/************************************************************************/
/*! \class RtAudio
- \brief Realtime audio i/o C++ class.
+ \brief Realtime audio i/o C++ classes.
RtAudio provides a common API (Application Programming Interface)
- for realtime audio input/output across Linux (native ALSA and
- OSS), SGI, Macintosh OS X (CoreAudio), and Windows (DirectSound
- and ASIO) operating systems.
+ for realtime audio input/output across Linux (native ALSA, Jack,
+ and OSS), SGI, Macintosh OS X (CoreAudio), and Windows
+ (DirectSound and ASIO) operating systems.
- RtAudio WWW site: http://www-ccrma.stanford.edu/~gary/rtaudio/
+ RtAudio WWW site: http://music.mcgill.ca/~gary/rtaudio/
RtAudio: a realtime audio i/o C++ class
- Copyright (c) 2001-2002 Gary P. Scavone
+ Copyright (c) 2001-2004 Gary P. Scavone
Permission is hereby granted, free of charge, to any person
obtaining a copy of this software and associated documentation files
*/
/************************************************************************/
-// RtAudio: Version 2.1.1, 24 October 2002
+// RtAudio: Version 3.0, 11 March 2004
#include "RtAudio.h"
-#include <vector>
-#include <stdio.h>
-#include <iostream.h>
+#include <iostream>
// Static variable definitions.
-const unsigned int RtAudio :: SAMPLE_RATES[] = {
+const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
+const unsigned int RtApi::SAMPLE_RATES[] = {
4000, 5512, 8000, 9600, 11025, 16000, 22050,
32000, 44100, 48000, 88200, 96000, 176400, 192000
};
-const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT8 = 1;
-const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT16 = 2;
-const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT24 = 4;
-const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT32 = 8;
-const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_FLOAT32 = 16;
-const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_FLOAT64 = 32;
#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
#define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
- #define MUTEX_LOCK(A) EnterCriticalSection(A)
+ #define MUTEX_DESTROY(A) DeleteCriticalSection(A);
+ #define MUTEX_LOCK(A) EnterCriticalSection(A)
#define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
#else // pthread API
#define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
+ #define MUTEX_DESTROY(A) pthread_mutex_destroy(A);
#define MUTEX_LOCK(A) pthread_mutex_lock(A)
#define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)
#endif
//
// *************************************************** //
-RtAudio :: RtAudio()
+RtAudio :: RtAudio( RtAudioApi api )
{
- initialize();
-
- if (nDevices <= 0) {
- sprintf(message, "RtAudio: no audio devices found!");
- error(RtError::NO_DEVICES_FOUND);
- }
+ initialize( api );
}
-RtAudio :: RtAudio(int *streamId,
- int outputDevice, int outputChannels,
- int inputDevice, int inputChannels,
- RTAUDIO_FORMAT format, int sampleRate,
- int *bufferSize, int numberOfBuffers)
+RtAudio :: RtAudio( int outputDevice, int outputChannels,
+ int inputDevice, int inputChannels,
+ RtAudioFormat format, int sampleRate,
+ int *bufferSize, int numberOfBuffers, RtAudioApi api )
{
- initialize();
-
- if (nDevices <= 0) {
- sprintf(message, "RtAudio: no audio devices found!");
- error(RtError::NO_DEVICES_FOUND);
- }
+ initialize( api );
try {
- *streamId = openStream(outputDevice, outputChannels, inputDevice, inputChannels,
- format, sampleRate, bufferSize, numberOfBuffers);
+ rtapi_->openStream( outputDevice, outputChannels,
+ inputDevice, inputChannels,
+ format, sampleRate,
+ bufferSize, numberOfBuffers );
}
catch (RtError &exception) {
- // deallocate the RTAUDIO_DEVICE structures
- if (devices) free(devices);
+ // Deallocate the RtApi instance.
+ delete rtapi_;
throw exception;
}
}
RtAudio :: ~RtAudio()
{
- // close any existing streams
- while ( streams.size() )
- closeStream( streams.begin()->first );
+ delete rtapi_;
+}
+
+void RtAudio :: openStream( int outputDevice, int outputChannels,
+ int inputDevice, int inputChannels,
+ RtAudioFormat format, int sampleRate,
+ int *bufferSize, int numberOfBuffers )
+{
+ rtapi_->openStream( outputDevice, outputChannels, inputDevice,
+ inputChannels, format, sampleRate,
+ bufferSize, numberOfBuffers );
+}
+
+void RtAudio::initialize( RtAudioApi api )
+{
+ rtapi_ = 0;
+
+ // First look for a compiled match to a specified API value. If one
+ // of these constructors throws an error, it will be passed up the
+ // inheritance chain.
+#if defined(__LINUX_JACK__)
+ if ( api == LINUX_JACK )
+ rtapi_ = new RtApiJack();
+#endif
+#if defined(__LINUX_ALSA__)
+ if ( api == LINUX_ALSA )
+ rtapi_ = new RtApiAlsa();
+#endif
+#if defined(__LINUX_OSS__)
+ if ( api == LINUX_OSS )
+ rtapi_ = new RtApiOss();
+#endif
+#if defined(__WINDOWS_ASIO__)
+ if ( api == WINDOWS_ASIO )
+ rtapi_ = new RtApiAsio();
+#endif
+#if defined(__WINDOWS_DS__)
+ if ( api == WINDOWS_DS )
+ rtapi_ = new RtApiDs();
+#endif
+#if defined(__IRIX_AL__)
+ if ( api == IRIX_AL )
+ rtapi_ = new RtApiAl();
+#endif
+#if defined(__MACOSX_CORE__)
+ if ( api == MACOSX_CORE )
+ rtapi_ = new RtApiCore();
+#endif
+
+ if ( rtapi_ ) return;
+ if ( api > 0 ) {
+ // No compiled support for specified API value.
+ throw RtError( "RtAudio: no compiled support for specified API argument!", RtError::INVALID_PARAMETER );
+ }
+
+ // No specified API ... search for "best" option.
+ try {
+#if defined(__LINUX_JACK__)
+ rtapi_ = new RtApiJack();
+#elif defined(__WINDOWS_ASIO__)
+ rtapi_ = new RtApiAsio();
+#elif defined(__IRIX_AL__)
+ rtapi_ = new RtApiAl();
+#elif defined(__MACOSX_CORE__)
+ rtapi_ = new RtApiCore();
+#else
+ ;
+#endif
+ }
+ catch (RtError &) {
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf(stderr, "\nRtAudio: no devices found for first api option (JACK, ASIO, Al, or CoreAudio).\n\n");
+#endif
+ rtapi_ = 0;
+ }
+
+ if ( rtapi_ ) return;
+
+// Try second API support
+ if ( rtapi_ == 0 ) {
+ try {
+#if defined(__LINUX_ALSA__)
+ rtapi_ = new RtApiAlsa();
+#elif defined(__WINDOWS_DS__)
+ rtapi_ = new RtApiDs();
+#else
+ ;
+#endif
+ }
+ catch (RtError &) {
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf(stderr, "\nRtAudio: no devices found for second api option (Alsa or DirectSound).\n\n");
+#endif
+ rtapi_ = 0;
+ }
+ }
+
+ if ( rtapi_ ) return;
+
+ // Try third API support
+ if ( rtapi_ == 0 ) {
+#if defined(__LINUX_OSS__)
+ try {
+ rtapi_ = new RtApiOss();
+ }
+ catch (RtError &error) {
+ rtapi_ = 0;
+ }
+#else
+ ;
+#endif
+ }
+
+ if ( rtapi_ == 0 ) {
+ // No devices found.
+ throw RtError( "RtAudio: no devices found for compiled audio APIs!", RtError::NO_DEVICES_FOUND );
+ }
+}
+
+RtApi :: RtApi()
+{
+ stream_.mode = UNINITIALIZED;
+ stream_.apiHandle = 0;
+ MUTEX_INITIALIZE(&stream_.mutex);
+}
- // deallocate the RTAUDIO_DEVICE structures
- if (devices) free(devices);
+RtApi :: ~RtApi()
+{
+ MUTEX_DESTROY(&stream_.mutex);
}
-int RtAudio :: openStream(int outputDevice, int outputChannels,
- int inputDevice, int inputChannels,
- RTAUDIO_FORMAT format, int sampleRate,
- int *bufferSize, int numberOfBuffers)
+void RtApi :: openStream( int outputDevice, int outputChannels,
+ int inputDevice, int inputChannels,
+ RtAudioFormat format, int sampleRate,
+ int *bufferSize, int numberOfBuffers )
{
- static int streamKey = 0; // Unique stream identifier ... OK for multiple instances.
+ if ( stream_.mode != UNINITIALIZED ) {
+ sprintf(message_, "RtApi: only one open stream allowed per class instance.");
+ error(RtError::INVALID_STREAM);
+ }
if (outputChannels < 1 && inputChannels < 1) {
- sprintf(message,"RtAudio: one or both 'channel' parameters must be greater than zero.");
+ sprintf(message_,"RtApi: one or both 'channel' parameters must be greater than zero.");
error(RtError::INVALID_PARAMETER);
}
if ( formatBytes(format) == 0 ) {
- sprintf(message,"RtAudio: 'format' parameter value is undefined.");
+ sprintf(message_,"RtApi: 'format' parameter value is undefined.");
error(RtError::INVALID_PARAMETER);
}
if ( outputChannels > 0 ) {
- if (outputDevice > nDevices || outputDevice < 0) {
- sprintf(message,"RtAudio: 'outputDevice' parameter value (%d) is invalid.", outputDevice);
+ if (outputDevice > nDevices_ || outputDevice < 0) {
+ sprintf(message_,"RtApi: 'outputDevice' parameter value (%d) is invalid.", outputDevice);
error(RtError::INVALID_PARAMETER);
}
}
if ( inputChannels > 0 ) {
- if (inputDevice > nDevices || inputDevice < 0) {
- sprintf(message,"RtAudio: 'inputDevice' parameter value (%d) is invalid.", inputDevice);
+ if (inputDevice > nDevices_ || inputDevice < 0) {
+ sprintf(message_,"RtApi: 'inputDevice' parameter value (%d) is invalid.", inputDevice);
error(RtError::INVALID_PARAMETER);
}
}
- // Allocate a new stream structure.
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) calloc(1, sizeof(RTAUDIO_STREAM));
- if (stream == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
- stream->mode = UNINITIALIZED;
- MUTEX_INITIALIZE(&stream->mutex);
-
+ clearStreamInfo();
bool result = FAILURE;
int device, defaultDevice = 0;
- STREAM_MODE mode;
+ StreamMode mode;
int channels;
if ( outputChannels > 0 ) {
else
device = outputDevice - 1;
- for (int i=-1; i<nDevices; i++) {
- if (i >= 0 ) {
+ for ( int i=-1; i<nDevices_; i++ ) {
+ if ( i >= 0 ) {
if ( i == defaultDevice ) continue;
device = i;
}
- if (devices[device].probed == false) {
+ if (devices_[device].probed == false) {
// If the device wasn't successfully probed before, try it
- // again now.
- clearDeviceInfo(&devices[device]);
- probeDeviceInfo(&devices[device]);
+ // (again) now.
+ clearDeviceInfo(&devices_[device]);
+ probeDeviceInfo(&devices_[device]);
}
- if ( devices[device].probed )
- result = probeDeviceOpen(device, stream, mode, channels, sampleRate,
+ if ( devices_[device].probed )
+ result = probeDeviceOpen(device, mode, channels, sampleRate,
format, bufferSize, numberOfBuffers);
- if (result == SUCCESS) break;
+ if ( result == SUCCESS ) break;
if ( outputDevice > 0 ) break;
+ clearStreamInfo();
}
}
else
device = inputDevice - 1;
- for (int i=-1; i<nDevices; i++) {
+ for (int i=-1; i<nDevices_; i++) {
if (i >= 0 ) {
if ( i == defaultDevice ) continue;
device = i;
}
- if (devices[device].probed == false) {
+ if (devices_[device].probed == false) {
// If the device wasn't successfully probed before, try it
- // again now.
- clearDeviceInfo(&devices[device]);
- probeDeviceInfo(&devices[device]);
+ // (again) now.
+ clearDeviceInfo(&devices_[device]);
+ probeDeviceInfo(&devices_[device]);
}
- if ( devices[device].probed )
- result = probeDeviceOpen(device, stream, mode, channels, sampleRate,
+ if ( devices_[device].probed )
+ result = probeDeviceOpen(device, mode, channels, sampleRate,
format, bufferSize, numberOfBuffers);
if (result == SUCCESS) break;
if ( outputDevice > 0 ) break;
}
}
- streams[++streamKey] = (void *) stream;
if ( result == SUCCESS )
- return streamKey;
+ return;
// If we get here, all attempted probes failed. Close any opened
- // devices and delete the allocated stream.
- closeStream(streamKey);
+ // devices and clear the stream structure.
+ if ( stream_.mode != UNINITIALIZED ) closeStream();
+ clearStreamInfo();
if ( ( outputDevice == 0 && outputChannels > 0 )
|| ( inputDevice == 0 && inputChannels > 0 ) )
- sprintf(message,"RtAudio: no devices found for given parameters.");
+ sprintf(message_,"RtApi: no devices found for given stream parameters.");
else
- sprintf(message,"RtAudio: unable to open specified device(s) with given stream parameters.");
+ sprintf(message_,"RtApi: unable to open specified device(s) with given stream parameters.");
error(RtError::INVALID_PARAMETER);
- return -1;
+ return;
}
-int RtAudio :: getDeviceCount(void)
+int RtApi :: getDeviceCount(void)
{
- return nDevices;
+ return devices_.size();
}
-void RtAudio :: getDeviceInfo(int device, RTAUDIO_DEVICE *info)
+RtAudioDeviceInfo RtApi :: getDeviceInfo( int device )
{
- if (device > nDevices || device < 1) {
- sprintf(message, "RtAudio: invalid device specifier (%d)!", device);
+ if (device > (int) devices_.size() || device < 1) {
+ sprintf(message_, "RtApi: invalid device specifier (%d)!", device);
error(RtError::INVALID_DEVICE);
}
+ RtAudioDeviceInfo info;
int deviceIndex = device - 1;
// If the device wasn't successfully probed before, try it now (or again).
- if (devices[deviceIndex].probed == false) {
- clearDeviceInfo(&devices[deviceIndex]);
- probeDeviceInfo(&devices[deviceIndex]);
- }
-
- // Clear the info structure.
- memset(info, 0, sizeof(RTAUDIO_DEVICE));
-
- strncpy(info->name, devices[deviceIndex].name, 128);
- info->probed = devices[deviceIndex].probed;
- if ( info->probed == true ) {
- info->maxOutputChannels = devices[deviceIndex].maxOutputChannels;
- info->maxInputChannels = devices[deviceIndex].maxInputChannels;
- info->maxDuplexChannels = devices[deviceIndex].maxDuplexChannels;
- info->minOutputChannels = devices[deviceIndex].minOutputChannels;
- info->minInputChannels = devices[deviceIndex].minInputChannels;
- info->minDuplexChannels = devices[deviceIndex].minDuplexChannels;
- info->hasDuplexSupport = devices[deviceIndex].hasDuplexSupport;
- info->nSampleRates = devices[deviceIndex].nSampleRates;
- if (info->nSampleRates == -1) {
- info->sampleRates[0] = devices[deviceIndex].sampleRates[0];
- info->sampleRates[1] = devices[deviceIndex].sampleRates[1];
- }
- else {
- for (int i=0; i<info->nSampleRates; i++)
- info->sampleRates[i] = devices[deviceIndex].sampleRates[i];
- }
- info->nativeFormats = devices[deviceIndex].nativeFormats;
- if ( deviceIndex == getDefaultOutputDevice() ||
- deviceIndex == getDefaultInputDevice() )
- info->isDefault = true;
- }
-
- return;
+ if (devices_[deviceIndex].probed == false) {
+ clearDeviceInfo(&devices_[deviceIndex]);
+ probeDeviceInfo(&devices_[deviceIndex]);
+ }
+
+ info.name.append( devices_[deviceIndex].name );
+ info.probed = devices_[deviceIndex].probed;
+ if ( info.probed == true ) {
+ info.outputChannels = devices_[deviceIndex].maxOutputChannels;
+ info.inputChannels = devices_[deviceIndex].maxInputChannels;
+ info.duplexChannels = devices_[deviceIndex].maxDuplexChannels;
+ for (unsigned int i=0; i<devices_[deviceIndex].sampleRates.size(); i++)
+ info.sampleRates.push_back( devices_[deviceIndex].sampleRates[i] );
+ info.nativeFormats = devices_[deviceIndex].nativeFormats;
+ if ( (deviceIndex == getDefaultOutputDevice()) ||
+ (deviceIndex == getDefaultInputDevice()) )
+ info.isDefault = true;
+ }
+
+ return info;
}
-char * const RtAudio :: getStreamBuffer(int streamId)
+char * const RtApi :: getStreamBuffer(void)
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- return stream->userBuffer;
+ verifyStream();
+ return stream_.userBuffer;
}
-#if defined(__LINUX_ALSA__) || defined(__LINUX_OSS__) || defined(__IRIX_AL__)
-
-extern "C" void *callbackHandler(void * ptr);
-
-void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData)
+int RtApi :: getDefaultInputDevice(void)
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- CALLBACK_INFO *info = (CALLBACK_INFO *) &stream->callbackInfo;
- if ( info->usingCallback ) {
- sprintf(message, "RtAudio: A callback is already set for this stream!");
- error(RtError::WARNING);
- return;
- }
-
- info->callback = (void *) callback;
- info->userData = userData;
- info->usingCallback = true;
- info->object = (void *) this;
- info->streamId = streamId;
-
- int err = pthread_create(&info->thread, NULL, callbackHandler, &stream->callbackInfo);
-
- if (err) {
- info->usingCallback = false;
- sprintf(message, "RtAudio: error starting callback thread!");
- error(RtError::THREAD_ERROR);
- }
+ // Should be implemented in subclasses if appropriate.
+ return 0;
}
-void RtAudio :: cancelStreamCallback(int streamId)
+int RtApi :: getDefaultOutputDevice(void)
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- if (stream->callbackInfo.usingCallback) {
-
- if (stream->state == STREAM_RUNNING)
- stopStream( streamId );
+ // Should be implemented in subclasses if appropriate.
+ return 0;
+}
- MUTEX_LOCK(&stream->mutex);
+void RtApi :: closeStream(void)
+{
+ // MUST be implemented in subclasses!
+}
- stream->callbackInfo.usingCallback = false;
- pthread_cancel(stream->callbackInfo.thread);
- pthread_join(stream->callbackInfo.thread, NULL);
- stream->callbackInfo.thread = 0;
- stream->callbackInfo.callback = NULL;
- stream->callbackInfo.userData = NULL;
+void RtApi :: probeDeviceInfo( RtApiDevice *info )
+{
+ // MUST be implemented in subclasses!
+}
- MUTEX_UNLOCK(&stream->mutex);
- }
+bool RtApi :: probeDeviceOpen( int device, StreamMode mode, int channels,
+ int sampleRate, RtAudioFormat format,
+ int *bufferSize, int numberOfBuffers )
+{
+ // MUST be implemented in subclasses!
+ return FAILURE;
}
-#endif
// *************************************************** //
//
//
// *************************************************** //
-#if defined(__MACOSX_CORE__)
-
-// The OS X CoreAudio API is designed to use a separate callback
-// procedure for each of its audio devices. A single RtAudio duplex
-// stream using two different devices is supported here, though it
-// cannot be guaranteed to always behave correctly because we cannot
-// synchronize these two callbacks. This same functionality can be
-// achieved with better synchrony by opening two separate streams for
-// the devices and using RtAudio blocking calls (i.e. tickStream()).
-//
-// The possibility of having multiple RtAudio streams accessing the
-// same CoreAudio device is not currently supported. The problem
-// involves the inability to install our callbackHandler function for
-// the same device more than once. I experimented with a workaround
-// for this, but it requires an additional buffer for mixing output
-// data before filling the CoreAudio device buffer. In the end, I
-// decided it wasn't worth supporting.
-//
-// Property listeners are currently not used. The issue is what could
-// be done if a critical stream parameter (buffer size, sample rate,
-// device disconnect) notification arrived. The listeners entail
-// quite a bit of extra code and most likely, a user program wouldn't
-// be prepared for the result anyway. Some initial listener code is
-// commented out.
-
-void RtAudio :: initialize(void)
-{
- OSStatus err = noErr;
- UInt32 dataSize;
- AudioDeviceID *deviceList = NULL;
- nDevices = 0;
-
- // Find out how many audio devices there are, if any.
- err = AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &dataSize, NULL);
- if (err != noErr) {
- sprintf(message, "RtAudio: OSX error getting device info!");
- error(RtError::SYSTEM_ERROR);
- }
-
- nDevices = dataSize / sizeof(AudioDeviceID);
- if (nDevices == 0) return;
+#if defined(__LINUX_OSS__)
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
+#include <unistd.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/soundcard.h>
+#include <errno.h>
+#include <math.h>
- // Make space for the devices we are about to get.
- deviceList = (AudioDeviceID *) malloc( dataSize );
- if (deviceList == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
+#define DAC_NAME "/dev/dsp"
+#define MAX_DEVICES 16
+#define MAX_CHANNELS 16
- // Get the array of AudioDeviceIDs.
- err = AudioHardwareGetProperty(kAudioHardwarePropertyDevices, &dataSize, (void *) deviceList);
- if (err != noErr) {
- free(deviceList);
- sprintf(message, "RtAudio: OSX error getting device properties!");
- error(RtError::SYSTEM_ERROR);
- }
+extern "C" void *ossCallbackHandler(void * ptr);
- // Write device identifiers to device structures and then
- // probe the device capabilities.
- for (int i=0; i<nDevices; i++) {
- devices[i].id[0] = deviceList[i];
- //probeDeviceInfo(&devices[i]);
- }
+RtApiOss :: RtApiOss()
+{
+ this->initialize();
- free(deviceList);
+ if (nDevices_ <= 0) {
+ sprintf(message_, "RtApiOss: no Linux OSS audio devices found!");
+ error(RtError::NO_DEVICES_FOUND);
+ }
}
-int RtAudio :: getDefaultInputDevice(void)
+RtApiOss :: ~RtApiOss()
{
- AudioDeviceID id;
- UInt32 dataSize = sizeof( AudioDeviceID );
+ if ( stream_.mode != UNINITIALIZED )
+ closeStream();
+}
- OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultInputDevice,
- &dataSize, &id );
+void RtApiOss :: initialize(void)
+{
+ // Count cards and devices
+ nDevices_ = 0;
- if (result != noErr) {
- sprintf( message, "RtAudio: OSX error getting default input device." );
- error(RtError::WARNING);
- return 0;
+ // We check /dev/dsp before probing devices. /dev/dsp is supposed to
+ // be a link to the "default" audio device, of the form /dev/dsp0,
+ // /dev/dsp1, etc... However, I've seen many cases where /dev/dsp was a
+ // real device, so we need to check for that. Also, sometimes the
+ // link is to /dev/dspx and other times just dspx. I'm not sure how
+ // the latter works, but it does.
+ char device_name[16];
+ struct stat dspstat;
+ int dsplink = -1;
+ int i = 0;
+ if (lstat(DAC_NAME, &dspstat) == 0) {
+ if (S_ISLNK(dspstat.st_mode)) {
+ i = readlink(DAC_NAME, device_name, sizeof(device_name));
+ if (i > 0) {
+ device_name[i] = '\0';
+ if (i > 8) { // check for "/dev/dspx"
+ if (!strncmp(DAC_NAME, device_name, 8))
+ dsplink = atoi(&device_name[8]);
+ }
+ else if (i > 3) { // check for "dspx"
+ if (!strncmp("dsp", device_name, 3))
+ dsplink = atoi(&device_name[3]);
+ }
+ }
+ else {
+ sprintf(message_, "RtApiOss: cannot read value of symbolic link %s.", DAC_NAME);
+ error(RtError::SYSTEM_ERROR);
+ }
+ }
}
-
- for ( int i=0; i<nDevices; i++ ) {
- if ( id == devices[i].id[0] ) return i;
+ else {
+ sprintf(message_, "RtApiOss: cannot stat %s.", DAC_NAME);
+ error(RtError::SYSTEM_ERROR);
}
- return 0;
-}
-
-int RtAudio :: getDefaultOutputDevice(void)
-{
- AudioDeviceID id;
- UInt32 dataSize = sizeof( AudioDeviceID );
+ // The OSS API doesn't provide a routine for determining the number
+ // of devices. Thus, we'll just pursue a brute force method. The
+ // idea is to start with /dev/dsp(0) and continue with higher device
+ // numbers until we reach MAX_DSP_DEVICES. This should tell us how
+ // many devices we have ... it is not a fullproof scheme, but hopefully
+ // it will work most of the time.
+ int fd = 0;
+ RtApiDevice device;
+ for (i=-1; i<MAX_DEVICES; i++) {
- OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultOutputDevice,
- &dataSize, &id );
+ // Probe /dev/dsp first, since it is supposed to be the default device.
+ if (i == -1)
+ sprintf(device_name, "%s", DAC_NAME);
+ else if (i == dsplink)
+ continue; // We've aready probed this device via /dev/dsp link ... try next device.
+ else
+ sprintf(device_name, "%s%d", DAC_NAME, i);
- if (result != noErr) {
- sprintf( message, "RtAudio: OSX error getting default output device." );
- error(RtError::WARNING);
- return 0;
- }
+ // First try to open the device for playback, then record mode.
+ fd = open(device_name, O_WRONLY | O_NONBLOCK);
+ if (fd == -1) {
+ // Open device for playback failed ... either busy or doesn't exist.
+ if (errno != EBUSY && errno != EAGAIN) {
+ // Try to open for capture
+ fd = open(device_name, O_RDONLY | O_NONBLOCK);
+ if (fd == -1) {
+ // Open device for record failed.
+ if (errno != EBUSY && errno != EAGAIN)
+ continue;
+ else {
+ sprintf(message_, "RtApiOss: OSS record device (%s) is busy.", device_name);
+ error(RtError::WARNING);
+ // still count it for now
+ }
+ }
+ }
+ else {
+ sprintf(message_, "RtApiOss: OSS playback device (%s) is busy.", device_name);
+ error(RtError::WARNING);
+ // still count it for now
+ }
+ }
- for ( int i=0; i<nDevices; i++ ) {
- if ( id == devices[i].id[0] ) return i;
+ if (fd >= 0) close(fd);
+ device.name.erase();
+ device.name.append( (const char *)device_name, strlen(device_name)+1);
+ devices_.push_back(device);
+ nDevices_++;
}
-
- return 0;
}
-static bool deviceSupportsFormat( AudioDeviceID id, bool isInput,
- AudioStreamBasicDescription *desc, bool isDuplex )
+void RtApiOss :: probeDeviceInfo(RtApiDevice *info)
{
- OSStatus result = noErr;
+ int i, fd, channels, mask;
+
+ // The OSS API doesn't provide a means for probing the capabilities
+ // of devices. Thus, we'll just pursue a brute force method.
+
+ // First try for playback
+ fd = open(info->name.c_str(), O_WRONLY | O_NONBLOCK);
+ if (fd == -1) {
+ // Open device failed ... either busy or doesn't exist
+ if (errno == EBUSY || errno == EAGAIN)
+ sprintf(message_, "RtApiOss: OSS playback device (%s) is busy and cannot be probed.",
+ info->name.c_str());
+ else
+ sprintf(message_, "RtApiOss: OSS playback device (%s) open error.", info->name.c_str());
+ error(RtError::DEBUG_WARNING);
+ goto capture_probe;
+ }
+
+ // We have an open device ... see how many channels it can handle
+ for (i=MAX_CHANNELS; i>0; i--) {
+ channels = i;
+ if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) {
+ // This would normally indicate some sort of hardware error, but under ALSA's
+ // OSS emulation, it sometimes indicates an invalid channel value. Further,
+ // the returned channel value is not changed. So, we'll ignore the possible
+ // hardware error.
+ continue; // try next channel number
+ }
+ // Check to see whether the device supports the requested number of channels
+ if (channels != i ) continue; // try next channel number
+ // If here, we found the largest working channel value
+ break;
+ }
+ info->maxOutputChannels = i;
+
+ // Now find the minimum number of channels it can handle
+ for (i=1; i<=info->maxOutputChannels; i++) {
+ channels = i;
+ if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
+ continue; // try next channel number
+ // If here, we found the smallest working channel value
+ break;
+ }
+ info->minOutputChannels = i;
+ close(fd);
+
+ capture_probe:
+ // Now try for capture
+ fd = open(info->name.c_str(), O_RDONLY | O_NONBLOCK);
+ if (fd == -1) {
+ // Open device for capture failed ... either busy or doesn't exist
+ if (errno == EBUSY || errno == EAGAIN)
+ sprintf(message_, "RtApiOss: OSS capture device (%s) is busy and cannot be probed.",
+ info->name.c_str());
+ else
+ sprintf(message_, "RtApiOss: OSS capture device (%s) open error.", info->name.c_str());
+ error(RtError::DEBUG_WARNING);
+ if (info->maxOutputChannels == 0)
+ // didn't open for playback either ... device invalid
+ return;
+ goto probe_parameters;
+ }
+
+ // We have the device open for capture ... see how many channels it can handle
+ for (i=MAX_CHANNELS; i>0; i--) {
+ channels = i;
+ if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) {
+ continue; // as above
+ }
+ // If here, we found a working channel value
+ break;
+ }
+ info->maxInputChannels = i;
+
+ // Now find the minimum number of channels it can handle
+ for (i=1; i<=info->maxInputChannels; i++) {
+ channels = i;
+ if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
+ continue; // try next channel number
+ // If here, we found the smallest working channel value
+ break;
+ }
+ info->minInputChannels = i;
+ close(fd);
+
+ if (info->maxOutputChannels == 0 && info->maxInputChannels == 0) {
+ sprintf(message_, "RtApiOss: device (%s) reports zero channels for input and output.",
+ info->name.c_str());
+ error(RtError::DEBUG_WARNING);
+ return;
+ }
+
+ // If device opens for both playback and capture, we determine the channels.
+ if (info->maxOutputChannels == 0 || info->maxInputChannels == 0)
+ goto probe_parameters;
+
+ fd = open(info->name.c_str(), O_RDWR | O_NONBLOCK);
+ if (fd == -1)
+ goto probe_parameters;
+
+ ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
+ ioctl(fd, SNDCTL_DSP_GETCAPS, &mask);
+ if (mask & DSP_CAP_DUPLEX) {
+ info->hasDuplexSupport = true;
+ // We have the device open for duplex ... see how many channels it can handle
+ for (i=MAX_CHANNELS; i>0; i--) {
+ channels = i;
+ if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
+ continue; // as above
+ // If here, we found a working channel value
+ break;
+ }
+ info->maxDuplexChannels = i;
+
+ // Now find the minimum number of channels it can handle
+ for (i=1; i<=info->maxDuplexChannels; i++) {
+ channels = i;
+ if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
+ continue; // try next channel number
+ // If here, we found the smallest working channel value
+ break;
+ }
+ info->minDuplexChannels = i;
+ }
+ close(fd);
+
+ probe_parameters:
+ // At this point, we need to figure out the supported data formats
+ // and sample rates. We'll proceed by openning the device in the
+ // direction with the maximum number of channels, or playback if
+ // they are equal. This might limit our sample rate options, but so
+ // be it.
+
+ if (info->maxOutputChannels >= info->maxInputChannels) {
+ fd = open(info->name.c_str(), O_WRONLY | O_NONBLOCK);
+ channels = info->maxOutputChannels;
+ }
+ else {
+ fd = open(info->name.c_str(), O_RDONLY | O_NONBLOCK);
+ channels = info->maxInputChannels;
+ }
+
+ if (fd == -1) {
+ // We've got some sort of conflict ... abort
+ sprintf(message_, "RtApiOss: device (%s) won't reopen during probe.",
+ info->name.c_str());
+ error(RtError::DEBUG_WARNING);
+ return;
+ }
+
+ // We have an open device ... set to maximum channels.
+ i = channels;
+ if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) {
+ // We've got some sort of conflict ... abort
+ close(fd);
+ sprintf(message_, "RtApiOss: device (%s) won't revert to previous channel setting.",
+ info->name.c_str());
+ error(RtError::DEBUG_WARNING);
+ return;
+ }
+
+ if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) {
+ close(fd);
+ sprintf(message_, "RtApiOss: device (%s) can't get supported audio formats.",
+ info->name.c_str());
+ error(RtError::DEBUG_WARNING);
+ return;
+ }
+
+ // Probe the supported data formats ... we don't care about endian-ness just yet.
+ int format;
+ info->nativeFormats = 0;
+#if defined (AFMT_S32_BE)
+ // This format does not seem to be in the 2.4 kernel version of OSS soundcard.h
+ if (mask & AFMT_S32_BE) {
+ format = AFMT_S32_BE;
+ info->nativeFormats |= RTAUDIO_SINT32;
+ }
+#endif
+#if defined (AFMT_S32_LE)
+ /* This format is not in the 2.4.4 kernel version of OSS soundcard.h */
+ if (mask & AFMT_S32_LE) {
+ format = AFMT_S32_LE;
+ info->nativeFormats |= RTAUDIO_SINT32;
+ }
+#endif
+ if (mask & AFMT_S8) {
+ format = AFMT_S8;
+ info->nativeFormats |= RTAUDIO_SINT8;
+ }
+ if (mask & AFMT_S16_BE) {
+ format = AFMT_S16_BE;
+ info->nativeFormats |= RTAUDIO_SINT16;
+ }
+ if (mask & AFMT_S16_LE) {
+ format = AFMT_S16_LE;
+ info->nativeFormats |= RTAUDIO_SINT16;
+ }
+
+ // Check that we have at least one supported format
+ if (info->nativeFormats == 0) {
+ close(fd);
+ sprintf(message_, "RtApiOss: device (%s) data format not supported by RtAudio.",
+ info->name.c_str());
+ error(RtError::DEBUG_WARNING);
+ return;
+ }
+
+ // Set the format
+ i = format;
+ if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1 || format != i) {
+ close(fd);
+ sprintf(message_, "RtApiOss: device (%s) error setting data format.",
+ info->name.c_str());
+ error(RtError::DEBUG_WARNING);
+ return;
+ }
+
+ // Probe the supported sample rates.
+ info->sampleRates.clear();
+ for (unsigned int k=0; k<MAX_SAMPLE_RATES; k++) {
+ int speed = SAMPLE_RATES[k];
+ if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) != -1 && speed == (int)SAMPLE_RATES[k])
+ info->sampleRates.push_back(speed);
+ }
+
+ if (info->sampleRates.size() == 0) {
+ close(fd);
+ sprintf(message_, "RtApiOss: no supported sample rates found for device (%s).",
+ info->name.c_str());
+ error(RtError::DEBUG_WARNING);
+ return;
+ }
+
+ // That's all ... close the device and return
+ close(fd);
+ info->probed = true;
+ return;
+}
+
+bool RtApiOss :: probeDeviceOpen(int device, StreamMode mode, int channels,
+ int sampleRate, RtAudioFormat format,
+ int *bufferSize, int numberOfBuffers)
+{
+ int buffers, buffer_bytes, device_channels, device_format;
+ int srate, temp, fd;
+ int *handle = (int *) stream_.apiHandle;
+
+ const char *name = devices_[device].name.c_str();
+
+ if (mode == OUTPUT)
+ fd = open(name, O_WRONLY | O_NONBLOCK);
+ else { // mode == INPUT
+ if (stream_.mode == OUTPUT && stream_.device[0] == device) {
+ // We just set the same device for playback ... close and reopen for duplex (OSS only).
+ close(handle[0]);
+ handle[0] = 0;
+ // First check that the number previously set channels is the same.
+ if (stream_.nUserChannels[0] != channels) {
+ sprintf(message_, "RtApiOss: input/output channels must be equal for OSS duplex device (%s).", name);
+ goto error;
+ }
+ fd = open(name, O_RDWR | O_NONBLOCK);
+ }
+ else
+ fd = open(name, O_RDONLY | O_NONBLOCK);
+ }
+
+ if (fd == -1) {
+ if (errno == EBUSY || errno == EAGAIN)
+ sprintf(message_, "RtApiOss: device (%s) is busy and cannot be opened.",
+ name);
+ else
+ sprintf(message_, "RtApiOss: device (%s) cannot be opened.", name);
+ goto error;
+ }
+
+ // Now reopen in blocking mode.
+ close(fd);
+ if (mode == OUTPUT)
+ fd = open(name, O_WRONLY | O_SYNC);
+ else { // mode == INPUT
+ if (stream_.mode == OUTPUT && stream_.device[0] == device)
+ fd = open(name, O_RDWR | O_SYNC);
+ else
+ fd = open(name, O_RDONLY | O_SYNC);
+ }
+
+ if (fd == -1) {
+ sprintf(message_, "RtApiOss: device (%s) cannot be opened.", name);
+ goto error;
+ }
+
+ // Get the sample format mask
+ int mask;
+ if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) {
+ close(fd);
+ sprintf(message_, "RtApiOss: device (%s) can't get supported audio formats.",
+ name);
+ goto error;
+ }
+
+ // Determine how to set the device format.
+ stream_.userFormat = format;
+ device_format = -1;
+ stream_.doByteSwap[mode] = false;
+ if (format == RTAUDIO_SINT8) {
+ if (mask & AFMT_S8) {
+ device_format = AFMT_S8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ }
+ else if (format == RTAUDIO_SINT16) {
+ if (mask & AFMT_S16_NE) {
+ device_format = AFMT_S16_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+#if BYTE_ORDER == LITTLE_ENDIAN
+ else if (mask & AFMT_S16_BE) {
+ device_format = AFMT_S16_BE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.doByteSwap[mode] = true;
+ }
+#else
+ else if (mask & AFMT_S16_LE) {
+ device_format = AFMT_S16_LE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.doByteSwap[mode] = true;
+ }
+#endif
+ }
+#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE)
+ else if (format == RTAUDIO_SINT32) {
+ if (mask & AFMT_S32_NE) {
+ device_format = AFMT_S32_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ }
+#if BYTE_ORDER == LITTLE_ENDIAN
+ else if (mask & AFMT_S32_BE) {
+ device_format = AFMT_S32_BE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ stream_.doByteSwap[mode] = true;
+ }
+#else
+ else if (mask & AFMT_S32_LE) {
+ device_format = AFMT_S32_LE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ stream_.doByteSwap[mode] = true;
+ }
+#endif
+ }
+#endif
+
+ if (device_format == -1) {
+ // The user requested format is not natively supported by the device.
+ if (mask & AFMT_S16_NE) {
+ device_format = AFMT_S16_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+#if BYTE_ORDER == LITTLE_ENDIAN
+ else if (mask & AFMT_S16_BE) {
+ device_format = AFMT_S16_BE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.doByteSwap[mode] = true;
+ }
+#else
+ else if (mask & AFMT_S16_LE) {
+ device_format = AFMT_S16_LE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.doByteSwap[mode] = true;
+ }
+#endif
+#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE)
+ else if (mask & AFMT_S32_NE) {
+ device_format = AFMT_S32_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ }
+#if BYTE_ORDER == LITTLE_ENDIAN
+ else if (mask & AFMT_S32_BE) {
+ device_format = AFMT_S32_BE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ stream_.doByteSwap[mode] = true;
+ }
+#else
+ else if (mask & AFMT_S32_LE) {
+ device_format = AFMT_S32_LE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ stream_.doByteSwap[mode] = true;
+ }
+#endif
+#endif
+ else if (mask & AFMT_S8) {
+ device_format = AFMT_S8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ }
+
+ if (stream_.deviceFormat[mode] == 0) {
+ // This really shouldn't happen ...
+ close(fd);
+ sprintf(message_, "RtApiOss: device (%s) data format not supported by RtAudio.",
+ name);
+ goto error;
+ }
+
+ // Determine the number of channels for this device. Note that the
+ // channel value requested by the user might be < min_X_Channels.
+ stream_.nUserChannels[mode] = channels;
+ device_channels = channels;
+ if (mode == OUTPUT) {
+ if (channels < devices_[device].minOutputChannels)
+ device_channels = devices_[device].minOutputChannels;
+ }
+ else { // mode == INPUT
+ if (stream_.mode == OUTPUT && stream_.device[0] == device) {
+ // We're doing duplex setup here.
+ if (channels < devices_[device].minDuplexChannels)
+ device_channels = devices_[device].minDuplexChannels;
+ }
+ else {
+ if (channels < devices_[device].minInputChannels)
+ device_channels = devices_[device].minInputChannels;
+ }
+ }
+ stream_.nDeviceChannels[mode] = device_channels;
+
+ // Attempt to set the buffer size. According to OSS, the minimum
+ // number of buffers is two. The supposed minimum buffer size is 16
+ // bytes, so that will be our lower bound. The argument to this
+ // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
+ // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
+ // We'll check the actual value used near the end of the setup
+ // procedure.
+ buffer_bytes = *bufferSize * formatBytes(stream_.deviceFormat[mode]) * device_channels;
+ if (buffer_bytes < 16) buffer_bytes = 16;
+ buffers = numberOfBuffers;
+ if (buffers < 2) buffers = 2;
+ temp = ((int) buffers << 16) + (int)(log10((double)buffer_bytes)/log10(2.0));
+ if (ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &temp)) {
+ close(fd);
+ sprintf(message_, "RtApiOss: error setting fragment size for device (%s).",
+ name);
+ goto error;
+ }
+ stream_.nBuffers = buffers;
+
+ // Set the data format.
+ temp = device_format;
+ if (ioctl(fd, SNDCTL_DSP_SETFMT, &device_format) == -1 || device_format != temp) {
+ close(fd);
+ sprintf(message_, "RtApiOss: error setting data format for device (%s).",
+ name);
+ goto error;
+ }
+
+ // Set the number of channels.
