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arts/flow/audioiooss.cpp

486 lines
11 KiB

/*
Copyright (C) 2000 Stefan Westerfeld
stefan@space.twc.de
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
#if defined(HAVE_SYS_SOUNDCARD_H)
#include <sys/soundcard.h>
#define COMPILE_AUDIOIO_OSS 1
#elif defined(HAVE_SOUNDCARD_H)
#include <soundcard.h>
#define COMPILE_AUDIOIO_OSS 1
#endif
/**
* only compile 'oss' AudioIO class if sys/soundcard.h or soundcard.h is present
*/
#ifdef COMPILE_AUDIOIO_OSS
#include <sys/types.h>
#include <sys/ioctl.h>
#include <sys/time.h>
#include <sys/stat.h>
#ifdef HAVE_SYS_SELECT_H
#include <sys/select.h> // Needed on some systems.
#endif
#include <assert.h>
#include <errno.h>
#include <fcntl.h>
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <iostream>
#include <algorithm>
#include <cstring>
#include "debug.h"
#include "audioio.h"
namespace Arts {
class AudioIOOSS : public AudioIO {
protected:
int audio_fd;
int requestedFragmentSize;
int requestedFragmentCount;
std::string findDefaultDevice();
int ossBits(int format);
public:
AudioIOOSS();
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(AudioIOOSS,"oss","Open Sound System");
}
using namespace std;
using namespace Arts;
/*
* Tries to figure out which is the OSS device we should write to
*/
string AudioIOOSS::findDefaultDevice()
{
static const char *device[] = {
"/dev/dsp", /* The usual device */
// how does access(2) deal with a directory?
// I don't know, but, since /dev/sound/dsp is a linux bogosity
#if defined(__linux__)
"/dev/sound/dsp", /* Linux with devfs-only installation */
#else
"/dev/sound", /* NetBSD */
#endif
"/dev/audio", /* OpenBSD */
0
};
for(int i = 0; device[i]; i++)
if(access(device[i],F_OK) == 0)
return device[i];
// Should this really return a valid device if we have access to none?
return device[0];
}
int AudioIOOSS::ossBits(int format)
{
arts_return_val_if_fail (format == AFMT_U8
|| format == AFMT_S16_LE
|| format == AFMT_S16_BE, 16);
return (format == AFMT_U8)?8:16;
}
AudioIOOSS::AudioIOOSS()
{
/*
* default parameters
*/
param(samplingRate) = 44100;
paramStr(deviceName) = findDefaultDevice();
requestedFragmentSize = param(fragmentSize) = 1024;
requestedFragmentCount = param(fragmentCount) = 7;
param(channels) = 2;
param(direction) = 2;
}
bool AudioIOOSS::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;
if(param(direction) == 3)
mode = O_RDWR|O_NDELAY;
else if(param(direction) == 2)
mode = O_WRONLY|O_NDELAY;
else
{
_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;
}
/*
* check device capabilities
*/
int device_caps;
if(ioctl(audio_fd,SNDCTL_DSP_GETCAPS,&device_caps) == -1)
device_caps=0;
string caps = "";
if(device_caps & DSP_CAP_DUPLEX) caps += "duplex ";
if(device_caps & DSP_CAP_REALTIME) caps += "realtime ";
if(device_caps & DSP_CAP_BATCH) caps += "batch ";
if(device_caps & DSP_CAP_COPROC) caps += "coproc ";
if(device_caps & DSP_CAP_TRIGGER) caps += "trigger ";
if(device_caps & DSP_CAP_MMAP) caps += "mmap ";
artsdebug("device capabilities: revision%d %s",
device_caps & DSP_CAP_REVISION, caps.c_str());
int requestedFormat = (_format == 8)?AFMT_U8:AFMT_S16_LE;
int gotFormat = requestedFormat;
if (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &gotFormat)==-1)
{
_error = "SNDCTL_DSP_SETFMT failed - ";
_error += strerror(errno);
close();
return false;
}
if (_format && (ossBits(gotFormat) != ossBits(requestedFormat)))
{
char details[80];
sprintf(details," (_format = %d, asked driver to give %d, got %d)",
_format, requestedFormat, gotFormat);
_error = "Can't set playback format";
_error += details;
close();
return false;
}
if(gotFormat == AFMT_U8)
_format = 8;
else if(gotFormat == AFMT_S16_LE)
_format = 16;
else if(gotFormat == AFMT_S16_BE)
_format = 17;
else
{
char details[80];
sprintf(details," (_format = %d, asked driver to give %d, got %d)",
_format, requestedFormat, gotFormat);
_error = "unknown format given by driver";
_error += details;
close();
return false;
}
int stereo=-1; /* 0=mono, 1=stereo */
if(_channels == 1)
{
stereo = 0;
}
if(_channels == 2)
{
stereo = 1;
}
if(stereo == -1)
{
_error = "internal error; set channels to 1 (mono) or 2 (stereo)";
close();
return false;
}
int requeststereo = stereo;
if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo)==-1)
{
_error = "SNDCTL_DSP_STEREO failed - ";
_error += strerror(errno);
close();
