diff options
| author | Gary Scavone <gary@music.mcgill.ca> | 2007-12-06 01:40:05 +0000 |
|---|---|---|
| committer | Stephen Sinclair <sinclair@music.mcgill.ca> | 2013-10-10 01:08:11 +0200 |
| commit | 72ee1e6be2d918af467fef76932231be731795e9 (patch) | |
| tree | cdfed5b0b96a65ebfa407691218ea2ac8d1a64be /RtAudio.cpp | |
Version 2.0
Diffstat (limited to 'RtAudio.cpp')
| -rw-r--r-- | RtAudio.cpp | 4997 |
1 files changed, 4997 insertions, 0 deletions
diff --git a/RtAudio.cpp b/RtAudio.cpp new file mode 100644 index 0000000..d2f8724 --- /dev/null +++ b/RtAudio.cpp @@ -0,0 +1,4997 @@ +/******************************************/ +/* + RtAudio - realtime sound I/O C++ class + Version 2.0 by Gary P. Scavone, 2001-2002. +*/ +/******************************************/ + +#include "RtAudio.h" +#include <vector> +#include <stdio.h> + +// 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; +#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(RtAudioError::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(RtAudioError::NO_DEVICES_FOUND); + } + + try { + *streamID = openStream(outputDevice, outputChannels, inputDevice, inputChannels, + format, sampleRate, bufferSize, numberOfBuffers); + } + catch (RtAudioError &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(RtAudioError::INVALID_PARAMETER); + } + + if ( formatBytes(format) == 0 ) { + sprintf(message,"RtAudio: 'format' parameter value is undefined."); + error(RtAudioError::INVALID_PARAMETER); + } + + if ( outputChannels > 0 ) { + if (outputDevice >= nDevices || outputDevice < 0) { + sprintf(message,"RtAudio: 'outputDevice' parameter value (%d) is invalid.", outputDevice); + error(RtAudioError::INVALID_PARAMETER); + } + } + + if ( inputChannels > 0 ) { + if (inputDevice >= nDevices || inputDevice < 0) { + sprintf(message,"RtAudio: 'inputDevice' parameter value (%d) is invalid.", inputDevice); + error(RtAudioError::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(RtAudioError::MEMORY_ERROR); + } + streams[++streamKey] = (void *) stream; + stream->mode = UNINITIALIZED; + + 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<nDevices; i++) { + if (devices[i].probed == false) { + // If the device wasn't successfully probed before, try it + // again now. + clearDeviceInfo(&devices[i]); + probeDeviceInfo(&devices[i]); + if (devices[i].probed == false) + continue; + } + result = probeDeviceOpen(i, stream, mode, channels, sampleRate, + format, bufferSize, numberOfBuffers); + if (result == SUCCESS) + break; + } + } + else { + result = probeDeviceOpen(device, stream, mode, channels, sampleRate, + format, bufferSize, numberOfBuffers); + } + } + + if ( inputChannels > 0 && result == SUCCESS ) { + + device = inputDevice; + mode = RECORD; + channels = inputChannels; + + if (device == 0) { // Try default device first. + for (int i=0; i<nDevices; i++) { + if (devices[i].probed == false) { + // If the device wasn't successfully probed before, try it + // again now. + clearDeviceInfo(&devices[i]); + probeDeviceInfo(&devices[i]); + if (devices[i].probed == false) + continue; + } + result = probeDeviceOpen(i, stream, mode, channels, sampleRate, + format, bufferSize, numberOfBuffers); + if (result == SUCCESS) + break; + } + } + else { + result = probeDeviceOpen(device, stream, mode, channels, sampleRate, + format, bufferSize, numberOfBuffers); + } + } + + if ( result == SUCCESS ) { + MUTEX_INITIALIZE(&stream->mutex); + return streamKey; + } + + // If we get here, all attempted probes failed. Close any opened + // devices and delete the allocated stream. + closeStream(streamKey); + sprintf(message,"RtAudio: no devices found for given parameters."); + error(RtAudioError::INVALID_PARAMETER); + + return -1; +} + +int RtAudio :: getDeviceCount(void) +{ + return nDevices; +} + +void RtAudio :: getDeviceInfo(int device, RTAUDIO_DEVICE *info) +{ + if (device >= nDevices || device < 0) { + sprintf(message, "RtAudio: invalid device specifier (%d)!", device); + error(RtAudioError::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; i<info->nSampleRates; 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; + +#if defined(__WINDOWS_DS_) + extern "C" unsigned __stdcall callbackHandler(void *ptr); +#else + extern "C" void *callbackHandler(void *ptr); +#endif + +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(RtAudioError::THREAD_ERROR); + } +} + +// *************************************************** // +// +// OS/API-specific methods. +// +// *************************************************** // + +#if defined(__LINUX_ALSA_) + +void RtAudio :: initialize(void) +{ + int card, err, device; + int devices_per_card[32] = {0}; + char name[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); + err = snd_ctl_open(&handle, name, 0); + if (err < 0) { + sprintf(message, "RtAudio: ALSA control open (%i): %s.", card, snd_strerror(err)); + error(RtAudioError::WARNING); + goto next_card; + } + err = snd_ctl_card_info(handle, info); + if (err < 0) { + sprintf(message, "RtAudio: ALSA control hardware info (%i): %s.", card, snd_strerror(err)); + error(RtAudioError::WARNING); + goto next_card; + } + device = -1; + while (1) { + err = snd_ctl_pcm_next_device(handle, &device); + if (err < 0) { + sprintf(message, "RtAudio: ALSA control next device (%i): %s.", card, snd_strerror(err)); + error(RtAudioError::WARNING); + break; + } + if (device < 0) + break; + nDevices++; + devices_per_card[card]++; + } + + 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(RtAudioError::MEMORY_ERROR); + } + + // Write device ascii identifiers to device structures and then + // probe the device capabilities. + card = 0; + device = 0; + for (int i=0; i<nDevices; i++) { + if (devices_per_card[card]) + sprintf(devices[i].name, "hw:%d,%d", card, device); + if (devices_per_card[card] <= device+1) { + card++; + device = 0; + } + else + device++; + probeDeviceInfo(&devices[i]); + } + + return; +} + +void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info) +{ + int err; + int open_mode = SND_PCM_ASYNC; + snd_pcm_t *handle; + snd_pcm_stream_t stream; + + // First try for playback + stream = SND_PCM_STREAM_PLAYBACK; + err = snd_pcm_open(&handle, info->name, stream, open_mode); + if (err < 0) { + sprintf(message, "RtAudio: ALSA pcm playback open (%s): %s.", + info->name, snd_strerror(err)); + error(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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 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) + 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(RtAudioError::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(RtAudioError::WARNING); + return; + } + + // Test a non-standard sample rate to see if continuous rate is supported. + int dir = 0; + if (snd_pcm_hw_params_test_rate(handle, params, 35500, dir) == 0) { + // It appears that continuous sample rate support is available. + info->nSampleRates = -1; + info->sampleRates[0] = snd_pcm_hw_params_get_rate_min(params, &dir); + info->sampleRates[1] = snd_pcm_hw_params_get_rate_max(params, &dir); + } + else { + // No continuous rate support ... test our discrete set of sample rate values. + info->nSampleRates = 0; + for (int i=0; i<MAX_SAMPLE_RATES; i++) { + if (snd_pcm_hw_params_test_rate(handle, params, SAMPLE_RATES[i], dir) == 0) { + info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i]; + info->nSampleRates++; + } + } + 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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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; i<channels; i++) + bufs[i] = (void *) (buffer + (i * offset)); + err = snd_pcm_writen(stream->handle[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(RtAudioError::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(RtAudioError::DRIVER_ERROR); + } + } + else { + sprintf(message, "RtAudio: ALSA error, current state is %s.", + snd_pcm_state_name(state)); + MUTEX_UNLOCK(&stream->mutex); + error(RtAudioError::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(RtAudioError::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; i<channels; i++) + bufs[i] = (void *) (buffer + (i * offset)); + err = snd_pcm_readn(stream->handle[1], bufs, stream->bufferSize); + } + else + err = snd_pcm_readi(stream->handle[1], buffer, stream->bufferSize); + + 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(RtAudioError::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(RtAudioError::DRIVER_ERROR); + } + } + else { + sprintf(message, "RtAudio: ALSA error, current state is %s.", + snd_pcm_state_name(state)); + MUTEX_UNLOCK(&stream->mutex); + error(RtAudioError::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(RtAudioError::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 (RtAudioError &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 <sys/stat.h> +#include <sys/types.h> +#include <sys/ioctl.h> +#include <unistd.h> +#include <fcntl.h> +#include <sys/soundcard.h> +#include <errno.h> +#include <math.h> + +#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(RtAudioError::SYSTEM_ERROR); + } + } + } + else { + sprintf(message, "RtAudio: cannot stat %s.", DAC_NAME); + error(RtAudioError::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<MAX_DEVICES; i++) { + + // Probe /dev/dsp first, since it is supposed to be the default device. + if (i == -1) + sprintf(device_name, "%s", DAC_NAME); + else if (i == dsplink) + continue; // We've aready probed this device via /dev/dsp link ... try next device. + else + sprintf(device_name, "%s%d", DAC_NAME, i); + + // First try to open the device for playback, then record mode. + fd = open(device_name, O_WRONLY | O_NONBLOCK); + if (fd == -1) { + // Open device for playback failed ... either busy or doesn't exist. + if (errno != EBUSY && errno != EAGAIN) { + // Try to open for capture + fd = open(device_name, O_RDONLY | O_NONBLOCK); + if (fd == -1) { + // Open device for record failed. + if (errno != EBUSY && errno != EAGAIN) + continue; + else { + sprintf(message, "RtAudio: OSS record device (%s) is busy.", device_name); + error(RtAudioError::WARNING); + // still count it for now + } + } + } + else { + sprintf(message, "RtAudio: OSS playback device (%s) is busy.", device_name); + error(RtAudioError::WARNING); + // still count it for now + } + } + + if (fd >= 0) close(fd); + strncpy(names[nDevices], device_name, 16); + nDevices++; + } + + if (nDevices == 0) return; + + // Allocate the DEVICE_CONTROL structures. + devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); + if (devices == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtAudioError::MEMORY_ERROR); + } + + // Write device ascii identifiers to device control structure and then probe capabilities. + for (i=0; i<nDevices; i++) { + strncpy(devices[i].name, names[i], 16); + probeDeviceInfo(&devices[i]); + } + + return; +} + +void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info) +{ + int i, fd, channels, mask; + + // The OSS API doesn't provide a means for probing the capabilities + // of devices. Thus, we'll just pursue a brute force method. + + // First try for playback + fd = open(info->name, 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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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; i<MAX_SAMPLE_RATES; i++) { + speed = SAMPLE_RATES[i]; + if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) != -1) { + info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i]; + info->nSampleRates++; + } + } + if (info->nSampleRates == 0) { + close(fd); + return; + } + goto finished; + } + 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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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, 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(RtAudioError::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(RtAudioError::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 (RtAudioError &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 <dsound.h> + +// 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(RtAudioError::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(RtAudioError::DRIVER_ERROR); + } + + count = ins + outs; + if (count == 0) return; + + std::vector<enum_info> info(count); + for (i=0; i<count; i++) { + info[i].name[0] = '\0'; + if (i < outs) info[i].isInput = false; + else info[i].isInput = true; + } + + // Get playback device info and check capabilities. + result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceInfoCallback, &info[0]); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to enumerate through sound playback devices: %s.", + getErrorString(result)); + error(RtAudioError::DRIVER_ERROR); + } + + // Get capture device info and check capabilities. + result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceInfoCallback, &info[0]); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to enumerate