+ temp = device_channels;
+ if (ioctl(fd, SNDCTL_DSP_CHANNELS, &device_channels) == -1 || device_channels != temp) {
+ close(fd);
+ sprintf(message_, "RtApiOss: error setting %d channels on device (%s).",
+ temp, name);
+ goto error;
+ }
+
+ // Set the sample rate.
+ srate = sampleRate;
+ temp = srate;
+ if (ioctl(fd, SNDCTL_DSP_SPEED, &srate) == -1) {
+ close(fd);
+ sprintf(message_, "RtApiOss: error setting sample rate = %d on device (%s).",
+ temp, name);
+ goto error;
+ }
+
+ // Verify the sample rate setup worked.
+ if (abs(srate - temp) > 100) {
+ close(fd);
+ sprintf(message_, "RtApiOss: error ... audio device (%s) doesn't support sample rate of %d.",
+ name, temp);
+ goto error;
+ }
+ stream_.sampleRate = sampleRate;
+
+ if (ioctl(fd, SNDCTL_DSP_GETBLKSIZE, &buffer_bytes) == -1) {
+ close(fd);
+ sprintf(message_, "RtApiOss: error getting buffer size for device (%s).",
+ name);
+ goto error;
+ }
+
+ // Save buffer size (in sample frames).
+ *bufferSize = buffer_bytes / (formatBytes(stream_.deviceFormat[mode]) * device_channels);
+ stream_.bufferSize = *bufferSize;
+
+ if (mode == INPUT && stream_.mode == OUTPUT &&
+ stream_.device[0] == device) {
+ // We're doing duplex setup here.
+ stream_.deviceFormat[0] = stream_.deviceFormat[1];
+ stream_.nDeviceChannels[0] = device_channels;
+ }
+
+ // Allocate the stream handles if necessary and then save.
+ if ( stream_.apiHandle == 0 ) {
+ handle = (int *) calloc(2, sizeof(int));
+ stream_.apiHandle = (void *) handle;
+ handle[0] = 0;
+ handle[1] = 0;
+ }
+ else {
+ handle = (int *) stream_.apiHandle;
+ }
+ handle[mode] = fd;
+
+ // Set flags for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if (stream_.userFormat != stream_.deviceFormat[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode])
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate necessary internal buffers
+ if ( stream_.nUserChannels[0] != stream_.nUserChannels[1] ) {
+
+ long buffer_bytes;
+ if (stream_.nUserChannels[0] >= stream_.nUserChannels[1])
+ buffer_bytes = stream_.nUserChannels[0];
+ else
+ buffer_bytes = stream_.nUserChannels[1];
+
+ buffer_bytes *= *bufferSize * formatBytes(stream_.userFormat);
+ if (stream_.userBuffer) free(stream_.userBuffer);
+ stream_.userBuffer = (char *) calloc(buffer_bytes, 1);
+ if (stream_.userBuffer == NULL) {
+ close(fd);
+ sprintf(message_, "RtApiOss: error allocating user buffer memory (%s).",
+ name);
+ goto error;
+ }
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ long buffer_bytes;
+ bool makeBuffer = true;
+ if ( mode == OUTPUT )
+ buffer_bytes = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
+ else { // mode == INPUT
+ buffer_bytes = stream_.nDeviceChannels[1] * formatBytes(stream_.deviceFormat[1]);
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ long bytes_out = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
+ if ( buffer_bytes < bytes_out ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ buffer_bytes *= *bufferSize;
+ if (stream_.deviceBuffer) free(stream_.deviceBuffer);
+ stream_.deviceBuffer = (char *) calloc(buffer_bytes, 1);
+ if (stream_.deviceBuffer == NULL) {
+ close(fd);
+ sprintf(message_, "RtApiOss: error allocating device buffer memory (%s).",
+ name);
+ goto error;
+ }
+ }
+ }
+
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+
+ if ( stream_.mode == OUTPUT && mode == INPUT ) {
+ stream_.mode = DUPLEX;
+ if (stream_.device[0] == device)
+ handle[0] = fd;
+ }
+ else
+ stream_.mode = mode;
+
+ return SUCCESS;
+
+ error:
+ if (handle) {
+ if (handle[0])
+ close(handle[0]);
+ free(handle);
+ stream_.apiHandle = 0;
+ }
+
+ if (stream_.userBuffer) {
+ free(stream_.userBuffer);
+ stream_.userBuffer = 0;
+ }
+
+ error(RtError::WARNING);
+ return FAILURE;
+}
+
+void RtApiOss :: closeStream()
+{
+ // We don't want an exception to be thrown here because this
+ // function is called by our class destructor. So, do our own
+ // stream check.
+ if ( stream_.mode == UNINITIALIZED ) {
+ sprintf(message_, "RtApiOss::closeStream(): no open stream to close!");
+ error(RtError::WARNING);
+ return;
+ }
+
+ int *handle = (int *) stream_.apiHandle;
+ if (stream_.state == STREAM_RUNNING) {
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX)
+ ioctl(handle[0], SNDCTL_DSP_RESET, 0);
+ else
+ ioctl(handle[1], SNDCTL_DSP_RESET, 0);
+ stream_.state = STREAM_STOPPED;
+ }
+
+ if (stream_.callbackInfo.usingCallback) {
+ stream_.callbackInfo.usingCallback = false;
+ pthread_join(stream_.callbackInfo.thread, NULL);
+ }
+
+ if (handle) {
+ if (handle[0]) close(handle[0]);
+ if (handle[1]) close(handle[1]);
+ free(handle);
+ stream_.apiHandle = 0;
+ }
+
+ if (stream_.userBuffer) {
+ free(stream_.userBuffer);
+ stream_.userBuffer = 0;
+ }
+
+ if (stream_.deviceBuffer) {
+ free(stream_.deviceBuffer);
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.mode = UNINITIALIZED;
+}
+
+void RtApiOss :: startStream()
+{
+ verifyStream();
+ if (stream_.state == STREAM_RUNNING) return;
+
+ MUTEX_LOCK(&stream_.mutex);
+
+ stream_.state = STREAM_RUNNING;
+
+ // No need to do anything else here ... OSS automatically starts
+ // when fed samples.
+
+ MUTEX_UNLOCK(&stream_.mutex);
+}
+
+void RtApiOss :: stopStream()
+{
+ verifyStream();
+ if (stream_.state == STREAM_STOPPED) return;
+
+ // Change the state before the lock to improve shutdown response
+ // when using a callback.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK(&stream_.mutex);
+
+ int err;
+ int *handle = (int *) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ err = ioctl(handle[0], SNDCTL_DSP_POST, 0);
+ //err = ioctl(handle[0], SNDCTL_DSP_SYNC, 0);
+ if (err < -1) {
+ sprintf(message_, "RtApiOss: error stopping device (%s).",
+ devices_[stream_.device[0]].name.c_str());
+ error(RtError::DRIVER_ERROR);
+ }
+ }
+ else {
+ err = ioctl(handle[1], SNDCTL_DSP_POST, 0);
+ //err = ioctl(handle[1], SNDCTL_DSP_SYNC, 0);
+ if (err < -1) {
+ sprintf(message_, "RtApiOss: error stopping device (%s).",
+ devices_[stream_.device[1]].name.c_str());
+ error(RtError::DRIVER_ERROR);
+ }
+ }
+
+ MUTEX_UNLOCK(&stream_.mutex);
+}
+
+void RtApiOss :: abortStream()
+{
+ stopStream();
+}
+
+int RtApiOss :: streamWillBlock()
+{
+ verifyStream();
+ if (stream_.state == STREAM_STOPPED) return 0;
+
+ MUTEX_LOCK(&stream_.mutex);
+
+ int bytes = 0, channels = 0, frames = 0;
+ audio_buf_info info;
+ int *handle = (int *) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ ioctl(handle[0], SNDCTL_DSP_GETOSPACE, &info);
+ bytes = info.bytes;
+ channels = stream_.nDeviceChannels[0];
+ }
+
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ ioctl(handle[1], SNDCTL_DSP_GETISPACE, &info);
+ if (stream_.mode == DUPLEX ) {
+ bytes = (bytes < info.bytes) ? bytes : info.bytes;
+ channels = stream_.nDeviceChannels[0];
+ }
+ else {
+ bytes = info.bytes;
+ channels = stream_.nDeviceChannels[1];
+ }
+ }
+
+ frames = (int) (bytes / (channels * formatBytes(stream_.deviceFormat[0])));
+ frames -= stream_.bufferSize;
+ if (frames < 0) frames = 0;
+
+ MUTEX_UNLOCK(&stream_.mutex);
+ return frames;
+}
+
+void RtApiOss :: tickStream()
+{
+ verifyStream();
+
+ int stopStream = 0;
+ if (stream_.state == STREAM_STOPPED) {
+ if (stream_.callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds
+ return;
+ }
+ else if (stream_.callbackInfo.usingCallback) {
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ stopStream = callback(stream_.userBuffer, stream_.bufferSize, stream_.callbackInfo.userData);
+ }
+
+ MUTEX_LOCK(&stream_.mutex);
+
+ // The state might change while waiting on a mutex.
+ if (stream_.state == STREAM_STOPPED)
+ goto unlock;
+
+ int result, *handle;
+ char *buffer;
+ int samples;
+ RtAudioFormat format;
+ handle = (int *) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+
+ // Setup parameters and do buffer conversion if necessary.
+ if (stream_.doConvertBuffer[0]) {
+ convertStreamBuffer(OUTPUT);
+ buffer = stream_.deviceBuffer;
+ samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
+ }
+ else {
+ buffer = stream_.userBuffer;
+ samples = stream_.bufferSize * stream_.nUserChannels[0];
+ format = stream_.userFormat;
+ }
+
+ // Do byte swapping if necessary.
+ if (stream_.doByteSwap[0])
+ byteSwapBuffer(buffer, samples, format);
+
+ // Write samples to device.
+ result = write(handle[0], buffer, samples * formatBytes(format));
+
+ if (result == -1) {
+ // This could be an underrun, but the basic OSS API doesn't provide a means for determining that.
+ sprintf(message_, "RtApiOss: audio write error for device (%s).",
+ devices_[stream_.device[0]].name.c_str());
+ error(RtError::DRIVER_ERROR);
+ }
+ }
+
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+
+ // Setup parameters.
+ if (stream_.doConvertBuffer[1]) {
+ buffer = stream_.deviceBuffer;
+ samples = stream_.bufferSize * stream_.nDeviceChannels[1];
+ format = stream_.deviceFormat[1];
+ }
+ else {
+ buffer = stream_.userBuffer;
+ samples = stream_.bufferSize * stream_.nUserChannels[1];
+ format = stream_.userFormat;
+ }
+
+ // Read samples from device.
+ result = read(handle[1], buffer, samples * formatBytes(format));
+
+ if (result == -1) {
+ // This could be an overrun, but the basic OSS API doesn't provide a means for determining that.
+ sprintf(message_, "RtApiOss: audio read error for device (%s).",
+ devices_[stream_.device[1]].name.c_str());
+ error(RtError::DRIVER_ERROR);
+ }
+
+ // Do byte swapping if necessary.
+ if (stream_.doByteSwap[1])
+ byteSwapBuffer(buffer, samples, format);
+
+ // Do buffer conversion if necessary.
+ if (stream_.doConvertBuffer[1])
+ convertStreamBuffer(INPUT);
+ }
+
+ unlock:
+ MUTEX_UNLOCK(&stream_.mutex);
+
+ if (stream_.callbackInfo.usingCallback && stopStream)
+ this->stopStream();
+}
+
+void RtApiOss :: setStreamCallback(RtAudioCallback callback, void *userData)
+{
+ verifyStream();
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ if ( info->usingCallback ) {
+ sprintf(message_, "RtApiOss: A callback is already set for this stream!");
+ error(RtError::WARNING);
+ return;
+ }
+
+ info->callback = (void *) callback;
+ info->userData = userData;
+ info->usingCallback = true;
+ info->object = (void *) this;
+
+ // Set the thread attributes for joinable and realtime scheduling
+ // priority. The higher priority will only take affect if the
+ // program is run as root or suid.
+ pthread_attr_t attr;
+ pthread_attr_init(&attr);
+ pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE);
+ pthread_attr_setschedpolicy(&attr, SCHED_RR);
+
+ int err = pthread_create(&(info->thread), &attr, ossCallbackHandler, &stream_.callbackInfo);
+ pthread_attr_destroy(&attr);
+ if (err) {
+ info->usingCallback = false;
+ sprintf(message_, "RtApiOss: error starting callback thread!");
+ error(RtError::THREAD_ERROR);
+ }
+}
+
+void RtApiOss :: cancelStreamCallback()
+{
+ verifyStream();
+
+ if (stream_.callbackInfo.usingCallback) {
+
+ if (stream_.state == STREAM_RUNNING)
+ stopStream();
+
+ MUTEX_LOCK(&stream_.mutex);
+
+ stream_.callbackInfo.usingCallback = false;
+ pthread_join(stream_.callbackInfo.thread, NULL);
+ stream_.callbackInfo.thread = 0;
+ stream_.callbackInfo.callback = NULL;
+ stream_.callbackInfo.userData = NULL;
+
+ MUTEX_UNLOCK(&stream_.mutex);
+ }
+}
+
+extern "C" void *ossCallbackHandler(void *ptr)
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiOss *object = (RtApiOss *) info->object;
+ bool *usingCallback = &info->usingCallback;
+
+ while ( *usingCallback ) {
+ pthread_testcancel();
+ try {
+ object->tickStream();
+ }
+ catch (RtError &exception) {
+ fprintf(stderr, "\nRtApiOss: callback thread error (%s) ... closing thread.\n\n",
+ exception.getMessageString());
+ break;
+ }
+ }
+
+ return 0;
+}
+
+//******************** End of __LINUX_OSS__ *********************//
+#endif
+
+#if defined(__MACOSX_CORE__)
+
+
+// The OS X CoreAudio API is designed to use a separate callback
+// procedure for each of its audio devices. A single RtAudio duplex
+// stream using two different devices is supported here, though it
+// cannot be guaranteed to always behave correctly because we cannot
+// synchronize these two callbacks. This same functionality can be
+// achieved with better synchrony by opening two separate streams for
+// the devices and using RtAudio blocking calls (i.e. tickStream()).
+//
+// A property listener is installed for over/underrun information.
+// However, no functionality is currently provided to allow property
+// listeners to trigger user handlers because it is unclear what could
+// be done if a critical stream parameter (buffer size, sample rate,
+// device disconnect) notification arrived. The listeners entail
+// quite a bit of extra code and most likely, a user program wouldn't
+// be prepared for the result anyway.
+
+// A structure to hold various information related to the CoreAuio API
+// implementation.
+struct CoreHandle {
+ UInt32 index[2];
+ bool stopStream;
+ bool xrun;
+ char *deviceBuffer;
+ pthread_cond_t condition;
+
+ CoreHandle()
+ :stopStream(false), xrun(false), deviceBuffer(0) {}
+};
+
+RtApiCore :: RtApiCore()
+{
+ this->initialize();
+
+ if (nDevices_ <= 0) {
+ sprintf(message_, "RtApiCore: no Macintosh OS-X Core Audio devices found!");
+ error(RtError::NO_DEVICES_FOUND);
+ }
+}
+
+RtApiCore :: ~RtApiCore()
+{
+ // The subclass destructor gets called before the base class
+ // destructor, so close an existing stream before deallocating
+ // apiDeviceId memory.
+ if ( stream_.mode != UNINITIALIZED ) closeStream();
+
+ // Free our allocated apiDeviceId memory.
+ AudioDeviceID *id;
+ for ( unsigned int i=0; i<devices_.size(); i++ ) {
+ id = (AudioDeviceID *) devices_[i].apiDeviceId;
+ if (id) free(id);
+ }
+}
+
+void RtApiCore :: initialize(void)
+{
+ OSStatus err = noErr;
+ UInt32 dataSize;
+ AudioDeviceID *deviceList = NULL;
+ nDevices_ = 0;
+
+ // Find out how many audio devices there are, if any.
+ err = AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &dataSize, NULL);
+ if (err != noErr) {
+ sprintf(message_, "RtApiCore: OS-X error getting device info!");
+ error(RtError::SYSTEM_ERROR);
+ }
+
+ nDevices_ = dataSize / sizeof(AudioDeviceID);
+ if (nDevices_ == 0) return;
+
+ // Make space for the devices we are about to get.
+ deviceList = (AudioDeviceID *) malloc( dataSize );
+ if (deviceList == NULL) {
+ sprintf(message_, "RtApiCore: memory allocation error during initialization!");
+ error(RtError::MEMORY_ERROR);
+ }
+
+ // Get the array of AudioDeviceIDs.
+ err = AudioHardwareGetProperty(kAudioHardwarePropertyDevices, &dataSize, (void *) deviceList);
+ if (err != noErr) {
+ free(deviceList);
+ sprintf(message_, "RtApiCore: OS-X error getting device properties!");
+ error(RtError::SYSTEM_ERROR);
+ }
+
+ // Create list of device structures and write device identifiers.
+ RtApiDevice device;
+ AudioDeviceID *id;
+ for (int i=0; i<nDevices_; i++) {
+ devices_.push_back(device);
+ id = (AudioDeviceID *) malloc( sizeof(AudioDeviceID) );
+ *id = deviceList[i];
+ devices_[i].apiDeviceId = (void *) id;
+ }
+
+ free(deviceList);
+}
+
+int RtApiCore :: getDefaultInputDevice(void)
+{
+ AudioDeviceID id, *deviceId;
+ UInt32 dataSize = sizeof( AudioDeviceID );
+
+ OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultInputDevice,
+ &dataSize, &id );
+
+ if (result != noErr) {
+ sprintf( message_, "RtApiCore: OS-X error getting default input device." );
+ error(RtError::WARNING);
+ return 0;
+ }
+
+ for ( int i=0; i<nDevices_; i++ ) {
+ deviceId = (AudioDeviceID *) devices_[i].apiDeviceId;
+ if ( id == *deviceId ) return i;
+ }
+
+ return 0;
+}
+
+int RtApiCore :: getDefaultOutputDevice(void)
+{
+ AudioDeviceID id, *deviceId;
+ UInt32 dataSize = sizeof( AudioDeviceID );
+
+ OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultOutputDevice,
+ &dataSize, &id );
+
+ if (result != noErr) {
+ sprintf( message_, "RtApiCore: OS-X error getting default output device." );
+ error(RtError::WARNING);
+ return 0;
+ }
+
+ for ( int i=0; i<nDevices_; i++ ) {
+ deviceId = (AudioDeviceID *) devices_[i].apiDeviceId;
+ if ( id == *deviceId ) return i;
+ }
+
+ return 0;
+}
+
+static bool deviceSupportsFormat( AudioDeviceID id, bool isInput,
+ AudioStreamBasicDescription *desc, bool isDuplex )
+{
+ OSStatus result = noErr;
UInt32 dataSize = sizeof( AudioStreamBasicDescription );
result = AudioDeviceGetProperty( id, 0, isInput,
return false;
}
-void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info)
+void RtApiCore :: probeDeviceInfo( RtApiDevice *info )
{
OSStatus err = noErr;
char name[256];
char fullname[512];
UInt32 dataSize = 256;
- err = AudioDeviceGetProperty( info->id[0], 0, false,
+ AudioDeviceID *id = (AudioDeviceID *) info->apiDeviceId;
+ err = AudioDeviceGetProperty( *id, 0, false,
kAudioDevicePropertyDeviceManufacturer,
&dataSize, name );
if (err != noErr) {
- sprintf( message, "RtAudio: OSX error getting device manufacturer." );
+ sprintf( message_, "RtApiCore: OS-X error getting device manufacturer." );
error(RtError::DEBUG_WARNING);
return;
}
strcat(fullname, ": " );
dataSize = 256;
- err = AudioDeviceGetProperty( info->id[0], 0, false,
+ err = AudioDeviceGetProperty( *id, 0, false,
kAudioDevicePropertyDeviceName,
&dataSize, name );
if (err != noErr) {
- sprintf( message, "RtAudio: OSX error getting device name." );
+ sprintf( message_, "RtApiCore: OS-X error getting device name." );
error(RtError::DEBUG_WARNING);
return;
}
strncat(fullname, name, 254);
- strncat(info->name, fullname, 128);
+ info->name.erase();
+ info->name.append( (const char *)fullname, strlen(fullname)+1);
// Get output channel information.
- unsigned int i, minChannels, maxChannels, nStreams = 0;
+ unsigned int i, minChannels = 0, maxChannels = 0, nStreams = 0;
AudioBufferList *bufferList = nil;
- err = AudioDeviceGetPropertyInfo( info->id[0], 0, false,
+ err = AudioDeviceGetPropertyInfo( *id, 0, false,
kAudioDevicePropertyStreamConfiguration,
&dataSize, NULL );
if (err == noErr && dataSize > 0) {
bufferList = (AudioBufferList *) malloc( dataSize );
if (bufferList == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
+ sprintf(message_, "RtApiCore: memory allocation error!");
error(RtError::DEBUG_WARNING);
return;
}
- err = AudioDeviceGetProperty( info->id[0], 0, false,
+ err = AudioDeviceGetProperty( *id, 0, false,
kAudioDevicePropertyStreamConfiguration,
&dataSize, bufferList );
if (err == noErr) {
}
}
}
+ free (bufferList);
+
if (err != noErr || dataSize <= 0) {
- sprintf( message, "RtAudio: OSX error getting output channels for device (%s).", info->name );
+ sprintf( message_, "RtApiCore: OS-X error getting output channels for device (%s).",
+ info->name.c_str() );
error(RtError::DEBUG_WARNING);
return;
}
- free (bufferList);
if ( nStreams ) {
if ( maxChannels > 0 )
info->maxOutputChannels = maxChannels;
// Get input channel information.
bufferList = nil;
- err = AudioDeviceGetPropertyInfo( info->id[0], 0, true,
+ err = AudioDeviceGetPropertyInfo( *id, 0, true,
kAudioDevicePropertyStreamConfiguration,
&dataSize, NULL );
if (err == noErr && dataSize > 0) {
bufferList = (AudioBufferList *) malloc( dataSize );
if (bufferList == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
+ sprintf(message_, "RtApiCore: memory allocation error!");
error(RtError::DEBUG_WARNING);
return;
}
- err = AudioDeviceGetProperty( info->id[0], 0, true,
+ err = AudioDeviceGetProperty( *id, 0, true,
kAudioDevicePropertyStreamConfiguration,
&dataSize, bufferList );
if (err == noErr) {
}
}
}
+ free (bufferList);
+
if (err != noErr || dataSize <= 0) {
- sprintf( message, "RtAudio: OSX error getting input channels for device (%s).", info->name );
+ sprintf( message_, "RtApiCore: OS-X error getting input channels for device (%s).",
+ info->name.c_str() );
error(RtError::DEBUG_WARNING);
return;
}
- free (bufferList);
if ( nStreams ) {
if ( maxChannels > 0 )
info->maxInputChannels = maxChannels;
if ( info->maxDuplexChannels > 0 ) isDuplex = true;
// Determine the supported sample rates.
- info->nSampleRates = 0;
- for (i=0; i<MAX_SAMPLE_RATES; i++) {
- description.mSampleRate = (double) SAMPLE_RATES[i];
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
- info->sampleRates[info->nSampleRates++] = SAMPLE_RATES[i];
+ info->sampleRates.clear();
+ for (unsigned int k=0; k<MAX_SAMPLE_RATES; k++) {
+ description.mSampleRate = (double) SAMPLE_RATES[k];
+ if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
+ info->sampleRates.push_back( SAMPLE_RATES[k] );
}
- if (info->nSampleRates == 0) {
- sprintf( message, "RtAudio: No supported sample rates found for OSX device (%s).", info->name );
+ if (info->sampleRates.size() == 0) {
+ sprintf( message_, "RtApiCore: No supported sample rates found for OS-X device (%s).",
+ info->name.c_str() );
error(RtError::DEBUG_WARNING);
return;
}
- // Check for continuous sample rate support.
- description.mSampleRate = kAudioStreamAnyRate;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) {
- info->sampleRates[1] = info->sampleRates[info->nSampleRates-1];
- info->nSampleRates = -1;
- }
-
// Determine the supported data formats.
info->nativeFormats = 0;
description.mFormatID = kAudioFormatLinearPCM;
description.mBitsPerChannel = 8;
description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
+ if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
info->nativeFormats |= RTAUDIO_SINT8;
else {
description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
+ if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
info->nativeFormats |= RTAUDIO_SINT8;
}
description.mBitsPerChannel = 16;
description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
+ if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
info->nativeFormats |= RTAUDIO_SINT16;
else {
description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
+ if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
info->nativeFormats |= RTAUDIO_SINT16;
}
description.mBitsPerChannel = 32;
description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
+ if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
info->nativeFormats |= RTAUDIO_SINT32;
else {
description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
+ if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
info->nativeFormats |= RTAUDIO_SINT32;
}
description.mBitsPerChannel = 24;
description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsAlignedHigh | kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
+ if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
info->nativeFormats |= RTAUDIO_SINT24;
else {
description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
+ if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
info->nativeFormats |= RTAUDIO_SINT24;
}
description.mBitsPerChannel = 32;
description.mFormatFlags = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
+ if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
info->nativeFormats |= RTAUDIO_FLOAT32;
else {
description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
+ if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
info->nativeFormats |= RTAUDIO_FLOAT32;
}
description.mBitsPerChannel = 64;
description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
+ if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
info->nativeFormats |= RTAUDIO_FLOAT64;
else {
description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
+ if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
info->nativeFormats |= RTAUDIO_FLOAT64;
}
// Check that we have at least one supported format.
if (info->nativeFormats == 0) {
- sprintf(message, "RtAudio: OSX PCM device (%s) data format not supported by RtAudio.",
- info->name);
+ sprintf(message_, "RtApiCore: OS-X device (%s) data format not supported by RtAudio.",
+ info->name.c_str());
error(RtError::DEBUG_WARNING);
return;
}
const AudioTimeStamp* inOutputTime,
void* infoPointer)
{
- CALLBACK_INFO *info = (CALLBACK_INFO *) infoPointer;
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
- RtAudio *object = (RtAudio *) info->object;
+ RtApiCore *object = (RtApiCore *) info->object;
try {
- object->callbackEvent( info->streamId, inDevice, (void *)inInputData, (void *)outOutputData );
+ object->callbackEvent( inDevice, (void *)inInputData, (void *)outOutputData );
}
catch (RtError &exception) {
- fprintf(stderr, "\nCallback handler error (%s)!\n\n", exception.getMessage());
+ fprintf(stderr, "\nRtApiCore: callback handler error (%s)!\n\n", exception.getMessageString());
return kAudioHardwareUnspecifiedError;
}
return kAudioHardwareNoError;
}
-/*
OSStatus deviceListener(AudioDeviceID inDevice,
UInt32 channel,
Boolean isInput,
AudioDevicePropertyID propertyID,
- void* infoPointer)
+ void* handlePointer)
{
- CALLBACK_INFO *info = (CALLBACK_INFO *) infoPointer;
-
- RtAudio *object = (RtAudio *) info->object;
- try {
- object->settingChange( info->streamId );
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nDevice listener error (%s)!\n\n", exception.getMessage());
- return kAudioHardwareUnspecifiedError;
+ CoreHandle *handle = (CoreHandle *) handlePointer;
+ if ( propertyID == kAudioDeviceProcessorOverload ) {
+ if ( isInput )
+ fprintf(stderr, "\nRtApiCore: OS-X audio input overrun detected!\n");
+ else
+ fprintf(stderr, "\nRtApiCore: OS-X audio output underrun detected!\n");
+ handle->xrun = true;
}
return kAudioHardwareNoError;
}
-*/
-bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
- STREAM_MODE mode, int channels,
- int sampleRate, RTAUDIO_FORMAT format,
- int *bufferSize, int numberOfBuffers)
+bool RtApiCore :: probeDeviceOpen( int device, StreamMode mode, int channels,
+ int sampleRate, RtAudioFormat format,
+ int *bufferSize, int numberOfBuffers )
{
- // Check to make sure we don't already have a stream accessing this device.
- RTAUDIO_STREAM *streamPtr;
- std::map<int, void *>::const_iterator i;
- for ( i=streams.begin(); i!=streams.end(); ++i ) {
- streamPtr = (RTAUDIO_STREAM *) i->second;
- if ( streamPtr->device[0] == device || streamPtr->device[1] == device ) {
- sprintf(message, "RtAudio: no current OS X support for multiple streams accessing the same device!");
- error(RtError::WARNING);
- return FAILURE;
- }
- }
-
// Setup for stream mode.
bool isInput = false;
- AudioDeviceID id = devices[device].id[0];
+ AudioDeviceID id = *((AudioDeviceID *) devices_[device].apiDeviceId);
if ( mode == INPUT ) isInput = true;
// Search for a stream which contains the desired number of channels.
OSStatus err = noErr;
UInt32 dataSize;
- unsigned int deviceChannels, nStreams;
+ unsigned int deviceChannels, nStreams = 0;
UInt32 iChannel = 0, iStream = 0;
AudioBufferList *bufferList = nil;
err = AudioDeviceGetPropertyInfo( id, 0, isInput,
if (err == noErr && dataSize > 0) {
bufferList = (AudioBufferList *) malloc( dataSize );
if (bufferList == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
+ sprintf(message_, "RtApiCore: memory allocation error in probeDeviceOpen()!");
error(RtError::DEBUG_WARNING);
return FAILURE;
}
&dataSize, bufferList );
if (err == noErr) {
- stream->deInterleave[mode] = false;
+ stream_.deInterleave[mode] = false;
nStreams = bufferList->mNumberBuffers;
for ( iStream=0; iStream<nStreams; iStream++ ) {
if ( bufferList->mBuffers[iStream].mNumberChannels >= (unsigned int) channels ) break;
if ( counter == channels ) {
iStream -= channels - 1;
iChannel -= channels - 1;
- stream->deInterleave[mode] = true;
+ stream_.deInterleave[mode] = true;
break;
}
iChannel += bufferList->mBuffers[iStream].mNumberChannels;
}
if (err != noErr || dataSize <= 0) {
if ( bufferList ) free( bufferList );
- sprintf( message, "RtAudio: OSX error getting channels for device (%s).", devices[device].name );
+ sprintf( message_, "RtApiCore: OS-X error getting channels for device (%s).",
+ devices_[device].name.c_str() );
error(RtError::DEBUG_WARNING);
return FAILURE;
}
if (iStream >= nStreams) {
free (bufferList);
- sprintf( message, "RtAudio: unable to find OSX audio stream on device (%s) for requested channels (%d).",
- devices[device].name, channels );
+ sprintf( message_, "RtApiCore: unable to find OS-X audio stream on device (%s) for requested channels (%d).",
+ devices_[device].name.c_str(), channels );
error(RtError::DEBUG_WARNING);
return FAILURE;
}
kAudioDevicePropertyBufferSizeRange,
&dataSize, &bufferRange);
if (err != noErr) {
- sprintf( message, "RtAudio: OSX error getting buffer size range for device (%s).",
- devices[device].name );
+ sprintf( message_, "RtApiCore: OS-X error getting buffer size range for device (%s).",
+ devices_[device].name.c_str() );
error(RtError::DEBUG_WARNING);
return FAILURE;
}
kAudioDevicePropertyBufferSize,
dataSize, &theSize);
if (err != noErr) {
- sprintf( message, "RtAudio: OSX error setting the buffer size for device (%s).",
- devices[device].name );
+ sprintf( message_, "RtApiCore: OS-X error setting the buffer size for device (%s).",
+ devices_[device].name.c_str() );
error(RtError::DEBUG_WARNING);
return FAILURE;
}
// If attempting to setup a duplex stream, the bufferSize parameter
// MUST be the same in both directions!
*bufferSize = bufferBytes / ( deviceChannels * formatBytes(RTAUDIO_FLOAT32) );
- if ( stream->mode == OUTPUT && mode == INPUT && *bufferSize != stream->bufferSize ) {
- sprintf( message, "RtAudio: OSX error setting buffer size for duplex stream on device (%s).",
- devices[device].name );
+ if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+ sprintf( message_, "RtApiCore: OS-X error setting buffer size for duplex stream on device (%s).",
+ devices_[device].name.c_str() );
error(RtError::DEBUG_WARNING);
return FAILURE;
}
- stream->bufferSize = *bufferSize;
- stream->nBuffers = 1;
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 1;
// Set the stream format description. Do for each channel in mono mode.
AudioStreamBasicDescription description;
dataSize = sizeof( AudioStreamBasicDescription );
- if ( stream->deInterleave[mode] ) nStreams = channels;
+ if ( stream_.deInterleave[mode] ) nStreams = channels;
else nStreams = 1;
for ( unsigned int i=0; i<nStreams; i++, iChannel++ ) {
kAudioDevicePropertyStreamFormat,
&dataSize, &description );
if (err != noErr) {
- sprintf( message, "RtAudio: OSX error getting stream format for device (%s).", devices[device].name );
+ sprintf( message_, "RtApiCore: OS-X error getting stream format for device (%s).",
+ devices_[device].name.c_str() );
error(RtError::DEBUG_WARNING);
return FAILURE;
}
kAudioDevicePropertyStreamFormat,
dataSize, &description );
if (err != noErr) {
- sprintf( message, "RtAudio: OSX error setting sample rate or data format for device (%s).", devices[device].name );
+ sprintf( message_, "RtApiCore: OS-X error setting sample rate or data format for device (%s).",
+ devices_[device].name.c_str() );
error(RtError::DEBUG_WARNING);
return FAILURE;
}
}
- // Check whether we need byte-swapping (assuming OS X host is big-endian).
+ // Check whether we need byte-swapping (assuming OS-X host is big-endian).
iChannel -= nStreams;
err = AudioDeviceGetProperty( id, iChannel, isInput,
kAudioDevicePropertyStreamFormat,
&dataSize, &description );
if (err != noErr) {
- sprintf( message, "RtAudio: OSX error getting stream format for device (%s).", devices[device].name );
+ sprintf( message_, "RtApiCore: OS-X error getting stream format for device (%s).", devices_[device].name.c_str() );
error(RtError::DEBUG_WARNING);
return FAILURE;
}
- stream->doByteSwap[mode] = false;
+ stream_.doByteSwap[mode] = false;
if ( !description.mFormatFlags & kLinearPCMFormatFlagIsBigEndian )
- stream->doByteSwap[mode] = true;
+ stream_.doByteSwap[mode] = true;
// From the CoreAudio documentation, PCM data must be supplied as
// 32-bit floats.
- stream->userFormat = format;
- stream->deviceFormat[mode] = RTAUDIO_FLOAT32;
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
- if ( stream->deInterleave[mode] )
- stream->nDeviceChannels[mode] = channels;
+ if ( stream_.deInterleave[mode] ) // mono mode
+ stream_.nDeviceChannels[mode] = channels;
+ else
+ stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
+ stream_.nUserChannels[mode] = channels;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if (stream_.userFormat != stream_.deviceFormat[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.nUserChannels[mode] > 1 && stream_.deInterleave[mode])
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate our CoreHandle structure for the stream.
+ CoreHandle *handle;
+ if ( stream_.apiHandle == 0 ) {
+ handle = (CoreHandle *) calloc(1, sizeof(CoreHandle));
+ if ( handle == NULL ) {
+ sprintf(message_, "RtApiCore: OS-X error allocating coreHandle memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
+ handle->index[0] = 0;
+ handle->index[1] = 0;
+ if ( pthread_cond_init(&handle->condition, NULL) ) {
+ sprintf(message_, "RtApiCore: error initializing pthread condition variable (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
+ stream_.apiHandle = (void *) handle;
+ }
else
- stream->nDeviceChannels[mode] = description.mChannelsPerFrame;
- stream->nUserChannels[mode] = channels;
-
- // Set handle and flags for buffer conversion.
- stream->handle[mode] = iStream;
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] > 1 && stream->deInterleave[mode])
- stream->doConvertBuffer[mode] = true;
+ handle = (CoreHandle *) stream_.apiHandle;
+ handle->index[mode] = iStream;
// Allocate necessary internal buffers.
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
+ if ( stream_.nUserChannels[0] != stream_.nUserChannels[1] ) {
long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
+ if (stream_.nUserChannels[0] >= stream_.nUserChannels[1])
+ buffer_bytes = stream_.nUserChannels[0];
else
- buffer_bytes = stream->nUserChannels[1];
-
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL)
- goto memory_error;
+ buffer_bytes = stream_.nUserChannels[1];
+
+ buffer_bytes *= *bufferSize * formatBytes(stream_.userFormat);
+ if (stream_.userBuffer) free(stream_.userBuffer);
+ stream_.userBuffer = (char *) calloc(buffer_bytes, 1);
+ if (stream_.userBuffer == NULL) {
+ sprintf(message_, "RtApiCore: OS-X error allocating user buffer memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
}
- if ( stream->deInterleave[mode] ) {
+ if ( stream_.deInterleave[mode] ) {
long buffer_bytes;
bool makeBuffer = true;
if ( mode == OUTPUT )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
+ buffer_bytes = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
else { // mode == INPUT
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == OUTPUT && stream->deviceBuffer ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
+ buffer_bytes = stream_.nDeviceChannels[1] * formatBytes(stream_.deviceFormat[1]);
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ long bytes_out = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
if ( buffer_bytes < bytes_out ) makeBuffer = false;
}
}
if ( makeBuffer ) {
buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL)
- goto memory_error;
+ if (stream_.deviceBuffer) free(stream_.deviceBuffer);
+ stream_.deviceBuffer = (char *) calloc(buffer_bytes, 1);
+ if (stream_.deviceBuffer == NULL) {
+ sprintf(message_, "RtApiCore: error allocating device buffer memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
- // If not de-interleaving, we point stream->deviceBuffer to the
+ // If not de-interleaving, we point stream_.deviceBuffer to the
// OS X supplied device buffer before doing any necessary data
// conversions. This presents a problem if we have a duplex
// stream using one device which needs de-interleaving and
// another device which doesn't. So, save a pointer to our own
- // device buffer in the CALLBACK_INFO structure.
- stream->callbackInfo.buffers = stream->deviceBuffer;
+ // device buffer in the CallbackInfo structure.
+ handle->deviceBuffer = stream_.deviceBuffer;
}
}
- stream->sampleRate = sampleRate;
- stream->device[mode] = device;
- stream->state = STREAM_STOPPED;
- stream->callbackInfo.object = (void *) this;
- stream->callbackInfo.waitTime = (unsigned long) (200000.0 * stream->bufferSize / stream->sampleRate);
- stream->callbackInfo.device[mode] = id;
- if ( stream->mode == OUTPUT && mode == INPUT && stream->device[0] == device )
+ stream_.sampleRate = sampleRate;
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ stream_.callbackInfo.object = (void *) this;
+
+ if ( stream_.mode == OUTPUT && mode == INPUT && stream_.device[0] == device )
// Only one callback procedure per device.
- stream->mode = DUPLEX;
+ stream_.mode = DUPLEX;
else {
- err = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream->callbackInfo );
+ err = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
if (err != noErr) {
- sprintf( message, "RtAudio: OSX error setting callback for device (%s).", devices[device].name );
+ sprintf( message_, "RtApiCore: OS-X error setting callback for device (%s).", devices_[device].name.c_str() );
error(RtError::DEBUG_WARNING);
return FAILURE;
}
- if ( stream->mode == OUTPUT && mode == INPUT )
- stream->mode = DUPLEX;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ stream_.mode = DUPLEX;
else
- stream->mode = mode;
+ stream_.mode = mode;
}
- // If we wanted to use property listeners, they would be setup here.
+ // Setup the device property listener for over/underload.
+ err = AudioDeviceAddPropertyListener( id, iChannel, isInput,
+ kAudioDeviceProcessorOverload,
+ deviceListener, (void *) handle );
return SUCCESS;
- memory_error:
- if (stream->userBuffer) {
- free(stream->userBuffer);
- stream->userBuffer = 0;
+ error:
+ if ( handle ) {
+ pthread_cond_destroy(&handle->condition);
+ free(handle);
+ stream_.apiHandle = 0;
}
- sprintf(message, "RtAudio: OSX error allocating buffer memory (%s).", devices[device].name);
- error(RtError::WARNING);
- return FAILURE;
-}
-
-void RtAudio :: cancelStreamCallback(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- if (stream->callbackInfo.usingCallback) {
-
- if (stream->state == STREAM_RUNNING)
- stopStream( streamId );
-
- MUTEX_LOCK(&stream->mutex);
-
- stream->callbackInfo.usingCallback = false;
- stream->callbackInfo.userData = NULL;
- stream->state = STREAM_STOPPED;
- stream->callbackInfo.callback = NULL;
- MUTEX_UNLOCK(&stream->mutex);
+ if (stream_.userBuffer) {
+ free(stream_.userBuffer);
+ stream_.userBuffer = 0;
}
+
+ error(RtError::WARNING);
+ return FAILURE;
}
-void RtAudio :: closeStream(int streamId)
+void RtApiCore :: closeStream()
{
// We don't want an exception to be thrown here because this
// function is called by our class destructor. So, do our own
- // streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
+ // stream check.
+ if ( stream_.mode == UNINITIALIZED ) {
+ sprintf(message_, "RtApiCore::closeStream(): no open stream to close!");
error(RtError::WARNING);
return;
}
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
-
- AudioDeviceID id;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- id = devices[stream->device[0]].id[0];
- if (stream->state == STREAM_RUNNING)
+ AudioDeviceID id = *( (AudioDeviceID *) devices_[stream_.device[0]].apiDeviceId );
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ if (stream_.state == STREAM_RUNNING)
AudioDeviceStop( id, callbackHandler );
AudioDeviceRemoveIOProc( id, callbackHandler );
}
- if (stream->mode == INPUT || ( stream->mode == DUPLEX && stream->device[0] != stream->device[1]) ) {
- id = devices[stream->device[1]].id[0];
- if (stream->state == STREAM_RUNNING)
+ id = *( (AudioDeviceID *) devices_[stream_.device[1]].apiDeviceId );
+ if (stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1]) ) {
+ if (stream_.state == STREAM_RUNNING)
AudioDeviceStop( id, callbackHandler );
AudioDeviceRemoveIOProc( id, callbackHandler );
}
- pthread_mutex_destroy(&stream->mutex);
+ if (stream_.userBuffer) {
+ free(stream_.userBuffer);
+ stream_.userBuffer = 0;
+ }
+
+ if ( stream_.deInterleave[0] || stream_.deInterleave[1] ) {
+ free(stream_.deviceBuffer);
+ stream_.deviceBuffer = 0;
+ }
- if (stream->userBuffer)
- free(stream->userBuffer);
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- if ( stream->deInterleave[0] || stream->deInterleave[1] )
- free(stream->callbackInfo.buffers);
+ // Destroy pthread condition variable and free the CoreHandle structure.