return false;
}
if (requeststereo != stereo)
{
_error = "audio device doesn't support number of requested channels";
close();
return false;
}
int speed = _samplingRate;
if (ioctl(audio_fd, SNDCTL_DSP_SPEED, &speed)==-1)
{
_error = "SNDCTL_DSP_SPEED failed - ";
_error += strerror(errno);
close();
return false;
}
/*
* 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(speed-_samplingRate) > tolerance)
{
_error = "can't set requested samplingrate";
char details[80];
sprintf(details," (requested rate %d, got rate %d)",
_samplingRate, speed);
_error += details;
close();
return false;
}
_samplingRate = speed;
/*
* set the fragment settings to what the user requested
*/
_fragmentSize = requestedFragmentSize;
_fragmentCount = requestedFragmentCount;
/*
* lower 16 bits are the fragment size (as 2^S)
* higher 16 bits are the number of fragments
*/
int frag_arg = 0;
int size = _fragmentSize;
while(size > 1) { size /= 2; frag_arg++; }
frag_arg += (_fragmentCount << 16);
if(ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_arg) == -1)
{
char buffer[1024];
_error = "can't set requested fragments settings";
sprintf(buffer,"size%d:count%d\n",_fragmentSize,_fragmentCount);
close();
return false;
}
/*
* now see what we really got as cards aren't required to supply what
* we asked for
*/
audio_buf_info info;
if(ioctl(audio_fd,SNDCTL_DSP_GETOSPACE, &info) == -1)
{
_error = "can't retrieve fragment settings";
close();
return false;
}
// update fragment settings with what we got
_fragmentSize = info.fragsize;
_fragmentCount = info.fragstotal;
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);
/*
* Workaround for broken kernel drivers: usually filling up the audio
* buffer is _only_ required if _fullDuplex is true. However, there
* are kernel drivers around (especially everything related to ES1370/1371)
* which will not trigger select()ing the file descriptor unless we have
* written something first.
*/
char *zbuffer = (char *)calloc(sizeof(char), _fragmentSize);
if(_format == 8)
for(int zpos = 0; zpos < _fragmentSize; zpos++)
zbuffer[zpos] |= 0x80;
for(int fill = 0; fill < _fragmentCount; fill++)
{
int len = write(zbuffer,_fragmentSize);
if(len != _fragmentSize)
{
arts_debug("AudioIOOSS: failed prefilling audio buffer (might cause synchronization problems in conjunction with full duplex)");
fill = _fragmentCount+1;
}
}
free(zbuffer);
/*
* Triggering - the original aRts code did this for full duplex:
*
* - stop audio i/o using SETTRIGGER(~(PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT))
* - fill buffer (see zbuffer code two lines above)
* - start audio i/o using SETTRIGGER(PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT)
*
* this should guarantee synchronous start of input/output. Today, it
* seems there are too many broken drivers around for this.
*/
if(device_caps & DSP_CAP_TRIGGER)
{
int enable_bits = 0;
if(param(direction) & 1) enable_bits |= PCM_ENABLE_INPUT;
if(param(direction) & 2) enable_bits |= PCM_ENABLE_OUTPUT;
if(ioctl(audio_fd,SNDCTL_DSP_SETTRIGGER, &enable_bits) == -1)
{
_error = "can't start sound i/o";
close();
return false;
}
}
return true;
}
void AudioIOOSS::close()
{
::close(audio_fd);
}
void AudioIOOSS::setParam(AudioParam p, int& value)
{
switch(p)
{
case fragmentSize:
param(p) = requestedFragmentSize = value;
break;
case fragmentCount:
param(p) = requestedFragmentCount = value;
break;
default:
param(p) = value;
break;
}
}
int AudioIOOSS::getParam(AudioParam p)
{
audio_buf_info info;
switch(p)
{
case canRead:
ioctl(audio_fd, SNDCTL_DSP_GETISPACE, &info);
return info.bytes;
break;
case canWrite:
ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &info);
return info.bytes;
break;
case selectReadFD:
return (param(direction) & directionRead)?audio_fd:-1;
break;
case selectWriteFD:
return (param(direction) & directionWrite)?audio_fd:-1;
break;
case autoDetect:
/* OSS works reasonable almost everywhere where it compiles */
return 10;
break;
default:
return param(p);
break;
}
}
int AudioIOOSS::read(void *buffer, int size)
{
arts_assert(audio_fd != 0);
int result;
do {
result = ::read(audio_fd,buffer,size);
} while(result == -1 && errno == EINTR);
return result;
}
int AudioIOOSS::write(void *buffer, int size)
{
arts_assert(audio_fd != 0);
int result;
do {
result = ::write(audio_fd,buffer,size);
} while(result == -1 && errno == EINTR);
return result;
}
#endif