path: root/flow/audioioossthreaded.cpp

    /*

    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., 51 Franklin Street, Fifth Floor,
    Boston, MA 02110-1301, 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