+ if ( handle ) {
+ pthread_cond_destroy(&handle->condition);
+ free( handle );
+ stream_.apiHandle = 0;
+ }
- free(stream);
- streams.erase(streamId);
+ stream_.mode = UNINITIALIZED;
}
-void RtAudio :: startStream(int streamId)
+void RtApiCore :: startStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+ if (stream_.state == STREAM_RUNNING) return;
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_RUNNING)
- goto unlock;
+ MUTEX_LOCK(&stream_.mutex);
OSStatus err;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ AudioDeviceID id;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
- err = AudioDeviceStart(devices[stream->device[0]].id[0], callbackHandler);
+ id = *( (AudioDeviceID *) devices_[stream_.device[0]].apiDeviceId );
+ err = AudioDeviceStart(id, callbackHandler);
if (err != noErr) {
- sprintf(message, "RtAudio: OSX error starting callback procedure on device (%s).",
- devices[stream->device[0]].name);
- MUTEX_UNLOCK(&stream->mutex);
+ sprintf(message_, "RtApiCore: OS-X error starting callback procedure on device (%s).",
+ devices_[stream_.device[0]].name.c_str());
+ MUTEX_UNLOCK(&stream_.mutex);
error(RtError::DRIVER_ERROR);
}
}
- if (stream->mode == INPUT || ( stream->mode == DUPLEX && stream->device[0] != stream->device[1]) ) {
+ if (stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1]) ) {
- err = AudioDeviceStart(devices[stream->device[1]].id[0], callbackHandler);
+ id = *( (AudioDeviceID *) devices_[stream_.device[1]].apiDeviceId );
+ err = AudioDeviceStart(id, callbackHandler);
if (err != noErr) {
- sprintf(message, "RtAudio: OSX error starting input callback procedure on device (%s).",
- devices[stream->device[0]].name);
- MUTEX_UNLOCK(&stream->mutex);
+ sprintf(message_, "RtApiCore: OS-X error starting input callback procedure on device (%s).",
+ devices_[stream_.device[0]].name.c_str());
+ MUTEX_UNLOCK(&stream_.mutex);
error(RtError::DRIVER_ERROR);
}
}
- stream->callbackInfo.streamId = streamId;
- stream->state = STREAM_RUNNING;
- stream->callbackInfo.blockTick = true;
- stream->callbackInfo.stopStream = false;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ handle->stopStream = false;
+ stream_.state = STREAM_RUNNING;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
}
-void RtAudio :: stopStream(int streamId)
+void RtApiCore :: stopStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+ if (stream_.state == STREAM_STOPPED) return;
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ // Change the state before the lock to improve shutdown response
+ // when using a callback.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK(&stream_.mutex);
OSStatus err;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ AudioDeviceID id;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
- err = AudioDeviceStop(devices[stream->device[0]].id[0], callbackHandler);
+ id = *( (AudioDeviceID *) devices_[stream_.device[0]].apiDeviceId );
+ err = AudioDeviceStop(id, callbackHandler);
if (err != noErr) {
- sprintf(message, "RtAudio: OSX error stopping callback procedure on device (%s).",
- devices[stream->device[0]].name);
- MUTEX_UNLOCK(&stream->mutex);
+ sprintf(message_, "RtApiCore: OS-X error stopping callback procedure on device (%s).",
+ devices_[stream_.device[0]].name.c_str());
+ MUTEX_UNLOCK(&stream_.mutex);
error(RtError::DRIVER_ERROR);
}
}
- if (stream->mode == INPUT || ( stream->mode == DUPLEX && stream->device[0] != stream->device[1]) ) {
+ if (stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1]) ) {
- err = AudioDeviceStop(devices[stream->device[1]].id[0], callbackHandler);
+ id = *( (AudioDeviceID *) devices_[stream_.device[1]].apiDeviceId );
+ err = AudioDeviceStop(id, callbackHandler);
if (err != noErr) {
- sprintf(message, "RtAudio: OSX error stopping input callback procedure on device (%s).",
- devices[stream->device[0]].name);
- MUTEX_UNLOCK(&stream->mutex);
+ sprintf(message_, "RtApiCore: OS-X error stopping input callback procedure on device (%s).",
+ devices_[stream_.device[0]].name.c_str());
+ MUTEX_UNLOCK(&stream_.mutex);
error(RtError::DRIVER_ERROR);
}
}
- stream->state = STREAM_STOPPED;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-void RtAudio :: abortStream(int streamId)
-{
- stopStream( streamId );
+ MUTEX_UNLOCK(&stream_.mutex);
}
-// I don't know how this function can be implemented.
-int RtAudio :: streamWillBlock(int streamId)
+void RtApiCore :: abortStream()
{
- sprintf(message, "RtAudio: streamWillBlock() cannot be implemented for OS X.");
- error(RtError::WARNING);
- return 0;
+ stopStream();
}
-void RtAudio :: tickStream(int streamId)
+void RtApiCore :: tickStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
- if (stream->state == STREAM_STOPPED)
- return;
+ if (stream_.state == STREAM_STOPPED) return;
- if (stream->callbackInfo.usingCallback) {
- sprintf(message, "RtAudio: tickStream() should not be used when a callback function is set!");
+ if (stream_.callbackInfo.usingCallback) {
+ sprintf(message_, "RtApiCore: tickStream() should not be used when a callback function is set!");
error(RtError::WARNING);
return;
}
- // Block waiting here until the user data is processed in callbackEvent().
- while ( stream->callbackInfo.blockTick )
- usleep(stream->callbackInfo.waitTime);
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK(&stream_.mutex);
- stream->callbackInfo.blockTick = true;
+ pthread_cond_wait(&handle->condition, &stream_.mutex);
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
}
-void RtAudio :: callbackEvent( int streamId, DEVICE_ID deviceId, void *inData, void *outData )
+void RtApiCore :: callbackEvent( AudioDeviceID deviceId, void *inData, void *outData )
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+
+ if (stream_.state == STREAM_STOPPED) return;
- CALLBACK_INFO *info;
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
AudioBufferList *inBufferList = (AudioBufferList *) inData;
AudioBufferList *outBufferList = (AudioBufferList *) outData;
- if (stream->state == STREAM_STOPPED) return;
-
- info = (CALLBACK_INFO *) &stream->callbackInfo;
- if ( !info->usingCallback ) {
- // Block waiting here until we get new user data in tickStream().
- while ( !info->blockTick )
- usleep(info->waitTime);
- }
- else if ( info->stopStream ) {
+ if ( info->usingCallback && handle->stopStream ) {
// Check if the stream should be stopped (via the previous user
// callback return value). We stop the stream here, rather than
// after the function call, so that output data can first be
// processed.
- this->stopStream(info->streamId);
+ this->stopStream();
return;
}
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK(&stream_.mutex);
// Invoke user callback first, to get fresh output data. Don't
- // invoke the user callback if duplex mode, the input/output devices
- // are different, and this function is called for the input device.
- if ( info->usingCallback && (stream->mode != DUPLEX || deviceId == info->device[0] ) ) {
- RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) info->callback;
- info->stopStream = callback(stream->userBuffer, stream->bufferSize, info->userData);
+ // invoke the user callback if duplex mode AND the input/output devices
+ // are different AND this function is called for the input device.
+ AudioDeviceID id = *( (AudioDeviceID *) devices_[stream_.device[0]].apiDeviceId );
+ if ( info->usingCallback && (stream_.mode != DUPLEX || deviceId == id ) ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ handle->stopStream = callback(stream_.userBuffer, stream_.bufferSize, info->userData);
+ if ( handle->xrun == true ) {
+ handle->xrun = false;
+ MUTEX_UNLOCK(&stream_.mutex);
+ return;
+ }
}
- if ( stream->mode == OUTPUT || ( stream->mode == DUPLEX && deviceId == info->device[0] ) ) {
+ if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == id ) ) {
- if (stream->doConvertBuffer[0]) {
+ if (stream_.doConvertBuffer[0]) {
- if ( !stream->deInterleave[0] )
- stream->deviceBuffer = (char *) outBufferList->mBuffers[stream->handle[0]].mData;
+ if ( !stream_.deInterleave[0] )
+ stream_.deviceBuffer = (char *) outBufferList->mBuffers[handle->index[0]].mData;
else
- stream->deviceBuffer = (char *) stream->callbackInfo.buffers;
-
- convertStreamBuffer(stream, OUTPUT);
- if ( stream->doByteSwap[0] )
- byteSwapBuffer(stream->deviceBuffer,
- stream->bufferSize * stream->nDeviceChannels[0],
- stream->deviceFormat[0]);
-
- if ( stream->deInterleave[0] ) {
- int bufferBytes = outBufferList->mBuffers[stream->handle[0]].mDataByteSize;
- for ( int i=0; i<stream->nDeviceChannels[0]; i++ ) {
- memcpy(outBufferList->mBuffers[stream->handle[0]+i].mData,
- &stream->deviceBuffer[i*bufferBytes], bufferBytes );
+ stream_.deviceBuffer = handle->deviceBuffer;
+
+ convertStreamBuffer(OUTPUT);
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer(stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[0],
+ stream_.deviceFormat[0]);
+
+ if ( stream_.deInterleave[0] ) {
+ int bufferBytes = outBufferList->mBuffers[handle->index[0]].mDataByteSize;
+ for ( int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+ memcpy(outBufferList->mBuffers[handle->index[0]+i].mData,
+ &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
}
}
}
else {
- if (stream->doByteSwap[0])
- byteSwapBuffer(stream->userBuffer,
- stream->bufferSize * stream->nUserChannels[0],
- stream->userFormat);
+ if (stream_.doByteSwap[0])
+ byteSwapBuffer(stream_.userBuffer,
+ stream_.bufferSize * stream_.nUserChannels[0],
+ stream_.userFormat);
- memcpy(outBufferList->mBuffers[stream->handle[0]].mData,
- stream->userBuffer,
- outBufferList->mBuffers[stream->handle[0]].mDataByteSize );
+ memcpy(outBufferList->mBuffers[handle->index[0]].mData,
+ stream_.userBuffer,
+ outBufferList->mBuffers[handle->index[0]].mDataByteSize );
}
}
- if ( stream->mode == INPUT || ( stream->mode == DUPLEX && deviceId == info->device[1] ) ) {
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == id ) ) {
- if (stream->doConvertBuffer[1]) {
+ if (stream_.doConvertBuffer[1]) {
- if ( stream->deInterleave[1] ) {
- stream->deviceBuffer = (char *) stream->callbackInfo.buffers;
- int bufferBytes = inBufferList->mBuffers[stream->handle[1]].mDataByteSize;
- for ( int i=0; i<stream->nDeviceChannels[1]; i++ ) {
- memcpy(&stream->deviceBuffer[i*bufferBytes],
- inBufferList->mBuffers[stream->handle[1]+i].mData, bufferBytes );
+ if ( stream_.deInterleave[1] ) {
+ stream_.deviceBuffer = (char *) handle->deviceBuffer;
+ int bufferBytes = inBufferList->mBuffers[handle->index[1]].mDataByteSize;
+ for ( int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
+ memcpy(&stream_.deviceBuffer[i*bufferBytes],
+ inBufferList->mBuffers[handle->index[1]+i].mData, bufferBytes );
}
}
else
- stream->deviceBuffer = (char *) inBufferList->mBuffers[stream->handle[1]].mData;
-
- if ( stream->doByteSwap[1] )
- byteSwapBuffer(stream->deviceBuffer,
- stream->bufferSize * stream->nDeviceChannels[1],
- stream->deviceFormat[1]);
- convertStreamBuffer(stream, INPUT);
-
- }
- else {
- memcpy(stream->userBuffer,
- inBufferList->mBuffers[stream->handle[1]].mData,
- inBufferList->mBuffers[stream->handle[1]].mDataByteSize );
-
- if (stream->doByteSwap[1])
- byteSwapBuffer(stream->userBuffer,
- stream->bufferSize * stream->nUserChannels[1],
- stream->userFormat);
- }
- }
-
- if ( !info->usingCallback && (stream->mode != DUPLEX || deviceId == info->device[1] ) )
- info->blockTick = false;
-
- MUTEX_UNLOCK(&stream->mutex);
-
-}
-
-void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- stream->callbackInfo.callback = (void *) callback;
- stream->callbackInfo.userData = userData;
- stream->callbackInfo.usingCallback = true;
-}
-
-//******************** End of __MACOSX_CORE__ *********************//
-
-#elif defined(__LINUX_ALSA__)
-
-#define MAX_DEVICES 16
-
-void RtAudio :: initialize(void)
-{
- int card, result, device;
- char name[32];
- const char *cardId;
- char deviceNames[MAX_DEVICES][32];
- snd_ctl_t *handle;
- snd_ctl_card_info_t *info;
- snd_ctl_card_info_alloca(&info);
-
- // Count cards and devices
- nDevices = 0;
- card = -1;
- snd_card_next(&card);
- while ( card >= 0 ) {
- sprintf(name, "hw:%d", card);
- result = snd_ctl_open(&handle, name, 0);
- if (result < 0) {
- sprintf(message, "RtAudio: ALSA control open (%i): %s.", card, snd_strerror(result));
- error(RtError::DEBUG_WARNING);
- goto next_card;
- }
- result = snd_ctl_card_info(handle, info);
- if (result < 0) {
- sprintf(message, "RtAudio: ALSA control hardware info (%i): %s.", card, snd_strerror(result));
- error(RtError::DEBUG_WARNING);
- goto next_card;
- }
- cardId = snd_ctl_card_info_get_id(info);
- device = -1;
- while (1) {
- result = snd_ctl_pcm_next_device(handle, &device);
- if (result < 0) {
- sprintf(message, "RtAudio: ALSA control next device (%i): %s.", card, snd_strerror(result));
- error(RtError::DEBUG_WARNING);
- break;
- }
- if (device < 0)
- break;
- if ( strlen(cardId) )
- sprintf( deviceNames[nDevices++], "hw:%s,%d", cardId, device );
- else
- sprintf( deviceNames[nDevices++], "hw:%d,%d", card, device );
- if ( nDevices > MAX_DEVICES ) break;
- }
- if ( nDevices > MAX_DEVICES ) break;
- next_card:
- snd_ctl_close(handle);
- snd_card_next(&card);
- }
-
- if (nDevices == 0) return;
-
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
-
- // Write device ascii identifiers to device structures and then
- // probe the device capabilities.
- for (int i=0; i<nDevices; i++) {
- strncpy(devices[i].name, deviceNames[i], 32);
- //probeDeviceInfo(&devices[i]);
- }
-}
-
-int RtAudio :: getDefaultInputDevice(void)
-{
- // No ALSA API functions for default devices.
- return 0;
-}
-
-int RtAudio :: getDefaultOutputDevice(void)
-{
- // No ALSA API functions for default devices.
- return 0;
-}
-
-void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info)
-{
- int err;
- int open_mode = SND_PCM_ASYNC;
- snd_pcm_t *handle;
- snd_ctl_t *chandle;
- snd_pcm_stream_t stream;
- snd_pcm_info_t *pcminfo;
- snd_pcm_info_alloca(&pcminfo);
- snd_pcm_hw_params_t *params;
- snd_pcm_hw_params_alloca(¶ms);
- char name[32];
- char *card;
-
- // Open the control interface for this card.
- strncpy( name, info->name, 32 );
- card = strtok(name, ",");
- err = snd_ctl_open(&chandle, card, 0);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA control open (%s): %s.", card, snd_strerror(err));
- error(RtError::DEBUG_WARNING);
- return;
- }
- unsigned int dev = (unsigned int) atoi( strtok(NULL, ",") );
-
- // First try for playback
- stream = SND_PCM_STREAM_PLAYBACK;
- snd_pcm_info_set_device(pcminfo, dev);
- snd_pcm_info_set_subdevice(pcminfo, 0);
- snd_pcm_info_set_stream(pcminfo, stream);
-
- if ((err = snd_ctl_pcm_info(chandle, pcminfo)) < 0) {
- if (err == -ENOENT) {
- sprintf(message, "RtAudio: ALSA pcm device (%s) doesn't handle output!", info->name);
- error(RtError::DEBUG_WARNING);
- }
- else {
- sprintf(message, "RtAudio: ALSA snd_ctl_pcm_info error for device (%s) output: %s",
- info->name, snd_strerror(err));
- error(RtError::DEBUG_WARNING);
- }
- goto capture_probe;
- }
-
- err = snd_pcm_open(&handle, info->name, stream, open_mode | SND_PCM_NONBLOCK );
- if (err < 0) {
- if ( err == EBUSY )
- sprintf(message, "RtAudio: ALSA pcm playback device (%s) is busy: %s.",
- info->name, snd_strerror(err));
- else
- sprintf(message, "RtAudio: ALSA pcm playback open (%s) error: %s.",
- info->name, snd_strerror(err));
- error(RtError::DEBUG_WARNING);
- goto capture_probe;
- }
-
- // We have an open device ... allocate the parameter structure.
- err = snd_pcm_hw_params_any(handle, params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA hardware probe error (%s): %s.",
- info->name, snd_strerror(err));
- error(RtError::WARNING);
- goto capture_probe;
- }
-
- // Get output channel information.
- info->minOutputChannels = snd_pcm_hw_params_get_channels_min(params);
- info->maxOutputChannels = snd_pcm_hw_params_get_channels_max(params);
-
- snd_pcm_close(handle);
-
- capture_probe:
- // Now try for capture
- stream = SND_PCM_STREAM_CAPTURE;
- snd_pcm_info_set_stream(pcminfo, stream);
-
- err = snd_ctl_pcm_info(chandle, pcminfo);
- snd_ctl_close(chandle);
- if ( err < 0 ) {
- if (err == -ENOENT) {
- sprintf(message, "RtAudio: ALSA pcm device (%s) doesn't handle input!", info->name);
- error(RtError::DEBUG_WARNING);
- }
- else {
- sprintf(message, "RtAudio: ALSA snd_ctl_pcm_info error for device (%s) input: %s",
- info->name, snd_strerror(err));
- error(RtError::DEBUG_WARNING);
- }
- if (info->maxOutputChannels == 0)
- // didn't open for playback either ... device invalid
- return;
- goto probe_parameters;
- }
-
- err = snd_pcm_open(&handle, info->name, stream, open_mode | SND_PCM_NONBLOCK);
- if (err < 0) {
- if ( err == EBUSY )
- sprintf(message, "RtAudio: ALSA pcm capture device (%s) is busy: %s.",
- info->name, snd_strerror(err));
- else
- sprintf(message, "RtAudio: ALSA pcm capture open (%s) error: %s.",
- info->name, snd_strerror(err));
- error(RtError::DEBUG_WARNING);
- if (info->maxOutputChannels == 0)
- // didn't open for playback either ... device invalid
- return;
- goto probe_parameters;
- }
-
- // We have an open capture device ... allocate the parameter structure.
- err = snd_pcm_hw_params_any(handle, params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA hardware probe error (%s): %s.",
- info->name, snd_strerror(err));
- error(RtError::WARNING);
- if (info->maxOutputChannels > 0)
- goto probe_parameters;
- else
- return;
- }
-
- // Get input channel information.
- info->minInputChannels = snd_pcm_hw_params_get_channels_min(params);
- info->maxInputChannels = snd_pcm_hw_params_get_channels_max(params);
-
- snd_pcm_close(handle);
-
- // If device opens for both playback and capture, we determine the channels.
- if (info->maxOutputChannels == 0 || info->maxInputChannels == 0)
- goto probe_parameters;
-
- info->hasDuplexSupport = true;
- info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ?
- info->maxInputChannels : info->maxOutputChannels;
- info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ?
- info->minInputChannels : info->minOutputChannels;
-
- probe_parameters:
- // At this point, we just need to figure out the supported data
- // formats and sample rates. We'll proceed by opening the device in
- // the direction with the maximum number of channels, or playback if
- // they are equal. This might limit our sample rate options, but so
- // be it.
-
- if (info->maxOutputChannels >= info->maxInputChannels)
- stream = SND_PCM_STREAM_PLAYBACK;
- else
- stream = SND_PCM_STREAM_CAPTURE;
-
- err = snd_pcm_open(&handle, info->name, stream, open_mode);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA pcm (%s) won't reopen during probe: %s.",
- info->name, snd_strerror(err));
- error(RtError::WARNING);
- return;
- }
-
- // We have an open device ... allocate the parameter structure.
- err = snd_pcm_hw_params_any(handle, params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA hardware reopen probe error (%s): %s.",
- info->name, snd_strerror(err));
- error(RtError::WARNING);
- return;
- }
-
- // Test a non-standard sample rate to see if continuous rate is supported.
- int dir = 0;
- if (snd_pcm_hw_params_test_rate(handle, params, 35500, dir) == 0) {
- // It appears that continuous sample rate support is available.
- info->nSampleRates = -1;
- info->sampleRates[0] = snd_pcm_hw_params_get_rate_min(params, &dir);
- info->sampleRates[1] = snd_pcm_hw_params_get_rate_max(params, &dir);
- }
- else {
- // No continuous rate support ... test our discrete set of sample rate values.
- info->nSampleRates = 0;
- for (int i=0; i<MAX_SAMPLE_RATES; i++) {
- if (snd_pcm_hw_params_test_rate(handle, params, SAMPLE_RATES[i], dir) == 0) {
- info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i];
- info->nSampleRates++;
- }
+ stream_.deviceBuffer = (char *) inBufferList->mBuffers[handle->index[1]].mData;
+
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer(stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[1],
+ stream_.deviceFormat[1]);
+ convertStreamBuffer(INPUT);
+
}
- if (info->nSampleRates == 0) {
- snd_pcm_close(handle);
- return;
+ else {
+ memcpy(stream_.userBuffer,
+ inBufferList->mBuffers[handle->index[1]].mData,
+ inBufferList->mBuffers[handle->index[1]].mDataByteSize );
+
+ if (stream_.doByteSwap[1])
+ byteSwapBuffer(stream_.userBuffer,
+ stream_.bufferSize * stream_.nUserChannels[1],
+ stream_.userFormat);
}
}
- // Probe the supported data formats ... we don't care about endian-ness just yet
- snd_pcm_format_t format;
- info->nativeFormats = 0;
- format = SND_PCM_FORMAT_S8;
- if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
- info->nativeFormats |= RTAUDIO_SINT8;
- format = SND_PCM_FORMAT_S16;
- if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
- info->nativeFormats |= RTAUDIO_SINT16;
- format = SND_PCM_FORMAT_S24;
- if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
- info->nativeFormats |= RTAUDIO_SINT24;
- format = SND_PCM_FORMAT_S32;
- if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
- info->nativeFormats |= RTAUDIO_SINT32;
- format = SND_PCM_FORMAT_FLOAT;
- if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
- info->nativeFormats |= RTAUDIO_FLOAT32;
- format = SND_PCM_FORMAT_FLOAT64;
- if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
- info->nativeFormats |= RTAUDIO_FLOAT64;
+ if ( !info->usingCallback && (stream_.mode != DUPLEX || deviceId == id ) )
+ pthread_cond_signal(&handle->condition);
- // Check that we have at least one supported format
- if (info->nativeFormats == 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA PCM device (%s) data format not supported by RtAudio.",
- info->name);
+ MUTEX_UNLOCK(&stream_.mutex);
+}
+
+void RtApiCore :: setStreamCallback(RtAudioCallback callback, void *userData)
+{
+ verifyStream();
+
+ if ( stream_.callbackInfo.usingCallback ) {
+ sprintf(message_, "RtApiCore: A callback is already set for this stream!");
error(RtError::WARNING);
return;
}
- // That's all ... close the device and return
- snd_pcm_close(handle);
- info->probed = true;
- return;
+ stream_.callbackInfo.callback = (void *) callback;
+ stream_.callbackInfo.userData = userData;
+ stream_.callbackInfo.usingCallback = true;
}
-bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
- STREAM_MODE mode, int channels,
- int sampleRate, RTAUDIO_FORMAT format,
- int *bufferSize, int numberOfBuffers)
+void RtApiCore :: cancelStreamCallback()
{
-#if defined(__RTAUDIO_DEBUG__)
- snd_output_t *out;
- snd_output_stdio_attach(&out, stderr, 0);
-#endif
+ verifyStream();
- // I'm not using the "plug" interface ... too much inconsistent behavior.
- const char *name = devices[device].name;
+ if (stream_.callbackInfo.usingCallback) {
- snd_pcm_stream_t alsa_stream;
- if (mode == OUTPUT)
- alsa_stream = SND_PCM_STREAM_PLAYBACK;
- else
- alsa_stream = SND_PCM_STREAM_CAPTURE;
+ if (stream_.state == STREAM_RUNNING)
+ stopStream();
- int err;
- snd_pcm_t *handle;
- int alsa_open_mode = SND_PCM_ASYNC;
- err = snd_pcm_open(&handle, name, alsa_stream, alsa_open_mode);
- if (err < 0) {
- sprintf(message,"RtAudio: ALSA pcm device (%s) won't open: %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
+ MUTEX_LOCK(&stream_.mutex);
- // Fill the parameter structure.
- snd_pcm_hw_params_t *hw_params;
- snd_pcm_hw_params_alloca(&hw_params);
- err = snd_pcm_hw_params_any(handle, hw_params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error getting parameter handle (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ stream_.callbackInfo.usingCallback = false;
+ stream_.callbackInfo.userData = NULL;
+ stream_.state = STREAM_STOPPED;
+ stream_.callbackInfo.callback = NULL;
+
+ MUTEX_UNLOCK(&stream_.mutex);
}
+}
-#if defined(__RTAUDIO_DEBUG__)
- fprintf(stderr, "\nRtAudio: ALSA dump hardware params just after device open:\n\n");
- snd_pcm_hw_params_dump(hw_params, out);
+
+//******************** End of __MACOSX_CORE__ *********************//
#endif
+#if defined(__LINUX_JACK__)
- // Set access ... try interleaved access first, then non-interleaved
- if ( !snd_pcm_hw_params_test_access( handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED) ) {
- err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
- }
- else if ( !snd_pcm_hw_params_test_access( handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED) ) {
- err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED);
- stream->deInterleave[mode] = true;
- }
- else {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA device (%s) access not supported by RtAudio.", name);
- error(RtError::WARNING);
- return FAILURE;
- }
+// JACK is a low-latency audio server, written primarily for the
+// GNU/Linux operating system. It can connect a number of different
+// applications to an audio device, as well as allowing them to share
+// audio between themselves.
+//
+// The JACK server must be running before RtApiJack can be instantiated.
+// RtAudio will report just a single "device", which is the JACK audio
+// server. The JACK server is typically started in a terminal as follows:
+//
+// .jackd -d alsa -d hw:0
+//
+// Many of the parameters normally set for a stream are fixed by the
+// JACK server and can be specified when the JACK server is started.
+// In particular,
+//
+// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
+//
+// specifies a sample rate of 44100 Hz, a buffer size of 512 sample
+// frames, and number of buffers = 4. Once the server is running, it
+// is not possible to override these values. If the values are not
+// specified in the command-line, the JACK server uses default values.
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting access ( (%s): %s.", name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
+#include <jack/jack.h>
+#include <unistd.h>
- // Determine how to set the device format.
- stream->userFormat = format;
- snd_pcm_format_t device_format;
+// A structure to hold various information related to the Jack API
+// implementation.
+struct JackHandle {
+ jack_client_t *client;
+ jack_port_t **ports[2];
+ bool clientOpen;
+ bool stopStream;
+ pthread_cond_t condition;
+
+ JackHandle()
+ :client(0), clientOpen(false), stopStream(false) {}
+};
- if (format == RTAUDIO_SINT8)
- device_format = SND_PCM_FORMAT_S8;
- else if (format == RTAUDIO_SINT16)
- device_format = SND_PCM_FORMAT_S16;
- else if (format == RTAUDIO_SINT24)
- device_format = SND_PCM_FORMAT_S24;
- else if (format == RTAUDIO_SINT32)
- device_format = SND_PCM_FORMAT_S32;
- else if (format == RTAUDIO_FLOAT32)
- device_format = SND_PCM_FORMAT_FLOAT;
- else if (format == RTAUDIO_FLOAT64)
- device_format = SND_PCM_FORMAT_FLOAT64;
+std::string jackmsg;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = format;
- goto set_format;
- }
+static void jackerror (const char *desc)
+{
+ jackmsg.erase();
+ jackmsg.append( desc, strlen(desc)+1 );
+}
- // The user requested format is not natively supported by the device.
- device_format = SND_PCM_FORMAT_FLOAT64;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_FLOAT64;
- goto set_format;
- }
+RtApiJack :: RtApiJack()
+{
+ this->initialize();
- device_format = SND_PCM_FORMAT_FLOAT;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_FLOAT32;
- goto set_format;
+ if (nDevices_ <= 0) {
+ sprintf(message_, "RtApiJack: no Linux Jack server found or connection error (jack: %s)!",
+ jackmsg.c_str());
+ error(RtError::NO_DEVICES_FOUND);
}
+}
- device_format = SND_PCM_FORMAT_S32;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- goto set_format;
- }
+RtApiJack :: ~RtApiJack()
+{
+ if ( stream_.mode != UNINITIALIZED ) closeStream();
+}
- device_format = SND_PCM_FORMAT_S24;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_SINT24;
- goto set_format;
- }
+void RtApiJack :: initialize(void)
+{
+ nDevices_ = 0;
- device_format = SND_PCM_FORMAT_S16;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- goto set_format;
- }
+ // Tell the jack server to call jackerror() when it experiences an
+ // error. This function saves the error message for subsequent
+ // reporting via the normal RtAudio error function.
+ jack_set_error_function( jackerror );
- device_format = SND_PCM_FORMAT_S8;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_SINT8;
- goto set_format;
- }
+ // Look for jack server and try to become a client.
+ jack_client_t *client;
+ if ( (client = jack_client_new( "RtApiJack" )) == 0)
+ return;
- // If we get here, no supported format was found.
- sprintf(message,"RtAudio: ALSA pcm device (%s) data format not supported by RtAudio.", name);
- snd_pcm_close(handle);
- error(RtError::WARNING);
- return FAILURE;
+ RtApiDevice device;
+ // Determine the name of the device.
+ device.name = "Jack Server";
+ devices_.push_back(device);
+ nDevices_++;
- set_format:
- err = snd_pcm_hw_params_set_format(handle, hw_params, device_format);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting format (%s): %s.",
- name, snd_strerror(err));
+ jack_client_close(client);
+}
+
+void RtApiJack :: probeDeviceInfo(RtApiDevice *info)
+{
+ // Look for jack server and try to become a client.
+ jack_client_t *client;
+ if ( (client = jack_client_new( "RtApiJack" )) == 0) {
+ sprintf(message_, "RtApiJack: error connecting to Linux Jack server in probeDeviceInfo() (jack: %s)!",
+ jackmsg.c_str());
error(RtError::WARNING);
- return FAILURE;
+ return;
}
- // Determine whether byte-swaping is necessary.
- stream->doByteSwap[mode] = false;
- if (device_format != SND_PCM_FORMAT_S8) {
- err = snd_pcm_format_cpu_endian(device_format);
- if (err == 0)
- stream->doByteSwap[mode] = true;
- else if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error getting format endian-ness (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
+ // Get the current jack server sample rate.
+ info->sampleRates.clear();
+ info->sampleRates.push_back( jack_get_sample_rate(client) );
+
+ // Count the available ports as device channels. Jack "input ports"
+ // equal RtAudio output channels.
+ const char **ports;
+ char *port;
+ unsigned int nChannels = 0;
+ ports = jack_get_ports( client, NULL, NULL, JackPortIsInput );
+ if ( ports ) {
+ port = (char *) ports[nChannels];
+ while ( port )
+ port = (char *) ports[++nChannels];
+ free( ports );
+ info->maxOutputChannels = nChannels;
+ info->minOutputChannels = 1;
+ }
+
+ // Jack "output ports" equal RtAudio input channels.
+ nChannels = 0;
+ ports = jack_get_ports( client, NULL, NULL, JackPortIsOutput );
+ if ( ports ) {
+ port = (char *) ports[nChannels];
+ while ( port )
+ port = (char *) ports[++nChannels];
+ free( ports );
+ info->maxInputChannels = nChannels;
+ info->minInputChannels = 1;
}
- // Set the sample rate.
- err = snd_pcm_hw_params_set_rate(handle, hw_params, (unsigned int)sampleRate, 0);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting sample rate (%d) on device (%s): %s.",
- sampleRate, name, snd_strerror(err));
+ if (info->maxOutputChannels == 0 && info->maxInputChannels == 0) {
+ jack_client_close(client);
+ sprintf(message_, "RtApiJack: error determining jack input/output channels!");
error(RtError::WARNING);
- return FAILURE;
+ return;
}
- // Determine the number of channels for this device. We support a possible
- // minimum device channel number > than the value requested by the user.
- stream->nUserChannels[mode] = channels;
- int device_channels = snd_pcm_hw_params_get_channels_max(hw_params);
- if (device_channels < channels) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: channels (%d) not supported by device (%s).",
- channels, name);
- error(RtError::WARNING);
- return FAILURE;
+ if (info->maxOutputChannels > 0 && info->maxInputChannels > 0) {
+ info->hasDuplexSupport = true;
+ info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ?
+ info->maxInputChannels : info->maxOutputChannels;
+ info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ?
+ info->minInputChannels : info->minOutputChannels;
}
- device_channels = snd_pcm_hw_params_get_channels_min(hw_params);
- if (device_channels < channels) device_channels = channels;
- stream->nDeviceChannels[mode] = device_channels;
+ // Get the jack data format type. There isn't much documentation
+ // regarding supported data formats in jack. I'm assuming here that
+ // the default type will always be a floating-point type, of length
+ // equal to either 4 or 8 bytes.
+ int sample_size = sizeof( jack_default_audio_sample_t );
+ if ( sample_size == 4 )
+ info->nativeFormats = RTAUDIO_FLOAT32;
+ else if ( sample_size == 8 )
+ info->nativeFormats = RTAUDIO_FLOAT64;
- // Set the device channels.
- err = snd_pcm_hw_params_set_channels(handle, hw_params, device_channels);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting channels (%d) on device (%s): %s.",
- device_channels, name, snd_strerror(err));
+ // Check that we have a supported format
+ if (info->nativeFormats == 0) {
+ jack_client_close(client);
+ sprintf(message_, "RtApiJack: error determining jack server data format!");
error(RtError::WARNING);
- return FAILURE;
+ return;
}
- // Set the buffer number, which in ALSA is referred to as the "period".
- int dir;
- int periods = numberOfBuffers;
- // Even though the hardware might allow 1 buffer, it won't work reliably.
- if (periods < 2) periods = 2;
- err = snd_pcm_hw_params_get_periods_min(hw_params, &dir);
- if (err > periods) periods = err;
- err = snd_pcm_hw_params_get_periods_max(hw_params, &dir);
- if (err < periods) periods = err;
+ jack_client_close(client);
+ info->probed = true;
+}
- err = snd_pcm_hw_params_set_periods(handle, hw_params, periods, 0);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting periods (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+int jackCallbackHandler(jack_nframes_t nframes, void *infoPointer)
+{
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
+ RtApiJack *object = (RtApiJack *) info->object;
+ try {
+ object->callbackEvent( (unsigned long) nframes );
+ }
+ catch (RtError &exception) {
+ fprintf(stderr, "\nRtApiJack: callback handler error (%s)!\n\n", exception.getMessageString());
+ return 0;
}
- // Set the buffer (or period) size.
- err = snd_pcm_hw_params_get_period_size_min(hw_params, &dir);
- if (err > *bufferSize) *bufferSize = err;
+ return 0;
+}
- err = snd_pcm_hw_params_set_period_size(handle, hw_params, *bufferSize, 0);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting period size (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+void jackShutdown(void *infoPointer)
+{
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
+ JackHandle *handle = (JackHandle *) info->apiInfo;
+ handle->clientOpen = false;
+ RtApiJack *object = (RtApiJack *) info->object;
+ try {
+ object->closeStream();
+ }
+ catch (RtError &exception) {
+ fprintf(stderr, "\nRtApiJack: jackShutdown error (%s)!\n\n", exception.getMessageString());
+ return;
}
- // If attempting to setup a duplex stream, the bufferSize parameter
- // MUST be the same in both directions!
- if ( stream->mode == OUTPUT && mode == INPUT && *bufferSize != stream->bufferSize ) {
- sprintf( message, "RtAudio: ALSA error setting buffer size for duplex stream on device (%s).",
- name );
+ fprintf(stderr, "\nRtApiJack: the Jack server is shutting down ... stream stopped and closed!!!\n\n");
+}
+
+int jackXrun( void * )
+{
+ fprintf(stderr, "\nRtApiJack: audio overrun/underrun reported!\n");
+ return 0;
+}
+
+bool RtApiJack :: probeDeviceOpen(int device, StreamMode mode, int channels,
+ int sampleRate, RtAudioFormat format,
+ int *bufferSize, int numberOfBuffers)
+{
+ // Compare the jack server channels to the requested number of channels.
+ if ( (mode == OUTPUT && devices_[device].maxOutputChannels < channels ) ||
+ (mode == INPUT && devices_[device].maxInputChannels < channels ) ) {
+ sprintf(message_, "RtApiJack: the Jack server does not support requested channels!");
error(RtError::DEBUG_WARNING);
return FAILURE;
}
- stream->bufferSize = *bufferSize;
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
- // Install the hardware configuration
- err = snd_pcm_hw_params(handle, hw_params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error installing hardware configuration (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ // Look for jack server and try to become a client (only do once per stream).
+ char label[32];
+ jack_client_t *client = 0;
+ if ( mode == OUTPUT || (mode == INPUT && stream_.mode != OUTPUT) ) {
+ snprintf(label, 32, "RtApiJack");
+ if ( (client = jack_client_new( (const char *) label )) == 0) {
+ sprintf(message_, "RtApiJack: cannot connect to Linux Jack server in probeDeviceOpen() (jack: %s)!",
+ jackmsg.c_str());
+ error(RtError::DEBUG_WARNING);
+ return FAILURE;
+ }
+ }
+ else {
+ // The handle must have been created on an earlier pass.
+ client = handle->client;
}
-#if defined(__RTAUDIO_DEBUG__)
- fprintf(stderr, "\nRtAudio: ALSA dump hardware params after installation:\n\n");
- snd_pcm_hw_params_dump(hw_params, out);
-#endif
-
- /*
- // Install the software configuration
- snd_pcm_sw_params_t *sw_params = NULL;
- snd_pcm_sw_params_alloca(&sw_params);
- snd_pcm_sw_params_current(handle, sw_params);
- err = snd_pcm_sw_params(handle, sw_params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error installing software configuration (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
+ // First, check the jack server sample rate.
+ int jack_rate;
+ jack_rate = (int) jack_get_sample_rate(client);
+ if ( sampleRate != jack_rate ) {
+ jack_client_close(client);
+ sprintf( message_, "RtApiJack: the requested sample rate (%d) is different than the JACK server rate (%d).",
+ sampleRate, jack_rate );
+ error(RtError::DEBUG_WARNING);
return FAILURE;
}
- */
-
- // Set handle and flags for buffer conversion
- stream->handle[mode] = handle;
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] > 1 && stream->deInterleave[mode])
- stream->doConvertBuffer[mode] = true;
+ stream_.sampleRate = jack_rate;
+
+ // The jack server seems to support just a single floating-point
+ // data type. Since we already checked it before, just use what we
+ // found then.
+ stream_.deviceFormat[mode] = devices_[device].nativeFormats;
+ stream_.userFormat = format;
+
+ // Jack always uses non-interleaved buffers. We'll need to
+ // de-interleave if we have more than one channel.
+ stream_.deInterleave[mode] = false;
+ if ( channels > 1 )
+ stream_.deInterleave[mode] = true;
+
+ // Jack always provides host byte-ordered data.
+ stream_.doByteSwap[mode] = false;
+
+ // Get the buffer size. The buffer size and number of buffers
+ // (periods) is set when the jack server is started.
+ stream_.bufferSize = (int) jack_get_buffer_size(client);
+ *bufferSize = stream_.bufferSize;
+
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+
+ stream_.doConvertBuffer[mode] = false;
+ if (stream_.userFormat != stream_.deviceFormat[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.deInterleave[mode])
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate our JackHandle structure for the stream.
+ if ( handle == 0 ) {
+ handle = (JackHandle *) calloc(1, sizeof(JackHandle));
+ if ( handle == NULL ) {
+ sprintf(message_, "RtApiJack: error allocating JackHandle memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
+ handle->ports[0] = 0;
+ handle->ports[1] = 0;
+ if ( pthread_cond_init(&handle->condition, NULL) ) {
+ sprintf(message_, "RtApiJack: error initializing pthread condition variable!");
+ goto error;
+ }
+ stream_.apiHandle = (void *) handle;
+ handle->client = client;
+ handle->clientOpen = true;
+ }
- // Allocate necessary internal buffers
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
+ // Allocate necessary internal buffers.
