From a3d2ee35944db4dd0a3a342bb7f2df69f229f45d Mon Sep 17 00:00:00 2001 From: Gary Scavone Date: Wed, 9 Oct 2013 23:44:33 +0200 Subject: Version 2.1 --- RtAudio.cpp | 12083 ++++++++++++++++++++++++++++++++++------------------------ 1 file changed, 7082 insertions(+), 5001 deletions(-) (limited to 'RtAudio.cpp') diff --git a/RtAudio.cpp b/RtAudio.cpp index fd116da..4a51baf 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -1,5001 +1,7082 @@ -/******************************************/ -/* - RtAudio - realtime sound I/O C++ class - by Gary P. Scavone, 2001-2002. -*/ -/******************************************/ - -#include "RtAudio.h" -#include -#include - -// Static variable definitions. -const unsigned int RtAudio :: SAMPLE_RATES[] = { - 4000, 5512, 8000, 9600, 11025, 16000, 22050, - 32000, 44100, 48000, 88200, 96000, 176400, 192000 -}; -const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT8 = 1; -const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT16 = 2; -const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT24 = 4; -const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT32 = 8; -const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_FLOAT32 = 16; -const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_FLOAT64 = 32; - -#if defined(__WINDOWS_DS__) - #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) - #define MUTEX_LOCK(A) EnterCriticalSection(A) - #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) - typedef unsigned THREAD_RETURN; - typedef unsigned (__stdcall THREAD_FUNCTION)(void *); -#else // pthread API - #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) - #define MUTEX_LOCK(A) pthread_mutex_lock(A) - #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A) - typedef void * THREAD_RETURN; -#endif - -// *************************************************** // -// -// Public common (OS-independent) methods. -// -// *************************************************** // - -RtAudio :: RtAudio() -{ - initialize(); - - if (nDevices <= 0) { - sprintf(message, "RtAudio: no audio devices found!"); - error(RtError::NO_DEVICES_FOUND); - } -} - -RtAudio :: RtAudio(int *streamId, - int outputDevice, int outputChannels, - int inputDevice, int inputChannels, - RTAUDIO_FORMAT format, int sampleRate, - int *bufferSize, int numberOfBuffers) -{ - initialize(); - - if (nDevices <= 0) { - sprintf(message, "RtAudio: no audio devices found!"); - error(RtError::NO_DEVICES_FOUND); - } - - try { - *streamId = openStream(outputDevice, outputChannels, inputDevice, inputChannels, - format, sampleRate, bufferSize, numberOfBuffers); - } - catch (RtError &exception) { - // deallocate the RTAUDIO_DEVICE structures - if (devices) free(devices); - error(exception.getType()); - } -} - -RtAudio :: ~RtAudio() -{ - // close any existing streams - while ( streams.size() ) - closeStream( streams.begin()->first ); - - // deallocate the RTAUDIO_DEVICE structures - if (devices) free(devices); -} - -int RtAudio :: openStream(int outputDevice, int outputChannels, - int inputDevice, int inputChannels, - RTAUDIO_FORMAT format, int sampleRate, - int *bufferSize, int numberOfBuffers) -{ - static int streamKey = 0; // Unique stream identifier ... OK for multiple instances. - - if (outputChannels < 1 && inputChannels < 1) { - sprintf(message,"RtAudio: one or both 'channel' parameters must be greater than zero."); - error(RtError::INVALID_PARAMETER); - } - - if ( formatBytes(format) == 0 ) { - sprintf(message,"RtAudio: 'format' parameter value is undefined."); - error(RtError::INVALID_PARAMETER); - } - - if ( outputChannels > 0 ) { - if (outputDevice >= nDevices || outputDevice < 0) { - sprintf(message,"RtAudio: 'outputDevice' parameter value (%d) is invalid.", outputDevice); - error(RtError::INVALID_PARAMETER); - } - } - - if ( inputChannels > 0 ) { - if (inputDevice >= nDevices || inputDevice < 0) { - sprintf(message,"RtAudio: 'inputDevice' parameter value (%d) is invalid.", inputDevice); - error(RtError::INVALID_PARAMETER); - } - } - - // Allocate a new stream structure. - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) calloc(1, sizeof(RTAUDIO_STREAM)); - if (stream == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::MEMORY_ERROR); - } - streams[++streamKey] = (void *) stream; - stream->mode = UNINITIALIZED; - MUTEX_INITIALIZE(&stream->mutex); - - bool result = SUCCESS; - int device; - STREAM_MODE mode; - int channels; - if ( outputChannels > 0 ) { - - device = outputDevice; - mode = PLAYBACK; - channels = outputChannels; - - if (device == 0) { // Try default device first. - for (int i=0; i 0 && result == SUCCESS ) { - - device = inputDevice; - mode = RECORD; - channels = inputChannels; - - if (device == 0) { // Try default device first. - for (int i=0; i= nDevices || device < 0) { - sprintf(message, "RtAudio: invalid device specifier (%d)!", device); - error(RtError::INVALID_DEVICE); - } - - // If the device wasn't successfully probed before, try it again. - if (devices[device].probed == false) { - clearDeviceInfo(&devices[device]); - probeDeviceInfo(&devices[device]); - } - - // Clear the info structure. - memset(info, 0, sizeof(RTAUDIO_DEVICE)); - - strncpy(info->name, devices[device].name, 128); - info->probed = devices[device].probed; - if ( info->probed == true ) { - info->maxOutputChannels = devices[device].maxOutputChannels; - info->maxInputChannels = devices[device].maxInputChannels; - info->maxDuplexChannels = devices[device].maxDuplexChannels; - info->minOutputChannels = devices[device].minOutputChannels; - info->minInputChannels = devices[device].minInputChannels; - info->minDuplexChannels = devices[device].minDuplexChannels; - info->hasDuplexSupport = devices[device].hasDuplexSupport; - info->nSampleRates = devices[device].nSampleRates; - if (info->nSampleRates == -1) { - info->sampleRates[0] = devices[device].sampleRates[0]; - info->sampleRates[1] = devices[device].sampleRates[1]; - } - else { - for (int i=0; inSampleRates; i++) - info->sampleRates[i] = devices[device].sampleRates[i]; - } - info->nativeFormats = devices[device].nativeFormats; - } - - return; -} - -char * const RtAudio :: getStreamBuffer(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - return stream->userBuffer; -} - -// This global structure is used to pass information to the thread -// function. I tried other methods but had intermittent errors due to -// variable persistence during thread startup. -struct { - RtAudio *object; - int streamId; -} thread_info; - -extern "C" THREAD_RETURN THREAD_TYPE callbackHandler(void * ptr); - -void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - stream->callback = callback; - stream->userData = userData; - stream->usingCallback = true; - thread_info.object = this; - thread_info.streamId = streamId; - - int err = 0; -#if defined(__WINDOWS_DS__) - unsigned thread_id; - stream->thread = _beginthreadex(NULL, 0, &callbackHandler, - &stream->usingCallback, 0, &thread_id); - if (stream->thread == 0) err = -1; - // When spawning multiple threads in quick succession, it appears to be - // necessary to wait a bit for each to initialize ... another windism! - Sleep(1); -#else - err = pthread_create(&stream->thread, NULL, callbackHandler, &stream->usingCallback); -#endif - - if (err) { - stream->usingCallback = false; - sprintf(message, "RtAudio: error starting callback thread!"); - error(RtError::THREAD_ERROR); - } -} - -// *************************************************** // -// -// OS/API-specific methods. -// -// *************************************************** // - -#if defined(__LINUX_ALSA__) - -#define MAX_DEVICES 16 - -void RtAudio :: initialize(void) -{ - int card, result, device; - char name[32]; - char deviceNames[MAX_DEVICES][32]; - snd_ctl_t *handle; - snd_ctl_card_info_t *info; - snd_ctl_card_info_alloca(&info); - - // Count cards and devices - nDevices = 0; - card = -1; - snd_card_next(&card); - while ( card >= 0 ) { - sprintf(name, "hw:%d", card); - result = snd_ctl_open(&handle, name, 0); - if (result < 0) { - sprintf(message, "RtAudio: ALSA control open (%i): %s.", card, snd_strerror(result)); - error(RtError::WARNING); - goto next_card; - } - result = snd_ctl_card_info(handle, info); - if (result < 0) { - sprintf(message, "RtAudio: ALSA control hardware info (%i): %s.", card, snd_strerror(result)); - error(RtError::WARNING); - goto next_card; - } - device = -1; - while (1) { - result = snd_ctl_pcm_next_device(handle, &device); - if (result < 0) { - sprintf(message, "RtAudio: ALSA control next device (%i): %s.", card, snd_strerror(result)); - error(RtError::WARNING); - break; - } - if (device < 0) - break; - sprintf( deviceNames[nDevices++], "hw:%d,%d", card, device ); - if ( nDevices > MAX_DEVICES ) break; - } - if ( nDevices > MAX_DEVICES ) break; - next_card: - snd_ctl_close(handle); - snd_card_next(&card); - } - - if (nDevices == 0) return; - - // Allocate the RTAUDIO_DEVICE structures. - devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); - if (devices == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::MEMORY_ERROR); - } - - // Write device ascii identifiers to device structures and then - // probe the device capabilities. - for (int i=0; iname, stream, open_mode); - if (err < 0) { - sprintf(message, "RtAudio: ALSA pcm playback open (%s): %s.", - info->name, snd_strerror(err)); - error(RtError::WARNING); - goto capture_probe; - } - - snd_pcm_hw_params_t *params; - snd_pcm_hw_params_alloca(¶ms); - - // We have an open device ... allocate the parameter structure. - err = snd_pcm_hw_params_any(handle, params); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA hardware probe error (%s): %s.", - info->name, snd_strerror(err)); - error(RtError::WARNING); - goto capture_probe; - } - - // Get output channel information. - info->minOutputChannels = snd_pcm_hw_params_get_channels_min(params); - info->maxOutputChannels = snd_pcm_hw_params_get_channels_max(params); - - snd_pcm_close(handle); - - capture_probe: - // Now try for capture - stream = SND_PCM_STREAM_CAPTURE; - err = snd_pcm_open(&handle, info->name, stream, open_mode); - if (err < 0) { - sprintf(message, "RtAudio: ALSA pcm capture open (%s): %s.", - info->name, snd_strerror(err)); - error(RtError::WARNING); - if (info->maxOutputChannels == 0) - // didn't open for playback either ... device invalid - return; - goto probe_parameters; - } - - // We have an open capture device ... allocate the parameter structure. - err = snd_pcm_hw_params_any(handle, params); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA hardware probe error (%s): %s.", - info->name, snd_strerror(err)); - error(RtError::WARNING); - if (info->maxOutputChannels > 0) - goto probe_parameters; - else - return; - } - - // Get input channel information. - info->minInputChannels = snd_pcm_hw_params_get_channels_min(params); - info->maxInputChannels = snd_pcm_hw_params_get_channels_max(params); - - // If device opens for both playback and capture, we determine the channels. - if (info->maxOutputChannels == 0 || info->maxInputChannels == 0) - goto probe_parameters; - - info->hasDuplexSupport = true; - info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ? - info->maxInputChannels : info->maxOutputChannels; - info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ? - info->minInputChannels : info->minOutputChannels; - - snd_pcm_close(handle); - - probe_parameters: - // At this point, we just need to figure out the supported data - // formats and sample rates. We'll proceed by opening the device in - // the direction with the maximum number of channels, or playback if - // they are equal. This might limit our sample rate options, but so - // be it. - - if (info->maxOutputChannels >= info->maxInputChannels) - stream = SND_PCM_STREAM_PLAYBACK; - else - stream = SND_PCM_STREAM_CAPTURE; - - err = snd_pcm_open(&handle, info->name, stream, open_mode); - if (err < 0) { - sprintf(message, "RtAudio: ALSA pcm (%s) won't reopen during probe: %s.", - info->name, snd_strerror(err)); - error(RtError::WARNING); - return; - } - - // We have an open device ... allocate the parameter structure. - err = snd_pcm_hw_params_any(handle, params); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA hardware reopen probe error (%s): %s.", - info->name, snd_strerror(err)); - error(RtError::WARNING); - return; - } - - // Test a non-standard sample rate to see if continuous rate is supported. - int dir = 0; - if (snd_pcm_hw_params_test_rate(handle, params, 35500, dir) == 0) { - // It appears that continuous sample rate support is available. - info->nSampleRates = -1; - info->sampleRates[0] = snd_pcm_hw_params_get_rate_min(params, &dir); - info->sampleRates[1] = snd_pcm_hw_params_get_rate_max(params, &dir); - } - else { - // No continuous rate support ... test our discrete set of sample rate values. - info->nSampleRates = 0; - for (int i=0; isampleRates[info->nSampleRates] = SAMPLE_RATES[i]; - info->nSampleRates++; - } - } - if (info->nSampleRates == 0) { - snd_pcm_close(handle); - return; - } - } - - // Probe the supported data formats ... we don't care about endian-ness just yet - snd_pcm_format_t format; - info->nativeFormats = 0; - format = SND_PCM_FORMAT_S8; - if (snd_pcm_hw_params_test_format(handle, params, format) == 0) - info->nativeFormats |= RTAUDIO_SINT8; - format = SND_PCM_FORMAT_S16; - if (snd_pcm_hw_params_test_format(handle, params, format) == 0) - info->nativeFormats |= RTAUDIO_SINT16; - format = SND_PCM_FORMAT_S24; - if (snd_pcm_hw_params_test_format(handle, params, format) == 0) - info->nativeFormats |= RTAUDIO_SINT24; - format = SND_PCM_FORMAT_S32; - if (snd_pcm_hw_params_test_format(handle, params, format) == 0) - info->nativeFormats |= RTAUDIO_SINT32; - format = SND_PCM_FORMAT_FLOAT; - if (snd_pcm_hw_params_test_format(handle, params, format) == 0) - info->nativeFormats |= RTAUDIO_FLOAT32; - format = SND_PCM_FORMAT_FLOAT64; - if (snd_pcm_hw_params_test_format(handle, params, format) == 0) - info->nativeFormats |= RTAUDIO_FLOAT64; - - // Check that we have at least one supported format - if (info->nativeFormats == 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA PCM device (%s) data format not supported by RtAudio.", - info->name); - error(RtError::WARNING); - return; - } - - // That's all ... close the device and return - snd_pcm_close(handle); - info->probed = true; - return; -} - -bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, - STREAM_MODE mode, int channels, - int sampleRate, RTAUDIO_FORMAT format, - int *bufferSize, int numberOfBuffers) -{ -#if defined(RTAUDIO_DEBUG) - snd_output_t *out; - snd_output_stdio_attach(&out, stderr, 0); -#endif - - // I'm not using the "plug" interface ... too much inconsistent behavior. - const char *name = devices[device].name; - - snd_pcm_stream_t alsa_stream; - if (mode == PLAYBACK) - alsa_stream = SND_PCM_STREAM_PLAYBACK; - else - alsa_stream = SND_PCM_STREAM_CAPTURE; - - int err; - snd_pcm_t *handle; - int alsa_open_mode = SND_PCM_ASYNC; - err = snd_pcm_open(&handle, name, alsa_stream, alsa_open_mode); - if (err < 0) { - sprintf(message,"RtAudio: ALSA pcm device (%s) won't open: %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } - - // Fill the parameter structure. - snd_pcm_hw_params_t *hw_params; - snd_pcm_hw_params_alloca(&hw_params); - err = snd_pcm_hw_params_any(handle, hw_params); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error getting parameter handle (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } - -#if defined(RTAUDIO_DEBUG) - fprintf(stderr, "\nRtAudio: ALSA dump hardware params just after device open:\n\n"); - snd_pcm_hw_params_dump(hw_params, out); -#endif - - // Set access ... try interleaved access first, then non-interleaved - err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); - if (err < 0) { - // No interleave support ... try non-interleave. - err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error setting access ( (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } - stream->deInterleave[mode] = true; - } - - // Determine how to set the device format. - stream->userFormat = format; - snd_pcm_format_t device_format; - - if (format == RTAUDIO_SINT8) - device_format = SND_PCM_FORMAT_S8; - else if (format == RTAUDIO_SINT16) - device_format = SND_PCM_FORMAT_S16; - else if (format == RTAUDIO_SINT24) - device_format = SND_PCM_FORMAT_S24; - else if (format == RTAUDIO_SINT32) - device_format = SND_PCM_FORMAT_S32; - else if (format == RTAUDIO_FLOAT32) - device_format = SND_PCM_FORMAT_FLOAT; - else if (format == RTAUDIO_FLOAT64) - device_format = SND_PCM_FORMAT_FLOAT64; - - if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { - stream->deviceFormat[mode] = format; - goto set_format; - } - - // The user requested format is not natively supported by the device. - device_format = SND_PCM_FORMAT_FLOAT64; - if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { - stream->deviceFormat[mode] = RTAUDIO_FLOAT64; - goto set_format; - } - - device_format = SND_PCM_FORMAT_FLOAT; - if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { - stream->deviceFormat[mode] = RTAUDIO_FLOAT32; - goto set_format; - } - - device_format = SND_PCM_FORMAT_S32; - if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { - stream->deviceFormat[mode] = RTAUDIO_SINT32; - goto set_format; - } - - device_format = SND_PCM_FORMAT_S24; - if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { - stream->deviceFormat[mode] = RTAUDIO_SINT24; - goto set_format; - } - - device_format = SND_PCM_FORMAT_S16; - if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { - stream->deviceFormat[mode] = RTAUDIO_SINT16; - goto set_format; - } - - device_format = SND_PCM_FORMAT_S8; - if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { - stream->deviceFormat[mode] = RTAUDIO_SINT8; - goto set_format; - } - - // If we get here, no supported format was found. - sprintf(message,"RtAudio: ALSA pcm device (%s) data format not supported by RtAudio.", name); - snd_pcm_close(handle); - error(RtError::WARNING); - return FAILURE; - - set_format: - err = snd_pcm_hw_params_set_format(handle, hw_params, device_format); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error setting format (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } - - // Determine whether byte-swaping is necessary. - stream->doByteSwap[mode] = false; - if (device_format != SND_PCM_FORMAT_S8) { - err = snd_pcm_format_cpu_endian(device_format); - if (err == 0) - stream->doByteSwap[mode] = true; - else if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error getting format endian-ness (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } - } - - // Determine the number of channels for this device. We support a possible - // minimum device channel number > than the value requested by the user. - stream->nUserChannels[mode] = channels; - int device_channels = snd_pcm_hw_params_get_channels_max(hw_params); - if (device_channels < channels) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: channels (%d) not supported by device (%s).", - channels, name); - error(RtError::WARNING); - return FAILURE; - } - - device_channels = snd_pcm_hw_params_get_channels_min(hw_params); - if (device_channels < channels) device_channels = channels; - stream->nDeviceChannels[mode] = device_channels; - - // Set the device channels. - err = snd_pcm_hw_params_set_channels(handle, hw_params, device_channels); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error setting channels (%d) on device (%s): %s.", - device_channels, name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } - - // Set the sample rate. - err = snd_pcm_hw_params_set_rate(handle, hw_params, (unsigned int)sampleRate, 0); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error setting sample rate (%d) on device (%s): %s.", - sampleRate, name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } - - // Set the buffer number, which in ALSA is referred to as the "period". - int dir; - int periods = numberOfBuffers; - // Even though the hardware might allow 1 buffer, it won't work reliably. - if (periods < 2) periods = 2; - err = snd_pcm_hw_params_get_periods_min(hw_params, &dir); - if (err > periods) periods = err; - - err = snd_pcm_hw_params_set_periods(handle, hw_params, periods, 0); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error setting periods (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } - - // Set the buffer (or period) size. - err = snd_pcm_hw_params_get_period_size_min(hw_params, &dir); - if (err > *bufferSize) *bufferSize = err; - - err = snd_pcm_hw_params_set_period_size(handle, hw_params, *bufferSize, 0); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error setting period size (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } - - stream->bufferSize = *bufferSize; - - // Install the hardware configuration - err = snd_pcm_hw_params(handle, hw_params); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error installing hardware configuration (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } - -#if defined(RTAUDIO_DEBUG) - fprintf(stderr, "\nRtAudio: ALSA dump hardware params after installation:\n\n"); - snd_pcm_hw_params_dump(hw_params, out); -#endif - - /* - // Install the software configuration - snd_pcm_sw_params_t *sw_params = NULL; - snd_pcm_sw_params_alloca(&sw_params); - snd_pcm_sw_params_current(handle, sw_params); - err = snd_pcm_sw_params(handle, sw_params); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error installing software configuration (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } - */ - - // Set handle and flags for buffer conversion - stream->handle[mode] = handle; - stream->doConvertBuffer[mode] = false; - if (stream->userFormat != stream->deviceFormat[mode]) - stream->doConvertBuffer[mode] = true; - if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode]) - stream->doConvertBuffer[mode] = true; - if (stream->nUserChannels[mode] > 1 && stream->deInterleave[mode]) - stream->doConvertBuffer[mode] = true; - - // Allocate necessary internal buffers - if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { - - long buffer_bytes; - if (stream->nUserChannels[0] >= stream->nUserChannels[1]) - buffer_bytes = stream->nUserChannels[0]; - else - buffer_bytes = stream->nUserChannels[1]; - - buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); - if (stream->userBuffer) free(stream->userBuffer); - stream->userBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->userBuffer == NULL) - goto memory_error; - } - - if ( stream->doConvertBuffer[mode] ) { - - long buffer_bytes; - bool makeBuffer = true; - if ( mode == PLAYBACK ) - buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - else { // mode == RECORD - buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); - if ( stream->mode == PLAYBACK ) { - long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - if ( buffer_bytes > bytes_out ) - buffer_bytes = (buffer_bytes > bytes_out) ? buffer_bytes : bytes_out; - else - makeBuffer = false; - } - } - - if ( makeBuffer ) { - buffer_bytes *= *bufferSize; - if (stream->deviceBuffer) free(stream->deviceBuffer); - stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->deviceBuffer == NULL) - goto memory_error; - } - } - - stream->device[mode] = device; - stream->state = STREAM_STOPPED; - if ( stream->mode == PLAYBACK && mode == RECORD ) - // We had already set up an output stream. - stream->mode = DUPLEX; - else - stream->mode = mode; - stream->nBuffers = periods; - stream->sampleRate = sampleRate; - - return SUCCESS; - - memory_error: - if (stream->handle[0]) { - snd_pcm_close(stream->handle[0]); - stream->handle[0] = 0; - } - if (stream->handle[1]) { - snd_pcm_close(stream->handle[1]); - stream->handle[1] = 0; - } - if (stream->userBuffer) { - free(stream->userBuffer); - stream->userBuffer = 0; - } - sprintf(message, "RtAudio: ALSA error allocating buffer memory (%s).", name); - error(RtError::WARNING); - return FAILURE; -} - -void RtAudio :: cancelStreamCallback(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - if (stream->usingCallback) { - stream->usingCallback = false; - pthread_cancel(stream->thread); - pthread_join(stream->thread, NULL); - stream->thread = 0; - stream->callback = NULL; - stream->userData = NULL; - } -} - -void RtAudio :: closeStream(int streamId) -{ - // We don't want an exception to be thrown here because this - // function is called by our class destructor. So, do our own - // streamId check. - if ( streams.find( streamId ) == streams.end() ) { - sprintf(message, "RtAudio: invalid stream identifier!"); - error(RtError::WARNING); - return; - } - - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; - - if (stream->usingCallback) { - pthread_cancel(stream->thread); - pthread_join(stream->thread, NULL); - } - - if (stream->state == STREAM_RUNNING) { - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) - snd_pcm_drop(stream->handle[0]); - if (stream->mode == RECORD || stream->mode == DUPLEX) - snd_pcm_drop(stream->handle[1]); - } - - pthread_mutex_destroy(&stream->mutex); - - if (stream->handle[0]) - snd_pcm_close(stream->handle[0]); - - if (stream->handle[1]) - snd_pcm_close(stream->handle[1]); - - if (stream->userBuffer) - free(stream->userBuffer); - - if (stream->deviceBuffer) - free(stream->deviceBuffer); - - free(stream); - streams.erase(streamId); -} - -void RtAudio :: startStream(int streamId) -{ - // This method calls snd_pcm_prepare if the device isn't already in that state. - - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - if (stream->state == STREAM_RUNNING) - goto unlock; - - int err; - snd_pcm_state_t state; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - state = snd_pcm_state(stream->handle[0]); - if (state != SND_PCM_STATE_PREPARED) { - err = snd_pcm_prepare(stream->handle[0]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error preparing pcm device (%s): %s.", - devices[stream->device[0]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - } - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - state = snd_pcm_state(stream->handle[1]); - if (state != SND_PCM_STATE_PREPARED) { - err = snd_pcm_prepare(stream->handle[1]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error preparing pcm device (%s): %s.", - devices[stream->device[1]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - } - stream->state = STREAM_RUNNING; - - unlock: - MUTEX_UNLOCK(&stream->mutex); -} - -void RtAudio :: stopStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - if (stream->state == STREAM_STOPPED) - goto unlock; - - int err; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - err = snd_pcm_drain(stream->handle[0]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", - devices[stream->device[0]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - err = snd_pcm_drain(stream->handle[1]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", - devices[stream->device[1]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - stream->state = STREAM_STOPPED; - - unlock: - MUTEX_UNLOCK(&stream->mutex); -} - -void RtAudio :: abortStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - if (stream->state == STREAM_STOPPED) - goto unlock; - - int err; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - err = snd_pcm_drop(stream->handle[0]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", - devices[stream->device[0]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - err = snd_pcm_drop(stream->handle[1]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", - devices[stream->device[1]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - stream->state = STREAM_STOPPED; - - unlock: - MUTEX_UNLOCK(&stream->mutex); -} - -int RtAudio :: streamWillBlock(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - int err = 0, frames = 0; - if (stream->state == STREAM_STOPPED) - goto unlock; - - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - err = snd_pcm_avail_update(stream->handle[0]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error getting available frames for device (%s): %s.", - devices[stream->device[0]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - - frames = err; - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - err = snd_pcm_avail_update(stream->handle[1]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error getting available frames for device (%s): %s.", - devices[stream->device[1]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - if (frames > err) frames = err; - } - - frames = stream->bufferSize - frames; - if (frames < 0) frames = 0; - - unlock: - MUTEX_UNLOCK(&stream->mutex); - return frames; -} - -void RtAudio :: tickStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - int stopStream = 0; - if (stream->state == STREAM_STOPPED) { - if (stream->usingCallback) usleep(50000); // sleep 50 milliseconds - return; - } - else if (stream->usingCallback) { - stopStream = stream->callback(stream->userBuffer, stream->bufferSize, stream->userData); - } - - MUTEX_LOCK(&stream->mutex); - - // The state might change while waiting on a mutex. - if (stream->state == STREAM_STOPPED) - goto unlock; - - int err; - char *buffer; - int channels; - RTAUDIO_FORMAT format; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - - // Setup parameters and do buffer conversion if necessary. - if (stream->doConvertBuffer[0]) { - convertStreamBuffer(stream, PLAYBACK); - buffer = stream->deviceBuffer; - channels = stream->nDeviceChannels[0]; - format = stream->deviceFormat[0]; - } - else { - buffer = stream->userBuffer; - channels = stream->nUserChannels[0]; - format = stream->userFormat; - } - - // Do byte swapping if necessary. - if (stream->doByteSwap[0]) - byteSwapBuffer(buffer, stream->bufferSize * channels, format); - - // Write samples to device in interleaved/non-interleaved format. - if (stream->deInterleave[0]) { - void *bufs[channels]; - size_t offset = stream->bufferSize * formatBytes(format); - for (int i=0; ihandle[0], bufs, stream->bufferSize); - } - else - err = snd_pcm_writei(stream->handle[0], buffer, stream->bufferSize); - - if (err < stream->bufferSize) { - // Either an error or underrun occured. - if (err == -EPIPE) { - snd_pcm_state_t state = snd_pcm_state(stream->handle[0]); - if (state == SND_PCM_STATE_XRUN) { - sprintf(message, "RtAudio: ALSA underrun detected."); - error(RtError::WARNING); - err = snd_pcm_prepare(stream->handle[0]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error preparing handle after underrun: %s.", - snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - else { - sprintf(message, "RtAudio: ALSA error, current state is %s.", - snd_pcm_state_name(state)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - goto unlock; - } - else { - sprintf(message, "RtAudio: ALSA audio write error for device (%s): %s.", - devices[stream->device[0]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - } - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - - // Setup parameters. - if (stream->doConvertBuffer[1]) { - buffer = stream->deviceBuffer; - channels = stream->nDeviceChannels[1]; - format = stream->deviceFormat[1]; - } - else { - buffer = stream->userBuffer; - channels = stream->nUserChannels[1]; - format = stream->userFormat; - } - - // Read samples from device in interleaved/non-interleaved format. - if (stream->deInterleave[1]) { - void *bufs[channels]; - size_t offset = stream->bufferSize * formatBytes(format); - for (int i=0; ihandle[1], bufs, stream->bufferSize); - } - else - err = snd_pcm_readi(stream->handle[1], buffer, stream->bufferSize); - - if (err < stream->bufferSize) { - // Either an error or underrun occured. - if (err == -EPIPE) { - snd_pcm_state_t state = snd_pcm_state(stream->handle[1]); - if (state == SND_PCM_STATE_XRUN) { - sprintf(message, "RtAudio: ALSA overrun detected."); - error(RtError::WARNING); - err = snd_pcm_prepare(stream->handle[1]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error preparing handle after overrun: %s.", - snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - else { - sprintf(message, "RtAudio: ALSA error, current state is %s.", - snd_pcm_state_name(state)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - goto unlock; - } - else { - sprintf(message, "RtAudio: ALSA audio read error for device (%s): %s.", - devices[stream->device[1]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - - // Do byte swapping if necessary. - if (stream->doByteSwap[1]) - byteSwapBuffer(buffer, stream->bufferSize * channels, format); - - // Do buffer conversion if necessary. - if (stream->doConvertBuffer[1]) - convertStreamBuffer(stream, RECORD); - } - - unlock: - MUTEX_UNLOCK(&stream->mutex); - - if (stream->usingCallback && stopStream) - this->stopStream(streamId); -} - -extern "C" void *callbackHandler(void *ptr) -{ - RtAudio *object = thread_info.object; - int stream = thread_info.streamId; - bool *usingCallback = (bool *) ptr; - - while ( *usingCallback ) { - pthread_testcancel(); - try { - object->tickStream(stream); - } - catch (RtError &exception) { - fprintf(stderr, "\nCallback thread error (%s) ... closing thread.