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486 lines
11 KiB
486 lines
11 KiB
/*
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Copyright (C) 2000 Stefan Westerfeld
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stefan@space.twc.de
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This library is free software; you can redistribute it and/or
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modify it under the terms of the GNU Library General Public
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License as published by the Free Software Foundation; either
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version 2 of the License, or (at your option) any later version.
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This library is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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Library General Public License for more details.
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You should have received a copy of the GNU Library General Public License
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along with this library; see the file COPYING.LIB. If not, write to
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the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
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Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#if defined(HAVE_SYS_SOUNDCARD_H)
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#include <sys/soundcard.h>
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#define COMPILE_AUDIOIO_OSS 1
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#elif defined(HAVE_SOUNDCARD_H)
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#include <soundcard.h>
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#define COMPILE_AUDIOIO_OSS 1
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#endif
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/**
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* only compile 'oss' AudioIO class if sys/soundcard.h or soundcard.h is present
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*/
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#ifdef COMPILE_AUDIOIO_OSS
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#include <sys/types.h>
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#include <sys/ioctl.h>
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#include <sys/time.h>
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#include <sys/stat.h>
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#ifdef HAVE_SYS_SELECT_H
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#include <sys/select.h> // Needed on some systems.
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#endif
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#include <assert.h>
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#include <errno.h>
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#include <fcntl.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <iostream>
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#include <algorithm>
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#include <cstring>
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#include "debug.h"
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#include "audioio.h"
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namespace Arts {
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class AudioIOOSS : public AudioIO {
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protected:
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int audio_fd;
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int requestedFragmentSize;
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int requestedFragmentCount;
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std::string findDefaultDevice();
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int ossBits(int format);
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public:
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AudioIOOSS();
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void setParam(AudioParam param, int& value);
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int getParam(AudioParam param);
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bool open();
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void close();
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int read(void *buffer, int size);
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int write(void *buffer, int size);
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};
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REGISTER_AUDIO_IO(AudioIOOSS,"oss","Open Sound System");
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}
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using namespace std;
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using namespace Arts;
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/*
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* Tries to figure out which is the OSS device we should write to
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*/
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string AudioIOOSS::findDefaultDevice()
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{
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static const char *device[] = {
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"/dev/dsp", /* The usual device */
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// how does access(2) deal with a directory?
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// I don't know, but, since /dev/sound/dsp is a linux bogosity
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#if defined(__linux__)
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"/dev/sound/dsp", /* Linux with devfs-only installation */
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#else
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"/dev/sound", /* NetBSD */
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#endif
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"/dev/audio", /* OpenBSD */
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0
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};
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for(int i = 0; device[i]; i++)
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if(access(device[i],F_OK) == 0)
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return device[i];
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// Should this really return a valid device if we have access to none?
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return device[0];
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}
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int AudioIOOSS::ossBits(int format)
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{
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arts_return_val_if_fail (format == AFMT_U8
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|| format == AFMT_S16_LE
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|| format == AFMT_S16_BE, 16);
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return (format == AFMT_U8)?8:16;
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}
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AudioIOOSS::AudioIOOSS()
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{
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/*
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* default parameters
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*/
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param(samplingRate) = 44100;
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paramStr(deviceName) = findDefaultDevice();
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requestedFragmentSize = param(fragmentSize) = 1024;
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requestedFragmentCount = param(fragmentCount) = 7;
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param(channels) = 2;
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param(direction) = 2;
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}
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bool AudioIOOSS::open()
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{
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string& _error = paramStr(lastError);
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string& _deviceName = paramStr(deviceName);
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int& _channels = param(channels);
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int& _fragmentSize = param(fragmentSize);
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int& _fragmentCount = param(fragmentCount);
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int& _samplingRate = param(samplingRate);
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int& _format = param(format);
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int mode;
