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+ /*
+
+ Copyright (C) 2000 Stefan Westerfeld
+ stefan@space.twc.de
+
+ Multithreading support by
+ Matthias Welwarsky <matze@stud.fbi.fh-darmstadt.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., 59 Temple Place - Suite 330,
+ Boston, MA 02111-1307, USA.
+
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#ifdef HAVE_SYS_SOUNDCARD_H
+#include <sys/soundcard.h>
+#define COMPILE_AUDIOIO_OSS 1
+#endif
+
+#ifdef 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
+ * also, only compile if libpthread was found
+ */
+#if defined(COMPILE_AUDIOIO_OSS) && defined(HAVE_LIBPTHREAD)
+#include <sys/types.h>
+#include <sys/ioctl.h>
+#include <sys/time.h>
+#include <sys/stat.h>
+
+#include <assert.h>
+#include <errno.h>
+#include <fcntl.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <iostream>
+#include <cstring>
+#include <algorithm>
+
+#include "debug.h"
+#include "audioio.h"
+#include "audiosubsys.h"
+#include "iomanager.h"
+#include "dispatcher.h"
+
+// thread safe data queue implementation
+#include "thread.h"
+#include "bufferqueue.h"
+
+namespace Arts {
+
+class AudioIOOSSThreaded : public AudioIO, public TimeNotify {
+private:
+ class ReaderThread : public Arts::Thread {
+ public:
+ void run();
+ bool isRunning() { return runThread; }
+ void stop() { runThread = false; }
+ void setParent(AudioIOOSSThreaded* instance) { parent = instance; }
+
+ private:
+ bool runThread;
+ AudioIOOSSThreaded* parent;
+ };
+
+ class WriterThread : public Arts::Thread {
+ public:
+ void run();
+ bool isRunning() { return runThread; }
+ void stop() { runThread = false; }
+ void setParent(AudioIOOSSThreaded* instance) { parent = instance; }
+
+ private:
+ bool runThread;
+ AudioIOOSSThreaded* parent;
+ };
+
+ friend class ReaderThread;
+ friend class WriterThread;
+
+ void startThread();
+ void stopThread();
+
+ BufferQueue readerQueue;
+ BufferQueue writerQueue;
+
+ ReaderThread readerThread;
+ WriterThread writerThread;
+
+protected:
+ int audio_fd;
+ int requestedFragmentSize;
+ int requestedFragmentCount;
+
+ std::string findDefaultDevice();
+ int ossBits(int format);
+
+public:
+ AudioIOOSSThreaded();
+
+ 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);
+
+ void notifyTime();
+};
+
+REGISTER_AUDIO_IO(AudioIOOSSThreaded,"toss","Threaded Open Sound System");
+}
+
+using namespace std;
+using namespace Arts;
+
+/*
+ * Tries to figure out which is the OSS device we should write to
+ */
+string AudioIOOSSThreaded::findDefaultDevice()
+{
+ static const char *device[] = {
+ "/dev/dsp", /* Linux (and lots of others) */
+ "/dev/sound/dsp0", /* Linux with devfs-only installation */
+ "/dev/audio", /* OpenBSD */
+ 0
+ };
+
+ for(int i = 0; device[i]; i++)
+ if(access(device[i],F_OK) == 0)
+ return device[i];
+
+ return device[0];
+}
+
+int AudioIOOSSThreaded::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;
+}
+
+AudioIOOSSThreaded::AudioIOOSSThreaded()
+{
+ /*
+ * default parameters
+ */
+ param(samplingRate) = 44100;
+ paramStr(deviceName) = findDefaultDevice();
+ requestedFragmentSize = param(fragmentSize) = 1024;
+ requestedFragmentCount = param(fragmentCount) = 7;
+ param(channels) = 2;
+ param(direction) = 2;
+}
+
+bool AudioIOOSSThreaded::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);
+
+ if (!SystemThreads::supported()) {
+ _error = "System does not support multithreading";
+ return false;
+ }
+
+ int mode;
+
+ if(param(direction) == 3)
+ mode = O_RDWR;
+ else if(param(direction) == 2)
+ mode = O_WRONLY;
+ 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
+ */
+ unsigned int frag_arg = 0;
+
+ unsigned int size = _fragmentSize;
+ while(size > 1) { size /= 2; frag_arg++; }
+ frag_arg += (_fragmentCount << 16);
+
+ //////////////////////////////////////////////////////////////////////
+ // MW
+ // stopping here because the fragment settings cannot be done
+ // is not sensible. Especially not as we check the settings in the
+ // very next step. We should handle this gracefully.