\n\n", - exception.getMessage()); - break; - } - } - - return 0; -} - -//******************** End of __LINUX_ALSA__ *********************// - -#elif defined(__LINUX_OSS__) - -#include -#include -#include -#include -#include -#include -#include -#include - -#define DAC_NAME "/dev/dsp" -#define MAX_DEVICES 16 -#define MAX_CHANNELS 16 - -void RtAudio :: initialize(void) -{ - // Count cards and devices - nDevices = 0; - - // We check /dev/dsp before probing devices. /dev/dsp is supposed to - // be a link to the "default" audio device, of the form /dev/dsp0, - // /dev/dsp1, etc... However, I've seen one case where /dev/dsp was a - // real device, so we need to check for that. Also, sometimes the - // link is to /dev/dspx and other times just dspx. I'm not sure how - // the latter works, but it does. - char device_name[16]; - struct stat dspstat; - int dsplink = -1; - int i = 0; - if (lstat(DAC_NAME, &dspstat) == 0) { - if (S_ISLNK(dspstat.st_mode)) { - i = readlink(DAC_NAME, device_name, sizeof(device_name)); - if (i > 0) { - device_name[i] = '\0'; - if (i > 8) { // check for "/dev/dspx" - if (!strncmp(DAC_NAME, device_name, 8)) - dsplink = atoi(&device_name[8]); - } - else if (i > 3) { // check for "dspx" - if (!strncmp("dsp", device_name, 3)) - dsplink = atoi(&device_name[3]); - } - } - else { - sprintf(message, "RtAudio: cannot read value of symbolic link %s.", DAC_NAME); - error(RtError::SYSTEM_ERROR); - } - } - } - else { - sprintf(message, "RtAudio: cannot stat %s.", DAC_NAME); - error(RtError::SYSTEM_ERROR); - } - - // The OSS API doesn't provide a routine for determining the number - // of devices. Thus, we'll just pursue a brute force method. The - // idea is to start with /dev/dsp(0) and continue with higher device - // numbers until we reach MAX_DSP_DEVICES. This should tell us how - // many devices we have ... it is not a fullproof scheme, but hopefully - // it will work most of the time. - - int fd = 0; - char names[MAX_DEVICES][16]; - for (i=-1; i= 0) close(fd); - strncpy(names[nDevices], device_name, 16); - nDevices++; - } - - if (nDevices == 0) return; - - // Allocate the RTAUDIO_DEVICE structures. - devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); - if (devices == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::MEMORY_ERROR); - } - - // Write device ascii identifiers to device control structure and then probe capabilities. - for (i=0; iname, O_WRONLY | O_NONBLOCK); - if (fd == -1) { - // Open device failed ... either busy or doesn't exist - if (errno == EBUSY || errno == EAGAIN) - sprintf(message, "RtAudio: OSS playback device (%s) is busy and cannot be probed.", - info->name); - else - sprintf(message, "RtAudio: OSS playback device (%s) open error.", info->name); - error(RtError::WARNING); - goto capture_probe; - } - - // We have an open device ... see how many channels it can handle - for (i=MAX_CHANNELS; i>0; i--) { - channels = i; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) { - // This would normally indicate some sort of hardware error, but under ALSA's - // OSS emulation, it sometimes indicates an invalid channel value. Further, - // the returned channel value is not changed. So, we'll ignore the possible - // hardware error. - continue; // try next channel number - } - // Check to see whether the device supports the requested number of channels - if (channels != i ) continue; // try next channel number - // If here, we found the largest working channel value - break; - } - info->maxOutputChannels = channels; - - // Now find the minimum number of channels it can handle - for (i=1; i<=info->maxOutputChannels; i++) { - channels = i; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) - continue; // try next channel number - // If here, we found the smallest working channel value - break; - } - info->minOutputChannels = channels; - close(fd); - - capture_probe: - // Now try for capture - fd = open(info->name, O_RDONLY | O_NONBLOCK); - if (fd == -1) { - // Open device for capture failed ... either busy or doesn't exist - if (errno == EBUSY || errno == EAGAIN) - sprintf(message, "RtAudio: OSS capture device (%s) is busy and cannot be probed.", - info->name); - else - sprintf(message, "RtAudio: OSS capture device (%s) open error.", info->name); - error(RtError::WARNING); - if (info->maxOutputChannels == 0) - // didn't open for playback either ... device invalid - return; - goto probe_parameters; - } - - // We have the device open for capture ... see how many channels it can handle - for (i=MAX_CHANNELS; i>0; i--) { - channels = i; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) { - continue; // as above - } - // If here, we found a working channel value - break; - } - info->maxInputChannels = channels; - - // Now find the minimum number of channels it can handle - for (i=1; i<=info->maxInputChannels; i++) { - channels = i; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) - continue; // try next channel number - // If here, we found the smallest working channel value - break; - } - info->minInputChannels = channels; - close(fd); - - // If device opens for both playback and capture, we determine the channels. - if (info->maxOutputChannels == 0 || info->maxInputChannels == 0) - goto probe_parameters; - - fd = open(info->name, O_RDWR | O_NONBLOCK); - if (fd == -1) - goto probe_parameters; - - ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0); - ioctl(fd, SNDCTL_DSP_GETCAPS, &mask); - if (mask & DSP_CAP_DUPLEX) { - info->hasDuplexSupport = true; - // We have the device open for duplex ... see how many channels it can handle - for (i=MAX_CHANNELS; i>0; i--) { - channels = i; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) - continue; // as above - // If here, we found a working channel value - break; - } - info->maxDuplexChannels = channels; - - // Now find the minimum number of channels it can handle - for (i=1; i<=info->maxDuplexChannels; i++) { - channels = i; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) - continue; // try next channel number - // If here, we found the smallest working channel value - break; - } - info->minDuplexChannels = channels; - } - close(fd); - - probe_parameters: - // At this point, we need to figure out the supported data formats - // and sample rates. We'll proceed by openning the device in the - // direction with the maximum number of channels, or playback if - // they are equal. This might limit our sample rate options, but so - // be it. - - if (info->maxOutputChannels >= info->maxInputChannels) { - fd = open(info->name, O_WRONLY | O_NONBLOCK); - channels = info->maxOutputChannels; - } - else { - fd = open(info->name, O_RDONLY | O_NONBLOCK); - channels = info->maxInputChannels; - } - - if (fd == -1) { - // We've got some sort of conflict ... abort - sprintf(message, "RtAudio: OSS device (%s) won't reopen during probe.", - info->name); - error(RtError::WARNING); - return; - } - - // We have an open device ... set to maximum channels. - i = channels; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) { - // We've got some sort of conflict ... abort - close(fd); - sprintf(message, "RtAudio: OSS device (%s) won't revert to previous channel setting.", - info->name); - error(RtError::WARNING); - return; - } - - if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) { - close(fd); - sprintf(message, "RtAudio: OSS device (%s) can't get supported audio formats.", - info->name); - error(RtError::WARNING); - return; - } - - // Probe the supported data formats ... we don't care about endian-ness just yet. - int format; - info->nativeFormats = 0; -#if defined (AFMT_S32_BE) - // This format does not seem to be in the 2.4 kernel version of OSS soundcard.h - if (mask & AFMT_S32_BE) { - format = AFMT_S32_BE; - info->nativeFormats |= RTAUDIO_SINT32; - } -#endif -#if defined (AFMT_S32_LE) - /* This format is not in the 2.4.4 kernel version of OSS soundcard.h */ - if (mask & AFMT_S32_LE) { - format = AFMT_S32_LE; - info->nativeFormats |= RTAUDIO_SINT32; - } -#endif - if (mask & AFMT_S8) { - format = AFMT_S8; - info->nativeFormats |= RTAUDIO_SINT8; - } - if (mask & AFMT_S16_BE) { - format = AFMT_S16_BE; - info->nativeFormats |= RTAUDIO_SINT16; - } - if (mask & AFMT_S16_LE) { - format = AFMT_S16_LE; - info->nativeFormats |= RTAUDIO_SINT16; - } - - // Check that we have at least one supported format - if (info->nativeFormats == 0) { - close(fd); - sprintf(message, "RtAudio: OSS device (%s) data format not supported by RtAudio.", - info->name); - error(RtError::WARNING); - return; - } - - // Set the format - i = format; - if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1 || format != i) { - close(fd); - sprintf(message, "RtAudio: OSS device (%s) error setting data format.", - info->name); - error(RtError::WARNING); - return; - } - - // Probe the supported sample rates ... first get lower limit - int speed = 1; - if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { - // If we get here, we're probably using an ALSA driver with OSS-emulation, - // which doesn't conform to the OSS specification. In this case, - // we'll probe our predefined list of sample rates for working values. - info->nSampleRates = 0; - for (i=0; isampleRates[info->nSampleRates] = SAMPLE_RATES[i]; - info->nSampleRates++; - } - } - if (info->nSampleRates == 0) { - close(fd); - return; - } - goto finished; - } - info->sampleRates[0] = speed; - - // Now get upper limit - speed = 1000000; - if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { - close(fd); - sprintf(message, "RtAudio: OSS device (%s) error setting sample rate.", - info->name); - error(RtError::WARNING); - return; - } - info->sampleRates[1] = speed; - info->nSampleRates = -1; - - finished: // That's all ... close the device and return - close(fd); - info->probed = true; - return; -} - -bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, - STREAM_MODE mode, int channels, - int sampleRate, RTAUDIO_FORMAT format, - int *bufferSize, int numberOfBuffers) -{ - int buffers, buffer_bytes, device_channels, device_format; - int srate, temp, fd; - - const char *name = devices[device].name; - - if (mode == PLAYBACK) - fd = open(name, O_WRONLY | O_NONBLOCK); - else { // mode == RECORD - if (stream->mode == PLAYBACK && stream->device[0] == device) { - // We just set the same device for playback ... close and reopen for duplex (OSS only). - close(stream->handle[0]); - stream->handle[0] = 0; - // First check that the number previously set channels is the same. - if (stream->nUserChannels[0] != channels) { - sprintf(message, "RtAudio: input/output channels must be equal for OSS duplex device (%s).", name); - goto error; - } - fd = open(name, O_RDWR | O_NONBLOCK); - } - else - fd = open(name, O_RDONLY | O_NONBLOCK); - } - - if (fd == -1) { - if (errno == EBUSY || errno == EAGAIN) - sprintf(message, "RtAudio: OSS device (%s) is busy and cannot be opened.", - name); - else - sprintf(message, "RtAudio: OSS device (%s) cannot be opened.", name); - goto error; - } - - // Now reopen in blocking mode. - close(fd); - if (mode == PLAYBACK) - fd = open(name, O_WRONLY | O_SYNC); - else { // mode == RECORD - if (stream->mode == PLAYBACK && stream->device[0] == device) - fd = open(name, O_RDWR | O_SYNC); - else - fd = open(name, O_RDONLY | O_SYNC); - } - - if (fd == -1) { - sprintf(message, "RtAudio: OSS device (%s) cannot be opened.", name); - goto error; - } - - // Get the sample format mask - int mask; - if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) { - close(fd); - sprintf(message, "RtAudio: OSS device (%s) can't get supported audio formats.", - name); - goto error; - } - - // Determine how to set the device format. - stream->userFormat = format; - device_format = -1; - stream->doByteSwap[mode] = false; - if (format == RTAUDIO_SINT8) { - if (mask & AFMT_S8) { - device_format = AFMT_S8; - stream->deviceFormat[mode] = RTAUDIO_SINT8; - } - } - else if (format == RTAUDIO_SINT16) { - if (mask & AFMT_S16_NE) { - device_format = AFMT_S16_NE; - stream->deviceFormat[mode] = RTAUDIO_SINT16; - } -#if BYTE_ORDER == LITTLE_ENDIAN - else if (mask & AFMT_S16_BE) { - device_format = AFMT_S16_BE; - stream->deviceFormat[mode] = RTAUDIO_SINT16; - stream->doByteSwap[mode] = true; - } -#else - else if (mask & AFMT_S16_LE) { - device_format = AFMT_S16_LE; - stream->deviceFormat[mode] = RTAUDIO_SINT16; - stream->doByteSwap[mode] = true; - } -#endif - } -#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE) - else if (format == RTAUDIO_SINT32) { - if (mask & AFMT_S32_NE) { - device_format = AFMT_S32_NE; - stream->deviceFormat[mode] = RTAUDIO_SINT32; - } -#if BYTE_ORDER == LITTLE_ENDIAN - else if (mask & AFMT_S32_BE) { - device_format = AFMT_S32_BE; - stream->deviceFormat[mode] = RTAUDIO_SINT32; - stream->doByteSwap[mode] = true; - } -#else - else if (mask & AFMT_S32_LE) { - device_format = AFMT_S32_LE; - stream->deviceFormat[mode] = RTAUDIO_SINT32; - stream->doByteSwap[mode] = true; - } -#endif - } -#endif - - if (device_format == -1) { - // The user requested format is not natively supported by the device. - if (mask & AFMT_S16_NE) { - device_format = AFMT_S16_NE; - stream->deviceFormat[mode] = RTAUDIO_SINT16; - } -#if BYTE_ORDER == LITTLE_ENDIAN - else if (mask & AFMT_S16_BE) { - device_format = AFMT_S16_BE; - stream->deviceFormat[mode] = RTAUDIO_SINT16; - stream->doByteSwap[mode] = true; - } -#else - else if (mask & AFMT_S16_LE) { - device_format = AFMT_S16_LE; - stream->deviceFormat[mode] = RTAUDIO_SINT16; - stream->doByteSwap[mode] = true; - } -#endif -#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE) - else if (mask & AFMT_S32_NE) { - device_format = AFMT_S32_NE; - stream->deviceFormat[mode] = RTAUDIO_SINT32; - } -#if BYTE_ORDER == LITTLE_ENDIAN - else if (mask & AFMT_S32_BE) { - device_format = AFMT_S32_BE; - stream->deviceFormat[mode] = RTAUDIO_SINT32; - stream->doByteSwap[mode] = true; - } -#else - else if (mask & AFMT_S32_LE) { - device_format = AFMT_S32_LE; - stream->deviceFormat[mode] = RTAUDIO_SINT32; - stream->doByteSwap[mode] = true; - } -#endif -#endif - else if (mask & AFMT_S8) { - device_format = AFMT_S8; - stream->deviceFormat[mode] = RTAUDIO_SINT8; - } - } - - if (stream->deviceFormat[mode] == 0) { - // This really shouldn't happen ... - close(fd); - sprintf(message, "RtAudio: OSS device (%s) data format not supported by RtAudio.", - name); - goto error; - } - - // Determine the number of channels for this device. Note that the - // channel value requested by the user might be < min_X_Channels. - stream->nUserChannels[mode] = channels; - device_channels = channels; - if (mode == PLAYBACK) { - if (channels < devices[device].minOutputChannels) - device_channels = devices[device].minOutputChannels; - } - else { // mode == RECORD - if (stream->mode == PLAYBACK && stream->device[0] == device) { - // We're doing duplex setup here. - if (channels < devices[device].minDuplexChannels) - device_channels = devices[device].minDuplexChannels; - } - else { - if (channels < devices[device].minInputChannels) - device_channels = devices[device].minInputChannels; - } - } - stream->nDeviceChannels[mode] = device_channels; - - // Attempt to set the buffer size. According to OSS, the minimum - // number of buffers is two. The supposed minimum buffer size is 16 - // bytes, so that will be our lower bound. The argument to this - // call is in the form 0xMMMMSSSS (hex), where the buffer size (in - // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. - // We'll check the actual value used near the end of the setup - // procedure. - buffer_bytes = *bufferSize * formatBytes(stream->deviceFormat[mode]) * device_channels; - if (buffer_bytes < 16) buffer_bytes = 16; - buffers = numberOfBuffers; - if (buffers < 2) buffers = 2; - temp = ((int) buffers << 16) + (int)(log10((double)buffer_bytes)/log10(2.0)); - if (ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &temp)) { - close(fd); - sprintf(message, "RtAudio: OSS error setting fragment size for device (%s).", - name); - goto error; - } - stream->nBuffers = buffers; - - // Set the data format. - temp = device_format; - if (ioctl(fd, SNDCTL_DSP_SETFMT, &device_format) == -1 || device_format != temp) { - close(fd); - sprintf(message, "RtAudio: OSS error setting data format for device (%s).", - name); - goto error; - } - - // Set the number of channels. - temp = device_channels; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &device_channels) == -1 || device_channels != temp) { - close(fd); - sprintf(message, "RtAudio: OSS error setting %d channels on device (%s).", - temp, name); - goto error; - } - - // Set the sample rate. - srate = sampleRate; - temp = srate; - if (ioctl(fd, SNDCTL_DSP_SPEED, &srate) == -1) { - close(fd); - sprintf(message, "RtAudio: OSS error setting sample rate = %d on device (%s).", - temp, name); - goto error; - } - - // Verify the sample rate setup worked. - if (abs(srate - temp) > 100) { - close(fd); - sprintf(message, "RtAudio: OSS error ... audio device (%s) doesn't support sample rate of %d.", - name, temp); - goto error; - } - stream->sampleRate = sampleRate; - - if (ioctl(fd, SNDCTL_DSP_GETBLKSIZE, &buffer_bytes) == -1) { - close(fd); - sprintf(message, "RtAudio: OSS error getting buffer size for device (%s).", - name); - goto error; - } - - // Save buffer size (in sample frames). - *bufferSize = buffer_bytes / (formatBytes(stream->deviceFormat[mode]) * device_channels); - stream->bufferSize = *bufferSize; - - if (mode == RECORD && stream->mode == PLAYBACK && - stream->device[0] == device) { - // We're doing duplex setup here. - stream->deviceFormat[0] = stream->deviceFormat[1]; - stream->nDeviceChannels[0] = device_channels; - } - - // Set flags for buffer conversion - stream->doConvertBuffer[mode] = false; - if (stream->userFormat != stream->deviceFormat[mode]) - stream->doConvertBuffer[mode] = true; - if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode]) - stream->doConvertBuffer[mode] = true; - - // Allocate necessary internal buffers - if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { - - long buffer_bytes; - if (stream->nUserChannels[0] >= stream->nUserChannels[1]) - buffer_bytes = stream->nUserChannels[0]; - else - buffer_bytes = stream->nUserChannels[1]; - - buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); - if (stream->userBuffer) free(stream->userBuffer); - stream->userBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->userBuffer == NULL) { - close(fd); - sprintf(message, "RtAudio: OSS error allocating user buffer memory (%s).", - name); - goto error; - } - } - - if ( stream->doConvertBuffer[mode] ) { - - long buffer_bytes; - bool makeBuffer = true; - if ( mode == PLAYBACK ) - buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - else { // mode == RECORD - buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); - if ( stream->mode == PLAYBACK ) { - long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - if ( buffer_bytes > bytes_out ) - buffer_bytes = (buffer_bytes > bytes_out) ? buffer_bytes : bytes_out; - else - makeBuffer = false; - } - } - - if ( makeBuffer ) { - buffer_bytes *= *bufferSize; - if (stream->deviceBuffer) free(stream->deviceBuffer); - stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->deviceBuffer == NULL) { - close(fd); - free(stream->userBuffer); - sprintf(message, "RtAudio: OSS error allocating device buffer memory (%s).", - name); - goto error; - } - } - } - - stream->device[mode] = device; - stream->handle[mode] = fd; - stream->state = STREAM_STOPPED; - if ( stream->mode == PLAYBACK && mode == RECORD ) { - stream->mode = DUPLEX; - if (stream->device[0] == device) - stream->handle[0] = fd; - } - else - stream->mode = mode; - - return SUCCESS; - - error: - if (stream->handle[0]) { - close(stream->handle[0]); - stream->handle[0] = 0; - } - error(RtError::WARNING); - return FAILURE; -} - -void RtAudio :: cancelStreamCallback(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - if (stream->usingCallback) { - stream->usingCallback = false; - pthread_cancel(stream->thread); - pthread_join(stream->thread, NULL); - stream->thread = 0; - stream->callback = NULL; - stream->userData = NULL; - } -} - -void RtAudio :: closeStream(int streamId) -{ - // We don't want an exception to be thrown here because this - // function is called by our class destructor. So, do our own - // streamId check. - if ( streams.find( streamId ) == streams.end() ) { - sprintf(message, "RtAudio: invalid stream identifier!"); - error(RtError::WARNING); - return; - } - - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; - - if (stream->usingCallback) { - pthread_cancel(stream->thread); - pthread_join(stream->thread, NULL); - } - - if (stream->state == STREAM_RUNNING) { - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) - ioctl(stream->handle[0], SNDCTL_DSP_RESET, 0); - if (stream->mode == RECORD || stream->mode == DUPLEX) - ioctl(stream->handle[1], SNDCTL_DSP_RESET, 0); - } - - pthread_mutex_destroy(&stream->mutex); - - if (stream->handle[0]) - close(stream->handle[0]); - - if (stream->handle[1]) - close(stream->handle[1]); - - if (stream->userBuffer) - free(stream->userBuffer); - - if (stream->deviceBuffer) - free(stream->deviceBuffer); - - free(stream); - streams.erase(streamId); -} - -void RtAudio :: startStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - stream->state = STREAM_RUNNING; - - // No need to do anything else here ... OSS automatically starts when fed samples. -} - -void RtAudio :: stopStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - if (stream->state == STREAM_STOPPED) - goto unlock; - - int err; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - err = ioctl(stream->handle[0], SNDCTL_DSP_SYNC, 0); - if (err < -1) { - sprintf(message, "RtAudio: OSS error stopping device (%s).", - devices[stream->device[0]].name); - error(RtError::DRIVER_ERROR); - } - } - else { - err = ioctl(stream->handle[1], SNDCTL_DSP_SYNC, 0); - if (err < -1) { - sprintf(message, "RtAudio: OSS error stopping device (%s).", - devices[stream->device[1]].name); - error(RtError::DRIVER_ERROR); - } - } - stream->state = STREAM_STOPPED; - - unlock: - MUTEX_UNLOCK(&stream->mutex); -} - -void RtAudio :: abortStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - if (stream->state == STREAM_STOPPED) - goto unlock; - - int err; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - err = ioctl(stream->handle[0], SNDCTL_DSP_RESET, 0); - if (err < -1) { - sprintf(message, "RtAudio: OSS error aborting device (%s).", - devices[stream->device[0]].name); - error(RtError::DRIVER_ERROR); - } - } - else { - err = ioctl(stream->handle[1], SNDCTL_DSP_RESET, 0); - if (err < -1) { - sprintf(message, "RtAudio: OSS error aborting device (%s).", - devices[stream->device[1]].name); - error(RtError::DRIVER_ERROR); - } - } - stream->state = STREAM_STOPPED; - - unlock: - MUTEX_UNLOCK(&stream->mutex); -} - -int RtAudio :: streamWillBlock(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - int bytes = 0, channels = 0, frames = 0; - if (stream->state == STREAM_STOPPED) - goto unlock; - - audio_buf_info info; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - ioctl(stream->handle[0], SNDCTL_DSP_GETOSPACE, &info); - bytes = info.bytes; - channels = stream->nDeviceChannels[0]; - } - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - ioctl(stream->handle[1], SNDCTL_DSP_GETISPACE, &info); - if (stream->mode == DUPLEX ) { - bytes = (bytes < info.bytes) ? bytes : info.bytes; - channels = stream->nDeviceChannels[0]; - } - else { - bytes = info.bytes; - channels = stream->nDeviceChannels[1]; - } - } - - frames = (int) (bytes / (channels * formatBytes(stream->deviceFormat[0]))); - frames -= stream->bufferSize; - if (frames < 0) frames = 0; - - unlock: - MUTEX_UNLOCK(&stream->mutex); - return frames; -} - -void RtAudio :: tickStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - int stopStream = 0; - if (stream->state == STREAM_STOPPED) { - if (stream->usingCallback) usleep(50000); // sleep 50 milliseconds - return; - } - else if (stream->usingCallback) { - stopStream = stream->callback(stream->userBuffer, stream->bufferSize, stream->userData); - } - - 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; - RTAUDIO_FORMAT format; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - - // Setup parameters and do buffer conversion if necessary. - if (stream->doConvertBuffer[0]) { - convertStreamBuffer(stream, PLAYBACK); - buffer = stream->deviceBuffer; - samples = stream->bufferSize * stream->nDeviceChannels[0]; - format = stream->deviceFormat[0]; - } - else { - buffer = stream->userBuffer; - samples = stream->bufferSize * stream->nUserChannels[0]; - format = stream->userFormat; - } - - // Do byte swapping if necessary. - if (stream->doByteSwap[0]) - byteSwapBuffer(buffer, samples, format); - - // Write samples to device. - result = write(stream->handle[0], buffer, samples * formatBytes(format)); - - if (result == -1) { - // This could be an underrun, but the basic OSS API doesn't provide a means for determining that. - sprintf(message, "RtAudio: OSS audio write error for device (%s).", - devices[stream->device[0]].name); - error(RtError::DRIVER_ERROR); - } - } - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - - // Setup parameters. - if (stream->doConvertBuffer[1]) { - buffer = stream->deviceBuffer; - samples = stream->bufferSize * stream->nDeviceChannels[1]; - format = stream->deviceFormat[1]; - } - else { - buffer = stream->userBuffer; - samples = stream->bufferSize * stream->nUserChannels[1]; - format = stream->userFormat; - } - - // Read samples from device. - result = read(stream->handle[1], buffer, samples * formatBytes(format)); - - if (result == -1) { - // This could be an overrun, but the basic OSS API doesn't provide a means for determining that. - sprintf(message, "RtAudio: OSS audio read error for device (%s).", - devices[stream->device[1]].name); - error(RtError::DRIVER_ERROR); - } - - // Do byte swapping if necessary. - if (stream->doByteSwap[1]) - byteSwapBuffer(buffer, samples, format); - - // Do buffer conversion if necessary. - if (stream->doConvertBuffer[1]) - convertStreamBuffer(stream, RECORD); - } - - unlock: - MUTEX_UNLOCK(&stream->mutex); - - if (stream->usingCallback && stopStream) - this->stopStream(streamId); -} - -extern "C" void *callbackHandler(void *ptr) -{ - RtAudio *object = thread_info.object; - int stream = thread_info.streamId; - bool *usingCallback = (bool *) ptr; - - while ( *usingCallback ) { - pthread_testcancel(); - try { - object->tickStream(stream); - } - catch (RtError &exception) { - fprintf(stderr, "\nCallback thread error (%s) ... closing thread.\n\n", - exception.getMessage()); - break; - } - } - - return 0; -} - -//******************** End of __LINUX_OSS__ *********************// - -#elif defined(__WINDOWS_DS__) // Windows DirectSound API - -#include - -// Declarations for utility functions, callbacks, and structures -// specific to the DirectSound implementation. -static bool CALLBACK deviceCountCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, - LPVOID lpContext); - -static bool CALLBACK deviceInfoCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, - LPVOID lpContext); - -static char* getErrorString(int code); - -struct enum_info { - char name[64]; - LPGUID id; - bool isInput; - bool isValid; -}; - -// RtAudio methods for DirectSound implementation. -void RtAudio :: initialize(void) -{ - int i, ins = 0, outs = 0, count = 0; - int index = 0; - HRESULT result; - nDevices = 0; - - // Count DirectSound devices. - result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceCountCallback, &outs); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to enumerate through sound playback devices: %s.", - getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - // Count DirectSoundCapture devices. - result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceCountCallback, &ins); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to enumerate through sound capture devices: %s.", - getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - count = ins + outs; - if (count == 0) return; - - std::vector info(count); - for (i=0; i 0) { - nDevices = 1; - index = 1; - } - - // Non-default devices are listed separately. - for (i=0; i= nDevices ) { - sprintf(message, "RtAudio: device (%s) indexing error in DirectSound probeDeviceInfo().", - info->name); - error(RtError::WARNING); - return; - } - - // Do capture probe first. If this is not the default device (index - // = 0) _and_ GUID = NULL, then the capture handle is invalid. - if ( index != 0 && info->id[1] == NULL ) - goto playback_probe; - - LPDIRECTSOUNDCAPTURE input; - result = DirectSoundCaptureCreate( info->id[0], &input, NULL ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Could not create DirectSound capture object (%s): %s.", - info->name, getErrorString(result)); - error(RtError::WARNING); - goto playback_probe; - } - - DSCCAPS in_caps; - in_caps.dwSize = sizeof(in_caps); - result = input->GetCaps( &in_caps ); - if ( FAILED(result) ) { - input->Release(); - sprintf(message, "RtAudio: Could not get DirectSound capture capabilities (%s): %s.", - info->name, getErrorString(result)); - error(RtError::WARNING); - goto playback_probe; - } - - // Get input channel information. - info->minInputChannels = 1; - info->maxInputChannels = in_caps.dwChannels; - - // Get sample rate and format information. - if( in_caps.dwChannels == 2 ) { - if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->nativeFormats |= RTAUDIO_SINT16; - if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->nativeFormats |= RTAUDIO_SINT16; - if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->nativeFormats |= RTAUDIO_SINT16; - if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->nativeFormats |= RTAUDIO_SINT8; - if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->nativeFormats |= RTAUDIO_SINT8; - if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->nativeFormats |= RTAUDIO_SINT8; - - if ( info->nativeFormats & RTAUDIO_SINT16 ) { - if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->sampleRates[info->nSampleRates++] = 11025; - if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->sampleRates[info->nSampleRates++] = 22050; - if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->sampleRates[info->nSampleRates++] = 44100; - } - else if ( info->nativeFormats & RTAUDIO_SINT8 ) { - if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->sampleRates[info->nSampleRates++] = 11025; - if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->sampleRates[info->nSampleRates++] = 22050; - if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->sampleRates[info->nSampleRates++] = 44100; - } - } - else if ( in_caps.dwChannels == 1 ) { - if( in_caps.dwFormats & WAVE_FORMAT_1M16 ) info->nativeFormats |= RTAUDIO_SINT16; - if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->nativeFormats |= RTAUDIO_SINT16; - if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->nativeFormats |= RTAUDIO_SINT16; - if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->nativeFormats |= RTAUDIO_SINT8; - if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->nativeFormats |= RTAUDIO_SINT8; - if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->nativeFormats |= RTAUDIO_SINT8; - - if ( info->nativeFormats & RTAUDIO_SINT16 ) { - if( in_caps.dwFormats & WAVE_FORMAT_1M16 ) info->sampleRates[info->nSampleRates++] = 11025; - if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->sampleRates[info->nSampleRates++] = 22050; - if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->sampleRates[info->nSampleRates++] = 44100; - } - else if ( info->nativeFormats & RTAUDIO_SINT8 ) { - if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->sampleRates[info->nSampleRates++] = 11025; - if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->sampleRates[info->nSampleRates++] = 22050; - if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->sampleRates[info->nSampleRates++] = 44100; - } - } - else info->minInputChannels = 0; // technically, this would be an error - - input->Release(); - - playback_probe: - LPDIRECTSOUND output; - DSCAPS out_caps; - - // Now do playback probe. If this is not the default device (index - // = 0) _and_ GUID = NULL, then the playback handle is invalid. - if ( index != 0 && info->id[0] == NULL ) - goto check_parameters; - - result = DirectSoundCreate( info->id[0], &output, NULL ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Could not create DirectSound playback object (%s): %s.", - info->name, getErrorString(result)); - error(RtError::WARNING); - goto check_parameters; - } - - out_caps.dwSize = sizeof(out_caps); - result = output->GetCaps( &out_caps ); - if ( FAILED(result) ) { - output->Release(); - sprintf(message, "RtAudio: Could not get DirectSound playback capabilities (%s): %s.", - info->name, getErrorString(result)); - error(RtError::WARNING); - goto check_parameters; - } - - // Get output channel information. - info->minOutputChannels = 1; - info->maxOutputChannels = ( out_caps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; - - // Get sample rate information. Use capture device rate information - // if it exists. - if ( info->nSampleRates == 0 ) { - info->sampleRates[0] = (int) out_caps.dwMinSecondarySampleRate; - info->sampleRates[1] = (int) out_caps.dwMaxSecondarySampleRate; - if ( out_caps.dwFlags & DSCAPS_CONTINUOUSRATE ) - info->nSampleRates = -1; - else if ( out_caps.dwMinSecondarySampleRate == out_caps.dwMaxSecondarySampleRate ) { - if ( out_caps.dwMinSecondarySampleRate == 0 ) { - // This is a bogus driver report ... fake the range and cross - // your fingers. - info->sampleRates[0] = 11025; - info->sampleRates[1] = 48000; - info->nSampleRates = -1; /* continuous range */ - sprintf(message, "RtAudio: bogus sample rates reported by DirectSound driver ... using defaults (%s).", - info->name); - error(RtError::WARNING); - } - else { - info->nSampleRates = 1; - } - } - else if ( (out_caps.dwMinSecondarySampleRate < 1000.0) && - (out_caps.dwMaxSecondarySampleRate > 50000.0) ) { - // This is a bogus driver report ... support for only two - // distant rates. We'll assume this is a range. - info->nSampleRates = -1; - sprintf(message, "RtAudio: bogus sample rates reported by DirectSound driver ... using range (%s).", - info->name); - error(RtError::WARNING); - } - else info->nSampleRates = 2; - } - else { - // Check input rates against output rate range - for ( int i=info->nSampleRates-1; i>=0; i-- ) { - if ( info->sampleRates[i] <= out_caps.dwMaxSecondarySampleRate ) - break; - info->nSampleRates--; - } - while ( info->sampleRates[0] < out_caps.dwMinSecondarySampleRate ) { - info->nSampleRates--; - for ( int i=0; inSampleRates; i++) - info->sampleRates[i] = info->sampleRates[i+1]; - if ( info->nSampleRates <= 0 ) break; - } - } - - // Get format information. - if ( out_caps.dwFlags & DSCAPS_PRIMARY16BIT ) info->nativeFormats |= RTAUDIO_SINT16; - if ( out_caps.dwFlags & DSCAPS_PRIMARY8BIT ) info->nativeFormats |= RTAUDIO_SINT8; - - output->Release(); - - check_parameters: - if ( info->maxInputChannels == 0 && info->maxOutputChannels == 0 ) - return; - if ( info->nSampleRates == 0 || info->nativeFormats == 0 ) - return; - - // Determine duplex status. - if (info->maxInputChannels < info->maxOutputChannels) - info->maxDuplexChannels = info->maxInputChannels; - else - info->maxDuplexChannels = info->maxOutputChannels; - if (info->minInputChannels < info->minOutputChannels) - info->minDuplexChannels = info->minInputChannels; - else - info->minDuplexChannels = info->minOutputChannels; - - if ( info->maxDuplexChannels > 0 ) info->hasDuplexSupport = true; - else info->hasDuplexSupport = false; - - info->probed = true; - - return; -} - -bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, - STREAM_MODE mode, int channels, - int sampleRate, RTAUDIO_FORMAT format, - int *bufferSize, int numberOfBuffers) -{ - HRESULT result; - HWND hWnd = GetForegroundWindow(); - // According to a note in PortAudio, using GetDesktopWindow() - // instead of GetForegroundWindow() is supposed to avoid problems - // that occur when the application's window is not the foreground - // 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(). - long buffer_size; - LPVOID audioPtr; - DWORD dataLen; - int nBuffers; - - // 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. - if (numberOfBuffers < 2) - nBuffers = 2; - else - nBuffers = numberOfBuffers; - - // Define the wave format structure (16-bit PCM, srate, channels) - WAVEFORMATEX waveFormat; - ZeroMemory(&waveFormat, sizeof(WAVEFORMATEX)); - waveFormat.wFormatTag = WAVE_FORMAT_PCM; - waveFormat.nChannels = channels; - waveFormat.nSamplesPerSec = (unsigned long) sampleRate; - - // Determine the data format. - if ( devices[device].nativeFormats ) { // 8-bit and/or 16-bit support - if ( format == RTAUDIO_SINT8 ) { - if ( devices[device].nativeFormats & RTAUDIO_SINT8 ) - waveFormat.wBitsPerSample = 8; - else - waveFormat.wBitsPerSample = 16; - } - else { - if ( devices[device].nativeFormats & RTAUDIO_SINT16 ) - waveFormat.wBitsPerSample = 16; - else - waveFormat.wBitsPerSample = 8; - } - } - else { - sprintf(message, "RtAudio: no reported data formats for DirectSound device (%s).", - devices[device].name); - error(RtError::WARNING); - return