through sound capture devices: %s.", + getErrorString(result)); + error(RtAudioError::DRIVER_ERROR); + } + + // Parse the devices and check validity. Devices are considered + // invalid if they cannot be opened, they report no supported data + // formats, or they report < 1 supported channels. + for (i=0; i<count; i++) { + if (info[i].isValid && info[i].id == NULL ) // default device + nDevices++; + } + + // We group the default input and output devices together (as one + // device) . + if (nDevices > 0) { + nDevices = 1; + index = 1; + } + + // Non-default devices are listed separately. + for (i=0; i<count; i++) { + if (info[i].isValid && info[i].id != NULL ) + 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(RtAudioError::MEMORY_ERROR); + } + + // Initialize the GUIDs to NULL for later validation. + for (i=0; i<nDevices; i++) { + devices[i].id[0] = NULL; + devices[i].id[1] = NULL; + } + + // Rename the default device(s). + if (index) + strcpy(devices[0].name, "Default Input/Output Devices"); + + // Copy the names and GUIDs to our devices structures. + for (i=0; i<count; i++) { + if (info[i].isValid && info[i].id != NULL ) { + strncpy(devices[index].name, info[i].name, 64); + if (info[i].isInput) + devices[index].id[1] = info[i].id; + else + devices[index].id[0] = info[i].id; + index++; + } + } + + for (i=0;i<nDevices; i++) + probeDeviceInfo(&devices[i]); + + return; +} + +void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info) +{ + HRESULT result; + + // Get the device index so that we can check the device handle. + int index; + for (index=0; index<nDevices; index++) + if ( info == &devices[index] ) break; + + if ( index >= nDevices ) { + sprintf(message, "RtAudio: device (%s) indexing error in DirectSound probeDeviceInfo().", + info->name); + error(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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; i<info->nSampleRates; 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(RtAudioError::WARNING); + return FAILURE; + } + + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + + if ( mode == PLAYBACK ) { + + 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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::WARNING); + return FAILURE; + } + + stream->handle[0].object = (void *) object; + stream->handle[0].buffer = (void *) buffer; + stream->nDeviceChannels[0] = channels; + } + + if ( mode == RECORD ) { + + 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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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; i<stream->nBuffers; i++) { + + // Find out where the read and "safe write" pointers are. + result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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 frames = 0; + int channels = 1; + if (stream->state == STREAM_STOPPED) + goto unlock; + + HRESULT result; + DWORD currentPos, safePos; + 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(RtAudioError::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(RtAudioError::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) + goto unlock; + + HRESULT result; + DWORD currentPos, safePos; + LPVOID buffer1, buffer2; + DWORD bufferSize1, bufferSize2; + 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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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); + } + + unlock: + 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 (RtAudioError &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 <unistd.h> +#include <errno.h> + +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(RtAudioError::DRIVER_ERROR); + } + + if (nDevices <= 0) return; + + ALvalue *vls = (ALvalue *) new ALvalue[nDevices]; + + // Add one for our default input/output devices. + nDevices++; + + // Allocate the DEVICE_CONTROL structures. + devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); + if (devices == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtAudioError::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(RtAudioError::DRIVER_ERROR); + } + + for (i=0; i<outs; i++) { + if (alGetParams(vls[i].i, pvs, 1) < 0) { + sprintf(message, "RtAudio: AL error querying output devices: %s.", + alGetErrorString(oserror())); + error(RtAudioError::DRIVER_ERROR); + } + strncpy(devices[i+1].name, name, 