+ if ( stream_.nUserChannels[0] != stream_.nUserChannels[1] ) {
long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
+ if (stream_.nUserChannels[0] >= stream_.nUserChannels[1])
+ buffer_bytes = stream_.nUserChannels[0];
else
- buffer_bytes = stream->nUserChannels[1];
-
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL)
- goto memory_error;
+ buffer_bytes = stream_.nUserChannels[1];
+
+ buffer_bytes *= *bufferSize * formatBytes(stream_.userFormat);
+ if (stream_.userBuffer) free(stream_.userBuffer);
+ stream_.userBuffer = (char *) calloc(buffer_bytes, 1);
+ if (stream_.userBuffer == NULL) {
+ sprintf(message_, "RtApiJack: error allocating user buffer memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
}
- if ( stream->doConvertBuffer[mode] ) {
+ if ( stream_.doConvertBuffer[mode] ) {
long buffer_bytes;
bool makeBuffer = true;
if ( mode == OUTPUT )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
+ buffer_bytes = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
else { // mode == INPUT
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == OUTPUT && stream->deviceBuffer ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
+ buffer_bytes = stream_.nDeviceChannels[1] * formatBytes(stream_.deviceFormat[1]);
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ long bytes_out = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
if ( buffer_bytes < bytes_out ) makeBuffer = false;
}
}
if ( makeBuffer ) {
buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL)
- goto memory_error;
+ if (stream_.deviceBuffer) free(stream_.deviceBuffer);
+ stream_.deviceBuffer = (char *) calloc(buffer_bytes, 1);
+ if (stream_.deviceBuffer == NULL) {
+ sprintf(message_, "RtApiJack: error allocating device buffer memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
}
}
- stream->device[mode] = device;
- stream->state = STREAM_STOPPED;
- if ( stream->mode == OUTPUT && mode == INPUT )
- // We had already set up an output stream.
- stream->mode = DUPLEX;
- else
- stream->mode = mode;
- stream->nBuffers = periods;
- stream->sampleRate = sampleRate;
+ // Allocate memory for the Jack ports (channels) identifiers.
+ handle->ports[mode] = (jack_port_t **) malloc (sizeof (jack_port_t *) * channels);
+ if ( handle->ports[mode] == NULL ) {
+ sprintf(message_, "RtApiJack: error allocating port handle memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
- return SUCCESS;
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ stream_.callbackInfo.usingCallback = false;
+ stream_.callbackInfo.object = (void *) this;
+ stream_.callbackInfo.apiInfo = (void *) handle;
- memory_error:
- if (stream->handle[0]) {
- snd_pcm_close(stream->handle[0]);
- stream->handle[0] = 0;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up the stream for output.
+ stream_.mode = DUPLEX;
+ else {
+ stream_.mode = mode;
+ jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
+ jack_set_xrun_callback( handle->client, jackXrun, NULL );
+ jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
}
- if (stream->handle[1]) {
- snd_pcm_close(stream->handle[1]);
- stream->handle[1] = 0;
+
+ return SUCCESS;
+
+ error:
+ if ( handle ) {
+ pthread_cond_destroy(&handle->condition);
+ if ( handle->clientOpen == true )
+ jack_client_close(handle->client);
+
+ if ( handle->ports[0] ) free(handle->ports[0]);
+ if ( handle->ports[1] ) free(handle->ports[1]);
+
+ free( handle );
+ stream_.apiHandle = 0;
}
- if (stream->userBuffer) {
- free(stream->userBuffer);
- stream->userBuffer = 0;
+
+ if (stream_.userBuffer) {
+ free(stream_.userBuffer);
+ stream_.userBuffer = 0;
}
- sprintf(message, "RtAudio: ALSA error allocating buffer memory (%s).", name);
+
error(RtError::WARNING);
return FAILURE;
}
-void RtAudio :: closeStream(int streamId)
+void RtApiJack :: closeStream()
{
// We don't want an exception to be thrown here because this
// function is called by our class destructor. So, do our own
- // streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
+ // stream check.
+ if ( stream_.mode == UNINITIALIZED ) {
+ sprintf(message_, "RtApiJack::closeStream(): no open stream to close!");
error(RtError::WARNING);
return;
}
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ if ( handle && handle->clientOpen == true ) {
+ if (stream_.state == STREAM_RUNNING)
+ jack_deactivate(handle->client);
- if (stream->callbackInfo.usingCallback) {
- pthread_cancel(stream->callbackInfo.thread);
- pthread_join(stream->callbackInfo.thread, NULL);
+ jack_client_close(handle->client);
}
- if (stream->state == STREAM_RUNNING) {
- if (stream->mode == OUTPUT || stream->mode == DUPLEX)
- snd_pcm_drop(stream->handle[0]);
- if (stream->mode == INPUT || stream->mode == DUPLEX)
- snd_pcm_drop(stream->handle[1]);
+ if ( handle ) {
+ if ( handle->ports[0] ) free(handle->ports[0]);
+ if ( handle->ports[1] ) free(handle->ports[1]);
+ pthread_cond_destroy(&handle->condition);
+ free( handle );
+ stream_.apiHandle = 0;
}
- pthread_mutex_destroy(&stream->mutex);
-
- if (stream->handle[0])
- snd_pcm_close(stream->handle[0]);
-
- if (stream->handle[1])
- snd_pcm_close(stream->handle[1]);
-
- if (stream->userBuffer)
- free(stream->userBuffer);
-
- if (stream->deviceBuffer)
- free(stream->deviceBuffer);
-
- free(stream);
- streams.erase(streamId);
-}
-
-void RtAudio :: startStream(int streamId)
-{
- // This method calls snd_pcm_prepare if the device isn't already in that state.
-
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_RUNNING)
- goto unlock;
-
- int err;
- snd_pcm_state_t state;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- state = snd_pcm_state(stream->handle[0]);
- if (state != SND_PCM_STATE_PREPARED) {
- err = snd_pcm_prepare(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error preparing pcm device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
+ if (stream_.userBuffer) {
+ free(stream_.userBuffer);
+ stream_.userBuffer = 0;
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- state = snd_pcm_state(stream->handle[1]);
- if (state != SND_PCM_STATE_PREPARED) {
- err = snd_pcm_prepare(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error preparing pcm device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
+ if (stream_.deviceBuffer) {
+ free(stream_.deviceBuffer);
+ stream_.deviceBuffer = 0;
}
- stream->state = STREAM_RUNNING;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ stream_.mode = UNINITIALIZED;
}
-void RtAudio :: stopStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- int err;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- err = snd_pcm_drain(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
-
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- err = snd_pcm_drain(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- stream->state = STREAM_STOPPED;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-void RtAudio :: abortStream(int streamId)
+void RtApiJack :: startStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
+ verifyStream();
+ if (stream_.state == STREAM_RUNNING) return;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ MUTEX_LOCK(&stream_.mutex);
- int err;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- err = snd_pcm_drop(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
+ char label[64];
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ for ( int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ snprintf(label, 64, "outport %d", i);
+ handle->ports[0][i] = jack_port_register(handle->client, (const char *)label,
+ JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
}
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- err = snd_pcm_drop(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ for ( int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ snprintf(label, 64, "inport %d", i);
+ handle->ports[1][i] = jack_port_register(handle->client, (const char *)label,
+ JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0);
}
}
- stream->state = STREAM_STOPPED;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-int RtAudio :: streamWillBlock(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
- int err = 0, frames = 0;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- err = snd_pcm_avail_update(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error getting available frames for device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
+ if (jack_activate(handle->client)) {
+ sprintf(message_, "RtApiJack: unable to activate JACK client!");
+ error(RtError::SYSTEM_ERROR);
}
- frames = err;
-
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- err = snd_pcm_avail_update(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error getting available frames for device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
+ const char **ports;
+ int result;
+ // Get the list of available ports.
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ ports = jack_get_ports(handle->client, NULL, NULL, JackPortIsPhysical|JackPortIsInput);
+ if ( ports == NULL) {
+ sprintf(message_, "RtApiJack: error determining available jack input ports!");
+ error(RtError::SYSTEM_ERROR);
+ }
+
+ // Now make the port connections. Since RtAudio wasn't designed to
+ // allow the user to select particular channels of a device, we'll
+ // just open the first "nChannels" ports.
+ for ( int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ result = 1;
+ if ( ports[i] )
+ result = jack_connect( handle->client, jack_port_name(handle->ports[0][i]), ports[i] );
+ if ( result ) {
+ free(ports);
+ sprintf(message_, "RtApiJack: error connecting output ports!");
+ error(RtError::SYSTEM_ERROR);
+ }
}
- if (frames > err) frames = err;
- }
-
- frames = stream->bufferSize - frames;
- if (frames < 0) frames = 0;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
- return frames;
-}
-
-void RtAudio :: tickStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- int stopStream = 0;
- if (stream->state == STREAM_STOPPED) {
- if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds
- return;
- }
- else if (stream->callbackInfo.usingCallback) {
- RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback;
- stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData);
+ free(ports);
}
- MUTEX_LOCK(&stream->mutex);
-
- // The state might change while waiting on a mutex.
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- int err;
- char *buffer;
- int channels;
- RTAUDIO_FORMAT format;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
-
- // Setup parameters and do buffer conversion if necessary.
- if (stream->doConvertBuffer[0]) {
- convertStreamBuffer(stream, OUTPUT);
- buffer = stream->deviceBuffer;
- channels = stream->nDeviceChannels[0];
- format = stream->deviceFormat[0];
- }
- else {
- buffer = stream->userBuffer;
- channels = stream->nUserChannels[0];
- format = stream->userFormat;
- }
-
- // Do byte swapping if necessary.
- if (stream->doByteSwap[0])
- byteSwapBuffer(buffer, stream->bufferSize * channels, format);
-
- // Write samples to device in interleaved/non-interleaved format.
- if (stream->deInterleave[0]) {
- void *bufs[channels];
- size_t offset = stream->bufferSize * formatBytes(format);
- for (int i=0; i<channels; i++)
- bufs[i] = (void *) (buffer + (i * offset));
- err = snd_pcm_writen(stream->handle[0], bufs, stream->bufferSize);
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ ports = jack_get_ports( handle->client, NULL, NULL, JackPortIsPhysical|JackPortIsOutput );
+ if ( ports == NULL) {
+ sprintf(message_, "RtApiJack: error determining available jack output ports!");
+ error(RtError::SYSTEM_ERROR);
}
- else
- err = snd_pcm_writei(stream->handle[0], buffer, stream->bufferSize);
- if (err < stream->bufferSize) {
- // Either an error or underrun occured.
- if (err == -EPIPE) {
- snd_pcm_state_t state = snd_pcm_state(stream->handle[0]);
- if (state == SND_PCM_STATE_XRUN) {
- sprintf(message, "RtAudio: ALSA underrun detected.");
- error(RtError::WARNING);
- err = snd_pcm_prepare(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error preparing handle after underrun: %s.",
- snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- else {
- sprintf(message, "RtAudio: ALSA error, current state is %s.",
- snd_pcm_state_name(state));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- goto unlock;
- }
- else {
- sprintf(message, "RtAudio: ALSA audio write error for device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
+ // Now make the port connections. See note above.
+ for ( int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ result = 1;
+ if ( ports[i] )
+ result = jack_connect( handle->client, ports[i], jack_port_name(handle->ports[1][i]) );
+ if ( result ) {
+ free(ports);
+ sprintf(message_, "RtApiJack: error connecting input ports!");
+ error(RtError::SYSTEM_ERROR);
}
}
+ free(ports);
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
-
- // Setup parameters.
- if (stream->doConvertBuffer[1]) {
- buffer = stream->deviceBuffer;
- channels = stream->nDeviceChannels[1];
- format = stream->deviceFormat[1];
- }
- else {
- buffer = stream->userBuffer;
- channels = stream->nUserChannels[1];
- format = stream->userFormat;
- }
+ handle->stopStream = false;
+ stream_.state = STREAM_RUNNING;
- // Read samples from device in interleaved/non-interleaved format.
- if (stream->deInterleave[1]) {
- void *bufs[channels];
- size_t offset = stream->bufferSize * formatBytes(format);
- for (int i=0; i<channels; i++)
- bufs[i] = (void *) (buffer + (i * offset));
- err = snd_pcm_readn(stream->handle[1], bufs, stream->bufferSize);
- }
- else
- err = snd_pcm_readi(stream->handle[1], buffer, stream->bufferSize);
+ MUTEX_UNLOCK(&stream_.mutex);
+}
- if (err < stream->bufferSize) {
- // Either an error or underrun occured.
- if (err == -EPIPE) {
- snd_pcm_state_t state = snd_pcm_state(stream->handle[1]);
- if (state == SND_PCM_STATE_XRUN) {
- sprintf(message, "RtAudio: ALSA overrun detected.");
- error(RtError::WARNING);
- err = snd_pcm_prepare(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error preparing handle after overrun: %s.",
- snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- else {
- sprintf(message, "RtAudio: ALSA error, current state is %s.",
- snd_pcm_state_name(state));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- goto unlock;
- }
- else {
- sprintf(message, "RtAudio: ALSA audio read error for device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
+void RtApiJack :: stopStream()
+{
+ verifyStream();
+ if (stream_.state == STREAM_STOPPED) return;
- // Do byte swapping if necessary.
- if (stream->doByteSwap[1])
- byteSwapBuffer(buffer, stream->bufferSize * channels, format);
+ // Change the state before the lock to improve shutdown response
+ // when using a callback.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK(&stream_.mutex);
- // Do buffer conversion if necessary.
- if (stream->doConvertBuffer[1])
- convertStreamBuffer(stream, INPUT);
- }
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ jack_deactivate(handle->client);
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
+}
- if (stream->callbackInfo.usingCallback && stopStream)
- this->stopStream(streamId);
+void RtApiJack :: abortStream()
+{
+ stopStream();
}
-extern "C" void *callbackHandler(void *ptr)
+void RtApiJack :: tickStream()
{
- CALLBACK_INFO *info = (CALLBACK_INFO *) ptr;
- RtAudio *object = (RtAudio *) info->object;
- int stream = info->streamId;
- bool *usingCallback = &info->usingCallback;
+ verifyStream();
- while ( *usingCallback ) {
- pthread_testcancel();
- try {
- object->tickStream(stream);
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n",
- exception.getMessage());
- break;
- }
- }
+ if (stream_.state == STREAM_STOPPED) return;
- return 0;
-}
+ if (stream_.callbackInfo.usingCallback) {
+ sprintf(message_, "RtApiJack: tickStream() should not be used when a callback function is set!");
+ error(RtError::WARNING);
+ return;
+ }
-//******************** End of __LINUX_ALSA__ *********************//
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
-#elif defined(__LINUX_OSS__)
+ MUTEX_LOCK(&stream_.mutex);
-#include <sys/stat.h>
-#include <sys/types.h>
-#include <sys/ioctl.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <sys/soundcard.h>
-#include <errno.h>
-#include <math.h>
+ pthread_cond_wait(&handle->condition, &stream_.mutex);
-#define DAC_NAME "/dev/dsp"
-#define MAX_DEVICES 16
-#define MAX_CHANNELS 16
+ MUTEX_UNLOCK(&stream_.mutex);
+}
-void RtAudio :: initialize(void)
+void RtApiJack :: callbackEvent( unsigned long nframes )
{
- // Count cards and devices
- nDevices = 0;
+ verifyStream();
- // We check /dev/dsp before probing devices. /dev/dsp is supposed to
- // be a link to the "default" audio device, of the form /dev/dsp0,
- // /dev/dsp1, etc... However, I've seen many cases where /dev/dsp was a
- // real device, so we need to check for that. Also, sometimes the
- // link is to /dev/dspx and other times just dspx. I'm not sure how
- // the latter works, but it does.
- char device_name[16];
- struct stat dspstat;
- int dsplink = -1;
- int i = 0;
- if (lstat(DAC_NAME, &dspstat) == 0) {
- if (S_ISLNK(dspstat.st_mode)) {
- i = readlink(DAC_NAME, device_name, sizeof(device_name));
- if (i > 0) {
- device_name[i] = '\0';
- if (i > 8) { // check for "/dev/dspx"
- if (!strncmp(DAC_NAME, device_name, 8))
- dsplink = atoi(&device_name[8]);
- }
- else if (i > 3) { // check for "dspx"
- if (!strncmp("dsp", device_name, 3))
- dsplink = atoi(&device_name[3]);
- }
- }
- else {
- sprintf(message, "RtAudio: cannot read value of symbolic link %s.", DAC_NAME);
- error(RtError::SYSTEM_ERROR);
- }
- }
- }
- else {
- sprintf(message, "RtAudio: cannot stat %s.", DAC_NAME);
- error(RtError::SYSTEM_ERROR);
+ if (stream_.state == STREAM_STOPPED) return;
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ if ( info->usingCallback && handle->stopStream ) {
+ // Check if the stream should be stopped (via the previous user
+ // callback return value). We stop the stream here, rather than
+ // after the function call, so that output data can first be
+ // processed.
+ this->stopStream();
+ return;
}
- // The OSS API doesn't provide a routine for determining the number
- // of devices. Thus, we'll just pursue a brute force method. The
- // idea is to start with /dev/dsp(0) and continue with higher device
- // numbers until we reach MAX_DSP_DEVICES. This should tell us how
- // many devices we have ... it is not a fullproof scheme, but hopefully
- // it will work most of the time.
+ MUTEX_LOCK(&stream_.mutex);
- int fd = 0;
- char names[MAX_DEVICES][16];
- for (i=-1; i<MAX_DEVICES; i++) {
+ // Invoke user callback first, to get fresh output data.
+ if ( info->usingCallback ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ handle->stopStream = callback(stream_.userBuffer, stream_.bufferSize, info->userData);
+ }
- // Probe /dev/dsp first, since it is supposed to be the default device.
- if (i == -1)
- sprintf(device_name, "%s", DAC_NAME);
- else if (i == dsplink)
- continue; // We've aready probed this device via /dev/dsp link ... try next device.
- else
- sprintf(device_name, "%s%d", DAC_NAME, i);
+ jack_default_audio_sample_t *jackbuffer;
+ long bufferBytes = nframes * sizeof (jack_default_audio_sample_t);
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- // First try to open the device for playback, then record mode.
- fd = open(device_name, O_WRONLY | O_NONBLOCK);
- if (fd == -1) {
- // Open device for playback failed ... either busy or doesn't exist.
- if (errno != EBUSY && errno != EAGAIN) {
- // Try to open for capture
- fd = open(device_name, O_RDONLY | O_NONBLOCK);
- if (fd == -1) {
- // Open device for record failed.
- if (errno != EBUSY && errno != EAGAIN)
- continue;
- else {
- sprintf(message, "RtAudio: OSS record device (%s) is busy.", device_name);
- error(RtError::WARNING);
- // still count it for now
- }
- }
- }
- else {
- sprintf(message, "RtAudio: OSS playback device (%s) is busy.", device_name);
- error(RtError::WARNING);
- // still count it for now
+ if (stream_.doConvertBuffer[0]) {
+ convertStreamBuffer(OUTPUT);
+
+ for ( int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(handle->ports[0][i],
+ (jack_nframes_t) nframes);
+ memcpy(jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
}
}
+ else { // single channel only
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(handle->ports[0][0],
+ (jack_nframes_t) nframes);
+ memcpy(jackbuffer, stream_.userBuffer, bufferBytes );
+ }
+ }
- if (fd >= 0) close(fd);
- strncpy(names[nDevices], device_name, 16);
- nDevices++;
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ if (stream_.doConvertBuffer[1]) {
+ for ( int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(handle->ports[1][i],
+ (jack_nframes_t) nframes);
+ memcpy(&stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
+ }
+ convertStreamBuffer(INPUT);
+ }
+ else { // single channel only
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(handle->ports[1][0],
+ (jack_nframes_t) nframes);
+ memcpy(stream_.userBuffer, jackbuffer, bufferBytes );
+ }
}
- if (nDevices == 0) return;
+ if ( !info->usingCallback )
+ pthread_cond_signal(&handle->condition);
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
+ MUTEX_UNLOCK(&stream_.mutex);
+}
+
+void RtApiJack :: setStreamCallback(RtAudioCallback callback, void *userData)
+{
+ verifyStream();
+
+ if ( stream_.callbackInfo.usingCallback ) {
+ sprintf(message_, "RtApiJack: A callback is already set for this stream!");
+ error(RtError::WARNING);
+ return;
}
- // Write device ascii identifiers to device control structure and then probe capabilities.
- for (i=0; i<nDevices; i++) {
- strncpy(devices[i].name, names[i], 16);
- //probeDeviceInfo(&devices[i]);
+ stream_.callbackInfo.callback = (void *) callback;
+ stream_.callbackInfo.userData = userData;
+ stream_.callbackInfo.usingCallback = true;
+}
+
+void RtApiJack :: cancelStreamCallback()
+{
+ verifyStream();
+
+ if (stream_.callbackInfo.usingCallback) {
+
+ if (stream_.state == STREAM_RUNNING)
+ stopStream();
+
+ MUTEX_LOCK(&stream_.mutex);
+
+ stream_.callbackInfo.usingCallback = false;
+ stream_.callbackInfo.userData = NULL;
+ stream_.state = STREAM_STOPPED;
+ stream_.callbackInfo.callback = NULL;
+
+ MUTEX_UNLOCK(&stream_.mutex);
}
+}
- return;
+#endif
+
+#if defined(__LINUX_ALSA__)
+
+#include <alsa/asoundlib.h>
+#include <unistd.h>
+#include <ctype.h>
+
+extern "C" void *alsaCallbackHandler(void * ptr);
+
+RtApiAlsa :: RtApiAlsa()
+{
+ this->initialize();
+
+ if (nDevices_ <= 0) {
+ sprintf(message_, "RtApiAlsa: no Linux ALSA audio devices found!");
+ error(RtError::NO_DEVICES_FOUND);
+ }
}
-int RtAudio :: getDefaultInputDevice(void)
+RtApiAlsa :: ~RtApiAlsa()
{
- // No OSS API functions for default devices.
- return 0;
+ if ( stream_.mode != UNINITIALIZED )
+ closeStream();
}
-int RtAudio :: getDefaultOutputDevice(void)
+void RtApiAlsa :: initialize(void)
{
- // No OSS API functions for default devices.
- return 0;
+ int card, subdevice, result;
+ char name[64];
+ const char *cardId;
+ snd_ctl_t *handle;
+ snd_ctl_card_info_t *info;
+ snd_ctl_card_info_alloca(&info);
+ RtApiDevice device;
+
+ // Count cards and devices
+ nDevices_ = 0;
+ card = -1;
+ snd_card_next(&card);
+ while ( card >= 0 ) {
+ sprintf(name, "hw:%d", card);
+ result = snd_ctl_open(&handle, name, 0);
+ if (result < 0) {
+ sprintf(message_, "RtApiAlsa: control open (%i): %s.", card, snd_strerror(result));
+ error(RtError::DEBUG_WARNING);
+ goto next_card;
+ }
+ result = snd_ctl_card_info(handle, info);
+ if (result < 0) {
+ sprintf(message_, "RtApiAlsa: control hardware info (%i): %s.", card, snd_strerror(result));
+ error(RtError::DEBUG_WARNING);
+ goto next_card;
+ }
+ cardId = snd_ctl_card_info_get_id(info);
+ subdevice = -1;
+ while (1) {
+ result = snd_ctl_pcm_next_device(handle, &subdevice);
+ if (result < 0) {
+ sprintf(message_, "RtApiAlsa: control next device (%i): %s.", card, snd_strerror(result));
+ error(RtError::DEBUG_WARNING);
+ break;
+ }
+ if (subdevice < 0)
+ break;
+ sprintf( name, "hw:%d,%d", card, subdevice );
+ // If a cardId exists and it contains at least one non-numeric
+ // character, use it to identify the device. This avoids a bug
+ // in ALSA such that a numeric string is interpreted as a device
+ // number.
+ for ( unsigned int i=0; i<strlen(cardId); i++ ) {
+ if ( !isdigit( cardId[i] ) ) {
+ sprintf( name, "hw:%s,%d", cardId, subdevice );
+ break;
+ }
+ }
+ device.name.erase();
+ device.name.append( (const char *)name, strlen(name)+1 );
+ devices_.push_back(device);
+ nDevices_++;
+ }
+ next_card:
+ snd_ctl_close(handle);
+ snd_card_next(&card);
+ }
}
-void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info)
+void RtApiAlsa :: probeDeviceInfo(RtApiDevice *info)
{
- int i, fd, channels, mask;
+ int err;
+ int open_mode = SND_PCM_ASYNC;
+ snd_pcm_t *handle;
+ snd_ctl_t *chandle;
+ snd_pcm_stream_t stream;
+ snd_pcm_info_t *pcminfo;
+ snd_pcm_info_alloca(&pcminfo);
+ snd_pcm_hw_params_t *params;
+ snd_pcm_hw_params_alloca(¶ms);
+ char name[64];
+ char *card;
- // The OSS API doesn't provide a means for probing the capabilities
- // of devices. Thus, we'll just pursue a brute force method.
+ // Open the control interface for this card.
+ strncpy( name, info->name.c_str(), 64 );
+ card = strtok(name, ",");
+ err = snd_ctl_open(&chandle, card, SND_CTL_NONBLOCK);
+ if (err < 0) {
+ sprintf(message_, "RtApiAlsa: control open (%s): %s.", card, snd_strerror(err));
+ error(RtError::DEBUG_WARNING);
+ return;
+ }
+ unsigned int dev = (unsigned int) atoi( strtok(NULL, ",") );
// First try for playback
- fd = open(info->name, O_WRONLY | O_NONBLOCK);
- if (fd == -1) {
- // Open device failed ... either busy or doesn't exist
- if (errno == EBUSY || errno == EAGAIN)
- sprintf(message, "RtAudio: OSS playback device (%s) is busy and cannot be probed.",
- info->name);
+ stream = SND_PCM_STREAM_PLAYBACK;
+ snd_pcm_info_set_device(pcminfo, dev);
+ snd_pcm_info_set_subdevice(pcminfo, 0);
+ snd_pcm_info_set_stream(pcminfo, stream);
+
+ if ((err = snd_ctl_pcm_info(chandle, pcminfo)) < 0) {
+ if (err == -ENOENT) {
+ sprintf(message_, "RtApiAlsa: pcm device (%s) doesn't handle output!", info->name.c_str());
+ error(RtError::DEBUG_WARNING);
+ }
+ else {
+ sprintf(message_, "RtApiAlsa: snd_ctl_pcm_info error for device (%s) output: %s",
+ info->name.c_str(), snd_strerror(err));
+ error(RtError::DEBUG_WARNING);
+ }
+ goto capture_probe;
+ }
+
+ err = snd_pcm_open(&handle, info->name.c_str(), stream, open_mode | SND_PCM_NONBLOCK );
+ if (err < 0) {
+ if ( err == EBUSY )
+ sprintf(message_, "RtApiAlsa: pcm playback device (%s) is busy: %s.",
+ info->name.c_str(), snd_strerror(err));
else
- sprintf(message, "RtAudio: OSS playback device (%s) open error.", info->name);
+ sprintf(message_, "RtApiAlsa: pcm playback open (%s) error: %s.",
+ info->name.c_str(), snd_strerror(err));
error(RtError::DEBUG_WARNING);
goto capture_probe;
}
- // We have an open device ... see how many channels it can handle
- for (i=MAX_CHANNELS; i>0; i--) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) {
- // This would normally indicate some sort of hardware error, but under ALSA's
- // OSS emulation, it sometimes indicates an invalid channel value. Further,
- // the returned channel value is not changed. So, we'll ignore the possible
- // hardware error.
- continue; // try next channel number
- }
- // Check to see whether the device supports the requested number of channels
- if (channels != i ) continue; // try next channel number
- // If here, we found the largest working channel value
- break;
+ // We have an open device ... allocate the parameter structure.
+ err = snd_pcm_hw_params_any(handle, params);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: hardware probe error (%s): %s.",
+ info->name.c_str(), snd_strerror(err));
+ error(RtError::WARNING);
+ goto capture_probe;
}
- info->maxOutputChannels = i;
- // Now find the minimum number of channels it can handle
- for (i=1; i<=info->maxOutputChannels; i++) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
- continue; // try next channel number
- // If here, we found the smallest working channel value
- break;
+ // Get output channel information.
+ unsigned int value;
+ err = snd_pcm_hw_params_get_channels_min(params, &value);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: hardware minimum channel probe error (%s): %s.",
+ info->name.c_str(), snd_strerror(err));
+ error(RtError::WARNING);
+ goto capture_probe;
}
- info->minOutputChannels = i;
- close(fd);
+ info->minOutputChannels = value;
+
+ err = snd_pcm_hw_params_get_channels_max(params, &value);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: hardware maximum channel probe error (%s): %s.",
+ info->name.c_str(), snd_strerror(err));
+ error(RtError::WARNING);
+ goto capture_probe;
+ }
+ info->maxOutputChannels = value;
+
+ snd_pcm_close(handle);
capture_probe:
// Now try for capture
- fd = open(info->name, O_RDONLY | O_NONBLOCK);
- if (fd == -1) {
- // Open device for capture failed ... either busy or doesn't exist
- if (errno == EBUSY || errno == EAGAIN)
- sprintf(message, "RtAudio: OSS capture device (%s) is busy and cannot be probed.",
- info->name);
+ stream = SND_PCM_STREAM_CAPTURE;
+ snd_pcm_info_set_stream(pcminfo, stream);
+
+ err = snd_ctl_pcm_info(chandle, pcminfo);
+ snd_ctl_close(chandle);
+ if ( err < 0 ) {
+ if (err == -ENOENT) {
+ sprintf(message_, "RtApiAlsa: pcm device (%s) doesn't handle input!", info->name.c_str());
+ error(RtError::DEBUG_WARNING);
+ }
+ else {
+ sprintf(message_, "RtApiAlsa: snd_ctl_pcm_info error for device (%s) input: %s",
+ info->name.c_str(), snd_strerror(err));
+ error(RtError::DEBUG_WARNING);
+ }
+ if (info->maxOutputChannels == 0)
+ // didn't open for playback either ... device invalid
+ return;
+ goto probe_parameters;
+ }
+
+ err = snd_pcm_open(&handle, info->name.c_str(), stream, open_mode | SND_PCM_NONBLOCK);
+ if (err < 0) {
+ if ( err == EBUSY )
+ sprintf(message_, "RtApiAlsa: pcm capture device (%s) is busy: %s.",
+ info->name.c_str(), snd_strerror(err));
else
- sprintf(message, "RtAudio: OSS capture device (%s) open error.", info->name);
+ sprintf(message_, "RtApiAlsa: pcm capture open (%s) error: %s.",
+ info->name.c_str(), snd_strerror(err));
error(RtError::DEBUG_WARNING);
if (info->maxOutputChannels == 0)
// didn't open for playback either ... device invalid
goto probe_parameters;
}
- // We have the device open for capture ... see how many channels it can handle
- for (i=MAX_CHANNELS; i>0; i--) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) {
- continue; // as above
- }
- // If here, we found a working channel value
- break;
+ // We have an open capture device ... allocate the parameter structure.
+ err = snd_pcm_hw_params_any(handle, params);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: hardware probe error (%s): %s.",
+ info->name.c_str(), snd_strerror(err));
+ error(RtError::WARNING);
+ if (info->maxOutputChannels > 0)
+ goto probe_parameters;
+ else
+ return;
}
- info->maxInputChannels = i;
- // Now find the minimum number of channels it can handle
- for (i=1; i<=info->maxInputChannels; i++) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
- continue; // try next channel number
- // If here, we found the smallest working channel value
- break;
+ // Get input channel information.
+ err = snd_pcm_hw_params_get_channels_min(params, &value);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: hardware minimum in channel probe error (%s): %s.",
+ info->name.c_str(), snd_strerror(err));
+ error(RtError::WARNING);
+ if (info->maxOutputChannels > 0)
+ goto probe_parameters;
+ else
+ return;
}
- info->minInputChannels = i;
- close(fd);
+ info->minInputChannels = value;
- if (info->maxOutputChannels == 0 && info->maxInputChannels == 0) {
- sprintf(message, "RtAudio: OSS device (%s) reports zero channels for input and output.",
- info->name);
- error(RtError::DEBUG_WARNING);
- return;
+ err = snd_pcm_hw_params_get_channels_max(params, &value);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: hardware maximum in channel probe error (%s): %s.",
+ info->name.c_str(), snd_strerror(err));
+ error(RtError::WARNING);
+ if (info->maxOutputChannels > 0)
+ goto probe_parameters;
+ else
+ return;
}
+ info->maxInputChannels = value;
+
+ snd_pcm_close(handle);
// If device opens for both playback and capture, we determine the channels.
if (info->maxOutputChannels == 0 || info->maxInputChannels == 0)
goto probe_parameters;
- fd = open(info->name, O_RDWR | O_NONBLOCK);
- if (fd == -1)
- goto probe_parameters;
-
- ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
- ioctl(fd, SNDCTL_DSP_GETCAPS, &mask);
- if (mask & DSP_CAP_DUPLEX) {
- info->hasDuplexSupport = true;
- // We have the device open for duplex ... see how many channels it can handle
- for (i=MAX_CHANNELS; i>0; i--) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
- continue; // as above
- // If here, we found a working channel value
- break;
- }
- info->maxDuplexChannels = i;
-
- // Now find the minimum number of channels it can handle
- for (i=1; i<=info->maxDuplexChannels; i++) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
- continue; // try next channel number
- // If here, we found the smallest working channel value
- break;
- }
- info->minDuplexChannels = i;
- }
- close(fd);
+ info->hasDuplexSupport = true;
+ info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ?
+ info->maxInputChannels : info->maxOutputChannels;
+ info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ?
+ info->minInputChannels : info->minOutputChannels;
probe_parameters:
- // At this point, we need to figure out the supported data formats
- // and sample rates. We'll proceed by openning the device in the
- // direction with the maximum number of channels, or playback if
+ // At this point, we just need to figure out the supported data
+ // formats and sample rates. We'll proceed by opening the device in
+ // the direction with the maximum number of channels, or playback if
// they are equal. This might limit our sample rate options, but so
// be it.
- if (info->maxOutputChannels >= info->maxInputChannels) {
- fd = open(info->name, O_WRONLY | O_NONBLOCK);
- channels = info->maxOutputChannels;
- }
- else {
- fd = open(info->name, O_RDONLY | O_NONBLOCK);
- channels = info->maxInputChannels;
- }
+ if (info->maxOutputChannels >= info->maxInputChannels)
+ stream = SND_PCM_STREAM_PLAYBACK;
+ else
+ stream = SND_PCM_STREAM_CAPTURE;
- if (fd == -1) {
- // We've got some sort of conflict ... abort
- sprintf(message, "RtAudio: OSS device (%s) won't reopen during probe.",
- info->name);
- error(RtError::DEBUG_WARNING);
+ err = snd_pcm_open(&handle, info->name.c_str(), stream, open_mode);
+ if (err < 0) {
+ sprintf(message_, "RtApiAlsa: pcm (%s) won't reopen during probe: %s.",
+ info->name.c_str(), snd_strerror(err));
+ error(RtError::WARNING);
return;
}
- // We have an open device ... set to maximum channels.
- i = channels;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) {
- // We've got some sort of conflict ... abort
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) won't revert to previous channel setting.",
- info->name);
- error(RtError::DEBUG_WARNING);
+ // We have an open device ... allocate the parameter structure.
+ err = snd_pcm_hw_params_any(handle, params);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: hardware reopen probe error (%s): %s.",
+ info->name.c_str(), snd_strerror(err));
+ error(RtError::WARNING);
return;
}
- if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) {
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) can't get supported audio formats.",
- info->name);
+ // Test our discrete set of sample rate values.
+ int dir = 0;
+ info->sampleRates.clear();
+ for (unsigned int i=0; i<MAX_SAMPLE_RATES; i++) {
+ if (snd_pcm_hw_params_test_rate(handle, params, SAMPLE_RATES[i], dir) == 0)
+ info->sampleRates.push_back(SAMPLE_RATES[i]);
+ }
+ if (info->sampleRates.size() == 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: no supported sample rates found for device (%s).",
+ info->name.c_str());
error(RtError::DEBUG_WARNING);
return;
}
- // Probe the supported data formats ... we don't care about endian-ness just yet.
- int format;
+ // Probe the supported data formats ... we don't care about endian-ness just yet
+ snd_pcm_format_t format;
info->nativeFormats = 0;
-#if defined (AFMT_S32_BE)
- // This format does not seem to be in the 2.4 kernel version of OSS soundcard.h
- if (mask & AFMT_S32_BE) {
- format = AFMT_S32_BE;
- info->nativeFormats |= RTAUDIO_SINT32;
- }
-#endif
-#if defined (AFMT_S32_LE)
- /* This format is not in the 2.4.4 kernel version of OSS soundcard.h */
- if (mask & AFMT_S32_LE) {
- format = AFMT_S32_LE;
- info->nativeFormats |= RTAUDIO_SINT32;
- }
-#endif
- if (mask & AFMT_S8) {
- format = AFMT_S8;
+ format = SND_PCM_FORMAT_S8;
+ if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
info->nativeFormats |= RTAUDIO_SINT8;
- }
- if (mask & AFMT_S16_BE) {
- format = AFMT_S16_BE;
- info->nativeFormats |= RTAUDIO_SINT16;
- }
- if (mask & AFMT_S16_LE) {
- format = AFMT_S16_LE;
+ format = SND_PCM_FORMAT_S16;
+ if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
info->nativeFormats |= RTAUDIO_SINT16;
- }
+ format = SND_PCM_FORMAT_S24;
+ if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
+ info->nativeFormats |= RTAUDIO_SINT24;
+ format = SND_PCM_FORMAT_S32;
+ if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
+ info->nativeFormats |= RTAUDIO_SINT32;
+ format = SND_PCM_FORMAT_FLOAT;
+ if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
+ info->nativeFormats |= RTAUDIO_FLOAT32;
+ format = SND_PCM_FORMAT_FLOAT64;
+ if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
+ info->nativeFormats |= RTAUDIO_FLOAT64;
// Check that we have at least one supported format
if (info->nativeFormats == 0) {
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) data format not supported by RtAudio.",
- info->name);
- error(RtError::DEBUG_WARNING);
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: pcm device (%s) data format not supported by RtAudio.",
+ info->name.c_str());
+ error(RtError::WARNING);
return;
}
- // Set the format
- i = format;
- if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1 || format != i) {
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) error setting data format.",
- info->name);
- error(RtError::DEBUG_WARNING);
- return;
+ // That's all ... close the device and return
+ snd_pcm_close(handle);
+ info->probed = true;
+ return;
+}
+
+bool RtApiAlsa :: probeDeviceOpen( int device, StreamMode mode, int channels,
+ int sampleRate, RtAudioFormat format,
+ int *bufferSize, int numberOfBuffers )
+{
+#if defined(__RTAUDIO_DEBUG__)
+ snd_output_t *out;
+ snd_output_stdio_attach(&out, stderr, 0);
+#endif
+
+ // I'm not using the "plug" interface ... too much inconsistent behavior.
+ const char *name = devices_[device].name.c_str();
+
+ snd_pcm_stream_t alsa_stream;
+ if (mode == OUTPUT)
+ alsa_stream = SND_PCM_STREAM_PLAYBACK;
+ else
+ alsa_stream = SND_PCM_STREAM_CAPTURE;
+
+ int err;
+ snd_pcm_t *handle;
+ int alsa_open_mode = SND_PCM_ASYNC;
+ err = snd_pcm_open(&handle, name, alsa_stream, alsa_open_mode);
+ if (err < 0) {
+ sprintf(message_,"RtApiAlsa: pcm device (%s) won't open: %s.",
+ name, snd_strerror(err));
+ error(RtError::WARNING);
+ return FAILURE;
}
- // Probe the supported sample rates ... first get lower limit
- int speed = 1;
- if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) {
- // If we get here, we're probably using an ALSA driver with OSS-emulation,
- // which doesn't conform to the OSS specification. In this case,
- // we'll probe our predefined list of sample rates for working values.
- info->nSampleRates = 0;
- for (i=0; i<MAX_SAMPLE_RATES; i++) {
- speed = SAMPLE_RATES[i];
- if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) != -1) {
- info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i];
- info->nSampleRates++;
- }
- }
- if (info->nSampleRates == 0) {
- close(fd);
- return;
- }
- goto finished;
+ // Fill the parameter structure.
+ snd_pcm_hw_params_t *hw_params;
+ snd_pcm_hw_params_alloca(&hw_params);
+ err = snd_pcm_hw_params_any(handle, hw_params);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: error getting parameter handle (%s): %s.",
+ name, snd_strerror(err));
+ error(RtError::WARNING);
+ return FAILURE;
}
- info->sampleRates[0] = speed;
- // Now get upper limit
- speed = 1000000;
- if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) {
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) error setting sample rate.",
- info->name);
- error(RtError::DEBUG_WARNING);
- return;
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf(stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n");
+ snd_pcm_hw_params_dump(hw_params, out);
+#endif
+
+ // Set access ... try interleaved access first, then non-interleaved
+ if ( !snd_pcm_hw_params_test_access( handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED) ) {
+ err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
+ }
+ else if ( !snd_pcm_hw_params_test_access( handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED) ) {
+ err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED);
+ stream_.deInterleave[mode] = true;
+ }
+ else {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: device (%s) access not supported by RtAudio.", name);
+ error(RtError::WARNING);
+ return FAILURE;
}
- info->sampleRates[1] = speed;
- info->nSampleRates = -1;
- finished: // That's all ... close the device and return
- close(fd);
- info->probed = true;
- return;
-}
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: error setting access ( (%s): %s.", name, snd_strerror(err));
+ error(RtError::WARNING);
+ return FAILURE;
+ }
-bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
- STREAM_MODE mode, int channels,
- int sampleRate, RTAUDIO_FORMAT format,
- int *bufferSize, int numberOfBuffers)
-{
- int buffers, buffer_bytes, device_channels, device_format;
- int srate, temp, fd;
+ // Determine how to set the device format.