FAILURE; - } - - waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; - waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; - - if ( mode == PLAYBACK ) { - - if ( devices[device].maxOutputChannels < channels ) - return FAILURE; - - LPGUID id = devices[device].id[0]; - LPDIRECTSOUND object; - LPDIRECTSOUNDBUFFER buffer; - DSBUFFERDESC bufferDescription; - - result = DirectSoundCreate( id, &object, NULL ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Could not create DirectSound playback object (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); - return FAILURE; - } - - // Set cooperative level to DSSCL_EXCLUSIVE - result = object->SetCooperativeLevel(hWnd, DSSCL_EXCLUSIVE); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to set DirectSound cooperative level (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); - return FAILURE; - } - - // Even though we will write to the secondary buffer, we need to - // access the primary buffer to set the correct output format. - // The default is 8-bit, 22 kHz! - // Setup the DS primary buffer description. - ZeroMemory(&bufferDescription, sizeof(DSBUFFERDESC)); - bufferDescription.dwSize = sizeof(DSBUFFERDESC); - bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; - // Obtain the primary buffer - result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to access DS primary buffer (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); - return FAILURE; - } - - // Set the primary DS buffer sound format. - result = buffer->SetFormat(&waveFormat); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to set DS primary buffer format (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); - return FAILURE; - } - - // Setup the secondary DS buffer description. - buffer_size = channels * *bufferSize * nBuffers * waveFormat.wBitsPerSample / 8; - ZeroMemory(&bufferDescription, sizeof(DSBUFFERDESC)); - bufferDescription.dwSize = sizeof(DSBUFFERDESC); - bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | - DSBCAPS_GETCURRENTPOSITION2 | - DSBCAPS_LOCHARDWARE ); // Force hardware mixing - bufferDescription.dwBufferBytes = buffer_size; - 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 = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL); - if ( FAILED(result) ) { - bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | - DSBCAPS_GETCURRENTPOSITION2 | - DSBCAPS_LOCSOFTWARE ); // Force software mixing - result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to create secondary DS buffer (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); - return FAILURE; - } - } - - // Get the buffer size ... might be different from what we specified. - DSBCAPS dsbcaps; - dsbcaps.dwSize = sizeof(DSBCAPS); - buffer->GetCaps(&dsbcaps); - buffer_size = dsbcaps.dwBufferBytes; - - // Lock the DS buffer - result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to lock DS buffer (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); - return FAILURE; - } - - // Zero the DS buffer - ZeroMemory(audioPtr, dataLen); - - // Unlock the DS buffer - result = buffer->Unlock(audioPtr, dataLen, NULL, 0); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to unlock DS buffer(%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); - return FAILURE; - } - - stream->handle[0].object = (void *) object; - stream->handle[0].buffer = (void *) buffer; - stream->nDeviceChannels[0] = channels; - } - - if ( mode == RECORD ) { - - if ( devices[device].maxInputChannels < channels ) - return FAILURE; - - LPGUID id = devices[device].id[1]; - LPDIRECTSOUNDCAPTURE object; - LPDIRECTSOUNDCAPTUREBUFFER buffer; - DSCBUFFERDESC bufferDescription; - - result = DirectSoundCaptureCreate( id, &object, NULL ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Could not create DirectSound capture object (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); - return FAILURE; - } - - // Setup the secondary DS buffer description. - buffer_size = channels * *bufferSize * nBuffers * waveFormat.wBitsPerSample / 8; - ZeroMemory(&bufferDescription, sizeof(DSCBUFFERDESC)); - bufferDescription.dwSize = sizeof(DSCBUFFERDESC); - bufferDescription.dwFlags = 0; - bufferDescription.dwReserved = 0; - bufferDescription.dwBufferBytes = buffer_size; - bufferDescription.lpwfxFormat = &waveFormat; - - // Create the capture buffer. - result = object->CreateCaptureBuffer(&bufferDescription, &buffer, NULL); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to create DS capture buffer (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); - return FAILURE; - } - - // Lock the capture buffer - result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); - return FAILURE; - } - - // Zero the buffer - ZeroMemory(audioPtr, dataLen); - - // Unlock the buffer - result = buffer->Unlock(audioPtr, dataLen, NULL, 0); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); - return FAILURE; - } - - stream->handle[1].object = (void *) object; - stream->handle[1].buffer = (void *) buffer; - stream->nDeviceChannels[1] = channels; - } - - stream->userFormat = format; - if ( waveFormat.wBitsPerSample == 8 ) - stream->deviceFormat[mode] = RTAUDIO_SINT8; - else - stream->deviceFormat[mode] = RTAUDIO_SINT16; - stream->nUserChannels[mode] = channels; - *bufferSize = buffer_size / (channels * nBuffers * waveFormat.wBitsPerSample / 8); - stream->bufferSize = *bufferSize; - - // Set flags for buffer conversion - stream->doConvertBuffer[mode] = false; - if (stream->userFormat != stream->deviceFormat[mode]) - stream->doConvertBuffer[mode] = true; - if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode]) - stream->doConvertBuffer[mode] = true; - - // Allocate necessary internal buffers - if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { - - long buffer_bytes; - if (stream->nUserChannels[0] >= stream->nUserChannels[1]) - buffer_bytes = stream->nUserChannels[0]; - else - buffer_bytes = stream->nUserChannels[1]; - - buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); - if (stream->userBuffer) free(stream->userBuffer); - stream->userBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->userBuffer == NULL) - goto memory_error; - } - - if ( stream->doConvertBuffer[mode] ) { - - long buffer_bytes; - bool makeBuffer = true; - if ( mode == PLAYBACK ) - buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - else { // mode == RECORD - buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); - if ( stream->mode == PLAYBACK ) { - long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - if ( buffer_bytes > bytes_out ) - buffer_bytes = (buffer_bytes > bytes_out) ? buffer_bytes : bytes_out; - else - makeBuffer = false; - } - } - - if ( makeBuffer ) { - buffer_bytes *= *bufferSize; - if (stream->deviceBuffer) free(stream->deviceBuffer); - stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->deviceBuffer == NULL) - goto memory_error; - } - } - - stream->device[mode] = device; - stream->state = STREAM_STOPPED; - if ( stream->mode == PLAYBACK && mode == RECORD ) - // We had already set up an output stream. - stream->mode = DUPLEX; - else - stream->mode = mode; - stream->nBuffers = nBuffers; - stream->sampleRate = sampleRate; - - return SUCCESS; - - memory_error: - if (stream->handle[0].object) { - LPDIRECTSOUND object = (LPDIRECTSOUND) stream->handle[0].object; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; - if (buffer) { - buffer->Release(); - stream->handle[0].buffer = NULL; - } - object->Release(); - stream->handle[0].object = NULL; - } - if (stream->handle[1].object) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) stream->handle[1].object; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; - if (buffer) { - buffer->Release(); - stream->handle[1].buffer = NULL; - } - object->Release(); - stream->handle[1].object = NULL; - } - if (stream->userBuffer) { - free(stream->userBuffer); - stream->userBuffer = 0; - } - sprintf(message, "RtAudio: error allocating buffer memory (%s).", - devices[device].name); - error(RtError::WARNING); - return FAILURE; -} - -void RtAudio :: cancelStreamCallback(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - if (stream->usingCallback) { - stream->usingCallback = false; - WaitForSingleObject( (HANDLE)stream->thread, INFINITE ); - CloseHandle( (HANDLE)stream->thread ); - stream->thread = 0; - stream->callback = NULL; - stream->userData = NULL; - } -} - -void RtAudio :: closeStream(int streamId) -{ - // We don't want an exception to be thrown here because this - // function is called by our class destructor. So, do our own - // streamId check. - if ( streams.find( streamId ) == streams.end() ) { - sprintf(message, "RtAudio: invalid stream identifier!"); - error(RtError::WARNING); - return; - } - - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; - - if (stream->usingCallback) { - stream->usingCallback = false; - WaitForSingleObject( (HANDLE)stream->thread, INFINITE ); - CloseHandle( (HANDLE)stream->thread ); - } - - DeleteCriticalSection(&stream->mutex); - - if (stream->handle[0].object) { - LPDIRECTSOUND object = (LPDIRECTSOUND) stream->handle[0].object; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; - if (buffer) { - buffer->Stop(); - buffer->Release(); - } - object->Release(); - } - - if (stream->handle[1].object) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) stream->handle[1].object; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; - if (buffer) { - buffer->Stop(); - buffer->Release(); - } - object->Release(); - } - - if (stream->userBuffer) - free(stream->userBuffer); - - if (stream->deviceBuffer) - free(stream->deviceBuffer); - - free(stream); - streams.erase(streamId); -} - -void RtAudio :: startStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - if (stream->state == STREAM_RUNNING) - goto unlock; - - HRESULT result; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; - result = buffer->Play(0, 0, DSBPLAY_LOOPING ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to start DS buffer (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - } - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; - result = buffer->Start(DSCBSTART_LOOPING ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to start DS capture buffer (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - } - stream->state = STREAM_RUNNING; - - unlock: - MUTEX_UNLOCK(&stream->mutex); -} - -void RtAudio :: stopStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - if (stream->state == STREAM_STOPPED) { - MUTEX_UNLOCK(&stream->mutex); - return; - } - - // 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 - // concept is less clear so we'll repeat what we do in the - // abortStream() case. - HRESULT result; - DWORD dsBufferSize; - LPVOID buffer1 = NULL; - LPVOID buffer2 = NULL; - DWORD bufferSize1 = 0; - DWORD bufferSize2 = 0; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - - DWORD currentPos, safePos; - long buffer_bytes = stream->bufferSize * stream->nDeviceChannels[0]; - buffer_bytes *= formatBytes(stream->deviceFormat[0]); - - LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; - UINT nextWritePos = stream->handle[0].bufferPointer; - dsBufferSize = buffer_bytes * stream->nBuffers; - - // Write zeroes for nBuffer counts. - for (int i=0; inBuffers; i++) { - - // Find out where the read and "safe write" pointers are. - result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset - DWORD endWrite = nextWritePos + buffer_bytes; - - // Check whether the entire write region is behind the play pointer. - while ( currentPos < endWrite ) { - float millis = (endWrite - currentPos) * 900.0; - millis /= ( formatBytes(stream->deviceFormat[0]) * stream->sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); - - // Wake up, find out where we are now - result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset - } - - // Lock free space in the buffer - result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to lock DS buffer during playback (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - // Zero the free space - ZeroMemory(buffer1, bufferSize1); - if (buffer2 != NULL) ZeroMemory(buffer2, bufferSize2); - - // Update our buffer offset and unlock sound buffer - dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to unlock DS buffer during playback (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize; - stream->handle[0].bufferPointer = nextWritePos; - } - - // If we play again, start at the beginning of the buffer. - stream->handle[0].bufferPointer = 0; - } - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; - buffer1 = NULL; - bufferSize1 = 0; - - result = buffer->Stop(); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to stop DS capture buffer (%s): %s", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - dsBufferSize = stream->bufferSize * stream->nDeviceChannels[1]; - dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers; - - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock(0, dsBufferSize, &buffer1, &bufferSize1, NULL, NULL, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - // Zero the DS buffer - ZeroMemory(buffer1, bufferSize1); - - // Unlock the DS buffer - result = buffer->Unlock(buffer1, bufferSize1, NULL, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - // If we start recording again, we must begin at beginning of buffer. - stream->handle[1].bufferPointer = 0; - } - stream->state = STREAM_STOPPED; - - MUTEX_UNLOCK(&stream->mutex); -} - -void RtAudio :: abortStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - if (stream->state == STREAM_STOPPED) - goto unlock; - - HRESULT result; - long dsBufferSize; - LPVOID audioPtr; - DWORD dataLen; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; - result = buffer->Stop(); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to stop DS buffer (%s): %s", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - dsBufferSize = stream->bufferSize * stream->nDeviceChannels[0]; - dsBufferSize *= formatBytes(stream->deviceFormat[0]) * stream->nBuffers; - - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to lock DS buffer (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - // Zero the DS buffer - ZeroMemory(audioPtr, dataLen); - - // Unlock the DS buffer - result = buffer->Unlock(audioPtr, dataLen, NULL, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to unlock DS buffer (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - // If we start playing again, we must begin at beginning of buffer. - stream->handle[0].bufferPointer = 0; - } - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; - audioPtr = NULL; - dataLen = 0; - - result = buffer->Stop(); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to stop DS capture buffer (%s): %s", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - dsBufferSize = stream->bufferSize * stream->nDeviceChannels[1]; - dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers; - - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - // Zero the DS buffer - ZeroMemory(audioPtr, dataLen); - - // Unlock the DS buffer - result = buffer->Unlock(audioPtr, dataLen, NULL, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - // If we start recording again, we must begin at beginning of buffer. - stream->handle[1].bufferPointer = 0; - } - stream->state = STREAM_STOPPED; - - unlock: - MUTEX_UNLOCK(&stream->mutex); -} - -int RtAudio :: streamWillBlock(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - int channels; - int frames = 0; - if (stream->state == STREAM_STOPPED) - goto unlock; - - HRESULT result; - DWORD currentPos, safePos; - channels = 1; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - - LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; - UINT nextWritePos = stream->handle[0].bufferPointer; - channels = stream->nDeviceChannels[0]; - DWORD dsBufferSize = stream->bufferSize * channels; - dsBufferSize *= formatBytes(stream->deviceFormat[0]) * stream->nBuffers; - - // Find out where the read and "safe write" pointers are. - result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset - frames = currentPos - nextWritePos; - frames /= channels * formatBytes(stream->deviceFormat[0]); - } - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - - LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; - UINT nextReadPos = stream->handle[1].bufferPointer; - channels = stream->nDeviceChannels[1]; - DWORD dsBufferSize = stream->bufferSize * channels; - dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers; - - // Find out where the write and "safe read" pointers are. - result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset - - if (stream->mode == DUPLEX ) { - // Take largest value of the two. - int temp = safePos - nextReadPos; - temp /= channels * formatBytes(stream->deviceFormat[1]); - frames = ( temp > frames ) ? temp : frames; - } - else { - frames = safePos - nextReadPos; - frames /= channels * formatBytes(stream->deviceFormat[1]); - } - } - - frames = stream->bufferSize - frames; - if (frames < 0) frames = 0; - - unlock: - MUTEX_UNLOCK(&stream->mutex); - return frames; -} - -void RtAudio :: tickStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - int stopStream = 0; - if (stream->state == STREAM_STOPPED) { - if (stream->usingCallback) Sleep(50); // sleep 50 milliseconds - return; - } - else if (stream->usingCallback) { - stopStream = stream->callback(stream->userBuffer, stream->bufferSize, stream->userData); - } - - MUTEX_LOCK(&stream->mutex); - - // The state might change while waiting on a mutex. - if (stream->state == STREAM_STOPPED) { - MUTEX_UNLOCK(&stream->mutex); - if (stream->usingCallback && stopStream) - this->stopStream(streamId); - } - - HRESULT result; - DWORD currentPos, safePos; - LPVOID buffer1 = NULL; - LPVOID buffer2 = NULL; - DWORD bufferSize1 = 0; - DWORD bufferSize2 = 0; - char *buffer; - long buffer_bytes; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - - // Setup parameters and do buffer conversion if necessary. - if (stream->doConvertBuffer[0]) { - convertStreamBuffer(stream, PLAYBACK); - buffer = stream->deviceBuffer; - buffer_bytes = stream->bufferSize * stream->nDeviceChannels[0]; - buffer_bytes *= formatBytes(stream->deviceFormat[0]); - } - else { - buffer = stream->userBuffer; - buffer_bytes = stream->bufferSize * stream->nUserChannels[0]; - buffer_bytes *= formatBytes(stream->userFormat); - } - - // No byte swapping necessary in DirectSound implementation. - - LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; - UINT nextWritePos = stream->handle[0].bufferPointer; - DWORD dsBufferSize = buffer_bytes * stream->nBuffers; - - // Find out where the read and "safe write" pointers are. - result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset - DWORD endWrite = nextWritePos + buffer_bytes; - - // Check whether the entire write region is behind the play pointer. - while ( currentPos < endWrite ) { - // 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. - float millis = (endWrite - currentPos) * 900.0; - millis /= ( formatBytes(stream->deviceFormat[0]) * stream->sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); - - // Wake up, find out where we are now - result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset - } - - // Lock free space in the buffer - result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to lock DS buffer during playback (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_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) ) { - sprintf(message, "RtAudio: Unable to unlock DS buffer during playback (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize; - stream->handle[0].bufferPointer = nextWritePos; - } - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - - // Setup parameters. - if (stream->doConvertBuffer[1]) { - buffer = stream->deviceBuffer; - buffer_bytes = stream->bufferSize * stream->nDeviceChannels[1]; - buffer_bytes *= formatBytes(stream->deviceFormat[1]); - } - else { - buffer = stream->userBuffer; - buffer_bytes = stream->bufferSize * stream->nUserChannels[1]; - buffer_bytes *= formatBytes(stream->userFormat); - } - - LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; - UINT nextReadPos = stream->handle[1].bufferPointer; - DWORD dsBufferSize = buffer_bytes * stream->nBuffers; - - // Find out where the write and "safe read" pointers are. - result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset - DWORD endRead = nextReadPos + buffer_bytes; - - // Check whether the entire write region is behind the play pointer. - while ( safePos < endRead ) { - // See comments for playback. - float millis = (endRead - safePos) * 900.0; - millis /= ( formatBytes(stream->deviceFormat[1]) * stream->sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); - - // Wake up, find out where we are now - result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset - } - - // Lock free space in the buffer - result = dsBuffer->Lock (nextReadPos, buffer_bytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to lock DS buffer during capture (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - // Copy our buffer into the DS buffer - CopyMemory(buffer, buffer1, bufferSize1); - if (buffer2 != NULL) CopyMemory(buffer+bufferSize1, buffer2, bufferSize2); - - // Update our buffer offset and unlock sound buffer - nextReadPos = (nextReadPos + bufferSize1 + bufferSize2) % dsBufferSize; - dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to unlock DS buffer during capture (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - stream->handle[1].bufferPointer = nextReadPos; - - // No byte swapping necessary in DirectSound implementation. - - // Do buffer conversion if necessary. - if (stream->doConvertBuffer[1]) - convertStreamBuffer(stream, RECORD); - } - - MUTEX_UNLOCK(&stream->mutex); - - if (stream->usingCallback && stopStream) - this->stopStream(streamId); -} - -// Definitions for utility functions and callbacks -// specific to the DirectSound implementation. - -extern "C" unsigned __stdcall callbackHandler(void *ptr) -{ - RtAudio *object = thread_info.object; - int stream = thread_info.streamId; - bool *usingCallback = (bool *) ptr; - - while ( *usingCallback ) { - try { - object->tickStream(stream); - } - catch (RtError &exception) { - fprintf(stderr, "\nCallback thread error (%s) ... closing thread.\n\n", - exception.getMessage()); - break; - } - } - - _endthreadex( 0 ); - return 0; -} - -static bool CALLBACK deviceCountCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, - LPVOID lpContext) -{ - int *pointer = ((int *) lpContext); - (*pointer)++; - - return true; -} - -static bool CALLBACK deviceInfoCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, - LPVOID lpContext) -{ - enum_info *info = ((enum_info *) lpContext); - while (strlen(info->name) > 0) info++; - - strncpy(info->name, lpcstrDescription, 64); - info->id = lpguid; - - HRESULT hr; - info->isValid = false; - 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->isValid = true; - } - 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->isValid = true; - } - object->Release(); - } - - return true; -} - -static char* getErrorString(int code) -{ - switch (code) { - - case DSERR_ALLOCATED: - return "Direct Sound already allocated"; - - case DSERR_CONTROLUNAVAIL: - return "Direct Sound control unavailable"; - - case DSERR_INVALIDPARAM: - return "Direct Sound invalid parameter"; - - case DSERR_INVALIDCALL: - return "Direct Sound invalid call"; - - case DSERR_GENERIC: - return "Direct Sound generic error"; - - case DSERR_PRIOLEVELNEEDED: - return "Direct Sound Priority level needed"; - - case DSERR_OUTOFMEMORY: - return "Direct Sound out of memory"; - - case DSERR_BADFORMAT: - return "Direct Sound bad format"; - - case DSERR_UNSUPPORTED: - return "Direct Sound unsupported error"; - - case DSERR_NODRIVER: - return "Direct Sound no driver error"; - - case DSERR_ALREADYINITIALIZED: - return "Direct Sound already initialized"; - - case DSERR_NOAGGREGATION: - return "Direct Sound no aggregation"; - - case DSERR_BUFFERLOST: - return "Direct Sound buffer lost"; - - case DSERR_OTHERAPPHASPRIO: - return "Direct Sound other app has priority"; - - case DSERR_UNINITIALIZED: - return "Direct Sound uninitialized"; - - default: - return "Direct Sound unknown error"; - } -} - -//******************** End of __WINDOWS_DS__ *********************// - -#elif defined(__IRIX_AL__) // SGI's AL API for IRIX - -#include -#include - -void RtAudio :: initialize(void) -{ - - // Count cards and devices - nDevices = 0; - - // Determine the total number of input and output devices. - nDevices = alQueryValues(AL_SYSTEM, AL_DEVICES, 0, 0, 0, 0); - if (nDevices < 0) { - sprintf(message, "RtAudio: AL error counting devices: %s.", - alGetErrorString(oserror())); - error(RtError::DRIVER_ERROR); - } - - if (nDevices <= 0) return; - - ALvalue *vls = (ALvalue *) new ALvalue[nDevices]; - - // Add one for our default input/output devices. - nDevices++; - - // Allocate the RTAUDIO_DEVICE structures. - devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); - if (devices == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::MEMORY_ERROR); - } - - // Write device ascii identifiers to device info structure. - char name[32]; - int outs, ins, i; - ALpv pvs[1]; - pvs[0].param = AL_NAME; - pvs[0].value.ptr = name; - pvs[0].sizeIn = 32; - - strcpy(devices[0].name, "Default Input/Output Devices"); - - outs = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, vls, nDevices-1, 0, 0); - if (outs < 0) { - sprintf(message, "RtAudio: AL error getting output devices: %s.", - alGetErrorString(oserror())); - error(RtError::DRIVER_ERROR); - } - - for (i=0; iname, "Default Input/Output Devices", 28) ) { - result = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, &value, 1, 0, 0); - if (result < 0) { - sprintf(message, "RtAudio: AL error getting default output device id: %s.", - alGetErrorString(oserror())); - error(RtError::WARNING); - } - else - resource = value.i; - } - else - resource = info->id[0]; - - if (resource > 0) { - - // Probe output device parameters. - result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0); - if (result < 0) { - sprintf(message, "RtAudio: AL error getting device (%s) channels: %s.", - info->name, alGetErrorString(oserror())); - error(RtError::WARNING); - } - else { - info->maxOutputChannels = value.i; - info->minOutputChannels = 1; - } - - result = alGetParamInfo(resource, AL_RATE, &pinfo); - if (result < 0) { - sprintf(message, "RtAudio: AL error getting device (%s) rates: %s.", - info->name, alGetErrorString(oserror())); - error(RtError::WARNING); - } - else { - info->nSampleRates = 0; - for (i=0; i= pinfo.min.i && SAMPLE_RATES[i] <= pinfo.max.i ) { - info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i]; - info->nSampleRates++; - } - } - } - - // The AL library supports all our formats, except 24-bit and 32-bit ints. - info->nativeFormats = (RTAUDIO_FORMAT) 51; - } - - // Now get input resource ID if it exists. - if ( !strncmp(info->name, "Default Input/Output Devices", 28) ) { - result = alQueryValues(AL_SYSTEM, AL_DEFAULT_INPUT, &value, 1, 0, 0); - if (result < 0) { - sprintf(message, "RtAudio: AL error getting default input device id: %s.", - alGetErrorString(oserror())); - error(RtError::WARNING); - } - else - resource = value.i; - } - else - resource = info->id[1]; - - if (resource > 0) { - - // Probe input device parameters. - result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0); - if (result < 0) { - sprintf(message, "RtAudio: AL error getting device (%s) channels: %s.", - info->name, alGetErrorString(oserror())); - error(RtError::WARNING); - } - else { - info->maxInputChannels = value.i; - info->minInputChannels = 1; - } - - result = alGetParamInfo(resource, AL_RATE, &pinfo); - if (result < 0) { - sprintf(message, "RtAudio: AL error getting device (%s) rates: %s.", - info->name, alGetErrorString(oserror())); - error(RtError::WARNING); - } - else { - // In the case of the default device, these values will - // overwrite the rates determined for the output device. Since - // the input device is most likely to be more limited than the - // output device, this is ok. - info->nSampleRates = 0; - for (i=0; i= pinfo.min.i && SAMPLE_RATES[i] <= pinfo.max.i ) { - info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i]; - info->nSampleRates++; - } - } - } - - // The AL library supports all our formats, except 24-bit and 32-bit ints. - info->nativeFormats = (RTAUDIO_FORMAT) 51; - } - - if ( info->maxInputChannels == 0 && info->maxOutputChannels == 0 ) - return; - if ( info->nSampleRates == 0 ) - return; - - // Determine duplex status. - if (info->maxInputChannels < info->maxOutputChannels) - info->maxDuplexChannels = info->maxInputChannels; - else - info->maxDuplexChannels = info->maxOutputChannels; - if (info->minInputChannels < info->minOutputChannels) - info->minDuplexChannels = info->minInputChannels; - else - info->minDuplexChannels = info->minOutputChannels; - - if ( info->maxDuplexChannels > 0 ) info->hasDuplexSupport = true; - else info->hasDuplexSupport = false; - - info->probed = true; - - return; -} - -bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, - STREAM_MODE mode, int channels, - int sampleRate, RTAUDIO_FORMAT format, - int *bufferSize, int numberOfBuffers) -{ - int result, resource, nBuffers; - ALconfig al_config; - ALport port; - ALpv pvs[2]; - - // Get a new ALconfig structure. - al_config = alNewConfig(); - if ( !al_config ) { - sprintf(message,"RtAudio: can't get AL config: %s.", - alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; - } - - // Set the channels. - result = alSetChannels(al_config, channels); - if ( result < 0 ) { - sprintf(message,"RtAudio: can't set %d channels in AL config: %s.", - channels, alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; - } - - // Set the queue (buffer) size. - if ( numberOfBuffers < 1 ) - nBuffers = 1; - else - nBuffers = numberOfBuffers; - long buffer_size = *bufferSize * nBuffers; - result = alSetQueueSize(al_config, buffer_size); // in sample frames - if ( result < 0 ) { - sprintf(message,"RtAudio: can't set buffer size (%ld) in AL config: %s.", - buffer_size, alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; - } - - // Set the data format. - stream->userFormat = format; - stream->deviceFormat[mode] = format; - if (format == RTAUDIO_SINT8) { - result = alSetSampFmt(al_config, AL_SAMPFMT_TWOSCOMP); - result = alSetWidth(al_config, AL_SAMPLE_8); - } - else if (format == RTAUDIO_SINT16) { - result = alSetSampFmt(al_config, AL_SAMPFMT_TWOSCOMP); - result = alSetWidth(al_config, AL_SAMPLE_16); - } - else if (format == RTAUDIO_SINT24) { - // Our 24-bit format assumes the upper 3 bytes of a 4 byte word. - // The AL library uses the lower 3 bytes, so we'll need to do our - // own conversion. - result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT); - stream->deviceFormat[mode] = RTAUDIO_FLOAT32; - } - else if (format == RTAUDIO_SINT32) { - // The AL library doesn't seem to support the 32-bit integer - // format, so we'll need to do our own conversion. - result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT); - stream->deviceFormat[mode] = RTAUDIO_FLOAT32; - } - else if (format == RTAUDIO_FLOAT32) - result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT); - else if (format == RTAUDIO_FLOAT64) - result = alSetSampFmt(al_config, AL_SAMPFMT_DOUBLE); - - if ( result == -1 ) { - sprintf(message,"RtAudio: AL error setting sample format in AL config: %s.", - alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; - } - - if (mode == PLAYBACK) { - - // Set our device. - if (device == 0) - resource = AL_DEFAULT_OUTPUT; - else - resource = devices[device].id[0]; - result = alSetDevice(al_config, resource); - if ( result == -1 ) { - sprintf(message,"RtAudio: AL error setting device (%s) in AL config: %s.", - devices[device].name, alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; - } - - // Open the port. - port = alOpenPort("RtAudio Output Port", "w", al_config); - if( !port ) { - sprintf(message,"RtAudio: AL error opening output port: %s.", - alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; - } - - // Set the sample rate - pvs[0].param = AL_MASTER_CLOCK; - pvs[0].value.i = AL_CRYSTAL_MCLK_TYPE; - pvs[1].param = AL_RATE; - pvs[1].value.ll = alDoubleToFixed((double)sampleRate); - result = alSetParams(resource, pvs, 2); - if ( result < 0 ) { - alClosePort(port); - sprintf(message,"RtAudio: AL error setting sample rate (%d) for device (%s): %s.", - sampleRate, devices[device].name, alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; - } - } - else { // mode == RECORD - - // Set our device. - if (device == 0) - resource = AL_DEFAULT_INPUT; - else - resource = devices[device].id[1]; - result = alSetDevice(al_config, resource); - if ( result == -1 ) { - sprintf(message,"RtAudio: AL error setting device (%s) in AL config: %s.", - devices[device].name, alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; - } - - // Open the port. - port = alOpenPort("RtAudio Output Port", "r", al_config); - if( !port ) { - sprintf(message,"RtAudio: AL error opening input port: %s.", - alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; - } - - // Set the sample rate - pvs[0].param = AL_MASTER_CLOCK; - pvs[0].value.i = AL_CRYSTAL_MCLK_TYPE; - pvs[1].param = AL_RATE; - pvs[1].value.ll = alDoubleToFixed((double)sampleRate); - result = alSetParams(resource, pvs, 2); - if ( result < 0 ) { - alClosePort(port); - sprintf(message,"RtAudio: AL error setting sample rate (%d) for device (%s): %s.", - sampleRate, devices[device].name, alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; - } - } - - alFreeConfig(al_config); - - stream->nUserChannels[mode] = channels; - stream->nDeviceChannels[mode] = channels; - - // Set handle and flags for buffer conversion - stream->handle[mode] = port; - stream->doConvertBuffer[mode] = false; - if (stream->userFormat != stream->deviceFormat[mode]) - stream->doConvertBuffer[mode] = true; - - // Allocate necessary internal buffers - if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { - - long buffer_bytes; - if (stream->nUserChannels[0] >= stream->nUserChannels[1]) - buffer_bytes = stream->nUserChannels[0]; - else - buffer_bytes = stream->nUserChannels[1]; - - buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); - if (stream->userBuffer) free(stream->userBuffer); - stream->userBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->userBuffer == NULL) - goto memory_error; - } - - if ( stream->doConvertBuffer[mode] ) { - - long buffer_bytes; - bool makeBuffer = true; - if ( mode == PLAYBACK ) - buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - else { // mode == RECORD - buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); - if ( stream->mode == PLAYBACK ) { - long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - if ( buffer_bytes > bytes_out ) - buffer_bytes = (buffer_bytes > bytes_out) ? buffer_bytes : bytes_out; - else - makeBuffer = false; - } - } - - if ( makeBuffer ) { - buffer_bytes *= *bufferSize; - if (stream->deviceBuffer) free(stream->deviceBuffer); - stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->deviceBuffer == NULL) - goto memory_error; - } - } - - stream->device[mode] = device; - stream->state = STREAM_STOPPED; - if ( stream->mode == PLAYBACK && mode == RECORD ) - // We had already set up an output stream. - stream->mode = DUPLEX; - else - stream->mode = mode; - stream->nBuffers = nBuffers; - stream->bufferSize = *bufferSize; - stream->sampleRate = sampleRate; - - return SUCCESS; - - memory_error: - if (stream->handle[0]) { - alClosePort(stream->handle[0]); - stream->handle[0] = 0; - } - if (stream->handle[1]) { - alClosePort(stream->handle[1]); - stream->handle[1] = 0; - } - if (stream->userBuffer) { - free(stream->userBuffer); - stream->userBuffer = 0; - } - sprintf(message, "RtAudio: ALSA error allocating buffer memory for device (%s).", - devices[device].name); - error(RtError::WARNING); - return FAILURE; -} - -void RtAudio :: cancelStreamCallback(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - if (stream->usingCallback) { - stream->usingCallback = false; - pthread_cancel(stream->thread); - pthread_join(stream->thread, NULL); - stream->thread = 0; - stream->callback = NULL; - stream->userData = NULL; - } -} - -void RtAudio :: closeStream(int streamId) -{ - // We don't want an exception to be thrown here because this - // function is called by our class destructor. So, do our own - // streamId check. - if ( streams.find( streamId ) == streams.end() ) { - sprintf(message, "RtAudio: invalid stream identifier!"); - error(RtError::WARNING); - return; - } - - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; - - if (stream->usingCallback) { - pthread_cancel(stream->thread); - pthread_join(stream->thread, NULL); - } - - pthread_mutex_destroy(&stream->mutex); - - if (stream->handle[0]) - alClosePort(stream->handle[0]); - - if (stream->handle[1]) - alClosePort(stream->handle[1]); - - if (stream->userBuffer) - free(stream->userBuffer); - - if (stream->deviceBuffer) - free(stream->deviceBuffer); - - free(stream); - streams.erase(streamId); -} - -void RtAudio :: startStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - if (stream->state == STREAM_RUNNING) - return; - - // The AL port is ready as soon as it is opened. - stream->state = STREAM_RUNNING; -} - -void RtAudio :: stopStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - if (stream->state == STREAM_STOPPED) - goto unlock; - - int result; - int buffer_size = stream->bufferSize * stream->nBuffers; - - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) - alZeroFrames(stream->handle[0], buffer_size); - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - result = alDiscardFrames(stream->handle[1], buffer_size); - if (result == -1) { - sprintf(message, "RtAudio: AL error draining stream device (%s): %s.", - devices[stream->device[1]].name, alGetErrorString(oserror())); - error(RtError::DRIVER_ERROR); - } - } - stream->state = STREAM_STOPPED; - - unlock: - MUTEX_UNLOCK(&stream->mutex); -} - -void RtAudio :: abortStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - if (stream->state == STREAM_STOPPED) - goto unlock; - - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - - int buffer_size = stream->bufferSize * stream->nBuffers; - int result = alDiscardFrames(stream->handle[0], buffer_size); - if (result == -1) { - sprintf(message, "RtAudio: AL error aborting stream device (%s): %s.", - devices[stream->device[0]].name, alGetErrorString(oserror())); - error(RtError::DRIVER_ERROR); - } - } - - // There is no clear action to take on the input stream, since the - // port will continue to run in any event. - stream->state = STREAM_STOPPED; - - unlock: - MUTEX_UNLOCK(&stream->mutex); -} - -int RtAudio :: streamWillBlock(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - int frames = 0; - if (stream->state == STREAM_STOPPED) - goto unlock; - - int err = 0; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - err = alGetFillable(stream->handle[0]); - if (err < 0) { - sprintf(message, "RtAudio: AL error getting available frames for stream (%s): %s.", - devices[stream->device[0]].name, alGetErrorString(oserror())); - error(RtError::DRIVER_ERROR); - } - } - - frames = err; - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - err = alGetFilled(stream->handle[1]); - if (err < 0) { - sprintf(message, "RtAudio: AL error getting available frames for stream (%s): %s.", - devices[stream->device[1]].name, alGetErrorString(oserror())); - error(RtError::DRIVER_ERROR); - } - if (frames > err) frames = err; - } - - frames = stream->bufferSize - frames; - if (frames < 0) frames = 0; - - unlock: - MUTEX_UNLOCK(&stream->mutex); - return frames; -} - -void RtAudio :: tickStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - int stopStream = 0; - if (stream->state == STREAM_STOPPED) { - if (stream->usingCallback) usleep(50000); // sleep 50 milliseconds - return; - } - else if (stream->usingCallback) { - stopStream = stream->callback(stream->userBuffer, stream->bufferSize, stream->userData); - } - - MUTEX_LOCK(&stream->mutex); - - // The state might change while waiting on a mutex. - if (stream->state == STREAM_STOPPED) - goto unlock; - - char *buffer; - int channels; - RTAUDIO_FORMAT format; - if (stream->mode == PLAYBACK || stream->mode == DUPLEX) { - - // Setup parameters and do buffer conversion if necessary. - if (stream->doConvertBuffer[0]) { - convertStreamBuffer(stream, PLAYBACK); - buffer = stream->deviceBuffer; - channels = stream->nDeviceChannels[0]; - format = stream->deviceFormat[0]; - } - else { - buffer = stream->userBuffer; - channels = stream->nUserChannels[0]; - format = stream->userFormat; - } - - // Do byte swapping if necessary. - if (stream->doByteSwap[0]) - byteSwapBuffer(buffer, stream->bufferSize * channels, format); - - // Write interleaved samples to device. - alWriteFrames(stream->handle[0], buffer, stream->bufferSize); - } - - if (stream->mode == RECORD || stream->mode == DUPLEX) { - - // Setup parameters. - if (stream->doConvertBuffer[1]) { - buffer = stream->deviceBuffer; - channels = stream->nDeviceChannels[1]; - format = stream->deviceFormat[1]; - } - else { - buffer = stream->userBuffer; - channels = stream->nUserChannels[1]; - format = stream->userFormat; - } - - // Read interleaved samples from device. - alReadFrames(stream->handle[1], buffer, stream->bufferSize); - - // Do byte swapping if necessary. - if (stream->doByteSwap[1]) - byteSwapBuffer(buffer, stream->bufferSize * channels, format); - - // Do buffer conversion if necessary. - if (stream->doConvertBuffer[1]) - convertStreamBuffer(stream, RECORD); - } - - unlock: - MUTEX_UNLOCK(&stream->mutex); - - if (stream->usingCallback && stopStream) - this->stopStream(streamId); -} - -extern "C" void *callbackHandler(void *ptr) -{ - RtAudio *object = thread_info.object; - int stream = thread_info.streamId; - bool *usingCallback = (bool *) ptr; - - while ( *usingCallback ) { - pthread_testcancel(); - try { - object->tickStream(stream); - } - catch (RtError &exception) { - fprintf(stderr, "\nCallback thread error (%s) ... closing thread.\n\n", - exception.getMessage()); - break; - } - } - - return 0; -} - -//******************** End of __IRIX_AL__ *********************// - -#endif - - -// *************************************************** // -// -// Private common (OS-independent) RtAudio methods. -// -// *************************************************** // - -// This method can be modified to control the behavior of error -// message reporting and throwing. -void RtAudio :: error(RtError::TYPE type) -{ - if (type == RtError::WARNING) { -#if defined(RTAUDIO_DEBUG) - fprintf(stderr, "\n%s\n\n", message); - else if (type == RtError::DEBUG_WARNING) { - fprintf(stderr, "\n%s\n\n", message); -#endif - } - else { - fprintf(stderr, "\n%s\n\n", message); - throw RtError(message, type); - } -} - -void *RtAudio :: verifyStream(int streamId) -{ - // Verify the stream key. - if ( streams.find( streamId ) == streams.end() ) { - sprintf(message, "RtAudio: invalid stream identifier!"); - error(RtError::INVALID_STREAM); - } - - return streams[streamId]; -} - -void RtAudio :: clearDeviceInfo(RTAUDIO_DEVICE *info) -{ - // Don't clear the name or DEVICE_ID fields here ... they are - // typically set prior to a call of this function. - info->probed = false; - info->maxOutputChannels = 0; - info->maxInputChannels = 0; - info->maxDuplexChannels = 0; - info->minOutputChannels = 0; - info->minInputChannels = 0; - info->minDuplexChannels = 0; - info->hasDuplexSupport = false; - info->nSampleRates = 0; - for (int i=0; isampleRates[i] = 0; - info->nativeFormats = 0; -} - -int RtAudio :: formatBytes(RTAUDIO_FORMAT 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; - - sprintf(message,"RtAudio: undefined format in formatBytes()."); - error(RtError::WARNING); - - return 0; -} - -void RtAudio :: convertStreamBuffer(RTAUDIO_STREAM *stream, STREAM_MODE mode) -{ - // This method does format conversion, input/output channel compensation, and - // data interleaving/deinterleaving. 24-bit integers are assumed to occupy - // the upper three bytes of a 32-bit integer. - - int j, channels_in, channels_out, channels; - RTAUDIO_FORMAT format_in, format_out; - char *input, *output; - - if (mode == RECORD) { // convert device to user buffer - input = stream->deviceBuffer; - output = stream->userBuffer; - channels_in = stream->nDeviceChannels[1]; - channels_out = stream->nUserChannels[1]; - format_in = stream->deviceFormat[1]; - format_out = stream->userFormat; - } - else { // convert user to device buffer - input = stream->userBuffer; - output = stream->deviceBuffer; - channels_in = stream->nUserChannels[0]; - channels_out = stream->nDeviceChannels[0]; - format_in = stream->userFormat; - format_out = stream->deviceFormat[0]; - - // clear our device buffer when in/out duplex device channels are different - if ( stream->mode == DUPLEX && - stream->nDeviceChannels[0] != stream->nDeviceChannels[1] ) - memset(output, 0, stream->bufferSize * channels_out * formatBytes(format_out)); - } - - channels = (channels_in < channels_out) ? channels_in : channels_out; - - // Set up the interleave/deinterleave offsets - std::vector offset_in(channels); - std::vector offset_out(channels); - if (mode == RECORD && stream->deInterleave[1]) { - for (int k=0; kbufferSize; - offset_out[k] = k; - } - } - else if (mode == PLAYBACK && stream->deInterleave[0]) { - for (int k=0; kbufferSize; - } - } - else { - for (int k=0; kbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; j> 16) & 0x0000ffff); - } - in += channels_in; - out += channels_out; - } - } - else if (format_in == RTAUDIO_SINT32) { - INT32 *in = (INT32 *)input; - for (int i=0; ibufferSize; i++) { - for (j=0; j> 16) & 0x0000ffff); - } - in += channels_in; - out += channels_out; - } - } - else if (format_in == RTAUDIO_FLOAT32) { - FLOAT32 *in = (FLOAT32 *)input; - for (int i=0; ibufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; j> 8) & 0x00ff); - } - in += channels_in; - out += channels_out; - } - } - else if (format_in == RTAUDIO_SINT24) { - INT32 *in = (INT32 *)input; - for (int i=0; ibufferSize; i++) { - for (j=0; j> 24) & 0x000000ff); - } - in += channels_in; - out += channels_out; - } - } - else if (format_in == RTAUDIO_SINT32) { - INT32 *in = (INT32 *)input; - for (int i=0; ibufferSize; i++) { - for (j=0; j> 24) & 0x000000ff); - } - in += channels_in; - out += channels_out; - } - } - else if (format_in == RTAUDIO_FLOAT32) { - FLOAT32 *in = (FLOAT32 *)input; - for (int i=0; ibufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; j +#include +#include + +// Static variable definitions. +const unsigned int RtAudio :: SAMPLE_RATES[] = { + 4000, 5512, 8000, 9600, 11025, 16000, 22050, + 32000, 44100, 48000, 88200, 96000, 176400, 192000 +}; +const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT8 = 1; +const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT16 = 2; +const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT24 = 4; +const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT32 = 8; +const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_FLOAT32 = 16; +const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_FLOAT64 = 32; + +#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) + #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) + #define MUTEX_LOCK(A) EnterCriticalSection(A) + #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) +#else // pthread API + #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) + #define MUTEX_LOCK(A) pthread_mutex_lock(A) + #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A) +#endif + +// *************************************************** // +// +// Public common (OS-independent) methods. +// +// *************************************************** // + +RtAudio :: RtAudio() +{ + initialize(); + + if (nDevices <= 0) { + sprintf(message, "RtAudio: no audio devices found!"); + error(RtError::NO_DEVICES_FOUND); + } +} + +RtAudio :: RtAudio(int *streamId, + int outputDevice, int outputChannels, + int inputDevice, int inputChannels, + RTAUDIO_FORMAT format, int sampleRate, + int *bufferSize, int numberOfBuffers) +{ + initialize(); + + if (nDevices <= 0) { + sprintf(message, "RtAudio: no audio devices found!"); + error(RtError::NO_DEVICES_FOUND); + } + + try { + *streamId = openStream(outputDevice, outputChannels, inputDevice, inputChannels, + format, sampleRate, bufferSize, numberOfBuffers); + } + catch (RtError &exception) { + // deallocate the RTAUDIO_DEVICE structures + if (devices) free(devices); + throw exception; + } +} + +RtAudio :: ~RtAudio() +{ + // close any existing streams + while ( streams.size() ) + closeStream( streams.begin()->first ); + + // deallocate the RTAUDIO_DEVICE structures + if (devices) free(devices); +} + +int RtAudio :: openStream(int outputDevice, int outputChannels, + int inputDevice, int inputChannels, + RTAUDIO_FORMAT format, int sampleRate, + int *bufferSize, int numberOfBuffers) +{ + static int streamKey = 0; // Unique stream identifier ... OK for multiple instances. + + if (outputChannels < 1 && inputChannels < 1) { + sprintf(message,"RtAudio: one or both 'channel' parameters must be greater than zero."); + error(RtError::INVALID_PARAMETER); + } + + if ( formatBytes(format) == 0 ) { + sprintf(message,"RtAudio: 'format' parameter value is undefined."); + error(RtError::INVALID_PARAMETER); + } + + if ( outputChannels > 0 ) { + if (outputDevice > nDevices || outputDevice < 0) { + sprintf(message,"RtAudio: 'outputDevice' parameter value (%d) is invalid.", outputDevice); + error(RtError::INVALID_PARAMETER); + } + } + + if ( inputChannels > 0 ) { + if (inputDevice > nDevices || inputDevice < 0) { + sprintf(message,"RtAudio: 'inputDevice' parameter value (%d) is invalid.", inputDevice); + error(RtError::INVALID_PARAMETER); + } + } + + // Allocate a new stream structure. + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) calloc(1, sizeof(RTAUDIO_STREAM)); + if (stream == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + stream->mode = UNINITIALIZED; + MUTEX_INITIALIZE(&stream->mutex); + + bool result = FAILURE; + int device, defaultDevice = 0; + STREAM_MODE mode; + int channels; + if ( outputChannels > 0 ) { + + mode = OUTPUT; + channels = outputChannels; + + if ( outputDevice == 0 ) { // Try default device first. + defaultDevice = getDefaultOutputDevice(); + device = defaultDevice; + } + else + device = outputDevice - 1; + + for (int i=-1; i= 0 ) { + if ( i == defaultDevice ) continue; + device = i; + } + if (devices[device].probed == false) { + // If the device wasn't successfully probed before, try it + // again now. + clearDeviceInfo(&devices[device]); + probeDeviceInfo(&devices[device]); + } + if ( devices[device].probed ) + result = probeDeviceOpen(device, stream, mode, channels, sampleRate, + format, bufferSize, numberOfBuffers); + if (result == SUCCESS) break; + if ( outputDevice > 0 ) break; + } + } + + if ( inputChannels > 0 && ( result == SUCCESS || outputChannels <= 0 ) ) { + + mode = INPUT; + channels = inputChannels; + + if ( inputDevice == 0 ) { // Try default device first. + defaultDevice = getDefaultInputDevice(); + device = defaultDevice; + } + else + device = inputDevice - 1; + + for (int i=-1; i= 0 ) { + if ( i == defaultDevice ) continue; + device = i; + } + if (devices[device].probed == false) { + // If the device wasn't successfully probed before, try it + // again now. + clearDeviceInfo(&devices[device]); + probeDeviceInfo(&devices[device]); + } + if ( devices[device].probed ) + result = probeDeviceOpen(device, stream, mode, channels, sampleRate, + format, bufferSize, numberOfBuffers); + if (result == SUCCESS) break; + if ( outputDevice > 0 ) break; + } + } + + streams[++streamKey] = (void *) stream; + if ( result == SUCCESS ) + return streamKey; + + // If we get here, all attempted probes failed. Close any opened + // devices and delete the allocated stream. + closeStream(streamKey); + if ( ( outputDevice == 0 && outputChannels > 0 ) + || ( inputDevice == 0 && inputChannels > 0 ) ) + sprintf(message,"RtAudio: no devices found for given parameters."); + else + sprintf(message,"RtAudio: unable to open specified device(s) with given stream parameters."); + error(RtError::INVALID_PARAMETER); + + return -1; +} + +int RtAudio :: getDeviceCount(void) +{ + return nDevices; +} + +void RtAudio :: getDeviceInfo(int device, RTAUDIO_DEVICE *info) +{ + if (device > nDevices || device < 1) { + sprintf(message, "RtAudio: invalid device specifier (%d)!", device); + error(RtError::INVALID_DEVICE); + } + + int deviceIndex = device - 1; + + // If the device wasn't successfully probed before, try it now (or again). + if (devices[deviceIndex].probed == false) { + clearDeviceInfo(&devices[deviceIndex]); + probeDeviceInfo(&devices[deviceIndex]); + } + + // Clear the info structure. + memset(info, 0, sizeof(RTAUDIO_DEVICE)); + + strncpy(info->name, devices[deviceIndex].name, 128); + info->probed = devices[deviceIndex].probed; + if ( info->probed == true ) { + info->maxOutputChannels = devices[deviceIndex].maxOutputChannels; + info->maxInputChannels = devices[deviceIndex].maxInputChannels; + info->maxDuplexChannels = devices[deviceIndex].maxDuplexChannels; + info->minOutputChannels = devices[deviceIndex].minOutputChannels; + info->minInputChannels = devices[deviceIndex].minInputChannels; + info->minDuplexChannels = devices[deviceIndex].minDuplexChannels; + info->hasDuplexSupport = devices[deviceIndex].hasDuplexSupport; + info->nSampleRates = devices[deviceIndex].nSampleRates; + if (info->nSampleRates == -1) { + info->sampleRates[0] = devices[deviceIndex].sampleRates[0]; + info->sampleRates[1] = devices[deviceIndex].sampleRates[1]; + } + else { + for (int i=0; inSampleRates; i++) + info->sampleRates[i] = devices[deviceIndex].sampleRates[i]; + } + info->nativeFormats = devices[deviceIndex].nativeFormats; + if ( deviceIndex == getDefaultOutputDevice() || + deviceIndex == getDefaultInputDevice() ) + info->isDefault = true; + } + + return; +} + +char * const RtAudio :: getStreamBuffer(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + return stream->userBuffer; +} + +#if defined(__LINUX_ALSA__) || defined(__LINUX_OSS__) || defined(__IRIX_AL__) + +extern "C" void *callbackHandler(void * ptr); + +void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + CALLBACK_INFO *info = (CALLBACK_INFO *) &stream->callbackInfo; + if ( info->usingCallback ) { + sprintf(message, "RtAudio: A callback is already set for this stream!"); + error(RtError::WARNING); + return; + } + + info->callback = (void *) callback; + info->userData = userData; + info->usingCallback = true; + info->object = (void *) this; + info->streamId = streamId; + + int err = pthread_create(&info->thread, NULL, callbackHandler, &stream->callbackInfo); + + if (err) { + info->usingCallback = false; + sprintf(message, "RtAudio: error starting callback thread!"); + error(RtError::THREAD_ERROR); + } +} + +void RtAudio :: cancelStreamCallback(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + if (stream->callbackInfo.usingCallback) { + + if (stream->state == STREAM_RUNNING) + stopStream( streamId ); + + MUTEX_LOCK(&stream->mutex); + + stream->callbackInfo.usingCallback = false; + pthread_cancel(stream->callbackInfo.thread); + pthread_join(stream->callbackInfo.thread, NULL); + stream->callbackInfo.thread = 0; + stream->callbackInfo.callback = NULL; + stream->callbackInfo.userData = NULL; + + MUTEX_UNLOCK(&stream->mutex); + } +} + +#endif + +// *************************************************** // +// +// 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. This same functionality can be +// achieved with better synchrony by opening two separate streams for +// the devices and using RtAudio blocking calls (i.e. tickStream()). +// +// The possibility of having multiple RtAudio streams accessing the +// same CoreAudio device is not currently supported. The problem +// involves the inability to install our callbackHandler function for +// the same device more than once. I experimented with a workaround +// for this, but it requires an additional buffer for mixing output +// data before filling the CoreAudio device buffer. In the end, I +// decided it wasn't worth supporting. +// +// Property listeners are currently not used. The issue is what could +// be done if a critical stream parameter (buffer size, sample rate, +// device disconnect) notification arrived. The listeners entail +// quite a bit of extra code and most likely, a user program wouldn't +// be prepared for the result anyway. Some initial listener code is +// commented out. + +void RtAudio :: initialize(void) +{ + OSStatus err = noErr; + UInt32 dataSize; + AudioDeviceID *deviceList = NULL; + nDevices = 0; + + // Find out how many audio devices there are, if any. + err = AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &dataSize, NULL); + if (err != noErr) { + sprintf(message, "RtAudio: OSX error getting device info!"); + error(RtError::SYSTEM_ERROR); + } + + nDevices = dataSize / sizeof(AudioDeviceID); + if (nDevices == 0) return; + + // Allocate the RTAUDIO_DEVICE structures. + devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); + if (devices == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + + // Make space for the devices we are about to get. + deviceList = (AudioDeviceID *) malloc( dataSize ); + if (deviceList == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + + // Get the array of AudioDeviceIDs. + err = AudioHardwareGetProperty(kAudioHardwarePropertyDevices, &dataSize, (void *) deviceList); + if (err != noErr) { + free(deviceList); + sprintf(message, "RtAudio: OSX error getting device properties!"); + error(RtError::SYSTEM_ERROR); + } + + // Write device identifiers to device structures and then + // probe the device capabilities. + for (int i=0; iid[0], 0, false, + kAudioDevicePropertyDeviceManufacturer, + &dataSize, name ); + if (err != noErr) { + sprintf( message, "RtAudio: OSX error getting device manufacturer." ); + error(RtError::DEBUG_WARNING); + return; + } + strncpy(fullname, name, 256); + strcat(fullname, ": " ); + + dataSize = 256; + err = AudioDeviceGetProperty( info->id[0], 0, false, + kAudioDevicePropertyDeviceName, + &dataSize, name ); + if (err != noErr) { + sprintf( message, "RtAudio: OSX error getting device name." ); + error(RtError::DEBUG_WARNING); + return; + } + strncat(fullname, name, 254); + strncat(info->name, fullname, 128); + + // Get output channel information. + unsigned int i, minChannels, maxChannels, nStreams = 0; + AudioBufferList *bufferList = nil; + err = AudioDeviceGetPropertyInfo( info->id[0], 0, false, + kAudioDevicePropertyStreamConfiguration, + &dataSize, NULL ); + if (err == noErr && dataSize > 0) { + bufferList = (AudioBufferList *) malloc( dataSize ); + if (bufferList == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::DEBUG_WARNING); + return; + } + + err = AudioDeviceGetProperty( info->id[0], 0, false, + kAudioDevicePropertyStreamConfiguration, + &dataSize, bufferList ); + if (err == noErr) { + maxChannels = 0; + minChannels = 1000; + nStreams = bufferList->mNumberBuffers; + for ( i=0; imBuffers[i].mNumberChannels; + if ( bufferList->mBuffers[i].mNumberChannels < minChannels ) + minChannels = bufferList->mBuffers[i].mNumberChannels; + } + } + } + if (err != noErr || dataSize <= 0) { + sprintf( message, "RtAudio: OSX error getting output channels for device (%s).", info->name ); + error(RtError::DEBUG_WARNING); + return; + } + + free (bufferList); + if ( nStreams ) { + if ( maxChannels > 0 ) + info->maxOutputChannels = maxChannels; + if ( minChannels > 0 ) + info->minOutputChannels = minChannels; + } + + // Get input channel information. + bufferList = nil; + err = AudioDeviceGetPropertyInfo( info->id[0], 0, true, + kAudioDevicePropertyStreamConfiguration, + &dataSize, NULL ); + if (err == noErr && dataSize > 0) { + bufferList = (AudioBufferList *) malloc( dataSize ); + if (bufferList == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::DEBUG_WARNING); + return; + } + err = AudioDeviceGetProperty( info->id[0], 0, true, + kAudioDevicePropertyStreamConfiguration, + &dataSize, bufferList ); + if (err == noErr) { + maxChannels = 0; + minChannels = 1000; + nStreams = bufferList->mNumberBuffers; + for ( i=0; imBuffers[i].mNumberChannels < minChannels ) + minChannels = bufferList->mBuffers[i].mNumberChannels; + maxChannels += bufferList->mBuffers[i].mNumberChannels; + } + } + } + if (err != noErr || dataSize <= 0) { + sprintf( message, "RtAudio: OSX error getting input channels for device (%s).", info->name ); + error(RtError::DEBUG_WARNING); + return; + } + + free (bufferList); + if ( nStreams ) { + if ( maxChannels > 0 ) + info->maxInputChannels = maxChannels; + if ( minChannels > 0 ) + info->minInputChannels = minChannels; + } + + // If device opens for both playback and capture, we determine the channels. + if (info->maxOutputChannels > 0 && info->maxInputChannels > 0) { + info->hasDuplexSupport = true; + info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ? + info->maxInputChannels : info->maxOutputChannels; + info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ? + info->minInputChannels : info->minOutputChannels; + } + + // Probe the device sample rate and data format parameters. The + // core audio query mechanism is performed on a "stream" + // description, which can have a variable number of channels and + // apply to input or output only. + + // Create a stream description structure. + AudioStreamBasicDescription description; + dataSize = sizeof( AudioStreamBasicDescription ); + memset(&description, 0, sizeof(AudioStreamBasicDescription)); + bool isInput = false; + if ( info->maxOutputChannels == 0 ) isInput = true; + bool isDuplex = false; + if ( info->maxDuplexChannels > 0 ) isDuplex = true; + + // Determine the supported sample rates. + info->nSampleRates = 0; + for (i=0; iid[0], isInput, &description, isDuplex ) ) + info->sampleRates[info->nSampleRates++] = SAMPLE_RATES[i]; + } + + if (info->nSampleRates == 0) { + sprintf( message, "RtAudio: No supported sample rates found for OSX device (%s).", info->name ); + error(RtError::DEBUG_WARNING); + return; + } + + // Check for continuous sample rate support. + description.mSampleRate = kAudioStreamAnyRate; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) { + info->sampleRates[1] = info->sampleRates[info->nSampleRates-1]; + info->nSampleRates = -1; + } + + // Determine the supported data formats. + info->nativeFormats = 0; + description.mFormatID = kAudioFormatLinearPCM; + description.mBitsPerChannel = 8; + description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT8; + else { + description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT8; + } + + description.mBitsPerChannel = 16; + description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT16; + else { + description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT16; + } + + description.mBitsPerChannel = 32; + description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT32; + else { + description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT32; + } + + description.mBitsPerChannel = 24; + description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsAlignedHigh | kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT24; + else { + description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT24; + } + + description.mBitsPerChannel = 32; + description.mFormatFlags = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_FLOAT32; + else { + description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_FLOAT32; + } + + description.mBitsPerChannel = 64; + description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_FLOAT64; + else { + description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_FLOAT64; + } + + // Check that we have at least one supported format. + if (info->nativeFormats == 0) { + sprintf(message, "RtAudio: OSX PCM device (%s) data format not supported by RtAudio.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + info->probed = true; +} + +OSStatus callbackHandler(AudioDeviceID inDevice, + const AudioTimeStamp* inNow, + const AudioBufferList* inInputData, + const AudioTimeStamp* inInputTime, + AudioBufferList* outOutputData, + const AudioTimeStamp* inOutputTime, + void* infoPointer) +{ + CALLBACK_INFO *info = (CALLBACK_INFO *) infoPointer; + + RtAudio *object = (RtAudio *) info->object; + try { + object->callbackEvent( info->streamId, inDevice, (void *)inInputData, (void *)outOutputData ); + } + catch (RtError &exception) { + fprintf(stderr, "\nCallback handler error (%s)!\n\n", exception.getMessage()); + return kAudioHardwareUnspecifiedError; + } + + return kAudioHardwareNoError; +} + +/* +OSStatus deviceListener(AudioDeviceID inDevice, + UInt32 channel, + Boolean isInput, + AudioDevicePropertyID propertyID, + void* infoPointer) +{ + CALLBACK_INFO *info = (CALLBACK_INFO *) infoPointer; + + RtAudio *object = (RtAudio *) info->object; + try { + object->settingChange( info->streamId ); + } + catch (RtError &exception) { + fprintf(stderr, "\nDevice listener error (%s)!\n\n", exception.getMessage()); + return kAudioHardwareUnspecifiedError; + } + + return kAudioHardwareNoError; +} +*/ + +bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, + STREAM_MODE mode, int channels, + int sampleRate, RTAUDIO_FORMAT format, + int *bufferSize, int numberOfBuffers) +{ + // Check to make sure we don't already have a stream accessing this device. + RTAUDIO_STREAM *streamPtr; + std::map::const_iterator i; + for ( i=streams.begin(); i!