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if(param(direction) == 3)
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mode = O_RDWR|O_NDELAY;
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else if(param(direction) == 2)
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mode = O_WRONLY|O_NDELAY;
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else
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{
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_error = "invalid direction";
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return false;
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}
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audio_fd = ::open(_deviceName.c_str(), mode, 0);
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if(audio_fd == -1)
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{
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_error = "device ";
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_error += _deviceName.c_str();
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_error += " can't be opened (";
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_error += strerror(errno);
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_error += ")";
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return false;
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}
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/*
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* check device capabilities
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*/
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int device_caps;
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if(ioctl(audio_fd,SNDCTL_DSP_GETCAPS,&device_caps) == -1)
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device_caps=0;
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string caps = "";
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if(device_caps & DSP_CAP_DUPLEX) caps += "duplex ";
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if(device_caps & DSP_CAP_REALTIME) caps += "realtime ";
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if(device_caps & DSP_CAP_BATCH) caps += "batch ";
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if(device_caps & DSP_CAP_COPROC) caps += "coproc ";
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if(device_caps & DSP_CAP_TRIGGER) caps += "trigger ";
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if(device_caps & DSP_CAP_MMAP) caps += "mmap ";
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artsdebug("device capabilities: revision%d %s",
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device_caps & DSP_CAP_REVISION, caps.c_str());
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int requestedFormat = (_format == 8)?AFMT_U8:AFMT_S16_LE;
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int gotFormat = requestedFormat;
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if (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &gotFormat)==-1)
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{
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_error = "SNDCTL_DSP_SETFMT failed - ";
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_error += strerror(errno);
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close();
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return false;
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}
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if (_format && (ossBits(gotFormat) != ossBits(requestedFormat)))
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{
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char details[80];
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sprintf(details," (_format = %d, asked driver to give %d, got %d)",
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_format, requestedFormat, gotFormat);
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_error = "Can't set playback format";
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_error += details;
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close();
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return false;
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}
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if(gotFormat == AFMT_U8)
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_format = 8;
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else if(gotFormat == AFMT_S16_LE)
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_format = 16;
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else if(gotFormat == AFMT_S16_BE)
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_format = 17;
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else
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{
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char details[80];
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sprintf(details," (_format = %d, asked driver to give %d, got %d)",
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_format, requestedFormat, gotFormat);
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_error = "unknown format given by driver";
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_error += details;
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close();
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return false;
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}
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int stereo=-1; /* 0=mono, 1=stereo */
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if(_channels == 1)
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{
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stereo = 0;
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}
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if(_channels == 2)
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{
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stereo = 1;
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}
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if(stereo == -1)
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{
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_error = "internal error; set channels to 1 (mono) or 2 (stereo)";
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close();
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return false;
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}
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int requeststereo = stereo;
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if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo)==-1)
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{
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_error = "SNDCTL_DSP_STEREO failed - ";
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_error += strerror(errno);
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close();
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return false;
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}
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if (requeststereo != stereo)
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{
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_error = "audio device doesn't support number of requested channels";
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close();
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return false;
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}
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int speed = _samplingRate;
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if (ioctl(audio_fd, SNDCTL_DSP_SPEED, &speed)==-1)
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{
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_error = "SNDCTL_DSP_SPEED failed - ";
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_error += strerror(errno);
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close();
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return false;
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}
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/*
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* Some soundcards seem to be able to only supply "nearly" the requested
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* sampling rate, especially PAS 16 cards seem to quite radical supplying
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* something different than the requested sampling rate ;)
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*
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* So we have a quite large tolerance here (when requesting 44100 Hz, it
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* will accept anything between 38690 Hz and 49510 Hz). Most parts of the
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* aRts code will do resampling where appropriate, so it shouldn't affect
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* sound quality.
