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godot/drivers/rtaudio/RtAudio.cpp

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#ifdef RTAUDIO_ENABLED
/************************************************************************/
/*! \class RtAudio
\brief Realtime audio i/o C++ classes.
RtAudio provides a common API (Application Programming Interface)
for realtime audio input/output across Linux (native ALSA, Jack,
and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
(DirectSound and ASIO) operating systems.
RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
RtAudio: realtime audio i/o C++ classes
Copyright (c) 2001-2009 Gary P. Scavone
Permission is hereby granted, free of charge, to any person
obtaining a copy of this software and associated documentation files
(the "Software"), to deal in the Software without restriction,
including without limitation the rights to use, copy, modify, merge,
publish, distribute, sublicense, and/or sell copies of the Software,
and to permit persons to whom the Software is furnished to do so,
subject to the following conditions:
The above copyright notice and this permission notice shall be
included in all copies or substantial portions of the Software.
Any person wishing to distribute modifications to the Software is
asked to send the modifications to the original developer so that
they can be incorporated into the canonical version. This is,
however, not a binding provision of this license.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
/************************************************************************/
// RtAudio: Version 4.0.6
#include "RtAudio.h"
#include <iostream>
#include <cstdlib>
#include <cstring>
#include <limits.h>
// Static variable definitions.
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
};
#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
#define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
#define MUTEX_DESTROY(A) DeleteCriticalSection(A)
#define MUTEX_LOCK(A) EnterCriticalSection(A)
#define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
// 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)
#else
#define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
#define MUTEX_DESTROY(A) abs(*A) // dummy definitions
#endif
// *************************************************** //
//
// RtAudio definitions.
//
// *************************************************** //
void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()
{
apis.clear();
// The order here will control the order of RtAudio's API search in
// the constructor.
#if defined(__UNIX_JACK__)
apis.push_back( UNIX_JACK );
#endif
#if defined(__LINUX_ALSA__)
apis.push_back( LINUX_ALSA );
#endif
#if defined(__LINUX_OSS__)
apis.push_back( LINUX_OSS );
#endif
#if defined(__WINDOWS_ASIO__)
apis.push_back( WINDOWS_ASIO );
#endif
#if defined(__WINDOWS_DS__)
apis.push_back( WINDOWS_DS );
#endif
#if defined(__MACOSX_CORE__)
apis.push_back( MACOSX_CORE );
#endif
#if defined(__RTAUDIO_DUMMY__)
apis.push_back( RTAUDIO_DUMMY );
#endif
}
void RtAudio :: openRtApi( RtAudio::Api api )
{
#if defined(__UNIX_JACK__)
if ( api == UNIX_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(__MACOSX_CORE__)
if ( api == MACOSX_CORE )
rtapi_ = new RtApiCore();
#endif
#if defined(__RTAUDIO_DUMMY__)
if ( api == RTAUDIO_DUMMY )
rtapi_ = new RtApiDummy();
#endif
}
RtAudio :: RtAudio( RtAudio::Api api ) throw()
{
rtapi_ = 0;
if ( api != UNSPECIFIED ) {
// Attempt to open the specified API.
openRtApi( api );
if ( rtapi_ ) return;
// No compiled support for specified API value. Issue a debug
// warning and continue as if no API was specified.
std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
}
// Iterate through the compiled APIs and return as soon as we find
// one with at least one device or we reach the end of the list.
std::vector< RtAudio::Api > apis;
getCompiledApi( apis );
for ( unsigned int i=0; i<apis.size(); i++ ) {
openRtApi( apis[i] );
if ( rtapi_->getDeviceCount() ) break;
}
if ( rtapi_ ) return;
// It should not be possible to get here because the preprocessor
// definition __RTAUDIO_DUMMY__ is automatically defined if no
// API-specific definitions are passed to the compiler. But just in
// case something weird happens, we'll print out an error message.
std::cerr << "\nRtAudio: no compiled API support found ... critical error!!\n\n";
}
RtAudio :: ~RtAudio() throw()
{
delete rtapi_;
}
void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
RtAudio::StreamParameters *inputParameters,
RtAudioFormat format, unsigned int sampleRate,
unsigned int *bufferFrames,
RtAudioCallback callback, void *userData,
RtAudio::StreamOptions *options )
{
return rtapi_->openStream( outputParameters, inputParameters, format,
sampleRate, bufferFrames, callback,
userData, options );
}
// *************************************************** //
//
// Public RtApi definitions (see end of file for
// private or protected utility functions).
//
// *************************************************** //
RtApi :: RtApi()
{
stream_.state = STREAM_CLOSED;
stream_.mode = UNINITIALIZED;
stream_.apiHandle = 0;
stream_.userBuffer[0] = 0;
stream_.userBuffer[1] = 0;
MUTEX_INITIALIZE( &stream_.mutex );
showWarnings_ = true;
}
RtApi :: ~RtApi()
{
MUTEX_DESTROY( &stream_.mutex );
}
void RtApi :: openStream( RtAudio::StreamParameters *oParams,
RtAudio::StreamParameters *iParams,
RtAudioFormat format, unsigned int sampleRate,
unsigned int *bufferFrames,
RtAudioCallback callback, void *userData,
RtAudio::StreamOptions *options )
{
if ( stream_.state != STREAM_CLOSED ) {
errorText_ = "RtApi::openStream: a stream is already open!";
error( RtError::INVALID_USE );
}
if ( oParams && oParams->nChannels < 1 ) {
errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
error( RtError::INVALID_USE );
}
if ( iParams && iParams->nChannels < 1 ) {
errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
error( RtError::INVALID_USE );
}
if ( oParams == NULL && iParams == NULL ) {
errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
error( RtError::INVALID_USE );
}
if ( formatBytes(format) == 0 ) {
errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
error( RtError::INVALID_USE );
}
unsigned int nDevices = getDeviceCount();
unsigned int oChannels = 0;
if ( oParams ) {
oChannels = oParams->nChannels;
if ( oParams->deviceId >= nDevices ) {
errorText_ = "RtApi::openStream: output device parameter value is invalid.";
error( RtError::INVALID_USE );
}
}
unsigned int iChannels = 0;
if ( iParams ) {
iChannels = iParams->nChannels;
if ( iParams->deviceId >= nDevices ) {
errorText_ = "RtApi::openStream: input device parameter value is invalid.";
error( RtError::INVALID_USE );
}
}
clearStreamInfo();
bool result;
if ( oChannels > 0 ) {
result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
sampleRate, format, bufferFrames, options );
if ( result == false ) error( RtError::SYSTEM_ERROR );
}
if ( iChannels > 0 ) {
result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
sampleRate, format, bufferFrames, options );
if ( result == false ) {
if ( oChannels > 0 ) closeStream();
error( RtError::SYSTEM_ERROR );
}
}
stream_.callbackInfo.callback = (void *) callback;
stream_.callbackInfo.userData = userData;
if ( options ) options->numberOfBuffers = stream_.nBuffers;
stream_.state = STREAM_STOPPED;
}
unsigned int RtApi :: getDefaultInputDevice( void )
{
// Should be implemented in subclasses if possible.
return 0;
}
unsigned int RtApi :: getDefaultOutputDevice( void )
{
// Should be implemented in subclasses if possible.
return 0;
}
void RtApi :: closeStream( void )
{
// MUST be implemented in subclasses!
return;
}
bool RtApi :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options )
{
// MUST be implemented in subclasses!
return FAILURE;
}
void RtApi :: tickStreamTime( void )
{
// Subclasses that do not provide their own implementation of
// getStreamTime should call this function once per buffer I/O to
// provide basic stream time support.
stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
#if defined( HAVE_GETTIMEOFDAY )
gettimeofday( &stream_.lastTickTimestamp, NULL );
#endif
}
long RtApi :: getStreamLatency( void )
{
verifyStream();
long totalLatency = 0;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
totalLatency = stream_.latency[0];
if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
totalLatency += stream_.latency[1];
return totalLatency;
}
double RtApi :: getStreamTime( void )
{
verifyStream();
#if defined( HAVE_GETTIMEOFDAY )
// Return a very accurate estimate of the stream time by
// adding in the elapsed time since the last tick.
struct timeval then;
struct timeval now;
if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
return stream_.streamTime;
gettimeofday( &now, NULL );
then = stream_.lastTickTimestamp;
return stream_.streamTime +
((now.tv_sec + 0.000001 * now.tv_usec) -
(then.tv_sec + 0.000001 * then.tv_usec));
#else
return stream_.streamTime;
#endif
}
unsigned int RtApi :: getStreamSampleRate( void )
{
verifyStream();
return stream_.sampleRate;
}
// *************************************************** //
//
// OS/API-specific methods.
//
// *************************************************** //
#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.
//
// 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. However, we do provide a flag
// to the client callback function to inform of an over/underrun.
//
// The mechanism for querying and setting system parameters was
// updated (and perhaps simplified) in OS-X version 10.4. However,
// since 10.4 support is not necessarily available to all users, I've
// decided not to update the respective code at this time. Perhaps
// this will happen when Apple makes 10.4 free for everyone. :-)
// A structure to hold various information related to the CoreAudio API
// implementation.
struct CoreHandle {
AudioDeviceID id[2]; // device ids
#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
AudioDeviceIOProcID procId[2];
#endif
UInt32 iStream[2]; // device stream index (or first if using multiple)
UInt32 nStreams[2]; // number of streams to use
bool xrun[2];
char *deviceBuffer;
pthread_cond_t condition;
int drainCounter; // Tracks callback counts when draining
bool internalDrain; // Indicates if stop is initiated from callback or not.
CoreHandle()
:deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
};
RtApiCore :: RtApiCore()
{
// Nothing to do here.
}
RtApiCore :: ~RtApiCore()
{
// The subclass destructor gets called before the base class
// destructor, so close an existing stream before deallocating
// apiDeviceId memory.
if ( stream_.state != STREAM_CLOSED ) closeStream();
}
unsigned int RtApiCore :: getDeviceCount( void )
{
// Find out how many audio devices there are, if any.
UInt32 dataSize;
OSStatus result = AudioHardwareGetPropertyInfo( kAudioHardwarePropertyDevices, &dataSize, NULL );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
error( RtError::WARNING );
return 0;
}
return dataSize / sizeof( AudioDeviceID );
}
unsigned int RtApiCore :: getDefaultInputDevice( void )
{
unsigned int nDevices = getDeviceCount();
if ( nDevices <= 1 ) return 0;
AudioDeviceID id;
UInt32 dataSize = sizeof( AudioDeviceID );
OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultInputDevice,
&dataSize, &id );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
error( RtError::WARNING );
return 0;
}
dataSize *= nDevices;
AudioDeviceID deviceList[ nDevices ];
result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
error( RtError::WARNING );
return 0;
}
for ( unsigned int i=0; i<nDevices; i++ )
if ( id == deviceList[i] ) return i;
errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
error( RtError::WARNING );
return 0;
}
unsigned int RtApiCore :: getDefaultOutputDevice( void )
{
unsigned int nDevices = getDeviceCount();
if ( nDevices <= 1 ) return 0;
AudioDeviceID id;
UInt32 dataSize = sizeof( AudioDeviceID );
OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultOutputDevice,
&dataSize, &id );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
error( RtError::WARNING );
return 0;
}
dataSize *= nDevices;
AudioDeviceID deviceList[ nDevices ];
result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
error( RtError::WARNING );
return 0;
}
for ( unsigned int i=0; i<nDevices; i++ )
if ( id == deviceList[i] ) return i;
errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
error( RtError::WARNING );
return 0;
}
RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
{
RtAudio::DeviceInfo info;
info.probed = false;
// Get device ID
unsigned int nDevices = getDeviceCount();
if ( nDevices == 0 ) {
errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
error( RtError::INVALID_USE );
}
if ( device >= nDevices ) {
errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
error( RtError::INVALID_USE );
}
AudioDeviceID deviceList[ nDevices ];
UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
if ( result != noErr ) {
errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
error( RtError::WARNING );
return info;
}
AudioDeviceID id = deviceList[ device ];
// Get the device name.
info.name.erase();
char name[256];
dataSize = 256;
result = AudioDeviceGetProperty( id, 0, false,
kAudioDevicePropertyDeviceManufacturer,
&dataSize, name );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
info.name.append( (const char *)name, strlen(name) );
info.name.append( ": " );
dataSize = 256;
result = AudioDeviceGetProperty( id, 0, false,
kAudioDevicePropertyDeviceName,
&dataSize, name );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
info.name.append( (const char *)name, strlen(name) );
// Get the output stream "configuration".
AudioBufferList *bufferList = nil;
result = AudioDeviceGetPropertyInfo( id, 0, false,
kAudioDevicePropertyStreamConfiguration,
&dataSize, NULL );
if (result != noErr || dataSize == 0) {
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
// Allocate the AudioBufferList.
bufferList = (AudioBufferList *) malloc( dataSize );
if ( bufferList == NULL ) {
errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
error( RtError::WARNING );
return info;
}
result = AudioDeviceGetProperty( id, 0, false,
kAudioDevicePropertyStreamConfiguration,
&dataSize, bufferList );
if ( result != noErr ) {
free( bufferList );
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
// Get output channel information.
unsigned int i, nStreams = bufferList->mNumberBuffers;
for ( i=0; i<nStreams; i++ )
info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
free( bufferList );
// Get the input stream "configuration".
result = AudioDeviceGetPropertyInfo( id, 0, true,
kAudioDevicePropertyStreamConfiguration,
&dataSize, NULL );
if (result != noErr || dataSize == 0) {
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
// Allocate the AudioBufferList.
bufferList = (AudioBufferList *) malloc( dataSize );
if ( bufferList == NULL ) {
errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
error( RtError::WARNING );
return info;
}
result = AudioDeviceGetProperty( id, 0, true,
kAudioDevicePropertyStreamConfiguration,
&dataSize, bufferList );
if ( result != noErr ) {
free( bufferList );
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
// Get input channel information.
nStreams = bufferList->mNumberBuffers;
for ( i=0; i<nStreams; i++ )
info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
free( bufferList );
// If device opens for both playback and capture, we determine the channels.
if ( info.outputChannels > 0 && info.inputChannels > 0 )
info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
// Probe the device sample rates.
bool isInput = false;
if ( info.outputChannels == 0 ) isInput = true;
// Determine the supported sample rates.
result = AudioDeviceGetPropertyInfo( id, 0, isInput,
kAudioDevicePropertyAvailableNominalSampleRates,
&dataSize, NULL );
if ( result != kAudioHardwareNoError || dataSize == 0 ) {
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
UInt32 nRanges = dataSize / sizeof( AudioValueRange );
AudioValueRange rangeList[ nRanges ];
result = AudioDeviceGetProperty( id, 0, isInput,
kAudioDevicePropertyAvailableNominalSampleRates,
&dataSize, &rangeList );
if ( result != kAudioHardwareNoError ) {
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
Float64 minimumRate = 100000000.0, maximumRate = 0.0;
for ( UInt32 i=0; i<nRanges; i++ ) {
if ( rangeList[i].mMinimum < minimumRate ) minimumRate = rangeList[i].mMinimum;
if ( rangeList[i].mMaximum > maximumRate ) maximumRate = rangeList[i].mMaximum;
}
info.sampleRates.clear();
for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate )
info.sampleRates.push_back( SAMPLE_RATES[k] );
}
if ( info.sampleRates.size() == 0 ) {
errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
// CoreAudio always uses 32-bit floating point data for PCM streams.
// Thus, any other "physical" formats supported by the device are of
// no interest to the client.
info.nativeFormats = RTAUDIO_FLOAT32;
if ( getDefaultOutputDevice() == device )
info.isDefaultOutput = true;
if ( getDefaultInputDevice() == device )
info.isDefaultInput = true;
info.probed = true;
return info;
}
OSStatus callbackHandler( AudioDeviceID inDevice,
const AudioTimeStamp* inNow,
const AudioBufferList* inInputData,
const AudioTimeStamp* inInputTime,
AudioBufferList* outOutputData,
const AudioTimeStamp* inOutputTime,
void* infoPointer )
{
CallbackInfo *info = (CallbackInfo *) infoPointer;
RtApiCore *object = (RtApiCore *) info->object;
if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
return kAudioHardwareUnspecifiedError;
else
return kAudioHardwareNoError;
}
OSStatus deviceListener( AudioDeviceID inDevice,
UInt32 channel,
Boolean isInput,
AudioDevicePropertyID propertyID,
void* handlePointer )
{
CoreHandle *handle = (CoreHandle *) handlePointer;
if ( propertyID == kAudioDeviceProcessorOverload ) {
if ( isInput )
handle->xrun[1] = true;
else
handle->xrun[0] = true;
}
return kAudioHardwareNoError;
}
static bool hasProperty( AudioDeviceID id, UInt32 channel, bool isInput, AudioDevicePropertyID property )
{
OSStatus result = AudioDeviceGetPropertyInfo( id, channel, isInput, property, NULL, NULL );
return result == 0;
}
bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options )
{
// Get device ID
unsigned int nDevices = getDeviceCount();
if ( nDevices == 0 ) {
// This should not happen because a check is made before this function is called.
errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
return FAILURE;
}
if ( device >= nDevices ) {
// This should not happen because a check is made before this function is called.
errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
return FAILURE;
}
AudioDeviceID deviceList[ nDevices ];
UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
if ( result != noErr ) {
errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
return FAILURE;
}
AudioDeviceID id = deviceList[ device ];
// Setup for stream mode.
bool isInput = false;
if ( mode == INPUT ) isInput = true;
// Set or disable "hog" mode.
dataSize = sizeof( UInt32 );
UInt32 doHog = 0;
if ( options && options->flags & RTAUDIO_HOG_DEVICE ) doHog = 1;
result = AudioHardwareSetProperty( kAudioHardwarePropertyHogModeIsAllowed, dataSize, &doHog );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
errorText_ = errorStream_.str();
return FAILURE;
}
// Get the stream "configuration".
AudioBufferList *bufferList;
result = AudioDeviceGetPropertyInfo( id, 0, isInput,
kAudioDevicePropertyStreamConfiguration,
&dataSize, NULL );
if (result != noErr || dataSize == 0) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
// Allocate the AudioBufferList.
bufferList = (AudioBufferList *) malloc( dataSize );
if ( bufferList == NULL ) {
errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
return FAILURE;
}
result = AudioDeviceGetProperty( id, 0, isInput,
kAudioDevicePropertyStreamConfiguration,
&dataSize, bufferList );
if ( result != noErr ) {
free( bufferList );
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
// Search for one or more streams that contain the desired number of
// channels. CoreAudio devices can have an arbitrary number of
// streams and each stream can have an arbitrary number of channels.
// For each stream, a single buffer of interleaved samples is
// provided. RtAudio prefers the use of one stream of interleaved
// data or multiple consecutive single-channel streams. However, we
// now support multiple consecutive multi-channel streams of
// interleaved data as well.
UInt32 iStream, offsetCounter = firstChannel;
UInt32 nStreams = bufferList->mNumberBuffers;
bool monoMode = false;
bool foundStream = false;
// First check that the device supports the requested number of
// channels.
UInt32 deviceChannels = 0;
for ( iStream=0; iStream<nStreams; iStream++ )
deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
if ( deviceChannels < ( channels + firstChannel ) ) {
free( bufferList );
errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
errorText_ = errorStream_.str();
return FAILURE;
}
// Look for a single stream meeting our needs.
UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
for ( iStream=0; iStream<nStreams; iStream++ ) {
streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
if ( streamChannels >= channels + offsetCounter ) {
firstStream = iStream;
channelOffset = offsetCounter;
foundStream = true;
break;
}
if ( streamChannels > offsetCounter ) break;
offsetCounter -= streamChannels;
}
// If we didn't find a single stream above, then we should be able
// to meet the channel specification with multiple streams.
if ( foundStream == false ) {
monoMode = true;
offsetCounter = firstChannel;
for ( iStream=0; iStream<nStreams; iStream++ ) {
streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
if ( streamChannels > offsetCounter ) break;
offsetCounter -= streamChannels;
}
firstStream = iStream;
channelOffset = offsetCounter;
Int32 channelCounter = channels + offsetCounter - streamChannels;
if ( streamChannels > 1 ) monoMode = false;
while ( channelCounter > 0 ) {
streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
if ( streamChannels > 1 ) monoMode = false;
channelCounter -= streamChannels;
streamCount++;
}
}
free( bufferList );
// Determine the buffer size.
AudioValueRange bufferRange;
dataSize = sizeof( AudioValueRange );
result = AudioDeviceGetProperty( id, 0, isInput,
kAudioDevicePropertyBufferFrameSizeRange,
&dataSize, &bufferRange );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
// Set the buffer size. For multiple streams, I'm assuming we only
// need to make this setting for the master channel.
