endian )
 {
-	if( _little_endian )
-	{
-		for( fpab_t frame = 0; frame < _frames; ++frame )
-		{
-			for( ch_cnt_t chnl = 0; chnl < channels(); ++chnl )
-			{
-				( _output_buffer + frame * channels() )[chnl] =
-						static_cast<int_sample_t>(
-						mixer::clip( _ab[frame][chnl] *
-						_master_gain ) *
-						OUTPUT_SAMPLE_MULTIPLIER );
-			}
-		}
-	}
-	else
+	if( _convert_endian )
 	{
 		Uint16 temp;
 		for( fpab_t frame = 0; frame < _frames; ++frame )
@@ -331,6 +317,20 @@ Uint32 FASTCALL audioDevice::convertToS16( const surroundSampleFrame * _ab,
 			}
 		}
 	}
+	else
+	{
+		for( fpab_t frame = 0; frame < _frames; ++frame )
+		{
+			for( ch_cnt_t chnl = 0; chnl < channels(); ++chnl )
+			{
+				( _output_buffer + frame * channels() )[chnl] =
+						static_cast<int_sample_t>(
+						mixer::clip( _ab[frame][chnl] *
+						_master_gain ) *
+						OUTPUT_SAMPLE_MULTIPLIER );
+			}
+		}
+	}
 
 	return( _frames * channels() * BYTES_PER_INT_SAMPLE );
 }

src/audio/audio_file_wave.cpp

@@ -99,7 +99,7 @@ void FASTCALL audioFileWave::writeBuffer( const surroundSampleFrame * _ab,
 	int_sample_t * outbuf = bufferAllocator::alloc<int_sample_t>(
 							_frames * channels() );
 	Uint32 bytes = convertToS16( _ab, _frames, _master_gain, outbuf,
-							TRUE );
+							!isLittleEndian() );
 	writeData( outbuf, bytes );
 
 	bufferAllocator::free( outbuf );

src/audio/audio_sdl.cpp

@@ -47,6 +47,7 @@
 #include "debug.h"
 #include "config_mgr.h"
 #include "gui_templates.h"
+#include "templates.h"
 
 
 
@@ -55,10 +56,16 @@ audioSDL::audioSDL( const sample_rate_t _sample_rate, bool & _success_ful,
 	audioDevice( _sample_rate, DEFAULT_CHANNELS, _mixer ),
 	m_outBuf( bufferAllocator::alloc<surroundSampleFrame>(
 				getMixer()->framesPerAudioBuffer() ) ),
+	m_convertedBuf_pos( 0 ),
 	m_convertEndian( FALSE )
 {
 	_success_ful = FALSE;
 
+	m_convertedBuf_size = getMixer()->framesPerAudioBuffer() * channels()
+						* sizeof( int_sample_t );
+	m_convertedBuf = (Uint8 *)bufferAllocator::allocBytes(
+							m_convertedBuf_size );
+
 /*	// if device is set, we set AUDIODEV-environment-variable, so that
 	// SDL can evaluate and use it
 	QString dev = configManager::inst()->value( "audiosdl", "device" );
@@ -110,6 +117,7 @@ audioSDL::~audioSDL()
 	stopProcessing();
 	SDL_CloseAudio();
 	SDL_Quit();
+	bufferAllocator::free( m_convertedBuf );
 	bufferAllocator::free( m_outBuf );
 }
 
@@ -145,12 +153,33 @@ void audioSDL::sdlAudioCallback( void * _udata, Uint8 * _buf, int _len )
 	assert( _this != NULL );
 #endif
 
-	const fpab_t frames = _this->getNextBuffer( _this->m_outBuf );
+	_this->sdlAudioCallback( _buf, _len );
+}
 
-	_this->convertToS16( _this->m_outBuf, frames,
-						_this->getMixer()->masterGain(),
-						(int_sample_t *)( _buf ),
-						_this->m_convertEndian );
+
+
+
+void audioSDL::sdlAudioCallback( Uint8 * _buf, int _len )
+{
+	while( _len )
+	{
+		if( m_convertedBuf_pos == 0 )
+		{
+			const fpab_t frames = getNextBuffer( m_outBuf );
+
+			convertToS16( m_outBuf, frames,
+						getMixer()->masterGain(),
+						(int_sample_t *)m_convertedBuf,
+						m_convertEndian );
+		}
+		int min_len = tMin( _len, m_convertedBuf_size
+							- m_convertedBuf_pos );
+		memcpy( _buf, m_convertedBuf + m_convertedBuf_pos, min_len );
+		_buf += min_len;
+		_len -= min_len;
+		m_convertedBuf_pos += min_len;
+		m_convertedBuf_pos %= m_convertedBuf_size;
+	}
 }