+ //
+ ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_arg);
+
+ /*
+ * 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;
+
+ fprintf(stderr, "buffering: fragsize:%d fragstotal:%d\n",
+ _fragmentSize, _fragmentCount);
+
+ 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);
+
+ // set the chunk size of the buffer queues.
+ readerQueue.setChunkSize(_fragmentSize);
+ writerQueue.setChunkSize(_fragmentSize);
+
+ /*
+ * 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(audio_fd,zbuffer,_fragmentSize);
+ if(len != _fragmentSize)
+ {
+ arts_debug("AudioIOOSSThreaded: 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 of sound i/o operation";
+
+ close();
+ return false;
+ }
+ }
+
+ startThread();
+
+ Dispatcher::the()->ioManager()->addTimer(10, this);
+
+ return true;
+}
+
+void AudioIOOSSThreaded::close()
+{
+ fprintf(stderr, "AudioIO::close(): entering\n");
+ Dispatcher::the()->ioManager()->removeTimer(this);
+
+ ::close(audio_fd);
+ stopThread();
+ fprintf(stderr, "AudioIO::close(): leaving\n");
+}
+
+void AudioIOOSSThreaded::notifyTime()
+{
+ int& _direction = param(direction);
+
+ for(int i=0; i < 3; i++) {
+ int todo = 0;
+
+ if((_direction & directionRead) && getParam(canRead) > 0)
+ todo |= AudioSubSystem::ioRead;
+
+ if((_direction & directionWrite) && getParam(canWrite) > 0)
+ todo |= AudioSubSystem::ioWrite;
+
+ //fprintf(stderr, "AudioIO: can write %d, fragment size %d, todo %d\n",
+ // getParam(canWrite), _fragmentSize, todo);
+
+ if(!todo) // break endless loop
+ break;
+
+ AudioSubSystem::the()->handleIO(todo);
+ }
+}
+
+void AudioIOOSSThreaded::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 AudioIOOSSThreaded::getParam(AudioParam p)
+{
+ switch(p) {
+ case canRead:
+ return readerQueue.bufferedChunks() * readerQueue.chunkSize();
+ break;
+
+ case canWrite:
+ return writerQueue.freeChunks() * writerQueue.chunkSize();
+ break;
+
+ case autoDetect:
+ /* It compiles, but Posix Threads don't work everywhere */
+ return 4;
+ break;
+
+ default:
+ return param(p);
+ break;
+ }
+}
+
+int AudioIOOSSThreaded::read(void *buffer, int size)
+{
+ int copied = 0;
+ int copySize;
+ ByteBuffer* tmpBuf = 0;
+ while (size > 0) {
+ if (!tmpBuf)
+ tmpBuf = readerQueue.waitProduced();
+
+ copySize = (size > tmpBuf->size())? tmpBuf->size():size;
+ memcpy(((char*)buffer)+copied, tmpBuf->get(), copySize);
+
+ if (tmpBuf->push(copySize) == 0) {
+ tmpBuf = 0;
+ readerQueue.consumed();
+ }
+
+ copied += copySize;
+ size -= copySize;
+ }
+ return copied;
+}
+
+int AudioIOOSSThreaded::write(void *buffer, int size)
+{
+ int copied = 0;
+ while (size > 0) {
+ int copySize = (size > writerQueue.chunkSize())?writerQueue.chunkSize():size;
+ if (!writerQueue.freeChunks())
+ fprintf(stderr, "AudioIO::write will block!\n");
