diff options
| author | Gary Scavone <gary@music.mcgill.ca> | 2013-10-09 23:44:33 +0200 |
|---|---|---|
| committer | Stephen Sinclair <sinclair@music.mcgill.ca> | 2013-10-10 01:08:39 +0200 |
| commit | a3d2ee35944db4dd0a3a342bb7f2df69f229f45d (patch) | |
| tree | 1a1199a30b0db8a0306dceef0c15c9f9e3e72629 /RtAudio.cpp | |
| parent | 8eb71e693530726068addf6b8088aea0fd340f2a (diff) | |
Version 2.1
Diffstat (limited to 'RtAudio.cpp')
| -rw-r--r-- | RtAudio.cpp | 12083 |
1 files changed, 7082 insertions, 5001 deletions
diff --git a/RtAudio.cpp b/RtAudio.cpp index fd116da..4a51baf 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -1,5001 +1,7082 @@ -/******************************************/
-/*
- RtAudio - realtime sound I/O C++ class
- by Gary P. Scavone, 2001-2002.
-*/
-/******************************************/
-
-#include "RtAudio.h"
-#include <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;
- typedef unsigned (__stdcall THREAD_FUNCTION)(void *);
-#else // pthread API
- #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
- #define MUTEX_LOCK(A) pthread_mutex_lock(A)
- #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)
- typedef void * THREAD_RETURN;
-#endif
-
-// *************************************************** //
-//
-// Public common (OS-independent) methods.
-//
-// *************************************************** //
-
-RtAudio :: RtAudio()
-{
- initialize();
-
- if (nDevices <= 0) {
- sprintf(message, "RtAudio: no audio devices found!");
- error(RtError::NO_DEVICES_FOUND);
- }
-}
-
-RtAudio :: RtAudio(int *streamId,
- int outputDevice, int outputChannels,
- int inputDevice, int inputChannels,
- RTAUDIO_FORMAT format, int sampleRate,
- int *bufferSize, int numberOfBuffers)
-{
- initialize();
-
- if (nDevices <= 0) {
- sprintf(message, "RtAudio: no audio devices found!");
- error(RtError::NO_DEVICES_FOUND);
- }
-
- try {
- *streamId = openStream(outputDevice, outputChannels, inputDevice, inputChannels,
- format, sampleRate, bufferSize, numberOfBuffers);
- }
- catch (RtError &exception) {
- // deallocate the RTAUDIO_DEVICE structures
- if (devices) free(devices);
- error(exception.getType());
- }
-}
-
-RtAudio :: ~RtAudio()
-{
- // close any existing streams
- while ( streams.size() )
- closeStream( streams.begin()->first );
-
- // deallocate the RTAUDIO_DEVICE structures
- if (devices) free(devices);
-}
-
-int RtAudio :: openStream(int outputDevice, int outputChannels,
- int inputDevice, int inputChannels,
- RTAUDIO_FORMAT format, int sampleRate,
- int *bufferSize, int numberOfBuffers)
-{
- static int streamKey = 0; // Unique stream identifier ... OK for multiple instances.
-
- if (outputChannels < 1 && inputChannels < 1) {
- sprintf(message,"RtAudio: one or both 'channel' parameters must be greater than zero.");
- error(RtError::INVALID_PARAMETER);
- }
-
- if ( formatBytes(format) == 0 ) {
- sprintf(message,"RtAudio: 'format' parameter value is undefined.");
- error(RtError::INVALID_PARAMETER);
- }
-
- if ( outputChannels > 0 ) {
- if (outputDevice >= nDevices || outputDevice < 0) {
- sprintf(message,"RtAudio: 'outputDevice' parameter value (%d) is invalid.", outputDevice);
- error(RtError::INVALID_PARAMETER);
- }
- }
-
- if ( inputChannels > 0 ) {
- if (inputDevice >= nDevices || inputDevice < 0) {
- sprintf(message,"RtAudio: 'inputDevice' parameter value (%d) is invalid.", inputDevice);
- error(RtError::INVALID_PARAMETER);
- }
- }
-
- // Allocate a new stream structure.
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) calloc(1, sizeof(RTAUDIO_STREAM));
- if (stream == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
- streams[++streamKey] = (void *) stream;
- stream->mode = UNINITIALIZED;
- MUTEX_INITIALIZE(&stream->mutex);
-
- bool result = SUCCESS;
- int device;
- STREAM_MODE mode;
- int channels;
- if ( outputChannels > 0 ) {
-
- device = outputDevice;
- mode = PLAYBACK;
- channels = outputChannels;
-
- if (device == 0) { // Try default device first.
