Renaming of files in preparation for code style tools.

Signed-off-by: Michele Calgaro <[email protected]>

1 files changed, 0 insertions, 390 deletions

diff --git a/flow/audioioaix.cc b/flow/audioioaix.cc
deleted file mode 100644
index b5c288b..0000000
--- a/flow/audioioaix.cc
+++ /dev/null
@@ -1,390 +0,0 @@
-/*
-
-    Copyright (C) 2001 Carsten Griwodz
-                       [email protected]
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-  
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-   
-    You should have received a copy of the GNU Library General Public License
-    along with this library; see the file COPYING.LIB.  If not, write to
-    the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
-    Boston, MA 02110-1301, USA.
-
-*/
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-#ifdef _AIX
-
-/*
- * The audio header files exist even if there is not soundcard the
- * the AIX machine. You won't be able to compile this code on AIX3
- * which had ACPA support, so /dev/acpa is not checked here.
- * I have no idea whether the Ultimedia Audio Adapter is actually
- * working or what it is right now.
- * For PCI machines including PowerSeries 850, baud or paud should
- * work. The DSP (MWave?) of the 850 laptops may need microcode
- * download. This is not implemented.
- */
-
-#include <assert.h>
-#include <stdlib.h>
-#include <stdio.h>
-#include <string.h>
-#include <errno.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <sys/time.h>
-#include <sys/ioctl.h>
-#include <sys/stat.h>
-#include <sys/machine.h>
-#undef BIG_ENDIAN
-#include <sys/audio.h>
-
-#ifndef AUDIO_BIG_ENDIAN
-#define AUDIO_BIG_ENDIAN BIG_ENDIAN
-#endif
-
-#include "debug.h"
-#include "audioio.h"
-
-namespace Arts {
-
-class AudioIOAIX : public AudioIO {
-    int openDevice();
-
-protected:
-    int audio_fd;
-
-public:
-    AudioIOAIX();
-
-    void setParam(AudioParam param, int& value);
-    int getParam(AudioParam param);
-
-    bool open();
-    void close();
-    int read(void *buffer, int size);
-    int write(void *buffer, int size);
-};
-
-REGISTER_AUDIO_IO(AudioIOAIX,"paud","Personal Audio Device");
-};
-
-using namespace std;
-using namespace Arts;
-
-int AudioIOAIX::openDevice()
-{
-	char devname[14];
-	int  fd;
-	for ( int dev=0; dev<4; dev++ )
-	{
-		for ( int chan=1; chan<8; chan++ )
-		{
-			sprintf(devname,"/dev/paud%d/%d",dev,chan);
-			fd = ::open (devname, O_WRONLY, 0);
-			if ( fd >= 0 )
-			{
-				paramStr(deviceName) = devname;
-				return fd;
-			}
-			sprintf(devname,"/dev/baud%d/%d",dev,chan);
-			fd = ::open (devname, O_WRONLY, 0);
-			if ( fd >= 0 )
-			{
-				paramStr(deviceName) = devname;
-				return fd;
-			}
-		}
-	}
-	return -1;
-}
-
-AudioIOAIX::AudioIOAIX()
-{
-	int fd = openDevice();
-	if( fd >= 0 )
-	{
-		audio_status audioStatus;
-		memset( &audioStatus, 0, sizeof(audio_status) );
-		ioctl(fd, AUDIO_STATUS, &audioStatus);
-
-		audio_buffer audioBuffer;
-		memset( &audioBuffer, 0, sizeof(audio_buffer) );
-		ioctl(fd, AUDIO_BUFFER, &audioBuffer);
-
-		::close( fd );
-
-		/*
-		 * default parameters
-		 */
-		param(samplingRate)  = audioStatus.srate;
-		param(fragmentSize)  = audioStatus.bsize;
-		param(fragmentCount) = audioBuffer.write_buf_cap / audioStatus.bsize;
-		param(channels)      = audioStatus.channels;
-		param(direction)     = 2;
-
-		param(format)        = ( audioStatus.bits_per_sample==8 ) ? 8
-			: ( ( audioStatus.flags & AUDIO_BIG_ENDIAN ) ? 17 : 16 );
-	}
-}
-
-bool AudioIOAIX::open()
-{
-	string& _error = paramStr(lastError);
-	string& _deviceName = paramStr(deviceName);
-	int& _channels = param(channels);
-	int& _fragmentSize = param(fragmentSize);
-	int& _fragmentCount = param(fragmentCount);
-	int& _samplingRate = param(samplingRate);
-	int& _format = param(format);
-
-	int mode;
-
-	switch( param(direction) )
-	{
-		case 1 : mode = O_RDONLY | O_NDELAY; break;
-		case 2 : mode = O_WRONLY | O_NDELAY; break;
-		case 3 :
-				 _error = "open device twice to RDWR";
-				 return false;
-		default :
-				 _error = "invalid direction";
-				 return false;
-	}
-
-	audio_fd = ::open(_deviceName.c_str(), mode, 0);
-
-	if(audio_fd == -1)
-	{
-		_error = "device ";
-		_error += _deviceName.c_str();
-		_error += " can't be opened (";
-		_error += strerror(errno);
-		_error += ")";
-		return false;
-	}
-
-	if( (_channels!=1) && (_channels!=2) )
-	{
-		_error = "internal error; set channels to 1 (mono) or 2 (stereo)";
-
-		close();
-		return false;
-	}
-
-	// int requeststereo = stereo;
-
-	// int speed = _samplingRate;
-
-	audio_init audioInit;
-	memset( &audioInit, 0, sizeof(audio_init) );
-	audioInit.srate = _samplingRate;