UInt32 theSize = (UInt32) *bufferSize;
dataSize = sizeof( UInt32 );
result = AudioDeviceSetProperty( id, NULL, 0, isInput,
kAudioDevicePropertyBufferFrameSize,
dataSize, &theSize );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
// If attempting to setup a duplex stream, the bufferSize parameter
// MUST be the same in both directions!
*bufferSize = theSize;
if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
stream_.bufferSize = *bufferSize;
stream_.nBuffers = 1;
// Get the stream ID(s) so we can set the stream format. We'll have
// to do this for each stream.
AudioStreamID streamIDs[ nStreams ];
dataSize = nStreams * sizeof( AudioStreamID );
result = AudioDeviceGetProperty( id, 0, isInput,
kAudioDevicePropertyStreams,
&dataSize, &streamIDs );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream ID(s) for device (" << device << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
// Now set the stream format. Also, check the physical format of the
// device and change that if necessary.
AudioStreamBasicDescription description;
dataSize = sizeof( AudioStreamBasicDescription );
bool updateFormat;
for ( UInt32 i=0; i<streamCount; i++ ) {
result = AudioStreamGetProperty( streamIDs[firstStream+i], 0,
kAudioStreamPropertyVirtualFormat,
&dataSize, &description );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
// Set the sample rate and data format id. However, only make the
// change if the sample rate is not within 1.0 of the desired
// rate and the format is not linear pcm.
updateFormat = false;
if ( fabs( description.mSampleRate - (double)sampleRate ) > 1.0 ) {
description.mSampleRate = (double) sampleRate;
updateFormat = true;
}
if ( description.mFormatID != kAudioFormatLinearPCM ) {
description.mFormatID = kAudioFormatLinearPCM;
updateFormat = true;
}
if ( updateFormat ) {
result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0,
kAudioStreamPropertyVirtualFormat,
dataSize, &description );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
}
// Now check the physical format.
result = AudioStreamGetProperty( streamIDs[firstStream+i], 0,
kAudioStreamPropertyPhysicalFormat,
&dataSize, &description );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 24 ) {
description.mFormatID = kAudioFormatLinearPCM;
AudioStreamBasicDescription testDescription = description;
unsigned long formatFlags;
// We'll try higher bit rates first and then work our way down.
testDescription.mBitsPerChannel = 32;
formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger;
testDescription.mFormatFlags = formatFlags;
result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
if ( result == noErr ) continue;
testDescription = description;
testDescription.mBitsPerChannel = 32;
formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger) & ~kLinearPCMFormatFlagIsFloat;
testDescription.mFormatFlags = formatFlags;
result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
if ( result == noErr ) continue;
testDescription = description;
testDescription.mBitsPerChannel = 24;
testDescription.mFormatFlags = formatFlags;
result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
if ( result == noErr ) continue;
testDescription = description;
testDescription.mBitsPerChannel = 16;
testDescription.mFormatFlags = formatFlags;
result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
if ( result == noErr ) continue;
testDescription = description;
testDescription.mBitsPerChannel = 8;
testDescription.mFormatFlags = formatFlags;
result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
}
}
// Get the stream latency. There can be latency in both the device
// and the stream. First, attempt to get the device latency on the
// master channel or the first open channel. Errors that might
// occur here are not deemed critical.
// ***** CHECK THIS ***** //
UInt32 latency, channel = 0;
dataSize = sizeof( UInt32 );
AudioDevicePropertyID property = kAudioDevicePropertyLatency;
if ( hasProperty( id, channel, isInput, property ) == true ) {
result = AudioDeviceGetProperty( id, channel, isInput, property, &dataSize, &latency );
if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
else {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
}
}
// Now try to get the stream latency. For multiple streams, I assume the
// latency is equal for each.
result = AudioStreamGetProperty( streamIDs[firstStream], 0, property, &dataSize, &latency );
if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] += latency;
else {
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream latency for device (" << device << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
}
// Byte-swapping: According to AudioHardware.h, the stream data will
// always be presented in native-endian format, so we should never
// need to byte swap.
stream_.doByteSwap[mode] = false;
// From the CoreAudio documentation, PCM data must be supplied as
// 32-bit floats.
stream_.userFormat = format;
stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
if ( streamCount == 1 )
stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
else // multiple streams
stream_.nDeviceChannels[mode] = channels;
stream_.nUserChannels[mode] = channels;
stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream
if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
else stream_.userInterleaved = true;
stream_.deviceInterleaved[mode] = true;
if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
// 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 ( streamCount == 1 ) {
if ( stream_.nUserChannels[mode] > 1 &&
stream_.userInterleaved != stream_.deviceInterleaved[mode] )
stream_.doConvertBuffer[mode] = true;
}
else if ( monoMode && stream_.userInterleaved )
stream_.doConvertBuffer[mode] = true;
// Allocate our CoreHandle structure for the stream.
CoreHandle *handle = 0;
if ( stream_.apiHandle == 0 ) {
try {
handle = new CoreHandle;
}
catch ( std::bad_alloc& ) {
errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
goto error;
}
if ( pthread_cond_init( &handle->condition, NULL ) ) {
errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
goto error;
}
stream_.apiHandle = (void *) handle;
}
else
handle = (CoreHandle *) stream_.apiHandle;
handle->iStream[mode] = firstStream;
handle->nStreams[mode] = streamCount;
handle->id[mode] = id;
// Allocate necessary internal buffers.
unsigned long bufferBytes;
bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
if ( stream_.userBuffer[mode] == NULL ) {
errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
goto error;
}
// If possible, we will make use of the CoreAudio stream buffers as
// "device buffers". However, we can't do this if using multiple
// streams.
if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
bool makeBuffer = true;
bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
if ( mode == INPUT ) {
if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
if ( bufferBytes <= bytesOut ) makeBuffer = false;
}
}
if ( makeBuffer ) {
bufferBytes *= *bufferSize;
if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
if ( stream_.deviceBuffer == NULL ) {
errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
goto error;
}
}
}
stream_.sampleRate = sampleRate;
stream_.device[mode] = device;
stream_.state = STREAM_STOPPED;
stream_.callbackInfo.object = (void *) this;
// Setup the buffer conversion information structure.
if ( stream_.doConvertBuffer[mode] ) {
if ( streamCount > 1 ) setConvertInfo( mode, 0 );
else setConvertInfo( mode, channelOffset );
}
if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
// Only one callback procedure per device.
stream_.mode = DUPLEX;
else {
#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
#else
// deprecated in favor of AudioDeviceCreateIOProcID()
result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
#endif
if ( result != noErr ) {
errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
errorText_ = errorStream_.str();
goto error;
}
if ( stream_.mode == OUTPUT && mode == INPUT )
stream_.mode = DUPLEX;
else
stream_.mode = mode;
}
// Setup the device property listener for over/underload.
result = AudioDeviceAddPropertyListener( id, 0, isInput,
kAudioDeviceProcessorOverload,
deviceListener, (void *) handle );
return SUCCESS;
error:
if ( handle ) {
pthread_cond_destroy( &handle->condition );
delete handle;
stream_.apiHandle = 0;
}
for ( int i=0; i<2; i++ ) {
if ( stream_.userBuffer[i] ) {
free( stream_.userBuffer[i] );
stream_.userBuffer[i] = 0;
}
}
if ( stream_.deviceBuffer ) {
free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
return FAILURE;
}
void RtApiCore :: closeStream( void )
{
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiCore::closeStream(): no open stream to close!";
error( RtError::WARNING );
return;
}
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
if ( stream_.state == STREAM_RUNNING )
AudioDeviceStop( handle->id[0], callbackHandler );
#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
#else
// deprecated in favor of AudioDeviceDestroyIOProcID()
AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
#endif
}
if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
if ( stream_.state == STREAM_RUNNING )
AudioDeviceStop( handle->id[1], callbackHandler );
#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
#else
// deprecated in favor of AudioDeviceDestroyIOProcID()
AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
#endif
}
for ( int i=0; i<2; i++ ) {
if ( stream_.userBuffer[i] ) {
free( stream_.userBuffer[i] );
stream_.userBuffer[i] = 0;
}
}
if ( stream_.deviceBuffer ) {
free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
// Destroy pthread condition variable.
pthread_cond_destroy( &handle->condition );
delete handle;
stream_.apiHandle = 0;
stream_.mode = UNINITIALIZED;
stream_.state = STREAM_CLOSED;
}
void RtApiCore :: startStream( void )
{
verifyStream();
if ( stream_.state == STREAM_RUNNING ) {
errorText_ = "RtApiCore::startStream(): the stream is already running!";
error( RtError::WARNING );
return;
}
MUTEX_LOCK( &stream_.mutex );
OSStatus result = noErr;
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
result = AudioDeviceStart( handle->id[0], callbackHandler );
if ( result != noErr ) {
errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
errorText_ = errorStream_.str();
goto unlock;
}
}
if ( stream_.mode == INPUT ||
( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
result = AudioDeviceStart( handle->id[1], callbackHandler );
if ( result != noErr ) {
errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
errorText_ = errorStream_.str();
goto unlock;
}
}
handle->drainCounter = 0;
handle->internalDrain = false;
stream_.state = STREAM_RUNNING;
unlock:
MUTEX_UNLOCK( &stream_.mutex );
if ( result == noErr ) return;
error( RtError::SYSTEM_ERROR );
}
void RtApiCore :: stopStream( void )
{
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
error( RtError::WARNING );
return;
}
MUTEX_LOCK( &stream_.mutex );
if ( stream_.state == STREAM_STOPPED ) {
MUTEX_UNLOCK( &stream_.mutex );
return;
}
OSStatus result = noErr;
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
if ( handle->drainCounter == 0 ) {
handle->drainCounter = 1;
pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
}
result = AudioDeviceStop( handle->id[0], callbackHandler );
if ( result != noErr ) {
errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
errorText_ = errorStream_.str();
goto unlock;
}
}
if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
result = AudioDeviceStop( handle->id[1], callbackHandler );
if ( result != noErr ) {
errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
errorText_ = errorStream_.str();
goto unlock;
}
}
stream_.state = STREAM_STOPPED;
unlock:
MUTEX_UNLOCK( &stream_.mutex );
if ( result == noErr ) return;
error( RtError::SYSTEM_ERROR );
}
void RtApiCore :: abortStream( void )
{
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
error( RtError::WARNING );
return;
}
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
handle->drainCounter = 1;
stopStream();
}
bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
const AudioBufferList *inBufferList,
const AudioBufferList *outBufferList )
{
if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
error( RtError::WARNING );
return FAILURE;
}
CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
// Check if we were draining the stream and signal is finished.
if ( handle->drainCounter > 3 ) {
if ( handle->internalDrain == false )
pthread_cond_signal( &handle->condition );
else
stopStream();
return SUCCESS;
}
MUTEX_LOCK( &stream_.mutex );
// The state might change while waiting on a mutex.
if ( stream_.state == STREAM_STOPPED ) {
MUTEX_UNLOCK( &stream_.mutex );
return SUCCESS;
}
AudioDeviceID outputDevice = handle->id[0];
// Invoke user callback to get fresh output data UNLESS we are
// draining stream or duplex mode AND the input/output devices are
// different AND this function is called for the input device.
if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
RtAudioCallback callback = (RtAudioCallback) info->callback;
double streamTime = getStreamTime();
RtAudioStreamStatus status = 0;
if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
status |= RTAUDIO_OUTPUT_UNDERFLOW;
handle->xrun[0] = false;
}
if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
status |= RTAUDIO_INPUT_OVERFLOW;
handle->xrun[1] = false;
}
handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
stream_.bufferSize, streamTime, status, info->userData );
if ( handle->drainCounter == 2 ) {
MUTEX_UNLOCK( &stream_.mutex );
abortStream();
return SUCCESS;
}
else if ( handle->drainCounter == 1 )
handle->internalDrain = true;
}
if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
if ( handle->drainCounter > 1 ) { // write zeros to the output stream
if ( handle->nStreams[0] == 1 ) {
memset( outBufferList->mBuffers[handle->iStream[0]].mData,
0,
outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
}
else { // fill multiple streams with zeros
for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
0,
outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
}
}
}
else if ( handle->nStreams[0] == 1 ) {
if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
stream_.userBuffer[0], stream_.convertInfo[0] );
}
else { // copy from user buffer
memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
stream_.userBuffer[0],
outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
}
}
else { // fill multiple streams
Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
if ( stream_.doConvertBuffer[0] ) {
convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
inBuffer = (Float32 *) stream_.deviceBuffer;
}
if ( stream_.deviceInterleaved[0] == false ) { // mono mode
UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
(void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
}
}
else { // fill multiple multi-channel streams with interleaved data
UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
Float32 *out, *in;
bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
UInt32 inChannels = stream_.nUserChannels[0];
if ( stream_.doConvertBuffer[0] ) {
inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
inChannels = stream_.nDeviceChannels[0];
}
if ( inInterleaved ) inOffset = 1;
else inOffset = stream_.bufferSize;
channelsLeft = inChannels;
for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
in = inBuffer;
out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
outJump = 0;
// Account for possible channel offset in first stream
if ( i == 0 && stream_.channelOffset[0] > 0 ) {
streamChannels -= stream_.channelOffset[0];
outJump = stream_.channelOffset[0];
out += outJump;
}
// Account for possible unfilled channels at end of the last stream
if ( streamChannels > channelsLeft ) {
outJump = streamChannels - channelsLeft;
streamChannels = channelsLeft;
}
// Determine input buffer offsets and skips
if ( inInterleaved ) {
inJump = inChannels;
in += inChannels - channelsLeft;
}
else {
inJump = 1;
in += (inChannels - channelsLeft) * inOffset;
}
for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
for ( unsigned int j=0; j<streamChannels; j++ ) {
*out++ = in[j*inOffset];
}
out += outJump;
in += inJump;
}
channelsLeft -= streamChannels;
}
}
}
if ( handle->drainCounter ) {
handle->drainCounter++;
goto unlock;
}
}
AudioDeviceID inputDevice;
inputDevice = handle->id[1];
if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
if ( handle->nStreams[1] == 1 ) {
if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
convertBuffer( stream_.userBuffer[1],
(char *) inBufferList->mBuffers[handle->iStream[1]].mData,
stream_.convertInfo[1] );
}
else { // copy to user buffer
memcpy( stream_.userBuffer[1],
inBufferList->mBuffers[handle->iStream[1]].mData,
inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
}
}
else { // read from multiple streams
Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
if ( stream_.deviceInterleaved[1] == false ) { // mono mode
UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
memcpy( (void *)&outBuffer[i*stream_.bufferSize],
inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
}
}
else { // read from multiple multi-channel streams
UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
Float32 *out, *in;
bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
UInt32 outChannels = stream_.nUserChannels[1];
if ( stream_.doConvertBuffer[1] ) {
outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
outChannels = stream_.nDeviceChannels[1];
}
if ( outInterleaved ) outOffset = 1;
else outOffset = stream_.bufferSize;
channelsLeft = outChannels;
for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
out = outBuffer;
in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
inJump = 0;
// Account for possible channel offset in first stream
if ( i == 0 && stream_.channelOffset[1] > 0 ) {
streamChannels -= stream_.channelOffset[1];
inJump = stream_.channelOffset[1];
in += inJump;
}
// Account for possible unread channels at end of the last stream
if ( streamChannels > channelsLeft ) {
inJump = streamChannels - channelsLeft;
streamChannels = channelsLeft;
}
// Determine output buffer offsets and skips
if ( outInterleaved ) {
outJump = outChannels;
out += outChannels - channelsLeft;
}
else {
outJump = 1;
out += (outChannels - channelsLeft) * outOffset;
}
for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
for ( unsigned int j=0; j<streamChannels; j++ ) {
out[j*outOffset] = *in++;
}
out += outJump;
in += inJump;
}
channelsLeft -= streamChannels;
}
}
if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
convertBuffer( stream_.userBuffer[1],
stream_.deviceBuffer,
stream_.convertInfo[1] );
}
}
}
unlock:
MUTEX_UNLOCK( &stream_.mutex );
RtApi::tickStreamTime();
return SUCCESS;
}
const char* RtApiCore :: getErrorCode( OSStatus code )
{
switch( code ) {
case kAudioHardwareNotRunningError:
return "kAudioHardwareNotRunningError";
case kAudioHardwareUnspecifiedError:
return "kAudioHardwareUnspecifiedError";
case kAudioHardwareUnknownPropertyError:
return "kAudioHardwareUnknownPropertyError";
case kAudioHardwareBadPropertySizeError:
return "kAudioHardwareBadPropertySizeError";
case kAudioHardwareIllegalOperationError:
return "kAudioHardwareIllegalOperationError";
case kAudioHardwareBadObjectError:
return "kAudioHardwareBadObjectError";
case kAudioHardwareBadDeviceError:
return "kAudioHardwareBadDeviceError";
case kAudioHardwareBadStreamError:
return "kAudioHardwareBadStreamError";
case kAudioHardwareUnsupportedOperationError:
return "kAudioHardwareUnsupportedOperationError";
case kAudioDeviceUnsupportedFormatError:
return "kAudioDeviceUnsupportedFormatError";
case kAudioDevicePermissionsError:
return "kAudioDevicePermissionsError";
default:
return "CoreAudio unknown error";
}
}
//******************** End of __MACOSX_CORE__ *********************//
#endif
#if defined(__UNIX_JACK__)
// JACK is a low-latency audio server, originally written for the
// GNU/Linux operating system and now also ported to OS-X. It can
// connect a number of different applications to an audio device, as
// well as allowing them to share audio between themselves.
//
// When using JACK with RtAudio, "devices" refer to JACK clients that
// have ports connected to the server. The JACK server is typically
// started in a terminal as follows:
//
// .jackd -d alsa -d hw:0
//
// or through an interface program such as qjackctl. 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.
//
// The JACK server does not have to be running when an instance of
// RtApiJack is created, though the function getDeviceCount() will
// report 0 devices found until JACK has been started. When no
// devices are available (i.e., the JACK server is not running), a
// stream cannot be opened.
#include <jack/jack.h>
#include <unistd.h>
// A structure to hold various information related to the Jack API
// implementation.
struct JackHandle {
jack_client_t *client;
jack_port_t **ports[2];
std::string deviceName[2];
bool xrun[2];
pthread_cond_t condition;
int drainCounter; // Tracks callback counts when draining
bool internalDrain; // Indicates if stop is initiated from callback or not.
JackHandle()
:client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
};
void jackSilentError( const char * ) {};
RtApiJack :: RtApiJack()
{
// Nothing to do here.
#if !defined(__RTAUDIO_DEBUG__)
// Turn off Jack's internal error reporting.
jack_set_error_function( &jackSilentError );
#endif
}
RtApiJack :: ~RtApiJack()
{
if ( stream_.state != STREAM_CLOSED ) closeStream();
}
unsigned int RtApiJack :: getDeviceCount( void )
{
// See if we can become a jack client.
jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption;
jack_status_t *status = NULL;
jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
if ( client == 0 ) return 0;
const char **ports;
std::string port, previousPort;
unsigned int nChannels = 0, nDevices = 0;
ports = jack_get_ports( client, NULL, NULL, 0 );
if ( ports ) {
// Parse the port names up to the first colon (:).
size_t iColon = 0;
do {
port = (char *) ports[ nChannels ];
iColon = port.find(":");
if ( iColon != std::string::npos ) {
port = port.substr( 0, iColon + 1 );
if ( port != previousPort ) {
nDevices++;
previousPort = port;
}
}
} while ( ports[++nChannels] );
free( ports );
}
jack_client_close( client );
return nDevices;
}
RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
{
RtAudio::DeviceInfo info;
info.probed = false;
jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption
jack_status_t *status = NULL;
jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
if ( client == 0 ) {
errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
error( RtError::WARNING );
return info;
}
const char **ports;
std::string port, previousPort;
unsigned int nPorts = 0, nDevices = 0;
ports = jack_get_ports( client, NULL, NULL, 0 );
if ( ports ) {
// Parse the port names up to the first colon (:).
size_t iColon = 0;
do {
port = (char *) ports[ nPorts ];
iColon = port.find(":");
if ( iColon != std::string::npos ) {
port = port.substr( 0, iColon );
if ( port != previousPort ) {
if ( nDevices == device ) info.name = port;
nDevices++;
previousPort = port;
}
}
} while ( ports[++nPorts] );
free( ports );
}
if ( device >= nDevices ) {
errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
error( RtError::INVALID_USE );
}
// Get the current jack server sample rate.
info.sampleRates.clear();
info.sampleRates.push_back( jack_get_sample_rate( client ) );
// Count the available ports containing the client name as device
// channels. Jack "input ports" equal RtAudio output channels.
unsigned int nChannels = 0;
ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
if ( ports ) {
while ( ports[ nChannels ] ) nChannels++;
free( ports );
info.outputChannels = nChannels;
}
// Jack "output ports" equal RtAudio input channels.
nChannels = 0;
ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
if ( ports ) {
while ( ports[ nChannels ] ) nChannels++;
free( ports );
info.inputChannels = nChannels;
}
if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
jack_client_close(client);
errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
error( RtError::WARNING );
return info;
}
// If device opens for both playback and capture, we determine the channels.
if ( info.outputChannels > 0 && info.inputChannels > 0 )
info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
// Jack always uses 32-bit floats.
info.nativeFormats = RTAUDIO_FLOAT32;
// Jack doesn't provide default devices so we'll use the first available one.
if ( device == 0 && info.outputChannels > 0 )
info.isDefaultOutput = true;
if ( device == 0 && info.inputChannels > 0 )
info.isDefaultInput = true;
jack_client_close(client);
info.probed = true;
return info;
}
int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
{
CallbackInfo *info = (CallbackInfo *) infoPointer;
RtApiJack *object = (RtApiJack *) info->object;
if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
return 0;
}
void jackShutdown( void *infoPointer )
{
CallbackInfo *info = (CallbackInfo *) infoPointer;
RtApiJack *object = (RtApiJack *) info->object;
// Check current stream state. If stopped, then we'll assume this
// was called as a result of a call to RtApiJack::stopStream (the
// deactivation of a client handle causes this function to be called).