- for (int i=0; i<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 )
- 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(RtError::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(RtError::INVALID_DEVICE);
- }
-
- // If the device wasn't successfully probed before, try it again.
- if (devices[device].probed == false) {
- clearDeviceInfo(&devices[device]);
- probeDeviceInfo(&devices[device]);
- }
-
- // Clear the info structure.
- memset(info, 0, sizeof(RTAUDIO_DEVICE));
-
- strncpy(info->name, devices[device].name, 128);
- info->probed = devices[device].probed;
- if ( info->probed == true ) {
- info->maxOutputChannels = devices[device].maxOutputChannels;
- info->maxInputChannels = devices[device].maxInputChannels;
- info->maxDuplexChannels = devices[device].maxDuplexChannels;
- info->minOutputChannels = devices[device].minOutputChannels;
- info->minInputChannels = devices[device].minInputChannels;
- info->minDuplexChannels = devices[device].minDuplexChannels;
- info->hasDuplexSupport = devices[device].hasDuplexSupport;
- info->nSampleRates = devices[device].nSampleRates;
- if (info->nSampleRates == -1) {
- info->sampleRates[0] = devices[device].sampleRates[0];
- info->sampleRates[1] = devices[device].sampleRates[1];
- }
- else {
- for (int i=0; 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;
-
-extern "C" THREAD_RETURN THREAD_TYPE callbackHandler(void * ptr);
-
-void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- stream->callback = callback;
- stream->userData = userData;
- stream->usingCallback = true;
- thread_info.object = this;
- thread_info.streamId = streamId;
-
- int err = 0;
-#if defined(__WINDOWS_DS__)
- unsigned thread_id;
- stream->thread = _beginthreadex(NULL, 0, &callbackHandler,
- &stream->usingCallback, 0, &thread_id);
- if (stream->thread == 0) err = -1;
- // When spawning multiple threads in quick succession, it appears to be
- // necessary to wait a bit for each to initialize ... another windism!
- Sleep(1);
-#else
- err = pthread_create(&stream->thread, NULL, callbackHandler, &stream->usingCallback);
-#endif
-
- if (err) {
- stream->usingCallback = false;
- sprintf(message, "RtAudio: error starting callback thread!");
- error(RtError::THREAD_ERROR);
- }
-}
-
-// *************************************************** //
-//
-// OS/API-specific methods.
-//
-// *************************************************** //
-
-#if defined(__LINUX_ALSA__)
-
-#define MAX_DEVICES 16
-
-void RtAudio :: initialize(void)
-{
- int card, result, device;
- char name[32];
- char deviceNames[MAX_DEVICES][32];
- snd_ctl_t *handle;
- snd_ctl_card_info_t *info;
- snd_ctl_card_info_alloca(&info);
-
- // Count cards and devices
- nDevices = 0;
- card = -1;
- snd_card_next(&card);
- while ( card >= 0 ) {
- sprintf(name, "hw:%d", card);
- result = snd_ctl_open(&handle, name, 0);
- if (result < 0) {
- sprintf(message, "RtAudio: ALSA control open (%i): %s.", card, snd_strerror(result));
- error(RtError::WARNING);
- goto next_card;
- }
- result = snd_ctl_card_info(handle, info);
- if (result < 0) {
- sprintf(message, "RtAudio: ALSA control hardware info (%i): %s.", card, snd_strerror(result));
- error(RtError::WARNING);
- goto next_card;
- }
- device = -1;
- while (1) {
- result = snd_ctl_pcm_next_device(handle, &device);
- if (result < 0) {
- sprintf(message, "RtAudio: ALSA control next device (%i): %s.", card, snd_strerror(result));
- error(RtError::WARNING);
- break;
- }
- if (device < 0)
- break;
- sprintf( deviceNames[nDevices++], "hw:%d,%d", card, device );
- if ( nDevices > MAX_DEVICES ) break;
- }
- if ( nDevices > MAX_DEVICES ) break;
- next_card:
- snd_ctl_close(handle);
- snd_card_next(&card);
- }
-
- if (nDevices == 0) return;
-
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
-
- // Write device ascii identifiers to device structures and then
- // probe the device capabilities.
- for (int i=0; i<nDevices; i++) {
- strncpy(devices[i].name, deviceNames[i], 32);
- 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(RtError::WARNING);
- goto capture_probe;
- }
-
- snd_pcm_hw_params_t *params;
- snd_pcm_hw_params_alloca(¶ms);
-
- // We have an open device ... allocate the parameter structure.