-	audioInit.bits_per_sample = ((_format==8)?8:16);
-	audioInit.bsize = _fragmentSize;
-	audioInit.mode = PCM;
-	audioInit.channels = _channels;
-	audioInit.flags = 0;
-	audioInit.flags |= (_format==17) ? AUDIO_BIG_ENDIAN : 0;
-	audioInit.flags |= (_format==8)  ? 0 : SIGNED;
-	audioInit.operation = (param(direction)==1) ? RECORD : PLAY;
-
-	if ( ioctl(audio_fd, AUDIO_INIT, &audioInit) < 0 )
-	{
-		_error = "AUDIO_INIT failed - ";
-		_error += strerror(errno);
-		switch ( audioInit.rc )
-		{
-			case 1 :
-				_error += "Couldn't set audio format: DSP can't do play requests";
-				break;
-			case 2 :
-				_error += "Couldn't set audio format: DSP can't do record requests";
-				break;
-			case 4 :
-				_error += "Couldn't set audio format: request was invalid";
-				break;
-			case 5 :
-				_error += "Couldn't set audio format: conflict with open's flags";
-				break;
-			case 6 :
-				_error += "Couldn't set audio format: out of DSP MIPS or memory";
-				break;
-			default :
-				_error += "Couldn't set audio format: not documented in sys/audio.h";
-				break;
-		}
-
-		close();
-		return false;
-	}
-
-	if (audioInit.channels != _channels)
-	{
-		_error = "audio device doesn't support number of requested channels";
-		close();
-		return false;
-	}
-
-	switch( _format )
-	{
-		case 8 :
-			if (audioInit.flags&AUDIO_BIG_ENDIAN==1)
-			{
-				_error = "setting little endian format failed";
-				close();
-				return false;
-			}
-			if (audioInit.flags&SIGNED==1)
-			{
-				_error = "setting unsigned format failed";
-				close();
-				return false;
-			}
-			break;
-		case 16 :
-			if (audioInit.flags&AUDIO_BIG_ENDIAN==1)
-			{
-				_error = "setting little endian format failed";
-				close();
-				return false;
-			}
-			if (audioInit.flags&SIGNED==0)
-			{
-				_error = "setting signed format failed";
-				close();
-				return false;
-			}
-			break;
-		case 17 :
-			if (audioInit.flags&AUDIO_BIG_ENDIAN==0)
-			{
-				_error = "setting big endian format failed";
-				close();
-				return false;
-			}
-			if (audioInit.flags&SIGNED==0)
-			{
-				_error = "setting signed format failed";
-				close();
-				return false;
-			}
-			break;
-		default :
-			break;
-	}
-
-	/*
-	 * Some soundcards seem to be able to only supply "nearly" the requested
-	 * sampling rate, especially PAS 16 cards seem to quite radical supplying
-	 * something different than the requested sampling rate ;)
-	 *
-	 * So we have a quite large tolerance here (when requesting 44100 Hz, it
-	 * will accept anything between 38690 Hz and 49510 Hz). Most parts of the
-	 * aRts code will do resampling where appropriate, so it shouldn't affect
-	 * sound quality.
-	 */
-	int tolerance = _samplingRate/10+1000;
-
-	if (abs(audioInit.srate - _samplingRate) > tolerance)
-	{  
-		_error = "can't set requested samplingrate";
-
-		char details[80];
-		sprintf(details," (requested rate %d, got rate %ld)",
-				_samplingRate, audioInit.srate);
-		_error += details;
-
-		close();
-		return false;
-	} 
-	_samplingRate = audioInit.srate;
-
-	_fragmentSize = audioInit.bsize;
-	_fragmentCount = audioInit.bsize / audioInit.bits_per_sample;
-
-	audio_buffer buffer_info;
-	ioctl(audio_fd, AUDIO_BUFFER, &buffer_info);
-	_fragmentCount = buffer_info.write_buf_cap / audioInit.bsize;
-
-
-	artsdebug("buffering: %d fragments with %d bytes "
-			"(audio latency is %1.1f ms)", _fragmentCount, _fragmentSize,
-			(float)(_fragmentSize*_fragmentCount) /
-			(float)(2.0 * _samplingRate * _channels)*1000.0);
-
-	return true;
-}
-
-void AudioIOAIX::close()
-{
-	::close(audio_fd);
-}
-
-void AudioIOAIX::setParam(AudioParam p, int& value)
-{
-	param(p) = value;
-}
-
-int AudioIOAIX::getParam(AudioParam p)
-{
-	audio_buffer info;
-	switch(p)
-	{
-		case canRead:
-			ioctl(audio_fd, AUDIO_BUFFER, &info);
-			return (info.read_buf_cap - info.read_buf_size);
-			break;
-
-		case canWrite:
-			ioctl(audio_fd, AUDIO_BUFFER, &info);
-			return (info.write_buf_cap - info.write_buf_size);
-			break;
-
-		case selectReadFD:
-			return (param(direction) & directionRead)?audio_fd:-1;
-			break;
-
-		case selectWriteFD:
-			return (param(direction) & directionWrite)?audio_fd:-1;
-			break;
-
-		case autoDetect:
-			/* You may prefer OSS if it works, e.g. on 43P 240
-			 * or you may prefer UMS, if anyone bothers to write
-			 * a module for it.
-			 */
-			return 2;
-			break;
-
-		default:
-			return param(p);
-			break;
-	}
-}
-
-int AudioIOAIX::read(void *buffer, int size)
-{
-	arts_assert(audio_fd != 0);
-	return ::read(audio_fd,buffer,size);
-}
-
-int AudioIOAIX::write(void *buffer, int size)
-{
-	arts_assert(audio_fd != 0);
-	return ::write(audio_fd,buffer,size);
-}
-
-#endif
-