// If not, we'll assume the Jack server is shutting down or some
// other problem occurred and we should close the stream.
if ( object->isStreamRunning() == false ) return;
object->closeStream();
std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
}
int jackXrun( void *infoPointer )
{
JackHandle *handle = (JackHandle *) infoPointer;
if ( handle->ports[0] ) handle->xrun[0] = true;
if ( handle->ports[1] ) handle->xrun[1] = true;
return 0;
}
bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options )
{
JackHandle *handle = (JackHandle *) stream_.apiHandle;
// Look for jack server and try to become a client (only do once per stream).
jack_client_t *client = 0;
if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption;
jack_status_t *status = NULL;
if ( options && !options->streamName.empty() )
client = jack_client_open( options->streamName.c_str(), jackoptions, status );
else
client = jack_client_open( "RtApiJack", jackoptions, status );
if ( client == 0 ) {
errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
error( RtError::WARNING );
return FAILURE;
}
}
else {
// The handle must have been created on an earlier pass.
client = handle->client;
}
const char **ports;
std::string port, previousPort, deviceName;
unsigned int nPorts = 0, nDevices = 0;
ports = jack_get_ports( client, NULL, NULL, 0 );
if ( ports ) {
// Parse the port names up to the first colon (:).
size_t iColon = 0;
do {
port = (char *) ports[ nPorts ];
iColon = port.find(":");
if ( iColon != std::string::npos ) {
port = port.substr( 0, iColon );
if ( port != previousPort ) {
if ( nDevices == device ) deviceName = port;
nDevices++;
previousPort = port;
}
}
} while ( ports[++nPorts] );
free( ports );
}
if ( device >= nDevices ) {
errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
return FAILURE;
}
// Count the available ports containing the client name as device
// channels. Jack "input ports" equal RtAudio output channels.
unsigned int nChannels = 0;
unsigned long flag = JackPortIsInput;
if ( mode == INPUT ) flag = JackPortIsOutput;
ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
if ( ports ) {
while ( ports[ nChannels ] ) nChannels++;
free( ports );
}
// Compare the jack ports for specified client to the requested number of channels.
if ( nChannels < (channels + firstChannel) ) {
errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
// Check the jack server sample rate.
unsigned int jackRate = jack_get_sample_rate( client );
if ( sampleRate != jackRate ) {
jack_client_close( client );
errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
stream_.sampleRate = jackRate;
// Get the latency of the JACK port.
ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
if ( ports[ firstChannel ] )
stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
free( ports );
// The jack server always uses 32-bit floating-point data.
stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
stream_.userFormat = format;
if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
else stream_.userInterleaved = true;
// Jack always uses non-interleaved buffers.
stream_.deviceInterleaved[mode] = false;
// 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;
// Set flags for buffer conversion.
stream_.doConvertBuffer[mode] = false;
if ( stream_.userFormat != stream_.deviceFormat[mode] )
stream_.doConvertBuffer[mode] = true;
if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
stream_.nUserChannels[mode] > 1 )
stream_.doConvertBuffer[mode] = true;
// Allocate our JackHandle structure for the stream.
if ( handle == 0 ) {
try {
handle = new JackHandle;
}
catch ( std::bad_alloc& ) {
errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
goto error;
}
if ( pthread_cond_init(&handle->condition, NULL) ) {
errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
goto error;
}
stream_.apiHandle = (void *) handle;
handle->client = client;
}
handle->deviceName[mode] = deviceName;
// Allocate necessary internal buffers.
unsigned long bufferBytes;
bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
if ( stream_.userBuffer[mode] == NULL ) {
errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
goto error;
}
if ( stream_.doConvertBuffer[mode] ) {
bool makeBuffer = true;
if ( mode == OUTPUT )
bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
else { // mode == INPUT
bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
if ( bufferBytes < bytesOut ) makeBuffer = false;
}
}
if ( makeBuffer ) {
bufferBytes *= *bufferSize;
if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
if ( stream_.deviceBuffer == NULL ) {
errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
goto error;
}
}
}
// 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 ) {
errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
goto error;
}
stream_.device[mode] = device;
stream_.channelOffset[mode] = firstChannel;
stream_.state = STREAM_STOPPED;
stream_.callbackInfo.object = (void *) this;
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, (void *) &handle );
jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
}
// Register our ports.
char label[64];
if ( mode == OUTPUT ) {
for ( unsigned 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 );
}
}
else {
for ( unsigned 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 );
}
}
// Setup the buffer conversion information structure. We don't use
// buffers to do channel offsets, so we override that parameter
// here.
if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
return SUCCESS;
error:
if ( handle ) {
pthread_cond_destroy( &handle->condition );
jack_client_close( handle->client );
if ( handle->ports[0] ) free( handle->ports[0] );
if ( handle->ports[1] ) free( handle->ports[1] );
delete handle;
stream_.apiHandle = 0;
}
for ( int i=0; i<2; i++ ) {
if ( stream_.userBuffer[i] ) {
free( stream_.userBuffer[i] );
stream_.userBuffer[i] = 0;
}
}
if ( stream_.deviceBuffer ) {
free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
return FAILURE;
}
void RtApiJack :: closeStream( void )
{
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiJack::closeStream(): no open stream to close!";
error( RtError::WARNING );
return;
}
JackHandle *handle = (JackHandle *) stream_.apiHandle;
if ( handle ) {
if ( stream_.state == STREAM_RUNNING )
jack_deactivate( handle->client );
jack_client_close( handle->client );
}
if ( handle ) {
if ( handle->ports[0] ) free( handle->ports[0] );
if ( handle->ports[1] ) free( handle->ports[1] );
pthread_cond_destroy( &handle->condition );
delete handle;
stream_.apiHandle = 0;
}
for ( int i=0; i<2; i++ ) {
if ( stream_.userBuffer[i] ) {
free( stream_.userBuffer[i] );
stream_.userBuffer[i] = 0;
}
}
if ( stream_.deviceBuffer ) {
free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
stream_.mode = UNINITIALIZED;
stream_.state = STREAM_CLOSED;
}
void RtApiJack :: startStream( void )
{
verifyStream();
if ( stream_.state == STREAM_RUNNING ) {
errorText_ = "RtApiJack::startStream(): the stream is already running!";
error( RtError::WARNING );
return;
}
MUTEX_LOCK(&stream_.mutex);
JackHandle *handle = (JackHandle *) stream_.apiHandle;
int result = jack_activate( handle->client );
if ( result ) {
errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
goto unlock;
}
const char **ports;
// Get the list of available ports.
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
result = 1;
ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
if ( ports == NULL) {
errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
goto unlock;
}
// 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 with offset.
for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
result = 1;
if ( ports[ stream_.channelOffset[0] + i ] )
result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
if ( result ) {
free( ports );
errorText_ = "RtApiJack::startStream(): error connecting output ports!";
goto unlock;
}
}
free(ports);
}
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
result = 1;
ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
if ( ports == NULL) {
errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
goto unlock;
}
// Now make the port connections. See note above.
for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
result = 1;
if ( ports[ stream_.channelOffset[1] + i ] )
result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
if ( result ) {
free( ports );
errorText_ = "RtApiJack::startStream(): error connecting input ports!";
goto unlock;
}
}
free(ports);
}
handle->drainCounter = 0;
handle->internalDrain = false;
stream_.state = STREAM_RUNNING;
unlock:
MUTEX_UNLOCK(&stream_.mutex);
if ( result == 0 ) return;
error( RtError::SYSTEM_ERROR );
}
void RtApiJack :: stopStream( void )
{
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
error( RtError::WARNING );
return;
}
MUTEX_LOCK( &stream_.mutex );
if ( stream_.state == STREAM_STOPPED ) {
MUTEX_UNLOCK( &stream_.mutex );
return;
}
JackHandle *handle = (JackHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
if ( handle->drainCounter == 0 ) {
handle->drainCounter = 1;
pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
}
}
jack_deactivate( handle->client );
stream_.state = STREAM_STOPPED;
MUTEX_UNLOCK( &stream_.mutex );
}
void RtApiJack :: abortStream( void )
{
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
error( RtError::WARNING );
return;
}
JackHandle *handle = (JackHandle *) stream_.apiHandle;
handle->drainCounter = 1;
stopStream();
}
// This function will be called by a spawned thread when the user
// callback function signals that the stream should be stopped or
// aborted. It is necessary to handle it this way because the
// callbackEvent() function must return before the jack_deactivate()
// function will return.
extern "C" void *jackStopStream( void *ptr )
{
CallbackInfo *info = (CallbackInfo *) ptr;
RtApiJack *object = (RtApiJack *) info->object;
object->stopStream();
pthread_exit( NULL );
}
bool RtApiJack :: callbackEvent( unsigned long nframes )
{
if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
error( RtError::WARNING );
return FAILURE;
}
if ( stream_.bufferSize != nframes ) {
errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
error( RtError::WARNING );
return FAILURE;
}
CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
JackHandle *handle = (JackHandle *) stream_.apiHandle;
// Check if we were draining the stream and signal is finished.
if ( handle->drainCounter > 3 ) {
if ( handle->internalDrain == true ) {
ThreadHandle id;
pthread_create( &id, NULL, jackStopStream, info );
}
else
pthread_cond_signal( &handle->condition );
return SUCCESS;
}
MUTEX_LOCK( &stream_.mutex );
// The state might change while waiting on a mutex.
if ( stream_.state == STREAM_STOPPED ) {
MUTEX_UNLOCK( &stream_.mutex );
return SUCCESS;
}
// Invoke user callback first, to get fresh output data.
if ( handle->drainCounter == 0 ) {
RtAudioCallback callback = (RtAudioCallback) info->callback;
double streamTime = getStreamTime();
RtAudioStreamStatus status = 0;
if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
status |= RTAUDIO_OUTPUT_UNDERFLOW;
handle->xrun[0] = false;
}
if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
status |= RTAUDIO_INPUT_OVERFLOW;
handle->xrun[1] = false;
}
handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
stream_.bufferSize, streamTime, status, info->userData );
if ( handle->drainCounter == 2 ) {
MUTEX_UNLOCK( &stream_.mutex );
ThreadHandle id;
pthread_create( &id, NULL, jackStopStream, info );
return SUCCESS;
}
else if ( handle->drainCounter == 1 )
handle->internalDrain = true;
}
jack_default_audio_sample_t *jackbuffer;
unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
if ( handle->drainCounter > 0 ) { // write zeros to the output stream
for ( unsigned 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 );
memset( jackbuffer, 0, bufferBytes );
}
}
else if ( stream_.doConvertBuffer[0] ) {
convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
for ( unsigned 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 { // no buffer conversion
for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
}
}
if ( handle->drainCounter ) {
handle->drainCounter++;
goto unlock;
}
}
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
if ( stream_.doConvertBuffer[1] ) {
for ( unsigned 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 );
}
convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
}
else { // no buffer conversion
for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
}
}
}
unlock:
MUTEX_UNLOCK(&stream_.mutex);
RtApi::tickStreamTime();
return SUCCESS;
}
//******************** End of __UNIX_JACK__ *********************//
#endif
#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 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 "asiosys.h"
#include "asio.h"
#include "iasiothiscallresolver.h"
#include "asiodrivers.h"
#include <cmath>
AsioDrivers drivers;
ASIOCallbacks asioCallbacks;
ASIODriverInfo driverInfo;
CallbackInfo *asioCallbackInfo;
bool asioXRun;
struct AsioHandle {
int drainCounter; // Tracks callback counts when draining
bool internalDrain; // Indicates if stop is initiated from callback or not.
ASIOBufferInfo *bufferInfos;
HANDLE condition;
AsioHandle()
:drainCounter(0), internalDrain(false), bufferInfos(0) {}
};
// Function declarations (definitions at end of section)
static const char* getAsioErrorString( ASIOError result );
void sampleRateChanged( ASIOSampleRate sRate );
long asioMessages( long selector, long value, void* message, double* opt );
RtApiAsio :: RtApiAsio()
{
// ASIO cannot run on a multi-threaded appartment. You can call
// CoInitialize beforehand, but it must be for appartment threading
// (in which case, CoInitilialize will return S_FALSE here).
coInitialized_ = false;
HRESULT hr = CoInitialize( NULL );
if ( FAILED(hr) ) {
errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
error( RtError::WARNING );
}
coInitialized_ = true;
drivers.removeCurrentDriver();
driverInfo.asioVersion = 2;
// See note in DirectSound implementation about GetDesktopWindow().
driverInfo.sysRef = GetForegroundWindow();
}
RtApiAsio :: ~RtApiAsio()
{
if ( stream_.state != STREAM_CLOSED ) closeStream();
if ( coInitialized_ ) CoUninitialize();
}
unsigned int RtApiAsio :: getDeviceCount( void )
{
return (unsigned int) drivers.asioGetNumDev();
}
RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
{
RtAudio::DeviceInfo info;
info.probed = false;
// Get device ID
unsigned int nDevices = getDeviceCount();
if ( nDevices == 0 ) {
errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
error( RtError::INVALID_USE );
}
if ( device >= nDevices ) {
errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
error( RtError::INVALID_USE );
}
// If a stream is already open, we cannot probe other devices. Thus, use the saved results.
if ( stream_.state != STREAM_CLOSED ) {
if ( device >= devices_.size() ) {
errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
error( RtError::WARNING );
return info;
}
return devices_[ device ];
}
char driverName[32];
ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
if ( result != ASE_OK ) {
errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
info.name = driverName;
if ( !drivers.loadDriver( driverName ) ) {
errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
result = ASIOInit( &driverInfo );
if ( result != ASE_OK ) {
errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
// Determine the device channel information.
long inputChannels, outputChannels;
result = ASIOGetChannels( &inputChannels, &outputChannels );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
info.outputChannels = outputChannels;
info.inputChannels = inputChannels;
if ( info.outputChannels > 0 && info.inputChannels > 0 )
info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
// Determine the supported sample rates.
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.push_back( SAMPLE_RATES[i] );
}
// Determine supported data types ... just check first channel and assume rest are the same.
ASIOChannelInfo channelInfo;
channelInfo.channel = 0;
channelInfo.isInput = true;
if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
result = ASIOGetChannelInfo( &channelInfo );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
info.nativeFormats = 0;
if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
info.nativeFormats |= RTAUDIO_SINT16;
else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
info.nativeFormats |= RTAUDIO_SINT32;
else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
info.nativeFormats |= RTAUDIO_FLOAT32;
else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
info.nativeFormats |= RTAUDIO_FLOAT64;
if ( getDefaultOutputDevice() == device )
info.isDefaultOutput = true;
if ( getDefaultInputDevice() == device )
info.isDefaultInput = true;
info.probed = true;
drivers.removeCurrentDriver();
return info;
}
void bufferSwitch( long index, ASIOBool processNow )
{
RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
object->callbackEvent( index );
}
void RtApiAsio :: saveDeviceInfo( void )
{
devices_.clear();
unsigned int nDevices = getDeviceCount();
devices_.resize( nDevices );
for ( unsigned int i=0; i<nDevices; i++ )
devices_[i] = getDeviceInfo( i );
}
bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options )
{
// For ASIO, a duplex stream MUST use the same driver.
if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
return FAILURE;
}
char driverName[32];
ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
if ( result != ASE_OK ) {
errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
// The getDeviceInfo() function will not work when a stream is open
// because ASIO does not allow multiple devices to run at the same
// time. Thus, we'll probe the system before opening a stream and
// save the results for use by getDeviceInfo().
this->saveDeviceInfo();
// Only load the driver once for duplex stream.
if ( mode != INPUT || stream_.mode != OUTPUT ) {
if ( !drivers.loadDriver( driverName ) ) {
errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
result = ASIOInit( &driverInfo );
if ( result != ASE_OK ) {
errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
}
// Check the device channel count.
long inputChannels, outputChannels;
result = ASIOGetChannels( &inputChannels, &outputChannels );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
stream_.nDeviceChannels[mode] = channels;
stream_.nUserChannels[mode] = channels;
stream_.channelOffset[mode] = firstChannel;
// Verify the sample rate is supported.
result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
// Get the current sample rate
ASIOSampleRate currentRate;
result = ASIOGetSampleRate( &currentRate );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
errorText_ = errorStream_.str();
return FAILURE;
}
// Set the sample rate only if necessary
if ( currentRate != sampleRate ) {
result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
}
// Determine the driver data type.
ASIOChannelInfo channelInfo;
channelInfo.channel = 0;
if ( mode == OUTPUT ) channelInfo.isInput = false;
else channelInfo.isInput = true;
result = ASIOGetChannelInfo( &channelInfo );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
errorText_ = errorStream_.str();
return FAILURE;
}
// Assuming WINDOWS host is always little-endian.
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;
}
else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
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;
}
else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
}
if ( stream_.deviceFormat[mode] == 0 ) {
drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
errorText_ = errorStream_.str();
return FAILURE;
}
// Set the buffer size. For a duplex stream, this will end up
// setting the buffer size based on the input constraints, which
// should be ok.
long minSize, maxSize, preferSize, granularity;
result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
if ( result != ASE_OK ) {
drivers.removeCurrentDriver();
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
errorText_ = errorStream_.str();
return FAILURE;
}
if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
else if ( granularity == -1 ) {
// Make sure bufferSize is a power of two.
int log2_of_min_size = 0;
int log2_of_max_size = 0;
for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
}
long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
int min_delta_num = log2_of_min_size;
for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
if (current_delta < min_delta) {
min_delta = current_delta;
min_delta_num = i;
}
}
*bufferSize = ( (unsigned int)1 << min_delta_num );
if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
}
else if ( granularity != 0 ) {
// Set to an even multiple of granularity, rounding up.
*bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
}
if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {
drivers.removeCurrentDriver();
errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
return FAILURE;
}
stream_.bufferSize = *bufferSize;
stream_.nBuffers = 2;
if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
else stream_.userInterleaved = true;
// ASIO always uses non-interleaved buffers.
stream_.deviceInterleaved[mode] = false;
// Allocate, if necessary, our AsioHandle structure for the stream.
AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
if ( handle == 0 ) {
try {
handle = new AsioHandle;
}
catch ( std::bad_alloc& ) {
//if ( handle == NULL ) {
drivers.removeCurrentDriver();
errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
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.
long inputLatency, outputLatency;
if ( mode == INPUT && stream_.mode == OUTPUT ) {
ASIODisposeBuffers();
if ( handle->bufferInfos ) free( handle->bufferInfos );
}
// Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
bool buffersAllocated = false;
unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
if ( handle->bufferInfos == NULL ) {
errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
errorText_ = errorStream_.str();
goto error;
}
ASIOBufferInfo *infos;
infos = handle->bufferInfos;
for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
infos->isInput = ASIOFalse;
infos->channelNum = i + stream_.channelOffset[0];
infos->buffers[0] = infos->buffers[1] = 0;
}
for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
infos->isInput = ASIOTrue;
infos->channelNum = i + stream_.channelOffset[1];
infos->buffers[0] = infos->buffers[1] = 0;
}
// Set up the ASIO callback structure and create the ASIO data buffers.
asioCallbacks.bufferSwitch = &bufferSwitch;
asioCallbacks.sampleRateDidChange = &sampleRateChanged;
asioCallbacks.asioMessage = &asioMessages;
asioCallbacks.bufferSwitchTimeInfo = NULL;
result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
if ( result != ASE_OK ) {
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
errorText_ = errorStream_.str();
goto error;
}
buffersAllocated = true;
// Set flags for buffer conversion.
stream_.doConvertBuffer[mode] = false;
if ( stream_.userFormat != stream_.deviceFormat[mode] )
stream_.doConvertBuffer[mode] = true;
if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
stream_.nUserChannels[mode] > 1 )
stream_.doConvertBuffer[mode] = true;
// Allocate necessary internal buffers
unsigned long bufferBytes;
bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
if ( stream_.userBuffer[mode] == NULL ) {
errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
goto error;
}
if ( stream_.doConvertBuffer[mode] ) {
bool makeBuffer = true;
bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
if ( mode == INPUT ) {
if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
if ( bufferBytes <= bytesOut ) makeBuffer = false;
}
}
if ( makeBuffer ) {
bufferBytes *= *bufferSize;
if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
if ( stream_.deviceBuffer == NULL ) {
errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
goto error;
}
}
}
stream_.sampleRate = sampleRate;
stream_.device[mode] = device;
stream_.state = STREAM_STOPPED;
asioCallbackInfo = &stream_.callbackInfo;
stream_.callbackInfo.object = (void *) this;
if ( stream_.mode == OUTPUT && mode == INPUT )
// We had already set up an output stream.
stream_.mode = DUPLEX;
else
stream_.mode = mode;
// Determine device latencies
result = ASIOGetLatencies( &inputLatency, &outputLatency );
if ( result != ASE_OK ) {
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
errorText_ = errorStream_.str();
error( RtError::WARNING); // warn but don't fail
}
else {
stream_.latency[0] = outputLatency;
stream_.latency[1] = inputLatency;
}
// Setup the buffer conversion information structure. We don't use
// buffers to do channel offsets, so we override that parameter
// here.
if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
return SUCCESS;
error:
if ( buffersAllocated )
ASIODisposeBuffers();
drivers.removeCurrentDriver();
if ( handle ) {
CloseHandle( handle->condition );
if ( handle->bufferInfos )
free( handle->bufferInfos );
delete handle;
stream_.apiHandle = 0;
}
for ( int i=0; i<2; i++ ) {
if ( stream_.userBuffer[i] ) {
free( stream_.userBuffer[i] );
stream_.userBuffer[i] = 0;
}
}
if ( stream_.deviceBuffer ) {
free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
return FAILURE;
}
void RtApiAsio :: closeStream()
{
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
error( RtError::WARNING );
return;
}
if ( stream_.state == STREAM_RUNNING ) {
stream_.state = STREAM_STOPPED;
ASIOStop();
}
ASIODisposeBuffers();
drivers.removeCurrentDriver();
AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
if ( handle ) {
CloseHandle( handle->condition );
if ( handle->bufferInfos )
free( handle->bufferInfos );
delete handle;
stream_.apiHandle = 0;
}
for ( int i=0; i<2; i++ ) {
if ( stream_.userBuffer[i] ) {
free( stream_.userBuffer[i] );
stream_.userBuffer[i] = 0;
}
}
if ( stream_.deviceBuffer ) {
free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
stream_.mode = UNINITIALIZED;
stream_.state = STREAM_CLOSED;
}
void RtApiAsio :: startStream()
{
verifyStream();
if ( stream_.state == STREAM_RUNNING ) {
errorText_ = "RtApiAsio::startStream(): the stream is already running!";
error( RtError::WARNING );
return;
}
MUTEX_LOCK( &stream_.mutex );
AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
ASIOError result = ASIOStart();
if ( result != ASE_OK ) {
errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
errorText_ = errorStream_.str();
goto unlock;
}
handle->drainCounter = 0;
handle->internalDrain = false;
stream_.state = STREAM_RUNNING;
asioXRun = false;
unlock:
MUTEX_UNLOCK( &stream_.mutex );
if ( result == ASE_OK ) return;
error( RtError::SYSTEM_ERROR );
}
void RtApiAsio :: stopStream()
{
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
error( RtError::WARNING );
return;
}
MUTEX_LOCK( &stream_.mutex );
if ( stream_.state == STREAM_STOPPED ) {
MUTEX_UNLOCK( &stream_.mutex );
return;
}
AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
if ( handle->drainCounter == 0 ) {
handle->drainCounter = 1;
MUTEX_UNLOCK( &stream_.mutex );
WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled
ResetEvent( handle->condition );
MUTEX_LOCK( &stream_.mutex );
}
}
ASIOError result = ASIOStop();
if ( result != ASE_OK ) {
errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
errorText_ = errorStream_.str();
}
stream_.state = STREAM_STOPPED;
MUTEX_UNLOCK( &stream_.mutex );
if ( result == ASE_OK ) return;
error( RtError::SYSTEM_ERROR );
}
void RtApiAsio :: abortStream()
{
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
error( RtError::WARNING );
return;
}
// The following lines were commented-out because some behavior was
// noted where the device buffers need to be zeroed to avoid
// continuing sound, even when the device buffers are completely
// disposed. So now, calling abort is the same as calling stop.
// AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
// handle->drainCounter = 1;
stopStream();
}
bool RtApiAsio :: callbackEvent( long bufferIndex )
{
if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
error( RtError::WARNING );
return FAILURE;
}
CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
// Check if we were draining the stream and signal is finished.
if ( handle->drainCounter > 3 ) {
if ( handle->internalDrain == false )
SetEvent( handle->condition );
else
stopStream();
return SUCCESS;
}
MUTEX_LOCK( &stream_.mutex );
// The state might change while waiting on a mutex.
if ( stream_.state == STREAM_STOPPED ) goto unlock;
// Invoke user callback to get fresh output data UNLESS we are
// draining stream.
if ( handle->drainCounter == 0 ) {
RtAudioCallback callback = (RtAudioCallback) info->callback;
double streamTime = getStreamTime();
RtAudioStreamStatus status = 0;
if ( stream_.mode != INPUT && asioXRun == true ) {
status |= RTAUDIO_OUTPUT_UNDERFLOW;
asioXRun = false;
}
if ( stream_.mode != OUTPUT && asioXRun == true ) {
status |= RTAUDIO_INPUT_OVERFLOW;
asioXRun = false;
}
handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
stream_.bufferSize, streamTime, status, info->userData );
if ( handle->drainCounter == 2 ) {
MUTEX_UNLOCK( &stream_.mutex );
abortStream();
return SUCCESS;
}
else if ( handle->drainCounter == 1 )
handle->internalDrain = true;
}
unsigned int nChannels, bufferBytes, i, j;
nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
if ( handle->drainCounter > 1 ) { // write zeros to the output stream
for ( i=0, j=0; i<nChannels; i++ ) {
if ( handle->bufferInfos[i].isInput != ASIOTrue )
memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
}
}
else if ( stream_.doConvertBuffer[0] ) {
convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
if ( stream_.doByteSwap[0] )
byteSwapBuffer( stream_.deviceBuffer,
stream_.bufferSize * stream_.nDeviceChannels[0],
stream_.deviceFormat[0] );
for ( i=0, j=0; i<nChannels; i++ ) {
if ( handle->bufferInfos[i].isInput != ASIOTrue )
memcpy( handle->bufferInfos[i].buffers[bufferIndex],
&stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
}
}
else {
if ( stream_.doByteSwap[0] )
byteSwapBuffer( stream_.userBuffer[0],
stream_.bufferSize * stream_.nUserChannels[0],
stream_.userFormat );
for ( i=0, j=0; i<nChannels; i++ ) {
if ( handle->bufferInfos[i].isInput != ASIOTrue )
memcpy( handle->bufferInfos[i].buffers[bufferIndex],
&stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
}
}
if ( handle->drainCounter ) {
handle->drainCounter++;
goto unlock;
}
}
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
if (stream_.doConvertBuffer[1]) {
// Always interleave ASIO input data.
for ( i=0, j=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] );
convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
}
else {
for ( i=0, j=0; i<nChannels; i++ ) {
if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
memcpy( &stream_.userBuffer[1][bufferBytes*j++],
handle->bufferInfos[i].buffers[bufferIndex],
bufferBytes );
}
}
if ( stream_.doByteSwap[1] )
byteSwapBuffer( stream_.userBuffer[1],
stream_.bufferSize * stream_.nUserChannels[1],
stream_.userFormat );
}
}
unlock:
// The following call was suggested by Malte Clasen. While the API
// documentation indicates it should not be required, some device
// drivers apparently do not function correctly without it.
ASIOOutputReady();
MUTEX_UNLOCK( &stream_.mutex );
RtApi::tickStreamTime();
return SUCCESS;
}
void sampleRateChanged( ASIOSampleRate sRate )
{
// The ASIO documentation says that this usually only happens during
// external sync. Audio processing is not stopped by the driver,
// actual sample rate might not have even changed, maybe only the
// sample rate status of an AES/EBU or S/PDIF digital input at the
// audio device.
RtApi *object = (RtApi *) asioCallbackInfo->object;
try {
object->stopStream();
}
catch ( RtError &exception ) {
std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
return;
}
std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
}
long asioMessages( long selector, long value, void* message, double* opt )
{
long ret = 0;
switch( selector ) {
case kAsioSelectorSupported:
if ( value == kAsioResetRequest
|| value == kAsioEngineVersion
|| value == kAsioResyncRequest
|| value == kAsioLatenciesChanged
// The following three were added for ASIO 2.0, you don't
// necessarily have to support them.
|| value == kAsioSupportsTimeInfo
|| value == kAsioSupportsTimeCode
|| value == kAsioSupportsInputMonitor)
ret = 1L;
break;
case kAsioResetRequest:
// Defer the task and perform the reset of the driver during the
// next "safe" situation. You cannot reset the driver right now,
// as this code is called from the driver. Reset the driver is
// done by completely destruct is. I.e. ASIOStop(),
// ASIODisposeBuffers(), Destruction Afterwards you initialize the
// driver again.
std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
ret = 1L;
break;
case kAsioResyncRequest:
// This informs the application that the driver encountered some
// non-fatal data loss. It is used for synchronization purposes
// of different media. Added mainly to work around the Win16Mutex
// problems in Windows 95/98 with the Windows Multimedia system,
// which could lose data because the Mutex was held too long by
// another thread. However a driver can issue it in other
// situations, too.
// std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
asioXRun = true;
ret = 1L;
break;
case kAsioLatenciesChanged:
// This will inform the host application that the drivers were
// latencies changed. Beware, it this does not mean that the
// buffer sizes have changed! You might need to update internal
// delay data.
std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
ret = 1L;
break;
case kAsioEngineVersion:
// Return the supported ASIO version of the host application. If
// a host application does not implement this selector, ASIO 1.0
// is assumed by the driver.
ret = 2L;
break;
case kAsioSupportsTimeInfo:
// Informs the driver whether the
// asioCallbacks.bufferSwitchTimeInfo() callback is supported.
// For compatibility with ASIO 1.0 drivers the host application
// should always support the "old" bufferSwitch method, too.
ret = 0;
break;
case kAsioSupportsTimeCode:
// Informs the driver whether application is interested in time
// code info. If an application does not need to know about time
// code, the driver has less work to do.
ret = 0;
break;
}
return ret;
}
static const char* getAsioErrorString( ASIOError result )
{
struct Messages
{
ASIOError value;
const char*message;
};
static Messages m[] =
{
{ ASE_NotPresent, "Hardware input or output is not present or available." },
{ ASE_HWMalfunction, "Hardware is malfunctioning." },
{ ASE_InvalidParameter, "Invalid input parameter." },
{ ASE_InvalidMode, "Invalid mode." },
{ ASE_SPNotAdvancing, "Sample position not advancing." },
{ ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
{ ASE_NoMemory, "Not enough memory to complete the request." }
};
for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
if ( m[i].value == result ) return m[i].message;
return "Unknown error.";
}
//******************** End of __WINDOWS_ASIO__ *********************//
#endif
#if defined(__WINDOWS_DS__) // Windows DirectSound API
// Modified by Robin Davies, October 2005
// - Improvements to DirectX pointer chasing.
// - Backdoor RtDsStatistics hook provides DirectX performance information.
// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
// - Auto-call CoInitialize for DSOUND and ASIO platforms.
// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
#include <dsound.h>
#include <assert.h>
#if defined(__MINGW32__)
// missing from latest mingw winapi
#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
#endif
#define MINIMUM_DEVICE_BUFFER_SIZE 32768
#ifdef _MSC_VER // if Microsoft Visual C++
#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
#endif
static inline DWORD dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
{
if ( laterPointer > earlierPointer )
return laterPointer - earlierPointer;
else
return laterPointer - earlierPointer + bufferSize;
}
static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
{
if ( pointer > bufferSize ) pointer -= bufferSize;
if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
if ( pointer < earlierPointer ) pointer += bufferSize;
return pointer >= earlierPointer && pointer < laterPointer;
}
// A structure to hold various information related to the DirectSound
// API implementation.
struct DsHandle {
unsigned int drainCounter; // Tracks callback counts when draining
bool internalDrain; // Indicates if stop is initiated from callback or not.
void *id[2];
void *buffer[2];
bool xrun[2];
UINT bufferPointer[2];
DWORD dsBufferSize[2];
DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
HANDLE condition;
DsHandle()
:drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
};
/*
RtApiDs::RtDsStatistics RtApiDs::statistics;
// Provides a backdoor hook to monitor for DirectSound read overruns and write underruns.
RtApiDs::RtDsStatistics RtApiDs::getDsStatistics()
{
RtDsStatistics s = statistics;
// update the calculated fields.
if ( s.inputFrameSize != 0 )
s.latency += s.readDeviceSafeLeadBytes * 1.0 / s.inputFrameSize / s.sampleRate;
if ( s.outputFrameSize != 0 )
s.latency += (s.writeDeviceSafeLeadBytes + s.writeDeviceBufferLeadBytes) * 1.0 / s.outputFrameSize / s.sampleRate;
return s;
}
*/
// Declarations for utility functions, callbacks, and structures
// specific to the DirectSound implementation.
static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
LPCTSTR description,
LPCTSTR module,
LPVOID lpContext );
static char* getErrorString( int code );
extern "C" unsigned __stdcall callbackHandler( void *ptr );
struct EnumInfo {
bool isInput;
bool getDefault;
bool findIndex;
unsigned int counter;
unsigned int index;
LPGUID id;
std::string name;
EnumInfo()
: isInput(false), getDefault(false), findIndex(false), counter(0), index(0) {}
};
RtApiDs :: RtApiDs()
{
// Dsound will run both-threaded. If CoInitialize fails, then just
// accept whatever the mainline chose for a threading model.
coInitialized_ = false;
HRESULT hr = CoInitialize( NULL );
if ( !FAILED( hr ) ) coInitialized_ = true;
}
RtApiDs :: ~RtApiDs()
{
if ( coInitialized_ ) CoUninitialize(); // balanced call.
if ( stream_.state != STREAM_CLOSED ) closeStream();
}
unsigned int RtApiDs :: getDefaultInputDevice( void )
{
// Count output devices.
EnumInfo info;
HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return 0;
}
// Now enumerate input devices until we find the id = NULL.
info.isInput = true;
info.getDefault = true;
result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return 0;
}
if ( info.counter > 0 ) return info.counter - 1;
return 0;
}
unsigned int RtApiDs :: getDefaultOutputDevice( void )
{
// Enumerate output devices until we find the id = NULL.
EnumInfo info;
info.getDefault = true;
HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return 0;
}
if ( info.counter > 0 ) return info.counter - 1;
return 0;
}
unsigned int RtApiDs :: getDeviceCount( void )
{
// Count DirectSound devices.
EnumInfo info;
HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
errorText_ = errorStream_.str();
error( RtError::WARNING );
}
// Count DirectSoundCapture devices.
info.isInput = true;
result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
errorText_ = errorStream_.str();
error( RtError::WARNING );
}
return info.counter;
}
RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
{
// Because DirectSound always enumerates input and output devices
// separately (and because we don't attempt to combine devices
// internally), none of our "devices" will ever be duplex.
RtAudio::DeviceInfo info;
info.probed = false;
// Enumerate through devices to find the id (if it exists). Note
// that we have to do the output enumeration first, even if this is
// an input device, in order for the device counter to be correct.
EnumInfo dsinfo;
dsinfo.findIndex = true;
dsinfo.index = device;
HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating output devices!";
errorText_ = errorStream_.str();
error( RtError::WARNING );
}
if ( dsinfo.name.empty() ) goto probeInput;
LPDIRECTSOUND output;
DSCAPS outCaps;
result = DirectSoundCreate( dsinfo.id, &output, NULL );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
outCaps.dwSize = sizeof( outCaps );
result = output->GetCaps( &outCaps );
if ( FAILED( result ) ) {
output->Release();
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
// Get output channel information.
info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
// Get sample rate information.
info.sampleRates.clear();
for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate )
info.sampleRates.push_back( SAMPLE_RATES[k] );
}
// Get format information.
if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
output->Release();
if ( getDefaultOutputDevice() == device )
info.isDefaultOutput = true;
// Copy name and return.
info.name = dsinfo.name;
info.probed = true;
return info;
probeInput:
dsinfo.isInput = true;
result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!";
errorText_ = errorStream_.str();
error( RtError::WARNING );
}
if ( dsinfo.name.empty() ) return info;
LPDIRECTSOUNDCAPTURE input;
result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
DSCCAPS inCaps;
inCaps.dwSize = sizeof( inCaps );
result = input->GetCaps( &inCaps );
if ( FAILED( result ) ) {
input->Release();
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
// Get input channel information.
info.inputChannels = inCaps.dwChannels;
// Get sample rate and format information.
if ( inCaps.dwChannels == 2 ) {
if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
if ( info.nativeFormats & RTAUDIO_SINT16 ) {
if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.sampleRates.push_back( 11025 );
if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.sampleRates.push_back( 22050 );
if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.sampleRates.push_back( 44100 );
if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.sampleRates.push_back( 96000 );
}
else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.sampleRates.push_back( 11025 );
if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.sampleRates.push_back( 22050 );
if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.sampleRates.push_back( 44100 );
if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.sampleRates.push_back( 44100 );
}
}
else if ( inCaps.dwChannels == 1 ) {
if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
if ( info.nativeFormats & RTAUDIO_SINT16 ) {
if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.sampleRates.push_back( 11025 );
if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.sampleRates.push_back( 22050 );
if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.sampleRates.push_back( 44100 );
if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.sampleRates.push_back( 96000 );
}
else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.sampleRates.push_back( 11025 );
if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.sampleRates.push_back( 22050 );
if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.sampleRates.push_back( 44100 );
if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.sampleRates.push_back( 96000 );
}
}
else info.inputChannels = 0; // technically, this would be an error
input->Release();
if ( info.inputChannels == 0 ) return info;
if ( getDefaultInputDevice() == device )
info.isDefaultInput = true;
// Copy name and return.
info.name = dsinfo.name;
info.probed = true;
return info;
}
bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options )
{
if ( channels + firstChannel > 2 ) {
errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
return FAILURE;
}
// Enumerate through devices to find the id (if it exists). Note
// that we have to do the output enumeration first, even if this is
// an input device, in order for the device counter to be correct.
EnumInfo dsinfo;
dsinfo.findIndex = true;
dsinfo.index = device;
HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!";
errorText_ = errorStream_.str();
return FAILURE;
}
if ( mode == OUTPUT ) {
if ( dsinfo.name.empty() ) {
errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
errorText_ = errorStream_.str();
return FAILURE;
}
}
else { // mode == INPUT
dsinfo.isInput = true;
HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!";
errorText_ = errorStream_.str();
return FAILURE;
}
if ( dsinfo.name.empty() ) {
errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
errorText_ = errorStream_.str();
return FAILURE;
}
}
// 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
// window. Also, if the application window closes before the
// DirectSound buffer, DirectSound can crash. However, for console
// applications, no sound was produced when using GetDesktopWindow().
HWND hWnd = GetForegroundWindow();
// Check the numberOfBuffers parameter and limit the lowest value to
// two. This is a judgement call and a value of two is probably too
// low for capture, but it should work for playback.
int nBuffers = 0;
if ( options ) nBuffers = options->numberOfBuffers;
if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
if ( nBuffers < 2 ) nBuffers = 3;
// Create the wave format structure. The data format setting will
// be determined later.
