diff options
Diffstat (limited to 'RtAudio.cpp')
| -rw-r--r-- | RtAudio.cpp | 439 |
1 files changed, 262 insertions, 177 deletions
diff --git a/RtAudio.cpp b/RtAudio.cpp index 1f6a597..f7c918d 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -37,13 +37,7 @@ */ /************************************************************************/ -// RtAudio: Version 3.0.2 (14 October 2005) - -// Modified by Robin Davies, 1 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. +// RtAudio: Version 3.0.3 (18 November 2005) #include "RtAudio.h" #include <iostream> @@ -2742,11 +2736,35 @@ void RtApiJack :: initialize(void) if ( (client = jack_client_new( "RtApiJack" )) == 0) return; - RtApiDevice device; - // Determine the name of the device. - device.name = "Jack Server"; - devices_.push_back(device); - nDevices_++; + /* + RtApiDevice device; + // Determine the name of the device. + device.name = "Jack Server"; + devices_.push_back(device); + nDevices_++; + */ + const char **ports; + std::string port, prevPort; + unsigned int nChannels = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + port = (char *) ports[ nChannels ]; + unsigned int colonPos = 0; + do { + port = (char *) ports[ nChannels ]; + if ( (colonPos = port.find(":")) != std::string::npos ) { + port = port.substr( 0, colonPos+1 ); + if ( port != prevPort ) { + RtApiDevice device; + device.name = port; + devices_.push_back( device ); + nDevices_++; + prevPort = port; + } + } + } while ( ports[++nChannels] ); + free( ports ); + } jack_client_close(client); } @@ -2755,7 +2773,7 @@ void RtApiJack :: probeDeviceInfo(RtApiDevice *info) { // Look for jack server and try to become a client. jack_client_t *client; - if ( (client = jack_client_new( "RtApiJack" )) == 0) { + if ( (client = jack_client_new( "RtApiJack_Probe" )) == 0) { sprintf(message_, "RtApiJack: error connecting to Linux Jack server in probeDeviceInfo() (jack: %s)!", jackmsg.c_str()); error(RtError::WARNING); @@ -2771,7 +2789,7 @@ void RtApiJack :: probeDeviceInfo(RtApiDevice *info) const char **ports; char *port; unsigned int nChannels = 0; - ports = jack_get_ports( client, NULL, NULL, JackPortIsInput ); + ports = jack_get_ports( client, info->name.c_str(), NULL, JackPortIsInput ); if ( ports ) { port = (char *) ports[nChannels]; while ( port ) @@ -2783,7 +2801,7 @@ void RtApiJack :: probeDeviceInfo(RtApiDevice *info) // Jack "output ports" equal RtAudio input channels. nChannels = 0; - ports = jack_get_ports( client, NULL, NULL, JackPortIsOutput ); + ports = jack_get_ports( client, info->name.c_str(), NULL, JackPortIsOutput ); if ( ports ) { port = (char *) ports[nChannels]; while ( port ) @@ -3178,7 +3196,7 @@ void RtApiJack :: startStream() int result; // Get the list of available ports. if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - ports = jack_get_ports(handle->client, NULL, NULL, JackPortIsPhysical|JackPortIsInput); + ports = jack_get_ports(handle->client, devices_[stream_.device[0]].name.c_str(), NULL, JackPortIsInput); if ( ports == NULL) { sprintf(message_, "RtApiJack: error determining available jack input ports!"); error(RtError::SYSTEM_ERROR); @@ -3201,7 +3219,7 @@ void RtApiJack :: startStream() } if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - ports = jack_get_ports( handle->client, NULL, NULL, JackPortIsPhysical|JackPortIsOutput ); + ports = jack_get_ports( handle->client, devices_[stream_.device[1]].name.c_str(), NULL, JackPortIsOutput ); if ( ports == NULL) { sprintf(message_, "RtApiJack: error determining available jack output ports!"); error(RtError::SYSTEM_ERROR); @@ -4209,6 +4227,72 @@ void RtApiAlsa :: closeStream() stream_.mode = UNINITIALIZED; } +// Pump a bunch of zeros into the output buffer. This is needed only when we +// are doing duplex operations. +bool RtApiAlsa :: primeOutputBuffer() +{ + int err; + char *buffer; + int channels; + snd_pcm_t **handle; + RtAudioFormat format; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + handle = (snd_pcm_t **) apiInfo->handles; + + if (stream_.mode == DUPLEX) { + + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + convertBuffer( stream_.deviceBuffer, apiInfo->tempBuffer, stream_.convertInfo[0] ); + channels = stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + channels = stream_.nUserChannels[0]; + format = stream_.userFormat; + } + + buffer = new char[stream_.bufferSize * formatBytes(format) * channels]; + bzero(buffer, stream_.bufferSize * formatBytes(format) * channels); + + for (int i=0; i<stream_.nBuffers; i++) { + // Write samples to device in interleaved/non-interleaved format. + if (stream_.deInterleave[0]) { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes(format); + for (int i=0; i<channels; i++) + bufs[i] = (void *) (buffer + (i * offset)); + err = snd_pcm_writen(handle[0], bufs, stream_.bufferSize); + } + else + err = snd_pcm_writei(handle[0], buffer, stream_.bufferSize); + + if (err < stream_.bufferSize) { + // Either an error or underrun occured. + if (err == -EPIPE) { + snd_pcm_state_t state = snd_pcm_state(handle[0]); + if (state == SND_PCM_STATE_XRUN) { + sprintf(message_, "RtApiAlsa: underrun detected while priming output buffer."); + return false; + } + else { + sprintf(message_, "RtApiAlsa: primeOutputBuffer() error, current state is %s.", + snd_pcm_state_name(state)); + return false; + } + } + else { + sprintf(message_, "RtApiAlsa: audio write error for device (%s): %s.", + devices_[stream_.device[0]].name.c_str(), snd_strerror(err)); + return false; + } + } + } + } + + return true; +} + void RtApiAlsa :: startStream() { // This method calls snd_pcm_prepare if the device isn't already in that state. @@ -4232,6 +4316,11 @@ void RtApiAlsa :: startStream() MUTEX_UNLOCK(&stream_.mutex); error(RtError::DRIVER_ERROR); } + // Reprime output buffer if needed + if ( (stream_.mode == DUPLEX) && ( !primeOutputBuffer() ) ) { + MUTEX_UNLOCK(&stream_.mutex); + error(RtError::DRIVER_ERROR); + } } } @@ -4247,6 +4336,11 @@ void RtApiAlsa :: startStream() } } } + + if ( (stream_.mode == DUPLEX) && ( !primeOutputBuffer() ) ) { + MUTEX_UNLOCK(&stream_.mutex); + error(RtError::DRIVER_ERROR); + } stream_.state = STREAM_RUNNING; MUTEX_UNLOCK(&stream_.mutex); @@ -4441,6 +4535,11 @@ void RtApiAlsa :: tickStream() MUTEX_UNLOCK(&stream_.mutex); error(RtError::DRIVER_ERROR); } + // Reprime output buffer if needed. + if ( (stream_.mode == DUPLEX) && ( !primeOutputBuffer() ) ) { + MUTEX_UNLOCK(&stream_.mutex); + error(RtError::DRIVER_ERROR); + } } else { sprintf(message_, "RtApiAlsa: tickStream() error, current state is %s.", @@ -4639,6 +4738,7 @@ extern "C" void *alsaCallbackHandler(void *ptr) #include "asio/asiosys.h" #include "asio/asio.h" +#include "asio/iasiothiscallresolver.h" #include "asio/asiodrivers.h" #include <math.h> @@ -4656,7 +4756,7 @@ struct AsioHandle { :stopStream(false), bufferInfos(0) {} }; -static const char*GetAsioErrorString(ASIOError result) +static const char* GetAsioErrorString( ASIOError result ) { struct Messages { @@ -4674,10 +4774,9 @@ static const char*GetAsioErrorString(ASIOError result) { ASE_NoMemory, "Not enough memory to complete the request." } }; - for (int i = 0; i < sizeof(m)/sizeof(m[0]); ++i) - { + for (unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i) if (m[i].value == result) return m[i].message; - } + return "Unknown error."; } @@ -4695,11 +4794,9 @@ RtApiAsio :: RtApiAsio() RtApiAsio :: ~RtApiAsio() { if ( stream_.mode != UNINITIALIZED ) closeStream(); + if ( coInitialized ) - { CoUninitialize(); - } - } void RtApiAsio :: initialize(void) @@ -4709,8 +4806,7 @@ void RtApiAsio :: initialize(void) // for appartment