+ stream_.userFormat = format;
+ snd_pcm_format_t device_format = SND_PCM_FORMAT_UNKNOWN;
+
+ if (format == RTAUDIO_SINT8)
+ device_format = SND_PCM_FORMAT_S8;
+ else if (format == RTAUDIO_SINT16)
+ device_format = SND_PCM_FORMAT_S16;
+ else if (format == RTAUDIO_SINT24)
+ device_format = SND_PCM_FORMAT_S24;
+ else if (format == RTAUDIO_SINT32)
+ device_format = SND_PCM_FORMAT_S32;
+ else if (format == RTAUDIO_FLOAT32)
+ device_format = SND_PCM_FORMAT_FLOAT;
+ else if (format == RTAUDIO_FLOAT64)
+ device_format = SND_PCM_FORMAT_FLOAT64;
- const char *name = devices[device].name;
+ if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
+ stream_.deviceFormat[mode] = format;
+ goto set_format;
+ }
- if (mode == OUTPUT)
- fd = open(name, O_WRONLY | O_NONBLOCK);
- else { // mode == INPUT
- if (stream->mode == OUTPUT && stream->device[0] == device) {
- // We just set the same device for playback ... close and reopen for duplex (OSS only).
- close(stream->handle[0]);
- stream->handle[0] = 0;
- // First check that the number previously set channels is the same.
- if (stream->nUserChannels[0] != channels) {
- sprintf(message, "RtAudio: input/output channels must be equal for OSS duplex device (%s).", name);
- goto error;
- }
- fd = open(name, O_RDWR | O_NONBLOCK);
- }
- else
- fd = open(name, O_RDONLY | O_NONBLOCK);
+ // The user requested format is not natively supported by the device.
+ device_format = SND_PCM_FORMAT_FLOAT64;
+ if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+ goto set_format;
+ }
+
+ device_format = SND_PCM_FORMAT_FLOAT;
+ if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ goto set_format;
+ }
+
+ device_format = SND_PCM_FORMAT_S32;
+ if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ goto set_format;
}
- if (fd == -1) {
- if (errno == EBUSY || errno == EAGAIN)
- sprintf(message, "RtAudio: OSS device (%s) is busy and cannot be opened.",
- name);
- else
- sprintf(message, "RtAudio: OSS device (%s) cannot be opened.", name);
- goto error;
+ device_format = SND_PCM_FORMAT_S24;
+ if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ goto set_format;
}
- // Now reopen in blocking mode.
- close(fd);
- if (mode == OUTPUT)
- fd = open(name, O_WRONLY | O_SYNC);
- else { // mode == INPUT
- if (stream->mode == OUTPUT && stream->device[0] == device)
- fd = open(name, O_RDWR | O_SYNC);
- else
- fd = open(name, O_RDONLY | O_SYNC);
+ device_format = SND_PCM_FORMAT_S16;
+ if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ goto set_format;
}
- if (fd == -1) {
- sprintf(message, "RtAudio: OSS device (%s) cannot be opened.", name);
- goto error;
+ device_format = SND_PCM_FORMAT_S8;
+ if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ goto set_format;
}
- // Get the sample format mask
- int mask;
- if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) {
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) can't get supported audio formats.",
- name);
- goto error;
+ // If we get here, no supported format was found.
+ sprintf(message_,"RtApiAlsa: pcm device (%s) data format not supported by RtAudio.", name);
+ snd_pcm_close(handle);
+ error(RtError::WARNING);
+ return FAILURE;
+
+ set_format:
+ err = snd_pcm_hw_params_set_format(handle, hw_params, device_format);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: error setting format (%s): %s.",
+ name, snd_strerror(err));
+ error(RtError::WARNING);
+ return FAILURE;
}
- // Determine how to set the device format.
- stream->userFormat = format;
- device_format = -1;
- stream->doByteSwap[mode] = false;
- if (format == RTAUDIO_SINT8) {
- if (mask & AFMT_S8) {
- device_format = AFMT_S8;
- stream->deviceFormat[mode] = RTAUDIO_SINT8;
+ // Determine whether byte-swaping is necessary.
+ stream_.doByteSwap[mode] = false;
+ if (device_format != SND_PCM_FORMAT_S8) {
+ err = snd_pcm_format_cpu_endian(device_format);
+ if (err == 0)
+ stream_.doByteSwap[mode] = true;
+ else if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: error getting format endian-ness (%s): %s.",
+ name, snd_strerror(err));
+ error(RtError::WARNING);
+ return FAILURE;
}
}
- else if (format == RTAUDIO_SINT16) {
- if (mask & AFMT_S16_NE) {
- device_format = AFMT_S16_NE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- }
-#if BYTE_ORDER == LITTLE_ENDIAN
- else if (mask & AFMT_S16_BE) {
- device_format = AFMT_S16_BE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->doByteSwap[mode] = true;
- }
-#else
- else if (mask & AFMT_S16_LE) {
- device_format = AFMT_S16_LE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->doByteSwap[mode] = true;
- }
-#endif
+
+ // Set the sample rate.
+ err = snd_pcm_hw_params_set_rate(handle, hw_params, (unsigned int)sampleRate, 0);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: error setting sample rate (%d) on device (%s): %s.",
+ sampleRate, name, snd_strerror(err));
+ error(RtError::WARNING);
+ return FAILURE;
}
-#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE)
- else if (format == RTAUDIO_SINT32) {
- if (mask & AFMT_S32_NE) {
- device_format = AFMT_S32_NE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- }
-#if BYTE_ORDER == LITTLE_ENDIAN
- else if (mask & AFMT_S32_BE) {
- device_format = AFMT_S32_BE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- stream->doByteSwap[mode] = true;
- }
-#else
- else if (mask & AFMT_S32_LE) {
- device_format = AFMT_S32_LE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- stream->doByteSwap[mode] = true;
- }
-#endif
+
+ // Determine the number of channels for this device. We support a possible
+ // minimum device channel number > than the value requested by the user.
+ stream_.nUserChannels[mode] = channels;
+ unsigned int value;
+ err = snd_pcm_hw_params_get_channels_max(hw_params, &value);
+ int device_channels = value;
+ if (err < 0 || device_channels < channels) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: channels (%d) not supported by device (%s).",
+ channels, name);
+ error(RtError::WARNING);
+ return FAILURE;
}
-#endif
- if (device_format == -1) {
- // The user requested format is not natively supported by the device.
- if (mask & AFMT_S16_NE) {
- device_format = AFMT_S16_NE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- }
-#if BYTE_ORDER == LITTLE_ENDIAN
- else if (mask & AFMT_S16_BE) {
- device_format = AFMT_S16_BE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->doByteSwap[mode] = true;
- }
-#else
- else if (mask & AFMT_S16_LE) {
- device_format = AFMT_S16_LE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->doByteSwap[mode] = true;
- }
-#endif
-#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE)
- else if (mask & AFMT_S32_NE) {
- device_format = AFMT_S32_NE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- }
-#if BYTE_ORDER == LITTLE_ENDIAN
- else if (mask & AFMT_S32_BE) {
- device_format = AFMT_S32_BE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- stream->doByteSwap[mode] = true;
- }
-#else
- else if (mask & AFMT_S32_LE) {
- device_format = AFMT_S32_LE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- stream->doByteSwap[mode] = true;
- }
-#endif
-#endif
- else if (mask & AFMT_S8) {
- device_format = AFMT_S8;
- stream->deviceFormat[mode] = RTAUDIO_SINT8;
- }
+ err = snd_pcm_hw_params_get_channels_min(hw_params, &value);
+ if (err < 0 ) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: error getting min channels count on device (%s).", name);
+ error(RtError::WARNING);
+ return FAILURE;
}
+ device_channels = value;
+ if (device_channels < channels) device_channels = channels;
+ stream_.nDeviceChannels[mode] = device_channels;
- if (stream->deviceFormat[mode] == 0) {
- // This really shouldn't happen ...
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) data format not supported by RtAudio.",
- name);
- goto error;
+ // Set the device channels.
+ err = snd_pcm_hw_params_set_channels(handle, hw_params, device_channels);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: error setting channels (%d) on device (%s): %s.",
+ device_channels, name, snd_strerror(err));
+ error(RtError::WARNING);
+ return FAILURE;
}
- // Determine the number of channels for this device. Note that the
- // channel value requested by the user might be < min_X_Channels.
- stream->nUserChannels[mode] = channels;
- device_channels = channels;
- if (mode == OUTPUT) {
- if (channels < devices[device].minOutputChannels)
- device_channels = devices[device].minOutputChannels;
+ // Set the buffer number, which in ALSA is referred to as the "period".
+ int dir;
+ unsigned int periods = numberOfBuffers;
+ // Even though the hardware might allow 1 buffer, it won't work reliably.
+ if (periods < 2) periods = 2;
+ err = snd_pcm_hw_params_get_periods_min(hw_params, &value, &dir);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: error getting min periods on device (%s): %s.",
+ name, snd_strerror(err));
+ error(RtError::WARNING);
+ return FAILURE;
}
- else { // mode == INPUT
- if (stream->mode == OUTPUT && stream->device[0] == device) {
- // We're doing duplex setup here.
- if (channels < devices[device].minDuplexChannels)
- device_channels = devices[device].minDuplexChannels;
- }
- else {
- if (channels < devices[device].minInputChannels)
- device_channels = devices[device].minInputChannels;
- }
+ if (value > periods) periods = value;
+ err = snd_pcm_hw_params_get_periods_max(hw_params, &value, &dir);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: error getting max periods on device (%s): %s.",
+ name, snd_strerror(err));
+ error(RtError::WARNING);
+ return FAILURE;
}
- stream->nDeviceChannels[mode] = device_channels;
+ if (value < periods) periods = value;
- // Attempt to set the buffer size. According to OSS, the minimum
- // number of buffers is two. The supposed minimum buffer size is 16
- // bytes, so that will be our lower bound. The argument to this
- // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
- // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
- // We'll check the actual value used near the end of the setup
- // procedure.
- buffer_bytes = *bufferSize * formatBytes(stream->deviceFormat[mode]) * device_channels;
- if (buffer_bytes < 16) buffer_bytes = 16;
- buffers = numberOfBuffers;
- if (buffers < 2) buffers = 2;
- temp = ((int) buffers << 16) + (int)(log10((double)buffer_bytes)/log10(2.0));
- if (ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &temp)) {
- close(fd);
- sprintf(message, "RtAudio: OSS error setting fragment size for device (%s).",
- name);
- goto error;
+ err = snd_pcm_hw_params_set_periods(handle, hw_params, periods, 0);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: error setting periods (%s): %s.",
+ name, snd_strerror(err));
+ error(RtError::WARNING);
+ return FAILURE;
}
- stream->nBuffers = buffers;
- // Set the data format.
- temp = device_format;
- if (ioctl(fd, SNDCTL_DSP_SETFMT, &device_format) == -1 || device_format != temp) {
- close(fd);
- sprintf(message, "RtAudio: OSS error setting data format for device (%s).",
- name);
- goto error;
+ // Set the buffer (or period) size.
+ snd_pcm_uframes_t period_size;
+ err = snd_pcm_hw_params_get_period_size_min(hw_params, &period_size, &dir);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: error getting period size (%s): %s.",
+ name, snd_strerror(err));
+ error(RtError::WARNING);
+ return FAILURE;
}
+ if (*bufferSize < (int) period_size) *bufferSize = (int) period_size;
- // Set the number of channels.
- temp = device_channels;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &device_channels) == -1 || device_channels != temp) {
- close(fd);
- sprintf(message, "RtAudio: OSS error setting %d channels on device (%s).",
- temp, name);
- goto error;
+ err = snd_pcm_hw_params_set_period_size(handle, hw_params, *bufferSize, 0);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: error setting period size (%s): %s.",
+ name, snd_strerror(err));
+ error(RtError::WARNING);
+ return FAILURE;
}
- // Set the sample rate.
- srate = sampleRate;
- temp = srate;
- if (ioctl(fd, SNDCTL_DSP_SPEED, &srate) == -1) {
- close(fd);
- sprintf(message, "RtAudio: OSS error setting sample rate = %d on device (%s).",
- temp, name);
- goto error;
+ // If attempting to setup a duplex stream, the bufferSize parameter
+ // MUST be the same in both directions!
+ if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+ sprintf( message_, "RtApiAlsa: error setting buffer size for duplex stream on device (%s).",
+ name );
+ error(RtError::DEBUG_WARNING);
+ return FAILURE;
}
- // Verify the sample rate setup worked.
- if (abs(srate - temp) > 100) {
- close(fd);
- sprintf(message, "RtAudio: OSS error ... audio device (%s) doesn't support sample rate of %d.",
- name, temp);
- goto error;
- }
- stream->sampleRate = sampleRate;
+ stream_.bufferSize = *bufferSize;
- if (ioctl(fd, SNDCTL_DSP_GETBLKSIZE, &buffer_bytes) == -1) {
- close(fd);
- sprintf(message, "RtAudio: OSS error getting buffer size for device (%s).",
- name);
- goto error;
+ // Install the hardware configuration
+ err = snd_pcm_hw_params(handle, hw_params);
+ if (err < 0) {
+ snd_pcm_close(handle);
+ sprintf(message_, "RtApiAlsa: error installing hardware configuration (%s): %s.",
+ name, snd_strerror(err));
+ error(RtError::WARNING);
+ return FAILURE;
}
- // Save buffer size (in sample frames).
- *bufferSize = buffer_bytes / (formatBytes(stream->deviceFormat[mode]) * device_channels);
- stream->bufferSize = *bufferSize;
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
+ snd_pcm_hw_params_dump(hw_params, out);
+#endif
- if (mode == INPUT && stream->mode == OUTPUT &&
- stream->device[0] == device) {
- // We're doing duplex setup here.
- stream->deviceFormat[0] = stream->deviceFormat[1];
- stream->nDeviceChannels[0] = device_channels;
+ // Allocate the stream handle if necessary and then save.
+ snd_pcm_t **handles;
+ if ( stream_.apiHandle == 0 ) {
+ handles = (snd_pcm_t **) calloc(2, sizeof(snd_pcm_t *));
+ if ( handle == NULL ) {
+ sprintf(message_, "RtApiAlsa: error allocating handle memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
+ stream_.apiHandle = (void *) handles;
+ handles[0] = 0;
+ handles[1] = 0;
+ }
+ else {
+ handles = (snd_pcm_t **) stream_.apiHandle;
}
+ handles[mode] = handle;
// Set flags for buffer conversion
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode])
- stream->doConvertBuffer[mode] = true;
+ stream_.doConvertBuffer[mode] = false;
+ if (stream_.userFormat != stream_.deviceFormat[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.nUserChannels[mode] > 1 && stream_.deInterleave[mode])
+ stream_.doConvertBuffer[mode] = true;
// Allocate necessary internal buffers
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
+ if ( stream_.nUserChannels[0] != stream_.nUserChannels[1] ) {
long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
+ if (stream_.nUserChannels[0] >= stream_.nUserChannels[1])
+ buffer_bytes = stream_.nUserChannels[0];
else
- buffer_bytes = stream->nUserChannels[1];
-
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL) {
- close(fd);
- sprintf(message, "RtAudio: OSS error allocating user buffer memory (%s).",
- name);
+ buffer_bytes = stream_.nUserChannels[1];
+
+ buffer_bytes *= *bufferSize * formatBytes(stream_.userFormat);
+ if (stream_.userBuffer) free(stream_.userBuffer);
+ stream_.userBuffer = (char *) calloc(buffer_bytes, 1);
+ if (stream_.userBuffer == NULL) {
+ sprintf(message_, "RtApiAlsa: error allocating user buffer memory (%s).",
+ devices_[device].name.c_str());
goto error;
}
}
- if ( stream->doConvertBuffer[mode] ) {
+ if ( stream_.doConvertBuffer[mode] ) {
long buffer_bytes;
bool makeBuffer = true;
if ( mode == OUTPUT )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
+ buffer_bytes = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
else { // mode == INPUT
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == OUTPUT && stream->deviceBuffer ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
+ buffer_bytes = stream_.nDeviceChannels[1] * formatBytes(stream_.deviceFormat[1]);
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ long bytes_out = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
if ( buffer_bytes < bytes_out ) makeBuffer = false;
}
}
if ( makeBuffer ) {
buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL) {
- close(fd);
- free(stream->userBuffer);
- sprintf(message, "RtAudio: OSS error allocating device buffer memory (%s).",
- name);
+ if (stream_.deviceBuffer) free(stream_.deviceBuffer);
+ stream_.deviceBuffer = (char *) calloc(buffer_bytes, 1);
+ if (stream_.deviceBuffer == NULL) {
+ sprintf(message_, "RtApiAlsa: error allocating device buffer memory (%s).",
+ devices_[device].name.c_str());
goto error;
}
}
}
- stream->device[mode] = device;
- stream->handle[mode] = fd;
- stream->state = STREAM_STOPPED;
- if ( stream->mode == OUTPUT && mode == INPUT ) {
- stream->mode = DUPLEX;
- if (stream->device[0] == device)
- stream->handle[0] = fd;
- }
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
else
- stream->mode = mode;
+ stream_.mode = mode;
+ stream_.nBuffers = periods;
+ stream_.sampleRate = sampleRate;
return SUCCESS;
error:
- if (stream->handle[0]) {
- close(stream->handle[0]);
- stream->handle[0] = 0;
+ if (handles) {
+ if (handles[0])
+ snd_pcm_close(handles[0]);
+ if (handles[1])
+ snd_pcm_close(handles[1]);
+ free(handles);
+ stream_.apiHandle = 0;
+ }
+
+ if (stream_.userBuffer) {
+ free(stream_.userBuffer);
+ stream_.userBuffer = 0;
}
+
error(RtError::WARNING);
return FAILURE;
}
-void RtAudio :: closeStream(int streamId)
+void RtApiAlsa :: closeStream()
{
// We don't want an exception to be thrown here because this
// function is called by our class destructor. So, do our own
- // streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
+ // stream check.
+ if ( stream_.mode == UNINITIALIZED ) {
+ sprintf(message_, "RtApiAlsa::closeStream(): no open stream to close!");
error(RtError::WARNING);
return;
}
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
-
- if (stream->callbackInfo.usingCallback) {
- pthread_cancel(stream->callbackInfo.thread);
- pthread_join(stream->callbackInfo.thread, NULL);
+ snd_pcm_t **handle = (snd_pcm_t **) stream_.apiHandle;
+ if (stream_.state == STREAM_RUNNING) {
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX)
+ snd_pcm_drop(handle[0]);
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX)
+ snd_pcm_drop(handle[1]);
+ stream_.state = STREAM_STOPPED;
}
- if (stream->state == STREAM_RUNNING) {
- if (stream->mode == OUTPUT || stream->mode == DUPLEX)
- ioctl(stream->handle[0], SNDCTL_DSP_RESET, 0);
- if (stream->mode == INPUT || stream->mode == DUPLEX)
- ioctl(stream->handle[1], SNDCTL_DSP_RESET, 0);
+ if (stream_.callbackInfo.usingCallback) {
+ stream_.callbackInfo.usingCallback = false;
+ pthread_join(stream_.callbackInfo.thread, NULL);
}
- pthread_mutex_destroy(&stream->mutex);
-
- if (stream->handle[0])
- close(stream->handle[0]);
-
- if (stream->handle[1])
- close(stream->handle[1]);
+ if (handle) {
+ if (handle[0]) snd_pcm_close(handle[0]);
+ if (handle[1]) snd_pcm_close(handle[1]);
+ free(handle);
+ handle = 0;
+ }
- if (stream->userBuffer)
- free(stream->userBuffer);
+ if (stream_.userBuffer) {
+ free(stream_.userBuffer);
+ stream_.userBuffer = 0;
+ }
- if (stream->deviceBuffer)
- free(stream->deviceBuffer);
+ if (stream_.deviceBuffer) {
+ free(stream_.deviceBuffer);
+ stream_.deviceBuffer = 0;
+ }
- free(stream);
- streams.erase(streamId);
+ stream_.mode = UNINITIALIZED;
}
-void RtAudio :: startStream(int streamId)
+void RtApiAlsa :: startStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ // This method calls snd_pcm_prepare if the device isn't already in that state.
- MUTEX_LOCK(&stream->mutex);
+ verifyStream();
+ if (stream_.state == STREAM_RUNNING) return;
- stream->state = STREAM_RUNNING;
+ MUTEX_LOCK(&stream_.mutex);
- // No need to do anything else here ... OSS automatically starts
- // when fed samples.
+ int err;
+ snd_pcm_state_t state;
+ snd_pcm_t **handle = (snd_pcm_t **) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ state = snd_pcm_state(handle[0]);
+ if (state != SND_PCM_STATE_PREPARED) {
+ err = snd_pcm_prepare(handle[0]);
+ if (err < 0) {
+ sprintf(message_, "RtApiAlsa: error preparing pcm device (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), snd_strerror(err));
+ MUTEX_UNLOCK(&stream_.mutex);
+ error(RtError::DRIVER_ERROR);
+ }
+ }
+ }
+
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ state = snd_pcm_state(handle[1]);
+ if (state != SND_PCM_STATE_PREPARED) {
+ err = snd_pcm_prepare(handle[1]);
+ if (err < 0) {
+ sprintf(message_, "RtApiAlsa: error preparing pcm device (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), snd_strerror(err));
+ MUTEX_UNLOCK(&stream_.mutex);
+ error(RtError::DRIVER_ERROR);
+ }
+ }
+ }
+ stream_.state = STREAM_RUNNING;
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
}
-void RtAudio :: stopStream(int streamId)
+void RtApiAlsa :: stopStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+ if (stream_.state == STREAM_STOPPED) return;
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ // Change the state before the lock to improve shutdown response
+ // when using a callback.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK(&stream_.mutex);
int err;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- err = ioctl(stream->handle[0], SNDCTL_DSP_SYNC, 0);
- if (err < -1) {
- sprintf(message, "RtAudio: OSS error stopping device (%s).",
- devices[stream->device[0]].name);
+ snd_pcm_t **handle = (snd_pcm_t **) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ err = snd_pcm_drain(handle[0]);
+ if (err < 0) {
+ sprintf(message_, "RtApiAlsa: error draining pcm device (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), snd_strerror(err));
+ MUTEX_UNLOCK(&stream_.mutex);
error(RtError::DRIVER_ERROR);
}
}
- else {
- err = ioctl(stream->handle[1], SNDCTL_DSP_SYNC, 0);
- if (err < -1) {
- sprintf(message, "RtAudio: OSS error stopping device (%s).",
- devices[stream->device[1]].name);
+
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ err = snd_pcm_drain(handle[1]);
+ if (err < 0) {
+ sprintf(message_, "RtApiAlsa: error draining pcm device (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), snd_strerror(err));
+ MUTEX_UNLOCK(&stream_.mutex);
error(RtError::DRIVER_ERROR);
}
}
- stream->state = STREAM_STOPPED;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
}
-void RtAudio :: abortStream(int streamId)
+void RtApiAlsa :: abortStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
+ verifyStream();
+ if (stream_.state == STREAM_STOPPED) return;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ // Change the state before the lock to improve shutdown response
+ // when using a callback.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK(&stream_.mutex);
int err;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- err = ioctl(stream->handle[0], SNDCTL_DSP_RESET, 0);
- if (err < -1) {
- sprintf(message, "RtAudio: OSS error aborting device (%s).",
- devices[stream->device[0]].name);
+ snd_pcm_t **handle = (snd_pcm_t **) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ err = snd_pcm_drop(handle[0]);
+ if (err < 0) {
+ sprintf(message_, "RtApiAlsa: error draining pcm device (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), snd_strerror(err));
+ MUTEX_UNLOCK(&stream_.mutex);
error(RtError::DRIVER_ERROR);
}
}
- else {
- err = ioctl(stream->handle[1], SNDCTL_DSP_RESET, 0);
- if (err < -1) {
- sprintf(message, "RtAudio: OSS error aborting device (%s).",
- devices[stream->device[1]].name);
+
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ err = snd_pcm_drop(handle[1]);
+ if (err < 0) {
+ sprintf(message_, "RtApiAlsa: error draining pcm device (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), snd_strerror(err));
+ MUTEX_UNLOCK(&stream_.mutex);
error(RtError::DRIVER_ERROR);
}
}
- stream->state = STREAM_STOPPED;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
}
-int RtAudio :: streamWillBlock(int streamId)
+int RtApiAlsa :: streamWillBlock()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+ if (stream_.state == STREAM_STOPPED) return 0;
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK(&stream_.mutex);
- int bytes = 0, channels = 0, frames = 0;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- audio_buf_info info;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- ioctl(stream->handle[0], SNDCTL_DSP_GETOSPACE, &info);
- bytes = info.bytes;
- channels = stream->nDeviceChannels[0];
+ int err = 0, frames = 0;
+ snd_pcm_t **handle = (snd_pcm_t **) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ err = snd_pcm_avail_update(handle[0]);
+ if (err < 0) {
+ sprintf(message_, "RtApiAlsa: error getting available frames for device (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), snd_strerror(err));
+ MUTEX_UNLOCK(&stream_.mutex);
+ error(RtError::DRIVER_ERROR);
+ }
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- ioctl(stream->handle[1], SNDCTL_DSP_GETISPACE, &info);
- if (stream->mode == DUPLEX ) {
- bytes = (bytes < info.bytes) ? bytes : info.bytes;
- channels = stream->nDeviceChannels[0];
- }
- else {
- bytes = info.bytes;
- channels = stream->nDeviceChannels[1];
+ frames = err;
+
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ err = snd_pcm_avail_update(handle[1]);
+ if (err < 0) {
+ sprintf(message_, "RtApiAlsa: error getting available frames for device (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), snd_strerror(err));
+ MUTEX_UNLOCK(&stream_.mutex);
+ error(RtError::DRIVER_ERROR);
}
+ if (frames > err) frames = err;
}
- frames = (int) (bytes / (channels * formatBytes(stream->deviceFormat[0])));
- frames -= stream->bufferSize;
+ frames = stream_.bufferSize - frames;
if (frames < 0) frames = 0;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
return frames;
}
-void RtAudio :: tickStream(int streamId)
+void RtApiAlsa :: tickStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
int stopStream = 0;
- if (stream->state == STREAM_STOPPED) {
- if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds
+ if (stream_.state == STREAM_STOPPED) {
+ if (stream_.callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds
return;
}
- else if (stream->callbackInfo.usingCallback) {
- RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback;
- stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData);
+ else if (stream_.callbackInfo.usingCallback) {
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ stopStream = callback(stream_.userBuffer, stream_.bufferSize, stream_.callbackInfo.userData);
}
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK(&stream_.mutex);
// The state might change while waiting on a mutex.
- if (stream->state == STREAM_STOPPED)
+ if (stream_.state == STREAM_STOPPED)
goto unlock;
- int result;
+ int err;
char *buffer;
- int samples;
- RTAUDIO_FORMAT format;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ int channels;
+ snd_pcm_t **handle;
+ RtAudioFormat format;
+ handle = (snd_pcm_t **) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
// Setup parameters and do buffer conversion if necessary.
- if (stream->doConvertBuffer[0]) {
- convertStreamBuffer(stream, OUTPUT);
- buffer = stream->deviceBuffer;
- samples = stream->bufferSize * stream->nDeviceChannels[0];
- format = stream->deviceFormat[0];
+ if (stream_.doConvertBuffer[0]) {
+ convertStreamBuffer(OUTPUT);
+ buffer = stream_.deviceBuffer;
+ channels = stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
}
else {
- buffer = stream->userBuffer;
- samples = stream->bufferSize * stream->nUserChannels[0];
- format = stream->userFormat;
+ buffer = stream_.userBuffer;
+ channels = stream_.nUserChannels[0];
+ format = stream_.userFormat;
}
// Do byte swapping if necessary.
- if (stream->doByteSwap[0])
- byteSwapBuffer(buffer, samples, format);
+ if (stream_.doByteSwap[0])
+ byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
- // Write samples to device.
- result = write(stream->handle[0], buffer, samples * formatBytes(format));
+ // Write samples to device in interleaved/non-interleaved format.
+ if (stream_.deInterleave[0]) {
+ void *bufs[channels];
+ size_t offset = stream_.bufferSize * formatBytes(format);
+ for (int i=0; i<channels; i++)
+ bufs[i] = (void *) (buffer + (i * offset));
+ err = snd_pcm_writen(handle[0], bufs, stream_.bufferSize);
+ }
+ else
+ err = snd_pcm_writei(handle[0], buffer, stream_.bufferSize);
- if (result == -1) {
- // This could be an underrun, but the basic OSS API doesn't provide a means for determining that.
- sprintf(message, "RtAudio: OSS audio write error for device (%s).",
- devices[stream->device[0]].name);
- error(RtError::DRIVER_ERROR);
+ if (err < stream_.bufferSize) {
+ // Either an error or underrun occured.
+ if (err == -EPIPE) {
+ snd_pcm_state_t state = snd_pcm_state(handle[0]);
+ if (state == SND_PCM_STATE_XRUN) {
+ sprintf(message_, "RtApiAlsa: underrun detected.");
+ error(RtError::WARNING);
+ err = snd_pcm_prepare(handle[0]);
+ if (err < 0) {
+ sprintf(message_, "RtApiAlsa: error preparing handle after underrun: %s.",
+ snd_strerror(err));
+ MUTEX_UNLOCK(&stream_.mutex);
+ error(RtError::DRIVER_ERROR);
+ }
+ }
+ else {
+ sprintf(message_, "RtApiAlsa: tickStream() error, current state is %s.",
+ snd_pcm_state_name(state));
+ MUTEX_UNLOCK(&stream_.mutex);
+ error(RtError::DRIVER_ERROR);
+ }
+ goto unlock;
+ }
+ else {
+ sprintf(message_, "RtApiAlsa: audio write error for device (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), snd_strerror(err));
+ MUTEX_UNLOCK(&stream_.mutex);
+ error(RtError::DRIVER_ERROR);
+ }
}
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
// Setup parameters.
- if (stream->doConvertBuffer[1]) {
- buffer = stream->deviceBuffer;
- samples = stream->bufferSize * stream->nDeviceChannels[1];
- format = stream->deviceFormat[1];
+ if (stream_.doConvertBuffer[1]) {
+ buffer = stream_.deviceBuffer;
+ channels = stream_.nDeviceChannels[1];
+ format = stream_.deviceFormat[1];
}
else {
- buffer = stream->userBuffer;
- samples = stream->bufferSize * stream->nUserChannels[1];
- format = stream->userFormat;
+ buffer = stream_.userBuffer;
+ channels = stream_.nUserChannels[1];
+ format = stream_.userFormat;
}
- // Read samples from device.
- result = read(stream->handle[1], buffer, samples * formatBytes(format));
+ // Read samples from device in interleaved/non-interleaved format.
+ if (stream_.deInterleave[1]) {
+ void *bufs[channels];
+ size_t offset = stream_.bufferSize * formatBytes(format);
+ for (int i=0; i<channels; i++)
+ bufs[i] = (void *) (buffer + (i * offset));
+ err = snd_pcm_readn(handle[1], bufs, stream_.bufferSize);
+ }
+ else
+ err = snd_pcm_readi(handle[1], buffer, stream_.bufferSize);
- if (result == -1) {
- // This could be an overrun, but the basic OSS API doesn't provide a means for determining that.
- sprintf(message, "RtAudio: OSS audio read error for device (%s).",
- devices[stream->device[1]].name);
- error(RtError::DRIVER_ERROR);
+ if (err < stream_.bufferSize) {
+ // Either an error or underrun occured.
+ if (err == -EPIPE) {
+ snd_pcm_state_t state = snd_pcm_state(handle[1]);
+ if (state == SND_PCM_STATE_XRUN) {
+ sprintf(message_, "RtApiAlsa: overrun detected.");
+ error(RtError::WARNING);
+ err = snd_pcm_prepare(handle[1]);
+ if (err < 0) {
+ sprintf(message_, "RtApiAlsa: error preparing handle after overrun: %s.",
+ snd_strerror(err));
+ MUTEX_UNLOCK(&stream_.mutex);
+ error(RtError::DRIVER_ERROR);
+ }
+ }
+ else {
+ sprintf(message_, "RtApiAlsa: tickStream() error, current state is %s.",
+ snd_pcm_state_name(state));
+ MUTEX_UNLOCK(&stream_.mutex);
+ error(RtError::DRIVER_ERROR);
+ }
+ goto unlock;
+ }
+ else {
+ sprintf(message_, "RtApiAlsa: audio read error for device (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), snd_strerror(err));
+ MUTEX_UNLOCK(&stream_.mutex);
+ error(RtError::DRIVER_ERROR);
+ }
}
// Do byte swapping if necessary.
- if (stream->doByteSwap[1])
- byteSwapBuffer(buffer, samples, format);
+ if (stream_.doByteSwap[1])
+ byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
// Do buffer conversion if necessary.
- if (stream->doConvertBuffer[1])
- convertStreamBuffer(stream, INPUT);
+ if (stream_.doConvertBuffer[1])
+ convertStreamBuffer(INPUT);
}
unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
- if (stream->callbackInfo.usingCallback && stopStream)
- this->stopStream(streamId);
+ if (stream_.callbackInfo.usingCallback && stopStream)
+ this->stopStream();
}
-extern "C" void *callbackHandler(void *ptr)
+void RtApiAlsa :: setStreamCallback(RtAudioCallback callback, void *userData)
+{
+ verifyStream();
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ if ( info->usingCallback ) {
+ sprintf(message_, "RtApiAlsa: A callback is already set for this stream!");
+ error(RtError::WARNING);
+ return;
+ }
+
+ info->callback = (void *) callback;
+ info->userData = userData;
+ info->usingCallback = true;
+ info->object = (void *) this;
+
+ // Set the thread attributes for joinable and realtime scheduling
+ // priority. The higher priority will only take affect if the
+ // program is run as root or suid.
+ pthread_attr_t attr;
+ pthread_attr_init(&attr);
+ pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE);
+ pthread_attr_setschedpolicy(&attr, SCHED_RR);
+
+ int err = pthread_create(&info->thread, &attr, alsaCallbackHandler, &stream_.callbackInfo);
+ pthread_attr_destroy(&attr);
+ if (err) {
+ info->usingCallback = false;
+ sprintf(message_, "RtApiAlsa: error starting callback thread!");
+ error(RtError::THREAD_ERROR);
+ }
+}
+
+void RtApiAlsa :: cancelStreamCallback()
+{
+ verifyStream();
+
+ if (stream_.callbackInfo.usingCallback) {
+
+ if (stream_.state == STREAM_RUNNING)
+ stopStream();
+
+ MUTEX_LOCK(&stream_.mutex);
+
+ stream_.callbackInfo.usingCallback = false;
+ pthread_join(stream_.callbackInfo.thread, NULL);
+ stream_.callbackInfo.thread = 0;
+ stream_.callbackInfo.callback = NULL;
+ stream_.callbackInfo.userData = NULL;
+
+ MUTEX_UNLOCK(&stream_.mutex);
+ }
+}
+
+extern "C" void *alsaCallbackHandler(void *ptr)
{
- CALLBACK_INFO *info = (CALLBACK_INFO *) ptr;
- RtAudio *object = (RtAudio *) info->object;
- int stream = info->streamId;
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiAlsa *object = (RtApiAlsa *) info->object;
bool *usingCallback = &info->usingCallback;
while ( *usingCallback ) {
- pthread_testcancel();
try {
- object->tickStream(stream);
+ object->tickStream();
}
catch (RtError &exception) {
- fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n",
- exception.getMessage());
+ fprintf(stderr, "\nRtApiAlsa: callback thread error (%s) ... closing thread.\n\n",
+ exception.getMessageString());
break;
}
}
- return 0;
+ pthread_exit(NULL);
}
+//******************** End of __LINUX_ALSA__ *********************//
+#endif
-//******************** End of __LINUX_OSS__ *********************//
-
-#elif defined(__WINDOWS_ASIO__) // ASIO API on Windows
+#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
// The ASIO API is designed around a callback scheme, so this
-// implementation is similar to that used for OS X CoreAudio. The
-// primary constraint with ASIO is that it only allows access to a
-// single driver at a time. Thus, it is not possible to have more
-// than one simultaneous RtAudio stream.
+// implementation is similar to that used for OS-X CoreAudio and Linux
+// Jack. The primary constraint with ASIO is that it only allows
+// access to a single driver at a time. Thus, it is not possible to
+// have more than one simultaneous RtAudio stream.
//
// This implementation also requires a number of external ASIO files
// and a few global variables. The ASIO callback scheme does not
// allow for the passing of user data, so we must create a global
// pointer to our callbackInfo structure.
+//
+// On unix systems, we make use of a pthread condition variable.
+// Since there is no equivalent in Windows, I hacked something based
+// on information found in
+// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
#include "asio/asiosys.h"
#include "asio/asio.h"
AsioDrivers drivers;
ASIOCallbacks asioCallbacks;
-CALLBACK_INFO *asioCallbackInfo;
ASIODriverInfo driverInfo;
+CallbackInfo *asioCallbackInfo;
+
+struct AsioHandle {
+ bool stopStream;
+ ASIOBufferInfo *bufferInfos;
+ HANDLE condition;
+
+ AsioHandle()
+ :stopStream(false), bufferInfos(0) {}
+};
-void RtAudio :: initialize(void)
+RtApiAsio :: RtApiAsio()
{
- nDevices = drivers.asioGetNumDev();
- if (nDevices <= 0) return;
+ this->initialize();
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
+ if (nDevices_ <= 0) {
+ sprintf(message_, "RtApiAsio: no Windows ASIO audio drivers found!");
+ error(RtError::NO_DEVICES_FOUND);
}
+}
- // Write device driver names to device structures and then probe the
- // device capabilities.
- for (int i=0; i<nDevices; i++) {
- if ( drivers.asioGetDriverName( i, devices[i].name, 128 ) == 0 )
- //probeDeviceInfo(&devices[i]);
- ;
+RtApiAsio :: ~RtApiAsio()
+{
+ if ( stream_.mode != UNINITIALIZED ) closeStream();
+}
+
+void RtApiAsio :: initialize(void)
+{
+ nDevices_ = drivers.asioGetNumDev();
+ if (nDevices_ <= 0) return;
+
+ // Create device structures and write device driver names to each.
+ RtApiDevice device;
+ char name[128];
+ for (int i=0; i<nDevices_; i++) {
+ if ( drivers.asioGetDriverName( i, name, 128 ) == 0 ) {
+ device.name.erase();
+ device.name.append( (const char *)name, strlen(name)+1);
+ devices_.push_back(device);
+ }
else {
- sprintf(message, "RtAudio: error getting ASIO driver name for device index %d!", i);
+ sprintf(message_, "RtApiAsio: error getting driver name for device index %d!", i);
error(RtError::WARNING);
}
}
+ nDevices_ = (int) devices_.size();
+
drivers.removeCurrentDriver();
driverInfo.asioVersion = 2;
// See note in DirectSound implementation about GetDesktopWindow().
driverInfo.sysRef = GetForegroundWindow();
}
-int RtAudio :: getDefaultInputDevice(void)
-{
- return 0;
-}
-
-int RtAudio :: getDefaultOutputDevice(void)
-{
- return 0;
-}
-
-void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info)
+void RtApiAsio :: probeDeviceInfo(RtApiDevice *info)
{
// Don't probe if a stream is already open.
- if ( streams.size() > 0 ) {
- sprintf(message, "RtAudio: unable to probe ASIO driver while a stream is open.");
+ if ( stream_.mode != UNINITIALIZED ) {
+ sprintf(message_, "RtApiAsio: unable to probe driver while a stream is open.");
error(RtError::DEBUG_WARNING);
return;
}
- if ( !drivers.loadDriver( info->name ) ) {
- sprintf(message, "RtAudio: ASIO error loading driver (%s).", info->name);
+ if ( !drivers.loadDriver( (char *)info->name.c_str() ) ) {
+ sprintf(message_, "RtApiAsio: error loading driver (%s).", info->name.c_str());
error(RtError::DEBUG_WARNING);
return;
}
sprintf(details, "driver/hardware not present");
else
sprintf(details, "unspecified");
- sprintf(message, "RtAudio: ASIO error (%s) initializing driver (%s).", details, info->name);
+ sprintf(message_, "RtApiAsio: error (%s) initializing driver (%s).", details, info->name.c_str());
error(RtError::DEBUG_WARNING);
return;
}
result = ASIOGetChannels( &inputChannels, &outputChannels );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO error getting input/output channel count (%s).", info->name);
+ sprintf(message_, "RtApiAsio: error getting input/output channel count (%s).", info->name.c_str());
error(RtError::DEBUG_WARNING);
return;
}
}
// Determine the supported sample rates.
- info->nSampleRates = 0;
- for (int i=0; i<MAX_SAMPLE_RATES; i++) {
+ info->sampleRates.clear();
+ for (unsigned int i=0; i<MAX_SAMPLE_RATES; i++) {
result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
if ( result == ASE_OK )
- info->sampleRates[info->nSampleRates++] = SAMPLE_RATES[i];
+ info->sampleRates.push_back( SAMPLE_RATES[i] );
}
- if (info->nSampleRates == 0) {
+ if (info->sampleRates.size() == 0) {
drivers.removeCurrentDriver();
- sprintf( message, "RtAudio: No supported sample rates found for ASIO driver (%s).", info->name );
+ sprintf( message_, "RtApiAsio: No supported sample rates found for driver (%s).", info->name.c_str() );
error(RtError::DEBUG_WARNING);
return;
}
result = ASIOGetChannelInfo( &channelInfo );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO error getting driver (%s) channel information.", info->name);
+ sprintf(message_, "RtApiAsio: error getting driver (%s) channel information.", info->name.c_str());
error(RtError::DEBUG_WARNING);
return;
}
// Check that we have at least one supported format.
if (info->nativeFormats == 0) {
drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) data format not supported by RtAudio.",
- info->name);
+ sprintf(message_, "RtApiAsio: driver (%s) data format not supported by RtAudio.",
+ info->name.c_str());
error(RtError::DEBUG_WARNING);
return;
}
void bufferSwitch(long index, ASIOBool processNow)
{
- RtAudio *object = (RtAudio *) asioCallbackInfo->object;
+ RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
try {
- object->callbackEvent( asioCallbackInfo->streamId, index, (void *)NULL, (void *)NULL );
+ object->callbackEvent( index );
}
catch (RtError &exception) {
- fprintf(stderr, "\nCallback handler error (%s)!\n\n", exception.getMessage());
+ fprintf(stderr, "\nRtApiAsio: callback handler error (%s)!\n\n", exception.getMessageString());
return;
}
RtAudio *object = (RtAudio *) asioCallbackInfo->object;
try {
- object->stopStream( asioCallbackInfo->streamId );
+ object->stopStream();
}
catch (RtError &exception) {
- fprintf(stderr, "\nRtAudio: sampleRateChanged() error (%s)!\n\n", exception.getMessage());
+ fprintf(stderr, "\nRtApiAsio: sampleRateChanged() error (%s)!\n\n", exception.getMessageString());
return;
}
- fprintf(stderr, "\nRtAudio: ASIO driver reports sample rate changed to %d ... stream stopped!!!", (int) sRate);
+ fprintf(stderr, "\nRtApiAsio: driver reports sample rate changed to %d ... stream stopped!!!", (int) sRate);
}
long asioMessages(long selector, long value, void* message, double* opt)
// done by completely destruct is. I.e. ASIOStop(),
// ASIODisposeBuffers(), Destruction Afterwards you initialize the
// driver again.