- err = snd_pcm_hw_params_any(handle, params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA hardware probe error (%s): %s.",
- info->name, snd_strerror(err));
- error(RtError::WARNING);
- goto capture_probe;
- }
-
- // Get output channel information.
- info->minOutputChannels = snd_pcm_hw_params_get_channels_min(params);
- info->maxOutputChannels = snd_pcm_hw_params_get_channels_max(params);
-
- snd_pcm_close(handle);
-
- capture_probe:
- // Now try for capture
- stream = SND_PCM_STREAM_CAPTURE;
- err = snd_pcm_open(&handle, info->name, stream, open_mode);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA pcm capture open (%s): %s.",
- info->name, snd_strerror(err));
- error(RtError::WARNING);
- if (info->maxOutputChannels == 0)
- // didn't open for playback either ... device invalid
- return;
- goto probe_parameters;
- }
-
- // We have an open capture device ... allocate the parameter structure.
- err = snd_pcm_hw_params_any(handle, params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA hardware probe error (%s): %s.",
- info->name, snd_strerror(err));
- error(RtError::WARNING);
- if (info->maxOutputChannels > 0)
- goto probe_parameters;
- else
- return;
- }
-
- // Get input channel information.
- info->minInputChannels = snd_pcm_hw_params_get_channels_min(params);
- info->maxInputChannels = snd_pcm_hw_params_get_channels_max(params);
-
- // If device opens for both playback and capture, we determine the channels.
- if (info->maxOutputChannels == 0 || info->maxInputChannels == 0)
- goto probe_parameters;
-
- info->hasDuplexSupport = true;
- info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ?
- info->maxInputChannels : info->maxOutputChannels;
- info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ?
- info->minInputChannels : info->minOutputChannels;
-
- snd_pcm_close(handle);
-
- probe_parameters:
- // At this point, we just need to figure out the supported data
- // formats and sample rates. We'll proceed by opening the device in
- // the direction with the maximum number of channels, or playback if
- // they are equal. This might limit our sample rate options, but so
- // be it.
-
- if (info->maxOutputChannels >= info->maxInputChannels)
- stream = SND_PCM_STREAM_PLAYBACK;
- else
- stream = SND_PCM_STREAM_CAPTURE;
-
- err = snd_pcm_open(&handle, info->name, stream, open_mode);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA pcm (%s) won't reopen during probe: %s.",
- info->name, snd_strerror(err));
- error(RtError::WARNING);
- return;
- }
-
- // We have an open device ... allocate the parameter structure.
- err = snd_pcm_hw_params_any(handle, params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA hardware reopen probe error (%s): %s.",
- info->name, snd_strerror(err));
- error(RtError::WARNING);
- return;
- }
-
- // Test a non-standard sample rate to see if continuous rate is supported.
- int dir = 0;
- if (snd_pcm_hw_params_test_rate(handle, params, 35500, dir) == 0) {
- // It appears that continuous sample rate support is available.
- info->nSampleRates = -1;
- info->sampleRates[0] = snd_pcm_hw_params_get_rate_min(params, &dir);
- info->sampleRates[1] = snd_pcm_hw_params_get_rate_max(params, &dir);
- }
- else {
- // No continuous rate support ... test our discrete set of sample rate values.
- info->nSampleRates = 0;
- for (int i=0; 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(RtError::WARNING);
- return;
- }
-
- // That's all ... close the device and return
- snd_pcm_close(handle);
- info->probed = true;
- return;
-}
-
-bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
- STREAM_MODE mode, int channels,
- int sampleRate, RTAUDIO_FORMAT format,
- int *bufferSize, int numberOfBuffers)
-{
-#if defined(RTAUDIO_DEBUG)
- snd_output_t *out;
- snd_output_stdio_attach(&out, stderr, 0);
-#endif
-
- // I'm not using the "plug" interface ... too much inconsistent behavior.
- const char *name = devices[device].name;
-
- snd_pcm_stream_t alsa_stream;
- if (mode == PLAYBACK)
- alsa_stream = SND_PCM_STREAM_PLAYBACK;
- else
- alsa_stream = SND_PCM_STREAM_CAPTURE;
-
- int err;
- snd_pcm_t *handle;
- int alsa_open_mode = SND_PCM_ASYNC;
- err = snd_pcm_open(&handle, name, alsa_stream, alsa_open_mode);
- if (err < 0) {
- sprintf(message,"RtAudio: ALSA pcm device (%s) won't open: %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Fill the parameter structure.