threading (in which case, CoInitilialize will return S_FALSE here). coInitialized = false; HRESULT hr = CoInitialize(NULL); - if (FAILED(hr)) - { + if ( FAILED(hr) ) { sprintf(message_,"RtApiAsio: ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"); } coInitialized = true; @@ -5573,29 +5669,24 @@ void RtApiAsio :: callbackEvent(long bufferIndex) static inline DWORD dsPointerDifference(DWORD laterPointer,DWORD earlierPointer,DWORD bufferSize) { if (laterPointer > earlierPointer) - { return laterPointer-earlierPointer; - } else - { + 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; } - #undef GENERATE_DEBUG_LOG // Define this to generate a debug timing log file in c:/rtaudiolog.txt" #ifdef GENERATE_DEBUG_LOG @@ -5637,38 +5728,34 @@ RtApiDs::RtDsStatistics RtApiDs::getDsStatistics() 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; - } + 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 deviceCountCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, + LPCTSTR description, + LPCTSTR module, LPVOID lpContext); static bool CALLBACK deviceInfoCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, + LPCTSTR description, + LPCTSTR module, LPVOID lpContext); static bool CALLBACK defaultDeviceCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, + LPCTSTR description, + LPCTSTR module, LPVOID lpContext); static bool CALLBACK deviceIdCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, + LPCTSTR description, + LPCTSTR module, LPVOID lpContext); static char* getErrorString(int code); @@ -5676,7 +5763,7 @@ static char* getErrorString(int code); extern "C" unsigned __stdcall callbackHandler(void *ptr); struct enum_info { - char name[64]; + std::string name; LPGUID id; bool isInput; bool isValid; @@ -5684,13 +5771,12 @@ struct enum_info { RtApiDs :: RtApiDs() { - // Dsound will run both-threaded. If CoInitialize fails, then just accept whatever the mainline - // chose for a threading model. + // 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)) { + if ( !FAILED(hr) ) coInitialized = true; - } this->initialize(); @@ -5703,16 +5789,14 @@ RtApiDs :: RtApiDs() RtApiDs :: ~RtApiDs() { if (coInitialized) - { CoUninitialize(); // balanced call. - } + if ( stream_.mode != UNINITIALIZED ) closeStream(); } int RtApiDs :: getDefaultInputDevice(void) { enum_info info; - info.name[0] = '\0'; // Enumerate through devices to find the default output. HRESULT result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)defaultDeviceCallback, &info); @@ -5724,10 +5808,9 @@ int RtApiDs :: getDefaultInputDevice(void) } for ( int i=0; i<nDevices_; i++ ) { - if ( strncmp( info.name, devices_[i].name.c_str(), 64 ) == 0 ) return i; + if ( info.name == devices_[i].name ) return i; } - return 0; } @@ -5746,7 +5829,7 @@ int RtApiDs :: getDefaultOutputDevice(void) } for ( int i=0; i<nDevices_; i++ ) - if ( strncmp( info.name, devices_[i].name.c_str(), 64 ) == 0 ) return i; + if ( info.name == devices_[i].name ) return i; return 0; } @@ -5778,7 +5861,6 @@ void RtApiDs :: initialize(void) std::vector<enum_info> info(count); for (i=0; i<count; i++) { - info[i].name[0] = '\0'; if (i < outs) info[i].isInput = false; else info[i].isInput = true; } @@ -5804,11 +5886,10 @@ void RtApiDs :: initialize(void) // opened, they report < 1 supported channels, or they report no // supported data (capture only). RtApiDevice device; - int index = 0; for (i=0; i<count; i++) { if ( info[i].isValid ) { device.name.erase(); - device.name.append( (const char *)info[i].name, strlen(info[i].name)+1); + device.name = info[i].name; devices_.push_back(device); } } @@ -5820,7 +5901,7 @@ void RtApiDs :: initialize(void) void RtApiDs :: probeDeviceInfo(RtApiDevice *info) { enum_info dsinfo; - strncpy( dsinfo.name, info->name.c_str(), 64 ); + dsinfo.name = info->name; dsinfo.isValid = false; // Enumerate through input devices to find the id (if it exists). @@ -6073,32 +6154,25 @@ bool RtApiDs :: probeDeviceOpen( int device, StreamMode mode, int channels, waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; - // Determine the device buffer size. By default, 32k, - // but we will grow it to make allowances for very large softare buffer sizes. + // 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; buffer_size = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this - - // poisonously large buffer lead time? Then increase the device buffer size accordingly. - while (dsPointerLeadTime *2U > (DWORD)buffer_size) - { - buffer_size *= 2; - } - - - enum_info dsinfo; void *ohandle = 0, *bhandle = 0; - strncpy( dsinfo.name, devices_[device].name.c_str(), 64 ); + // strncpy( dsinfo.name, devices_[device].name.c_str(), 64 ); + dsinfo.name = devices_[device].name; dsinfo.isValid = false; if ( mode == OUTPUT ) { - dsPointerLeadTime = (numberOfBuffers) * - (*bufferSize) * - (waveFormat.wBitsPerSample / 8) - *channels; + dsPointerLeadTime = numberOfBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; + + // If the user wants an even bigger buffer, increase the device buffer size accordingly. + while ( dsPointerLeadTime * 2U > (DWORD)buffer_size ) + buffer_size *= 2; if ( devices_[device].maxOutputChannels < channels ) { sprintf(message_, "RtApiDs: requested channels (%d) > than supported (%d) by device (%s).", @@ -6595,18 +6669,16 @@ void RtApiDs :: startStream() verifyStream(); if (stream_.state == STREAM_RUNNING) 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. + // 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 *handles = (DsHandle *) stream_.apiHandle; timeBeginPeriod(1); - memset(&statistics,0,sizeof(statistics)); statistics.sampleRate = stream_.sampleRate; statistics.writeDeviceBufferLeadBytes = handles[0].dsPointerLeadTime ; @@ -6614,8 +6686,7 @@ void RtApiDs :: startStream() buffersRolling = false; duplexPrerollBytes = 0; - if (stream_.mode == DUPLEX) - { + 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]); } @@ -6629,9 +6700,8 @@ void RtApiDs :: startStream() statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0]) *stream_.nDeviceChannels[0]; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer; - result = buffer->Play(0, 0, DSBPLAY_LOOPING ); + result = buffer->Play( 0, 0, DSBPLAY_LOOPING ); if ( FAILED(result) ) { sprintf(message_, "RtApiDs: Unable to start buffer (%s): %s.", devices_[stream_.device[0]].name.c_str(), getErrorString(result)); @@ -6661,34 +6731,30 @@ void RtApiDs :: stopStream() verifyStream(); if (stream_.state == STREAM_STOPPED) return; - // Change the state before the lock to improve shutdown response // when using a callback. stream_.state = STREAM_STOPPED; MUTEX_LOCK(&stream_.mutex); - timeEndPeriod(1); // revert to normal scheduler frequency on lesser windows. #ifdef GENERATE_DEBUG_LOG - // write the timing log to a .TSV file for analysis in Excel. + // Write the timing log to a .TSV file for analysis in Excel. unlink("c:/rtaudiolog.txt"); std::ofstream os("c:/rtaudiolog.txt"); os << "writeTime\treadDelay\tnextWritePointer\tnextReadPointer\tcurrentWritePointer\tsafeWritePointer\tcurrentReadPointer\tsafeReadPointer" << std::endl; - for (int i = 0; i < currentDebugLogEntry ; ++i) - { + for (int i = 0; i < currentDebugLogEntry ; ++i) { TTickRecord &r = debugLog[i]; - os - << r.writeTime-debugLog[0].writeTime << "\t" << (r.readTime-r.writeTime) << "\t" - << r.nextWritePointer % BUFFER_SIZE << "\t" << r.nextReadPointer % BUFFER_SIZE - << "\t" << r.currentWritePointer % BUFFER_SIZE << "\t" << r.safeWritePointer % BUFFER_SIZE - << "\t" << r.currentReadPointer % BUFFER_SIZE << "\t" << r.safeReadPointer % BUFFER_SIZE << std::endl; + os << r.writeTime-debugLog[0].writeTime << "\t" << (r.readTime-r.writeTime) << "\t" + << r.nextWritePointer % BUFFER_SIZE << "\t" << r.nextReadPointer % BUFFER_SIZE + << "\t" << r.currentWritePointer % BUFFER_SIZE << "\t" << r.safeWritePointer % BUFFER_SIZE + << "\t" << r.currentReadPointer % BUFFER_SIZE << "\t" << r.safeReadPointer % BUFFER_SIZE << std::endl; } #endif // There is no specific DirectSound API call to "drain" a buffer - // before stopping. We can hack this for playback by writing zeroes - // for another bufferSize * nBuffers frames. For capture, the + // before stopping. We can hack this for playback by writing + // buffers of zeroes over the entire buffer. For capture, the // concept is less clear so we'll repeat what we do in the // abortStream() case. HRESULT result; @@ -6701,27 +6767,26 @@ void RtApiDs :: stopStream() if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) { DWORD currentPos, safePos; - long buffer_bytes = stream_.bufferSize * stream_.nDeviceChannels[0] - * formatBytes(stream_.deviceFormat[0]); - + long buffer_bytes = stream_.bufferSize * stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]); LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer; - long nextWritePos = handles[0].bufferPointer; + DWORD nextWritePos = handles[0].bufferPointer; dsBufferSize = handles[0].dsBufferSize; + DWORD dsBytesWritten = 0; - // Write zeroes for nBuffer counts. - for (int i=0; i<stream_.nBuffers; i++) { + // Write zeroes for at least dsBufferSize bytes. + while ( dsBytesWritten < dsBufferSize ) { // Find out where the read and "safe write" pointers are. - result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); + result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos ); if ( FAILED(result) ) { sprintf(message_, "RtApiDs: Unable to get current position (%s): %s.", devices_[stream_.device[0]].name.c_str(), getErrorString(result)); error(RtError::DRIVER_ERROR); } - // Chase nextWritePos. - if ( currentPos < (DWORD)nextWritePos ) currentPos += dsBufferSize; // unwrap offset + // Chase nextWritePosition. + if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset DWORD endWrite = nextWritePos + buffer_bytes; // Check whether the entire write region is behind the play pointer. @@ -6738,11 +6803,12 @@ void RtApiDs :: stopStream() devices_[stream_.device[0]].name.c_str(), getErrorString(result)); error(RtError::DRIVER_ERROR); } + if ( currentPos < (DWORD)nextWritePos ) currentPos += dsBufferSize; // unwrap offset } // Lock free space in the buffer - result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1, + result = dsBuffer->Lock( nextWritePos, buffer_bytes, &buffer1, &bufferSize1, &buffer2, &bufferSize2, 0); if ( FAILED(result) ) { sprintf(message_, "RtApiDs: Unable to lock buffer during playback (%s): %s.", @@ -6751,11 +6817,11 @@ void RtApiDs :: stopStream() } // Zero the free space - ZeroMemory(buffer1, bufferSize1); - if (buffer2 != NULL) ZeroMemory(buffer2, bufferSize2); + ZeroMemory( buffer1, bufferSize1 ); + if (buffer2 != NULL) ZeroMemory( buffer2, bufferSize2 ); // Update our buffer offset and unlock sound buffer - dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2); + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); if ( FAILED(result) ) { sprintf(message_, "RtApiDs: Unable to unlock buffer during playback (%s): %s.", devices_[stream_.device[0]].name.c_str(), getErrorString(result)); @@ -6763,6 +6829,15 @@ void RtApiDs :: stopStream() } nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize; handles[0].bufferPointer = nextWritePos; + dsBytesWritten += buffer_bytes; + } + + // OK, now stop the buffer. + result = dsBuffer->Stop(); + if ( FAILED(result) ) { + sprintf(message_, "RtApiDs: Unable to stop buffer (%s): %s", + devices_[stream_.device[0]].name.c_str(), getErrorString(result)); + error(RtError::DRIVER_ERROR); } // If