WAVEFORMATEX waveFormat;
ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nChannels = channels + firstChannel;
waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
// Determine the device buffer size. By default, 32k, but we will
// grow it to make allowances for very large software buffer sizes.
DWORD dsBufferSize = 0;
DWORD dsPointerLeadTime = 0;
long bufferBytes = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this
void *ohandle = 0, *bhandle = 0;
if ( mode == OUTPUT ) {
LPDIRECTSOUND output;
result = DirectSoundCreate( dsinfo.id, &output, NULL );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
DSCAPS outCaps;
outCaps.dwSize = sizeof( outCaps );
result = output->GetCaps( &outCaps );
if ( FAILED( result ) ) {
output->Release();
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
// Check channel information.
if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsinfo.name << ") does not support stereo playback.";
errorText_ = errorStream_.str();
return FAILURE;
}
// Check format information. Use 16-bit format unless not
// supported or user requests 8-bit.
if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
!( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
waveFormat.wBitsPerSample = 16;
stream_.deviceFormat[mode] = RTAUDIO_SINT16;
}
else {
waveFormat.wBitsPerSample = 8;
stream_.deviceFormat[mode] = RTAUDIO_SINT8;
}
stream_.userFormat = format;
// Update wave format structure and buffer information.
waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
// If the user wants an even bigger buffer, increase the device buffer size accordingly.
while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes )
bufferBytes *= 2;
// Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
// result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
// Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
if ( FAILED( result ) ) {
output->Release();
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
// Even though we will write to the secondary buffer, we need to
// access the primary buffer to set the correct output format
// (since the default is 8-bit, 22 kHz!). Setup the DS primary
// buffer description.
DSBUFFERDESC bufferDescription;
ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
bufferDescription.dwSize = sizeof( DSBUFFERDESC );
bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
// Obtain the primary buffer
LPDIRECTSOUNDBUFFER buffer;
result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
if ( FAILED( result ) ) {
output->Release();
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
// Set the primary DS buffer sound format.
result = buffer->SetFormat( &waveFormat );
if ( FAILED( result ) ) {
output->Release();
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
// Setup the secondary DS buffer description.
dsBufferSize = (DWORD) bufferBytes;
ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
bufferDescription.dwSize = sizeof( DSBUFFERDESC );
bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
DSBCAPS_GLOBALFOCUS |
DSBCAPS_GETCURRENTPOSITION2 |
DSBCAPS_LOCHARDWARE ); // Force hardware mixing
bufferDescription.dwBufferBytes = bufferBytes;
bufferDescription.lpwfxFormat = &waveFormat;
// Try to create the secondary DS buffer. If that doesn't work,
// try to use software mixing. Otherwise, there's a problem.
result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
if ( FAILED( result ) ) {
bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
DSBCAPS_GLOBALFOCUS |
DSBCAPS_GETCURRENTPOSITION2 |
DSBCAPS_LOCSOFTWARE ); // Force software mixing
result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
if ( FAILED( result ) ) {
output->Release();
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
}
// Get the buffer size ... might be different from what we specified.
DSBCAPS dsbcaps;
dsbcaps.dwSize = sizeof( DSBCAPS );
result = buffer->GetCaps( &dsbcaps );
if ( FAILED( result ) ) {
output->Release();
buffer->Release();
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
bufferBytes = dsbcaps.dwBufferBytes;
// Lock the DS buffer
LPVOID audioPtr;
DWORD dataLen;
result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 );
if ( FAILED( result ) ) {
output->Release();
buffer->Release();
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
// Zero the DS buffer
ZeroMemory( audioPtr, dataLen );
// Unlock the DS buffer
result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
if ( FAILED( result ) ) {
output->Release();
buffer->Release();
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
dsBufferSize = bufferBytes;
ohandle = (void *) output;
bhandle = (void *) buffer;
}
if ( mode == INPUT ) {
LPDIRECTSOUNDCAPTURE input;
result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
DSCCAPS inCaps;
inCaps.dwSize = sizeof( inCaps );
result = input->GetCaps( &inCaps );
if ( FAILED( result ) ) {
input->Release();
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
// Check channel information.
if ( inCaps.dwChannels < channels + firstChannel ) {
errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
return FAILURE;
}
// Check format information. Use 16-bit format unless user
// requests 8-bit.
DWORD deviceFormats;
if ( channels + firstChannel == 2 ) {
deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
waveFormat.wBitsPerSample = 8;
stream_.deviceFormat[mode] = RTAUDIO_SINT8;
}
else { // assume 16-bit is supported
waveFormat.wBitsPerSample = 16;
stream_.deviceFormat[mode] = RTAUDIO_SINT16;
}
}
else { // channel == 1
deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
waveFormat.wBitsPerSample = 8;
stream_.deviceFormat[mode] = RTAUDIO_SINT8;
}
else { // assume 16-bit is supported
waveFormat.wBitsPerSample = 16;
stream_.deviceFormat[mode] = RTAUDIO_SINT16;
}
}
stream_.userFormat = format;
// Update wave format structure and buffer information.
waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
// If the user wants an even bigger buffer, increase the device buffer size accordingly.
while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes )
bufferBytes *= 2;
// Setup the secondary DS buffer description.
dsBufferSize = bufferBytes;
DSCBUFFERDESC bufferDescription;
ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
bufferDescription.dwFlags = 0;
bufferDescription.dwReserved = 0;
bufferDescription.dwBufferBytes = bufferBytes;
bufferDescription.lpwfxFormat = &waveFormat;
// Create the capture buffer.
LPDIRECTSOUNDCAPTUREBUFFER buffer;
result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
if ( FAILED( result ) ) {
input->Release();
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
// Get the buffer size ... might be different from what we specified.
DSCBCAPS dscbcaps;
dscbcaps.dwSize = sizeof( DSCBCAPS );
result = buffer->GetCaps( &dscbcaps );
if ( FAILED( result ) ) {
input->Release();
buffer->Release();
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
bufferBytes = dscbcaps.dwBufferBytes;
// Lock the capture buffer
LPVOID audioPtr;
DWORD dataLen;
result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 );
if ( FAILED( result ) ) {
input->Release();
buffer->Release();
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
// Zero the buffer
ZeroMemory( audioPtr, dataLen );
// Unlock the buffer
result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
if ( FAILED( result ) ) {
input->Release();
buffer->Release();
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsinfo.name << ")!";
errorText_ = errorStream_.str();
return FAILURE;
}
dsBufferSize = bufferBytes;
ohandle = (void *) input;
bhandle = (void *) buffer;
}
// Set various stream parameters
DsHandle *handle = 0;
stream_.nDeviceChannels[mode] = channels + firstChannel;
stream_.nUserChannels[mode] = channels;
stream_.bufferSize = *bufferSize;
stream_.channelOffset[mode] = firstChannel;
stream_.deviceInterleaved[mode] = true;
if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
else stream_.userInterleaved = true;
// Set flag for buffer conversion
stream_.doConvertBuffer[mode] = false;
if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
stream_.doConvertBuffer[mode] = true;
if (stream_.userFormat != stream_.deviceFormat[mode])
stream_.doConvertBuffer[mode] = true;
if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
stream_.nUserChannels[mode] > 1 )
stream_.doConvertBuffer[mode] = true;
// Allocate necessary internal buffers
bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
if ( stream_.userBuffer[mode] == NULL ) {
errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
goto error;
}
if ( stream_.doConvertBuffer[mode] ) {
bool makeBuffer = true;
bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
if ( mode == INPUT ) {
if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
}
}
if ( makeBuffer ) {
bufferBytes *= *bufferSize;
if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
if ( stream_.deviceBuffer == NULL ) {
errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
goto error;
}
}
}
// Allocate our DsHandle structures for the stream.
if ( stream_.apiHandle == 0 ) {
try {
handle = new DsHandle;
}
catch ( std::bad_alloc& ) {
errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
goto error;
}
// Create a manual-reset event.
handle->condition = CreateEvent( NULL, // no security
TRUE, // manual-reset
FALSE, // non-signaled initially
NULL ); // unnamed
stream_.apiHandle = (void *) handle;
}
else
handle = (DsHandle *) stream_.apiHandle;
handle->id[mode] = ohandle;
handle->buffer[mode] = bhandle;
handle->dsBufferSize[mode] = dsBufferSize;
handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
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 = nBuffers;
stream_.sampleRate = sampleRate;
// Setup the buffer conversion information structure.
if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
// Setup the callback thread.
unsigned threadId;
stream_.callbackInfo.object = (void *) this;
stream_.callbackInfo.isRunning = true;
stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
&stream_.callbackInfo, 0, &threadId );
if ( stream_.callbackInfo.thread == 0 ) {
errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
goto error;
}
// Boost DS thread priority
SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
return SUCCESS;
error:
if ( handle ) {
if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
if ( buffer ) buffer->Release();
object->Release();
}
if ( handle->buffer[1] ) {
LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
if ( buffer ) buffer->Release();
object->Release();
}
CloseHandle( handle->condition );
delete handle;
stream_.apiHandle = 0;
}
for ( int i=0; i<2; i++ ) {
if ( stream_.userBuffer[i] ) {
free( stream_.userBuffer[i] );
stream_.userBuffer[i] = 0;
}
}
if ( stream_.deviceBuffer ) {
free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
return FAILURE;
}
void RtApiDs :: closeStream()
{
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiDs::closeStream(): no open stream to close!";
error( RtError::WARNING );
return;
}
// Stop the callback thread.
stream_.callbackInfo.isRunning = false;
WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
CloseHandle( (HANDLE) stream_.callbackInfo.thread );
DsHandle *handle = (DsHandle *) stream_.apiHandle;
if ( handle ) {
if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
if ( buffer ) {
buffer->Stop();
buffer->Release();
}
object->Release();
}
if ( handle->buffer[1] ) {
LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
if ( buffer ) {
buffer->Stop();
buffer->Release();
}
object->Release();
}
CloseHandle( handle->condition );
delete handle;
stream_.apiHandle = 0;
}
for ( int i=0; i<2; i++ ) {
if ( stream_.userBuffer[i] ) {
free( stream_.userBuffer[i] );
stream_.userBuffer[i] = 0;
}
}
if ( stream_.deviceBuffer ) {
free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
stream_.mode = UNINITIALIZED;
stream_.state = STREAM_CLOSED;
}
void RtApiDs :: startStream()
{
verifyStream();
if ( stream_.state == STREAM_RUNNING ) {
errorText_ = "RtApiDs::startStream(): the stream is already running!";
error( RtError::WARNING );
return;
}
// Increase scheduler frequency on lesser windows (a side-effect of
// increasing timer accuracy). On greater windows (Win2K or later),
// this is already in effect.
MUTEX_LOCK( &stream_.mutex );
DsHandle *handle = (DsHandle *) stream_.apiHandle;
timeBeginPeriod( 1 );
/*
memset( &statistics, 0, sizeof( statistics ) );
statistics.sampleRate = stream_.sampleRate;
statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0];
*/
buffersRolling = false;
duplexPrerollBytes = 0;
if ( stream_.mode == DUPLEX ) {
// 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
}
HRESULT result = 0;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
//statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[0];
LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
errorText_ = errorStream_.str();
goto unlock;
}
}
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
//statistics.inputFrameSize = formatBytes( stream_.deviceFormat[1]) * stream_.nDeviceChannels[1];
LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
result = buffer->Start( DSCBSTART_LOOPING );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
errorText_ = errorStream_.str();
goto unlock;
}
}
handle->drainCounter = 0;
handle->internalDrain = false;
stream_.state = STREAM_RUNNING;
unlock:
MUTEX_UNLOCK( &stream_.mutex );
if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
}
void RtApiDs :: stopStream()
{
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
error( RtError::WARNING );
return;
}
MUTEX_LOCK( &stream_.mutex );
if ( stream_.state == STREAM_STOPPED ) {
MUTEX_UNLOCK( &stream_.mutex );
return;
}
HRESULT result = 0;
LPVOID audioPtr;
DWORD dataLen;
DsHandle *handle = (DsHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
if ( handle->drainCounter == 0 ) {
handle->drainCounter = 1;
MUTEX_UNLOCK( &stream_.mutex );
WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled
ResetEvent( handle->condition );
MUTEX_LOCK( &stream_.mutex );
}
// Stop the buffer and clear memory
LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
result = buffer->Stop();
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
errorText_ = errorStream_.str();
goto unlock;
}
// 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, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
errorText_ = errorStream_.str();
goto unlock;
}
// Zero the DS buffer
ZeroMemory( audioPtr, dataLen );
// Unlock the DS buffer
result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
errorText_ = errorStream_.str();
goto unlock;
}
// If we start playing again, we must begin at beginning of buffer.
handle->bufferPointer[0] = 0;
}
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
audioPtr = NULL;
dataLen = 0;
result = buffer->Stop();
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
errorText_ = errorStream_.str();
goto unlock;
}
// 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, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
errorText_ = errorStream_.str();
goto unlock;
}
// Zero the DS buffer
ZeroMemory( audioPtr, dataLen );
// Unlock the DS buffer
result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
errorText_ = errorStream_.str();
goto unlock;
}
// If we start recording again, we must begin at beginning of buffer.
handle->bufferPointer[1] = 0;
}
unlock:
timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
stream_.state = STREAM_STOPPED;
MUTEX_UNLOCK( &stream_.mutex );
if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
}
void RtApiDs :: abortStream()
{
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
error( RtError::WARNING );
return;
}
DsHandle *handle = (DsHandle *) stream_.apiHandle;
handle->drainCounter = 1;
stopStream();
}
void RtApiDs :: callbackEvent()
{
if ( stream_.state == STREAM_STOPPED ) {
Sleep(50); // sleep 50 milliseconds
return;
}
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
error( RtError::WARNING );
return;
}
CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
DsHandle *handle = (DsHandle *) stream_.apiHandle;
// Check if we were draining the stream and signal is finished.
if ( handle->drainCounter > stream_.nBuffers + 2 ) {
if ( handle->internalDrain == false )
SetEvent( handle->condition );
else
stopStream();
return;
}
MUTEX_LOCK( &stream_.mutex );
// The state might change while waiting on a mutex.
if ( stream_.state == STREAM_STOPPED ) {
MUTEX_UNLOCK( &stream_.mutex );
return;
}
// Invoke user callback to get fresh output data UNLESS we are
// draining stream.
if ( handle->drainCounter == 0 ) {
RtAudioCallback callback = (RtAudioCallback) info->callback;
double streamTime = getStreamTime();
RtAudioStreamStatus status = 0;
if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
status |= RTAUDIO_OUTPUT_UNDERFLOW;
handle->xrun[0] = false;
}
if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
status |= RTAUDIO_INPUT_OVERFLOW;
handle->xrun[1] = false;
}
handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
stream_.bufferSize, streamTime, status, info->userData );
if ( handle->drainCounter == 2 ) {
MUTEX_UNLOCK( &stream_.mutex );
abortStream();
return;
}
else if ( handle->drainCounter == 1 )
handle->internalDrain = true;
}
HRESULT result;
DWORD currentWritePos, safeWritePos;
DWORD currentReadPos, safeReadPos;
DWORD leadPos;
UINT nextWritePos;
#ifdef GENERATE_DEBUG_LOG
DWORD writeTime, readTime;
#endif
LPVOID buffer1 = NULL;
LPVOID buffer2 = NULL;
DWORD bufferSize1 = 0;
DWORD bufferSize2 = 0;
char *buffer;
long bufferBytes;
if ( stream_.mode == DUPLEX && !buffersRolling ) {
//assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
// It takes a while for the devices to get rolling. As a result,
// there's no guarantee that the capture and write device pointers
// will move in lockstep. Wait here for both devices to start
// rolling, and then set our buffer pointers accordingly.
// e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
// bytes later than the write buffer.
// Stub: a serious risk of having a pre-emptive scheduling round
// take place between the two GetCurrentPosition calls... but I'm
// really not sure how to solve the problem. Temporarily boost to
// Realtime priority, maybe; but I'm not sure what priority the
// DirectSound service threads run at. We *should* be roughly
// within a ms or so of correct.
LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
DWORD initialWritePos, initialSafeWritePos;
DWORD initialReadPos, initialSafeReadPos;
result = dsWriteBuffer->GetCurrentPosition( &initialWritePos, &initialSafeWritePos );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
result = dsCaptureBuffer->GetCurrentPosition( &initialReadPos, &initialSafeReadPos );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
while ( true ) {
result = dsWriteBuffer->GetCurrentPosition( &currentWritePos, &safeWritePos );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
result = dsCaptureBuffer->GetCurrentPosition( &currentReadPos, &safeReadPos );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
if ( safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos ) break;
Sleep( 1 );
}
//assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
buffersRolling = true;
handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] );
handle->bufferPointer[1] = safeReadPos;
}
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
if ( handle->drainCounter > 1 ) { // write zeros to the output stream
bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
bufferBytes *= formatBytes( stream_.userFormat );
memset( stream_.userBuffer[0], 0, bufferBytes );
}
// Setup parameters and do buffer conversion if necessary.
if ( stream_.doConvertBuffer[0] ) {
buffer = stream_.deviceBuffer;
convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
bufferBytes *= formatBytes( stream_.deviceFormat[0] );
}
else {
buffer = stream_.userBuffer[0];
bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
bufferBytes *= formatBytes( stream_.userFormat );
}
// No byte swapping necessary in DirectSound implementation.
// Ahhh ... windoze. 16-bit data is signed but 8-bit data is
// unsigned. So, we need to convert our signed 8-bit data here to
// unsigned.
if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
DWORD dsBufferSize = handle->dsBufferSize[0];
nextWritePos = handle->bufferPointer[0];
DWORD endWrite;
while ( true ) {
// Find out where the read and "safe write" pointers are.
result = dsBuffer->GetCurrentPosition( &currentWritePos, &safeWritePos );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
leadPos = safeWritePos + handle->dsPointerLeadTime[0];
if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize;
if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset
endWrite = nextWritePos + bufferBytes;
// Check whether the entire write region is behind the play pointer.
if ( leadPos >= endWrite ) break;
// If we are here, then we must wait until the play pointer gets
// beyond the write region. The approach here is to use the
// Sleep() function to suspend operation until safePos catches
// up. Calculate number of milliseconds to wait as:
// time = distance * (milliseconds/second) * fudgefactor /
// ((bytes/sample) * (samples/second))
// A "fudgefactor" less than 1 is used because it was found
// that sleeping too long was MUCH worse than sleeping for
// several shorter periods.
double millis = ( endWrite - leadPos ) * 900.0;
millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
if ( millis < 1.0 ) millis = 1.0;
if ( millis > 50.0 ) {
static int nOverruns = 0;
++nOverruns;
}
Sleep( (DWORD) millis );
}
//if ( statistics.writeDeviceSafeLeadBytes < dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ) ) {
// statistics.writeDeviceSafeLeadBytes = dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] );
//}
if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize )
|| dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) {
// We've strayed into the forbidden zone ... resync the read pointer.
//++statistics.numberOfWriteUnderruns;
handle->xrun[0] = true;
nextWritePos = safeWritePos + handle->dsPointerLeadTime[0] - bufferBytes + dsBufferSize;
while ( nextWritePos >= dsBufferSize ) nextWritePos -= dsBufferSize;
handle->bufferPointer[0] = nextWritePos;
endWrite = nextWritePos + bufferBytes;
}
// Lock free space in the buffer
result = dsBuffer->Lock( nextWritePos, bufferBytes, &buffer1,
&bufferSize1, &buffer2, &bufferSize2, 0 );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
// Copy our buffer into the DS buffer
CopyMemory( buffer1, buffer, bufferSize1 );
if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
// Update our buffer offset and unlock sound buffer
dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
nextWritePos = ( nextWritePos + bufferSize1 + bufferSize2 ) % dsBufferSize;
handle->bufferPointer[0] = nextWritePos;
if ( handle->drainCounter ) {
handle->drainCounter++;
goto unlock;
}
}
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
// Setup parameters.
if ( stream_.doConvertBuffer[1] ) {
buffer = stream_.deviceBuffer;
bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
bufferBytes *= formatBytes( stream_.deviceFormat[1] );
}
else {
buffer = stream_.userBuffer[1];
bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
bufferBytes *= formatBytes( stream_.userFormat );
}
LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
long nextReadPos = handle->bufferPointer[1];
DWORD dsBufferSize = handle->dsBufferSize[1];
// Find out where the write and "safe read" pointers are.
result = dsBuffer->GetCurrentPosition( &currentReadPos, &safeReadPos );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset
DWORD endRead = nextReadPos + bufferBytes;
// Handling depends on whether we are INPUT or DUPLEX.