- snd_pcm_hw_params_t *hw_params;
- snd_pcm_hw_params_alloca(&hw_params);
- err = snd_pcm_hw_params_any(handle, hw_params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error getting parameter handle (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
-
-#if defined(RTAUDIO_DEBUG)
- fprintf(stderr, "\nRtAudio: ALSA dump hardware params just after device open:\n\n");
- snd_pcm_hw_params_dump(hw_params, out);
-#endif
-
- // Set access ... try interleaved access first, then non-interleaved
- err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
- if (err < 0) {
- // No interleave support ... try non-interleave.
- err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting access ( (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
- stream->deInterleave[mode] = true;
- }
-
- // Determine how to set the device format.
- stream->userFormat = format;
- snd_pcm_format_t device_format;
-
- if (format == RTAUDIO_SINT8)
- device_format = SND_PCM_FORMAT_S8;
- else if (format == RTAUDIO_SINT16)
- device_format = SND_PCM_FORMAT_S16;
- else if (format == RTAUDIO_SINT24)
- device_format = SND_PCM_FORMAT_S24;
- else if (format == RTAUDIO_SINT32)
- device_format = SND_PCM_FORMAT_S32;
- else if (format == RTAUDIO_FLOAT32)
- device_format = SND_PCM_FORMAT_FLOAT;
- else if (format == RTAUDIO_FLOAT64)
- device_format = SND_PCM_FORMAT_FLOAT64;
-
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = format;
- goto set_format;
- }
-
- // The user requested format is not natively supported by the device.
- device_format = SND_PCM_FORMAT_FLOAT64;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_FLOAT64;
- goto set_format;
- }
-
- device_format = SND_PCM_FORMAT_FLOAT;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_FLOAT32;
- goto set_format;
- }
-
- device_format = SND_PCM_FORMAT_S32;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- goto set_format;
- }
-
- device_format = SND_PCM_FORMAT_S24;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_SINT24;
- goto set_format;
- }
-
- device_format = SND_PCM_FORMAT_S16;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- goto set_format;
- }
-
- device_format = SND_PCM_FORMAT_S8;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_SINT8;
- goto set_format;
- }
-
- // If we get here, no supported format was found.
- sprintf(message,"RtAudio: ALSA pcm device (%s) data format not supported by RtAudio.", name);
- snd_pcm_close(handle);
- error(RtError::WARNING);
- return FAILURE;
-
- set_format:
- err = snd_pcm_hw_params_set_format(handle, hw_params, device_format);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting format (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Determine whether byte-swaping is necessary.
- stream->doByteSwap[mode] = false;
- if (device_format != SND_PCM_FORMAT_S8) {
- err = snd_pcm_format_cpu_endian(device_format);
- if (err == 0)
- stream->doByteSwap[mode] = true;
- else if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error getting format endian-ness (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
- }
-
- // Determine the number of channels for this device. We support a possible
- // minimum device channel number > than the value requested by the user.
- stream->nUserChannels[mode] = channels;
- int device_channels = snd_pcm_hw_params_get_channels_max(hw_params);
- if (device_channels < channels) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: channels (%d) not supported by device (%s).",
- channels, name);
- error(RtError::WARNING);
- return FAILURE;
- }
-
- device_channels = snd_pcm_hw_params_get_channels_min(hw_params);
- if (device_channels < channels) device_channels = channels;
- stream->nDeviceChannels[mode] = device_channels;
-
- // Set the device channels.
- err = snd_pcm_hw_params_set_channels(handle, hw_params, device_channels);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting channels (%d) on device (%s): %s.",
- device_channels, name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Set the sample rate.
- err = snd_pcm_hw_params_set_rate(handle, hw_params, (unsigned int)sampleRate, 0);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting sample rate (%d) on device (%s): %s.",
- sampleRate, name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Set the buffer number, which in ALSA is referred to as the "period".
- int dir;
- int periods = numberOfBuffers;
- // Even though the hardware might allow 1 buffer, it won't work reliably.
- if (periods < 2) periods = 2;
- err = snd_pcm_hw_params_get_periods_min(hw_params, &dir);
- if (err > periods) periods = err;
-
- err = snd_pcm_hw_params_set_periods(handle, hw_params, periods, 0);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting periods (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Set the buffer (or period) size.