- fprintf(stderr, "\nRtAudio: ASIO driver reset requested!!!");
+ fprintf(stderr, "\nRtApiAsio: driver reset requested!!!");
ret = 1L;
break;
case kAsioResyncRequest:
// which could lose data because the Mutex was held too long by
// another thread. However a driver can issue it in other
// situations, too.
- fprintf(stderr, "\nRtAudio: ASIO driver resync requested!!!");
+ fprintf(stderr, "\nRtApiAsio: driver resync requested!!!");
ret = 1L;
break;
case kAsioLatenciesChanged:
// latencies changed. Beware, it this does not mean that the
// buffer sizes have changed! You might need to update internal
// delay data.
- fprintf(stderr, "\nRtAudio: ASIO driver latency may have changed!!!");
+ fprintf(stderr, "\nRtApiAsio: driver latency may have changed!!!");
ret = 1L;
break;
case kAsioEngineVersion:
return ret;
}
-bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
- STREAM_MODE mode, int channels,
- int sampleRate, RTAUDIO_FORMAT format,
- int *bufferSize, int numberOfBuffers)
+bool RtApiAsio :: probeDeviceOpen(int device, StreamMode mode, int channels,
+ int sampleRate, RtAudioFormat format,
+ int *bufferSize, int numberOfBuffers)
{
- // Don't attempt to load another driver if a stream is already open.
- if ( streams.size() > 0 ) {
- sprintf(message, "RtAudio: unable to load ASIO driver while a stream is open.");
- error(RtError::WARNING);
- return FAILURE;
- }
-
// For ASIO, a duplex stream MUST use the same driver.
- if ( mode == INPUT && stream->mode == OUTPUT && stream->device[0] != device ) {
- sprintf(message, "RtAudio: ASIO duplex stream must use the same device for input and output.");
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
+ sprintf(message_, "RtApiAsio: duplex stream must use the same device for input and output.");
error(RtError::WARNING);
return FAILURE;
}
// Only load the driver once for duplex stream.
ASIOError result;
- if ( mode != INPUT || stream->mode != OUTPUT ) {
- if ( !drivers.loadDriver( devices[device].name ) ) {
- sprintf(message, "RtAudio: ASIO error loading driver (%s).", devices[device].name);
+ if ( mode != INPUT || stream_.mode != OUTPUT ) {
+ if ( !drivers.loadDriver( (char *)devices_[device].name.c_str() ) ) {
+ sprintf(message_, "RtApiAsio: error loading driver (%s).", devices_[device].name.c_str());
error(RtError::DEBUG_WARNING);
return FAILURE;
}
sprintf(details, "driver/hardware not present");
else
sprintf(details, "unspecified");
- sprintf(message, "RtAudio: ASIO error (%s) initializing driver (%s).", details, devices[device].name);
+ sprintf(message_, "RtApiAsio: error (%s) initializing driver (%s).", details, devices_[device].name.c_str());
error(RtError::DEBUG_WARNING);
return FAILURE;
}
result = ASIOGetChannels( &inputChannels, &outputChannels );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO error getting input/output channel count (%s).",
- devices[device].name);
+ sprintf(message_, "RtApiAsio: error getting input/output channel count (%s).",
+ devices_[device].name.c_str());
error(RtError::DEBUG_WARNING);
return FAILURE;
}
if ( ( mode == OUTPUT && channels > outputChannels) ||
( mode == INPUT && channels > inputChannels) ) {
drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) does not support requested channel count (%d).",
- devices[device].name, channels);
+ sprintf(message_, "RtApiAsio: driver (%s) does not support requested channel count (%d).",
+ devices_[device].name.c_str(), channels);
error(RtError::DEBUG_WARNING);
return FAILURE;
}
- stream->nDeviceChannels[mode] = channels;
- stream->nUserChannels[mode] = channels;
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
// Verify the sample rate is supported.
result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) does not support requested sample rate (%d).",
- devices[device].name, sampleRate);
+ sprintf(message_, "RtApiAsio: driver (%s) does not support requested sample rate (%d).",
+ devices_[device].name.c_str(), sampleRate);
error(RtError::DEBUG_WARNING);
return FAILURE;
}
result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) error setting sample rate (%d).",
- devices[device].name, sampleRate);
+ sprintf(message_, "RtApiAsio: driver (%s) error setting sample rate (%d).",
+ devices_[device].name.c_str(), sampleRate);
error(RtError::DEBUG_WARNING);
return FAILURE;
}
result = ASIOGetChannelInfo( &channelInfo );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) error getting data format.",
- devices[device].name);
+ sprintf(message_, "RtApiAsio: driver (%s) error getting data format.",
+ devices_[device].name.c_str());
error(RtError::DEBUG_WARNING);
return FAILURE;
}
// Assuming WINDOWS host is always little-endian.
- stream->doByteSwap[mode] = false;
- stream->userFormat = format;
- stream->deviceFormat[mode] = 0;
+ stream_.doByteSwap[mode] = false;
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = 0;
if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- if ( channelInfo.type == ASIOSTInt16MSB ) stream->doByteSwap[mode] = true;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
}
else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- if ( channelInfo.type == ASIOSTInt32MSB ) stream->doByteSwap[mode] = true;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
}
else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
- stream->deviceFormat[mode] = RTAUDIO_FLOAT32;
- if ( channelInfo.type == ASIOSTFloat32MSB ) stream->doByteSwap[mode] = true;
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
}
else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
- stream->deviceFormat[mode] = RTAUDIO_FLOAT64;
- if ( channelInfo.type == ASIOSTFloat64MSB ) stream->doByteSwap[mode] = true;
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+ if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
}
- if ( stream->deviceFormat[mode] == 0 ) {
+ if ( stream_.deviceFormat[mode] == 0 ) {
drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) data format not supported by RtAudio.",
- devices[device].name);
+ sprintf(message_, "RtApiAsio: driver (%s) data format not supported by RtAudio.",
+ devices_[device].name.c_str());
error(RtError::DEBUG_WARNING);
return FAILURE;
}
result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) error getting buffer size.",
- devices[device].name);
+ sprintf(message_, "RtApiAsio: driver (%s) error getting buffer size.",
+ devices_[device].name.c_str());
error(RtError::DEBUG_WARNING);
return FAILURE;
}
else if ( *bufferSize > maxSize ) *bufferSize = maxSize;
else if ( granularity == -1 ) {
// Make sure bufferSize is a power of two.
- double power = log10( *bufferSize ) / log10( 2.0 );
- *bufferSize = pow( 2.0, floor(power+0.5) );
+ double power = log10( (double) *bufferSize ) / log10( 2.0 );
+ *bufferSize = (int) pow( 2.0, floor(power+0.5) );
if ( *bufferSize < minSize ) *bufferSize = minSize;
else if ( *bufferSize > maxSize ) *bufferSize = maxSize;
else *bufferSize = preferSize;
}
- if ( mode == INPUT && stream->mode == OUTPUT && stream->bufferSize != *bufferSize )
- cout << "possible input/output buffersize discrepancy" << endl;
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize )
+ std::cerr << "Possible input/output buffersize discrepancy!" << std::endl;
- stream->bufferSize = *bufferSize;
- stream->nBuffers = 2;
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 2;
// ASIO always uses deinterleaved channels.
- stream->deInterleave[mode] = true;
+ stream_.deInterleave[mode] = true;
- // Create the ASIO internal buffers. Since RtAudio sets up input
- // and output separately, we'll have to dispose of previously
- // created output buffers for a duplex stream.
- if ( mode == INPUT && stream->mode == OUTPUT ) {
- free(stream->callbackInfo.buffers);
- result = ASIODisposeBuffers();
- if ( result != ASE_OK ) {
+ // Allocate, if necessary, our AsioHandle structure for the stream.
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( handle == 0 ) {
+ handle = (AsioHandle *) calloc(1, sizeof(AsioHandle));
+ if ( handle == NULL ) {
drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) error disposing previously allocated buffers.",
- devices[device].name);
+ sprintf(message_, "RtApiAsio: error allocating AsioHandle memory (%s).",
+ devices_[device].name.c_str());
error(RtError::DEBUG_WARNING);
return FAILURE;
}
+ handle->bufferInfos = 0;
+ // Create a manual-reset event.
+ handle->condition = CreateEvent(NULL, // no security
+ TRUE, // manual-reset
+ FALSE, // non-signaled initially
+ NULL); // unnamed
+ stream_.apiHandle = (void *) handle;
+ }
+
+ // Create the ASIO internal buffers. Since RtAudio sets up input
+ // and output separately, we'll have to dispose of previously
+ // created output buffers for a duplex stream.
+ if ( mode == INPUT && stream_.mode == OUTPUT ) {
+ ASIODisposeBuffers();
+ if ( handle->bufferInfos ) free( handle->bufferInfos );
}
// Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
- int i, nChannels = stream->nDeviceChannels[0] + stream->nDeviceChannels[1];
- stream->callbackInfo.buffers = 0;
- ASIOBufferInfo *bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
- stream->callbackInfo.buffers = (void *) bufferInfos;
- ASIOBufferInfo *infos = bufferInfos;
- for ( i=0; i<stream->nDeviceChannels[1]; i++, infos++ ) {
- infos->isInput = ASIOTrue;
+ int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+ handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
+ if (handle->bufferInfos == NULL) {
+ sprintf(message_, "RtApiAsio: error allocating bufferInfo memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
+ ASIOBufferInfo *infos;
+ infos = handle->bufferInfos;
+ for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
+ infos->isInput = ASIOFalse;
infos->channelNum = i;
infos->buffers[0] = infos->buffers[1] = 0;
}
-
- for ( i=0; i<stream->nDeviceChannels[0]; i++, infos++ ) {
- infos->isInput = ASIOFalse;
+ for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
+ infos->isInput = ASIOTrue;
infos->channelNum = i;
infos->buffers[0] = infos->buffers[1] = 0;
}
asioCallbacks.sampleRateDidChange = &sampleRateChanged;
asioCallbacks.asioMessage = &asioMessages;
asioCallbacks.bufferSwitchTimeInfo = NULL;
- result = ASIOCreateBuffers( bufferInfos, nChannels, stream->bufferSize, &asioCallbacks);
+ result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks);
if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) error creating buffers.",
- devices[device].name);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ sprintf(message_, "RtApiAsio: driver (%s) error creating buffers.",
+ devices_[device].name.c_str());
+ goto error;
}
// Set flags for buffer conversion.
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] > 1 && stream->deInterleave[mode])
- stream->doConvertBuffer[mode] = true;
+ stream_.doConvertBuffer[mode] = false;
+ if (stream_.userFormat != stream_.deviceFormat[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.nUserChannels[mode] > 1 && stream_.deInterleave[mode])
+ stream_.doConvertBuffer[mode] = true;
// Allocate necessary internal buffers
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
+ if ( stream_.nUserChannels[0] != stream_.nUserChannels[1] ) {
long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
+ if (stream_.nUserChannels[0] >= stream_.nUserChannels[1])
+ buffer_bytes = stream_.nUserChannels[0];
else
- buffer_bytes = stream->nUserChannels[1];
-
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL)
- goto memory_error;
+ buffer_bytes = stream_.nUserChannels[1];
+
+ buffer_bytes *= *bufferSize * formatBytes(stream_.userFormat);
+ if (stream_.userBuffer) free(stream_.userBuffer);
+ stream_.userBuffer = (char *) calloc(buffer_bytes, 1);
+ if (stream_.userBuffer == NULL) {
+ sprintf(message_, "RtApiAsio: error allocating user buffer memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
}
- if ( stream->doConvertBuffer[mode] ) {
+ if ( stream_.doConvertBuffer[mode] ) {
long buffer_bytes;
bool makeBuffer = true;
if ( mode == OUTPUT )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
+ buffer_bytes = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
else { // mode == INPUT
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == OUTPUT && stream->deviceBuffer ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
+ buffer_bytes = stream_.nDeviceChannels[1] * formatBytes(stream_.deviceFormat[1]);
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ long bytes_out = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
if ( buffer_bytes < bytes_out ) makeBuffer = false;
}
}
if ( makeBuffer ) {
buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL)
- goto memory_error;
+ if (stream_.deviceBuffer) free(stream_.deviceBuffer);
+ stream_.deviceBuffer = (char *) calloc(buffer_bytes, 1);
+ if (stream_.deviceBuffer == NULL) {
+ sprintf(message_, "RtApiAsio: error allocating device buffer memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
}
}
- stream->device[mode] = device;
- stream->state = STREAM_STOPPED;
- if ( stream->mode == OUTPUT && mode == INPUT )
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
// We had already set up an output stream.
- stream->mode = DUPLEX;
+ stream_.mode = DUPLEX;
else
- stream->mode = mode;
- stream->sampleRate = sampleRate;
- asioCallbackInfo = &stream->callbackInfo;
- stream->callbackInfo.object = (void *) this;
- stream->callbackInfo.waitTime = (unsigned long) (200.0 * stream->bufferSize / stream->sampleRate);
+ stream_.mode = mode;
+ stream_.sampleRate = sampleRate;
+ asioCallbackInfo = &stream_.callbackInfo;
+ stream_.callbackInfo.object = (void *) this;
return SUCCESS;
- memory_error:
+ error:
ASIODisposeBuffers();
drivers.removeCurrentDriver();
- if (stream->callbackInfo.buffers)
- free(stream->callbackInfo.buffers);
- stream->callbackInfo.buffers = 0;
+ if ( handle ) {
+ CloseHandle( handle->condition );
+ if ( handle->bufferInfos )
+ free( handle->bufferInfos );
+ free( handle );
+ stream_.apiHandle = 0;
+ }
- if (stream->userBuffer) {
- free(stream->userBuffer);
- stream->userBuffer = 0;
+ if (stream_.userBuffer) {
+ free(stream_.userBuffer);
+ stream_.userBuffer = 0;
}
- sprintf(message, "RtAudio: error allocating buffer memory (%s).",
- devices[device].name);
+
error(RtError::WARNING);
return FAILURE;
}
-void RtAudio :: cancelStreamCallback(int streamId)
+void RtApiAsio :: closeStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ // We don't want an exception to be thrown here because this
+ // function is called by our class destructor. So, do our own
+ // streamId check.
+ if ( stream_.mode == UNINITIALIZED ) {
+ sprintf(message_, "RtApiAsio::closeStream(): no open stream to close!");
+ error(RtError::WARNING);
+ return;
+ }
- if (stream->callbackInfo.usingCallback) {
+ if (stream_.state == STREAM_RUNNING)
+ ASIOStop();
- if (stream->state == STREAM_RUNNING)
- stopStream( streamId );
+ ASIODisposeBuffers();
+ drivers.removeCurrentDriver();
- MUTEX_LOCK(&stream->mutex);
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( handle ) {
+ CloseHandle( handle->condition );
+ if ( handle->bufferInfos )
+ free( handle->bufferInfos );
+ free( handle );
+ stream_.apiHandle = 0;
+ }
- stream->callbackInfo.usingCallback = false;
- stream->callbackInfo.userData = NULL;
- stream->state = STREAM_STOPPED;
- stream->callbackInfo.callback = NULL;
+ if (stream_.userBuffer) {
+ free(stream_.userBuffer);
+ stream_.userBuffer = 0;
+ }
- MUTEX_UNLOCK(&stream->mutex);
+ if (stream_.deviceBuffer) {
+ free(stream_.deviceBuffer);
+ stream_.deviceBuffer = 0;
}
+
+ stream_.mode = UNINITIALIZED;
}
-void RtAudio :: closeStream(int streamId)
+void RtApiAsio :: setStreamCallback(RtAudioCallback callback, void *userData)
{
- // We don't want an exception to be thrown here because this
- // function is called by our class destructor. So, do our own
- // streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
+ verifyStream();
+
+ if ( stream_.callbackInfo.usingCallback ) {
+ sprintf(message_, "RtApiAsio: A callback is already set for this stream!");
error(RtError::WARNING);
return;
}
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
-
- if (stream->state == STREAM_RUNNING)
- ASIOStop();
+ stream_.callbackInfo.callback = (void *) callback;
+ stream_.callbackInfo.userData = userData;
+ stream_.callbackInfo.usingCallback = true;
+}
- ASIODisposeBuffers();
- //ASIOExit();
- drivers.removeCurrentDriver();
+void RtApiAsio :: cancelStreamCallback()
+{
+ verifyStream();
- DeleteCriticalSection(&stream->mutex);
+ if (stream_.callbackInfo.usingCallback) {
- if (stream->callbackInfo.buffers)
- free(stream->callbackInfo.buffers);
+ if (stream_.state == STREAM_RUNNING)
+ stopStream();
- if (stream->userBuffer)
- free(stream->userBuffer);
+ MUTEX_LOCK(&stream_.mutex);
- if (stream->deviceBuffer)
- free(stream->deviceBuffer);
+ stream_.callbackInfo.usingCallback = false;
+ stream_.callbackInfo.userData = NULL;
+ stream_.state = STREAM_STOPPED;
+ stream_.callbackInfo.callback = NULL;
- free(stream);
- streams.erase(streamId);
+ MUTEX_UNLOCK(&stream_.mutex);
+ }
}
-void RtAudio :: startStream(int streamId)
+void RtApiAsio :: startStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
+ verifyStream();
+ if (stream_.state == STREAM_RUNNING) return;
- if (stream->state == STREAM_RUNNING) {
- MUTEX_UNLOCK(&stream->mutex);
- return;
- }
+ MUTEX_LOCK(&stream_.mutex);
- stream->callbackInfo.blockTick = true;
- stream->callbackInfo.stopStream = false;
- stream->callbackInfo.streamId = streamId;
ASIOError result = ASIOStart();
if ( result != ASE_OK ) {
- sprintf(message, "RtAudio: ASIO error starting device (%s).",
- devices[stream->device[0]].name);
- MUTEX_UNLOCK(&stream->mutex);
+ sprintf(message_, "RtApiAsio: error starting device (%s).",
+ devices_[stream_.device[0]].name.c_str());
+ MUTEX_UNLOCK(&stream_.mutex);
error(RtError::DRIVER_ERROR);
}
- stream->state = STREAM_RUNNING;
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ handle->stopStream = false;
+ stream_.state = STREAM_RUNNING;
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
}
-void RtAudio :: stopStream(int streamId)
+void RtApiAsio :: stopStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+ if (stream_.state == STREAM_STOPPED) return;
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED) {
- MUTEX_UNLOCK(&stream->mutex);
- return;
- }
+ // Change the state before the lock to improve shutdown response
+ // when using a callback.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK(&stream_.mutex);
ASIOError result = ASIOStop();
if ( result != ASE_OK ) {
- sprintf(message, "RtAudio: ASIO error stopping device (%s).",
- devices[stream->device[0]].name);
- MUTEX_UNLOCK(&stream->mutex);
+ sprintf(message_, "RtApiAsio: error stopping device (%s).",
+ devices_[stream_.device[0]].name.c_str());
+ MUTEX_UNLOCK(&stream_.mutex);
error(RtError::DRIVER_ERROR);
}
- stream->state = STREAM_STOPPED;
-
- MUTEX_UNLOCK(&stream->mutex);
-}
-void RtAudio :: abortStream(int streamId)
-{
- stopStream( streamId );
+ MUTEX_UNLOCK(&stream_.mutex);
}
-// I don't know how this function can be implemented.
-int RtAudio :: streamWillBlock(int streamId)
+void RtApiAsio :: abortStream()
{
- sprintf(message, "RtAudio: streamWillBlock() cannot be implemented for ASIO.");
- error(RtError::WARNING);
- return 0;
+ stopStream();
}
-void RtAudio :: tickStream(int streamId)
+void RtApiAsio :: tickStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
- if (stream->state == STREAM_STOPPED)
+ if (stream_.state == STREAM_STOPPED)
return;
- if (stream->callbackInfo.usingCallback) {
- sprintf(message, "RtAudio: tickStream() should not be used when a callback function is set!");
+ if (stream_.callbackInfo.usingCallback) {
+ sprintf(message_, "RtApiAsio: tickStream() should not be used when a callback function is set!");
error(RtError::WARNING);
return;
}
- // Block waiting here until the user data is processed in callbackEvent().
- while ( stream->callbackInfo.blockTick )
- Sleep(stream->callbackInfo.waitTime);
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK(&stream_.mutex);
- stream->callbackInfo.blockTick = true;
-
- MUTEX_UNLOCK(&stream->mutex);
+ // Release the stream_mutex here and wait for the event
+ // to become signaled by the callback process.
+ MUTEX_UNLOCK(&stream_.mutex);
+ WaitForMultipleObjects(1, &handle->condition, FALSE, INFINITE);
+ ResetEvent( handle->condition );
}
-void RtAudio :: callbackEvent(int streamId, int bufferIndex, void *inData, void *outData)
+void RtApiAsio :: callbackEvent(long bufferIndex)
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+
+ if (stream_.state == STREAM_STOPPED) return;
- CALLBACK_INFO *info = asioCallbackInfo;
- if ( !info->usingCallback ) {
- // Block waiting here until we get new user data in tickStream().
- while ( !info->blockTick )
- Sleep(info->waitTime);
- }
- else if ( info->stopStream ) {
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( info->usingCallback && handle->stopStream ) {
// Check if the stream should be stopped (via the previous user
// callback return value). We stop the stream here, rather than
// after the function call, so that output data can first be
// processed.
- this->stopStream(asioCallbackInfo->streamId);
+ this->stopStream();
return;
}
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK(&stream_.mutex);
// Invoke user callback first, to get fresh output data.
if ( info->usingCallback ) {
- RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) info->callback;
- if ( callback(stream->userBuffer, stream->bufferSize, info->userData) )
- info->stopStream = true;
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ if ( callback(stream_.userBuffer, stream_.bufferSize, info->userData) )
+ handle->stopStream = true;
}
- int nChannels = stream->nDeviceChannels[0] + stream->nDeviceChannels[1];
- int bufferBytes;
- ASIOBufferInfo *bufferInfos = (ASIOBufferInfo *) info->buffers;
- if ( stream->mode == OUTPUT || stream->mode == DUPLEX ) {
+ int bufferBytes, j;
+ int nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- bufferBytes = stream->bufferSize * formatBytes(stream->deviceFormat[0]);
- if (stream->doConvertBuffer[0]) {
+ bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[0]);
+ if (stream_.doConvertBuffer[0]) {
- convertStreamBuffer(stream, OUTPUT);
- if ( stream->doByteSwap[0] )
- byteSwapBuffer(stream->deviceBuffer,
- stream->bufferSize * stream->nDeviceChannels[0],
- stream->deviceFormat[0]);
+ convertStreamBuffer(OUTPUT);
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer(stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[0],
+ stream_.deviceFormat[0]);
// Always de-interleave ASIO output data.
- for ( int i=0; i<stream->nDeviceChannels[0]; i++, bufferInfos++ ) {
- memcpy(bufferInfos->buffers[bufferIndex],
- &stream->deviceBuffer[i*bufferBytes], bufferBytes );
+ j = 0;
+ for ( int i=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memcpy(handle->bufferInfos[i].buffers[bufferIndex],
+ &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
}
}
else { // single channel only
- if (stream->doByteSwap[0])
- byteSwapBuffer(stream->userBuffer,
- stream->bufferSize * stream->nUserChannels[0],
- stream->userFormat);
+ if (stream_.doByteSwap[0])
+ byteSwapBuffer(stream_.userBuffer,
+ stream_.bufferSize * stream_.nUserChannels[0],
+ stream_.userFormat);
- memcpy(bufferInfos->buffers[bufferIndex], stream->userBuffer, bufferBytes );
+ for ( int i=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue ) {
+ memcpy(handle->bufferInfos[i].buffers[bufferIndex], stream_.userBuffer, bufferBytes );
+ break;
+ }
+ }
}
}
- if ( stream->mode == INPUT || stream->mode == DUPLEX ) {
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- bufferBytes = stream->bufferSize * formatBytes(stream->deviceFormat[1]);
- if (stream->doConvertBuffer[1]) {
+ bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
+ if (stream_.doConvertBuffer[1]) {
// Always interleave ASIO input data.
- for ( int i=0; i<stream->nDeviceChannels[1]; i++, bufferInfos++ )
- memcpy(&stream->deviceBuffer[i*bufferBytes], bufferInfos->buffers[bufferIndex], bufferBytes );
+ j = 0;
+ for ( int i=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput == ASIOTrue )
+ memcpy(&stream_.deviceBuffer[j++*bufferBytes],
+ handle->bufferInfos[i].buffers[bufferIndex],
+ bufferBytes );
+ }
- if ( stream->doByteSwap[1] )
- byteSwapBuffer(stream->deviceBuffer,
- stream->bufferSize * stream->nDeviceChannels[1],
- stream->deviceFormat[1]);
- convertStreamBuffer(stream, INPUT);
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer(stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[1],
+ stream_.deviceFormat[1]);
+ convertStreamBuffer(INPUT);
}
else { // single channel only
- memcpy(stream->userBuffer, bufferInfos->buffers[bufferIndex], bufferBytes );
+ for ( int i=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
+ memcpy(stream_.userBuffer,
+ handle->bufferInfos[i].buffers[bufferIndex],
+ bufferBytes );
+ break;
+ }
+ }
- if (stream->doByteSwap[1])
- byteSwapBuffer(stream->userBuffer,
- stream->bufferSize * stream->nUserChannels[1],
- stream->userFormat);
+ if (stream_.doByteSwap[1])
+ byteSwapBuffer(stream_.userBuffer,
+ stream_.bufferSize * stream_.nUserChannels[1],
+ stream_.userFormat);
}
}
if ( !info->usingCallback )
- info->blockTick = false;
-
- MUTEX_UNLOCK(&stream->mutex);
-}
+ SetEvent( handle->condition );
-void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- stream->callbackInfo.callback = (void *) callback;
- stream->callbackInfo.userData = userData;
- stream->callbackInfo.usingCallback = true;
+ MUTEX_UNLOCK(&stream_.mutex);
}
//******************** End of __WINDOWS_ASIO__ *********************//
+#endif
-#elif defined(__WINDOWS_DS__) // Windows DirectSound API
+#if defined(__WINDOWS_DS__) // Windows DirectSound API
#include <dsound.h>
+// A structure to hold various information related to the DirectSound
+// API implementation.
+struct DsHandle {
+ void *object;
+ void *buffer;
+ UINT bufferPointer;
+};
+
// Declarations for utility functions, callbacks, and structures
// specific to the DirectSound implementation.
static bool CALLBACK deviceCountCallback(LPGUID lpguid,
static char* getErrorString(int code);
+extern "C" unsigned __stdcall callbackHandler(void *ptr);
+
struct enum_info {
char name[64];
LPGUID id;
bool isValid;
};
-int RtAudio :: getDefaultInputDevice(void)
+RtApiDs :: RtApiDs()
+{
+ this->initialize();
+
+ if (nDevices_ <= 0) {
+ sprintf(message_, "RtApiDs: no Windows DirectSound audio devices found!");
+ error(RtError::NO_DEVICES_FOUND);
+ }
+}
+
+RtApiDs :: ~RtApiDs()
+{
+ if ( stream_.mode != UNINITIALIZED ) closeStream();
+}
+
+int RtApiDs :: getDefaultInputDevice(void)
{
enum_info info;
info.name[0] = '\0';
// Enumerate through devices to find the default output.
HRESULT result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)defaultDeviceCallback, &info);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Error performing default input device enumeration: %s.",
+ sprintf(message_, "RtApiDs: Error performing default input device enumeration: %s.",
getErrorString(result));
error(RtError::WARNING);
return 0;
}
- for ( int i=0; i<nDevices; i++ )
- if ( strncmp( devices[i].name, info.name, 64 ) == 0 ) return i;
+ for ( int i=0; i<nDevices_; i++ ) {
+ if ( strncmp( info.name, devices_[i].name.c_str(), 64 ) == 0 ) return i;
+ }
+
return 0;
}
-int RtAudio :: getDefaultOutputDevice(void)
+int RtApiDs :: getDefaultOutputDevice(void)
{
enum_info info;
info.name[0] = '\0';
// Enumerate through devices to find the default output.
HRESULT result = DirectSoundEnumerate((LPDSENUMCALLBACK)defaultDeviceCallback, &info);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Error performing default output device enumeration: %s.",
+ sprintf(message_, "RtApiDs: Error performing default output device enumeration: %s.",
getErrorString(result));
error(RtError::WARNING);
return 0;
}
- for ( int i=0; i<nDevices; i++ )
- if ( strncmp(devices[i].name, info.name, 64 ) == 0 ) return i;
+ for ( int i=0; i<nDevices_; i++ )
+ if ( strncmp( info.name, devices_[i].name.c_str(), 64 ) == 0 ) return i;
return 0;
}
-void RtAudio :: initialize(void)
+void RtApiDs :: initialize(void)
{
int i, ins = 0, outs = 0, count = 0;
HRESULT result;
- nDevices = 0;
+ nDevices_ = 0;
// Count DirectSound devices.
result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceCountCallback, &outs);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to enumerate through sound playback devices: %s.",
+ sprintf(message_, "RtApiDs: Unable to enumerate through sound playback devices: %s.",
getErrorString(result));
error(RtError::DRIVER_ERROR);
}
// Count DirectSoundCapture devices.
result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceCountCallback, &ins);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to enumerate through sound capture devices: %s.",
+ sprintf(message_, "RtApiDs: Unable to enumerate through sound capture devices: %s.",
getErrorString(result));
error(RtError::DRIVER_ERROR);
}
// Get playback device info and check capabilities.
result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceInfoCallback, &info[0]);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to enumerate through sound playback devices: %s.",
+ sprintf(message_, "RtApiDs: Unable to enumerate through sound playback devices: %s.",
getErrorString(result));
error(RtError::DRIVER_ERROR);
}
// Get capture device info and check capabilities.
result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceInfoCallback, &info[0]);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to enumerate through sound capture devices: %s.",
+ sprintf(message_, "RtApiDs: Unable to enumerate through sound capture devices: %s.",
getErrorString(result));
error(RtError::DRIVER_ERROR);
}
- // Parse the devices and check validity. Devices are considered
- // invalid if they cannot be opened, they report < 1 supported
- // channels, or they report no supported data (capture only).
- for (i=0; i<count; i++)
- if ( info[i].isValid ) nDevices++;
-
- if (nDevices == 0) return;
-
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
-
- // Copy the names to our devices structures.
+ // Create device structures for valid devices and write device names
+ // to each. Devices are considered invalid if they cannot be
+ // opened, they report < 1 supported channels, or they report no
+ // supported data (capture only).
+ RtApiDevice device;
int index = 0;
for (i=0; i<count; i++) {
- if ( info[i].isValid )
- strncpy(devices[index++].name, info[i].name, 64);
+ if ( info[i].isValid ) {
+ device.name.erase();
+ device.name.append( (const char *)info[i].name, strlen(info[i].name)+1);
+ devices_.push_back(device);
+ }
}
- //for (i=0;i<nDevices; i++)
- //probeDeviceInfo(&devices[i]);
-
+ nDevices_ = devices_.size();
return;
}
-void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info)
+void RtApiDs :: probeDeviceInfo(RtApiDevice *info)
{
enum_info dsinfo;
- strncpy( dsinfo.name, info->name, 64 );
+ strncpy( dsinfo.name, info->name.c_str(), 64 );
dsinfo.isValid = false;
// Enumerate through input devices to find the id (if it exists).
HRESULT result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Error performing input device id enumeration: %s.",
+ sprintf(message_, "RtApiDs: Error performing input device id enumeration: %s.",
getErrorString(result));
error(RtError::WARNING);
return;
LPDIRECTSOUNDCAPTURE input;
result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Could not create DirectSound capture object (%s): %s.",
- info->name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Could not create capture object (%s): %s.",
+ info->name.c_str(), getErrorString(result));
error(RtError::WARNING);
goto playback_probe;
}
result = input->GetCaps( &in_caps );
if ( FAILED(result) ) {
input->Release();
- sprintf(message, "RtAudio: Could not get DirectSound capture capabilities (%s): %s.",
- info->name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Could not get capture capabilities (%s): %s.",
+ info->name.c_str(), getErrorString(result));
error(RtError::WARNING);
goto playback_probe;
}
info->maxInputChannels = in_caps.dwChannels;
// Get sample rate and format information.
+ info->sampleRates.clear();
if( in_caps.dwChannels == 2 ) {
if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->nativeFormats |= RTAUDIO_SINT16;
if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->nativeFormats |= RTAUDIO_SINT16;
if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->nativeFormats |= RTAUDIO_SINT8;
if ( info->nativeFormats & RTAUDIO_SINT16 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->sampleRates[info->nSampleRates++] = 11025;
- if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->sampleRates[info->nSampleRates++] = 22050;
- if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->sampleRates[info->nSampleRates++] = 44100;
+ if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->sampleRates.push_back( 11025 );
+ if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->sampleRates.push_back( 22050 );
+ if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->sampleRates.push_back( 44100 );
}
else if ( info->nativeFormats & RTAUDIO_SINT8 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->sampleRates[info->nSampleRates++] = 11025;
- if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->sampleRates[info->nSampleRates++] = 22050;
- if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->sampleRates[info->nSampleRates++] = 44100;
+ if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->sampleRates.push_back( 11025 );
+ if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->sampleRates.push_back( 22050 );
+ if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->sampleRates.push_back( 44100 );
}
}
else if ( in_caps.dwChannels == 1 ) {
if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->nativeFormats |= RTAUDIO_SINT8;
if ( info->nativeFormats & RTAUDIO_SINT16 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1M16 ) info->sampleRates[info->nSampleRates++] = 11025;
- if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->sampleRates[info->nSampleRates++] = 22050;
- if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->sampleRates[info->nSampleRates++] = 44100;
+ if( in_caps.dwFormats & WAVE_FORMAT_1M16 ) info->sampleRates.push_back( 11025 );
+ if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->sampleRates.push_back( 22050 );
+ if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->sampleRates.push_back( 44100 );
}
else if ( info->nativeFormats & RTAUDIO_SINT8 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->sampleRates[info->nSampleRates++] = 11025;
- if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->sampleRates[info->nSampleRates++] = 22050;
- if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->sampleRates[info->nSampleRates++] = 44100;
+ if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->sampleRates.push_back( 11025 );
+ if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->sampleRates.push_back( 22050 );
+ if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->sampleRates.push_back( 44100 );
}
}
else info->minInputChannels = 0; // technically, this would be an error
// Enumerate through output devices to find the id (if it exists).
result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Error performing output device id enumeration: %s.",
+ sprintf(message_, "RtApiDs: Error performing output device id enumeration: %s.",
getErrorString(result));
error(RtError::WARNING);
return;
DSCAPS out_caps;
result = DirectSoundCreate( dsinfo.id, &output, NULL );
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Could not create DirectSound playback object (%s): %s.",
- info->name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Could not create playback object (%s): %s.",
+ info->name.c_str(), getErrorString(result));
error(RtError::WARNING);
goto check_parameters;
}
result = output->GetCaps( &out_caps );
if ( FAILED(result) ) {
output->Release();
- sprintf(message, "RtAudio: Could not get DirectSound playback capabilities (%s): %s.",
- info->name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Could not get playback capabilities (%s): %s.",
+ info->name.c_str(), getErrorString(result));
error(RtError::WARNING);
goto check_parameters;
}
// Get sample rate information. Use capture device rate information
// if it exists.
- if ( info->nSampleRates == 0 ) {
- info->sampleRates[0] = (int) out_caps.dwMinSecondarySampleRate;
- info->sampleRates[1] = (int) out_caps.dwMaxSecondarySampleRate;
- if ( out_caps.dwFlags & DSCAPS_CONTINUOUSRATE )
- info->nSampleRates = -1;
- else if ( out_caps.dwMinSecondarySampleRate == out_caps.dwMaxSecondarySampleRate ) {
- if ( out_caps.dwMinSecondarySampleRate == 0 ) {
- // This is a bogus driver report ... fake the range and cross
- // your fingers.
- info->sampleRates[0] = 11025;
- info->sampleRates[1] = 48000;
- info->nSampleRates = -1; /* continuous range */
- sprintf(message, "RtAudio: bogus sample rates reported by DirectSound driver ... using defaults (%s).",
- info->name);
- error(RtError::DEBUG_WARNING);
- }
- else {
- info->nSampleRates = 1;
- }
- }
- else if ( (out_caps.dwMinSecondarySampleRate < 1000.0) &&
- (out_caps.dwMaxSecondarySampleRate > 50000.0) ) {
- // This is a bogus driver report ... support for only two
- // distant rates. We'll assume this is a range.
- info->nSampleRates = -1;
- sprintf(message, "RtAudio: bogus sample rates reported by DirectSound driver ... using range (%s).",
- info->name);
- error(RtError::WARNING);
- }
- else info->nSampleRates = 2;
+ if ( info->sampleRates.size() == 0 ) {
+ info->sampleRates.push_back( (int) out_caps.dwMinSecondarySampleRate );
+ info->sampleRates.push_back( (int) out_caps.dwMaxSecondarySampleRate );
}
else {
- // Check input rates against output rate range
- for ( int i=info->nSampleRates-1; i>=0; i-- ) {
- if ( info->sampleRates[i] <= out_caps.dwMaxSecondarySampleRate )
- break;
- info->nSampleRates--;
+ // Check input rates against output rate range.
+ for ( unsigned int i=info->sampleRates.size()-1; i>=0; i-- ) {
+ if ( (unsigned int) info->sampleRates[i] > out_caps.dwMaxSecondarySampleRate )
+ info->sampleRates.erase( info->sampleRates.begin() + i );
}
- while ( info->sampleRates[0] < out_caps.dwMinSecondarySampleRate ) {
- info->nSampleRates--;
- for ( int i=0; i<info->nSampleRates; i++)
- info->sampleRates[i] = info->sampleRates[i+1];
- if ( info->nSampleRates <= 0 ) break;
+ while ( info->sampleRates.size() > 0 &&
+ ((unsigned int) info->sampleRates[0] < out_caps.dwMinSecondarySampleRate) ) {
+ info->sampleRates.erase( info->sampleRates.begin() );
}
}
output->Release();
check_parameters:
- if ( info->maxInputChannels == 0 && info->maxOutputChannels == 0 )
+ if ( info->maxInputChannels == 0 && info->maxOutputChannels == 0 ) {
+ sprintf(message_, "RtApiDs: no reported input or output channels for device (%s).",
+ info->name.c_str());
+ error(RtError::DEBUG_WARNING);
return;
- if ( info->nSampleRates == 0 || info->nativeFormats == 0 )
+ }
+ if ( info->sampleRates.size() == 0 || info->nativeFormats == 0 ) {
+ sprintf(message_, "RtApiDs: no reported sample rates or data formats for device (%s).",
+ info->name.c_str());
+ error(RtError::DEBUG_WARNING);
return;
+ }
// Determine duplex status.
if (info->maxInputChannels < info->maxOutputChannels)
return;
}
-bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
- STREAM_MODE mode, int channels,
- int sampleRate, RTAUDIO_FORMAT format,
- int *bufferSize, int numberOfBuffers)
+bool RtApiDs :: probeDeviceOpen( int device, StreamMode mode, int channels,
+ int sampleRate, RtAudioFormat format,
+ int *bufferSize, int numberOfBuffers)
{
HRESULT result;
HWND hWnd = GetForegroundWindow();
+
// According to a note in PortAudio, using GetDesktopWindow()
// instead of GetForegroundWindow() is supposed to avoid problems
// that occur when the application's window is not the foreground
waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
// Determine the data format.