32); + devices[i+1].id[0] = vls[i].i; + } + + ins = alQueryValues(AL_SYSTEM, AL_DEFAULT_INPUT, &vls[outs], nDevices-outs-1, 0, 0); + if (ins < 0) { + sprintf(message, "RtAudio: AL error getting input devices: %s.", + alGetErrorString(oserror())); + error(RtAudioError::DRIVER_ERROR); + } + + for (i=outs; i<ins+outs; i++) { + if (alGetParams(vls[i].i, pvs, 1) < 0) { + sprintf(message, "RtAudio: AL error querying input devices: %s.", + alGetErrorString(oserror())); + error(RtAudioError::DRIVER_ERROR); + } + strncpy(devices[i+1].name, name, 32); + devices[i+1].id[1] = vls[i].i; + } + + delete [] vls; + + return; +} + +void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info) +{ + int resource, result, i; + ALvalue value; + ALparamInfo pinfo; + + // Get output resource ID if it exists. + if ( !strncmp(info->name, "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(RtAudioError::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(RtAudioError::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(RtAudioError::WARNING); + } + else { + info->nSampleRates = 0; + for (i=0; i<MAX_SAMPLE_RATES; i++) { + if ( SAMPLE_RATES[i] >= pinfo.min.i && SAMPLE_RATES[i] <= pinfo.max.i ) { + info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i]; + info->nSampleRates++; + } + } + } + + // 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(RtAudioError::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(RtAudioError::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(RtAudioError::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<MAX_SAMPLE_RATES; i++) { + if ( SAMPLE_RATES[i] >= pinfo.min.i && SAMPLE_RATES[i] <= pinfo.max.i ) { + info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i]; + info->nSampleRates++; + } + } + } + + // 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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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(RtAudioError::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 (RtAudioError &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(RtAudioError::TYPE type) +{ + if (type == RtAudioError::WARNING) + fprintf(stderr, "\n%s\n\n", message); + else if (type == RtAudioError::DEBUG_WARNING) { +#if defined(RTAUDIO_DEBUG) + fprintf(stderr, "\n%s\n\n", message); +#endif + } + else + throw RtAudioError(message, type); +} + +void *RtAudio :: verifyStream(int streamID) +{ + // Verify the stream key. + if ( streams.find( streamID ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtAudioError::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; i<MAX_SAMPLE_RATES; i++) + info->sampleRates[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(RtAudioError::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<int> offset_in(channels); + std::vector<int> offset_out(channels); + if (mode == RECORD && stream->deInterleave[1]) { + for (int k=0; k<channels; k++) { + offset_in[k] = k * stream->bufferSize; + offset_out[k] = k; + } + } + else if (mode == PLAYBACK && stream->deInterleave[0]) { + for (int k=0; k<channels; k++) { + offset_in[k] = k; + offset_out[k] = k * stream->bufferSize; + } + } + else { + for (int k=0; k<channels; k++) { + offset_in[k] = k; + offset_out[k] = k; + } + } + + if (format_out == RTAUDIO_FLOAT64) { + FLOAT64 scale; + FLOAT64 *out = (FLOAT64 *)output; + + if (format_in == RTAUDIO_SINT8) { + signed char *in = (signed char *)input; + scale = 1.0 / 128.0; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (FLOAT64) in[offset_in[j]]; + out[offset_out[j]] *= scale; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT16) { + INT16 *in = (INT16 *)input; + scale = 1.0 / 32768.0; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (FLOAT64) in[offset_in[j]]; + out[offset_out[j]] *= scale; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT24) { + INT32 *in = (INT32 *)input; + scale = 1.0 / 2147483648.0; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (FLOAT64) (in[offset_in[j]] & 0xffffff00); + out[offset_out[j]] *= scale; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT32) { + INT32 *in = (INT32 *)input; + scale = 1.0 / 2147483648.0; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (FLOAT64) in[offset_in[j]]; + out[offset_out[j]] *= scale; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_FLOAT32) { + FLOAT32 *in = (FLOAT32 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (FLOAT64) in[offset_in[j]]; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_FLOAT64) { + // Channel compensation and/or (de)interleaving only. + FLOAT64 *in = (FLOAT64 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = in[offset_in[j]]; + } + in += channels_in; + out += channels_out; + } + } + } + else if (format_out == RTAUDIO_FLOAT32) { + FLOAT32 scale; + FLOAT32 *out = (FLOAT32 *)output; + + if (format_in == RTAUDIO_SINT8) { + signed char *in = (signed char *)input; + scale = 1.0 / 128.0; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (FLOAT32) in[offset_in[j]]; + out[offset_out[j]] *= scale; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT16) { + INT16 *in = (INT16 *)input; + scale = 1.0 / 32768.0; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (FLOAT32) in[offset_in[j]]; + out[offset_out[j]] *= scale; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT24) { + INT32 *in = (INT32 *)input; + scale = 1.0 / 2147483648.0; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (FLOAT32) (in[offset_in[j]] & 0xffffff00); + out[offset_out[j]] *= scale; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT32) { + INT32 *in = (INT32 *)input; + scale = 1.0 / 2147483648.0; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (FLOAT32) in[offset_in[j]]; + out[offset_out[j]] *= scale; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_FLOAT32) { + // Channel compensation and/or (de)interleaving only. + FLOAT32 *in = (FLOAT32 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = in[offset_in[j]]; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_FLOAT64) { + FLOAT64 *in = (FLOAT64 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (FLOAT32) in[offset_in[j]]; + } + in += channels_in; + out += channels_out; + } + } + } + else if (format_out == RTAUDIO_SINT32) { + INT32 *out = (INT32 *)output; + if (format_in == RTAUDIO_SINT8) { + signed char *in = (signed char *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT32) in[offset_in[j]]; + out[offset_out[j]] <<= 24; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT16) { + INT16 *in = (INT16 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT32) in[offset_in[j]]; + out[offset_out[j]] <<= 16; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT24) { + INT32 *in = (INT32 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT32) in[offset_in[j]]; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT32) { + // Channel compensation and/or (de)interleaving only. + INT32 *in = (INT32 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = in[offset_in[j]]; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_FLOAT32) { + FLOAT32 *in = (FLOAT32 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT32) (in[offset_in[j]] * 2147483647.0); + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_FLOAT64) { + FLOAT64 *in = (FLOAT64 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT32) (in[offset_in[j]] * 2147483647.0); + } + in += channels_in; + out += channels_out; + } + } + } + else if (format_out == RTAUDIO_SINT24) { + INT32 *out = (INT32 *)output; + if (format_in == RTAUDIO_SINT8) { + signed char *in = (signed char *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT32) in[offset_in[j]]; + out[offset_out[j]] <<= 24; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT16) { + INT16 *in = (INT16 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT32) in[offset_in[j]]; + out[offset_out[j]] <<= 16; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT24) { + // Channel compensation and/or (de)interleaving only. + INT32 *in = (INT32 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = in[offset_in[j]]; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT32) { + INT32 *in = (INT32 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT32) (in[offset_in[j]] & 0xffffff00); + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_FLOAT32) { + FLOAT32 *in = (FLOAT32 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT32) (in[offset_in[j]] * 2147483647.0); + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_FLOAT64) { + FLOAT64 *in = (FLOAT64 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT32) (in[offset_in[j]] * 2147483647.0); + } + in += channels_in; + out += channels_out; + } + } + } + else if (format_out == RTAUDIO_SINT16) { + INT16 *out = (INT16 *)output; + if (format_in == RTAUDIO_SINT8) { + signed char *in = (signed char *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT16) in[offset_in[j]]; + out[offset_out[j]] <<= 8; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT16) { + // Channel compensation and/or (de)interleaving only. + INT16 *in = (INT16 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = in[offset_in[j]]; + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT24) { + INT32 *in = (INT32 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT16) ((in[offset_in[j]] >> 16) & 0x0000ffff); + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT32) { + INT32 *in = (INT32 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT16) ((in[offset_in[j]] >> 16) & 0x0000ffff); + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_FLOAT32) { + FLOAT32 *in = (FLOAT32 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT16) (in[offset_in[j]] * 32767.0); + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_FLOAT64) { + FLOAT64 *in = (FLOAT64 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (INT16) (in[offset_in[j]] * 32767.0); + } + in += channels_in; + out += channels_out; + } + } + } + else if (format_out == RTAUDIO_SINT8) { + signed char *out = (signed char *)output; + if (format_in == RTAUDIO_SINT8) { + // Channel compensation and/or (de)interleaving only. + signed char *in = (signed char *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = in[offset_in[j]]; + } + in += channels_in; + out += channels_out; + } + } + if (format_in == RTAUDIO_SINT16) { + INT16 *in = (INT16 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (signed char) ((in[offset_in[j]] >> 8) & 0x00ff); + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT24) { + INT32 *in = (INT32 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (signed char) ((in[offset_in[j]] >> 24) & 0x000000ff); + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_SINT32) { + INT32 *in = (INT32 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (signed char) ((in[offset_in[j]] >> 24) & 0x000000ff); + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_FLOAT32) { + FLOAT32 *in = (FLOAT32 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (signed char) (in[offset_in[j]] * 127.0); + } + in += channels_in; + out += channels_out; + } + } + else if (format_in == RTAUDIO_FLOAT64) { + FLOAT64 *in = (FLOAT64 *)input; + for (int i=0; i<stream->bufferSize; i++) { + for (j=0; j<channels; j++) { + out[offset_out[j]] = (signed char) (in[offset_in[j]] * 127.0); + } + in += channels_in; + out += channels_out; + } + } + } +} + +void RtAudio :: byteSwapBuffer(char *buffer, int samples, RTAUDIO_FORMAT format) +{ + register char val; + register char *ptr; + + ptr = buffer; + if (format == RTAUDIO_SINT16) { + for (int i=0; i<samples; i++) { + // Swap 1st and 2nd bytes. + val = *(ptr); + *(ptr) = *(ptr+1); + *(ptr+1) = val; + + // Increment 2 bytes. + ptr += 2; + } + } + else if (format == RTAUDIO_SINT24 || + format == RTAUDIO_SINT32 || + format == RTAUDIO_FLOAT32) { + for (int i=0; i<samples; i++) { + // Swap 1st and 4th bytes. + val = *(ptr); + *(ptr) = *(ptr+3); + *(ptr+3) = val; + + // Swap 2nd and 3rd bytes. + ptr += 1; + val = *(ptr); + *(ptr) = *(ptr+1); + *(ptr+1) = val; + + // Increment 4 bytes. + ptr += 4; + } + } + else if (format == RTAUDIO_FLOAT64) { + for (int i=0; i<samples; i++) { + // Swap 1st and 8th bytes + val = *(ptr); + *(ptr) = *(ptr+7); + *(ptr+7) = val; + + // Swap 2nd and 7th bytes + ptr += 1; + val = *(ptr); + *(ptr) = *(ptr+5); + *(ptr+5) = val; + + // Swap 3rd and 6th bytes + ptr += 1; + val = *(ptr); + *(ptr) = *(ptr+3); + *(ptr+3) = val; + + // Swap 4th and 5th bytes + ptr += 1; + val = *(ptr); + *(ptr) = *(ptr+1); + *(ptr+1) = val; + + // Increment 8 bytes. + ptr += 8; + } + } +} + + +// *************************************************** // +// +// RtAudioError class definition. +// +// *************************************************** // + +RtAudioError :: RtAudioError(const char *p, TYPE tipe) +{ + type = tipe; + strncpy(error_message, p, 256); +} + +RtAudioError :: ~RtAudioError() +{ +} + +void RtAudioError :: printMessage() +{ + printf("\n%s\n\n", error_message); +} |