Renaming of files in preparation for code style tools.

Signed-off-by: Michele Calgaro <michele.calgaro@yahoo.it>
(cherry picked from commit 00d4f92b717fbcbed6f9eee361975d6ee5380d59)

1 files changed, 390 insertions, 0 deletions

diff --git a/flow/audioioaix.cpp b/flow/audioioaix.cpp
new file mode 100644
index 0000000..b5c288b
--- /dev/null
+++ b/flow/audioioaix.cpp
@@ -0,0 +1,390 @@
+/*
+
+    Copyright (C) 2001 Carsten Griwodz
+                       griff@ifi.uio.no
+
+    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
+