=streams.end(); ++i ) { + streamPtr = (RTAUDIO_STREAM *) i->second; + if ( streamPtr->device[0] == device || streamPtr->device[1] == device ) { + sprintf(message, "RtAudio: no current OS X support for multiple streams accessing the same device!"); + error(RtError::WARNING); + return FAILURE; + } + } + + // Setup for stream mode. + bool isInput = false; + AudioDeviceID id = devices[device].id[0]; + if ( mode == INPUT ) isInput = true; + + // Search for a stream which contains the desired number of channels. + OSStatus err = noErr; + UInt32 dataSize; + unsigned int deviceChannels, nStreams; + UInt32 iChannel = 0, iStream = 0; + AudioBufferList *bufferList = nil; + err = AudioDeviceGetPropertyInfo( id, 0, isInput, + kAudioDevicePropertyStreamConfiguration, + &dataSize, NULL ); + + if (err == noErr && dataSize > 0) { + bufferList = (AudioBufferList *) malloc( dataSize ); + if (bufferList == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + err = AudioDeviceGetProperty( id, 0, isInput, + kAudioDevicePropertyStreamConfiguration, + &dataSize, bufferList ); + + if (err == noErr) { + stream->deInterleave[mode] = false; + nStreams = bufferList->mNumberBuffers; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels >= (unsigned int) channels ) break; + iChannel += bufferList->mBuffers[iStream].mNumberChannels; + } + // If we didn't find a single stream above, see if we can meet + // the channel specification in mono mode (i.e. using separate + // non-interleaved buffers). This can only work if there are N + // consecutive one-channel streams, where N is the number of + // desired channels. + iChannel = 0; + if ( iStream >= nStreams && nStreams >= (unsigned int) channels ) { + int counter = 0; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels == 1 ) + counter++; + else + counter = 0; + if ( counter == channels ) { + iStream -= channels - 1; + iChannel -= channels - 1; + stream->deInterleave[mode] = true; + break; + } + iChannel += bufferList->mBuffers[iStream].mNumberChannels; + } + } + } + } + if (err != noErr || dataSize <= 0) { + if ( bufferList ) free( bufferList ); + sprintf( message, "RtAudio: OSX error getting channels for device (%s).", devices[device].name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + if (iStream >= nStreams) { + free (bufferList); + sprintf( message, "RtAudio: unable to find OSX audio stream on device (%s) for requested channels (%d).", + devices[device].name, channels ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + // This is ok even for mono mode ... it gets updated later. + deviceChannels = bufferList->mBuffers[iStream].mNumberChannels; + free (bufferList); + + // Determine the buffer size. + AudioValueRange bufferRange; + dataSize = sizeof(AudioValueRange); + err = AudioDeviceGetProperty( id, 0, isInput, + kAudioDevicePropertyBufferSizeRange, + &dataSize, &bufferRange); + if (err != noErr) { + sprintf( message, "RtAudio: OSX error getting buffer size range for device (%s).", + devices[device].name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + long bufferBytes = *bufferSize * deviceChannels * formatBytes(RTAUDIO_FLOAT32); + if (bufferRange.mMinimum > bufferBytes) bufferBytes = (int) bufferRange.mMinimum; + else if (bufferRange.mMaximum < bufferBytes) bufferBytes = (int) bufferRange.mMaximum; + + // Set the buffer size. For mono mode, I'm assuming we only need to + // make this setting for the first channel. + UInt32 theSize = (UInt32) bufferBytes; + dataSize = sizeof( UInt32); + err = AudioDeviceSetProperty(id, NULL, 0, isInput, + kAudioDevicePropertyBufferSize, + dataSize, &theSize); + if (err != noErr) { + sprintf( message, "RtAudio: OSX error setting the buffer size for device (%s).", + devices[device].name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + *bufferSize = bufferBytes / ( deviceChannels * formatBytes(RTAUDIO_FLOAT32) ); + if ( stream->mode == OUTPUT && mode == INPUT && *bufferSize != stream->bufferSize ) { + sprintf( message, "RtAudio: OSX error setting buffer size for duplex stream on device (%s).", + devices[device].name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + stream->bufferSize = *bufferSize; + stream->nBuffers = 1; + + // Set the stream format description. Do for each channel in mono mode. + AudioStreamBasicDescription description; + dataSize = sizeof( AudioStreamBasicDescription ); + if ( stream->deInterleave[mode] ) nStreams = channels; + else nStreams = 1; + for ( unsigned int i=0; idoByteSwap[mode] = false; + if ( !description.mFormatFlags & kLinearPCMFormatFlagIsBigEndian ) + stream->doByteSwap[mode] = true; + + // From the CoreAudio documentation, PCM data must be supplied as + // 32-bit floats. + stream->userFormat = format; + stream->deviceFormat[mode] = RTAUDIO_FLOAT32; + + if ( stream->deInterleave[mode] ) + stream->nDeviceChannels[mode] = channels; + else + stream->nDeviceChannels[mode] = description.mChannelsPerFrame; + stream->nUserChannels[mode] = channels; + + // Set handle and flags for buffer conversion. + stream->handle[mode] = iStream; + stream->doConvertBuffer[mode] = false; + if (stream->userFormat != stream->deviceFormat[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] > 1 && stream->deInterleave[mode]) + stream->doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers. + if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { + + long buffer_bytes; + if (stream->nUserChannels[0] >= stream->nUserChannels[1]) + buffer_bytes = stream->nUserChannels[0]; + else + buffer_bytes = stream->nUserChannels[1]; + + buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); + if (stream->userBuffer) free(stream->userBuffer); + stream->userBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->userBuffer == NULL) + goto memory_error; + } + + if ( stream->deInterleave[mode] ) { + + long buffer_bytes; + bool makeBuffer = true; + if ( mode == OUTPUT ) + buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + else { // mode == INPUT + buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); + if ( stream->mode == OUTPUT && stream->deviceBuffer ) { + long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + if ( buffer_bytes < bytes_out ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + buffer_bytes *= *bufferSize; + if (stream->deviceBuffer) free(stream->deviceBuffer); + stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->deviceBuffer == NULL) + goto memory_error; + + // If not de-interleaving, we point stream->deviceBuffer to the + // OS X supplied device buffer before doing any necessary data + // conversions. This presents a problem if we have a duplex + // stream using one device which needs de-interleaving and + // another device which doesn't. So, save a pointer to our own + // device buffer in the CALLBACK_INFO structure. + stream->callbackInfo.buffers = stream->deviceBuffer; + } + } + + stream->sampleRate = sampleRate; + stream->device[mode] = device; + stream->state = STREAM_STOPPED; + stream->callbackInfo.object = (void *) this; + stream->callbackInfo.waitTime = (unsigned long) (200000.0 * stream->bufferSize / stream->sampleRate); + stream->callbackInfo.device[mode] = id; + if ( stream->mode == OUTPUT && mode == INPUT && stream->device[0] == device ) + // Only one callback procedure per device. + stream->mode = DUPLEX; + else { + err = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream->callbackInfo ); + if (err != noErr) { + sprintf( message, "RtAudio: OSX error setting callback for device (%s).", devices[device].name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + if ( stream->mode == OUTPUT && mode == INPUT ) + stream->mode = DUPLEX; + else + stream->mode = mode; + } + + // If we wanted to use property listeners, they would be setup here. + + return SUCCESS; + + memory_error: + if (stream->userBuffer) { + free(stream->userBuffer); + stream->userBuffer = 0; + } + sprintf(message, "RtAudio: OSX error allocating buffer memory (%s).", devices[device].name); + error(RtError::WARNING); + return FAILURE; +} + +void RtAudio :: cancelStreamCallback(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + if (stream->callbackInfo.usingCallback) { + + if (stream->state == STREAM_RUNNING) + stopStream( streamId ); + + MUTEX_LOCK(&stream->mutex); + + stream->callbackInfo.usingCallback = false; + stream->callbackInfo.userData = NULL; + stream->state = STREAM_STOPPED; + stream->callbackInfo.callback = NULL; + + MUTEX_UNLOCK(&stream->mutex); + } +} + +void RtAudio :: closeStream(int streamId) +{ + // We don't want an exception to be thrown here because this + // function is called by our class destructor. So, do our own + // streamId check. + if ( streams.find( streamId ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtError::WARNING); + return; + } + + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + + AudioDeviceID id; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + id = devices[stream->device[0]].id[0]; + if (stream->state == STREAM_RUNNING) + AudioDeviceStop( id, callbackHandler ); + AudioDeviceRemoveIOProc( id, callbackHandler ); + } + + if (stream->mode == INPUT || ( stream->mode == DUPLEX && stream->device[0] != stream->device[1]) ) { + id = devices[stream->device[1]].id[0]; + if (stream->state == STREAM_RUNNING) + AudioDeviceStop( id, callbackHandler ); + AudioDeviceRemoveIOProc( id, callbackHandler ); + } + + pthread_mutex_destroy(&stream->mutex); + + if (stream->userBuffer) + free(stream->userBuffer); + + if ( stream->deInterleave[0] || stream->deInterleave[1] ) + free(stream->callbackInfo.buffers); + + free(stream); + streams.erase(streamId); +} + +void RtAudio :: startStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_RUNNING) + goto unlock; + + OSStatus err; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + err = AudioDeviceStart(devices[stream->device[0]].id[0], callbackHandler); + if (err != noErr) { + sprintf(message, "RtAudio: OSX error starting callback procedure on device (%s).", + devices[stream->device[0]].name); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + if (stream->mode == INPUT || ( stream->mode == DUPLEX && stream->device[0] != stream->device[1]) ) { + + err = AudioDeviceStart(devices[stream->device[1]].id[0], callbackHandler); + if (err != noErr) { + sprintf(message, "RtAudio: OSX error starting input callback procedure on device (%s).", + devices[stream->device[0]].name); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + stream->callbackInfo.streamId = streamId; + stream->state = STREAM_RUNNING; + stream->callbackInfo.blockTick = true; + stream->callbackInfo.stopStream = false; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: stopStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + OSStatus err; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + err = AudioDeviceStop(devices[stream->device[0]].id[0], callbackHandler); + if (err != noErr) { + sprintf(message, "RtAudio: OSX error stopping callback procedure on device (%s).", + devices[stream->device[0]].name); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + if (stream->mode == INPUT || ( stream->mode == DUPLEX && stream->device[0] != stream->device[1]) ) { + + err = AudioDeviceStop(devices[stream->device[1]].id[0], callbackHandler); + if (err != noErr) { + sprintf(message, "RtAudio: OSX error stopping input callback procedure on device (%s).", + devices[stream->device[0]].name); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: abortStream(int streamId) +{ + stopStream( streamId ); +} + +// I don't know how this function can be implemented. +int RtAudio :: streamWillBlock(int streamId) +{ + sprintf(message, "RtAudio: streamWillBlock() cannot be implemented for OS X."); + error(RtError::WARNING); + return 0; +} + +void RtAudio :: tickStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + if (stream->state == STREAM_STOPPED) + return; + + if (stream->callbackInfo.usingCallback) { + sprintf(message, "RtAudio: tickStream() should not be used when a callback function is set!"); + error(RtError::WARNING); + return; + } + + // Block waiting here until the user data is processed in callbackEvent(). + while ( stream->callbackInfo.blockTick ) + usleep(stream->callbackInfo.waitTime); + + MUTEX_LOCK(&stream->mutex); + + stream->callbackInfo.blockTick = true; + + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: callbackEvent( int streamId, DEVICE_ID deviceId, void *inData, void *outData ) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + CALLBACK_INFO *info; + AudioBufferList *inBufferList = (AudioBufferList *) inData; + AudioBufferList *outBufferList = (AudioBufferList *) outData; + + if (stream->state == STREAM_STOPPED) return; + + info = (CALLBACK_INFO *) &stream->callbackInfo; + if ( !info->usingCallback ) { + // Block waiting here until we get new user data in tickStream(). + while ( !info->blockTick ) + usleep(info->waitTime); + } + else if ( info->stopStream ) { + // Check if the stream should be stopped (via the previous user + // callback return value). We stop the stream here, rather than + // after the function call, so that output data can first be + // processed. + this->stopStream(info->streamId); + return; + } + + MUTEX_LOCK(&stream->mutex); + + if ( stream->mode == INPUT || ( stream->mode == DUPLEX && deviceId == info->device[1] ) ) { + + if (stream->doConvertBuffer[1]) { + + if ( stream->deInterleave[1] ) { + stream->deviceBuffer = (char *) stream->callbackInfo.buffers; + int bufferBytes = inBufferList->mBuffers[stream->handle[1]].mDataByteSize; + for ( int i=0; inDeviceChannels[1]; i++ ) { + memcpy(&stream->deviceBuffer[i*bufferBytes], + inBufferList->mBuffers[stream->handle[1]+i].mData, bufferBytes ); + } + } + else + stream->deviceBuffer = (char *) inBufferList->mBuffers[stream->handle[1]].mData; + + if ( stream->doByteSwap[1] ) + byteSwapBuffer(stream->deviceBuffer, + stream->bufferSize * stream->nDeviceChannels[1], + stream->deviceFormat[1]); + convertStreamBuffer(stream, INPUT); + + } + else { + memcpy(stream->userBuffer, + inBufferList->mBuffers[stream->handle[1]].mData, + inBufferList->mBuffers[stream->handle[1]].mDataByteSize ); + + if (stream->doByteSwap[1]) + byteSwapBuffer(stream->userBuffer, + stream->bufferSize * stream->nUserChannels[1], + stream->userFormat); + } + } + + // Don't invoke the user callback if duplex mode, the input/output + // devices are different, and this function is called for the output + // device. + if ( info->usingCallback && (stream->mode != DUPLEX || deviceId == info->device[1] ) ) { + RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) info->callback; + info->stopStream = callback(stream->userBuffer, stream->bufferSize, info->userData); + } + + if ( stream->mode == OUTPUT || ( stream->mode == DUPLEX && deviceId == info->device[0] ) ) { + + if (stream->doConvertBuffer[0]) { + + if ( !stream->deInterleave[0] ) + stream->deviceBuffer = (char *) outBufferList->mBuffers[stream->handle[0]].mData; + else + stream->deviceBuffer = (char *) stream->callbackInfo.buffers; + + convertStreamBuffer(stream, OUTPUT); + if ( stream->doByteSwap[0] ) + byteSwapBuffer(stream->deviceBuffer, + stream->bufferSize * stream->nDeviceChannels[0], + stream->deviceFormat[0]); + + if ( stream->deInterleave[0] ) { + int bufferBytes = outBufferList->mBuffers[stream->handle[0]].mDataByteSize; + for ( int i=0; inDeviceChannels[0]; i++ ) { + memcpy(outBufferList->mBuffers[stream->handle[0]+i].mData, + &stream->deviceBuffer[i*bufferBytes], bufferBytes ); + } + } + + } + else { + if (stream->doByteSwap[0]) + byteSwapBuffer(stream->userBuffer, + stream->bufferSize * stream->nUserChannels[0], + stream->userFormat); + + memcpy(outBufferList->mBuffers[stream->handle[0]].mData, + stream->userBuffer, + outBufferList->mBuffers[stream->handle[0]].mDataByteSize ); + } + } + + if ( !info->usingCallback && (stream->mode != DUPLEX || deviceId == info->device[1] ) ) + info->blockTick = false; + + MUTEX_UNLOCK(&stream->mutex); + +} + +void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + stream->callbackInfo.callback = (void *) callback; + stream->callbackInfo.userData = userData; + stream->callbackInfo.usingCallback = true; +} + +//******************** End of __MACOSX_CORE__ *********************// + +#elif defined(__LINUX_ALSA__) + +#define MAX_DEVICES 16 + +void RtAudio :: initialize(void) +{ + int card, result, device; + char name[32]; + const char *cardId; + char deviceNames[MAX_DEVICES][32]; + snd_ctl_t *handle; + snd_ctl_card_info_t *info; + snd_ctl_card_info_alloca(&info); + + // Count cards and devices + nDevices = 0; + card = -1; + snd_card_next(&card); + while ( card >= 0 ) { + sprintf(name, "hw:%d", card); + result = snd_ctl_open(&handle, name, 0); + if (result < 0) { + sprintf(message, "RtAudio: ALSA control open (%i): %s.", card, snd_strerror(result)); + error(RtError::DEBUG_WARNING); + goto next_card; + } + result = snd_ctl_card_info(handle, info); + if (result < 0) { + sprintf(message, "RtAudio: ALSA control hardware info (%i): %s.", card, snd_strerror(result)); + error(RtError::DEBUG_WARNING); + goto next_card; + } + cardId = snd_ctl_card_info_get_id(info); + device = -1; + while (1) { + result = snd_ctl_pcm_next_device(handle, &device); + if (result < 0) { + sprintf(message, "RtAudio: ALSA control next device (%i): %s.", card, snd_strerror(result)); + error(RtError::DEBUG_WARNING); + break; + } + if (device < 0) + break; + if ( strlen(cardId) ) + sprintf( deviceNames[nDevices++], "hw:%s,%d", cardId, device ); + else + sprintf( deviceNames[nDevices++], "hw:%d,%d", card, device ); + if ( nDevices > MAX_DEVICES ) break; + } + if ( nDevices > MAX_DEVICES ) break; + next_card: + snd_ctl_close(handle); + snd_card_next(&card); + } + + if (nDevices == 0) return; + + // Allocate the RTAUDIO_DEVICE structures. + devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); + if (devices == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + + // Write device ascii identifiers to device structures and then + // probe the device capabilities. + for (int i=0; iname, 32 ); + card = strtok(name, ","); + err = snd_ctl_open(&chandle, card, 0); + if (err < 0) { + sprintf(message, "RtAudio: ALSA control open (%s): %s.", card, snd_strerror(err)); + error(RtError::DEBUG_WARNING); + return; + } + unsigned int dev = (unsigned int) atoi( strtok(NULL, ",") ); + + // First try for playback + stream = SND_PCM_STREAM_PLAYBACK; + snd_pcm_info_set_device(pcminfo, dev); + snd_pcm_info_set_subdevice(pcminfo, 0); + snd_pcm_info_set_stream(pcminfo, stream); + + if ((err = snd_ctl_pcm_info(chandle, pcminfo)) < 0) { + if (err == -ENOENT) { + sprintf(message, "RtAudio: ALSA pcm device (%s) doesn't handle output!", info->name); + error(RtError::DEBUG_WARNING); + } + else { + sprintf(message, "RtAudio: ALSA snd_ctl_pcm_info error for device (%s) output: %s", + info->name, snd_strerror(err)); + error(RtError::DEBUG_WARNING); + } + goto capture_probe; + } + + err = snd_pcm_open(&handle, info->name, stream, open_mode | SND_PCM_NONBLOCK ); + if (err < 0) { + if ( err == EBUSY ) + sprintf(message, "RtAudio: ALSA pcm playback device (%s) is busy: %s.", + info->name, snd_strerror(err)); + else + sprintf(message, "RtAudio: ALSA pcm playback open (%s) error: %s.", + info->name, snd_strerror(err)); + error(RtError::DEBUG_WARNING); + goto capture_probe; + } + + // We have an open device ... allocate the parameter structure. + err = snd_pcm_hw_params_any(handle, params); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA hardware probe error (%s): %s.", + info->name, snd_strerror(err)); + error(RtError::WARNING); + goto capture_probe; + } + + // Get output channel information. + info->minOutputChannels = snd_pcm_hw_params_get_channels_min(params); + info->maxOutputChannels = snd_pcm_hw_params_get_channels_max(params); + + snd_pcm_close(handle); + + capture_probe: + // Now try for capture + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream(pcminfo, stream); + + err = snd_ctl_pcm_info(chandle, pcminfo); + snd_ctl_close(chandle); + if ( err < 0 ) { + if (err == -ENOENT) { + sprintf(message, "RtAudio: ALSA pcm device (%s) doesn't handle input!", info->name); + error(RtError::DEBUG_WARNING); + } + else { + sprintf(message, "RtAudio: ALSA snd_ctl_pcm_info error for device (%s) input: %s", + info->name, snd_strerror(err)); + error(RtError::DEBUG_WARNING); + } + if (info->maxOutputChannels == 0) + // didn't open for playback either ... device invalid + return; + goto probe_parameters; + } + + err = snd_pcm_open(&handle, info->name, stream, open_mode | SND_PCM_NONBLOCK); + if (err < 0) { + if ( err == EBUSY ) + sprintf(message, "RtAudio: ALSA pcm capture device (%s) is busy: %s.", + info->name, snd_strerror(err)); + else + sprintf(message, "RtAudio: ALSA pcm capture open (%s) error: %s.", + info->name, snd_strerror(err)); + error(RtError::DEBUG_WARNING); + if (info->maxOutputChannels == 0) + // didn't open for playback either ... device invalid + return; + goto probe_parameters; + } + + // We have an open capture device ... allocate the parameter structure. + err = snd_pcm_hw_params_any(handle, params); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA hardware probe error (%s): %s.", + info->name, snd_strerror(err)); + error(RtError::WARNING); + if (info->maxOutputChannels > 0) + goto probe_parameters; + else + return; + } + + // Get input channel information. + info->minInputChannels = snd_pcm_hw_params_get_channels_min(params); + info->maxInputChannels = snd_pcm_hw_params_get_channels_max(params); + + snd_pcm_close(handle); + + // If device opens for both playback and capture, we determine the channels. + if (info->maxOutputChannels == 0 || info->maxInputChannels == 0) + goto probe_parameters; + + info->hasDuplexSupport = true; + info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ? + info->maxInputChannels : info->maxOutputChannels; + info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ? + info->minInputChannels : info->minOutputChannels; + + probe_parameters: + // At this point, we just need to figure out the supported data + // formats and sample rates. We'll proceed by opening the device in + // the direction with the maximum number of channels, or playback if + // they are equal. This might limit our sample rate options, but so + // be it. + + if (info->maxOutputChannels >= info->maxInputChannels) + stream = SND_PCM_STREAM_PLAYBACK; + else + stream = SND_PCM_STREAM_CAPTURE; + + err = snd_pcm_open(&handle, info->name, stream, open_mode); + if (err < 0) { + sprintf(message, "RtAudio: ALSA pcm (%s) won't reopen during probe: %s.", + info->name, snd_strerror(err)); + error(RtError::WARNING); + return; + } + + // We have an open device ... allocate the parameter structure. + err = snd_pcm_hw_params_any(handle, params); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA hardware reopen probe error (%s): %s.", + info->name, snd_strerror(err)); + error(RtError::WARNING); + return; + } + + // Test a non-standard sample rate to see if continuous rate is supported. + int dir = 0; + if (snd_pcm_hw_params_test_rate(handle, params, 35500, dir) == 0) { + // It appears that continuous sample rate support is available. + info->nSampleRates = -1; + info->sampleRates[0] = snd_pcm_hw_params_get_rate_min(params, &dir); + info->sampleRates[1] = snd_pcm_hw_params_get_rate_max(params, &dir); + } + else { + // No continuous rate support ... test our discrete set of sample rate values. + info->nSampleRates = 0; + for (int i=0; isampleRates[info->nSampleRates] = SAMPLE_RATES[i]; + info->nSampleRates++; + } + } + if (info->nSampleRates == 0) { + snd_pcm_close(handle); + return; + } + } + + // Probe the supported data formats ... we don't care about endian-ness just yet + snd_pcm_format_t format; + info->nativeFormats = 0; + format = SND_PCM_FORMAT_S8; + if (snd_pcm_hw_params_test_format(handle, params, format) == 0) + info->nativeFormats |= RTAUDIO_SINT8; + format = SND_PCM_FORMAT_S16; + if (snd_pcm_hw_params_test_format(handle, params, format) == 0) + info->nativeFormats |= RTAUDIO_SINT16; + format = SND_PCM_FORMAT_S24; + if (snd_pcm_hw_params_test_format(handle, params, format) == 0) + info->nativeFormats |= RTAUDIO_SINT24; + format = SND_PCM_FORMAT_S32; + if (snd_pcm_hw_params_test_format(handle, params, format) == 0) + info->nativeFormats |= RTAUDIO_SINT32; + format = SND_PCM_FORMAT_FLOAT; + if (snd_pcm_hw_params_test_format(handle, params, format) == 0) + info->nativeFormats |= RTAUDIO_FLOAT32; + format = SND_PCM_FORMAT_FLOAT64; + if (snd_pcm_hw_params_test_format(handle, params, format) == 0) + info->nativeFormats |= RTAUDIO_FLOAT64; + + // Check that we have at least one supported format + if (info->nativeFormats == 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA PCM device (%s) data format not supported by RtAudio.", + info->name); + error(RtError::WARNING); + return; + } + + // That's all ... close the device and return + snd_pcm_close(handle); + info->probed = true; + return; +} + +bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, + STREAM_MODE mode, int channels, + int sampleRate, RTAUDIO_FORMAT format, + int *bufferSize, int numberOfBuffers) +{ +#if defined(__RTAUDIO_DEBUG__) + snd_output_t *out; + snd_output_stdio_attach(&out, stderr, 0); +#endif + + // I'm not using the "plug" interface ... too much inconsistent behavior. + const char *name = devices[device].name; + + snd_pcm_stream_t alsa_stream; + if (mode == OUTPUT) + alsa_stream = SND_PCM_STREAM_PLAYBACK; + else + alsa_stream = SND_PCM_STREAM_CAPTURE; + + int err; + snd_pcm_t *handle; + int alsa_open_mode = SND_PCM_ASYNC; + err = snd_pcm_open(&handle, name, alsa_stream, alsa_open_mode); + if (err < 0) { + sprintf(message,"RtAudio: ALSA pcm device (%s) won't open: %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + + // Fill the parameter structure. + snd_pcm_hw_params_t *hw_params; + snd_pcm_hw_params_alloca(&hw_params); + err = snd_pcm_hw_params_any(handle, hw_params); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error getting parameter handle (%s): %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf(stderr, "\nRtAudio: ALSA dump hardware params just after device open:\n\n"); + snd_pcm_hw_params_dump(hw_params, out); +#endif + + + // Set access ... try interleaved access first, then non-interleaved + if ( !snd_pcm_hw_params_test_access( handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED) ) { + err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); + } + else if ( !snd_pcm_hw_params_test_access( handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED) ) { + err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED); + stream->deInterleave[mode] = true; + } + else { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA device (%s) access not supported by RtAudio.", name); + error(RtError::WARNING); + return FAILURE; + } + + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error setting access ( (%s): %s.", name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + + // Determine how to set the device format. + stream->userFormat = format; + snd_pcm_format_t device_format; + + if (format == RTAUDIO_SINT8) + device_format = SND_PCM_FORMAT_S8; + else if (format == RTAUDIO_SINT16) + device_format = SND_PCM_FORMAT_S16; + else if (format == RTAUDIO_SINT24) + device_format = SND_PCM_FORMAT_S24; + else if (format == RTAUDIO_SINT32) + device_format = SND_PCM_FORMAT_S32; + else if (format == RTAUDIO_FLOAT32) + device_format = SND_PCM_FORMAT_FLOAT; + else if (format == RTAUDIO_FLOAT64) + device_format = SND_PCM_FORMAT_FLOAT64; + + if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { + stream->deviceFormat[mode] = format; + goto set_format; + } + + // The user requested format is not natively supported by the device. + device_format = SND_PCM_FORMAT_FLOAT64; + if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { + stream->deviceFormat[mode] = RTAUDIO_FLOAT64; + goto set_format; + } + + device_format = SND_PCM_FORMAT_FLOAT; + if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { + stream->deviceFormat[mode] = RTAUDIO_FLOAT32; + goto set_format; + } + + device_format = SND_PCM_FORMAT_S32; + if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { + stream->deviceFormat[mode] = RTAUDIO_SINT32; + goto set_format; + } + + device_format = SND_PCM_FORMAT_S24; + if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { + stream->deviceFormat[mode] = RTAUDIO_SINT24; + goto set_format; + } + + device_format = SND_PCM_FORMAT_S16; + if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { + stream->deviceFormat[mode] = RTAUDIO_SINT16; + goto set_format; + } + + device_format = SND_PCM_FORMAT_S8; + if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { + stream->deviceFormat[mode] = RTAUDIO_SINT8; + goto set_format; + } + + // If we get here, no supported format was found. + sprintf(message,"RtAudio: ALSA pcm device (%s) data format not supported by RtAudio.", name); + snd_pcm_close(handle); + error(RtError::WARNING); + return FAILURE; + + set_format: + err = snd_pcm_hw_params_set_format(handle, hw_params, device_format); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error setting format (%s): %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + + // Determine whether byte-swaping is necessary. + stream->doByteSwap[mode] = false; + if (device_format != SND_PCM_FORMAT_S8) { + err = snd_pcm_format_cpu_endian(device_format); + if (err == 0) + stream->doByteSwap[mode] = true; + else if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error getting format endian-ness (%s): %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + } + + // Set the sample rate. + err = snd_pcm_hw_params_set_rate(handle, hw_params, (unsigned int)sampleRate, 0); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error setting sample rate (%d) on device (%s): %s.", + sampleRate, name, snd_strerror(err)); + error(RtError::WARNING); + 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; + int device_channels = snd_pcm_hw_params_get_channels_max(hw_params); + if (device_channels < channels) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: channels (%d) not supported by device (%s).", + channels, name); + error(RtError::WARNING); + return FAILURE; + } + + device_channels = snd_pcm_hw_params_get_channels_min(hw_params); + if (device_channels < channels) device_channels = channels; + stream->nDeviceChannels[mode] = device_channels; + + // Set the device channels. + err = snd_pcm_hw_params_set_channels(handle, hw_params, device_channels); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error setting channels (%d) on device (%s): %s.", + device_channels, name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + + // Set the buffer number, which in ALSA is referred to as the "period". + int dir; + int periods = numberOfBuffers; + // Even though the hardware might allow 1 buffer, it won't work reliably. + if (periods < 2) periods = 2; + err = snd_pcm_hw_params_get_periods_min(hw_params, &dir); + if (err > periods) periods = err; + err = snd_pcm_hw_params_get_periods_max(hw_params, &dir); + if (err < periods) periods = err; + + err = snd_pcm_hw_params_set_periods(handle, hw_params, periods, 0); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error setting periods (%s): %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + + // Set the buffer (or period) size. + err = snd_pcm_hw_params_get_period_size_min(hw_params, &dir); + if (err > *bufferSize) *bufferSize = err; + + err = snd_pcm_hw_params_set_period_size(handle, hw_params, *bufferSize, 0); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error setting period size (%s): %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + if ( stream->mode == OUTPUT && mode == INPUT && *bufferSize != stream->bufferSize ) { + sprintf( message, "RtAudio: ALSA error setting buffer size for duplex stream on device (%s).", + name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + stream->bufferSize = *bufferSize; + + // Install the hardware configuration + err = snd_pcm_hw_params(handle, hw_params); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error installing hardware configuration (%s): %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf(stderr, "\nRtAudio: ALSA dump hardware params after installation:\n\n"); + snd_pcm_hw_params_dump(hw_params, out); +#endif + + /* + // Install the software configuration + snd_pcm_sw_params_t *sw_params = NULL; + snd_pcm_sw_params_alloca(&sw_params); + snd_pcm_sw_params_current(handle, sw_params); + err = snd_pcm_sw_params(handle, sw_params); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error installing software configuration (%s): %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + */ + + // Set handle and flags for buffer conversion + stream->handle[mode] = handle; + stream->doConvertBuffer[mode] = false; + if (stream->userFormat != stream->deviceFormat[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] > 1 && stream->deInterleave[mode]) + stream->doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { + + long buffer_bytes; + if (stream->nUserChannels[0] >= stream->nUserChannels[1]) + buffer_bytes = stream->nUserChannels[0]; + else + buffer_bytes = stream->nUserChannels[1]; + + buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); + if (stream->userBuffer) free(stream->userBuffer); + stream->userBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->userBuffer == NULL) + goto memory_error; + } + + if ( stream->doConvertBuffer[mode] ) { + + long buffer_bytes; + bool makeBuffer = true; + if ( mode == OUTPUT ) + buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + else { // mode == INPUT + buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); + if ( stream->mode == OUTPUT && stream->deviceBuffer ) { + long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + if ( buffer_bytes < bytes_out ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + buffer_bytes *= *bufferSize; + if (stream->deviceBuffer) free(stream->deviceBuffer); + stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->deviceBuffer == NULL) + goto memory_error; + } + } + + stream->device[mode] = device; + stream->state = STREAM_STOPPED; + if ( stream->mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream->mode = DUPLEX; + else + stream->mode = mode; + stream->nBuffers = periods; + stream->sampleRate = sampleRate; + + return SUCCESS; + + memory_error: + if (stream->handle[0]) { + snd_pcm_close(stream->handle[0]); + stream->handle[0] = 0; + } + if (stream->handle[1]) { + snd_pcm_close(stream->handle[1]); + stream->handle[1] = 0; + } + if (stream->userBuffer) { + free(stream->userBuffer); + stream->userBuffer = 0; + } + sprintf(message, "RtAudio: ALSA error allocating buffer memory (%s).", name); + error(RtError::WARNING); + return FAILURE; +} + +void RtAudio :: closeStream(int streamId) +{ + // We don't want an exception to be thrown here because this + // function is called by our class destructor. So, do our own + // streamId check. + if ( streams.find( streamId ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtError::WARNING); + return; + } + + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + + if (stream->callbackInfo.usingCallback) { + pthread_cancel(stream->callbackInfo.thread); + pthread_join(stream->callbackInfo.thread, NULL); + } + + if (stream->state == STREAM_RUNNING) { + if (stream->mode == OUTPUT || stream->mode == DUPLEX) + snd_pcm_drop(stream->handle[0]); + if (stream->mode == INPUT || stream->mode == DUPLEX) + snd_pcm_drop(stream->handle[1]); + } + + pthread_mutex_destroy(&stream->mutex); + + if (stream->handle[0]) + snd_pcm_close(stream->handle[0]); + + if (stream->handle[1]) + snd_pcm_close(stream->handle[1]); + + if (stream->userBuffer) + free(stream->userBuffer); + + if (stream->deviceBuffer) + free(stream->deviceBuffer); + + free(stream); + streams.erase(streamId); +} + +void RtAudio :: startStream(int streamId) +{ + // This method calls snd_pcm_prepare if the device isn't already in that state. + + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_RUNNING) + goto unlock; + + int err; + snd_pcm_state_t state; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + state = snd_pcm_state(stream->handle[0]); + if (state != SND_PCM_STATE_PREPARED) { + err = snd_pcm_prepare(stream->handle[0]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error preparing pcm device (%s): %s.", + devices[stream->device[0]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + state = snd_pcm_state(stream->handle[1]); + if (state != SND_PCM_STATE_PREPARED) { + err = snd_pcm_prepare(stream->handle[1]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error preparing pcm device (%s): %s.", + devices[stream->device[1]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + } + stream->state = STREAM_RUNNING; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: stopStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + int err; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + err = snd_pcm_drain(stream->handle[0]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", + devices[stream->device[0]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + err = snd_pcm_drain(stream->handle[1]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", + devices[stream->device[1]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: abortStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + int err; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + err = snd_pcm_drop(stream->handle[0]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", + devices[stream->device[0]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + err = snd_pcm_drop(stream->handle[1]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", + devices[stream->device[1]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +int RtAudio :: streamWillBlock(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + int err = 0, frames = 0; + if (stream->state == STREAM_STOPPED) + goto unlock; + + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + err = snd_pcm_avail_update(stream->handle[0]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error getting available frames for device (%s): %s.", + devices[stream->device[0]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + frames = err; + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + err = snd_pcm_avail_update(stream->handle[1]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error getting available frames for device (%s): %s.", + devices[stream->device[1]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + if (frames > err) frames = err; + } + + frames = stream->bufferSize - frames; + if (frames < 0) frames = 0; + + unlock: + MUTEX_UNLOCK(&stream->mutex); + return frames; +} + +void RtAudio :: tickStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + int stopStream = 0; + if (stream->state == STREAM_STOPPED) { + if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds + return; + } + else if (stream->callbackInfo.usingCallback) { + RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback; + stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData); + } + + MUTEX_LOCK(&stream->mutex); + + // The state might change while waiting on a mutex. + if (stream->state == STREAM_STOPPED) + goto unlock; + + int err; + char *buffer; + int channels; + RTAUDIO_FORMAT format; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + // Setup parameters and do buffer conversion if necessary. + if (stream->doConvertBuffer[0]) { + convertStreamBuffer(stream, OUTPUT); + buffer = stream->deviceBuffer; + channels = stream->nDeviceChannels[0]; + format = stream->deviceFormat[0]; + } + else { + buffer = stream->userBuffer; + channels = stream->nUserChannels[0]; + format = stream->userFormat; + } + + // Do byte swapping if necessary. + if (stream->doByteSwap[0]) + byteSwapBuffer(buffer, stream->bufferSize * channels, format); + + // Write samples to device in interleaved/non-interleaved format. + if (stream->deInterleave[0]) { + void *bufs[channels]; + size_t