--- /dev/null
+/******************************************/
+/*
+ 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);
+}
projects / rtaudio-cdist.git / commitdiff