we play again, start at the beginning of the buffer. @@ -6821,6 +6896,8 @@ void RtApiDs :: abortStream() stream_.state = STREAM_STOPPED; MUTEX_LOCK(&stream_.mutex); + timeEndPeriod(1); // revert to normal scheduler frequency on lesser windows. + HRESULT result; long dsBufferSize; LPVOID audioPtr; @@ -7001,6 +7078,7 @@ void RtApiDs :: tickStream() #ifdef GENERATE_DEBUG_LOG DWORD writeTime, readTime; #endif + LPVOID buffer1 = NULL; LPVOID buffer2 = NULL; DWORD bufferSize1 = 0; @@ -7010,29 +7088,29 @@ void RtApiDs :: tickStream() long buffer_bytes; DsHandle *handles = (DsHandle *) stream_.apiHandle; - if (stream_.mode == DUPLEX && !buffersRolling) - { + if (stream_.mode == DUPLEX && !buffersRolling) { assert(handles[0].dsBufferSize == handles[1].dsBufferSize); - // 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. + // 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. + // 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) handles[0].buffer; LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer; - DWORD initialWritePos, initialSafeWritePos; DWORD initialReadPos, initialSafeReadPos;; - result = dsWriteBuffer->GetCurrentPosition(&initialWritePos, &initialSafeWritePos); if ( FAILED(result) ) { sprintf(message_, "RtApiDs: Unable to get current position (%s): %s.", @@ -7045,8 +7123,7 @@ void RtApiDs :: tickStream() devices_[stream_.device[1]].name.c_str(), getErrorString(result)); error(RtError::DRIVER_ERROR); } - while (true) - { + while (true) { result = dsWriteBuffer->GetCurrentPosition(¤tWritePos, &safeWritePos); if ( FAILED(result) ) { sprintf(message_, "RtApiDs: Unable to get current position (%s): %s.", @@ -7059,16 +7136,14 @@ void RtApiDs :: tickStream() devices_[stream_.device[1]].name.c_str(), getErrorString(result)); error(RtError::DRIVER_ERROR); } - if (safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos) - { + if (safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos) { break; } Sleep(1); } - assert(handles[0].dsBufferSize == handles[1].dsBufferSize); + assert( handles[0].dsBufferSize == handles[1].dsBufferSize ); - UINT writeBufferLead = (safeWritePos-safeReadPos + handles[0].dsBufferSize) % handles[0].dsBufferSize; buffersRolling = true; handles[0].bufferPointer = (safeWritePos + handles[0].dsPointerLeadTime); handles[1].bufferPointer = safeReadPos; @@ -7104,8 +7179,7 @@ void RtApiDs :: tickStream() nextWritePos = handles[0].bufferPointer; DWORD endWrite; - while (true) - { + while ( true ) { // Find out where the read and "safe write" pointers are. result = dsBuffer->GetCurrentPosition(¤tWritePos, &safeWritePos); if ( FAILED(result) ) { @@ -7115,16 +7189,11 @@ void RtApiDs :: tickStream() } leadPos = safeWritePos + handles[0].dsPointerLeadTime; - if (leadPos > dsBufferSize) { - leadPos -= dsBufferSize; - } + if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize; if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset - - endWrite = nextWritePos + buffer_bytes; - // Check whether the entire write region is behind the play pointer. - + // 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 @@ -7139,28 +7208,24 @@ void RtApiDs :: tickStream() 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) { + if ( millis > 50.0 ) { static int nOverruns = 0; ++nOverruns; } Sleep( (DWORD) millis ); - // Sleep( (DWORD) 2); } + #ifdef GENERATE_DEBUG_LOG writeTime = timeGetTime(); #endif - if (statistics.writeDeviceSafeLeadBytes < dsPointerDifference(safeWritePos,currentWritePos,handles[0].dsBufferSize)) - { + + if (statistics.writeDeviceSafeLeadBytes < dsPointerDifference(safeWritePos,currentWritePos,handles[0].dsBufferSize)) { statistics.writeDeviceSafeLeadBytes = dsPointerDifference(safeWritePos,currentWritePos,handles[0].dsBufferSize); } - if ( - dsPointerBetween(nextWritePos,safeWritePos,currentWritePos,dsBufferSize) - || dsPointerBetween(endWrite,safeWritePos,currentWritePos,dsBufferSize) - ) - { - // we've strayed into the forbidden zone. - // resync the read pointer. + if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize ) + || dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) { + // We've strayed into the forbidden zone ... resync the read pointer. ++statistics.numberOfWriteUnderruns; nextWritePos = safeWritePos + handles[0].dsPointerLeadTime-buffer_bytes+dsBufferSize; while (nextWritePos >= dsBufferSize) nextWritePos-= dsBufferSize; @@ -7169,8 +7234,8 @@ void RtApiDs :: tickStream() } // Lock free space in the buffer - result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0); + result = dsBuffer->Lock( nextWritePos, buffer_bytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); if ( FAILED(result) ) { sprintf(message_, "RtApiDs: Unable to lock buffer during playback (%s): %s.", devices_[stream_.device[0]].name.c_str(), getErrorString(result)); @@ -7182,7 +7247,7 @@ void RtApiDs :: tickStream() if (buffer2 != NULL) CopyMemory(buffer2, buffer+bufferSize1, bufferSize2); // Update our buffer offset and unlock sound buffer - dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2); + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); if ( FAILED(result) ) { sprintf(message_, "RtApiDs: Unable to unlock buffer during playback (%s): %s.", devices_[stream_.device[0]].name.c_str(), getErrorString(result)); @@ -7230,9 +7295,9 @@ void RtApiDs :: tickStream() // 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. + // 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) { @@ -7381,8 +7446,8 @@ extern "C" unsigned __stdcall callbackHandler(void *ptr) } static bool CALLBACK deviceCountCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, + LPCTSTR description, + LPCTSTR module, LPVOID lpContext) { int *pointer = ((int *) lpContext); @@ -7391,22 +7456,41 @@ static bool CALLBACK deviceCountCallback(LPGUID lpguid, return true; } +#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 deviceInfoCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, + LPCTSTR description, + LPCTSTR module, LPVOID lpContext) { enum_info *info = ((enum_info *) lpContext); - while (strlen(info->name) > 0) info++; + while ( !info->name.empty() ) info++; - strncpy(info->name, lpcstrDescription, 64); + info->name = convertTChar( description ); info->id = lpguid; - HRESULT hr; + HRESULT hr; info->isValid = false; if (info->isInput == true) { - DSCCAPS caps; - LPDIRECTSOUNDCAPTURE object; + DSCCAPS caps; + LPDIRECTSOUNDCAPTURE object; hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); if( hr != DS_OK ) return true; @@ -7420,8 +7504,8 @@ static bool CALLBACK deviceInfoCallback(LPGUID lpguid, object->Release(); } else { - DSCAPS caps; - LPDIRECTSOUND object; + DSCAPS caps; + LPDIRECTSOUND object; hr = DirectSoundCreate( lpguid, &object, NULL ); if( hr != DS_OK ) return true; @@ -7438,14 +7522,14 @@ static bool CALLBACK deviceInfoCallback(LPGUID lpguid, } static bool CALLBACK defaultDeviceCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, + LPCTSTR description, + LPCTSTR module, LPVOID lpContext) { enum_info *info = ((enum_info *) lpContext); if ( lpguid == NULL ) { - strncpy(info->name, lpcstrDescription, 64); + info->name = convertTChar( description ); return false; } @@ -7453,13 +7537,14 @@ static bool CALLBACK defaultDeviceCallback(LPGUID lpguid, } static bool CALLBACK deviceIdCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, + LPCTSTR description, + LPCTSTR module, LPVOID lpContext) { enum_info *info = ((enum_info *) lpContext); - if ( strncmp( info->name, lpcstrDescription, 64 ) == 0 ) { + std::string s = convertTChar( description ); + if ( info->name == s ) { info->id = lpguid; info->isValid = true; return false; |