offset = stream->bufferSize * formatBytes(format); + for (int i=0; ihandle[0], bufs, stream->bufferSize); + } + else + err = snd_pcm_writei(stream->handle[0], buffer, stream->bufferSize); + + if (err < stream->bufferSize) { + // Either an error or underrun occured. + if (err == -EPIPE) { + snd_pcm_state_t state = snd_pcm_state(stream->handle[0]); + if (state == SND_PCM_STATE_XRUN) { + sprintf(message, "RtAudio: ALSA underrun detected."); + error(RtError::WARNING); + err = snd_pcm_prepare(stream->handle[0]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error preparing handle after underrun: %s.", + snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + else { + sprintf(message, "RtAudio: ALSA error, current state is %s.", + snd_pcm_state_name(state)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + goto unlock; + } + else { + sprintf(message, "RtAudio: ALSA audio write error for device (%s): %s.", + devices[stream->device[0]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + + // Setup parameters. + if (stream->doConvertBuffer[1]) { + buffer = stream->deviceBuffer; + channels = stream->nDeviceChannels[1]; + format = stream->deviceFormat[1]; + } + else { + buffer = stream->userBuffer; + channels = stream->nUserChannels[1]; + format = stream->userFormat; + } + + // Read samples from device in interleaved/non-interleaved format. + if (stream->deInterleave[1]) { + void *bufs[channels]; + size_t offset = stream->bufferSize * formatBytes(format); + for (int i=0; ihandle[1], bufs, stream->bufferSize); + } + else + err = snd_pcm_readi(stream->handle[1], buffer, stream->bufferSize); + + if (err < stream->bufferSize) { + // Either an error or underrun occured. + if (err == -EPIPE) { + snd_pcm_state_t state = snd_pcm_state(stream->handle[1]); + if (state == SND_PCM_STATE_XRUN) { + sprintf(message, "RtAudio: ALSA overrun detected."); + error(RtError::WARNING); + err = snd_pcm_prepare(stream->handle[1]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error preparing handle after overrun: %s.", + snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + else { + sprintf(message, "RtAudio: ALSA error, current state is %s.", + snd_pcm_state_name(state)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + goto unlock; + } + else { + sprintf(message, "RtAudio: ALSA audio read error for device (%s): %s.", + devices[stream->device[1]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + // Do byte swapping if necessary. + if (stream->doByteSwap[1]) + byteSwapBuffer(buffer, stream->bufferSize * channels, format); + + // Do buffer conversion if necessary. + if (stream->doConvertBuffer[1]) + convertStreamBuffer(stream, INPUT); + } + + unlock: + MUTEX_UNLOCK(&stream->mutex); + + if (stream->callbackInfo.usingCallback && stopStream) + this->stopStream(streamId); +} + +extern "C" void *callbackHandler(void *ptr) +{ + CALLBACK_INFO *info = (CALLBACK_INFO *) ptr; + RtAudio *object = (RtAudio *) info->object; + int stream = info->streamId; + bool *usingCallback = &info->usingCallback; + + while ( *usingCallback ) { + pthread_testcancel(); + try { + object->tickStream(stream); + } + catch (RtError &exception) { + fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n", + exception.getMessage()); + break; + } + } + + return 0; +} + +//******************** End of __LINUX_ALSA__ *********************// + +#elif defined(__LINUX_OSS__) + +#include +#include +#include +#include +#include +#include +#include +#include + +#define DAC_NAME "/dev/dsp" +#define MAX_DEVICES 16 +#define MAX_CHANNELS 16 + +void RtAudio :: initialize(void) +{ + // Count cards and devices + nDevices = 0; + + // We check /dev/dsp before probing devices. /dev/dsp is supposed to + // be a link to the "default" audio device, of the form /dev/dsp0, + // /dev/dsp1, etc... However, I've seen many cases where /dev/dsp was a + // real device, so we need to check for that. Also, sometimes the + // link is to /dev/dspx and other times just dspx. I'm not sure how + // the latter works, but it does. + char device_name[16]; + struct stat dspstat; + int dsplink = -1; + int i = 0; + if (lstat(DAC_NAME, &dspstat) == 0) { + if (S_ISLNK(dspstat.st_mode)) { + i = readlink(DAC_NAME, device_name, sizeof(device_name)); + if (i > 0) { + device_name[i] = '\0'; + if (i > 8) { // check for "/dev/dspx" + if (!strncmp(DAC_NAME, device_name, 8)) + dsplink = atoi(&device_name[8]); + } + else if (i > 3) { // check for "dspx" + if (!strncmp("dsp", device_name, 3)) + dsplink = atoi(&device_name[3]); + } + } + else { + sprintf(message, "RtAudio: cannot read value of symbolic link %s.", DAC_NAME); + error(RtError::SYSTEM_ERROR); + } + } + } + else { + sprintf(message, "RtAudio: cannot stat %s.", DAC_NAME); + error(RtError::SYSTEM_ERROR); + } + + // The OSS API doesn't provide a routine for determining the number + // of devices. Thus, we'll just pursue a brute force method. The + // idea is to start with /dev/dsp(0) and continue with higher device + // numbers until we reach MAX_DSP_DEVICES. This should tell us how + // many devices we have ... it is not a fullproof scheme, but hopefully + // it will work most of the time. + + int fd = 0; + char names[MAX_DEVICES][16]; + for (i=-1; i= 0) close(fd); + strncpy(names[nDevices], device_name, 16); + nDevices++; + } + + if (nDevices == 0) return; + + // Allocate the RTAUDIO_DEVICE structures. + devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); + if (devices == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + + // Write device ascii identifiers to device control structure and then probe capabilities. + for (i=0; iname, O_WRONLY | O_NONBLOCK); + if (fd == -1) { + // Open device failed ... either busy or doesn't exist + if (errno == EBUSY || errno == EAGAIN) + sprintf(message, "RtAudio: OSS playback device (%s) is busy and cannot be probed.", + info->name); + else + sprintf(message, "RtAudio: OSS playback device (%s) open error.", info->name); + error(RtError::DEBUG_WARNING); + goto capture_probe; + } + + // We have an open device ... see how many channels it can handle + for (i=MAX_CHANNELS; i>0; i--) { + channels = i; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) { + // This would normally indicate some sort of hardware error, but under ALSA's + // OSS emulation, it sometimes indicates an invalid channel value. Further, + // the returned channel value is not changed. So, we'll ignore the possible + // hardware error. + continue; // try next channel number + } + // Check to see whether the device supports the requested number of channels + if (channels != i ) continue; // try next channel number + // If here, we found the largest working channel value + break; + } + info->maxOutputChannels = i; + + // Now find the minimum number of channels it can handle + for (i=1; i<=info->maxOutputChannels; i++) { + channels = i; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) + continue; // try next channel number + // If here, we found the smallest working channel value + break; + } + info->minOutputChannels = i; + close(fd); + + capture_probe: + // Now try for capture + fd = open(info->name, O_RDONLY | O_NONBLOCK); + if (fd == -1) { + // Open device for capture failed ... either busy or doesn't exist + if (errno == EBUSY || errno == EAGAIN) + sprintf(message, "RtAudio: OSS capture device (%s) is busy and cannot be probed.", + info->name); + else + sprintf(message, "RtAudio: OSS capture device (%s) open error.", info->name); + error(RtError::DEBUG_WARNING); + if (info->maxOutputChannels == 0) + // didn't open for playback either ... device invalid + return; + goto probe_parameters; + } + + // We have the device open for capture ... see how many channels it can handle + for (i=MAX_CHANNELS; i>0; i--) { + channels = i; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) { + continue; // as above + } + // If here, we found a working channel value + break; + } + info->maxInputChannels = i; + + // Now find the minimum number of channels it can handle + for (i=1; i<=info->maxInputChannels; i++) { + channels = i; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) + continue; // try next channel number + // If here, we found the smallest working channel value + break; + } + info->minInputChannels = i; + close(fd); + + if (info->maxOutputChannels == 0 && info->maxInputChannels == 0) { + sprintf(message, "RtAudio: OSS device (%s) reports zero channels for input and output.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + // If device opens for both playback and capture, we determine the channels. + if (info->maxOutputChannels == 0 || info->maxInputChannels == 0) + goto probe_parameters; + + fd = open(info->name, O_RDWR | O_NONBLOCK); + if (fd == -1) + goto probe_parameters; + + ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0); + ioctl(fd, SNDCTL_DSP_GETCAPS, &mask); + if (mask & DSP_CAP_DUPLEX) { + info->hasDuplexSupport = true; + // We have the device open for duplex ... see how many channels it can handle + for (i=MAX_CHANNELS; i>0; i--) { + channels = i; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) + continue; // as above + // If here, we found a working channel value + break; + } + info->maxDuplexChannels = i; + + // Now find the minimum number of channels it can handle + for (i=1; i<=info->maxDuplexChannels; i++) { + channels = i; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) + continue; // try next channel number + // If here, we found the smallest working channel value + break; + } + info->minDuplexChannels = i; + } + close(fd); + + probe_parameters: + // At this point, we need to figure out the supported data formats + // and sample rates. We'll proceed by openning the device in the + // direction with the maximum number of channels, or playback if + // they are equal. This might limit our sample rate options, but so + // be it. + + if (info->maxOutputChannels >= info->maxInputChannels) { + fd = open(info->name, O_WRONLY | O_NONBLOCK); + channels = info->maxOutputChannels; + } + else { + fd = open(info->name, O_RDONLY | O_NONBLOCK); + channels = info->maxInputChannels; + } + + if (fd == -1) { + // We've got some sort of conflict ... abort + sprintf(message, "RtAudio: OSS device (%s) won't reopen during probe.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + // We have an open device ... set to maximum channels. + i = channels; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) { + // We've got some sort of conflict ... abort + close(fd); + sprintf(message, "RtAudio: OSS device (%s) won't revert to previous channel setting.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) { + close(fd); + sprintf(message, "RtAudio: OSS device (%s) can't get supported audio formats.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + // Probe the supported data formats ... we don't care about endian-ness just yet. + int format; + info->nativeFormats = 0; +#if defined (AFMT_S32_BE) + // This format does not seem to be in the 2.4 kernel version of OSS soundcard.h + if (mask & AFMT_S32_BE) { + format = AFMT_S32_BE; + info->nativeFormats |= RTAUDIO_SINT32; + } +#endif +#if defined (AFMT_S32_LE) + /* This format is not in the 2.4.4 kernel version of OSS soundcard.h */ + if (mask & AFMT_S32_LE) { + format = AFMT_S32_LE; + info->nativeFormats |= RTAUDIO_SINT32; + } +#endif + if (mask & AFMT_S8) { + format = AFMT_S8; + info->nativeFormats |= RTAUDIO_SINT8; + } + if (mask & AFMT_S16_BE) { + format = AFMT_S16_BE; + info->nativeFormats |= RTAUDIO_SINT16; + } + if (mask & AFMT_S16_LE) { + format = AFMT_S16_LE; + info->nativeFormats |= RTAUDIO_SINT16; + } + + // Check that we have at least one supported format + if (info->nativeFormats == 0) { + close(fd); + sprintf(message, "RtAudio: OSS device (%s) data format not supported by RtAudio.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + // Set the format + i = format; + if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1 || format != i) { + close(fd); + sprintf(message, "RtAudio: OSS device (%s) error setting data format.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + // Probe the supported sample rates ... first get lower limit + int speed = 1; + if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { + // If we get here, we're probably using an ALSA driver with OSS-emulation, + // which doesn't conform to the OSS specification. In this case, + // we'll probe our predefined list of sample rates for working values. + info->nSampleRates = 0; + for (i=0; isampleRates[info->nSampleRates] = SAMPLE_RATES[i]; + info->nSampleRates++; + } + } + if (info->nSampleRates == 0) { + close(fd); + return; + } + goto finished; + } + info->sampleRates[0] = speed; + + // Now get upper limit + speed = 1000000; + if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { + close(fd); + sprintf(message, "RtAudio: OSS device (%s) error setting sample rate.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + info->sampleRates[1] = speed; + info->nSampleRates = -1; + + finished: // That's all ... close the device and return + close(fd); + info->probed = true; + return; +} + +bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, + STREAM_MODE mode, int channels, + int sampleRate, RTAUDIO_FORMAT format, + int *bufferSize, int numberOfBuffers) +{ + int buffers, buffer_bytes, device_channels, device_format; + int srate, temp, fd; + + const char *name = devices[device].name; + + if (mode == OUTPUT) + fd = open(name, O_WRONLY | O_NONBLOCK); + else { // mode == INPUT + if (stream->mode == OUTPUT && stream->device[0] == device) { + // We just set the same device for playback ... close and reopen for duplex (OSS only). + close(stream->handle[0]); + stream->handle[0] = 0; + // First check that the number previously set channels is the same. + if (stream->nUserChannels[0] != channels) { + sprintf(message, "RtAudio: input/output channels must be equal for OSS duplex device (%s).", name); + goto error; + } + fd = open(name, O_RDWR | O_NONBLOCK); + } + else + fd = open(name, O_RDONLY | O_NONBLOCK); + } + + if (fd == -1) { + if (errno == EBUSY || errno == EAGAIN) + sprintf(message, "RtAudio: OSS device (%s) is busy and cannot be opened.", + name); + else + sprintf(message, "RtAudio: OSS device (%s) cannot be opened.", name); + goto error; + } + + // Now reopen in blocking mode. + close(fd); + if (mode == OUTPUT) + fd = open(name, O_WRONLY | O_SYNC); + else { // mode == INPUT + if (stream->mode == OUTPUT && stream->device[0] == device) + fd = open(name, O_RDWR | O_SYNC); + else + fd = open(name, O_RDONLY | O_SYNC); + } + + if (fd == -1) { + sprintf(message, "RtAudio: OSS device (%s) cannot be opened.", name); + goto error; + } + + // Get the sample format mask + int mask; + if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) { + close(fd); + sprintf(message, "RtAudio: OSS device (%s) can't get supported audio formats.", + name); + goto error; + } + + // Determine how to set the device format. + stream->userFormat = format; + device_format = -1; + stream->doByteSwap[mode] = false; + if (format == RTAUDIO_SINT8) { + if (mask & AFMT_S8) { + device_format = AFMT_S8; + stream->deviceFormat[mode] = RTAUDIO_SINT8; + } + } + else if (format == RTAUDIO_SINT16) { + if (mask & AFMT_S16_NE) { + device_format = AFMT_S16_NE; + stream->deviceFormat[mode] = RTAUDIO_SINT16; + } +#if BYTE_ORDER == LITTLE_ENDIAN + else if (mask & AFMT_S16_BE) { + device_format = AFMT_S16_BE; + stream->deviceFormat[mode] = RTAUDIO_SINT16; + stream->doByteSwap[mode] = true; + } +#else + else if (mask & AFMT_S16_LE) { + device_format = AFMT_S16_LE; + stream->deviceFormat[mode] = RTAUDIO_SINT16; + stream->doByteSwap[mode] = true; + } +#endif + } +#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE) + else if (format == RTAUDIO_SINT32) { + if (mask & AFMT_S32_NE) { + device_format = AFMT_S32_NE; + stream->deviceFormat[mode] = RTAUDIO_SINT32; + } +#if BYTE_ORDER == LITTLE_ENDIAN + else if (mask & AFMT_S32_BE) { + device_format = AFMT_S32_BE; + stream->deviceFormat[mode] = RTAUDIO_SINT32; + stream->doByteSwap[mode] = true; + } +#else + else if (mask & AFMT_S32_LE) { + device_format = AFMT_S32_LE; + stream->deviceFormat[mode] = RTAUDIO_SINT32; + stream->doByteSwap[mode] = true; + } +#endif + } +#endif + + if (device_format == -1) { + // The user requested format is not natively supported by the device. + if (mask & AFMT_S16_NE) { + device_format = AFMT_S16_NE; + stream->deviceFormat[mode] = RTAUDIO_SINT16; + } +#if BYTE_ORDER == LITTLE_ENDIAN + else if (mask & AFMT_S16_BE) { + device_format = AFMT_S16_BE; + stream->deviceFormat[mode] = RTAUDIO_SINT16; + stream->doByteSwap[mode] = true; + } +#else + else if (mask & AFMT_S16_LE) { + device_format = AFMT_S16_LE; + stream->deviceFormat[mode] = RTAUDIO_SINT16; + stream->doByteSwap[mode] = true; + } +#endif +#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE) + else if (mask & AFMT_S32_NE) { + device_format = AFMT_S32_NE; + stream->deviceFormat[mode] = RTAUDIO_SINT32; + } +#if BYTE_ORDER == LITTLE_ENDIAN + else if (mask & AFMT_S32_BE) { + device_format = AFMT_S32_BE; + stream->deviceFormat[mode] = RTAUDIO_SINT32; + stream->doByteSwap[mode] = true; + } +#else + else if (mask & AFMT_S32_LE) { + device_format = AFMT_S32_LE; + stream->deviceFormat[mode] = RTAUDIO_SINT32; + stream->doByteSwap[mode] = true; + } +#endif +#endif + else if (mask & AFMT_S8) { + device_format = AFMT_S8; + stream->deviceFormat[mode] = RTAUDIO_SINT8; + } + } + + if (stream->deviceFormat[mode] == 0) { + // This really shouldn't happen ... + close(fd); + sprintf(message, "RtAudio: OSS device (%s) data format not supported by RtAudio.", + name); + goto error; + } + + // Determine the number of channels for this device. Note that the + // channel value requested by the user might be < min_X_Channels. + stream->nUserChannels[mode] = channels; + device_channels = channels; + if (mode == OUTPUT) { + if (channels < devices[device].minOutputChannels) + device_channels = devices[device].minOutputChannels; + } + else { // mode == INPUT + if (stream->mode == OUTPUT && stream->device[0] == device) { + // We're doing duplex setup here. + if (channels < devices[device].minDuplexChannels) + device_channels = devices[device].minDuplexChannels; + } + else { + if (channels < devices[device].minInputChannels) + device_channels = devices[device].minInputChannels; + } + } + stream->nDeviceChannels[mode] = device_channels; + + // Attempt to set the buffer size. According to OSS, the minimum + // number of buffers is two. The supposed minimum buffer size is 16 + // bytes, so that will be our lower bound. The argument to this + // call is in the form 0xMMMMSSSS (hex), where the buffer size (in + // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. + // We'll check the actual value used near the end of the setup + // procedure. + buffer_bytes = *bufferSize * formatBytes(stream->deviceFormat[mode]) * device_channels; + if (buffer_bytes < 16) buffer_bytes = 16; + buffers = numberOfBuffers; + if (buffers < 2) buffers = 2; + temp = ((int) buffers << 16) + (int)(log10((double)buffer_bytes)/log10(2.0)); + if (ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &temp)) { + close(fd); + sprintf(message, "RtAudio: OSS error setting fragment size for device (%s).", + name); + goto error; + } + stream->nBuffers = buffers; + + // Set the data format. + temp = device_format; + if (ioctl(fd, SNDCTL_DSP_SETFMT, &device_format) == -1 || device_format != temp) { + close(fd); + sprintf(message, "RtAudio: OSS error setting data format for device (%s).", + name); + goto error; + } + + // Set the number of channels. + temp = device_channels; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &device_channels) == -1 || device_channels != temp) { + close(fd); + sprintf(message, "RtAudio: OSS error setting %d channels on device (%s).", + temp, name); + goto error; + } + + // Set the sample rate. + srate = sampleRate; + temp = srate; + if (ioctl(fd, SNDCTL_DSP_SPEED, &srate) == -1) { + close(fd); + sprintf(message, "RtAudio: OSS error setting sample rate = %d on device (%s).", + temp, name); + goto error; + } + + // Verify the sample rate setup worked. + if (abs(srate - temp) > 100) { + close(fd); + sprintf(message, "RtAudio: OSS error ... audio device (%s) doesn't support sample rate of %d.", + name, temp); + goto error; + } + stream->sampleRate = sampleRate; + + if (ioctl(fd, SNDCTL_DSP_GETBLKSIZE, &buffer_bytes) == -1) { + close(fd); + sprintf(message, "RtAudio: OSS error getting buffer size for device (%s).", + name); + goto error; + } + + // Save buffer size (in sample frames). + *bufferSize = buffer_bytes / (formatBytes(stream->deviceFormat[mode]) * device_channels); + stream->bufferSize = *bufferSize; + + if (mode == INPUT && stream->mode == OUTPUT && + stream->device[0] == device) { + // We're doing duplex setup here. + stream->deviceFormat[0] = stream->deviceFormat[1]; + stream->nDeviceChannels[0] = device_channels; + } + + // Set flags for buffer conversion + stream->doConvertBuffer[mode] = false; + if (stream->userFormat != stream->deviceFormat[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode]) + stream->doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { + + long buffer_bytes; + if (stream->nUserChannels[0] >= stream->nUserChannels[1]) + buffer_bytes = stream->nUserChannels[0]; + else + buffer_bytes = stream->nUserChannels[1]; + + buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); + if (stream->userBuffer) free(stream->userBuffer); + stream->userBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->userBuffer == NULL) { + close(fd); + sprintf(message, "RtAudio: OSS error allocating user buffer memory (%s).", + name); + goto error; + } + } + + if ( stream->doConvertBuffer[mode] ) { + + long buffer_bytes; + bool makeBuffer = true; + if ( mode == OUTPUT ) + buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + else { // mode == INPUT + buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); + if ( stream->mode == OUTPUT && stream->deviceBuffer ) { + long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + if ( buffer_bytes < bytes_out ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + buffer_bytes *= *bufferSize; + if (stream->deviceBuffer) free(stream->deviceBuffer); + stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->deviceBuffer == NULL) { + close(fd); + free(stream->userBuffer); + sprintf(message, "RtAudio: OSS error allocating device buffer memory (%s).", + name); + goto error; + } + } + } + + stream->device[mode] = device; + stream->handle[mode] = fd; + stream->state = STREAM_STOPPED; + if ( stream->mode == OUTPUT && mode == INPUT ) { + stream->mode = DUPLEX; + if (stream->device[0] == device) + stream->handle[0] = fd; + } + else + stream->mode = mode; + + return SUCCESS; + + error: + if (stream->handle[0]) { + close(stream->handle[0]); + stream->handle[0] = 0; + } + error(RtError::WARNING); + return FAILURE; +} + +void RtAudio :: closeStream(int streamId) +{ + // We don't want an exception to be thrown here because this + // function is called by our class destructor. So, do our own + // streamId check. + if ( streams.find( streamId ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtError::WARNING); + return; + } + + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + + if (stream->callbackInfo.usingCallback) { + pthread_cancel(stream->callbackInfo.thread); + pthread_join(stream->callbackInfo.thread, NULL); + } + + if (stream->state == STREAM_RUNNING) { + if (stream->mode == OUTPUT || stream->mode == DUPLEX) + ioctl(stream->handle[0], SNDCTL_DSP_RESET, 0); + if (stream->mode == INPUT || stream->mode == DUPLEX) + ioctl(stream->handle[1], SNDCTL_DSP_RESET, 0); + } + + pthread_mutex_destroy(&stream->mutex); + + if (stream->handle[0]) + close(stream->handle[0]); + + if (stream->handle[1]) + close(stream->handle[1]); + + if (stream->userBuffer) + free(stream->userBuffer); + + if (stream->deviceBuffer) + free(stream->deviceBuffer); + + free(stream); + streams.erase(streamId); +} + +void RtAudio :: startStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + stream->state = STREAM_RUNNING; + + // No need to do anything else here ... OSS automatically starts + // when fed samples. + + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: stopStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + int err; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + err = ioctl(stream->handle[0], SNDCTL_DSP_SYNC, 0); + if (err < -1) { + sprintf(message, "RtAudio: OSS error stopping device (%s).", + devices[stream->device[0]].name); + error(RtError::DRIVER_ERROR); + } + } + else { + err = ioctl(stream->handle[1], SNDCTL_DSP_SYNC, 0); + if (err < -1) { + sprintf(message, "RtAudio: OSS error stopping device (%s).", + devices[stream->device[1]].name); + error(RtError::DRIVER_ERROR); + } + } + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: abortStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + int err; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + err = ioctl(stream->handle[0], SNDCTL_DSP_RESET, 0); + if (err < -1) { + sprintf(message, "RtAudio: OSS error aborting device (%s).", + devices[stream->device[0]].name); + error(RtError::DRIVER_ERROR); + } + } + else { + err = ioctl(stream->handle[1], SNDCTL_DSP_RESET, 0); + if (err < -1) { + sprintf(message, "RtAudio: OSS error aborting device (%s).", + devices[stream->device[1]].name); + error(RtError::DRIVER_ERROR); + } + } + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +int RtAudio :: streamWillBlock(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + int bytes = 0, channels = 0, frames = 0; + if (stream->state == STREAM_STOPPED) + goto unlock; + + audio_buf_info info; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + ioctl(stream->handle[0], SNDCTL_DSP_GETOSPACE, &info); + bytes = info.bytes; + channels = stream->nDeviceChannels[0]; + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + ioctl(stream->handle[1], SNDCTL_DSP_GETISPACE, &info); + if (stream->mode == DUPLEX ) { + bytes = (bytes < info.bytes) ? bytes : info.bytes; + channels = stream->nDeviceChannels[0]; + } + else { + bytes = info.bytes; + channels = stream->nDeviceChannels[1]; + } + } + + frames = (int) (bytes / (channels * formatBytes(stream->deviceFormat[0]))); + frames -= stream->bufferSize; + if (frames < 0) frames = 0; + + unlock: + MUTEX_UNLOCK(&stream->mutex); + return frames; +} + +void RtAudio :: tickStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + int stopStream = 0; + if (stream->state == STREAM_STOPPED) { + if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds + return; + } + else if (stream->callbackInfo.usingCallback) { + RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback; + stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData); + } + + 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; + RTAUDIO_FORMAT format; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + // Setup parameters and do buffer conversion if necessary. + if (stream->doConvertBuffer[0]) { + convertStreamBuffer(stream, OUTPUT); + buffer = stream->deviceBuffer; + samples = stream->bufferSize * stream->nDeviceChannels[0]; + format = stream->deviceFormat[0]; + } + else { + buffer = stream->userBuffer; + samples = stream->bufferSize * stream->nUserChannels[0]; + format = stream->userFormat; + } + + // Do byte swapping if necessary. + if (stream->doByteSwap[0]) + byteSwapBuffer(buffer, samples, format); + + // Write samples to device. + result = write(stream->handle[0], buffer, samples * formatBytes(format)); + + if (result == -1) { + // This could be an underrun, but the basic OSS API doesn't provide a means for determining that. + sprintf(message, "RtAudio: OSS audio write error for device (%s).", + devices[stream->device[0]].name); + error(RtError::DRIVER_ERROR); + } + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + + // Setup parameters. + if (stream->doConvertBuffer[1]) { + buffer = stream->deviceBuffer; + samples = stream->bufferSize * stream->nDeviceChannels[1]; + format = stream->deviceFormat[1]; + } + else { + buffer = stream->userBuffer; + samples = stream->bufferSize * stream->nUserChannels[1]; + format = stream->userFormat; + } + + // Read samples from device. + result = read(stream->handle[1], buffer, samples * formatBytes(format)); + + if (result == -1) { + // This could be an overrun, but the basic OSS API doesn't provide a means for determining that. + sprintf(message, "RtAudio: OSS audio read error for device (%s).", + devices[stream->device[1]].name); + error(RtError::DRIVER_ERROR); + } + + // Do byte swapping if necessary. + if (stream->doByteSwap[1]) + byteSwapBuffer(buffer, samples, format); + + // Do buffer conversion if necessary. + if (stream->doConvertBuffer[1]) + convertStreamBuffer(stream, INPUT); + } + + unlock: + MUTEX_UNLOCK(&stream->mutex); + + if (stream->callbackInfo.usingCallback && stopStream) + this->stopStream(streamId); +} + +extern "C" void *callbackHandler(void *ptr) +{ + CALLBACK_INFO *info = (CALLBACK_INFO *) ptr; + RtAudio *object = (RtAudio *) info->object; + int stream = info->streamId; + bool *usingCallback = &info->usingCallback; + + while ( *usingCallback ) { + pthread_testcancel(); + try { + object->tickStream(stream); + } + catch (RtError &exception) { + fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n", + exception.getMessage()); + break; + } + } + + return 0; +} + + +//******************** End of __LINUX_OSS__ *********************// + +#elif defined(__WINDOWS_ASIO__) // ASIO API on Windows + +// The ASIO API is designed around a callback scheme, so this +// implementation is similar to that used for OS X CoreAudio. The +// primary constraint with ASIO is that it only allows access to a +// single driver at a time. Thus, it is not possible to have more +// than one simultaneous RtAudio stream. +// +// 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. + +#include "asio/asiosys.h" +#include "asio/asio.h" +#include "asio/asiodrivers.h" +#include + +AsioDrivers drivers; +ASIOCallbacks asioCallbacks; +CALLBACK_INFO *asioCallbackInfo; +ASIODriverInfo driverInfo; + +void RtAudio :: initialize(void) +{ + nDevices = drivers.asioGetNumDev(); + if (nDevices <= 0) return; + + // Allocate the RTAUDIO_DEVICE structures. + devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); + if (devices == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + + // Write device driver names to device structures and then probe the + // device capabilities. + for (int i=0; i 0 ) { + sprintf(message, "RtAudio: unable to probe ASIO driver while a stream is open."); + error(RtError::DEBUG_WARNING); + return; + } + + if ( !drivers.loadDriver( info->name ) ) { + sprintf(message, "RtAudio: ASIO error loading driver (%s).", info->name); + error(RtError::DEBUG_WARNING); + return; + } + + ASIOError result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + char details[32]; + if ( result == ASE_HWMalfunction ) + sprintf(details, "hardware malfunction"); + else if ( result == ASE_NoMemory ) + sprintf(details, "no memory"); + else if ( result == ASE_NotPresent ) + sprintf(details, "driver/hardware not present"); + else + sprintf(details, "unspecified"); + sprintf(message, "RtAudio: ASIO error (%s) initializing driver (%s).", details, info->name); + error(RtError::DEBUG_WARNING); + return; + } + + // Determine the device channel information. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO error getting input/output channel count (%s).", info->name); + error(RtError::DEBUG_WARNING); + return; + } + + info->maxOutputChannels = outputChannels; + if ( outputChannels > 0 ) info->minOutputChannels = 1; + + info->maxInputChannels = inputChannels; + if ( inputChannels > 0 ) info->minInputChannels = 1; + + // If device opens for both playback and capture, we determine the channels. + if (info->maxOutputChannels > 0 && info->maxInputChannels > 0) { + info->hasDuplexSupport = true; + info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ? + info->maxInputChannels : info->maxOutputChannels; + info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ? + info->minInputChannels : info->minOutputChannels; + } + + // Determine the supported sample rates. + info->nSampleRates = 0; + for (int i=0; isampleRates[info->nSampleRates++] = SAMPLE_RATES[i]; + } + + if (info->nSampleRates == 0) { + drivers.removeCurrentDriver(); + sprintf( message, "RtAudio: No supported sample rates found for ASIO driver (%s).", info->name ); + error(RtError::DEBUG_WARNING); + return; + } + + // Determine supported data types ... just check first channel and assume rest are the same. + ASIOChannelInfo channelInfo; + channelInfo.channel = 0; + channelInfo.isInput = true; + if ( info->maxInputChannels <= 0 ) channelInfo.isInput = false; + result = ASIOGetChannelInfo( &channelInfo ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO error getting driver (%s) channel information.", info->name); + error(RtError::DEBUG_WARNING); + return; + } + + 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; + + // Check that we have at least one supported format. + if (info->nativeFormats == 0) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) data format not supported by RtAudio.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + info->probed = true; + drivers.removeCurrentDriver(); +} + +void bufferSwitch(long index, ASIOBool processNow) +{ + RtAudio *object = (RtAudio *) asioCallbackInfo->object; + try { + object->callbackEvent( asioCallbackInfo->streamId, index, (void *)NULL, (void *)NULL ); + } + catch (RtError &exception) { + fprintf(stderr, "\nCallback handler error (%s)!\n\n", exception.getMessage()); + return; + } + + return; +} + +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. + + RtAudio *object = (RtAudio *) asioCallbackInfo->object; + try { + object->stopStream( asioCallbackInfo->streamId ); + } + catch (RtError &exception) { + fprintf(stderr, "\nRtAudio: sampleRateChanged() error (%s)!\n\n", exception.getMessage()); + return; + } + + fprintf(stderr, "\nRtAudio: ASIO driver reports sample rate changed to %d ... stream stopped!!!", (int) sRate); +} + +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. + fprintf(stderr, "\nRtAudio: ASIO driver reset requested!!!"); + 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. + fprintf(stderr, "\nRtAudio: ASIO driver resync requested!!!"); + 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. + fprintf(stderr, "\nRtAudio: ASIO driver latency may have changed!!!"); + 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 wether 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; +} + +bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, + STREAM_MODE mode, int channels, + int sampleRate, RTAUDIO_FORMAT format, + int *bufferSize, int numberOfBuffers) +{ + // Don't attempt to load another driver if a stream is already open. + if ( streams.size() > 0 ) { + sprintf(message, "RtAudio: unable to load ASIO driver while a stream is open."); + error(RtError::WARNING); + return FAILURE; + } + + // For ASIO, a duplex stream MUST use the same driver. + if ( mode == INPUT && stream->mode == OUTPUT && stream->device[0] != device ) { + sprintf(message, "RtAudio: ASIO duplex stream must use the same device for input and output."); + error(RtError::WARNING); + return FAILURE; + } + + // Only load the driver once for duplex stream. + ASIOError result; + if ( mode != INPUT || stream->mode != OUTPUT ) { + if ( !drivers.loadDriver( devices[device].name ) ) { + sprintf(message, "RtAudio: ASIO error loading driver (%s).", devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + char details[32]; + if ( result == ASE_HWMalfunction ) + sprintf(details, "hardware malfunction"); + else if ( result == ASE_NoMemory ) + sprintf(details, "no memory"); + else if ( result == ASE_NotPresent ) + sprintf(details, "driver/hardware not present"); + else + sprintf(details, "unspecified"); + sprintf(message, "RtAudio: ASIO error (%s) initializing driver (%s).", details, devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + } + + // Check the device channel count. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO error getting input/output channel count (%s).", + devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + if ( ( mode == OUTPUT && channels > outputChannels) || + ( mode == INPUT && channels > inputChannels) ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) does not support requested channel count (%d).", + devices[device].name, channels); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + stream->nDeviceChannels[mode] = channels; + stream->nUserChannels[mode] = channels; + + // Verify the sample rate is supported. + result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) does not support requested sample rate (%d).", + devices[device].name, sampleRate); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + // Set the sample rate. + result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) error setting sample rate (%d).", + devices[device].name, sampleRate); + error(RtError::DEBUG_WARNING); + 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(); + sprintf(message, "RtAudio: ASIO driver (%s) error getting data format.", + devices[device].name); + error(RtError::DEBUG_WARNING); + 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(); + sprintf(message, "RtAudio: ASIO driver (%s) data format not supported by RtAudio.", + devices[device].name); + error(RtError::DEBUG_WARNING); + 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(); + sprintf(message, "RtAudio: ASIO driver (%s) error getting buffer size.", + devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + if ( *bufferSize < minSize ) *bufferSize = minSize; + else if ( *bufferSize > maxSize ) *bufferSize = maxSize; + else if ( granularity == -1 ) { + // Make sure bufferSize is a power of two. + double power = log10( *bufferSize ) / log10( 2.0 ); + *bufferSize = pow( 2.0, floor(power+0.5) ); + if ( *bufferSize < minSize ) *bufferSize = minSize; + else if ( *bufferSize > maxSize ) *bufferSize = maxSize; + else *bufferSize = preferSize; + } + + if ( mode == INPUT && stream->mode == OUTPUT && stream->bufferSize != *bufferSize ) + cout << "possible input/output buffersize discrepancy" << endl; + + stream->bufferSize = *bufferSize; + stream->nBuffers = 2; + + // ASIO always uses deinterleaved channels. + stream->deInterleave[mode] = true; + + // Create the ASIO internal buffers. Since RtAudio sets up input + // and output separately, we'll have to dispose of previously + // created output buffers for a duplex stream. + if ( mode == INPUT && stream->mode == OUTPUT ) { + free(stream->callbackInfo.buffers); + result = ASIODisposeBuffers(); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) error disposing previously allocated buffers.", + devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + } + + // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. + int i, nChannels = stream->nDeviceChannels[0] + stream->nDeviceChannels[1]; + stream->callbackInfo.buffers = 0; + ASIOBufferInfo *bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); + stream->callbackInfo.buffers = (void *) bufferInfos; + ASIOBufferInfo *infos = bufferInfos; + for ( i=0; inDeviceChannels[1]; i++, infos++ ) { + infos->isInput = ASIOTrue; + infos->channelNum = i; + infos->buffers[0] = infos->buffers[1] = 0; + } + + for ( i=0; inDeviceChannels[0]; i++, infos++ ) { + infos->isInput = ASIOFalse; + infos->channelNum = i; + 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( bufferInfos, nChannels, stream->bufferSize, &asioCallbacks); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) error creating buffers.", + devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + // Set flags for buffer conversion. + stream->doConvertBuffer[mode] = false; + if (stream->userFormat != stream->deviceFormat[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] > 1 && stream->deInterleave[mode]) + stream->doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { + + long buffer_bytes; + if (stream->nUserChannels[0] >= stream->nUserChannels[1]) + buffer_bytes = stream->nUserChannels[0]; + else + buffer_bytes = stream->nUserChannels[1]; + + buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); + if (stream->userBuffer) free(stream->userBuffer); + stream->userBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->userBuffer == NULL) + goto memory_error; + } + + if ( stream->doConvertBuffer[mode] ) { + + long buffer_bytes; + bool makeBuffer = true; + if ( mode == OUTPUT ) + buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + else { // mode == INPUT + buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); + if ( stream->mode == OUTPUT && stream->deviceBuffer ) { + long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + if ( buffer_bytes < bytes_out ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + buffer_bytes *= *bufferSize; + if (stream->deviceBuffer) free(stream->deviceBuffer); + stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->deviceBuffer == NULL) + goto memory_error; + } + } + + stream->device[mode] = device; + stream->state = STREAM_STOPPED; + if ( stream->mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream->mode = DUPLEX; + else + stream->mode = mode; + stream->sampleRate = sampleRate; + asioCallbackInfo = &stream->callbackInfo; + stream->callbackInfo.object = (void *) this; + stream->callbackInfo.waitTime = (unsigned long) (200.0 * stream->bufferSize / stream->sampleRate); + + return SUCCESS; + + memory_error: + ASIODisposeBuffers(); + drivers.removeCurrentDriver(); + + if (stream->callbackInfo.buffers) + free(stream->callbackInfo.buffers); + stream->callbackInfo.buffers = 0; + + if (stream->userBuffer) { + free(stream->userBuffer); + stream->userBuffer = 0; + } + sprintf(message, "RtAudio: error allocating buffer memory (%s).", + devices[device].name); + error(RtError::WARNING); + return FAILURE; +} + +void RtAudio :: cancelStreamCallback(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + if (stream->callbackInfo.usingCallback) { + + if (stream->state == STREAM_RUNNING) + stopStream( streamId ); + + MUTEX_LOCK(&stream->mutex); + + stream->callbackInfo.usingCallback = false; + stream->callbackInfo.userData = NULL; + stream->state = STREAM_STOPPED; + stream->callbackInfo.callback = NULL; + + MUTEX_UNLOCK(&stream->mutex); + } +} + +void RtAudio :: closeStream(int streamId) +{ + // We don't want an exception to be thrown here because this + // function is called by our class destructor. So, do our own + // streamId check. + if ( streams.find( streamId ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtError::WARNING); + return; + } + + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + + if (stream->state == STREAM_RUNNING) + ASIOStop(); + + ASIODisposeBuffers(); + //ASIOExit(); + drivers.removeCurrentDriver(); + + DeleteCriticalSection(&stream->mutex); + + if (stream->callbackInfo.buffers) + free(stream->callbackInfo.buffers); + + if (stream->userBuffer) + free(stream->userBuffer); + + if (stream->deviceBuffer) + free(stream->deviceBuffer); + + free(stream); + streams.erase(streamId); +} + +void RtAudio :: startStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_RUNNING) { + MUTEX_UNLOCK(&stream->mutex); + return; + } + + stream->callbackInfo.blockTick = true; + stream->callbackInfo.stopStream = false; + stream->callbackInfo.streamId = streamId; + ASIOError result = ASIOStart(); + if ( result != ASE_OK ) { + sprintf(message, "RtAudio: ASIO error starting device (%s).", + devices[stream->device[0]].name); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + stream->state = STREAM_RUNNING; + + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: stopStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) { + MUTEX_UNLOCK(&stream->mutex); + return; + } + + ASIOError result = ASIOStop(); + if ( result != ASE_OK ) { + sprintf(message, "RtAudio: ASIO error stopping device (%s).", + devices[stream->device[0]].name); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + stream->state = STREAM_STOPPED; + + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: abortStream(int streamId) +{ + stopStream( streamId ); +} + +// I don't know how this function can be implemented. +int RtAudio :: streamWillBlock(int streamId) +{ + sprintf(message, "RtAudio: streamWillBlock() cannot be implemented for ASIO."); + error(RtError::WARNING); + return 0; +} + +void RtAudio :: tickStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + if (stream->state == STREAM_STOPPED) + return; + + if (stream->callbackInfo.usingCallback) { + sprintf(message, "RtAudio: tickStream() should not be used when a callback function is set!"); + error(RtError::WARNING); + return; + } + + // Block waiting here until the user data is processed in callbackEvent(). + while ( stream->callbackInfo.blockTick ) + Sleep(stream->callbackInfo.waitTime); + + MUTEX_LOCK(&stream->mutex); + + stream->callbackInfo.blockTick = true; + + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: callbackEvent(int streamId, int bufferIndex, void *inData, void *outData) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + CALLBACK_INFO *info = asioCallbackInfo; + if ( !info->usingCallback ) { + // Block waiting here until we get new user data in tickStream(). + while ( !info->blockTick ) + Sleep(info->waitTime); + } + else if ( info->stopStream ) { + // Check if the stream should be stopped (via the previous user + // callback return value). We stop the stream here, rather than + // after the function call, so that output data can first be + // processed. + this->stopStream(asioCallbackInfo->streamId); + return; + } + + MUTEX_LOCK(&stream->mutex); + int nChannels = stream->nDeviceChannels[0] + stream->nDeviceChannels[1]; + int bufferBytes; + ASIOBufferInfo *bufferInfos = (ASIOBufferInfo *) info->buffers; + if ( stream->mode == INPUT || stream->mode == DUPLEX ) { + + bufferBytes = stream->bufferSize * formatBytes(stream->deviceFormat[1]); + if (stream->doConvertBuffer[1]) { + + // Always interleave ASIO input data. + for ( int i=0; inDeviceChannels[1]; i++, bufferInfos++ ) + memcpy(&stream->deviceBuffer[i*bufferBytes], bufferInfos->buffers[bufferIndex], bufferBytes ); + + if ( stream->doByteSwap[1] ) + byteSwapBuffer(stream->deviceBuffer, + stream->bufferSize * stream->nDeviceChannels[1], + stream->deviceFormat[1]); + convertStreamBuffer(stream, INPUT); + + } + else { // single channel only + memcpy(stream->userBuffer, bufferInfos->buffers[bufferIndex], bufferBytes ); + + if (stream->doByteSwap[1]) + byteSwapBuffer(stream->userBuffer, + stream->bufferSize * stream->nUserChannels[1], + stream->userFormat); + } + } + + if ( info->usingCallback ) { + RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) info->callback; + if ( callback(stream->userBuffer, stream->bufferSize, info->userData) ) + info->stopStream = true; + } + + if ( stream->mode == OUTPUT || stream->mode == DUPLEX ) { + + bufferBytes = stream->bufferSize * formatBytes(stream->deviceFormat[0]); + if (stream->doConvertBuffer[0]) { + + convertStreamBuffer(stream, OUTPUT); + if ( stream->doByteSwap[0] ) + byteSwapBuffer(stream->deviceBuffer, + stream->bufferSize * stream->nDeviceChannels[0], + stream->deviceFormat[0]); + + // Always de-interleave ASIO output data. + for ( int i=0; inDeviceChannels[0]; i++, bufferInfos++ ) { + memcpy(bufferInfos->buffers[bufferIndex], + &stream->deviceBuffer[i*bufferBytes], bufferBytes ); + } + } + else { // single channel only + + if (stream->doByteSwap[0]) + byteSwapBuffer(stream->userBuffer, + stream->bufferSize * stream->nUserChannels[0], + stream->userFormat); + + memcpy(bufferInfos->buffers[bufferIndex], stream->userBuffer, bufferBytes ); + } + } + + if ( !info->usingCallback ) + info->blockTick = false; + + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + stream->callbackInfo.callback = (void *) callback; + stream->callbackInfo.userData = userData; + stream->callbackInfo.usingCallback = true; +} + +//******************** End of __WINDOWS_ASIO__ *********************// + +#elif defined(__WINDOWS_DS__) // Windows DirectSound API + +#include + +// Declarations for utility functions, callbacks, and structures +// specific to the DirectSound implementation. +static bool CALLBACK deviceCountCallback(LPGUID lpguid, + LPCSTR lpcstrDescription, + LPCSTR lpcstrModule, + LPVOID lpContext); + +static bool CALLBACK deviceInfoCallback(LPGUID lpguid, + LPCSTR lpcstrDescription, + LPCSTR lpcstrModule, + LPVOID lpContext); + +static bool CALLBACK defaultDeviceCallback(LPGUID lpguid, + LPCSTR lpcstrDescription, + LPCSTR lpcstrModule, + LPVOID lpContext); + +static bool CALLBACK deviceIdCallback(LPGUID lpguid, + LPCSTR lpcstrDescription, + LPCSTR lpcstrModule, + LPVOID lpContext); + +static char* getErrorString(int code); + +struct enum_info { + char name[64]; + LPGUID id; + bool isInput; + bool isValid; +}; + +int RtAudio :: getDefaultInputDevice(void) +{ + enum_info info; + info.name[0] = '\0'; + + // Enumerate through devices to find the default output. + HRESULT result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)defaultDeviceCallback, &info); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Error performing default input device enumeration: %s.", + getErrorString(result)); + error(RtError::WARNING); + return 0; + } + + for ( int i=0; i info(count); + for (i=0; iname, 64 ); + dsinfo.isValid = false; + + // Enumerate through input devices to find the id (if it exists). + HRESULT result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Error performing input device id enumeration: %s.", + getErrorString(result)); + error(RtError::WARNING); + return; + } + + // Do capture probe first. + if ( dsinfo.isValid == false ) + goto playback_probe; + + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Could not create DirectSound capture object (%s): %s.", + info->name, getErrorString(result)); + error(RtError::WARNING); + goto playback_probe; + } + + DSCCAPS in_caps; + in_caps.dwSize = sizeof(in_caps); + result = input->GetCaps( &in_caps ); + if ( FAILED(result) ) { + input->Release(); + sprintf(message, "RtAudio: Could not get DirectSound capture capabilities (%s): %s.", + info->name, getErrorString(result)); + error(RtError::WARNING); + goto playback_probe; + } + + // Get input channel information. + info->minInputChannels = 1; + info->maxInputChannels = in_caps.dwChannels; + + // Get sample rate and format information. + if( in_caps.dwChannels == 2 ) { + if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->nativeFormats |= RTAUDIO_SINT16; + if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->nativeFormats |= RTAUDIO_SINT16; + if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->nativeFormats |= RTAUDIO_SINT16; + if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->nativeFormats |= RTAUDIO_SINT8; + if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->nativeFormats |= RTAUDIO_SINT8; + if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->nativeFormats |= RTAUDIO_SINT8; + + if ( info->nativeFormats & RTAUDIO_SINT16 ) { + if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->sampleRates[info->nSampleRates++] = 11025; + if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->sampleRates[info->nSampleRates++] = 22050; + if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->sampleRates[info->nSampleRates++] = 44100; + } + else if ( info->nativeFormats & RTAUDIO_SINT8 ) { + if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->sampleRates[info->nSampleRates++] = 11025; + if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->sampleRates[info->nSampleRates++] = 22050; + if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->sampleRates[info->nSampleRates++] = 44100; + } + } + else if ( in_caps.dwChannels == 1 ) { + if( in_caps.dwFormats & WAVE_FORMAT_1M16 ) info->nativeFormats |= RTAUDIO_SINT16; + if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->nativeFormats |= RTAUDIO_SINT16; + if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->nativeFormats |= RTAUDIO_SINT16; + if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->nativeFormats |= RTAUDIO_SINT8; + if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->nativeFormats |= RTAUDIO_SINT8; + if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->nativeFormats |= RTAUDIO_SINT8; + + if ( info->nativeFormats & RTAUDIO_SINT16 ) { + if( in_caps.dwFormats & WAVE_FORMAT_1M16 ) info->sampleRates[info->nSampleRates++] = 11025; + if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->sampleRates[info->nSampleRates++] = 22050; + if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->sampleRates[info->nSampleRates++] = 44100; + } + else if ( info->nativeFormats & RTAUDIO_SINT8 ) { + if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->sampleRates[info->nSampleRates++] = 11025; + if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->sampleRates[info->nSampleRates++] = 22050; + if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->sampleRates[info->nSampleRates++] = 44100; + } + } + else info->minInputChannels = 0; // technically, this would be an error + + input->Release(); + + playback_probe: + + dsinfo.isValid = false; + + // Enumerate through output devices to find the id (if it exists). + result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Error performing output device id enumeration: %s.", + getErrorString(result)); + error(RtError::WARNING); + return; + } + + // Now do playback probe. + if ( dsinfo.isValid == false ) + goto check_parameters; + + LPDIRECTSOUND output; + DSCAPS out_caps; + result = DirectSoundCreate( dsinfo.id, &output, NULL ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Could not create DirectSound playback object (%s): %s.", + info->name, getErrorString(result)); + error(RtError::WARNING); + goto check_parameters; + } + + out_caps.dwSize = sizeof(out_caps); + result = output->GetCaps( &out_caps ); + if ( FAILED(result) ) { + output->Release(); + sprintf(message, "RtAudio: Could not get DirectSound playback capabilities (%s): %s.", + info->name, getErrorString(result)); + error(RtError::WARNING); + goto check_parameters; + } + + // Get output channel information. + info->minOutputChannels = 1; + info->maxOutputChannels = ( out_caps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; + + // Get sample rate information. Use capture device rate information + // if it exists. + if ( info->nSampleRates == 0 ) { + info->sampleRates[0] = (int) out_caps.dwMinSecondarySampleRate; + info->sampleRates[1] = (int) out_caps.dwMaxSecondarySampleRate; + if ( out_caps.dwFlags & DSCAPS_CONTINUOUSRATE ) + info->nSampleRates = -1; + else if ( out_caps.dwMinSecondarySampleRate == out_caps.dwMaxSecondarySampleRate ) { + if ( out_caps.dwMinSecondarySampleRate == 0 ) { + // This is a bogus driver report ... fake the range and cross + // your fingers. + info->sampleRates[0] = 11025; + info->sampleRates[1] = 48000; + info->nSampleRates = -1; /* continuous range */ + sprintf(message, "RtAudio: bogus sample rates reported by DirectSound driver ... using defaults (%s).", + info->name); + error(RtError::DEBUG_WARNING); + } + else { + info->nSampleRates = 1; + } + } + else if ( (out_caps.dwMinSecondarySampleRate < 1000.0) && + (out_caps.dwMaxSecondarySampleRate > 50000.0) ) { + // This is a bogus driver report ... support for only two + // distant rates. We'll assume this is a range. + info->nSampleRates = -1; + sprintf(message, "RtAudio: bogus sample rates reported by DirectSound driver ... using range (%s).", + info->name); + error(RtError::WARNING); + } + else info->nSampleRates = 2; + } + else { + // Check input rates against output rate range + for ( int i=info->nSampleRates-1; i>=0; i-- ) { + if ( info->sampleRates[i] <= out_caps.dwMaxSecondarySampleRate ) + break; + info->nSampleRates--; + } + while ( info->sampleRates[0] < out_caps.dwMinSecondarySampleRate ) { + info->nSampleRates--; + for ( int i=0; inSampleRates; i++) + info->sampleRates[i] = info->sampleRates[i+1]; + if ( info->nSampleRates <= 0 ) break; + } + } + + // Get format information. + if ( out_caps.dwFlags & DSCAPS_PRIMARY16BIT ) info->nativeFormats |= RTAUDIO_SINT16; + if ( out_caps.dwFlags & DSCAPS_PRIMARY8BIT ) info->nativeFormats |= RTAUDIO_SINT8; + + output->Release(); + + check_parameters: + if ( info->maxInputChannels == 0 && info->maxOutputChannels == 0 ) + return; + if ( info->nSampleRates == 0 || info->nativeFormats == 0 ) + return; + + // Determine duplex status. + if (info->maxInputChannels < info->maxOutputChannels) + info->maxDuplexChannels = info->maxInputChannels; + else + info->maxDuplexChannels = info->maxOutputChannels; + if (info->minInputChannels < info->minOutputChannels) + info->minDuplexChannels = info->minInputChannels; + else + info->minDuplexChannels = info->minOutputChannels; + + if ( info->maxDuplexChannels > 0 ) info->hasDuplexSupport = true; + else info->hasDuplexSupport = false; + + info->probed = true; + + return; +} + +bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, + STREAM_MODE mode, int channels, + int sampleRate, RTAUDIO_FORMAT format, + int *bufferSize, int numberOfBuffers) +{ + HRESULT result; + HWND hWnd = GetForegroundWindow(); + // According to a note in PortAudio, using GetDesktopWindow() + // instead of GetForegroundWindow() is supposed to avoid problems + // that occur when the application's window is not the foreground + // 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(). + long buffer_size; + LPVOID audioPtr; + DWORD dataLen; + int nBuffers; + + // 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. + if (numberOfBuffers < 2) + nBuffers = 2; + else + nBuffers = numberOfBuffers; + + // Define the wave format structure (16-bit PCM, srate, channels) + WAVEFORMATEX waveFormat; + ZeroMemory(&waveFormat, sizeof(WAVEFORMATEX)); + waveFormat.wFormatTag = WAVE_FORMAT_PCM; + waveFormat.nChannels = channels; + waveFormat.nSamplesPerSec = (unsigned long) sampleRate; + + // Determine the data format. + if ( devices[device].nativeFormats ) { // 8-bit and/or 16-bit support + if ( format == RTAUDIO_SINT8 ) { + if ( devices[device].nativeFormats & RTAUDIO_SINT8 ) + waveFormat.wBitsPerSample = 8; + else + waveFormat.wBitsPerSample = 16; + } + else { + if ( devices[device].nativeFormats & RTAUDIO_SINT16 ) + waveFormat.wBitsPerSample = 16; + else + waveFormat.wBitsPerSample = 8; + } + } + else { + sprintf(message, "RtAudio: no reported data formats for DirectSound device (%s).", + devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + + enum_info dsinfo; + strncpy( dsinfo.name, devices[device].name, 64 ); + dsinfo.isValid = false; + if ( mode == OUTPUT ) { + + if ( devices[device].maxOutputChannels < channels ) + return FAILURE; + + // Enumerate through output devices to find the id (if it exists). + result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Error performing output device id enumeration: %s.", + getErrorString(result)); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + if ( dsinfo.isValid == false ) { + sprintf(message, "RtAudio: DS output device (%s) id not found!", devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + LPGUID id = dsinfo.id; + LPDIRECTSOUND object; + LPDIRECTSOUNDBUFFER buffer; + DSBUFFERDESC bufferDescription; + + result = DirectSoundCreate( id, &object, NULL ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Could not create DirectSound playback object (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + // Set cooperative level to DSSCL_EXCLUSIVE + result = object->SetCooperativeLevel(hWnd, DSSCL_EXCLUSIVE); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to set DirectSound cooperative level (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + // Even though we will write to the secondary buffer, we need to + // access the primary buffer to set the correct output format. + // The default is 8-bit, 22 kHz! + // Setup the DS primary buffer description. + ZeroMemory(&bufferDescription, sizeof(DSBUFFERDESC)); + bufferDescription.dwSize = sizeof(DSBUFFERDESC); + bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; + // Obtain the primary buffer + result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to access DS primary buffer (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + // Set the primary DS buffer sound format. + result = buffer->SetFormat(&waveFormat); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to set DS primary buffer format (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + // Setup the secondary DS buffer description. + buffer_size = channels * *bufferSize * nBuffers * waveFormat.wBitsPerSample / 8; + ZeroMemory(&bufferDescription, sizeof(DSBUFFERDESC)); + bufferDescription.dwSize = sizeof(DSBUFFERDESC); + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCHARDWARE ); // Force hardware mixing + bufferDescription.dwBufferBytes = buffer_size; + 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 = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL); + if ( FAILED(result) ) { + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCSOFTWARE ); // Force software mixing + result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to create secondary DS buffer (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + } + + // Get the buffer size ... might be different from what we specified. + DSBCAPS dsbcaps; + dsbcaps.dwSize = sizeof(DSBCAPS); + buffer->GetCaps(&dsbcaps); + buffer_size = dsbcaps.dwBufferBytes; + + // Lock the DS buffer + result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to lock DS buffer (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + // Zero the DS buffer + ZeroMemory(audioPtr, dataLen); + + // Unlock the DS buffer + result = buffer->Unlock(audioPtr, dataLen, NULL, 0); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to unlock DS buffer(%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + stream->handle[0].object = (void *) object; + stream->handle[0].buffer = (void *) buffer; + stream->nDeviceChannels[0] = channels; + } + + if ( mode == INPUT ) { + + if ( devices[device].maxInputChannels < channels ) + return FAILURE; + + // Enumerate through input devices to find the id (if it exists). + result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Error performing input device id enumeration: %s.", + getErrorString(result)); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + if ( dsinfo.isValid == false ) { + sprintf(message, "RtAudio: DS input device (%s) id not found!", devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + LPGUID id = dsinfo.id; + LPDIRECTSOUNDCAPTURE object; + LPDIRECTSOUNDCAPTUREBUFFER buffer; + DSCBUFFERDESC bufferDescription; + + result = DirectSoundCaptureCreate( id, &object, NULL ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Could not create DirectSound capture object (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + // Setup the secondary DS buffer description. + buffer_size = channels * *bufferSize * nBuffers * waveFormat.wBitsPerSample / 8; + ZeroMemory(&bufferDescription, sizeof(DSCBUFFERDESC)); + bufferDescription.dwSize = sizeof(DSCBUFFERDESC); + bufferDescription.dwFlags = 0; + bufferDescription.dwReserved = 0; + bufferDescription.dwBufferBytes = buffer_size; + bufferDescription.lpwfxFormat = &waveFormat; + + // Create the capture buffer. + result = object->CreateCaptureBuffer(&bufferDescription, &buffer, NULL); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to create DS capture buffer (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + // Lock the capture buffer + result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + // Zero the buffer + ZeroMemory(audioPtr, dataLen); + + // Unlock the buffer + result = buffer->Unlock(audioPtr, dataLen, NULL, 0); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + stream->handle[1].object = (void *) object; + stream->handle[1].buffer = (void *) buffer; + stream->nDeviceChannels[1] = channels; + } + + stream->userFormat = format; + if ( waveFormat.wBitsPerSample == 8 ) + stream->deviceFormat[mode] = RTAUDIO_SINT8; + else + stream->deviceFormat[mode] = RTAUDIO_SINT16; + stream->nUserChannels[mode] = channels; + *bufferSize = buffer_size / (channels * nBuffers * waveFormat.wBitsPerSample / 8); + stream->bufferSize = *bufferSize; + + // Set flags for buffer conversion + stream->doConvertBuffer[mode] = false; + if (stream->userFormat != stream->deviceFormat[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode]) + stream->doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { + + long buffer_bytes; + if (stream->nUserChannels[0] >= stream->nUserChannels[1]) + buffer_bytes = stream->nUserChannels[0]; + else + buffer_bytes = stream->nUserChannels[1]; + + buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); + if (stream->userBuffer) free(stream->userBuffer); + stream->userBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->userBuffer == NULL) + goto memory_error; + } + + if ( stream->doConvertBuffer[mode] ) { + + long buffer_bytes; + bool makeBuffer = true; + if ( mode == OUTPUT ) + buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + else { // mode == INPUT + buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); + if ( stream->mode == OUTPUT && stream->deviceBuffer ) { + long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + if ( buffer_bytes < bytes_out ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + buffer_bytes *= *bufferSize; + if (stream->deviceBuffer) free(stream->deviceBuffer); + stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->deviceBuffer == NULL) + goto memory_error; + } + } + + stream->device[mode] = device; + stream->state = STREAM_STOPPED; + if ( stream->mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream->mode = DUPLEX; + else + stream->mode = mode; + stream->nBuffers = nBuffers; + stream->sampleRate = sampleRate; + + return SUCCESS; + + memory_error: + if (stream->handle[0].object) { + LPDIRECTSOUND object = (LPDIRECTSOUND) stream->handle[0].object; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; + if (buffer) { + buffer->Release(); + stream->handle[0].buffer = NULL; + } + object->Release(); + stream->handle[0].object = NULL; + } + if (stream->handle[1].object) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) stream->handle[1].object; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; + if (buffer) { + buffer->Release(); + stream->handle[1].buffer = NULL; + } + object->Release(); + stream->handle[1].object = NULL; + } + if (stream->userBuffer) { + free(stream->userBuffer); + stream->userBuffer = 0; + } + sprintf(message, "RtAudio: error allocating buffer memory (%s).", + devices[device].name); + error(RtError::WARNING); + return FAILURE; +} + +void RtAudio :: cancelStreamCallback(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + if (stream->callbackInfo.usingCallback) { + + if (stream->state == STREAM_RUNNING) + stopStream( streamId ); + + MUTEX_LOCK(&stream->mutex); + + stream->callbackInfo.usingCallback = false; + WaitForSingleObject( (HANDLE)stream->callbackInfo.thread, INFINITE ); + CloseHandle( (HANDLE)stream->callbackInfo.thread ); + stream->callbackInfo.thread = 0; + stream->callbackInfo.callback = NULL; + stream->callbackInfo.userData = NULL; + + MUTEX_UNLOCK(&stream->mutex); + } +} + +void RtAudio :: closeStream(int streamId) +{ + // We don't want an exception to be thrown here because this + // function is called by our class destructor. So, do our own + // streamId check. + if ( streams.find( streamId ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtError::WARNING); + return; + } + + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + + if (stream->callbackInfo.usingCallback) { + stream->callbackInfo.usingCallback = false; + WaitForSingleObject( (HANDLE)stream->callbackInfo.thread, INFINITE ); + CloseHandle( (HANDLE)stream->callbackInfo.thread ); + } + + DeleteCriticalSection(&stream->mutex); + + if (stream->handle[0].object) { + LPDIRECTSOUND object = (LPDIRECTSOUND) stream->handle[0].object; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; + if (buffer) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + + if (stream->handle[1].object) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) stream->handle[1].object; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; + if (buffer) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + + if (stream->userBuffer) + free(stream->userBuffer); + + if (stream->deviceBuffer) + free(stream->deviceBuffer); + + free(stream); + streams.erase(streamId); +} + +void RtAudio :: startStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_RUNNING) + goto unlock; + + HRESULT result; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; + result = buffer->Play(0, 0, DSBPLAY_LOOPING ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to start DS buffer (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; + result = buffer->Start(DSCBSTART_LOOPING ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to start DS capture buffer (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + } + stream->state = STREAM_RUNNING; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: stopStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) { + MUTEX_UNLOCK(&stream->mutex); + return; + } + + // 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 + // concept is less clear so we'll repeat what we do in the + // abortStream() case. + HRESULT result; + DWORD dsBufferSize; + LPVOID buffer1 = NULL; + LPVOID buffer2 = NULL; + DWORD bufferSize1 = 0; + DWORD bufferSize2 = 0; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + DWORD currentPos, safePos; + long buffer_bytes = stream->bufferSize * stream->nDeviceChannels[0]; + buffer_bytes *= formatBytes(stream->deviceFormat[0]); + + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; + UINT nextWritePos = stream->handle[0].bufferPointer; + dsBufferSize = buffer_bytes * stream->nBuffers; + + // Write zeroes for nBuffer counts. + for (int i=0; inBuffers; i++) { + + // Find out where the read and "safe write" pointers are. + result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset + DWORD endWrite = nextWritePos + buffer_bytes; + + // Check whether the entire write region is behind the play pointer. + while ( currentPos < endWrite ) { + float millis = (endWrite - currentPos) * 900.0; + millis /= ( formatBytes(stream->deviceFormat[0]) * stream->sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + + // Wake up, find out where we are now + result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset + } + + // Lock free space in the buffer + result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to lock DS buffer during playback (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // Zero the free space + ZeroMemory(buffer1, bufferSize1); + if (buffer2 != NULL) ZeroMemory(buffer2, bufferSize2); + + // Update our buffer offset and unlock sound buffer + dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to unlock DS buffer during playback (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize; + stream->handle[0].bufferPointer = nextWritePos; + } + + // If we play again, start at the beginning of the buffer. + stream->handle[0].bufferPointer = 0; + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; + buffer1 = NULL; + bufferSize1 = 0; + + result = buffer->Stop(); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to stop DS capture buffer (%s): %s", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + dsBufferSize = stream->bufferSize * stream->nDeviceChannels[1]; + dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers; + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock(0, dsBufferSize, &buffer1, &bufferSize1, NULL, NULL, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // Zero the DS buffer + ZeroMemory(buffer1, bufferSize1); + + // Unlock the DS buffer + result = buffer->Unlock(buffer1, bufferSize1, NULL, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // If we start recording again, we must begin at beginning of buffer. + stream->handle[1].bufferPointer = 0; + } + stream->state = STREAM_STOPPED; + + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: abortStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + HRESULT result; + long dsBufferSize; + LPVOID audioPtr; + DWORD dataLen; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; + result = buffer->Stop(); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to stop DS buffer (%s): %s", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + dsBufferSize = stream->bufferSize * stream->nDeviceChannels[0]; + dsBufferSize *= formatBytes(stream->deviceFormat[0]) * stream->nBuffers; + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to lock DS buffer (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // Zero the DS buffer + ZeroMemory(audioPtr, dataLen); + + // Unlock the DS buffer + result = buffer->Unlock(audioPtr, dataLen, NULL, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to unlock DS buffer (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // If we start playing again, we must begin at beginning of buffer. + stream->handle[0].bufferPointer = 0; + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; + audioPtr = NULL; + dataLen = 0; + + result = buffer->Stop(); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to stop DS capture buffer (%s): %s", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + dsBufferSize = stream->bufferSize * stream->nDeviceChannels[1]; + dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers; + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // Zero the DS buffer + ZeroMemory(audioPtr, dataLen); + + // Unlock the DS buffer + result = buffer->Unlock(audioPtr, dataLen, NULL, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // If we start recording again, we must begin at beginning of buffer. + stream->handle[1].bufferPointer = 0; + } + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +int RtAudio :: streamWillBlock(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + int channels; + int frames = 0; + if (stream->state == STREAM_STOPPED) + goto unlock; + + HRESULT result; + DWORD currentPos, safePos; + channels = 1; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; + UINT nextWritePos = stream->handle[0].bufferPointer; + channels = stream->nDeviceChannels[0]; + DWORD dsBufferSize = stream->bufferSize * channels; + dsBufferSize *= formatBytes(stream->deviceFormat[0]) * stream->nBuffers; + + // Find out where the read and "safe write" pointers are. + result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset + frames = currentPos - nextWritePos; + frames /= channels * formatBytes(stream->deviceFormat[0]); + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + + LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; + UINT nextReadPos = stream->handle[1].bufferPointer; + channels = stream->nDeviceChannels[1]; + DWORD dsBufferSize = stream->bufferSize * channels; + dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers; + + // Find out where the write and "safe read" pointers are. + result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset + + if (stream->mode == DUPLEX ) { + // Take largest value of the two. + int temp = safePos - nextReadPos; + temp /= channels * formatBytes(stream->deviceFormat[1]); + frames = ( temp > frames ) ? temp : frames; + } + else { + frames = safePos - nextReadPos; + frames /= channels * formatBytes(stream->deviceFormat[1]); + } + } + + frames = stream->bufferSize - frames; + if (frames < 0) frames = 0; + + unlock: + MUTEX_UNLOCK(&stream->mutex); + return frames; +} + +void RtAudio :: tickStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + int stopStream = 0; + if (stream->state == STREAM_STOPPED) { + if (stream->callbackInfo.usingCallback) Sleep(50); // sleep 50 milliseconds + return; + } + else if (stream->callbackInfo.usingCallback) { + RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback; + stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData); + } + + MUTEX_LOCK(&stream->mutex); + + // The state might change while waiting on a mutex. + if (stream->state == STREAM_STOPPED) { + MUTEX_UNLOCK(&stream->mutex); + return; + } + + HRESULT result; + DWORD currentPos, safePos; + LPVOID buffer1 = NULL; + LPVOID buffer2 = NULL; + DWORD bufferSize1 = 0; + DWORD bufferSize2 = 0; + char *buffer; + long buffer_bytes; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + // Setup parameters and do buffer conversion if necessary. + if (stream->doConvertBuffer[0]) { + convertStreamBuffer(stream, OUTPUT); + buffer = stream->deviceBuffer; + buffer_bytes = stream->bufferSize * stream->nDeviceChannels[0]; + buffer_bytes *= formatBytes(stream->deviceFormat[0]); + } + else { + buffer = stream->userBuffer; + buffer_bytes = stream->bufferSize * stream->nUserChannels[0]; + buffer_bytes *= formatBytes(stream->userFormat); + } + + // No byte swapping necessary in DirectSound implementation. + + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; + UINT nextWritePos = stream->handle[0].bufferPointer; + DWORD dsBufferSize = buffer_bytes * stream->nBuffers; + + // Find out where the read and "safe write" pointers are. + result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset + DWORD endWrite = nextWritePos + buffer_bytes; + + // Check whether the entire write region is behind the play pointer. + while ( currentPos < endWrite ) { + // 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. + float millis = (endWrite - currentPos) * 900.0; + millis /= ( formatBytes(stream->deviceFormat[0]) * stream->sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + + // Wake up, find out where we are now + result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset + } + + // Lock free space in the buffer + result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to lock DS buffer during playback (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_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) ) { + sprintf(message, "RtAudio: Unable to unlock DS buffer during playback (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize; + stream->handle[0].bufferPointer = nextWritePos; + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + + // Setup parameters. + if (stream->doConvertBuffer[1]) { + buffer = stream->deviceBuffer; + buffer_bytes = stream->bufferSize * stream->nDeviceChannels[1]; + buffer_bytes *= formatBytes(stream->deviceFormat[1]); + } + else { + buffer = stream->userBuffer; + buffer_bytes = stream->bufferSize * stream->nUserChannels[1]; + buffer_bytes *= formatBytes(stream->userFormat); + } + + LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; + UINT nextReadPos = stream->handle[1].bufferPointer; + DWORD dsBufferSize = buffer_bytes * stream->nBuffers; + + // Find out where the write and "safe read" pointers are. + result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset + DWORD endRead = nextReadPos + buffer_bytes; + + // Check whether the entire write region is behind the play pointer. + while ( safePos < endRead ) { + // See comments for playback. + float millis = (endRead - safePos) * 900.0; + millis /= ( formatBytes(stream->deviceFormat[1]) * stream->sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + + // Wake up, find out where we are now + result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset + } + + // Lock free space in the buffer + result = dsBuffer->Lock (nextReadPos, buffer_bytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to lock DS buffer during capture (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // Copy our buffer into the DS buffer + CopyMemory(buffer, buffer1, bufferSize1); + if (buffer2 != NULL) CopyMemory(buffer+bufferSize1, buffer2, bufferSize2); + + // Update our buffer offset and unlock sound buffer + nextReadPos = (nextReadPos + bufferSize1 + bufferSize2) % dsBufferSize; + dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to unlock DS buffer during capture (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + stream->handle[1].bufferPointer = nextReadPos; + + // No byte swapping necessary in DirectSound implementation. + + // Do buffer conversion if necessary. + if (stream->doConvertBuffer[1]) + convertStreamBuffer(stream, INPUT); + } + + MUTEX_UNLOCK(&stream->mutex); + + if (stream->callbackInfo.usingCallback && stopStream) + this->stopStream(streamId); +} + +// Definitions for utility functions and callbacks +// specific to the DirectSound implementation. + +extern "C" unsigned __stdcall callbackHandler(void *ptr) +{ + CALLBACK_INFO *info = (CALLBACK_INFO *) ptr; + RtAudio *object = (RtAudio *) info->object; + int stream = info->streamId; + bool *usingCallback = &info->usingCallback; + + while ( *usingCallback ) { + try { + object->tickStream(stream); + } + catch (RtError &exception) { + fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n", + exception.getMessage()); + break; + } + } + + _endthreadex( 0 ); + return 0; +} + +void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + CALLBACK_INFO *info = (CALLBACK_INFO *) &stream->callbackInfo; + if ( info->usingCallback ) { + sprintf(message, "RtAudio: A callback is already set for this stream!"); + error(RtError::WARNING); + return; + } + + info->callback = (void *) callback; + info->userData = userData; + info->usingCallback = true; + info->object = (void *) this; + info->streamId = streamId; + + unsigned thread_id; + info->thread = _beginthreadex(NULL, 0, &callbackHandler, + &stream->callbackInfo, 0, &thread_id); + if (info->thread == 0) { + info->usingCallback = false; + sprintf(message, "RtAudio: error starting callback thread!"); + error(RtError::THREAD_ERROR); + } + + // When spawning multiple threads in quick succession, it appears to be + // necessary to wait a bit for each to initialize ... another windoism! + Sleep(1); +} + +static bool CALLBACK deviceCountCallback(LPGUID lpguid, + LPCSTR lpcstrDescription, + LPCSTR lpcstrModule, + LPVOID lpContext) +{ + int *pointer = ((int *) lpContext); + (*pointer)++; + + return true; +} + +static bool CALLBACK deviceInfoCallback(LPGUID lpguid, + LPCSTR lpcstrDescription, + LPCSTR lpcstrModule, + LPVOID lpContext) +{ + enum_info *info = ((enum_info *) lpContext); + while (strlen(info->name) > 0) info++; + + strncpy(info->name, lpcstrDescription, 64); + info->id = lpguid; + + HRESULT hr; + info->isValid = false; + 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->isValid = true; + } + 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->isValid = true; + } + object->Release(); + } + + return true; +} + +static bool CALLBACK defaultDeviceCallback(LPGUID lpguid, + LPCSTR lpcstrDescription, + LPCSTR lpcstrModule, + LPVOID lpContext) +{ + enum_info *info = ((enum_info *) lpContext); + + if ( lpguid == NULL ) { + strncpy(info->name, lpcstrDescription, 64); + return false; + } + + return true; +} + +static bool CALLBACK deviceIdCallback(LPGUID lpguid, + LPCSTR lpcstrDescription, + LPCSTR lpcstrModule, + LPVOID lpContext) +{ + enum_info *info = ((enum_info *) lpContext); + + if ( strncmp( info->name, lpcstrDescription, 64 ) == 0 ) { + info->id = lpguid; + info->isValid = true; + return false; + } + + return true; +} + +static char* getErrorString(int code) +{ + switch (code) { + + case DSERR_ALLOCATED: + return "Direct Sound already allocated"; + + case DSERR_CONTROLUNAVAIL: + return "Direct Sound control unavailable"; + + case DSERR_INVALIDPARAM: + return "Direct Sound invalid parameter"; + + case DSERR_INVALIDCALL: + return "Direct Sound invalid call"; + + case DSERR_GENERIC: + return "Direct Sound generic error"; + + case DSERR_PRIOLEVELNEEDED: + return "Direct Sound Priority level needed"; + + case DSERR_OUTOFMEMORY: + return "Direct Sound out of memory"; + + case DSERR_BADFORMAT: + return "Direct Sound bad format"; + + case DSERR_UNSUPPORTED: + return "Direct Sound unsupported error"; + + case DSERR_NODRIVER: + return "Direct Sound no driver error"; + + case DSERR_ALREADYINITIALIZED: + return "Direct Sound already initialized"; + + case DSERR_NOAGGREGATION: + return "Direct Sound no aggregation"; + + case DSERR_BUFFERLOST: + return "Direct Sound buffer lost"; + + case DSERR_OTHERAPPHASPRIO: + return "Direct Sound other app has priority"; + + case DSERR_UNINITIALIZED: + return "Direct Sound uninitialized"; + + default: + return "Direct Sound unknown error"; + } +} + +//******************** End of __WINDOWS_DS__ *********************// + +#elif defined(__IRIX_AL__) // SGI's AL API for IRIX + +#include +#include + +void RtAudio :: initialize(void) +{ + // Count cards and devices + nDevices = 0; + + // Determine the total number of input and output devices. + nDevices = alQueryValues(AL_SYSTEM, AL_DEVICES, 0, 0, 0, 0); + if (nDevices < 0) { + sprintf(message, "RtAudio: AL error counting devices: %s.", + alGetErrorString(oserror())); + error(RtError::DRIVER_ERROR); + } + + if (nDevices <= 0) return; + + ALvalue *vls = (ALvalue *) new ALvalue[nDevices]; + + // Allocate the RTAUDIO_DEVICE structures. + devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); + if (devices == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + + // Write device ascii identifiers and resource ids to device info + // structure. + char name[32]; + int outs, ins, i; + ALpv pvs[1]; + pvs[0].param = AL_NAME; + pvs[0].value.ptr = name; + pvs[0].sizeIn = 32; + + outs = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, vls, nDevices, 0, 0); + if (outs < 0) { + sprintf(message, "RtAudio: AL error getting output devices: %s.", + alGetErrorString(oserror())); + error(RtError::DRIVER_ERROR); + } + + for (i=0; iid[0]; + if (resource > 0) { + + // Probe output device parameters. + result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0); + if (result < 0) { + sprintf(message, "RtAudio: AL error getting device (%s) channels: %s.", + info->name, alGetErrorString(oserror())); + error(RtError::WARNING); + } + else { + info->maxOutputChannels = value.i; + info->minOutputChannels = 1; + } + + result = alGetParamInfo(resource, AL_RATE, &pinfo); + if (result < 0) { + sprintf(message, "RtAudio: AL error getting device (%s) rates: %s.", + info->name, alGetErrorString(oserror())); + error(RtError::WARNING); + } + else { + info->nSampleRates = 0; + for (i=0; i= pinfo.min.i && SAMPLE_RATES[i] <= pinfo.max.i ) { + info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i]; + info->nSampleRates++; + } + } + } + + // The AL library supports all our formats, except 24-bit and 32-bit ints. + info->nativeFormats = (RTAUDIO_FORMAT) 51; + } + + // Now get input resource ID if it exists. + resource = info->id[1]; + if (resource > 0) { + + // Probe input device parameters. + result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0); + if (result < 0) { + sprintf(message, "RtAudio: AL error getting device (%s) channels: %s.", + info->name, alGetErrorString(oserror())); + error(RtError::WARNING); + } + else { + info->maxInputChannels = value.i; + info->minInputChannels = 1; + } + + result = alGetParamInfo(resource, AL_RATE, &pinfo); + if (result < 0) { + sprintf(message, "RtAudio: AL error getting device (%s) rates: %s.", + info->name, alGetErrorString(oserror())); + error(RtError::WARNING); + } + else { + // In the case of the default device, these values will + // overwrite the rates determined for the output device. Since + // the input device is most likely to be more limited than the + // output device, this is ok. + info->nSampleRates = 0; + for (i=0; i= pinfo.min.i && SAMPLE_RATES[i] <= pinfo.max.i ) { + info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i]; + info->nSampleRates++; + } + } + } + + // The AL library supports all our formats, except 24-bit and 32-bit ints. + info->nativeFormats = (RTAUDIO_FORMAT) 51; + } + + if ( info->maxInputChannels == 0 && info->maxOutputChannels == 0 ) + return; + if ( info->nSampleRates == 0 ) + return; + + // Determine duplex status. + if (info->maxInputChannels < info->maxOutputChannels) + info->maxDuplexChannels = info->maxInputChannels; + else + info->maxDuplexChannels = info->maxOutputChannels; + if (info->minInputChannels < info->minOutputChannels) + info->minDuplexChannels = info->minInputChannels; + else + info->minDuplexChannels = info->minOutputChannels; + + if ( info->maxDuplexChannels > 0 ) info->hasDuplexSupport = true; + else info->hasDuplexSupport = false; + + info->probed = true; + + return; +} + +bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, + STREAM_MODE mode, int channels, + int sampleRate, RTAUDIO_FORMAT format, + int *bufferSize, int numberOfBuffers) +{ + int result, resource, nBuffers; + ALconfig al_config; + ALport port; + ALpv pvs[2]; + + // Get a new ALconfig structure. + al_config = alNewConfig(); + if ( !al_config ) { + sprintf(message,"RtAudio: can't get AL config: %s.", + alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + + // Set the channels. + result = alSetChannels(al_config, channels); + if ( result < 0 ) { + sprintf(message,"RtAudio: can't set %d channels in AL config: %s.", + channels, alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + + // Attempt to set the queue size. The al API doesn't provide a + // means for querying the minimum/maximum buffer size of a device, + // so if the specified size doesn't work, take whatever the + // al_config structure returns. + if ( numberOfBuffers < 1 ) + nBuffers = 1; + else + nBuffers = numberOfBuffers; + long buffer_size = *bufferSize * nBuffers; + result = alSetQueueSize(al_config, buffer_size); // in sample frames + if ( result < 0 ) { + // Get the buffer size specified by the al_config and try that. + buffer_size = alGetQueueSize(al_config); + result = alSetQueueSize(al_config, buffer_size); + if ( result < 0 ) { + sprintf(message,"RtAudio: can't set buffer size (%ld) in AL config: %s.", + buffer_size, alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + *bufferSize = buffer_size / nBuffers; + } + + // Set the data format. + stream->userFormat = format; + stream->deviceFormat[mode] = format; + if (format == RTAUDIO_SINT8) { + result = alSetSampFmt(al_config, AL_SAMPFMT_TWOSCOMP); + result = alSetWidth(al_config, AL_SAMPLE_8); + } + else if (format == RTAUDIO_SINT16) { + result = alSetSampFmt(al_config, AL_SAMPFMT_TWOSCOMP); + result = alSetWidth(al_config, AL_SAMPLE_16); + } + else if (format == RTAUDIO_SINT24) { + // Our 24-bit format assumes the upper 3 bytes of a 4 byte word. + // The AL library uses the lower 3 bytes, so we'll need to do our + // own conversion. + result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT); + stream->deviceFormat[mode] = RTAUDIO_FLOAT32; + } + else if (format == RTAUDIO_SINT32) { + // The AL library doesn't seem to support the 32-bit integer + // format, so we'll need to do our own conversion. + result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT); + stream->deviceFormat[mode] = RTAUDIO_FLOAT32; + } + else if (format == RTAUDIO_FLOAT32) + result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT); + else if (format == RTAUDIO_FLOAT64) + result = alSetSampFmt(al_config, AL_SAMPFMT_DOUBLE); + + if ( result == -1 ) { + sprintf(message,"RtAudio: AL error setting sample format in AL config: %s.", + alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + + if (mode == OUTPUT) { + + // Set our device. + if (device == 0) + resource = AL_DEFAULT_OUTPUT; + else + resource = devices[device].id[0]; + result = alSetDevice(al_config, resource); + if ( result == -1 ) { + sprintf(message,"RtAudio: AL error setting device (%s) in AL config: %s.", + devices[device].name, alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + + // Open the port. + port = alOpenPort("RtAudio Output Port", "w", al_config); + if( !port ) { + sprintf(message,"RtAudio: AL error opening output port: %s.", + alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + + // Set the sample rate + pvs[0].param = AL_MASTER_CLOCK; + pvs[0].value.i = AL_CRYSTAL_MCLK_TYPE; + pvs[1].param = AL_RATE; + pvs[1].value.ll = alDoubleToFixed((double)sampleRate); + result = alSetParams(resource, pvs, 2); + if ( result < 0 ) { + alClosePort(port); + sprintf(message,"RtAudio: AL error setting sample rate (%d) for device (%s): %s.", + sampleRate, devices[device].name, alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + } + else { // mode == INPUT + + // Set our device. + if (device == 0) + resource = AL_DEFAULT_INPUT; + else + resource = devices[device].id[1]; + result = alSetDevice(al_config, resource); + if ( result == -1 ) { + sprintf(message,"RtAudio: AL error setting device (%s) in AL config: %s.", + devices[device].name, alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + + // Open the port. + port = alOpenPort("RtAudio Output Port", "r", al_config); + if( !port ) { + sprintf(message,"RtAudio: AL error opening input port: %s.", + alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + + // Set the sample rate + pvs[0].param = AL_MASTER_CLOCK; + pvs[0].value.i = AL_CRYSTAL_MCLK_TYPE; + pvs[1].param = AL_RATE; + pvs[1].value.ll = alDoubleToFixed((double)sampleRate); + result = alSetParams(resource, pvs, 2); + if ( result < 0 ) { + alClosePort(port); + sprintf(message,"RtAudio: AL error setting sample rate (%d) for device (%s): %s.", + sampleRate, devices[device].name, alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + } + + alFreeConfig(al_config); + + stream->nUserChannels[mode] = channels; + stream->nDeviceChannels[mode] = channels; + + // Set handle and flags for buffer conversion + stream->handle[mode] = port; + stream->doConvertBuffer[mode] = false; + if (stream->userFormat != stream->deviceFormat[mode]) + stream->doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { + + long buffer_bytes; + if (stream->nUserChannels[0] >= stream->nUserChannels[1]) + buffer_bytes = stream->nUserChannels[0]; + else + buffer_bytes = stream->nUserChannels[1]; + + buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); + if (stream->userBuffer) free(stream->userBuffer); + stream->userBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->userBuffer == NULL) + goto memory_error; + } + + if ( stream->doConvertBuffer[mode] ) { + + long buffer_bytes; + bool makeBuffer = true; + if ( mode == OUTPUT ) + buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + else { // mode == INPUT + buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); + if ( stream->mode == OUTPUT && stream->deviceBuffer ) { + long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + if ( buffer_bytes < bytes_out ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + buffer_bytes *= *bufferSize; + if (stream->deviceBuffer) free(stream->deviceBuffer); + stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->deviceBuffer == NULL) + goto memory_error; + } + } + + stream->device[mode] = device; + stream->state = STREAM_STOPPED; + if ( stream->mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream->mode = DUPLEX; + else + stream->mode = mode; + stream->nBuffers = nBuffers; + stream->bufferSize = *bufferSize; + stream->sampleRate = sampleRate; + + return SUCCESS; + + memory_error: + if (stream->handle[0]) { + alClosePort(stream->handle[0]); + stream->handle[0] = 0; + } + if (stream->handle[1]) { + alClosePort(stream->handle[1]); + stream->handle[1] = 0; + } + if (stream->userBuffer) { + free(stream->userBuffer); + stream->userBuffer = 0; + } + sprintf(message, "RtAudio: ALSA error allocating buffer memory for device (%s).", + devices[device].name); + error(RtError::WARNING); + return FAILURE; +} + +void RtAudio :: closeStream(int streamId) +{ + // We don't want an exception to be thrown here because this + // function is called by our class destructor. So, do our own + // streamId check. + if ( streams.find( streamId ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtError::WARNING); + return; + } + + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + + if (stream->callbackInfo.usingCallback) { + pthread_cancel(stream->callbackInfo.thread); + pthread_join(stream->callbackInfo.thread, NULL); + } + + pthread_mutex_destroy(&stream->mutex); + + if (stream->handle[0]) + alClosePort(stream->handle[0]); + + if (stream->handle[1]) + alClosePort(stream->handle[1]); + + if (stream->userBuffer) + free(stream->userBuffer); + + if (stream->deviceBuffer) + free(stream->deviceBuffer); + + free(stream); + streams.erase(streamId); +} + +void RtAudio :: startStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + if (stream->state == STREAM_RUNNING) + return; + + // The AL port is ready as soon as it is opened. + stream->state = STREAM_RUNNING; +} + +void RtAudio :: stopStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + int result; + int buffer_size = stream->bufferSize * stream->nBuffers; + + if (stream->mode == OUTPUT || stream->mode == DUPLEX) + alZeroFrames(stream->handle[0], buffer_size); + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + result = alDiscardFrames(stream->handle[1], buffer_size); + if (result == -1) { + sprintf(message, "RtAudio: AL error draining stream device (%s): %s.", + devices[stream->device[1]].name, alGetErrorString(oserror())); + error(RtError::DRIVER_ERROR); + } + } + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: abortStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + int buffer_size = stream->bufferSize * stream->nBuffers; + int result = alDiscardFrames(stream->handle[0], buffer_size); + if (result == -1) { + sprintf(message, "RtAudio: AL error aborting stream device (%s): %s.", + devices[stream->device[0]].name, alGetErrorString(oserror())); + error(RtError::DRIVER_ERROR); + } + } + + // There is no clear action to take on the input stream, since the + // port will continue to run in any event. + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +int RtAudio :: streamWillBlock(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + int frames = 0; + if (stream->state == STREAM_STOPPED) + goto unlock; + + int err = 0; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + err = alGetFillable(stream->handle[0]); + if (err < 0) { + sprintf(message, "RtAudio: AL error getting available frames for stream (%s): %s.", + devices[stream->device[0]].name, alGetErrorString(oserror())); + error(RtError::DRIVER_ERROR); + } + } + + frames = err; + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + err = alGetFilled(stream->handle[1]); + if (err < 0) { + sprintf(message, "RtAudio: AL error getting available frames for stream (%s): %s.", + devices[stream->device[1]].name, alGetErrorString(oserror())); + error(RtError::DRIVER_ERROR); + } + if (frames > err) frames = err; + } + + frames = stream->bufferSize - frames; + if (frames < 0) frames = 0; + + unlock: + MUTEX_UNLOCK(&stream->mutex); + return frames; +} + +void RtAudio :: tickStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + int stopStream = 0; + if (stream->state == STREAM_STOPPED) { + if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds + return; + } + else if (stream->callbackInfo.usingCallback) { + RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback; + stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData); + } + + MUTEX_LOCK(&stream->mutex); + + // The state might change while waiting on a mutex. + if (stream->state == STREAM_STOPPED) + goto unlock; + + char *buffer; + int channels; + RTAUDIO_FORMAT format; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + // Setup parameters and do buffer conversion if necessary. + if (stream->doConvertBuffer[0]) { + convertStreamBuffer(stream, OUTPUT); + buffer = stream->deviceBuffer; + channels = stream->nDeviceChannels[0]; + format = stream->deviceFormat[0]; + } + else { + buffer = stream->userBuffer; + channels = stream->nUserChannels[0]; + format = stream->userFormat; + } + + // Do byte swapping if necessary. + if (stream->doByteSwap[0]) + byteSwapBuffer(buffer, stream->bufferSize * channels, format); + + // Write interleaved samples to device. + alWriteFrames(stream->handle[0], buffer, stream->bufferSize); + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + + // Setup parameters. + if (stream->doConvertBuffer[1]) { + buffer = stream->deviceBuffer; + channels = stream->nDeviceChannels[1]; + format = stream->deviceFormat[1]; + } + else { + buffer = stream->userBuffer; + channels = stream->nUserChannels[1]; + format = stream->userFormat; + } + + // Read interleaved samples from device. + alReadFrames(stream->handle[1], buffer, stream->bufferSize); + + // Do byte swapping if necessary. + if (stream->doByteSwap[1]) + byteSwapBuffer(buffer, stream->bufferSize * channels, format); + + // Do buffer conversion if necessary. + if (stream->doConvertBuffer[1]) + convertStreamBuffer(stream, INPUT); + } + + unlock: + MUTEX_UNLOCK(&stream->mutex); + + if (stream->callbackInfo.usingCallback && stopStream) + this->stopStream(streamId); +} + +extern "C" void *callbackHandler(void *ptr) +{ + CALLBACK_INFO *info = (CALLBACK_INFO *) ptr; + RtAudio *object = (RtAudio *) info->object; + int stream = info->streamId; + bool *usingCallback = &info->usingCallback; + + while ( *usingCallback ) { + pthread_testcancel(); + try { + object->tickStream(stream); + } + catch (RtError &exception) { + fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n", + exception.getMessage()); + break; + } + } + + return 0; +} + +//******************** End of __IRIX_AL__ *********************// + +#endif + + +// *************************************************** // +// +// Private common (OS-independent) RtAudio methods. +// +// *************************************************** // + +// This method can be modified to control the behavior of error +// message reporting and throwing. +void RtAudio :: error(RtError::TYPE type) +{ + if (type == RtError::WARNING) { + fprintf(stderr, "\n%s\n\n", message); + } + else if (type == RtError::DEBUG_WARNING) { +#if defined(__RTAUDIO_DEBUG__) + fprintf(stderr, "\n%s\n\n", message); +#endif + } + else { + fprintf(stderr, "\n%s\n\n", message); + throw RtError(message, type); + } +} + +void *RtAudio :: verifyStream(int streamId) +{ + // Verify the stream key. + if ( streams.find( streamId ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtError::INVALID_STREAM); + } + + return streams[streamId]; +} + +void RtAudio :: clearDeviceInfo(RTAUDIO_DEVICE *info) +{ + // Don't clear the name or DEVICE_ID fields here ... they are + // typically set prior to a call of this function. + info->probed = false; + info->maxOutputChannels = 0; + info->maxInputChannels = 0; + info->maxDuplexChannels = 0; + info->minOutputChannels = 0; + info->minInputChannels = 0; + info->minDuplexChannels = 0; + info->hasDuplexSupport = false; + info->nSampleRates = 0; + for (int i=0; isampleRates[i] = 0; + info->nativeFormats = 0; +} + +int RtAudio :: formatBytes(RTAUDIO_FORMAT 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; + + sprintf(message,"RtAudio: undefined format in formatBytes()."); + error(RtError::WARNING); + + return 0; +} + +void RtAudio :: convertStreamBuffer(RTAUDIO_STREAM *stream, STREAM_MODE mode) +{ + // This method does format conversion, input/output channel compensation, and + // data interleaving/deinterleaving. 24-bit integers are assumed to occupy + // the upper three bytes of a 32-bit integer. + + int j, jump_in, jump_out, channels; + RTAUDIO_FORMAT format_in, format_out; + char *input, *output; + + if (mode == INPUT) { // convert device to user buffer + input = stream->deviceBuffer; + output = stream->userBuffer; + jump_in = stream->nDeviceChannels[1]; + jump_out = stream->nUserChannels[1]; + format_in = stream->deviceFormat[1]; + format_out = stream->userFormat; + } + else { // convert user to device buffer + input = stream->userBuffer; + output = stream->deviceBuffer; + jump_in = stream->nUserChannels[0]; + jump_out = stream->nDeviceChannels[0]; + format_in = stream->userFormat; + format_out = stream->deviceFormat[0]; + + // clear our device buffer when in/out duplex device channels are different + if ( stream->mode == DUPLEX && + stream->nDeviceChannels[0] != stream->nDeviceChannels[1] ) + memset(output, 0, stream->bufferSize * jump_out * formatBytes(format_out)); + } + + channels = (jump_in < jump_out) ? jump_in : jump_out; + + // Set up the interleave/deinterleave offsets + std::vector offset_in(channels); + std::vector offset_out(channels); + if (mode == INPUT && stream->deInterleave[1]) { + for (int k=0; kbufferSize; + offset_out[k] = k; + jump_in = 1; + } + } + else if (mode == OUTPUT && stream->deInterleave[0]) { + for (int k=0; kbufferSize; + jump_out = 1; + } + } + else { + for (int k=0; kbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; j> 16) & 0x0000ffff); + } + in += jump_in; + out += jump_out; + } + } + else if (format_in == RTAUDIO_SINT32) { + INT32 *in = (INT32 *)input; + for (int i=0; ibufferSize; i++) { + for (j=0; j> 16) & 0x0000ffff); + } + in += jump_in; + out += jump_out; + } + } + else if (format_in == RTAUDIO_FLOAT32) { + FLOAT32 *in = (FLOAT32 *)input; + for (int i=0; ibufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; j> 8) & 0x00ff); + } + in += jump_in; + out += jump_out; + } + } + else if (format_in == RTAUDIO_SINT24) { + INT32 *in = (INT32 *)input; + for (int i=0; ibufferSize; i++) { + for (j=0; j> 24) & 0x000000ff); + } + in += jump_in; + out += jump_out; + } + } + else if (format_in == RTAUDIO_SINT32) { + INT32 *in = (INT32 *)input; + for (int i=0; ibufferSize; i++) { + for (j=0; j> 24) & 0x000000ff); + } + in += jump_in; + out += jump_out; + } + } + else if (format_in == RTAUDIO_FLOAT32) { + FLOAT32 *in = (FLOAT32 *)input; + for (int i=0; ibufferSize; i++) { + for (j=0; jbufferSize; i++) { + for (j=0; j