- if ( devices[device].nativeFormats ) { // 8-bit and/or 16-bit support
+ if ( devices_[device].nativeFormats ) { // 8-bit and/or 16-bit support
if ( format == RTAUDIO_SINT8 ) {
- if ( devices[device].nativeFormats & RTAUDIO_SINT8 )
+ if ( devices_[device].nativeFormats & RTAUDIO_SINT8 )
waveFormat.wBitsPerSample = 8;
else
waveFormat.wBitsPerSample = 16;
}
else {
- if ( devices[device].nativeFormats & RTAUDIO_SINT16 )
+ if ( devices_[device].nativeFormats & RTAUDIO_SINT16 )
waveFormat.wBitsPerSample = 16;
else
waveFormat.wBitsPerSample = 8;
}
}
else {
- sprintf(message, "RtAudio: no reported data formats for DirectSound device (%s).",
- devices[device].name);
+ sprintf(message_, "RtApiDs: no reported data formats for device (%s).",
+ devices_[device].name.c_str());
error(RtError::DEBUG_WARNING);
return FAILURE;
}
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
enum_info dsinfo;
- strncpy( dsinfo.name, devices[device].name, 64 );
+ void *ohandle = 0, *bhandle = 0;
+ strncpy( dsinfo.name, devices_[device].name.c_str(), 64 );
dsinfo.isValid = false;
if ( mode == OUTPUT ) {
- if ( devices[device].maxOutputChannels < channels )
+ if ( devices_[device].maxOutputChannels < channels ) {
+ sprintf(message_, "RtApiDs: requested channels (%d) > than supported (%d) by device (%s).",
+ channels, devices_[device].maxOutputChannels, devices_[device].name.c_str());
+ error(RtError::DEBUG_WARNING);
return FAILURE;
+ }
// Enumerate through output devices to find the id (if it exists).
result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Error performing output device id enumeration: %s.",
+ sprintf(message_, "RtApiDs: Error performing output device id enumeration: %s.",
getErrorString(result));
error(RtError::DEBUG_WARNING);
return FAILURE;
}
if ( dsinfo.isValid == false ) {
- sprintf(message, "RtAudio: DS output device (%s) id not found!", devices[device].name);
+ sprintf(message_, "RtApiDs: output device (%s) id not found!", devices_[device].name.c_str());
error(RtError::DEBUG_WARNING);
return FAILURE;
}
result = DirectSoundCreate( id, &object, NULL );
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Could not create DirectSound playback object (%s): %s.",
- devices[device].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Could not create playback object (%s): %s.",
+ devices_[device].name.c_str(), getErrorString(result));
error(RtError::DEBUG_WARNING);
return FAILURE;
}
result = object->SetCooperativeLevel(hWnd, DSSCL_EXCLUSIVE);
if ( FAILED(result) ) {
object->Release();
- sprintf(message, "RtAudio: Unable to set DirectSound cooperative level (%s): %s.",
- devices[device].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to set cooperative level (%s): %s.",
+ devices_[device].name.c_str(), getErrorString(result));
error(RtError::WARNING);
return FAILURE;
}
// Even though we will write to the secondary buffer, we need to
- // access the primary buffer to set the correct output format.
- // The default is 8-bit, 22 kHz!
- // Setup the DS primary buffer description.
+ // access the primary buffer to set the correct output format
+ // (since the default is 8-bit, 22 kHz!). Setup the DS primary
+ // buffer description.
ZeroMemory(&bufferDescription, sizeof(DSBUFFERDESC));
bufferDescription.dwSize = sizeof(DSBUFFERDESC);
bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL);
if ( FAILED(result) ) {
object->Release();
- sprintf(message, "RtAudio: Unable to access DS primary buffer (%s): %s.",
- devices[device].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to access primary buffer (%s): %s.",
+ devices_[device].name.c_str(), getErrorString(result));
error(RtError::WARNING);
return FAILURE;
}
result = buffer->SetFormat(&waveFormat);
if ( FAILED(result) ) {
object->Release();
- sprintf(message, "RtAudio: Unable to set DS primary buffer format (%s): %s.",
- devices[device].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to set primary buffer format (%s): %s.",
+ devices_[device].name.c_str(), getErrorString(result));
error(RtError::WARNING);
return FAILURE;
}
result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL);
if ( FAILED(result) ) {
object->Release();
- sprintf(message, "RtAudio: Unable to create secondary DS buffer (%s): %s.",
- devices[device].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to create secondary DS buffer (%s): %s.",
+ devices_[device].name.c_str(), getErrorString(result));
error(RtError::WARNING);
return FAILURE;
}
result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0);
if ( FAILED(result) ) {
object->Release();
- sprintf(message, "RtAudio: Unable to lock DS buffer (%s): %s.",
- devices[device].name, getErrorString(result));
+ buffer->Release();
+ sprintf(message_, "RtApiDs: Unable to lock buffer (%s): %s.",
+ devices_[device].name.c_str(), getErrorString(result));
error(RtError::WARNING);
return FAILURE;
}
result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
if ( FAILED(result) ) {
object->Release();
- sprintf(message, "RtAudio: Unable to unlock DS buffer(%s): %s.",
- devices[device].name, getErrorString(result));
+ buffer->Release();
+ sprintf(message_, "RtApiDs: Unable to unlock buffer(%s): %s.",
+ devices_[device].name.c_str(), getErrorString(result));
error(RtError::WARNING);
return FAILURE;
}
- stream->handle[0].object = (void *) object;
- stream->handle[0].buffer = (void *) buffer;
- stream->nDeviceChannels[0] = channels;
+ ohandle = (void *) object;
+ bhandle = (void *) buffer;
+ stream_.nDeviceChannels[0] = channels;
}
if ( mode == INPUT ) {
- if ( devices[device].maxInputChannels < channels )
+ if ( devices_[device].maxInputChannels < channels )
return FAILURE;
// Enumerate through input devices to find the id (if it exists).
result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Error performing input device id enumeration: %s.",
+ sprintf(message_, "RtApiDs: Error performing input device id enumeration: %s.",
getErrorString(result));
error(RtError::DEBUG_WARNING);
return FAILURE;
}
if ( dsinfo.isValid == false ) {
- sprintf(message, "RtAudio: DS input device (%s) id not found!", devices[device].name);
+ sprintf(message_, "RtAudioDS: input device (%s) id not found!", devices_[device].name.c_str());
error(RtError::DEBUG_WARNING);
return FAILURE;
}
result = DirectSoundCaptureCreate( id, &object, NULL );
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Could not create DirectSound capture object (%s): %s.",
- devices[device].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Could not create capture object (%s): %s.",
+ devices_[device].name.c_str(), getErrorString(result));
error(RtError::WARNING);
return FAILURE;
}
result = object->CreateCaptureBuffer(&bufferDescription, &buffer, NULL);
if ( FAILED(result) ) {
object->Release();
- sprintf(message, "RtAudio: Unable to create DS capture buffer (%s): %s.",
- devices[device].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to create capture buffer (%s): %s.",
+ devices_[device].name.c_str(), getErrorString(result));
error(RtError::WARNING);
return FAILURE;
}
result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0);
if ( FAILED(result) ) {
object->Release();
- sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.",
- devices[device].name, getErrorString(result));
+ buffer->Release();
+ sprintf(message_, "RtApiDs: Unable to lock capture buffer (%s): %s.",
+ devices_[device].name.c_str(), getErrorString(result));
error(RtError::WARNING);
return FAILURE;
}
result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
if ( FAILED(result) ) {
object->Release();
- sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.",
- devices[device].name, getErrorString(result));
+ buffer->Release();
+ sprintf(message_, "RtApiDs: Unable to unlock capture buffer (%s): %s.",
+ devices_[device].name.c_str(), getErrorString(result));
error(RtError::WARNING);
return FAILURE;
}
- stream->handle[1].object = (void *) object;
- stream->handle[1].buffer = (void *) buffer;
- stream->nDeviceChannels[1] = channels;
+ ohandle = (void *) object;
+ bhandle = (void *) buffer;
+ stream_.nDeviceChannels[1] = channels;
}
- stream->userFormat = format;
+ stream_.userFormat = format;
if ( waveFormat.wBitsPerSample == 8 )
- stream->deviceFormat[mode] = RTAUDIO_SINT8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
else
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->nUserChannels[mode] = channels;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.nUserChannels[mode] = channels;
*bufferSize = buffer_size / (channels * nBuffers * waveFormat.wBitsPerSample / 8);
- stream->bufferSize = *bufferSize;
+ stream_.bufferSize = *bufferSize;
// Set flags for buffer conversion
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode])
- stream->doConvertBuffer[mode] = true;
+ stream_.doConvertBuffer[mode] = false;
+ if (stream_.userFormat != stream_.deviceFormat[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode])
+ stream_.doConvertBuffer[mode] = true;
// Allocate necessary internal buffers
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
+ if ( stream_.nUserChannels[0] != stream_.nUserChannels[1] ) {
long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
+ if (stream_.nUserChannels[0] >= stream_.nUserChannels[1])
+ buffer_bytes = stream_.nUserChannels[0];
else
- buffer_bytes = stream->nUserChannels[1];
-
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL)
- goto memory_error;
+ buffer_bytes = stream_.nUserChannels[1];
+
+ buffer_bytes *= *bufferSize * formatBytes(stream_.userFormat);
+ if (stream_.userBuffer) free(stream_.userBuffer);
+ stream_.userBuffer = (char *) calloc(buffer_bytes, 1);
+ if (stream_.userBuffer == NULL) {
+ sprintf(message_, "RtApiDs: error allocating user buffer memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
}
- if ( stream->doConvertBuffer[mode] ) {
+ if ( stream_.doConvertBuffer[mode] ) {
long buffer_bytes;
bool makeBuffer = true;
if ( mode == OUTPUT )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
+ buffer_bytes = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
else { // mode == INPUT
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == OUTPUT && stream->deviceBuffer ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
+ buffer_bytes = stream_.nDeviceChannels[1] * formatBytes(stream_.deviceFormat[1]);
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ long bytes_out = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
if ( buffer_bytes < bytes_out ) makeBuffer = false;
}
}
if ( makeBuffer ) {
buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL)
- goto memory_error;
+ if (stream_.deviceBuffer) free(stream_.deviceBuffer);
+ stream_.deviceBuffer = (char *) calloc(buffer_bytes, 1);
+ if (stream_.deviceBuffer == NULL) {
+ sprintf(message_, "RtApiDs: error allocating device buffer memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
}
}
- stream->device[mode] = device;
- stream->state = STREAM_STOPPED;
- if ( stream->mode == OUTPUT && mode == INPUT )
+ // Allocate our DsHandle structures for the stream.
+ DsHandle *handles;
+ if ( stream_.apiHandle == 0 ) {
+ handles = (DsHandle *) calloc(2, sizeof(DsHandle));
+ if ( handles == NULL ) {
+ sprintf(message_, "RtApiDs: Error allocating DsHandle memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
+ handles[0].object = 0;
+ handles[1].object = 0;
+ stream_.apiHandle = (void *) handles;
+ }
+ else
+ handles = (DsHandle *) stream_.apiHandle;
+ handles[mode].object = ohandle;
+ handles[mode].buffer = bhandle;
+
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
// We had already set up an output stream.
- stream->mode = DUPLEX;
+ stream_.mode = DUPLEX;
else
- stream->mode = mode;
- stream->nBuffers = nBuffers;
- stream->sampleRate = sampleRate;
+ stream_.mode = mode;
+ stream_.nBuffers = nBuffers;
+ stream_.sampleRate = sampleRate;
return SUCCESS;
- memory_error:
- if (stream->handle[0].object) {
- LPDIRECTSOUND object = (LPDIRECTSOUND) stream->handle[0].object;
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- if (buffer) {
- buffer->Release();
- stream->handle[0].buffer = NULL;
+ error:
+ if (handles) {
+ if (handles[0].object) {
+ LPDIRECTSOUND object = (LPDIRECTSOUND) handles[0].object;
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer;
+ if (buffer) buffer->Release();
+ object->Release();
}
- object->Release();
- stream->handle[0].object = NULL;
- }
- if (stream->handle[1].object) {
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) stream->handle[1].object;
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- if (buffer) {
- buffer->Release();
- stream->handle[1].buffer = NULL;
+ if (handles[1].object) {
+ LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handles[1].object;
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer;
+ if (buffer) buffer->Release();
+ object->Release();
}
- object->Release();
- stream->handle[1].object = NULL;
+ free(handles);
+ stream_.apiHandle = 0;
}
- if (stream->userBuffer) {
- free(stream->userBuffer);
- stream->userBuffer = 0;
+
+ if (stream_.userBuffer) {
+ free(stream_.userBuffer);
+ stream_.userBuffer = 0;
}
- sprintf(message, "RtAudio: error allocating buffer memory (%s).",
- devices[device].name);
+
error(RtError::WARNING);
return FAILURE;
}
-void RtAudio :: cancelStreamCallback(int streamId)
+void RtApiDs :: setStreamCallback(RtAudioCallback callback, void *userData)
+{
+ verifyStream();
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ if ( info->usingCallback ) {
+ sprintf(message_, "RtApiDs: A callback is already set for this stream!");
+ error(RtError::WARNING);
+ return;
+ }
+
+ info->callback = (void *) callback;
+ info->userData = userData;
+ info->usingCallback = true;
+ info->object = (void *) this;
+
+ unsigned thread_id;
+ info->thread = _beginthreadex(NULL, 0, &callbackHandler,
+ &stream_.callbackInfo, 0, &thread_id);
+ if (info->thread == 0) {
+ info->usingCallback = false;
+ sprintf(message_, "RtApiDs: error starting callback thread!");
+ error(RtError::THREAD_ERROR);
+ }
+
+ // When spawning multiple threads in quick succession, it appears to be
+ // necessary to wait a bit for each to initialize ... another windoism!
+ Sleep(1);
+}
+
+void RtApiDs :: cancelStreamCallback()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
- if (stream->callbackInfo.usingCallback) {
+ if (stream_.callbackInfo.usingCallback) {
- if (stream->state == STREAM_RUNNING)
- stopStream( streamId );
+ if (stream_.state == STREAM_RUNNING)
+ stopStream();
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK(&stream_.mutex);
- stream->callbackInfo.usingCallback = false;
- WaitForSingleObject( (HANDLE)stream->callbackInfo.thread, INFINITE );
- CloseHandle( (HANDLE)stream->callbackInfo.thread );
- stream->callbackInfo.thread = 0;
- stream->callbackInfo.callback = NULL;
- stream->callbackInfo.userData = NULL;
+ stream_.callbackInfo.usingCallback = false;
+ WaitForSingleObject( (HANDLE)stream_.callbackInfo.thread, INFINITE );
+ CloseHandle( (HANDLE)stream_.callbackInfo.thread );
+ stream_.callbackInfo.thread = 0;
+ stream_.callbackInfo.callback = NULL;
+ stream_.callbackInfo.userData = NULL;
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
}
}
-void RtAudio :: closeStream(int streamId)
+void RtApiDs :: closeStream()
{
// We don't want an exception to be thrown here because this
// function is called by our class destructor. So, do our own
// streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
+ if ( stream_.mode == UNINITIALIZED ) {
+ sprintf(message_, "RtApiDs::closeStream(): no open stream to close!");
error(RtError::WARNING);
return;
}
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
-
- if (stream->callbackInfo.usingCallback) {
- stream->callbackInfo.usingCallback = false;
- WaitForSingleObject( (HANDLE)stream->callbackInfo.thread, INFINITE );
- CloseHandle( (HANDLE)stream->callbackInfo.thread );
+ if (stream_.callbackInfo.usingCallback) {
+ stream_.callbackInfo.usingCallback = false;
+ WaitForSingleObject( (HANDLE)stream_.callbackInfo.thread, INFINITE );
+ CloseHandle( (HANDLE)stream_.callbackInfo.thread );
}
- DeleteCriticalSection(&stream->mutex);
-
- if (stream->handle[0].object) {
- LPDIRECTSOUND object = (LPDIRECTSOUND) stream->handle[0].object;
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- if (buffer) {
- buffer->Stop();
- buffer->Release();
+ DsHandle *handles = (DsHandle *) stream_.apiHandle;
+ if (handles) {
+ if (handles[0].object) {
+ LPDIRECTSOUND object = (LPDIRECTSOUND) handles[0].object;
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer;
+ if (buffer) {
+ buffer->Stop();
+ buffer->Release();
+ }
+ object->Release();
}
- object->Release();
- }
- if (stream->handle[1].object) {
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) stream->handle[1].object;
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- if (buffer) {
- buffer->Stop();
- buffer->Release();
+ if (handles[1].object) {
+ LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handles[1].object;
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer;
+ if (buffer) {
+ buffer->Stop();
+ buffer->Release();
+ }
+ object->Release();
}
- object->Release();
+ free(handles);
+ stream_.apiHandle = 0;
+ }
+
+ if (stream_.userBuffer) {
+ free(stream_.userBuffer);
+ stream_.userBuffer = 0;
}
- if (stream->userBuffer)
- free(stream->userBuffer);
-
- if (stream->deviceBuffer)
- free(stream->deviceBuffer);
+ if (stream_.deviceBuffer) {
+ free(stream_.deviceBuffer);
+ stream_.deviceBuffer = 0;
+ }
- free(stream);
- streams.erase(streamId);
+ stream_.mode = UNINITIALIZED;
}
-void RtAudio :: startStream(int streamId)
+void RtApiDs :: startStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+ if (stream_.state == STREAM_RUNNING) return;
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_RUNNING)
- goto unlock;
+ MUTEX_LOCK(&stream_.mutex);
HRESULT result;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
+ DsHandle *handles = (DsHandle *) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer;
result = buffer->Play(0, 0, DSBPLAY_LOOPING );
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to start DS buffer (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to start buffer (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer;
result = buffer->Start(DSCBSTART_LOOPING );
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to start DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to start capture buffer (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
}
- stream->state = STREAM_RUNNING;
+ stream_.state = STREAM_RUNNING;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
}
-void RtAudio :: stopStream(int streamId)
+void RtApiDs :: stopStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
+ verifyStream();
+ if (stream_.state == STREAM_STOPPED) return;
- if (stream->state == STREAM_STOPPED) {
- MUTEX_UNLOCK(&stream->mutex);
- return;
- }
+ // Change the state before the lock to improve shutdown response
+ // when using a callback.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK(&stream_.mutex);
// There is no specific DirectSound API call to "drain" a buffer
// before stopping. We can hack this for playback by writing zeroes
LPVOID buffer2 = NULL;
DWORD bufferSize1 = 0;
DWORD bufferSize2 = 0;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ DsHandle *handles = (DsHandle *) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
DWORD currentPos, safePos;
- long buffer_bytes = stream->bufferSize * stream->nDeviceChannels[0];
- buffer_bytes *= formatBytes(stream->deviceFormat[0]);
+ long buffer_bytes = stream_.bufferSize * stream_.nDeviceChannels[0];
+ buffer_bytes *= formatBytes(stream_.deviceFormat[0]);
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- UINT nextWritePos = stream->handle[0].bufferPointer;
- dsBufferSize = buffer_bytes * stream->nBuffers;
+ LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer;
+ UINT nextWritePos = handles[0].bufferPointer;
+ dsBufferSize = buffer_bytes * stream_.nBuffers;
// Write zeroes for nBuffer counts.
- for (int i=0; i<stream->nBuffers; i++) {
+ for (int i=0; i<stream_.nBuffers; i++) {
// Find out where the read and "safe write" pointers are.
result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to get current position (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
// Check whether the entire write region is behind the play pointer.
while ( currentPos < endWrite ) {
- float millis = (endWrite - currentPos) * 900.0;
- millis /= ( formatBytes(stream->deviceFormat[0]) * stream->sampleRate);
+ double millis = (endWrite - currentPos) * 900.0;
+ millis /= ( formatBytes(stream_.deviceFormat[0]) * stream_.sampleRate);
if ( millis < 1.0 ) millis = 1.0;
Sleep( (DWORD) millis );
// Wake up, find out where we are now
result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos );
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to get current position (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1,
&bufferSize1, &buffer2, &bufferSize2, 0);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS buffer during playback (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to lock buffer during playback (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
// Update our buffer offset and unlock sound buffer
dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS buffer during playback (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to unlock buffer during playback (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize;
- stream->handle[0].bufferPointer = nextWritePos;
+ handles[0].bufferPointer = nextWritePos;
}
// If we play again, start at the beginning of the buffer.
- stream->handle[0].bufferPointer = 0;
+ handles[0].bufferPointer = 0;
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer;
buffer1 = NULL;
bufferSize1 = 0;
result = buffer->Stop();
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to stop DS capture buffer (%s): %s",
- devices[stream->device[1]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to stop capture buffer (%s): %s",
+ devices_[stream_.device[1]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
- dsBufferSize = stream->bufferSize * stream->nDeviceChannels[1];
- dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers;
+ dsBufferSize = stream_.bufferSize * stream_.nDeviceChannels[1];
+ dsBufferSize *= formatBytes(stream_.deviceFormat[1]) * stream_.nBuffers;
// Lock the buffer and clear it so that if we start to play again,
// we won't have old data playing.
result = buffer->Lock(0, dsBufferSize, &buffer1, &bufferSize1, NULL, NULL, 0);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to lock capture buffer (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
// Unlock the DS buffer
result = buffer->Unlock(buffer1, bufferSize1, NULL, 0);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to unlock capture buffer (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
// If we start recording again, we must begin at beginning of buffer.
- stream->handle[1].bufferPointer = 0;
+ handles[1].bufferPointer = 0;
}
- stream->state = STREAM_STOPPED;
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
}
-void RtAudio :: abortStream(int streamId)
+void RtApiDs :: abortStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+ if (stream_.state == STREAM_STOPPED) return;
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ // Change the state before the lock to improve shutdown response
+ // when using a callback.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK(&stream_.mutex);
HRESULT result;
long dsBufferSize;
LPVOID audioPtr;
DWORD dataLen;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
+ DsHandle *handles = (DsHandle *) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer;
result = buffer->Stop();
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to stop DS buffer (%s): %s",
- devices[stream->device[0]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to stop buffer (%s): %s",
+ devices_[stream_.device[0]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
- dsBufferSize = stream->bufferSize * stream->nDeviceChannels[0];
- dsBufferSize *= formatBytes(stream->deviceFormat[0]) * stream->nBuffers;
+ dsBufferSize = stream_.bufferSize * stream_.nDeviceChannels[0];
+ dsBufferSize *= formatBytes(stream_.deviceFormat[0]) * stream_.nBuffers;
// Lock the buffer and clear it so that if we start to play again,
// we won't have old data playing.
result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS buffer (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to lock buffer (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
// Unlock the DS buffer
result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS buffer (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to unlock buffer (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
// If we start playing again, we must begin at beginning of buffer.
- stream->handle[0].bufferPointer = 0;
+ handles[0].bufferPointer = 0;
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer;
audioPtr = NULL;
dataLen = 0;
result = buffer->Stop();
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to stop DS capture buffer (%s): %s",
- devices[stream->device[1]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to stop capture buffer (%s): %s",
+ devices_[stream_.device[1]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
- dsBufferSize = stream->bufferSize * stream->nDeviceChannels[1];
- dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers;
+ dsBufferSize = stream_.bufferSize * stream_.nDeviceChannels[1];
+ dsBufferSize *= formatBytes(stream_.deviceFormat[1]) * stream_.nBuffers;
// Lock the buffer and clear it so that if we start to play again,
// we won't have old data playing.
result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to lock capture buffer (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
// Unlock the DS buffer
result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to unlock capture buffer (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
// If we start recording again, we must begin at beginning of buffer.
- stream->handle[1].bufferPointer = 0;
+ handles[1].bufferPointer = 0;
}
- stream->state = STREAM_STOPPED;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
}
-int RtAudio :: streamWillBlock(int streamId)
+int RtApiDs :: streamWillBlock()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+ if (stream_.state == STREAM_STOPPED) return 0;
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK(&stream_.mutex);
int channels;
int frames = 0;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
HRESULT result;
DWORD currentPos, safePos;
channels = 1;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ DsHandle *handles = (DsHandle *) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- UINT nextWritePos = stream->handle[0].bufferPointer;
- channels = stream->nDeviceChannels[0];
- DWORD dsBufferSize = stream->bufferSize * channels;
- dsBufferSize *= formatBytes(stream->deviceFormat[0]) * stream->nBuffers;
+ LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer;
+ UINT nextWritePos = handles[0].bufferPointer;
+ channels = stream_.nDeviceChannels[0];
+ DWORD dsBufferSize = stream_.bufferSize * channels;
+ dsBufferSize *= formatBytes(stream_.deviceFormat[0]) * stream_.nBuffers;
// Find out where the read and "safe write" pointers are.
result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to get current position (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
frames = currentPos - nextWritePos;
- frames /= channels * formatBytes(stream->deviceFormat[0]);
+ frames /= channels * formatBytes(stream_.deviceFormat[0]);
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
- LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- UINT nextReadPos = stream->handle[1].bufferPointer;
- channels = stream->nDeviceChannels[1];
- DWORD dsBufferSize = stream->bufferSize * channels;
- dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers;
+ LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer;
+ UINT nextReadPos = handles[1].bufferPointer;
+ channels = stream_.nDeviceChannels[1];
+ DWORD dsBufferSize = stream_.bufferSize * channels;
+ dsBufferSize *= formatBytes(stream_.deviceFormat[1]) * stream_.nBuffers;
// Find out where the write and "safe read" pointers are.
result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to get current capture position (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset
- if (stream->mode == DUPLEX ) {
+ if (stream_.mode == DUPLEX ) {
// Take largest value of the two.
int temp = safePos - nextReadPos;
- temp /= channels * formatBytes(stream->deviceFormat[1]);
+ temp /= channels * formatBytes(stream_.deviceFormat[1]);
frames = ( temp > frames ) ? temp : frames;
}
else {
frames = safePos - nextReadPos;
- frames /= channels * formatBytes(stream->deviceFormat[1]);
+ frames /= channels * formatBytes(stream_.deviceFormat[1]);
}
}
- frames = stream->bufferSize - frames;
+ frames = stream_.bufferSize - frames;
if (frames < 0) frames = 0;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
return frames;
}
-void RtAudio :: tickStream(int streamId)
+void RtApiDs :: tickStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
int stopStream = 0;
- if (stream->state == STREAM_STOPPED) {
- if (stream->callbackInfo.usingCallback) Sleep(50); // sleep 50 milliseconds
+ if (stream_.state == STREAM_STOPPED) {
+ if (stream_.callbackInfo.usingCallback) Sleep(50); // sleep 50 milliseconds
return;
}
- else if (stream->callbackInfo.usingCallback) {
- RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback;
- stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData);
+ else if (stream_.callbackInfo.usingCallback) {
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ stopStream = callback(stream_.userBuffer, stream_.bufferSize, stream_.callbackInfo.userData);
}
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK(&stream_.mutex);
// The state might change while waiting on a mutex.
- if (stream->state == STREAM_STOPPED) {
- MUTEX_UNLOCK(&stream->mutex);
+ if (stream_.state == STREAM_STOPPED) {
+ MUTEX_UNLOCK(&stream_.mutex);
return;
}
DWORD bufferSize2 = 0;
char *buffer;
long buffer_bytes;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ DsHandle *handles = (DsHandle *) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
// Setup parameters and do buffer conversion if necessary.
- if (stream->doConvertBuffer[0]) {
- convertStreamBuffer(stream, OUTPUT);
- buffer = stream->deviceBuffer;
- buffer_bytes = stream->bufferSize * stream->nDeviceChannels[0];
- buffer_bytes *= formatBytes(stream->deviceFormat[0]);
+ if (stream_.doConvertBuffer[0]) {
+ convertStreamBuffer(OUTPUT);
+ buffer = stream_.deviceBuffer;
+ buffer_bytes = stream_.bufferSize * stream_.nDeviceChannels[0];
+ buffer_bytes *= formatBytes(stream_.deviceFormat[0]);
}
else {
- buffer = stream->userBuffer;
- buffer_bytes = stream->bufferSize * stream->nUserChannels[0];
- buffer_bytes *= formatBytes(stream->userFormat);
+ buffer = stream_.userBuffer;
+ buffer_bytes = stream_.bufferSize * stream_.nUserChannels[0];
+ buffer_bytes *= formatBytes(stream_.userFormat);
}
// No byte swapping necessary in DirectSound implementation.
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- UINT nextWritePos = stream->handle[0].bufferPointer;
- DWORD dsBufferSize = buffer_bytes * stream->nBuffers;
+ LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer;
+ UINT nextWritePos = handles[0].bufferPointer;
+ DWORD dsBufferSize = buffer_bytes * stream_.nBuffers;
// Find out where the read and "safe write" pointers are.
result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to get current position (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
// A "fudgefactor" less than 1 is used because it was found
// that sleeping too long was MUCH worse than sleeping for
// several shorter periods.
- float millis = (endWrite - currentPos) * 900.0;
- millis /= ( formatBytes(stream->deviceFormat[0]) * stream->sampleRate);
+ double millis = (endWrite - currentPos) * 900.0;
+ millis /= ( formatBytes(stream_.deviceFormat[0]) * stream_.sampleRate);
if ( millis < 1.0 ) millis = 1.0;
Sleep( (DWORD) millis );
// Wake up, find out where we are now
result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos );
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to get current position (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1,
&bufferSize1, &buffer2, &bufferSize2, 0);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS buffer during playback (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to lock buffer during playback (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
// Update our buffer offset and unlock sound buffer
dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS buffer during playback (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to unlock buffer during playback (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize;
- stream->handle[0].bufferPointer = nextWritePos;
+ handles[0].bufferPointer = nextWritePos;
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
// Setup parameters.
- if (stream->doConvertBuffer[1]) {
- buffer = stream->deviceBuffer;
- buffer_bytes = stream->bufferSize * stream->nDeviceChannels[1];
- buffer_bytes *= formatBytes(stream->deviceFormat[1]);
+ if (stream_.doConvertBuffer[1]) {
+ buffer = stream_.deviceBuffer;
+ buffer_bytes = stream_.bufferSize * stream_.nDeviceChannels[1];
+ buffer_bytes *= formatBytes(stream_.deviceFormat[1]);
}
else {
- buffer = stream->userBuffer;
- buffer_bytes = stream->bufferSize * stream->nUserChannels[1];
- buffer_bytes *= formatBytes(stream->userFormat);
+ buffer = stream_.userBuffer;
+ buffer_bytes = stream_.bufferSize * stream_.nUserChannels[1];
+ buffer_bytes *= formatBytes(stream_.userFormat);
}
- LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- UINT nextReadPos = stream->handle[1].bufferPointer;
- DWORD dsBufferSize = buffer_bytes * stream->nBuffers;
+ LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer;
+ UINT nextReadPos = handles[1].bufferPointer;
+ DWORD dsBufferSize = buffer_bytes * stream_.nBuffers;
// Find out where the write and "safe read" pointers are.
result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to get current capture position (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
// Check whether the entire write region is behind the play pointer.
while ( safePos < endRead ) {
// See comments for playback.
- float millis = (endRead - safePos) * 900.0;
- millis /= ( formatBytes(stream->deviceFormat[1]) * stream->sampleRate);
+ double millis = (endRead - safePos) * 900.0;
+ millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.sampleRate);
if ( millis < 1.0 ) millis = 1.0;
Sleep( (DWORD) millis );
// Wake up, find out where we are now
result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos );
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to get current capture position (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
result = dsBuffer->Lock (nextReadPos, buffer_bytes, &buffer1,
&bufferSize1, &buffer2, &bufferSize2, 0);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS buffer during capture (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to lock buffer during capture (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
nextReadPos = (nextReadPos + bufferSize1 + bufferSize2) % dsBufferSize;
dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2);
if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS buffer during capture (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
+ sprintf(message_, "RtApiDs: Unable to unlock buffer during capture (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), getErrorString(result));
error(RtError::DRIVER_ERROR);
}
- stream->handle[1].bufferPointer = nextReadPos;
+ handles[1].bufferPointer = nextReadPos;
// No byte swapping necessary in DirectSound implementation.
// Do buffer conversion if necessary.
- if (stream->doConvertBuffer[1])
- convertStreamBuffer(stream, INPUT);
+ if (stream_.doConvertBuffer[1])
+ convertStreamBuffer(INPUT);
}
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
- if (stream->callbackInfo.usingCallback && stopStream)
- this->stopStream(streamId);
+ if (stream_.callbackInfo.usingCallback && stopStream)
+ this->stopStream();
}
// Definitions for utility functions and callbacks
extern "C" unsigned __stdcall callbackHandler(void *ptr)
{
- CALLBACK_INFO *info = (CALLBACK_INFO *) ptr;
- RtAudio *object = (RtAudio *) info->object;
- int stream = info->streamId;
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiDs *object = (RtApiDs *) info->object;
bool *usingCallback = &info->usingCallback;
while ( *usingCallback ) {
try {
- object->tickStream(stream);
+ object->tickStream();
}
catch (RtError &exception) {
- fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n",
- exception.getMessage());
+ fprintf(stderr, "\nRtApiDs: callback thread error (%s) ... closing thread.\n\n",
+ exception.getMessageString());
break;
}
}
return 0;
}
-void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- CALLBACK_INFO *info = (CALLBACK_INFO *) &stream->callbackInfo;
- if ( info->usingCallback ) {
- sprintf(message, "RtAudio: A callback is already set for this stream!");
- error(RtError::WARNING);
- return;
- }
-
- info->callback = (void *) callback;
- info->userData = userData;
- info->usingCallback = true;
- info->object = (void *) this;
- info->streamId = streamId;
-
- unsigned thread_id;
- info->thread = _beginthreadex(NULL, 0, &callbackHandler,
- &stream->callbackInfo, 0, &thread_id);
- if (info->thread == 0) {
- info->usingCallback = false;
- sprintf(message, "RtAudio: error starting callback thread!");
- error(RtError::THREAD_ERROR);
- }
-
- // When spawning multiple threads in quick succession, it appears to be
- // necessary to wait a bit for each to initialize ... another windoism!
- Sleep(1);
-}
-
static bool CALLBACK deviceCountCallback(LPGUID lpguid,
LPCSTR lpcstrDescription,
LPCSTR lpcstrModule,
}
//******************** End of __WINDOWS_DS__ *********************//
+#endif
-#elif defined(__IRIX_AL__) // SGI's AL API for IRIX
+#if defined(__IRIX_AL__) // SGI's AL API for IRIX
+#include <dmedia/audio.h>
#include <unistd.h>
#include <errno.h>
-void RtAudio :: initialize(void)
+extern "C" void *callbackHandler(void * ptr);
+
+RtApiAl :: RtApiAl()
+{
+ this->initialize();
+
+ if (nDevices_ <= 0) {
+ sprintf(message_, "RtApiAl: no Irix AL audio devices found!");
+ error(RtError::NO_DEVICES_FOUND);
+ }
+}
+
+RtApiAl :: ~RtApiAl()
+{
+ // The subclass destructor gets called before the base class
+ // destructor, so close any existing streams before deallocating
+ // apiDeviceId memory.
+ if ( stream_.mode != UNINITIALIZED ) closeStream();
+
+ // Free our allocated apiDeviceId memory.
+ long *id;
+ for ( unsigned int i=0; i<devices_.size(); i++ ) {
+ id = (long *) devices_[i].apiDeviceId;
+ if (id) free(id);
+ }
+}
+
+void RtApiAl :: initialize(void)
{
// Count cards and devices
- nDevices = 0;
+ nDevices_ = 0;
// Determine the total number of input and output devices.
- nDevices = alQueryValues(AL_SYSTEM, AL_DEVICES, 0, 0, 0, 0);
- if (nDevices < 0) {
- sprintf(message, "RtAudio: AL error counting devices: %s.",
+ nDevices_ = alQueryValues(AL_SYSTEM, AL_DEVICES, 0, 0, 0, 0);
+ if (nDevices_ < 0) {
+ sprintf(message_, "RtApiAl: error counting devices: %s.",
alGetErrorString(oserror()));
error(RtError::DRIVER_ERROR);
}
- if (nDevices <= 0) return;
+ if (nDevices_ <= 0) return;
- ALvalue *vls = (ALvalue *) new ALvalue[nDevices];
+ ALvalue *vls = (ALvalue *) new ALvalue[nDevices_];
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
-
- // Write device ascii identifiers and resource ids to device info
- // structure.
- char name[32];
+ // Create our list of devices and write their ascii identifiers and resource ids.
+ char name[64];
int outs, ins, i;
ALpv pvs[1];
pvs[0].param = AL_NAME;
pvs[0].value.ptr = name;
- pvs[0].sizeIn = 32;
+ pvs[0].sizeIn = 64;
+ RtApiDevice device;
+ long *id;
- outs = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, vls, nDevices, 0, 0);
+ outs = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, vls, nDevices_, 0, 0);
if (outs < 0) {
- sprintf(message, "RtAudio: AL error getting output devices: %s.",
+ delete [] vls;
+ sprintf(message_, "RtApiAl: error getting output devices: %s.",
alGetErrorString(oserror()));
error(RtError::DRIVER_ERROR);
}
for (i=0; i<outs; i++) {
if (alGetParams(vls[i].i, pvs, 1) < 0) {
- sprintf(message, "RtAudio: AL error querying output devices: %s.",
+ delete [] vls;
+ sprintf(message_, "RtApiAl: error querying output devices: %s.",
alGetErrorString(oserror()));
error(RtError::DRIVER_ERROR);
}
- strncpy(devices[i].name, name, 32);
- devices[i].id[0] = vls[i].i;
+ device.name.erase();
+ device.name.append( (const char *)name, strlen(name)+1);
+ devices_.push_back(device);
+ id = (long *) calloc(2, sizeof(long));
+ id[0] = vls[i].i;
+ devices_[i].apiDeviceId = (void *) id;
}
- ins = alQueryValues(AL_SYSTEM, AL_DEFAULT_INPUT, &vls[outs], nDevices-outs, 0, 0);
+ ins = alQueryValues(AL_SYSTEM, AL_DEFAULT_INPUT, &vls[outs], nDevices_-outs, 0, 0);
if (ins < 0) {
- sprintf(message, "RtAudio: AL error getting input devices: %s.",
+ delete [] vls;
+ sprintf(message_, "RtApiAl: error getting input devices: %s.",
alGetErrorString(oserror()));
error(RtError::DRIVER_ERROR);
}
for (i=outs; i<ins+outs; i++) {
if (alGetParams(vls[i].i, pvs, 1) < 0) {
- sprintf(message, "RtAudio: AL error querying input devices: %s.",
+ delete [] vls;
+ sprintf(message_, "RtApiAl: error querying input devices: %s.",
alGetErrorString(oserror()));
error(RtError::DRIVER_ERROR);
}
- strncpy(devices[i].name, name, 32);
- devices[i].id[1] = vls[i].i;
+ device.name.erase();
+ device.name.append( (const char *)name, strlen(name)+1);
+ devices_.push_back(device);
+ id = (long *) calloc(2, sizeof(long));
+ id[1] = vls[i].i;
+ devices_[i].apiDeviceId = (void *) id;
}
delete [] vls;
-
- return;
}
-int RtAudio :: getDefaultInputDevice(void)
+int RtApiAl :: getDefaultInputDevice(void)
{
ALvalue value;
+ long *id;
int result = alQueryValues(AL_SYSTEM, AL_DEFAULT_INPUT, &value, 1, 0, 0);
if (result < 0) {
- sprintf(message, "RtAudio: AL error getting default input device id: %s.",
+ sprintf(message_, "RtApiAl: error getting default input device id: %s.",
alGetErrorString(oserror()));
error(RtError::WARNING);
}
else {
- for ( int i=0; i<nDevices; i++ )
- if ( devices[i].id[1] == value.i ) return i;
+ for ( unsigned int i=0; i<devices_.size(); i++ ) {
+ id = (long *) devices_[i].apiDeviceId;
+ if ( id[1] == value.i ) return i;
+ }
}
return 0;
}
-int RtAudio :: getDefaultOutputDevice(void)
+int RtApiAl :: getDefaultOutputDevice(void)
{
ALvalue value;
+ long *id;
int result = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, &value, 1, 0, 0);
if (result < 0) {
- sprintf(message, "RtAudio: AL error getting default output device id: %s.",
+ sprintf(message_, "RtApiAl: error getting default output device id: %s.",
alGetErrorString(oserror()));
error(RtError::WARNING);
}
else {
- for ( int i=0; i<nDevices; i++ )
- if ( devices[i].id[0] == value.i ) return i;
+ for ( unsigned int i=0; i<devices_.size(); i++ ) {
+ id = (long *) devices_[i].apiDeviceId;
+ if ( id[0] == value.i ) return i;
+ }
}
return 0;
}
-void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info)
+void RtApiAl :: probeDeviceInfo(RtApiDevice *info)
{
- int resource, result, i;
+ int result;
+ long resource;
ALvalue value;
ALparamInfo pinfo;
// Get output resource ID if it exists.
- resource = info->id[0];
+ long *id = (long *) info->apiDeviceId;
+ resource = id[0];
if (resource > 0) {
// Probe output device parameters.
result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0);
if (result < 0) {
- sprintf(message, "RtAudio: AL error getting device (%s) channels: %s.",
- info->name, alGetErrorString(oserror()));
+ sprintf(message_, "RtApiAl: error getting device (%s) channels: %s.",
+ info->name.c_str(), alGetErrorString(oserror()));
error(RtError::WARNING);
}
else {
result = alGetParamInfo(resource, AL_RATE, &pinfo);
if (result < 0) {
- sprintf(message, "RtAudio: AL error getting device (%s) rates: %s.",
- info->name, alGetErrorString(oserror()));
+ sprintf(message_, "RtApiAl: error getting device (%s) rates: %s.",
+ info->name.c_str(), alGetErrorString(oserror()));
error(RtError::WARNING);
}
else {
- info->nSampleRates = 0;
- for (i=0; i<MAX_SAMPLE_RATES; i++) {
- if ( SAMPLE_RATES[i] >= pinfo.min.i && SAMPLE_RATES[i] <= pinfo.max.i ) {
- info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i];
- info->nSampleRates++;
- }
+ info->sampleRates.clear();
+ for (unsigned int k=0; k<MAX_SAMPLE_RATES; k++) {
+ if ( SAMPLE_RATES[k] >= pinfo.min.i && SAMPLE_RATES[k] <= pinfo.max.i )
+ info->sampleRates.push_back( SAMPLE_RATES[k] );
}
}
// The AL library supports all our formats, except 24-bit and 32-bit ints.
- info->nativeFormats = (RTAUDIO_FORMAT) 51;
+ info->nativeFormats = (RtAudioFormat) 51;
}
// Now get input resource ID if it exists.