// If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
// then a wait here will drag the write pointers into the forbidden zone.
//
// In DUPLEX mode, rather than wait, we will back off the read pointer until
// it's in a safe position. This causes dropouts, but it seems to be the only
// practical way to sync up the read and write pointers reliably, given the
// the very complex relationship between phase and increment of the read and write
// pointers.
//
// In order to minimize audible dropouts in DUPLEX mode, we will
// provide a pre-roll period of 0.5 seconds in which we return
// zeros from the read buffer while the pointers sync up.
if ( stream_.mode == DUPLEX ) {
if ( safeReadPos < endRead ) {
if ( duplexPrerollBytes <= 0 ) {
// Pre-roll time over. Be more agressive.
int adjustment = endRead-safeReadPos;
handle->xrun[1] = true;
//++statistics.numberOfReadOverruns;
// Two cases:
// - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
// and perform fine adjustments later.
// - small adjustments: back off by twice as much.
if ( adjustment >= 2*bufferBytes )
nextReadPos = safeReadPos-2*bufferBytes;
else
nextReadPos = safeReadPos-bufferBytes-adjustment;
//statistics.readDeviceSafeLeadBytes = currentReadPos-nextReadPos;
//if ( statistics.readDeviceSafeLeadBytes < 0) statistics.readDeviceSafeLeadBytes += dsBufferSize;
if ( nextReadPos < 0 ) nextReadPos += dsBufferSize;
}
else {
// In pre=roll time. Just do it.
nextReadPos = safeReadPos-bufferBytes;
while ( nextReadPos < 0 ) nextReadPos += dsBufferSize;
}
endRead = nextReadPos + bufferBytes;
}
}
else { // mode == INPUT
while ( safeReadPos < endRead ) {
// See comments for playback.
double millis = (endRead - safeReadPos) * 900.0;
millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
if ( millis < 1.0 ) millis = 1.0;
Sleep( (DWORD) millis );
// Wake up, find out where we are now
result = dsBuffer->GetCurrentPosition( &currentReadPos, &safeReadPos );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset
}
}
//if (statistics.readDeviceSafeLeadBytes < dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ) )
// statistics.readDeviceSafeLeadBytes = dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize );
// Lock free space in the buffer
result = dsBuffer->Lock( nextReadPos, bufferBytes, &buffer1,
&bufferSize1, &buffer2, &bufferSize2, 0 );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
if ( duplexPrerollBytes <= 0 ) {
// Copy our buffer into the DS buffer
CopyMemory( buffer, buffer1, bufferSize1 );
if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
}
else {
memset( buffer, 0, bufferSize1 );
if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
duplexPrerollBytes -= bufferSize1 + bufferSize2;
}
// Update our buffer offset and unlock sound buffer
nextReadPos = ( nextReadPos + bufferSize1 + bufferSize2 ) % dsBufferSize;
dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
if ( FAILED( result ) ) {
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
errorText_ = errorStream_.str();
error( RtError::SYSTEM_ERROR );
}
handle->bufferPointer[1] = nextReadPos;
// No byte swapping necessary in DirectSound implementation.
// If necessary, convert 8-bit data from unsigned to signed.
if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
// Do buffer conversion if necessary.
if ( stream_.doConvertBuffer[1] )
convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
}
#ifdef GENERATE_DEBUG_LOG
if ( currentDebugLogEntry < debugLog.size() )
{
TTickRecord &r = debugLog[currentDebugLogEntry++];
r.currentReadPointer = currentReadPos;
r.safeReadPointer = safeReadPos;
r.currentWritePointer = currentWritePos;
r.safeWritePointer = safeWritePos;
r.readTime = readTime;
r.writeTime = writeTime;
r.nextReadPointer = handles[1].bufferPointer;
r.nextWritePointer = handles[0].bufferPointer;
}
#endif
unlock:
MUTEX_UNLOCK( &stream_.mutex );
RtApi::tickStreamTime();
}
// Definitions for utility functions and callbacks
// specific to the DirectSound implementation.
extern "C" unsigned __stdcall callbackHandler( void *ptr )
{
CallbackInfo *info = (CallbackInfo *) ptr;
RtApiDs *object = (RtApiDs *) info->object;
bool* isRunning = &info->isRunning;
while ( *isRunning == true ) {
object->callbackEvent();
}
_endthreadex( 0 );
return 0;
}
#include "tchar.h"
std::string convertTChar( LPCTSTR name )
{
std::string s;
#if defined( UNICODE ) || defined( _UNICODE )
// Yes, this conversion doesn't make sense for two-byte characters
// but RtAudio is currently written to return an std::string of
// one-byte chars for the device name.
for ( unsigned int i=0; i<wcslen( name ); i++ )
s.push_back( name[i] );
#else
s.append( std::string( name ) );
#endif
return s;
}
static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
LPCTSTR description,
LPCTSTR module,
LPVOID lpContext )
{
EnumInfo *info = (EnumInfo *) lpContext;
HRESULT hr;
if ( info->isInput == true ) {
DSCCAPS caps;
LPDIRECTSOUNDCAPTURE object;
hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
if ( hr != DS_OK ) return TRUE;
caps.dwSize = sizeof(caps);
hr = object->GetCaps( &caps );
if ( hr == DS_OK ) {
if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
info->counter++;
}
object->Release();
}
else {
DSCAPS caps;
LPDIRECTSOUND object;
hr = DirectSoundCreate( lpguid, &object, NULL );
if ( hr != DS_OK ) return TRUE;
caps.dwSize = sizeof(caps);
hr = object->GetCaps( &caps );
if ( hr == DS_OK ) {
if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
info->counter++;
}
object->Release();
}
if ( info->getDefault && lpguid == NULL ) return FALSE;
if ( info->findIndex && info->counter > info->index ) {
info->id = lpguid;
info->name = convertTChar( description );
return FALSE;
}
return TRUE;
}
static char* getErrorString( int code )
{
switch ( code ) {
case DSERR_ALLOCATED:
return "Already allocated";
case DSERR_CONTROLUNAVAIL:
return "Control unavailable";
case DSERR_INVALIDPARAM:
return "Invalid parameter";
case DSERR_INVALIDCALL:
return "Invalid call";
case DSERR_GENERIC:
return "Generic error";
case DSERR_PRIOLEVELNEEDED:
return "Priority level needed";
case DSERR_OUTOFMEMORY:
return "Out of memory";
case DSERR_BADFORMAT:
return "The sample rate or the channel format is not supported";
case DSERR_UNSUPPORTED:
return "Not supported";
case DSERR_NODRIVER:
return "No driver";
case DSERR_ALREADYINITIALIZED:
return "Already initialized";
case DSERR_NOAGGREGATION:
return "No aggregation";
case DSERR_BUFFERLOST:
return "Buffer lost";
case DSERR_OTHERAPPHASPRIO:
return "Another application already has priority";
case DSERR_UNINITIALIZED:
return "Uninitialized";
default:
return "DirectSound unknown error";
}
}
//******************** End of __WINDOWS_DS__ *********************//
#endif
#if defined(__LINUX_ALSA__)
#include <alsa/asoundlib.h>
#include <unistd.h>
// A structure to hold various information related to the ALSA API
// implementation.
struct AlsaHandle {
snd_pcm_t *handles[2];
bool synchronized;
bool xrun[2];
pthread_cond_t runnable;
AlsaHandle()
:synchronized(false) { xrun[0] = false; xrun[1] = false; }
};
extern "C" void *alsaCallbackHandler( void * ptr );
RtApiAlsa :: RtApiAlsa()
{
// Nothing to do here.
}
RtApiAlsa :: ~RtApiAlsa()
{
if ( stream_.state != STREAM_CLOSED ) closeStream();
}
unsigned int RtApiAlsa :: getDeviceCount( void )
{
unsigned nDevices = 0;
int result, subdevice, card;
char name[64];
snd_ctl_t *handle;
// Count cards and devices
card = -1;
snd_card_next( &card );
while ( card >= 0 ) {
sprintf( name, "hw:%d", card );
result = snd_ctl_open( &handle, name, 0 );
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
error( RtError::WARNING );
goto nextcard;
}
subdevice = -1;
while( 1 ) {
result = snd_ctl_pcm_next_device( handle, &subdevice );
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
error( RtError::WARNING );
break;
}
if ( subdevice < 0 )
break;
nDevices++;
}
nextcard:
snd_ctl_close( handle );
snd_card_next( &card );
}
return nDevices;
}
RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
{
RtAudio::DeviceInfo info;
info.probed = false;
unsigned nDevices = 0;
int result, subdevice, card;
char name[64];
snd_ctl_t *chandle;
// Count cards and devices
card = -1;
snd_card_next( &card );
while ( card >= 0 ) {
sprintf( name, "hw:%d", card );
result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
error( RtError::WARNING );
goto nextcard;
}
subdevice = -1;
while( 1 ) {
result = snd_ctl_pcm_next_device( chandle, &subdevice );
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
error( RtError::WARNING );
break;
}
if ( subdevice < 0 ) break;
if ( nDevices == device ) {
sprintf( name, "hw:%d,%d", card, subdevice );
goto foundDevice;
}
nDevices++;
}
nextcard:
snd_ctl_close( chandle );
snd_card_next( &card );
}
if ( nDevices == 0 ) {
errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
error( RtError::INVALID_USE );
}
if ( device >= nDevices ) {
errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
error( RtError::INVALID_USE );
}
foundDevice:
// If a stream is already open, we cannot probe the stream devices.
// Thus, use the saved results.
if ( stream_.state != STREAM_CLOSED &&
( stream_.device[0] == device || stream_.device[1] == device ) ) {
if ( device >= devices_.size() ) {
errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
error( RtError::WARNING );
return info;
}
return devices_[ device ];
}
int openMode = SND_PCM_ASYNC;
snd_pcm_stream_t stream;
snd_pcm_info_t *pcminfo;
snd_pcm_info_alloca( &pcminfo );
snd_pcm_t *phandle;
snd_pcm_hw_params_t *params;
snd_pcm_hw_params_alloca( &params );
// First try for playback
stream = SND_PCM_STREAM_PLAYBACK;
snd_pcm_info_set_device( pcminfo, subdevice );
snd_pcm_info_set_subdevice( pcminfo, 0 );
snd_pcm_info_set_stream( pcminfo, stream );
result = snd_ctl_pcm_info( chandle, pcminfo );
if ( result < 0 ) {
// Device probably doesn't support playback.
goto captureProbe;
}
result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
error( RtError::WARNING );
goto captureProbe;
}
// The device is open ... fill the parameter structure.
result = snd_pcm_hw_params_any( phandle, params );
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
error( RtError::WARNING );
goto captureProbe;
}
// Get output channel information.
unsigned int value;
result = snd_pcm_hw_params_get_channels_max( params, &value );
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
error( RtError::WARNING );
goto captureProbe;
}
info.outputChannels = value;
snd_pcm_close( phandle );
captureProbe:
// Now try for capture
stream = SND_PCM_STREAM_CAPTURE;
snd_pcm_info_set_stream( pcminfo, stream );
result = snd_ctl_pcm_info( chandle, pcminfo );
snd_ctl_close( chandle );
if ( result < 0 ) {
// Device probably doesn't support capture.
if ( info.outputChannels == 0 ) return info;
goto probeParameters;
}
result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
error( RtError::WARNING );
if ( info.outputChannels == 0 ) return info;
goto probeParameters;
}
// The device is open ... fill the parameter structure.
result = snd_pcm_hw_params_any( phandle, params );
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
error( RtError::WARNING );
if ( info.outputChannels == 0 ) return info;
goto probeParameters;
}
result = snd_pcm_hw_params_get_channels_max( params, &value );
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
error( RtError::WARNING );
if ( info.outputChannels == 0 ) return info;
goto probeParameters;
}
info.inputChannels = value;
snd_pcm_close( phandle );
// If device opens for both playback and capture, we determine the channels.
if ( info.outputChannels > 0 && info.inputChannels > 0 )
info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
// ALSA doesn't provide default devices so we'll use the first available one.
if ( device == 0 && info.outputChannels > 0 )
info.isDefaultOutput = true;
if ( device == 0 && info.inputChannels > 0 )
info.isDefaultInput = true;
probeParameters:
// 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.outputChannels >= info.inputChannels )
stream = SND_PCM_STREAM_PLAYBACK;
else
stream = SND_PCM_STREAM_CAPTURE;
snd_pcm_info_set_stream( pcminfo, stream );
result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
// The device is open ... fill the parameter structure.
result = snd_pcm_hw_params_any( phandle, params );
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
// Test our discrete set of sample rate values.
info.sampleRates.clear();
for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 )
info.sampleRates.push_back( SAMPLE_RATES[i] );
}
if ( info.sampleRates.size() == 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
// 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( phandle, params, format ) == 0 )
info.nativeFormats |= RTAUDIO_SINT8;
format = SND_PCM_FORMAT_S16;
if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
info.nativeFormats |= RTAUDIO_SINT16;
format = SND_PCM_FORMAT_S24;
if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
info.nativeFormats |= RTAUDIO_SINT24;
format = SND_PCM_FORMAT_S32;
if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
info.nativeFormats |= RTAUDIO_SINT32;
format = SND_PCM_FORMAT_FLOAT;
if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
info.nativeFormats |= RTAUDIO_FLOAT32;
format = SND_PCM_FORMAT_FLOAT64;
if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
info.nativeFormats |= RTAUDIO_FLOAT64;
// Check that we have at least one supported format
if ( info.nativeFormats == 0 ) {
errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
// Get the device name
char *cardname;
result = snd_card_get_name( card, &cardname );
if ( result >= 0 )
sprintf( name, "hw:%s,%d", cardname, subdevice );
info.name = name;
// That's all ... close the device and return
snd_pcm_close( phandle );
info.probed = true;
return info;
}
void RtApiAlsa :: saveDeviceInfo( void )
{
devices_.clear();
unsigned int nDevices = getDeviceCount();
devices_.resize( nDevices );
for ( unsigned int i=0; i<nDevices; i++ )
devices_[i] = getDeviceInfo( i );
}
bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options )
{
#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.
unsigned nDevices = 0;
int result, subdevice, card;
char name[64];
snd_ctl_t *chandle;
// Count cards and devices
card = -1;
snd_card_next( &card );
while ( card >= 0 ) {
sprintf( name, "hw:%d", card );
result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
}
subdevice = -1;
while( 1 ) {
result = snd_ctl_pcm_next_device( chandle, &subdevice );
if ( result < 0 ) break;
if ( subdevice < 0 ) break;
if ( nDevices == device ) {
sprintf( name, "hw:%d,%d", card, subdevice );
snd_ctl_close( chandle );
goto foundDevice;
}
nDevices++;
}
snd_ctl_close( chandle );
snd_card_next( &card );
}
if ( nDevices == 0 ) {
// This should not happen because a check is made before this function is called.
errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
return FAILURE;
}
if ( device >= nDevices ) {
// This should not happen because a check is made before this function is called.
errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
return FAILURE;
}
foundDevice:
// The getDeviceInfo() function will not work for a device that is
// already open. Thus, we'll probe the system before opening a
// stream and save the results for use by getDeviceInfo().
if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
this->saveDeviceInfo();
snd_pcm_stream_t stream;
if ( mode == OUTPUT )
stream = SND_PCM_STREAM_PLAYBACK;
else
stream = SND_PCM_STREAM_CAPTURE;
snd_pcm_t *phandle;
int openMode = SND_PCM_ASYNC;
result = snd_pcm_open( &phandle, name, stream, openMode );
if ( result < 0 ) {
if ( mode == OUTPUT )
errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
else
errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
errorText_ = errorStream_.str();
return FAILURE;
}
// Fill the parameter structure.
snd_pcm_hw_params_t *hw_params;
snd_pcm_hw_params_alloca( &hw_params );
result = snd_pcm_hw_params_any( phandle, hw_params );
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
}
#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 ... check user preference.
if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
stream_.userInterleaved = false;
result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
if ( result < 0 ) {
result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
stream_.deviceInterleaved[mode] = true;
}
else
stream_.deviceInterleaved[mode] = false;
}
else {
stream_.userInterleaved = true;
result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
if ( result < 0 ) {
result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
stream_.deviceInterleaved[mode] = false;
}
else
stream_.deviceInterleaved[mode] = true;
}
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
}
// Determine how to set the device format.
stream_.userFormat = format;
snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
if ( format == RTAUDIO_SINT8 )
deviceFormat = SND_PCM_FORMAT_S8;
else if ( format == RTAUDIO_SINT16 )
deviceFormat = SND_PCM_FORMAT_S16;
else if ( format == RTAUDIO_SINT24 )
deviceFormat = SND_PCM_FORMAT_S24;
else if ( format == RTAUDIO_SINT32 )
deviceFormat = SND_PCM_FORMAT_S32;
else if ( format == RTAUDIO_FLOAT32 )
deviceFormat = SND_PCM_FORMAT_FLOAT;
else if ( format == RTAUDIO_FLOAT64 )
deviceFormat = SND_PCM_FORMAT_FLOAT64;
if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
stream_.deviceFormat[mode] = format;
goto setFormat;
}
// The user requested format is not natively supported by the device.
deviceFormat = SND_PCM_FORMAT_FLOAT64;
if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
goto setFormat;
}
deviceFormat = SND_PCM_FORMAT_FLOAT;
if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
goto setFormat;
}
deviceFormat = SND_PCM_FORMAT_S32;
if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
stream_.deviceFormat[mode] = RTAUDIO_SINT32;
goto setFormat;
}
deviceFormat = SND_PCM_FORMAT_S24;
if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
stream_.deviceFormat[mode] = RTAUDIO_SINT24;
goto setFormat;
}
deviceFormat = SND_PCM_FORMAT_S16;
if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
stream_.deviceFormat[mode] = RTAUDIO_SINT16;
goto setFormat;
}
deviceFormat = SND_PCM_FORMAT_S8;
if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
stream_.deviceFormat[mode] = RTAUDIO_SINT8;
goto setFormat;
}
// If we get here, no supported format was found.
errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
errorText_ = errorStream_.str();
return FAILURE;
setFormat:
result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
}
// Determine whether byte-swaping is necessary.
stream_.doByteSwap[mode] = false;
if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
result = snd_pcm_format_cpu_endian( deviceFormat );
if ( result == 0 )
stream_.doByteSwap[mode] = true;
else if (result < 0) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
}
}
// Set the sample rate.
result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
}
// 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;
result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
unsigned int deviceChannels = value;
if ( result < 0 || deviceChannels < channels + firstChannel ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
}
result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
}
deviceChannels = value;
if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
stream_.nDeviceChannels[mode] = deviceChannels;
// Set the device channels.
result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
}
// Set the buffer number, which in ALSA is referred to as the "period".
int totalSize, dir = 0;
unsigned int periods = 0;
if ( options ) periods = options->numberOfBuffers;
totalSize = *bufferSize * periods;
// Set the buffer (or period) size.
snd_pcm_uframes_t periodSize = *bufferSize;
result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
}
*bufferSize = periodSize;
if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
else periods = totalSize / *bufferSize;
// Even though the hardware might allow 1 buffer, it won't work reliably.
if ( periods < 2 ) periods = 2;
result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
}
// 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 ) {
errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
stream_.bufferSize = *bufferSize;
// Install the hardware configuration
result = snd_pcm_hw_params( phandle, hw_params );
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
}
#if defined(__RTAUDIO_DEBUG__)
fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
snd_pcm_hw_params_dump( hw_params, out );
#endif
// Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
snd_pcm_sw_params_t *sw_params = NULL;
snd_pcm_sw_params_alloca( &sw_params );
snd_pcm_sw_params_current( phandle, sw_params );
snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
// The following two settings were suggested by Theo Veenker
//snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
//snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
// here are two options for a fix
//snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
snd_pcm_uframes_t val;
snd_pcm_sw_params_get_boundary( sw_params, &val );
snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
result = snd_pcm_sw_params( phandle, sw_params );
if ( result < 0 ) {
snd_pcm_close( phandle );
errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
return FAILURE;
}
#if defined(__RTAUDIO_DEBUG__)
fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
snd_pcm_sw_params_dump( sw_params, out );
#endif
// 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_.userInterleaved != stream_.deviceInterleaved[mode] &&
stream_.nUserChannels[mode] > 1 )
stream_.doConvertBuffer[mode] = true;
// Allocate the ApiHandle if necessary and then save.