+ writerQueue.write(buffer, copySize);
+ copied += copySize;
+ size -= copySize;
+ }
+ return copied;
+}
+
+/*
+ * posix thread to feed the audio device
+ */
+
+void AudioIOOSSThreaded::WriterThread::run()
+{
+ fprintf(stderr, "AudioIOOSSThreaded::writerThread() thread started\n");
+
+ setPriority(45);
+
+ runThread = true;
+
+ ssize_t size;
+ ByteBuffer* tmpBuf = 0;
+ while (runThread) {
+
+ if (!tmpBuf) {
+ tmpBuf = parent->writerQueue.waitProduced();
+
+ if (!tmpBuf->size()) {
+ tmpBuf = NULL;
+ parent->writerQueue.consumed();
+ continue;
+ }
+ }
+
+ size = ::write(parent->audio_fd, tmpBuf->get(), tmpBuf->size());
+
+ if (size >= 0) {
+ if (tmpBuf->push(size) == 0) {
+ tmpBuf = NULL;
+ parent->writerQueue.consumed();
+ }
+ } else if (errno != EINTR) {
+ // this is a fatal error. we cannot write to the fd any more.
+ runThread = false;
+ fprintf(stderr, "AudioIOOSSTHreaded::writerThread() fatal error writing to audio_fd\n");
+ }
+ }
+ fprintf(stderr, "AudioIOOSSThreaded::writerThread() thread stopped\n");
+}
+
+/*
+ * posix thread to read the audio device
+ */
+void AudioIOOSSThreaded::ReaderThread::run()
+{
+ fprintf(stderr, "AudioIOOSSThreaded::readerThread() thread started\n");
+
+ runThread = true;
+
+ ByteBuffer* tmpBuf;
+ ssize_t size;
+ while (runThread) {
+ tmpBuf = parent->readerQueue.waitConsumed();
+
+ size = ::read(parent->audio_fd, tmpBuf->reset(), tmpBuf->maxSize());
+
+ if (size >= 0) {
+ tmpBuf->set(size);
+ parent->readerQueue.produced();
+ } else if (errno != EINTR) {
+ runThread = false;
+ fprintf(stderr, "AudioIOOSSTHreaded::readerThread() fatal error reading from audio_fd\n");
+ }
+ }
+ fprintf(stderr, "AudioIOOSSThreaded::readerThread() thread stopped\n");
+}
+
+void AudioIOOSSThreaded::startThread()
+{
+ fprintf(stderr, "AudioIOOSSThreaded::startThread(): entering\n");
+
+ if (param(direction) & directionWrite) {
+ writerThread.setParent(this);
+ writerThread.start();
+ }
+ if (param(direction) & directionRead) {
+ readerThread.setParent(this);
+ readerThread.start();
+ }
+
+ fprintf(stderr, "AudioIOOSSThreaded::startThread(): leaving\n");
+}
+
+void AudioIOOSSThreaded::stopThread()
+{
+ fprintf(stderr, "AudioIOOSSThreaded::stopThread() entering\n");
+
+ if (param(direction) & directionWrite) {
+ writerThread.stop();
+ // make sure there's something to write, avoids race condition
+ if (writerQueue.isEmpty())
+ writerQueue.write(NULL, 0);
+ fprintf(stderr, "waiting for writerThread to finish\n");
+ writerThread.waitDone();
+ writerQueue.clear();
+ }
+
+ if (param(direction) & directionRead) {
+ readerThread.stop();
+ fprintf(stderr, "waiting for readerThread to finish\n");
+ readerThread.waitDone();
+ readerQueue.clear();
+ }
+
+ fprintf(stderr, "AudioIOOSSThreaded::stopThread(): leaving\n");
+}
+
+#endif