- resource = info->id[1];
+ resource = id[1];
if (resource > 0) {
// Probe input device parameters.
result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0);
if (result < 0) {
- sprintf(message, "RtAudio: AL error getting device (%s) channels: %s.",
- info->name, alGetErrorString(oserror()));
+ sprintf(message_, "RtApiAl: error getting device (%s) channels: %s.",
+ info->name.c_str(), alGetErrorString(oserror()));
error(RtError::WARNING);
}
else {
result = alGetParamInfo(resource, AL_RATE, &pinfo);
if (result < 0) {
- sprintf(message, "RtAudio: AL error getting device (%s) rates: %s.",
- info->name, alGetErrorString(oserror()));
+ sprintf(message_, "RtApiAl: error getting device (%s) rates: %s.",
+ info->name.c_str(), alGetErrorString(oserror()));
error(RtError::WARNING);
}
else {
// overwrite the rates determined for the output device. Since
// the input device is most likely to be more limited than the
// output device, this is ok.
- info->nSampleRates = 0;
- for (i=0; i<MAX_SAMPLE_RATES; i++) {
- if ( SAMPLE_RATES[i] >= pinfo.min.i && SAMPLE_RATES[i] <= pinfo.max.i ) {
- info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i];
- info->nSampleRates++;
- }
+ info->sampleRates.clear();
+ for (unsigned int k=0; k<MAX_SAMPLE_RATES; k++) {
+ if ( SAMPLE_RATES[k] >= pinfo.min.i && SAMPLE_RATES[k] <= pinfo.max.i )
+ info->sampleRates.push_back( SAMPLE_RATES[k] );
}
}
// The AL library supports all our formats, except 24-bit and 32-bit ints.
- info->nativeFormats = (RTAUDIO_FORMAT) 51;
+ info->nativeFormats = (RtAudioFormat) 51;
}
if ( info->maxInputChannels == 0 && info->maxOutputChannels == 0 )
return;
- if ( info->nSampleRates == 0 )
+ if ( info->sampleRates.size() == 0 )
return;
// Determine duplex status.
return;
}
-bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
- STREAM_MODE mode, int channels,
- int sampleRate, RTAUDIO_FORMAT format,
+bool RtApiAl :: probeDeviceOpen(int device, StreamMode mode, int channels,
+ int sampleRate, RtAudioFormat format,
int *bufferSize, int numberOfBuffers)
{
- int result, resource, nBuffers;
+ int result, nBuffers;
+ long resource;
ALconfig al_config;
ALport port;
ALpv pvs[2];
+ long *id = (long *) devices_[device].apiDeviceId;
// Get a new ALconfig structure.
al_config = alNewConfig();
if ( !al_config ) {
- sprintf(message,"RtAudio: can't get AL config: %s.",
+ sprintf(message_,"RtApiAl: can't get AL config: %s.",
alGetErrorString(oserror()));
error(RtError::WARNING);
return FAILURE;
// Set the channels.
result = alSetChannels(al_config, channels);
if ( result < 0 ) {
- sprintf(message,"RtAudio: can't set %d channels in AL config: %s.",
+ alFreeConfig(al_config);
+ sprintf(message_,"RtApiAl: can't set %d channels in AL config: %s.",
channels, alGetErrorString(oserror()));
error(RtError::WARNING);
return FAILURE;
buffer_size = alGetQueueSize(al_config);
result = alSetQueueSize(al_config, buffer_size);
if ( result < 0 ) {
- sprintf(message,"RtAudio: can't set buffer size (%ld) in AL config: %s.",
+ alFreeConfig(al_config);
+ sprintf(message_,"RtApiAl: can't set buffer size (%ld) in AL config: %s.",
buffer_size, alGetErrorString(oserror()));
error(RtError::WARNING);
return FAILURE;
}
// Set the data format.
- stream->userFormat = format;
- stream->deviceFormat[mode] = format;
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = format;
if (format == RTAUDIO_SINT8) {
result = alSetSampFmt(al_config, AL_SAMPFMT_TWOSCOMP);
result = alSetWidth(al_config, AL_SAMPLE_8);
// The AL library uses the lower 3 bytes, so we'll need to do our
// own conversion.
result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT);
- stream->deviceFormat[mode] = RTAUDIO_FLOAT32;
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
}
else if (format == RTAUDIO_SINT32) {
// The AL library doesn't seem to support the 32-bit integer
// format, so we'll need to do our own conversion.
result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT);
- stream->deviceFormat[mode] = RTAUDIO_FLOAT32;
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
}
else if (format == RTAUDIO_FLOAT32)
result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT);
result = alSetSampFmt(al_config, AL_SAMPFMT_DOUBLE);
if ( result == -1 ) {
- sprintf(message,"RtAudio: AL error setting sample format in AL config: %s.",
+ alFreeConfig(al_config);
+ sprintf(message_,"RtApiAl: error setting sample format in AL config: %s.",
alGetErrorString(oserror()));
error(RtError::WARNING);
return FAILURE;
if (device == 0)
resource = AL_DEFAULT_OUTPUT;
else
- resource = devices[device].id[0];
+ resource = id[0];
result = alSetDevice(al_config, resource);
if ( result == -1 ) {
- sprintf(message,"RtAudio: AL error setting device (%s) in AL config: %s.",
- devices[device].name, alGetErrorString(oserror()));
+ alFreeConfig(al_config);
+ sprintf(message_,"RtApiAl: error setting device (%s) in AL config: %s.",
+ devices_[device].name.c_str(), alGetErrorString(oserror()));
error(RtError::WARNING);
return FAILURE;
}
// Open the port.
- port = alOpenPort("RtAudio Output Port", "w", al_config);
+ port = alOpenPort("RtApiAl Output Port", "w", al_config);
if( !port ) {
- sprintf(message,"RtAudio: AL error opening output port: %s.",
+ alFreeConfig(al_config);
+ sprintf(message_,"RtApiAl: error opening output port: %s.",
alGetErrorString(oserror()));
error(RtError::WARNING);
return FAILURE;
result = alSetParams(resource, pvs, 2);
if ( result < 0 ) {
alClosePort(port);
- sprintf(message,"RtAudio: AL error setting sample rate (%d) for device (%s): %s.",
- sampleRate, devices[device].name, alGetErrorString(oserror()));
+ alFreeConfig(al_config);
+ sprintf(message_,"RtApiAl: error setting sample rate (%d) for device (%s): %s.",
+ sampleRate, devices_[device].name.c_str(), alGetErrorString(oserror()));
error(RtError::WARNING);
return FAILURE;
}
if (device == 0)
resource = AL_DEFAULT_INPUT;
else
- resource = devices[device].id[1];
+ resource = id[1];
result = alSetDevice(al_config, resource);
if ( result == -1 ) {
- sprintf(message,"RtAudio: AL error setting device (%s) in AL config: %s.",
- devices[device].name, alGetErrorString(oserror()));
+ alFreeConfig(al_config);
+ sprintf(message_,"RtApiAl: error setting device (%s) in AL config: %s.",
+ devices_[device].name.c_str(), alGetErrorString(oserror()));
error(RtError::WARNING);
return FAILURE;
}
// Open the port.
- port = alOpenPort("RtAudio Output Port", "r", al_config);
+ port = alOpenPort("RtApiAl Input Port", "r", al_config);
if( !port ) {
- sprintf(message,"RtAudio: AL error opening input port: %s.",
+ alFreeConfig(al_config);
+ sprintf(message_,"RtApiAl: error opening input port: %s.",
alGetErrorString(oserror()));
error(RtError::WARNING);
return FAILURE;
result = alSetParams(resource, pvs, 2);
if ( result < 0 ) {
alClosePort(port);
- sprintf(message,"RtAudio: AL error setting sample rate (%d) for device (%s): %s.",
- sampleRate, devices[device].name, alGetErrorString(oserror()));
+ alFreeConfig(al_config);
+ sprintf(message_,"RtApiAl: error setting sample rate (%d) for device (%s): %s.",
+ sampleRate, devices_[device].name.c_str(), alGetErrorString(oserror()));
error(RtError::WARNING);
return FAILURE;
}
alFreeConfig(al_config);
- stream->nUserChannels[mode] = channels;
- stream->nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+ stream_.nDeviceChannels[mode] = channels;
+
+ // Save stream handle.
+ ALport *handle = (ALport *) stream_.apiHandle;
+ if ( handle == 0 ) {
+ handle = (ALport *) calloc(2, sizeof(ALport));
+ if ( handle == NULL ) {
+ sprintf(message_, "RtApiAl: Irix Al error allocating handle memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
+ stream_.apiHandle = (void *) handle;
+ handle[0] = 0;
+ handle[1] = 0;
+ }
+ handle[mode] = port;
- // Set handle and flags for buffer conversion
- stream->handle[mode] = port;
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
+ // Set flags for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if (stream_.userFormat != stream_.deviceFormat[mode])
+ stream_.doConvertBuffer[mode] = true;
// Allocate necessary internal buffers
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
+ if ( stream_.nUserChannels[0] != stream_.nUserChannels[1] ) {
long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
+ if (stream_.nUserChannels[0] >= stream_.nUserChannels[1])
+ buffer_bytes = stream_.nUserChannels[0];
else
- buffer_bytes = stream->nUserChannels[1];
-
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL)
- goto memory_error;
+ buffer_bytes = stream_.nUserChannels[1];
+
+ buffer_bytes *= *bufferSize * formatBytes(stream_.userFormat);
+ if (stream_.userBuffer) free(stream_.userBuffer);
+ stream_.userBuffer = (char *) calloc(buffer_bytes, 1);
+ if (stream_.userBuffer == NULL) {
+ sprintf(message_, "RtApiAl: error allocating user buffer memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
}
- if ( stream->doConvertBuffer[mode] ) {
+ if ( stream_.doConvertBuffer[mode] ) {
long buffer_bytes;
bool makeBuffer = true;
if ( mode == OUTPUT )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
+ buffer_bytes = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
else { // mode == INPUT
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == OUTPUT && stream->deviceBuffer ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
+ buffer_bytes = stream_.nDeviceChannels[1] * formatBytes(stream_.deviceFormat[1]);
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ long bytes_out = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
if ( buffer_bytes < bytes_out ) makeBuffer = false;
}
}
if ( makeBuffer ) {
buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL)
- goto memory_error;
+ if (stream_.deviceBuffer) free(stream_.deviceBuffer);
+ stream_.deviceBuffer = (char *) calloc(buffer_bytes, 1);
+ if (stream_.deviceBuffer == NULL) {
+ sprintf(message_, "RtApiAl: error allocating device buffer memory (%s).",
+ devices_[device].name.c_str());
+ goto error;
+ }
}
}
- stream->device[mode] = device;
- stream->state = STREAM_STOPPED;
- if ( stream->mode == OUTPUT && mode == INPUT )
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
// We had already set up an output stream.
- stream->mode = DUPLEX;
+ stream_.mode = DUPLEX;
else
- stream->mode = mode;
- stream->nBuffers = nBuffers;
- stream->bufferSize = *bufferSize;
- stream->sampleRate = sampleRate;
+ stream_.mode = mode;
+ stream_.nBuffers = nBuffers;
+ stream_.bufferSize = *bufferSize;
+ stream_.sampleRate = sampleRate;
return SUCCESS;
- memory_error:
- if (stream->handle[0]) {
- alClosePort(stream->handle[0]);
- stream->handle[0] = 0;
- }
- if (stream->handle[1]) {
- alClosePort(stream->handle[1]);
- stream->handle[1] = 0;
+ error:
+ if (handle) {
+ if (handle[0])
+ alClosePort(handle[0]);
+ if (handle[1])
+ alClosePort(handle[1]);
+ free(handle);
+ stream_.apiHandle = 0;
}
- if (stream->userBuffer) {
- free(stream->userBuffer);
- stream->userBuffer = 0;
+
+ if (stream_.userBuffer) {
+ free(stream_.userBuffer);
+ stream_.userBuffer = 0;
}
- sprintf(message, "RtAudio: ALSA error allocating buffer memory for device (%s).",
- devices[device].name);
+
error(RtError::WARNING);
return FAILURE;
}
-void RtAudio :: closeStream(int streamId)
+void RtApiAl :: closeStream()
{
// We don't want an exception to be thrown here because this
// function is called by our class destructor. So, do our own
// streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
+ if ( stream_.mode == UNINITIALIZED ) {
+ sprintf(message_, "RtApiAl::closeStream(): no open stream to close!");
error(RtError::WARNING);
return;
}
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
-
- if (stream->callbackInfo.usingCallback) {
- pthread_cancel(stream->callbackInfo.thread);
- pthread_join(stream->callbackInfo.thread, NULL);
+ ALport *handle = (ALport *) stream_.apiHandle;
+ if (stream_.state == STREAM_RUNNING) {
+ int buffer_size = stream_.bufferSize * stream_.nBuffers;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX)
+ alDiscardFrames(handle[0], buffer_size);
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX)
+ alDiscardFrames(handle[1], buffer_size);
+ stream_.state = STREAM_STOPPED;
}
- pthread_mutex_destroy(&stream->mutex);
-
- if (stream->handle[0])
- alClosePort(stream->handle[0]);
+ if (stream_.callbackInfo.usingCallback) {
+ stream_.callbackInfo.usingCallback = false;
+ pthread_join(stream_.callbackInfo.thread, NULL);
+ }
- if (stream->handle[1])
- alClosePort(stream->handle[1]);
+ if (handle) {
+ if (handle[0]) alClosePort(handle[0]);
+ if (handle[1]) alClosePort(handle[1]);
+ free(handle);
+ stream_.apiHandle = 0;
+ }
- if (stream->userBuffer)
- free(stream->userBuffer);
+ if (stream_.userBuffer) {
+ free(stream_.userBuffer);
+ stream_.userBuffer = 0;
+ }
- if (stream->deviceBuffer)
- free(stream->deviceBuffer);
+ if (stream_.deviceBuffer) {
+ free(stream_.deviceBuffer);
+ stream_.deviceBuffer = 0;
+ }
- free(stream);
- streams.erase(streamId);
+ stream_.mode = UNINITIALIZED;
}
-void RtAudio :: startStream(int streamId)
+void RtApiAl :: startStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+ if (stream_.state == STREAM_RUNNING) return;
- if (stream->state == STREAM_RUNNING)
- return;
+ MUTEX_LOCK(&stream_.mutex);
// The AL port is ready as soon as it is opened.
- stream->state = STREAM_RUNNING;
+ stream_.state = STREAM_RUNNING;
+
+ MUTEX_UNLOCK(&stream_.mutex);
}
-void RtAudio :: stopStream(int streamId)
+void RtApiAl :: stopStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
+ verifyStream();
+ if (stream_.state == STREAM_STOPPED) return;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ // Change the state before the lock to improve shutdown response
+ // when using a callback.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK(&stream_.mutex);
- int result;
- int buffer_size = stream->bufferSize * stream->nBuffers;
+ int result, buffer_size = stream_.bufferSize * stream_.nBuffers;
+ ALport *handle = (ALport *) stream_.apiHandle;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX)
- alZeroFrames(stream->handle[0], buffer_size);
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX)
+ alZeroFrames(handle[0], buffer_size);
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- result = alDiscardFrames(stream->handle[1], buffer_size);
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ result = alDiscardFrames(handle[1], buffer_size);
if (result == -1) {
- sprintf(message, "RtAudio: AL error draining stream device (%s): %s.",
- devices[stream->device[1]].name, alGetErrorString(oserror()));
+ sprintf(message_, "RtApiAl: error draining stream device (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), alGetErrorString(oserror()));
error(RtError::DRIVER_ERROR);
}
}
- stream->state = STREAM_STOPPED;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
}
-void RtAudio :: abortStream(int streamId)
+void RtApiAl :: abortStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
+ verifyStream();
+ if (stream_.state == STREAM_STOPPED) return;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ // Change the state before the lock to improve shutdown response
+ // when using a callback.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK(&stream_.mutex);
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ ALport *handle = (ALport *) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
- int buffer_size = stream->bufferSize * stream->nBuffers;
- int result = alDiscardFrames(stream->handle[0], buffer_size);
+ int buffer_size = stream_.bufferSize * stream_.nBuffers;
+ int result = alDiscardFrames(handle[0], buffer_size);
if (result == -1) {
- sprintf(message, "RtAudio: AL error aborting stream device (%s): %s.",
- devices[stream->device[0]].name, alGetErrorString(oserror()));
+ sprintf(message_, "RtApiAl: error aborting stream device (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), alGetErrorString(oserror()));
error(RtError::DRIVER_ERROR);
}
}
// There is no clear action to take on the input stream, since the
// port will continue to run in any event.
- stream->state = STREAM_STOPPED;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
}
-int RtAudio :: streamWillBlock(int streamId)
+int RtApiAl :: streamWillBlock()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
- MUTEX_LOCK(&stream->mutex);
+ if (stream_.state == STREAM_STOPPED) return 0;
- int frames = 0;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ MUTEX_LOCK(&stream_.mutex);
+ int frames = 0;
int err = 0;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- err = alGetFillable(stream->handle[0]);
+ ALport *handle = (ALport *) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ err = alGetFillable(handle[0]);
if (err < 0) {
- sprintf(message, "RtAudio: AL error getting available frames for stream (%s): %s.",
- devices[stream->device[0]].name, alGetErrorString(oserror()));
+ sprintf(message_, "RtApiAl: error getting available frames for stream (%s): %s.",
+ devices_[stream_.device[0]].name.c_str(), alGetErrorString(oserror()));
error(RtError::DRIVER_ERROR);
}
}
frames = err;
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- err = alGetFilled(stream->handle[1]);
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ err = alGetFilled(handle[1]);
if (err < 0) {
- sprintf(message, "RtAudio: AL error getting available frames for stream (%s): %s.",
- devices[stream->device[1]].name, alGetErrorString(oserror()));
+ sprintf(message_, "RtApiAl: error getting available frames for stream (%s): %s.",
+ devices_[stream_.device[1]].name.c_str(), alGetErrorString(oserror()));
error(RtError::DRIVER_ERROR);
}
if (frames > err) frames = err;
}
- frames = stream->bufferSize - frames;
+ frames = stream_.bufferSize - frames;
if (frames < 0) frames = 0;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
return frames;
}
-void RtAudio :: tickStream(int streamId)
+void RtApiAl :: tickStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
int stopStream = 0;
- if (stream->state == STREAM_STOPPED) {
- if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds
+ if (stream_.state == STREAM_STOPPED) {
+ if (stream_.callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds
return;
}
- else if (stream->callbackInfo.usingCallback) {
- RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback;
- stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData);
+ else if (stream_.callbackInfo.usingCallback) {
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ stopStream = callback(stream_.userBuffer, stream_.bufferSize, stream_.callbackInfo.userData);
}
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK(&stream_.mutex);
// The state might change while waiting on a mutex.
- if (stream->state == STREAM_STOPPED)
+ if (stream_.state == STREAM_STOPPED)
goto unlock;
char *buffer;
int channels;
- RTAUDIO_FORMAT format;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ RtAudioFormat format;
+ ALport *handle = (ALport *) stream_.apiHandle;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
// Setup parameters and do buffer conversion if necessary.
- if (stream->doConvertBuffer[0]) {
- convertStreamBuffer(stream, OUTPUT);
- buffer = stream->deviceBuffer;
- channels = stream->nDeviceChannels[0];
- format = stream->deviceFormat[0];
+ if (stream_.doConvertBuffer[0]) {
+ convertStreamBuffer(OUTPUT);
+ buffer = stream_.deviceBuffer;
+ channels = stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
}
else {
- buffer = stream->userBuffer;
- channels = stream->nUserChannels[0];
- format = stream->userFormat;
+ buffer = stream_.userBuffer;
+ channels = stream_.nUserChannels[0];
+ format = stream_.userFormat;
}
// Do byte swapping if necessary.
- if (stream->doByteSwap[0])
- byteSwapBuffer(buffer, stream->bufferSize * channels, format);
+ if (stream_.doByteSwap[0])
+ byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
// Write interleaved samples to device.
- alWriteFrames(stream->handle[0], buffer, stream->bufferSize);
+ alWriteFrames(handle[0], buffer, stream_.bufferSize);
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
+ if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
// Setup parameters.
- if (stream->doConvertBuffer[1]) {
- buffer = stream->deviceBuffer;
- channels = stream->nDeviceChannels[1];
- format = stream->deviceFormat[1];
+ if (stream_.doConvertBuffer[1]) {
+ buffer = stream_.deviceBuffer;
+ channels = stream_.nDeviceChannels[1];
+ format = stream_.deviceFormat[1];
}
else {
- buffer = stream->userBuffer;
- channels = stream->nUserChannels[1];
- format = stream->userFormat;
+ buffer = stream_.userBuffer;
+ channels = stream_.nUserChannels[1];
+ format = stream_.userFormat;
}
// Read interleaved samples from device.
- alReadFrames(stream->handle[1], buffer, stream->bufferSize);
+ alReadFrames(handle[1], buffer, stream_.bufferSize);
// Do byte swapping if necessary.
- if (stream->doByteSwap[1])
- byteSwapBuffer(buffer, stream->bufferSize * channels, format);
+ if (stream_.doByteSwap[1])
+ byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
// Do buffer conversion if necessary.
- if (stream->doConvertBuffer[1])
- convertStreamBuffer(stream, INPUT);
+ if (stream_.doConvertBuffer[1])
+ convertStreamBuffer(INPUT);
}
unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
+
+ if (stream_.callbackInfo.usingCallback && stopStream)
+ this->stopStream();
+}
+
+void RtApiAl :: setStreamCallback(RtAudioCallback callback, void *userData)
+{
+ verifyStream();
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ if ( info->usingCallback ) {
+ sprintf(message_, "RtApiAl: A callback is already set for this stream!");
+ error(RtError::WARNING);
+ return;
+ }
+
+ info->callback = (void *) callback;
+ info->userData = userData;
+ info->usingCallback = true;
+ info->object = (void *) this;
+
+ // Set the thread attributes for joinable and realtime scheduling
+ // priority. The higher priority will only take affect if the
+ // program is run as root or suid.
+ pthread_attr_t attr;
+ pthread_attr_init(&attr);
+ pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE);
+ pthread_attr_setschedpolicy(&attr, SCHED_RR);
+
+ int err = pthread_create(&info->thread, &attr, callbackHandler, &stream_.callbackInfo);
+ pthread_attr_destroy(&attr);
+ if (err) {
+ info->usingCallback = false;
+ sprintf(message_, "RtApiAl: error starting callback thread!");
+ error(RtError::THREAD_ERROR);
+ }
+}
+
+void RtApiAl :: cancelStreamCallback()
+{
+ verifyStream();
+
+ if (stream_.callbackInfo.usingCallback) {
- if (stream->callbackInfo.usingCallback && stopStream)
- this->stopStream(streamId);
+ if (stream_.state == STREAM_RUNNING)
+ stopStream();
+
+ MUTEX_LOCK(&stream_.mutex);
+
+ stream_.callbackInfo.usingCallback = false;
+ pthread_join(stream_.callbackInfo.thread, NULL);
+ stream_.callbackInfo.thread = 0;
+ stream_.callbackInfo.callback = NULL;
+ stream_.callbackInfo.userData = NULL;
+
+ MUTEX_UNLOCK(&stream_.mutex);
+ }
}
extern "C" void *callbackHandler(void *ptr)
{
- CALLBACK_INFO *info = (CALLBACK_INFO *) ptr;
- RtAudio *object = (RtAudio *) info->object;
- int stream = info->streamId;
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiAl *object = (RtApiAl *) info->object;
bool *usingCallback = &info->usingCallback;
while ( *usingCallback ) {
- pthread_testcancel();
try {
- object->tickStream(stream);
+ object->tickStream();
}
catch (RtError &exception) {
- fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n",
- exception.getMessage());
+ fprintf(stderr, "\nRtApiAl: callback thread error (%s) ... closing thread.\n\n",
+ exception.getMessageString());
break;
}
}
}
//******************** End of __IRIX_AL__ *********************//
-
#endif
// *************************************************** //
//
-// Private common (OS-independent) RtAudio methods.
+// Protected common (OS-independent) RtAudio methods.
//
// *************************************************** //
// This method can be modified to control the behavior of error
// message reporting and throwing.
-void RtAudio :: error(RtError::TYPE type)
+void RtApi :: error(RtError::Type type)
{
if (type == RtError::WARNING) {
- fprintf(stderr, "\n%s\n\n", message);
+ fprintf(stderr, "\n%s\n\n", message_);
}
else if (type == RtError::DEBUG_WARNING) {
#if defined(__RTAUDIO_DEBUG__)
- fprintf(stderr, "\n%s\n\n", message);
+ fprintf(stderr, "\n%s\n\n", message_);
#endif
}
else {
- fprintf(stderr, "\n%s\n\n", message);
- throw RtError(message, type);
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf(stderr, "\n%s\n\n", message_);
+#endif
+ throw RtError(std::string(message_), type);
}
}
-void *RtAudio :: verifyStream(int streamId)
+void RtApi :: verifyStream()
{
- // Verify the stream key.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
+ if ( stream_.mode == UNINITIALIZED ) {
+ sprintf(message_, "RtAudio: a stream was not previously opened!");
error(RtError::INVALID_STREAM);
}
-
- return streams[streamId];
}
-void RtAudio :: clearDeviceInfo(RTAUDIO_DEVICE *info)
+void RtApi :: clearDeviceInfo(RtApiDevice *info)
{
// Don't clear the name or DEVICE_ID fields here ... they are
// typically set prior to a call of this function.
info->minInputChannels = 0;
info->minDuplexChannels = 0;
info->hasDuplexSupport = false;
- info->nSampleRates = 0;
- for (int i=0; i<MAX_SAMPLE_RATES; i++)
- info->sampleRates[i] = 0;
+ info->sampleRates.clear();
info->nativeFormats = 0;
}
-int RtAudio :: formatBytes(RTAUDIO_FORMAT format)
+void RtApi :: clearStreamInfo()
+{
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_STOPPED;
+ stream_.sampleRate = 0;
+ stream_.bufferSize = 0;
+ stream_.nBuffers = 0;
+ stream_.userFormat = 0;
+ for ( int i=0; i<2; i++ ) {
+ stream_.device[i] = 0;
+ stream_.doConvertBuffer[i] = false;
+ stream_.deInterleave[i] = false;
+ stream_.doByteSwap[i] = false;
+ stream_.nUserChannels[i] = 0;
+ stream_.nDeviceChannels[i] = 0;
+ stream_.deviceFormat[i] = 0;
+ }
+}
+
+int RtApi :: formatBytes(RtAudioFormat format)
{
if (format == RTAUDIO_SINT16)
return 2;
else if (format == RTAUDIO_SINT8)
return 1;
- sprintf(message,"RtAudio: undefined format in formatBytes().");
+ sprintf(message_,"RtApi: undefined format in formatBytes().");
error(RtError::WARNING);
return 0;
}
-void RtAudio :: convertStreamBuffer(RTAUDIO_STREAM *stream, STREAM_MODE mode)
+void RtApi :: convertStreamBuffer( StreamMode mode )
{
// This method does format conversion, input/output channel compensation, and
// data interleaving/deinterleaving. 24-bit integers are assumed to occupy
// the upper three bytes of a 32-bit integer.
int j, jump_in, jump_out, channels;
- RTAUDIO_FORMAT format_in, format_out;
+ RtAudioFormat format_in, format_out;
char *input, *output;
if (mode == INPUT) { // convert device to user buffer
- input = stream->deviceBuffer;
- output = stream->userBuffer;
- jump_in = stream->nDeviceChannels[1];
- jump_out = stream->nUserChannels[1];
- format_in = stream->deviceFormat[1];
- format_out = stream->userFormat;
+ input = stream_.deviceBuffer;
+ output = stream_.userBuffer;
+ jump_in = stream_.nDeviceChannels[1];
+ jump_out = stream_.nUserChannels[1];
+ format_in = stream_.deviceFormat[1];
+ format_out = stream_.userFormat;
}
else { // convert user to device buffer
- input = stream->userBuffer;
- output = stream->deviceBuffer;
- jump_in = stream->nUserChannels[0];
- jump_out = stream->nDeviceChannels[0];
- format_in = stream->userFormat;
- format_out = stream->deviceFormat[0];
+ input = stream_.userBuffer;
+ output = stream_.deviceBuffer;
+ jump_in = stream_.nUserChannels[0];
+ jump_out = stream_.nDeviceChannels[0];
+ format_in = stream_.userFormat;
+ format_out = stream_.deviceFormat[0];
// clear our device buffer when in/out duplex device channels are different
- if ( stream->mode == DUPLEX &&
- stream->nDeviceChannels[0] != stream->nDeviceChannels[1] )
- memset(output, 0, stream->bufferSize * jump_out * formatBytes(format_out));
+ if ( stream_.mode == DUPLEX &&
+ stream_.nDeviceChannels[0] != stream_.nDeviceChannels[1] )
+ memset(output, 0, stream_.bufferSize * jump_out * formatBytes(format_out));
}
channels = (jump_in < jump_out) ? jump_in : jump_out;
// Set up the interleave/deinterleave offsets
std::vector<int> offset_in(channels);
std::vector<int> offset_out(channels);
- if (mode == INPUT && stream->deInterleave[1]) {
+ if (mode == INPUT && stream_.deInterleave[1]) {
for (int k=0; k<channels; k++) {
- offset_in[k] = k * stream->bufferSize;
+ offset_in[k] = k * stream_.bufferSize;
offset_out[k] = k;
jump_in = 1;
}
}
- else if (mode == OUTPUT && stream->deInterleave[0]) {
+ else if (mode == OUTPUT && stream_.deInterleave[0]) {
for (int k=0; k<channels; k++) {
offset_in[k] = k;
- offset_out[k] = k * stream->bufferSize;
+ offset_out[k] = k * stream_.bufferSize;
jump_out = 1;
}
}
}
if (format_out == RTAUDIO_FLOAT64) {
- FLOAT64 scale;
- FLOAT64 *out = (FLOAT64 *)output;
+ Float64 scale;
+ Float64 *out = (Float64 *)output;
if (format_in == RTAUDIO_SINT8) {
signed char *in = (signed char *)input;
scale = 1.0 / 128.0;
- for (int i=0; i<stream->bufferSize; i++) {
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT64) in[offset_in[j]];
+ out[offset_out[j]] = (Float64) in[offset_in[j]];
out[offset_out[j]] *= scale;
}
in += jump_in;
}
}
else if (format_in == RTAUDIO_SINT16) {
- INT16 *in = (INT16 *)input;
+ Int16 *in = (Int16 *)input;
scale = 1.0 / 32768.0;
- for (int i=0; i<stream->bufferSize; i++) {
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT64) in[offset_in[j]];
+ out[offset_out[j]] = (Float64) in[offset_in[j]];
out[offset_out[j]] *= scale;
}
in += jump_in;
}
}
else if (format_in == RTAUDIO_SINT24) {
- INT32 *in = (INT32 *)input;
+ Int32 *in = (Int32 *)input;
scale = 1.0 / 2147483648.0;
- for (int i=0; i<stream->bufferSize; i++) {
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT64) (in[offset_in[j]] & 0xffffff00);
+ out[offset_out[j]] = (Float64) (in[offset_in[j]] & 0xffffff00);
out[offset_out[j]] *= scale;
}
in += jump_in;
}
}
else if (format_in == RTAUDIO_SINT32) {
- INT32 *in = (INT32 *)input;
+ Int32 *in = (Int32 *)input;
scale = 1.0 / 2147483648.0;
- for (int i=0; i<stream->bufferSize; i++) {
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT64) in[offset_in[j]];
+ out[offset_out[j]] = (Float64) in[offset_in[j]];
out[offset_out[j]] *= scale;
}
in += jump_in;
}
}
else if (format_in == RTAUDIO_FLOAT32) {
- FLOAT32 *in = (FLOAT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Float32 *in = (Float32 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT64) in[offset_in[j]];
+ out[offset_out[j]] = (Float64) in[offset_in[j]];
}
in += jump_in;
out += jump_out;
}
else if (format_in == RTAUDIO_FLOAT64) {
// Channel compensation and/or (de)interleaving only.
- FLOAT64 *in = (FLOAT64 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Float64 *in = (Float64 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
out[offset_out[j]] = in[offset_in[j]];
}
}
}
else if (format_out == RTAUDIO_FLOAT32) {
- FLOAT32 scale;
- FLOAT32 *out = (FLOAT32 *)output;
+ Float32 scale;
+ Float32 *out = (Float32 *)output;
if (format_in == RTAUDIO_SINT8) {
signed char *in = (signed char *)input;
scale = 1.0 / 128.0;
- for (int i=0; i<stream->bufferSize; i++) {
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT32) in[offset_in[j]];
+ out[offset_out[j]] = (Float32) in[offset_in[j]];
out[offset_out[j]] *= scale;
}
in += jump_in;
}
}
else if (format_in == RTAUDIO_SINT16) {
- INT16 *in = (INT16 *)input;
+ Int16 *in = (Int16 *)input;
scale = 1.0 / 32768.0;
- for (int i=0; i<stream->bufferSize; i++) {
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT32) in[offset_in[j]];
+ out[offset_out[j]] = (Float32) in[offset_in[j]];
out[offset_out[j]] *= scale;
}
in += jump_in;
}
}
else if (format_in == RTAUDIO_SINT24) {
- INT32 *in = (INT32 *)input;
+ Int32 *in = (Int32 *)input;
scale = 1.0 / 2147483648.0;
- for (int i=0; i<stream->bufferSize; i++) {
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT32) (in[offset_in[j]] & 0xffffff00);
+ out[offset_out[j]] = (Float32) (in[offset_in[j]] & 0xffffff00);
out[offset_out[j]] *= scale;
}
in += jump_in;
}
}
else if (format_in == RTAUDIO_SINT32) {
- INT32 *in = (INT32 *)input;
+ Int32 *in = (Int32 *)input;
scale = 1.0 / 2147483648.0;
- for (int i=0; i<stream->bufferSize; i++) {
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT32) in[offset_in[j]];
+ out[offset_out[j]] = (Float32) in[offset_in[j]];
out[offset_out[j]] *= scale;
}
in += jump_in;
}
else if (format_in == RTAUDIO_FLOAT32) {
// Channel compensation and/or (de)interleaving only.
- FLOAT32 *in = (FLOAT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Float32 *in = (Float32 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
out[offset_out[j]] = in[offset_in[j]];
}
}
}
else if (format_in == RTAUDIO_FLOAT64) {
- FLOAT64 *in = (FLOAT64 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Float64 *in = (Float64 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT32) in[offset_in[j]];
+ out[offset_out[j]] = (Float32) in[offset_in[j]];
}
in += jump_in;
out += jump_out;
}
}
else if (format_out == RTAUDIO_SINT32) {
- INT32 *out = (INT32 *)output;
+ Int32 *out = (Int32 *)output;
if (format_in == RTAUDIO_SINT8) {
signed char *in = (signed char *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) in[offset_in[j]];
+ out[offset_out[j]] = (Int32) in[offset_in[j]];
out[offset_out[j]] <<= 24;
}
in += jump_in;
}
}
else if (format_in == RTAUDIO_SINT16) {
- INT16 *in = (INT16 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Int16 *in = (Int16 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) in[offset_in[j]];
+ out[offset_out[j]] = (Int32) in[offset_in[j]];
out[offset_out[j]] <<= 16;
}
in += jump_in;
}
}
else if (format_in == RTAUDIO_SINT24) {
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Int32 *in = (Int32 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) in[offset_in[j]];
+ out[offset_out[j]] = (Int32) in[offset_in[j]];
}
in += jump_in;
out += jump_out;
}
else if (format_in == RTAUDIO_SINT32) {
// Channel compensation and/or (de)interleaving only.
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Int32 *in = (Int32 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
out[offset_out[j]] = in[offset_in[j]];
}
}
}
else if (format_in == RTAUDIO_FLOAT32) {
- FLOAT32 *in = (FLOAT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Float32 *in = (Float32 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) (in[offset_in[j]] * 2147483647.0);
+ out[offset_out[j]] = (Int32) (in[offset_in[j]] * 2147483647.0);
}
in += jump_in;
out += jump_out;
}
}
else if (format_in == RTAUDIO_FLOAT64) {
- FLOAT64 *in = (FLOAT64 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Float64 *in = (Float64 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) (in[offset_in[j]] * 2147483647.0);
+ out[offset_out[j]] = (Int32) (in[offset_in[j]] * 2147483647.0);
}
in += jump_in;
out += jump_out;
}
}
else if (format_out == RTAUDIO_SINT24) {
- INT32 *out = (INT32 *)output;
+ Int32 *out = (Int32 *)output;
if (format_in == RTAUDIO_SINT8) {
signed char *in = (signed char *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) in[offset_in[j]];
+ out[offset_out[j]] = (Int32) in[offset_in[j]];
out[offset_out[j]] <<= 24;
}
in += jump_in;
}
}
else if (format_in == RTAUDIO_SINT16) {
- INT16 *in = (INT16 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Int16 *in = (Int16 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) in[offset_in[j]];
+ out[offset_out[j]] = (Int32) in[offset_in[j]];
out[offset_out[j]] <<= 16;
}
in += jump_in;
}
else if (format_in == RTAUDIO_SINT24) {
// Channel compensation and/or (de)interleaving only.
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Int32 *in = (Int32 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
out[offset_out[j]] = in[offset_in[j]];
}
}
}
else if (format_in == RTAUDIO_SINT32) {
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Int32 *in = (Int32 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) (in[offset_in[j]] & 0xffffff00);
+ out[offset_out[j]] = (Int32) (in[offset_in[j]] & 0xffffff00);
}
in += jump_in;
out += jump_out;
}
}
else if (format_in == RTAUDIO_FLOAT32) {
- FLOAT32 *in = (FLOAT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Float32 *in = (Float32 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) (in[offset_in[j]] * 2147483647.0);
+ out[offset_out[j]] = (Int32) (in[offset_in[j]] * 2147483647.0);
}
in += jump_in;
out += jump_out;
}
}
else if (format_in == RTAUDIO_FLOAT64) {
- FLOAT64 *in = (FLOAT64 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Float64 *in = (Float64 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) (in[offset_in[j]] * 2147483647.0);
+ out[offset_out[j]] = (Int32) (in[offset_in[j]] * 2147483647.0);
}
in += jump_in;
out += jump_out;
}
}
else if (format_out == RTAUDIO_SINT16) {
- INT16 *out = (INT16 *)output;
+ Int16 *out = (Int16 *)output;
if (format_in == RTAUDIO_SINT8) {
signed char *in = (signed char *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT16) in[offset_in[j]];
+ out[offset_out[j]] = (Int16) in[offset_in[j]];
out[offset_out[j]] <<= 8;
}
in += jump_in;
}
else if (format_in == RTAUDIO_SINT16) {
// Channel compensation and/or (de)interleaving only.
- INT16 *in = (INT16 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Int16 *in = (Int16 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
out[offset_out[j]] = in[offset_in[j]];
}
}
}
else if (format_in == RTAUDIO_SINT24) {
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Int32 *in = (Int32 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT16) ((in[offset_in[j]] >> 16) & 0x0000ffff);
+ out[offset_out[j]] = (Int16) ((in[offset_in[j]] >> 16) & 0x0000ffff);
}
in += jump_in;
out += jump_out;
}
}
else if (format_in == RTAUDIO_SINT32) {
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Int32 *in = (Int32 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT16) ((in[offset_in[j]] >> 16) & 0x0000ffff);
+ out[offset_out[j]] = (Int16) ((in[offset_in[j]] >> 16) & 0x0000ffff);
}
in += jump_in;
out += jump_out;
}
}
else if (format_in == RTAUDIO_FLOAT32) {
- FLOAT32 *in = (FLOAT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Float32 *in = (Float32 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT16) (in[offset_in[j]] * 32767.0);
+ out[offset_out[j]] = (Int16) (in[offset_in[j]] * 32767.0);
}
in += jump_in;
out += jump_out;
}
}
else if (format_in == RTAUDIO_FLOAT64) {
- FLOAT64 *in = (FLOAT64 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Float64 *in = (Float64 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT16) (in[offset_in[j]] * 32767.0);
+ out[offset_out[j]] = (Int16) (in[offset_in[j]] * 32767.0);
}
in += jump_in;
out += jump_out;
if (format_in == RTAUDIO_SINT8) {
// Channel compensation and/or (de)interleaving only.
signed char *in = (signed char *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
out[offset_out[j]] = in[offset_in[j]];
}
}
}
if (format_in == RTAUDIO_SINT16) {
- INT16 *in = (INT16 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Int16 *in = (Int16 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
out[offset_out[j]] = (signed char) ((in[offset_in[j]] >> 8) & 0x00ff);
}
}
}
else if (format_in == RTAUDIO_SINT24) {
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Int32 *in = (Int32 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
out[offset_out[j]] = (signed char) ((in[offset_in[j]] >> 24) & 0x000000ff);
}
}
}
else if (format_in == RTAUDIO_SINT32) {
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Int32 *in = (Int32 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
out[offset_out[j]] = (signed char) ((in[offset_in[j]] >> 24) & 0x000000ff);
}
}
}
else if (format_in == RTAUDIO_FLOAT32) {
- FLOAT32 *in = (FLOAT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Float32 *in = (Float32 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
out[offset_out[j]] = (signed char) (in[offset_in[j]] * 127.0);
}
}
}
else if (format_in == RTAUDIO_FLOAT64) {
- FLOAT64 *in = (FLOAT64 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
+ Float64 *in = (Float64 *)input;
+ for (int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<channels; j++) {
out[offset_out[j]] = (signed char) (in[offset_in[j]] * 127.0);
}
}
}
-void RtAudio :: byteSwapBuffer(char *buffer, int samples, RTAUDIO_FORMAT format)
+void RtApi :: byteSwapBuffer( char *buffer, int samples, RtAudioFormat format )
{
register char val;
register char *ptr;
}
}
}
-
-
-// *************************************************** //
-//
-// RtError class definition.
-//
-// *************************************************** //
-
-RtError :: RtError(const char *p, TYPE tipe)
-{
- type = tipe;
- strncpy(error_message, p, 256);
-}
-
-RtError :: ~RtError()
-{
-}
-
-void RtError :: printMessage()
-{
- printf("\n%s\n\n", error_message);
-}
projects / rtaudio-cdist.git / commitdiff