AlsaHandle *apiInfo = 0;
if ( stream_.apiHandle == 0 ) {
try {
apiInfo = (AlsaHandle *) new AlsaHandle;
}
catch ( std::bad_alloc& ) {
errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
goto error;
}
if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) {
errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
goto error;
}
stream_.apiHandle = (void *) apiInfo;
apiInfo->handles[0] = 0;
apiInfo->handles[1] = 0;
}
else {
apiInfo = (AlsaHandle *) stream_.apiHandle;
}
apiInfo->handles[mode] = phandle;
// Allocate necessary internal buffers.
unsigned long bufferBytes;
bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
if ( stream_.userBuffer[mode] == NULL ) {
errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
goto error;
}
if ( stream_.doConvertBuffer[mode] ) {
bool makeBuffer = true;
bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
if ( mode == INPUT ) {
if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
if ( bufferBytes <= bytesOut ) makeBuffer = false;
}
}
if ( makeBuffer ) {
bufferBytes *= *bufferSize;
if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
if ( stream_.deviceBuffer == NULL ) {
errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
goto error;
}
}
}
stream_.sampleRate = sampleRate;
stream_.nBuffers = periods;
stream_.device[mode] = device;
stream_.state = STREAM_STOPPED;
// Setup the buffer conversion information structure.
if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
// Setup thread if necessary.
if ( stream_.mode == OUTPUT && mode == INPUT ) {
// We had already set up an output stream.
stream_.mode = DUPLEX;
// Link the streams if possible.
apiInfo->synchronized = false;
if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
apiInfo->synchronized = true;
else {
errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
error( RtError::WARNING );
}
}
else {
stream_.mode = mode;
// Setup callback thread.
stream_.callbackInfo.object = (void *) this;
// Set the thread attributes for joinable and realtime scheduling
// priority (optional). The higher priority will only take affect
// if the program is run as root or suid. Note, under Linux
// processes with CAP_SYS_NICE privilege, a user can change
// scheduling policy and priority (thus need not be root). See
// POSIX "capabilities".
pthread_attr_t attr;
pthread_attr_init( &attr );
pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
struct sched_param param;
int priority = options->priority;
int min = sched_get_priority_min( SCHED_RR );
int max = sched_get_priority_max( SCHED_RR );
if ( priority < min ) priority = min;
else if ( priority > max ) priority = max;
param.sched_priority = priority;
pthread_attr_setschedparam( &attr, &param );
pthread_attr_setschedpolicy( &attr, SCHED_RR );
}
else
pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
#else
pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
#endif
stream_.callbackInfo.isRunning = true;
result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
pthread_attr_destroy( &attr );
if ( result ) {
stream_.callbackInfo.isRunning = false;
errorText_ = "RtApiAlsa::error creating callback thread!";
goto error;
}
}
return SUCCESS;
error:
if ( apiInfo ) {
pthread_cond_destroy( &apiInfo->runnable );
if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
delete apiInfo;
stream_.apiHandle = 0;
}
for ( int i=0; i<2; i++ ) {
if ( stream_.userBuffer[i] ) {
free( stream_.userBuffer[i] );
stream_.userBuffer[i] = 0;
}
}
if ( stream_.deviceBuffer ) {
free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
return FAILURE;
}
void RtApiAlsa :: closeStream()
{
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
error( RtError::WARNING );
return;
}
AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
stream_.callbackInfo.isRunning = false;
MUTEX_LOCK( &stream_.mutex );
if ( stream_.state == STREAM_STOPPED )
pthread_cond_signal( &apiInfo->runnable );
MUTEX_UNLOCK( &stream_.mutex );
pthread_join( stream_.callbackInfo.thread, NULL );
if ( stream_.state == STREAM_RUNNING ) {
stream_.state = STREAM_STOPPED;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
snd_pcm_drop( apiInfo->handles[0] );
if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
snd_pcm_drop( apiInfo->handles[1] );
}
if ( apiInfo ) {
pthread_cond_destroy( &apiInfo->runnable );
if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
delete apiInfo;
stream_.apiHandle = 0;
}
for ( int i=0; i<2; i++ ) {
if ( stream_.userBuffer[i] ) {
free( stream_.userBuffer[i] );
stream_.userBuffer[i] = 0;
}
}
if ( stream_.deviceBuffer ) {
free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
stream_.mode = UNINITIALIZED;
stream_.state = STREAM_CLOSED;
}
void RtApiAlsa :: startStream()
{
// This method calls snd_pcm_prepare if the device isn't already in that state.
verifyStream();
if ( stream_.state == STREAM_RUNNING ) {
errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
error( RtError::WARNING );
return;
}
MUTEX_LOCK( &stream_.mutex );
int result = 0;
snd_pcm_state_t state;
AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
state = snd_pcm_state( handle[0] );
if ( state != SND_PCM_STATE_PREPARED ) {
result = snd_pcm_prepare( handle[0] );
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
goto unlock;
}
}
}
if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
state = snd_pcm_state( handle[1] );
if ( state != SND_PCM_STATE_PREPARED ) {
result = snd_pcm_prepare( handle[1] );
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
goto unlock;
}
}
}
stream_.state = STREAM_RUNNING;
unlock:
MUTEX_UNLOCK( &stream_.mutex );
pthread_cond_signal( &apiInfo->runnable );
if ( result >= 0 ) return;
error( RtError::SYSTEM_ERROR );
}
void RtApiAlsa :: stopStream()
{
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
error( RtError::WARNING );
return;
}
MUTEX_LOCK( &stream_.mutex );
if ( stream_.state == STREAM_STOPPED ) {
MUTEX_UNLOCK( &stream_.mutex );
return;
}
int result = 0;
AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
if ( apiInfo->synchronized )
result = snd_pcm_drop( handle[0] );
else
result = snd_pcm_drain( handle[0] );
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
goto unlock;
}
}
if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
result = snd_pcm_drop( handle[1] );
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
goto unlock;
}
}
unlock:
stream_.state = STREAM_STOPPED;
MUTEX_UNLOCK( &stream_.mutex );
if ( result >= 0 ) return;
error( RtError::SYSTEM_ERROR );
}
void RtApiAlsa :: abortStream()
{
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
error( RtError::WARNING );
return;
}
MUTEX_LOCK( &stream_.mutex );
if ( stream_.state == STREAM_STOPPED ) {
MUTEX_UNLOCK( &stream_.mutex );
return;
}
int result = 0;
AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
result = snd_pcm_drop( handle[0] );
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
goto unlock;
}
}
if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
result = snd_pcm_drop( handle[1] );
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
goto unlock;
}
}
unlock:
stream_.state = STREAM_STOPPED;
MUTEX_UNLOCK( &stream_.mutex );
if ( result >= 0 ) return;
error( RtError::SYSTEM_ERROR );
}
void RtApiAlsa :: callbackEvent()
{
AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
if ( stream_.state == STREAM_STOPPED ) {
MUTEX_LOCK( &stream_.mutex );
pthread_cond_wait( &apiInfo->runnable, &stream_.mutex );
if ( stream_.state != STREAM_RUNNING ) {
MUTEX_UNLOCK( &stream_.mutex );
return;
}
MUTEX_UNLOCK( &stream_.mutex );
}
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
error( RtError::WARNING );
return;
}
int doStopStream = 0;
RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
double streamTime = getStreamTime();
RtAudioStreamStatus status = 0;
if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
status |= RTAUDIO_OUTPUT_UNDERFLOW;
apiInfo->xrun[0] = false;
}
if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
status |= RTAUDIO_INPUT_OVERFLOW;
apiInfo->xrun[1] = false;
}
doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
if ( doStopStream == 2 ) {
abortStream();
return;
}
MUTEX_LOCK( &stream_.mutex );
// The state might change while waiting on a mutex.
if ( stream_.state == STREAM_STOPPED ) goto unlock;
int result;
char *buffer;
int channels;
snd_pcm_t **handle;
snd_pcm_sframes_t frames;
RtAudioFormat format;
handle = (snd_pcm_t **) apiInfo->handles;
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[1];
channels = stream_.nUserChannels[1];
format = stream_.userFormat;
}
// Read samples from device in interleaved/non-interleaved format.
if ( stream_.deviceInterleaved[1] )
result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
else {
void *bufs[channels];
size_t offset = stream_.bufferSize * formatBytes( format );
for ( int i=0; i<channels; i++ )
bufs[i] = (void *) (buffer + (i * offset));
result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
}
if ( result < (int) stream_.bufferSize ) {
// Either an error or overrun occured.
if ( result == -EPIPE ) {
snd_pcm_state_t state = snd_pcm_state( handle[1] );
if ( state == SND_PCM_STATE_XRUN ) {
apiInfo->xrun[1] = true;
result = snd_pcm_prepare( handle[1] );
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
}
}
else {
errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
}
}
else {
errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
}
error( RtError::WARNING );
goto tryOutput;
}
// Do byte swapping if necessary.
if ( stream_.doByteSwap[1] )
byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
// Do buffer conversion if necessary.
if ( stream_.doConvertBuffer[1] )
convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
// Check stream latency
result = snd_pcm_delay( handle[1], &frames );
if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
}
tryOutput:
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
// Setup parameters and do buffer conversion if necessary.
if ( stream_.doConvertBuffer[0] ) {
buffer = stream_.deviceBuffer;
convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
channels = stream_.nDeviceChannels[0];
format = stream_.deviceFormat[0];
}
else {
buffer = stream_.userBuffer[0];
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_.deviceInterleaved[0] )
result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
else {
void *bufs[channels];
size_t offset = stream_.bufferSize * formatBytes( format );
for ( int i=0; i<channels; i++ )
bufs[i] = (void *) (buffer + (i * offset));
result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
}
if ( result < (int) stream_.bufferSize ) {
// Either an error or underrun occured.
if ( result == -EPIPE ) {
snd_pcm_state_t state = snd_pcm_state( handle[0] );
if ( state == SND_PCM_STATE_XRUN ) {
apiInfo->xrun[0] = true;
result = snd_pcm_prepare( handle[0] );
if ( result < 0 ) {
errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
}
}
else {
errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
}
}
else {
errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
errorText_ = errorStream_.str();
}
error( RtError::WARNING );
goto unlock;
}
// Check stream latency
result = snd_pcm_delay( handle[0], &frames );
if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
}
unlock:
MUTEX_UNLOCK( &stream_.mutex );
RtApi::tickStreamTime();
if ( doStopStream == 1 ) this->stopStream();
}
extern "C" void *alsaCallbackHandler( void *ptr )
{
CallbackInfo *info = (CallbackInfo *) ptr;
RtApiAlsa *object = (RtApiAlsa *) info->object;
bool *isRunning = &info->isRunning;
while ( *isRunning == true ) {
pthread_testcancel();
object->callbackEvent();
}
pthread_exit( NULL );
}
//******************** End of __LINUX_ALSA__ *********************//
#endif
#if defined(__LINUX_OSS__)
#include <unistd.h>
#include <sys/ioctl.h>
#include <unistd.h>
#include <fcntl.h>
#include "soundcard.h"
#include <errno.h>
#include <math.h>
extern "C" void *ossCallbackHandler(void * ptr);
// A structure to hold various information related to the OSS API
// implementation.
struct OssHandle {
int id[2]; // device ids
bool xrun[2];
bool triggered;
pthread_cond_t runnable;
OssHandle()
:triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
};
RtApiOss :: RtApiOss()
{
// Nothing to do here.
}
RtApiOss :: ~RtApiOss()
{
if ( stream_.state != STREAM_CLOSED ) closeStream();
}
unsigned int RtApiOss :: getDeviceCount( void )
{
int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
if ( mixerfd == -1 ) {
errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
error( RtError::WARNING );
return 0;
}
oss_sysinfo sysinfo;
if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
close( mixerfd );
errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
error( RtError::WARNING );
return 0;
}
close( mixerfd );
return sysinfo.numaudios;
}
RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
{
RtAudio::DeviceInfo info;
info.probed = false;
int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
if ( mixerfd == -1 ) {
errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
error( RtError::WARNING );
return info;
}
oss_sysinfo sysinfo;
int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
if ( result == -1 ) {
close( mixerfd );
errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
error( RtError::WARNING );
return info;
}
unsigned nDevices = sysinfo.numaudios;
if ( nDevices == 0 ) {
close( mixerfd );
errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
error( RtError::INVALID_USE );
}
if ( device >= nDevices ) {
close( mixerfd );
errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
error( RtError::INVALID_USE );
}
oss_audioinfo ainfo;
ainfo.dev = device;
result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
close( mixerfd );
if ( result == -1 ) {
errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
// Probe channels
if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
if ( ainfo.caps & PCM_CAP_DUPLEX ) {
if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
}
// Probe data formats ... do for input
unsigned long mask = ainfo.iformats;
if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
info.nativeFormats |= RTAUDIO_SINT16;
if ( mask & AFMT_S8 )
info.nativeFormats |= RTAUDIO_SINT8;
if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
info.nativeFormats |= RTAUDIO_SINT32;
if ( mask & AFMT_FLOAT )
info.nativeFormats |= RTAUDIO_FLOAT32;
if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
info.nativeFormats |= RTAUDIO_SINT24;
// Check that we have at least one supported format
if ( info.nativeFormats == 0 ) {
errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
errorText_ = errorStream_.str();
error( RtError::WARNING );
return info;
}
// Probe the supported sample rates.
info.sampleRates.clear();
if ( ainfo.nrates ) {
for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
info.sampleRates.push_back( SAMPLE_RATES[k] );
break;
}
}
}
}
else {
// Check min and max rate values;
for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] )
info.sampleRates.push_back( SAMPLE_RATES[k] );
}
}
if ( info.sampleRates.size() == 0 ) {
errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
errorText_ = errorStream_.str();
error( RtError::WARNING );
}
else {
info.probed = true;
info.name = ainfo.name;
}
return info;
}
bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
unsigned int firstChannel, unsigned int sampleRate,
RtAudioFormat format, unsigned int *bufferSize,
RtAudio::StreamOptions *options )
{
int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
if ( mixerfd == -1 ) {
errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
return FAILURE;
}
oss_sysinfo sysinfo;
int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
if ( result == -1 ) {
close( mixerfd );
errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
return FAILURE;
}
unsigned nDevices = sysinfo.numaudios;
if ( nDevices == 0 ) {
// This should not happen because a check is made before this function is called.
close( mixerfd );
errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
return FAILURE;
}
if ( device >= nDevices ) {
// This should not happen because a check is made before this function is called.
close( mixerfd );
errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
return FAILURE;
}
oss_audioinfo ainfo;
ainfo.dev = device;
result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
close( mixerfd );
if ( result == -1 ) {
errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
errorText_ = errorStream_.str();
return FAILURE;
}
// Check if device supports input or output
if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
if ( mode == OUTPUT )
errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
else
errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
errorText_ = errorStream_.str();
return FAILURE;
}
int flags = 0;
OssHandle *handle = (OssHandle *) stream_.apiHandle;
if ( mode == OUTPUT )
flags |= O_WRONLY;
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->id[0] );
handle->id[0] = 0;
if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
errorText_ = errorStream_.str();
return FAILURE;
}
// Check that the number previously set channels is the same.
if ( stream_.nUserChannels[0] != channels ) {
errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
flags |= O_RDWR;
}
else
flags |= O_RDONLY;
}
// Set exclusive access if specified.
if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
// Try to open the device.
int fd;
fd = open( ainfo.devnode, flags, 0 );
if ( fd == -1 ) {
if ( errno == EBUSY )
errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
else
errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
// For duplex operation, specifically set this mode (this doesn't seem to work).
/*
if ( flags | O_RDWR ) {
result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
if ( result == -1) {
errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
}
*/
// Check the device channel support.
stream_.nUserChannels[mode] = channels;
if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
close( fd );
errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
errorText_ = errorStream_.str();
return FAILURE;
}
// Set the number of channels.
int deviceChannels = channels + firstChannel;
result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
close( fd );
errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
stream_.nDeviceChannels[mode] = deviceChannels;
// Get the data format mask
int mask;
result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
if ( result == -1 ) {
close( fd );
errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
errorText_ = errorStream_.str();
return FAILURE;
}
// Determine how to set the device format.
stream_.userFormat = format;
int deviceFormat = -1;
stream_.doByteSwap[mode] = false;
if ( format == RTAUDIO_SINT8 ) {
if ( mask & AFMT_S8 ) {
deviceFormat = AFMT_S8;
stream_.deviceFormat[mode] = RTAUDIO_SINT8;
}
}
else if ( format == RTAUDIO_SINT16 ) {
if ( mask & AFMT_S16_NE ) {
deviceFormat = AFMT_S16_NE;
stream_.deviceFormat[mode] = RTAUDIO_SINT16;
}
else if ( mask & AFMT_S16_OE ) {
deviceFormat = AFMT_S16_OE;
stream_.deviceFormat[mode] = RTAUDIO_SINT16;
stream_.doByteSwap[mode] = true;
}
}
else if ( format == RTAUDIO_SINT24 ) {
if ( mask & AFMT_S24_NE ) {
deviceFormat = AFMT_S24_NE;
stream_.deviceFormat[mode] = RTAUDIO_SINT24;
}
else if ( mask & AFMT_S24_OE ) {
deviceFormat = AFMT_S24_OE;
stream_.deviceFormat[mode] = RTAUDIO_SINT24;
stream_.doByteSwap[mode] = true;
}
}
else if ( format == RTAUDIO_SINT32 ) {
if ( mask & AFMT_S32_NE ) {
deviceFormat = AFMT_S32_NE;
stream_.deviceFormat[mode] = RTAUDIO_SINT32;
}
else if ( mask & AFMT_S32_OE ) {
deviceFormat = AFMT_S32_OE;
stream_.deviceFormat[mode] = RTAUDIO_SINT32;
stream_.doByteSwap[mode] = true;
}
}
if ( deviceFormat == -1 ) {
// The user requested format is not natively supported by the device.
if ( mask & AFMT_S16_NE ) {
deviceFormat = AFMT_S16_NE;
stream_.deviceFormat[mode] = RTAUDIO_SINT16;
}
else if ( mask & AFMT_S32_NE ) {
deviceFormat = AFMT_S32_NE;
stream_.deviceFormat[mode] = RTAUDIO_SINT32;
}
else if ( mask & AFMT_S24_NE ) {
deviceFormat = AFMT_S24_NE;
stream_.deviceFormat[mode] = RTAUDIO_SINT24;
}
else if ( mask & AFMT_S16_OE ) {
deviceFormat = AFMT_S16_OE;
stream_.deviceFormat[mode] = RTAUDIO_SINT16;
stream_.doByteSwap[mode] = true;
}
else if ( mask & AFMT_S32_OE ) {
deviceFormat = AFMT_S32_OE;
stream_.deviceFormat[mode] = RTAUDIO_SINT32;
stream_.doByteSwap[mode] = true;
}
else if ( mask & AFMT_S24_OE ) {
deviceFormat = AFMT_S24_OE;
stream_.deviceFormat[mode] = RTAUDIO_SINT24;
stream_.doByteSwap[mode] = true;
}
else if ( mask & AFMT_S8) {
deviceFormat = AFMT_S8;
stream_.deviceFormat[mode] = RTAUDIO_SINT8;
}
}
if ( stream_.deviceFormat[mode] == 0 ) {
// This really shouldn't happen ...
close( fd );
errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
errorText_ = errorStream_.str();
return FAILURE;
}
// Set the data format.
int temp = deviceFormat;
result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
if ( result == -1 || deviceFormat != temp ) {
close( fd );
errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
// 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.
int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
int buffers = 0;
if ( options ) buffers = options->numberOfBuffers;
if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
if ( buffers < 2 ) buffers = 3;
temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
if ( result == -1 ) {
close( fd );
errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
stream_.nBuffers = buffers;
// Save buffer size (in sample frames).
*bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
stream_.bufferSize = *bufferSize;
// Set the sample rate.
int srate = sampleRate;
result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
if ( result == -1 ) {
close( fd );
errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
// Verify the sample rate setup worked.
if ( abs( srate - sampleRate ) > 100 ) {
close( fd );
errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
errorText_ = errorStream_.str();
return FAILURE;
}
stream_.sampleRate = sampleRate;
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] = deviceChannels;
}
// Set interleaving parameters.
stream_.userInterleaved = true;
stream_.deviceInterleaved[mode] = true;
if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
stream_.userInterleaved = false;
// 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_.userInterleaved != stream_.deviceInterleaved[mode] &&
stream_.nUserChannels[mode] > 1 )
stream_.doConvertBuffer[mode] = true;
// Allocate the stream handles if necessary and then save.
if ( stream_.apiHandle == 0 ) {
try {
handle = new OssHandle;
}
catch ( std::bad_alloc& ) {
errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
goto error;
}
if ( pthread_cond_init( &handle->runnable, NULL ) ) {
errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
goto error;
}
stream_.apiHandle = (void *) handle;
}
else {
handle = (OssHandle *) stream_.apiHandle;
}
handle->id[mode] = fd;
// Allocate necessary internal buffers.
unsigned long bufferBytes;
bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
if ( stream_.userBuffer[mode] == NULL ) {
errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
goto error;
}
if ( stream_.doConvertBuffer[mode] ) {
bool makeBuffer = true;
bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
if ( mode == INPUT ) {
if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
if ( bufferBytes <= bytesOut ) makeBuffer = false;
}
}
if ( makeBuffer ) {
bufferBytes *= *bufferSize;
if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
if ( stream_.deviceBuffer == NULL ) {
errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
goto error;
}
}
}
stream_.device[mode] = device;
stream_.state = STREAM_STOPPED;
// Setup the buffer conversion information structure.
if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
// Setup thread if necessary.
if ( stream_.mode == OUTPUT && mode == INPUT ) {
// We had already set up an output stream.
stream_.mode = DUPLEX;
if ( stream_.device[0] == device ) handle->id[0] = fd;
}
else {
stream_.mode = mode;
// Setup callback thread.
stream_.callbackInfo.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 );
#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
struct sched_param param;
int priority = options->priority;
int min = sched_get_priority_min( SCHED_RR );
int max = sched_get_priority_max( SCHED_RR );
if ( priority < min ) priority = min;
else if ( priority > max ) priority = max;
param.sched_priority = priority;
pthread_attr_setschedparam( &attr, &param );
pthread_attr_setschedpolicy( &attr, SCHED_RR );
}
else
pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
#else
pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
#endif
stream_.callbackInfo.isRunning = true;
result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
pthread_attr_destroy( &attr );
if ( result ) {
stream_.callbackInfo.isRunning = false;
errorText_ = "RtApiOss::error creating callback thread!";
goto error;
}
}
return SUCCESS;
error:
if ( handle ) {
pthread_cond_destroy( &handle->runnable );
if ( handle->id[0] ) close( handle->id[0] );
if ( handle->id[1] ) close( handle->id[1] );
delete handle;
stream_.apiHandle = 0;
}
for ( int i=0; i<2; i++ ) {
if ( stream_.userBuffer[i] ) {
free( stream_.userBuffer[i] );
stream_.userBuffer[i] = 0;
}
}
if ( stream_.deviceBuffer ) {
free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
return FAILURE;
}
void RtApiOss :: closeStream()
{
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiOss::closeStream(): no open stream to close!";
error( RtError::WARNING );
return;
}
OssHandle *handle = (OssHandle *) stream_.apiHandle;
stream_.callbackInfo.isRunning = false;
MUTEX_LOCK( &stream_.mutex );
if ( stream_.state == STREAM_STOPPED )
pthread_cond_signal( &handle->runnable );
MUTEX_UNLOCK( &stream_.mutex );
pthread_join( stream_.callbackInfo.thread, NULL );
if ( stream_.state == STREAM_RUNNING ) {
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
else
ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
stream_.state = STREAM_STOPPED;
}
if ( handle ) {
pthread_cond_destroy( &handle->runnable );
if ( handle->id[0] ) close( handle->id[0] );
if ( handle->id[1] ) close( handle->id[1] );
delete handle;
stream_.apiHandle = 0;
}
for ( int i=0; i<2; i++ ) {
if ( stream_.userBuffer[i] ) {
free( stream_.userBuffer[i] );
stream_.userBuffer[i] = 0;
}
}
if ( stream_.deviceBuffer ) {
free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
stream_.mode = UNINITIALIZED;
stream_.state = STREAM_CLOSED;
}
void RtApiOss :: startStream()
{
verifyStream();
if ( stream_.state == STREAM_RUNNING ) {
errorText_ = "RtApiOss::startStream(): the stream is already running!";
error( RtError::WARNING );
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 );
OssHandle *handle = (OssHandle *) stream_.apiHandle;
pthread_cond_signal( &handle->runnable );
}
void RtApiOss :: stopStream()
{
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
error( RtError::WARNING );
return;
}
MUTEX_LOCK( &stream_.mutex );
// The state might change while waiting on a mutex.
if ( stream_.state == STREAM_STOPPED ) {
MUTEX_UNLOCK( &stream_.mutex );
return;
}
int result = 0;
OssHandle *handle = (OssHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
// Flush the output with zeros a few times.
char *buffer;
int samples;
RtAudioFormat format;
if ( stream_.doConvertBuffer[0] ) {
buffer = stream_.deviceBuffer;
samples = stream_.bufferSize * stream_.nDeviceChannels[0];
format = stream_.deviceFormat[0];
}
else {
buffer = stream_.userBuffer[0];
samples = stream_.bufferSize * stream_.nUserChannels[0];
format = stream_.userFormat;
}
memset( buffer, 0, samples * formatBytes(format) );
for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
result = write( handle->id[0], buffer, samples * formatBytes(format) );
if ( result == -1 ) {
errorText_ = "RtApiOss::stopStream: audio write error.";
error( RtError::WARNING );
}
}
result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
if ( result == -1 ) {
errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
errorText_ = errorStream_.str();
goto unlock;
}
handle->triggered = false;
}
if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
if ( result == -1 ) {
errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
errorText_ = errorStream_.str();
goto unlock;
}
}
unlock:
stream_.state = STREAM_STOPPED;
MUTEX_UNLOCK( &stream_.mutex );
if ( result != -1 ) return;
error( RtError::SYSTEM_ERROR );
}
void RtApiOss :: abortStream()
{
verifyStream();
if ( stream_.state == STREAM_STOPPED ) {
errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
error( RtError::WARNING );
return;
}
MUTEX_LOCK( &stream_.mutex );
// The state might change while waiting on a mutex.
if ( stream_.state == STREAM_STOPPED ) {
MUTEX_UNLOCK( &stream_.mutex );
return;
}
int result = 0;
OssHandle *handle = (OssHandle *) stream_.apiHandle;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
if ( result == -1 ) {
errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
errorText_ = errorStream_.str();
goto unlock;
}
handle->triggered = false;
}
if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
if ( result == -1 ) {
errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
errorText_ = errorStream_.str();
goto unlock;
}
}
unlock:
stream_.state = STREAM_STOPPED;
MUTEX_UNLOCK( &stream_.mutex );
if ( result != -1 ) return;
error( RtError::SYSTEM_ERROR );
}
void RtApiOss :: callbackEvent()
{
OssHandle *handle = (OssHandle *) stream_.apiHandle;
if ( stream_.state == STREAM_STOPPED ) {
MUTEX_LOCK( &stream_.mutex );
pthread_cond_wait( &handle->runnable, &stream_.mutex );
if ( stream_.state != STREAM_RUNNING ) {
MUTEX_UNLOCK( &stream_.mutex );
return;
}
MUTEX_UNLOCK( &stream_.mutex );
}
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
error( RtError::WARNING );
return;
}
// Invoke user callback to get fresh output data.
int doStopStream = 0;
RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
double streamTime = getStreamTime();
RtAudioStreamStatus status = 0;
if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
status |= RTAUDIO_OUTPUT_UNDERFLOW;
handle->xrun[0] = false;
}
if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
status |= RTAUDIO_INPUT_OVERFLOW;
handle->xrun[1] = false;
}
doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
if ( doStopStream == 2 ) {
this->abortStream();
return;
}
MUTEX_LOCK( &stream_.mutex );
// The state might change while waiting on a mutex.
if ( stream_.state == STREAM_STOPPED ) goto unlock;
int result;
char *buffer;
int samples;
RtAudioFormat format;
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
// Setup parameters and do buffer conversion if necessary.
if ( stream_.doConvertBuffer[0] ) {
buffer = stream_.deviceBuffer;
convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
samples = stream_.bufferSize * stream_.nDeviceChannels[0];
format = stream_.deviceFormat[0];
}
else {
buffer = stream_.userBuffer[0];
samples = stream_.bufferSize * stream_.nUserChannels[0];
format = stream_.userFormat;
}
// Do byte swapping if necessary.
if ( stream_.doByteSwap[0] )
byteSwapBuffer( buffer, samples, format );
if ( stream_.mode == DUPLEX && handle->triggered == false ) {
int trig = 0;
ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
result = write( handle->id[0], buffer, samples * formatBytes(format) );
trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
handle->triggered = true;
}
else
// Write samples to device.
result = write( handle->id[0], buffer, samples * formatBytes(format) );
if ( result == -1 ) {
// We'll assume this is an underrun, though there isn't a
// specific means for determining that.
handle->xrun[0] = true;
errorText_ = "RtApiOss::callbackEvent: audio write error.";
error( RtError::WARNING );
// Continue on to input section.
}
}
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[1];
samples = stream_.bufferSize * stream_.nUserChannels[1];
format = stream_.userFormat;
}
// Read samples from device.
result = read( handle->id[1], buffer, samples * formatBytes(format) );
if ( result == -1 ) {
// We'll assume this is an overrun, though there isn't a
// specific means for determining that.
handle->xrun[1] = true;
errorText_ = "RtApiOss::callbackEvent: audio read error.";
error( RtError::WARNING );
goto unlock;
}
// Do byte swapping if necessary.
if ( stream_.doByteSwap[1] )
byteSwapBuffer( buffer, samples, format );
// Do buffer conversion if necessary.
if ( stream_.doConvertBuffer[1] )
convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
}
unlock:
MUTEX_UNLOCK( &stream_.mutex );
RtApi::tickStreamTime();
if ( doStopStream == 1 ) this->stopStream();
}
extern "C" void *ossCallbackHandler( void *ptr )
{
CallbackInfo *info = (CallbackInfo *) ptr;
RtApiOss *object = (RtApiOss *) info->object;
bool *isRunning = &info->isRunning;
while ( *isRunning == true ) {
pthread_testcancel();
object->callbackEvent();
}
pthread_exit( NULL );
}
//******************** End of __LINUX_OSS__ *********************//
#endif
// *************************************************** //
//
// Protected common (OS-independent) RtAudio methods.
//
// *************************************************** //
// This method can be modified to control the behavior of error
// message printing.
void RtApi :: error( RtError::Type type )
{
errorStream_.str(""); // clear the ostringstream
if ( type == RtError::WARNING && showWarnings_ == true )
std::cerr << '\n' << errorText_ << "\n\n";
else
throw( RtError( errorText_, type ) );
}
void RtApi :: verifyStream()
{
if ( stream_.state == STREAM_CLOSED ) {
errorText_ = "RtApi:: a stream is not open!";
error( RtError::INVALID_USE );
}
}
void RtApi :: clearStreamInfo()
{
stream_.mode = UNINITIALIZED;
stream_.state = STREAM_CLOSED;
stream_.sampleRate = 0;
stream_.bufferSize = 0;
stream_.nBuffers = 0;
stream_.userFormat = 0;
stream_.userInterleaved = true;
stream_.streamTime = 0.0;
stream_.apiHandle = 0;
stream_.deviceBuffer = 0;
stream_.callbackInfo.callback = 0;
stream_.callbackInfo.userData = 0;
stream_.callbackInfo.isRunning = false;
for ( int i=0; i<2; i++ ) {
stream_.device[i] = 11111;
stream_.doConvertBuffer[i] = false;
stream_.deviceInterleaved[i] = true;
stream_.doByteSwap[i] = false;
stream_.nUserChannels[i] = 0;
stream_.nDeviceChannels[i] = 0;
stream_.channelOffset[i] = 0;
stream_.deviceFormat[i] = 0;
stream_.latency[i] = 0;
stream_.userBuffer[i] = 0;
stream_.convertInfo[i].channels = 0;
stream_.convertInfo[i].inJump = 0;
stream_.convertInfo[i].outJump = 0;
stream_.convertInfo[i].inFormat = 0;
stream_.convertInfo[i].outFormat = 0;
stream_.convertInfo[i].inOffset.clear();
stream_.convertInfo[i].outOffset.clear();
}
}
unsigned int RtApi :: formatBytes( RtAudioFormat format )
{
if ( format == RTAUDIO_SINT16 )
return 2;
else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 ||
format == RTAUDIO_FLOAT32 )
return 4;
else if ( format == RTAUDIO_FLOAT64 )
return 8;
else if ( format == RTAUDIO_SINT8 )
return 1;
errorText_ = "RtApi::formatBytes: undefined format.";
error( RtError::WARNING );
return 0;
}
void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
{
if ( mode == INPUT ) { // convert device to user buffer
stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
stream_.convertInfo[mode].outFormat = stream_.userFormat;
}
else { // convert user to device buffer
stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
stream_.convertInfo[mode].inFormat = stream_.userFormat;
stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
}
if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
else
stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
// Set up the interleave/deinterleave offsets.
if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
( mode == INPUT && stream_.userInterleaved ) ) {
for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
stream_.convertInfo[mode].outOffset.push_back( k );
stream_.convertInfo[mode].inJump = 1;
}
}
else {
for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
stream_.convertInfo[mode].inOffset.push_back( k );
stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
stream_.convertInfo[mode].outJump = 1;
}
}
}
else { // no (de)interleaving
if ( stream_.userInterleaved ) {
for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
stream_.convertInfo[mode].inOffset.push_back( k );
stream_.convertInfo[mode].outOffset.push_back( k );
}
}
else {
for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
stream_.convertInfo[mode].inJump = 1;
stream_.convertInfo[mode].outJump = 1;
}
}
}
// Add channel offset.
if ( firstChannel > 0 ) {
if ( stream_.deviceInterleaved[mode] ) {
if ( mode == OUTPUT ) {
for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
stream_.convertInfo[mode].outOffset[k] += firstChannel;
}
else {
for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
stream_.convertInfo[mode].inOffset[k] += firstChannel;
}
}
else {
if ( mode == OUTPUT ) {
for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
}
else {
for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
}
}
}
}
void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
{
// This function 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.
// Clear our device buffer when in/out duplex device channels are different
if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
int j;
if (info.outFormat == RTAUDIO_FLOAT64) {
Float64 scale;
Float64 *out = (Float64 *)outBuffer;
if (info.inFormat == RTAUDIO_SINT8) {
signed char *in = (signed char *)inBuffer;
scale = 1.0 / 127.5;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT16) {
Int16 *in = (Int16 *)inBuffer;
scale = 1.0 / 32767.5;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT24) {
Int32 *in = (Int32 *)inBuffer;
scale = 1.0 / 8388607.5;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]] & 0x00ffffff);
out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT32) {
Int32 *in = (Int32 *)inBuffer;
scale = 1.0 / 2147483647.5;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_FLOAT32) {
Float32 *in = (Float32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_FLOAT64) {
// Channel compensation and/or (de)interleaving only.
Float64 *in = (Float64 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = in[info.inOffset[j]];
}
in += info.inJump;
out += info.outJump;
}
}
}
else if (info.outFormat == RTAUDIO_FLOAT32) {
Float32 scale;
Float32 *out = (Float32 *)outBuffer;
if (info.inFormat == RTAUDIO_SINT8) {
signed char *in = (signed char *)inBuffer;
scale = (Float32) ( 1.0 / 127.5 );
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT16) {
Int16 *in = (Int16 *)inBuffer;
scale = (Float32) ( 1.0 / 32767.5 );
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT24) {
Int32 *in = (Int32 *)inBuffer;
scale = (Float32) ( 1.0 / 8388607.5 );
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]] & 0x00ffffff);
out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT32) {
Int32 *in = (Int32 *)inBuffer;
scale = (Float32) ( 1.0 / 2147483647.5 );
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
out[info.outOffset[j]] += 0.5;
out[info.outOffset[j]] *= scale;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_FLOAT32) {
// Channel compensation and/or (de)interleaving only.
Float32 *in = (Float32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = in[info.inOffset[j]];
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_FLOAT64) {
Float64 *in = (Float64 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
}
in += info.inJump;
out += info.outJump;
}
}
}
else if (info.outFormat == RTAUDIO_SINT32) {
Int32 *out = (Int32 *)outBuffer;
if (info.inFormat == RTAUDIO_SINT8) {
signed char *in = (signed char *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
out[info.outOffset[j]] <<= 24;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT16) {
Int16 *in = (Int16 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
out[info.outOffset[j]] <<= 16;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT24) {
Int32 *in = (Int32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
out[info.outOffset[j]] <<= 8;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT32) {
// Channel compensation and/or (de)interleaving only.
Int32 *in = (Int32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = in[info.inOffset[j]];
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_FLOAT32) {
Float32 *in = (Float32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_FLOAT64) {
Float64 *in = (Float64 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
}
}
}
else if (info.outFormat == RTAUDIO_SINT24) {
Int32 *out = (Int32 *)outBuffer;
if (info.inFormat == RTAUDIO_SINT8) {
signed char *in = (signed char *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
out[info.outOffset[j]] <<= 16;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT16) {
Int16 *in = (Int16 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
out[info.outOffset[j]] <<= 8;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT24) {
// Channel compensation and/or (de)interleaving only.
Int32 *in = (Int32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = in[info.inOffset[j]];
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT32) {
Int32 *in = (Int32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
out[info.outOffset[j]] >>= 8;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_FLOAT32) {
Float32 *in = (Float32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_FLOAT64) {
Float64 *in = (Float64 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
}
}
}
else if (info.outFormat == RTAUDIO_SINT16) {
Int16 *out = (Int16 *)outBuffer;
if (info.inFormat == RTAUDIO_SINT8) {
signed char *in = (signed char *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
out[info.outOffset[j]] <<= 8;
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT16) {
// Channel compensation and/or (de)interleaving only.
Int16 *in = (Int16 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = in[info.inOffset[j]];
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT24) {
Int32 *in = (Int32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 8) & 0x0000ffff);
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT32) {
Int32 *in = (Int32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_FLOAT32) {
Float32 *in = (Float32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_FLOAT64) {
Float64 *in = (Float64 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
}
}
}
else if (info.outFormat == RTAUDIO_SINT8) {
signed char *out = (signed char *)outBuffer;
if (info.inFormat == RTAUDIO_SINT8) {
// Channel compensation and/or (de)interleaving only.
signed char *in = (signed char *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = in[info.inOffset[j]];
}
in += info.inJump;
out += info.outJump;
}
}
if (info.inFormat == RTAUDIO_SINT16) {
Int16 *in = (Int16 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT24) {
Int32 *in = (Int32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 16) & 0x000000ff);
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_SINT32) {
Int32 *in = (Int32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_FLOAT32) {
Float32 *in = (Float32 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
}
}
else if (info.inFormat == RTAUDIO_FLOAT64) {
Float64 *in = (Float64 *)inBuffer;
for (unsigned int i=0; i<stream_.bufferSize; i++) {
for (j=0; j<info.channels; j++) {
out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
}
in += info.inJump;
out += info.outJump;
}
}
}
}
//static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
{
register char val;
register char *ptr;
ptr = buffer;
if ( format == RTAUDIO_SINT16 ) {
for ( unsigned int i=0; i<samples; i++ ) {
// Swap 1st and 2nd bytes.
val = *(ptr);
*(ptr) = *(ptr+1);
*(ptr+1) = val;
// Increment 2 bytes.
ptr += 2;
}
}
else if ( format == RTAUDIO_SINT24 ||
format == RTAUDIO_SINT32 ||
format == RTAUDIO_FLOAT32 ) {
for ( unsigned int i=0; i<samples; i++ ) {
// Swap 1st and 4th bytes.
val = *(ptr);
*(ptr) = *(ptr+3);
*(ptr+3) = val;
// Swap 2nd and 3rd bytes.
ptr += 1;
val = *(ptr);
*(ptr) = *(ptr+1);
*(ptr+1) = val;
// Increment 3 more bytes.
ptr += 3;
}
}
else if ( format == RTAUDIO_FLOAT64 ) {
for ( unsigned int i=0; i<samples; i++ ) {
// Swap 1st and 8th bytes
val = *(ptr);
*(ptr) = *(ptr+7);
*(ptr+7) = val;
// Swap 2nd and 7th bytes
ptr += 1;
val = *(ptr);
*(ptr) = *(ptr+5);
*(ptr+5) = val;
// Swap 3rd and 6th bytes
ptr += 1;
val = *(ptr);
*(ptr) = *(ptr+3);
*(ptr+3) = val;
// Swap 4th and 5th bytes
ptr += 1;
val = *(ptr);
*(ptr) = *(ptr+1);
*(ptr+1) = val;
// Increment 5 more bytes.
ptr += 5;
}
}
}
// Indentation settings for Vim and Emacs
//
// Local Variables:
// c-basic-offset: 2
// indent-tabs-mode: nil
// End:
//
// vim: et sts=2 sw=2
#endif
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