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*/
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int tolerance = _samplingRate/10+1000;
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if (abs(speed-_samplingRate) > tolerance)
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{
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_error = "can't set requested samplingrate";
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char details[80];
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sprintf(details," (requested rate %d, got rate %d)",
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_samplingRate, speed);
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_error += details;
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close();
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return false;
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}
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_samplingRate = speed;
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/*
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* set the fragment settings to what the user requested
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*/
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_fragmentSize = requestedFragmentSize;
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_fragmentCount = requestedFragmentCount;
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/*
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* lower 16 bits are the fragment size (as 2^S)
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* higher 16 bits are the number of fragments
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*/
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int frag_arg = 0;
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int size = _fragmentSize;
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while(size > 1) { size /= 2; frag_arg++; }
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frag_arg += (_fragmentCount << 16);
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if(ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_arg) == -1)
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{
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char buffer[1024];
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_error = "can't set requested fragments settings";
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sprintf(buffer,"size%d:count%d\n",_fragmentSize,_fragmentCount);
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close();
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return false;
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}
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/*
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* now see what we really got as cards aren't required to supply what
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* we asked for
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*/
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audio_buf_info info;
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if(ioctl(audio_fd,SNDCTL_DSP_GETOSPACE, &info) == -1)
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{
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_error = "can't retrieve fragment settings";
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close();
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return false;
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}
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// update fragment settings with what we got
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_fragmentSize = info.fragsize;
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_fragmentCount = info.fragstotal;
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artsdebug("buffering: %d fragments with %d bytes "
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"(audio latency is %1.1f ms)", _fragmentCount, _fragmentSize,
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(float)(_fragmentSize*_fragmentCount) /
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(float)(2.0 * _samplingRate * _channels)*1000.0);
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/*
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* Workaround for broken kernel drivers: usually filling up the audio
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* buffer is _only_ required if _fullDuplex is true. However, there
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* are kernel drivers around (especially everything related to ES1370/1371)
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* which will not trigger select()ing the file descriptor unless we have
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* written something first.
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*/
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char *zbuffer = (char *)calloc(sizeof(char), _fragmentSize);
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if(_format == 8)
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for(int zpos = 0; zpos < _fragmentSize; zpos++)
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zbuffer[zpos] |= 0x80;
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for(int fill = 0; fill < _fragmentCount; fill++)
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{
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int len = write(zbuffer,_fragmentSize);
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if(len != _fragmentSize)
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{
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arts_debug("AudioIOOSS: failed prefilling audio buffer (might cause synchronization problems in conjunction with full duplex)");
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fill = _fragmentCount+1;
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}
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}
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free(zbuffer);
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/*
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* Triggering - the original aRts code did this for full duplex:
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*
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* - stop audio i/o using SETTRIGGER(~(PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT))
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* - fill buffer (see zbuffer code two lines above)
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* - start audio i/o using SETTRIGGER(PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT)
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*
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* this should guarantee synchronous start of input/output. Today, it
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* seems there are too many broken drivers around for this.
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*/
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if(device_caps & DSP_CAP_TRIGGER)
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{
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int enable_bits = 0;
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if(param(direction) & 1) enable_bits |= PCM_ENABLE_INPUT;
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if(param(direction) & 2) enable_bits |= PCM_ENABLE_OUTPUT;
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if(ioctl(audio_fd,SNDCTL_DSP_SETTRIGGER, &enable_bits) == -1)
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{
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_error = "can't start sound i/o";
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close();
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return false;
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}
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}
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return true;
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}
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void AudioIOOSS::close()
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{
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::close(audio_fd);
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}
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void AudioIOOSS::setParam(AudioParam p, int& value)
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{
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switch(p)
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{
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case fragmentSize:
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param(p) = requestedFragmentSize = value;
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break;
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case fragmentCount:
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param(p) = requestedFragmentCount = value;
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break;
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default:
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param(p) = value;
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break;
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}
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}
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int AudioIOOSS::getParam(AudioParam p)
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{
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audio_buf_info info;
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switch(p)
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{
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case canRead:
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ioctl(audio_fd, SNDCTL_DSP_GETISPACE, &info);
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return info.bytes;
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break;
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case canWrite:
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ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &info);
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return info.bytes;
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break;
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case selectReadFD:
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return (param(direction) & directionRead)?audio_fd:-1;
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break;
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case selectWriteFD:
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return (param(direction) & directionWrite)?audio_fd:-1;
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break;
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case autoDetect:
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/* OSS works reasonable almost everywhere where it compiles */
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return 10;
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break;
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default:
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return param(p);
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break;
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}
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}
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int AudioIOOSS::read(void *buffer, int size)
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{
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arts_assert(audio_fd != 0);
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int result;
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do {
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result = ::read(audio_fd,buffer,size);
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} while(result == -1 && errno == EINTR);
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return result;
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}
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int AudioIOOSS::write(void *buffer, int size)
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{
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arts_assert(audio_fd != 0);
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int result;
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do {
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result = ::write(audio_fd,buffer,size);
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} while(result == -1 && errno == EINTR);
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return result;
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}
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#endif
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