- err = snd_pcm_hw_params_get_period_size_min(hw_params, &dir);
- if (err > *bufferSize) *bufferSize = err;
-
- err = snd_pcm_hw_params_set_period_size(handle, hw_params, *bufferSize, 0);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting period size (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- stream->bufferSize = *bufferSize;
-
- // Install the hardware configuration
- err = snd_pcm_hw_params(handle, hw_params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error installing hardware configuration (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
-
-#if defined(RTAUDIO_DEBUG)
- fprintf(stderr, "\nRtAudio: ALSA dump hardware params after installation:\n\n");
- snd_pcm_hw_params_dump(hw_params, out);
-#endif
-
- /*
- // Install the software configuration
- snd_pcm_sw_params_t *sw_params = NULL;
- snd_pcm_sw_params_alloca(&sw_params);
- snd_pcm_sw_params_current(handle, sw_params);
- err = snd_pcm_sw_params(handle, sw_params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error installing software configuration (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
- */
-
- // Set handle and flags for buffer conversion
- stream->handle[mode] = handle;
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] > 1 && stream->deInterleave[mode])
- stream->doConvertBuffer[mode] = true;
-
- // Allocate necessary internal buffers
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
-
- long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
- else
- buffer_bytes = stream->nUserChannels[1];
-
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL)
- goto memory_error;
- }
-
- if ( stream->doConvertBuffer[mode] ) {
-
- long buffer_bytes;
- bool makeBuffer = true;
- if ( mode == PLAYBACK )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- else { // mode == RECORD
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == PLAYBACK ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- if ( buffer_bytes > bytes_out )
- buffer_bytes = (buffer_bytes > bytes_out) ? buffer_bytes : bytes_out;
- else
- makeBuffer = false;
- }
- }
-
- if ( makeBuffer ) {
- buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL)
- goto memory_error;
- }
- }
-
- stream->device[mode] = device;
- stream->state = STREAM_STOPPED;
- if ( stream->mode == PLAYBACK && mode == RECORD )
- // We had already set up an output stream.
- stream->mode = DUPLEX;
- else
- stream->mode = mode;
- stream->nBuffers = periods;
- stream->sampleRate = sampleRate;
-
- return SUCCESS;
-
- memory_error:
- if (stream->handle[0]) {
- snd_pcm_close(stream->handle[0]);
- stream->handle[0] = 0;
- }
- if (stream->handle[1]) {
- snd_pcm_close(stream->handle[1]);
- stream->handle[1] = 0;
- }
- if (stream->userBuffer) {
- free(stream->userBuffer);
- stream->userBuffer = 0;
- }
- sprintf(message, "RtAudio: ALSA error allocating buffer memory (%s).", name);
- error(RtError::WARNING);
- return FAILURE;
-}
-
-void RtAudio :: cancelStreamCallback(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- if (stream->usingCallback) {
- stream->usingCallback = false;
- pthread_cancel(stream->thread);
- pthread_join(stream->thread, NULL);
- stream->thread = 0;
- stream->callback = NULL;
- stream->userData = NULL;
- }
-}
-
-void RtAudio :: closeStream(int streamId)
-{
- // We don't want an exception to be thrown here because this
- // function is called by our class destructor. So, do our own
- // streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
- error(RtError::WARNING);
- return;
- }
-
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
-
- if (stream->usingCallback) {
- pthread_cancel(stream->thread);
- pthread_join(stream->thread, NULL);
- }
-
- if (stream->state == STREAM_RUNNING) {
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX)
- snd_pcm_drop(stream->handle[0]);
- if (stream->mode == RECORD || stream->mode == DUPLEX)
- snd_pcm_drop(stream->handle[1]);
- }
-
- pthread_mutex_destroy(&stream->mutex);
-
- if (stream->handle[0])
- snd_pcm_close(stream->handle[0]);
-
- if (stream->handle[1])
- snd_pcm_close(stream->handle[1]);
-
- if (stream->userBuffer)
- free(stream->userBuffer);
-
- if (stream->deviceBuffer)
- free(stream->deviceBuffer);
-
- free(stream);
- streams.erase(streamId);
-}
-
-void RtAudio :: startStream(int streamId)
-{
- // This method calls snd_pcm_prepare if the device isn't already in that state.
-
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_RUNNING)
- goto unlock;
-
- int err;
- snd_pcm_state_t state;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
- state = snd_pcm_state(stream->handle[0]);
- if (state != SND_PCM_STATE_PREPARED) {
- err = snd_pcm_prepare(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error preparing pcm device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- }
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
- state = snd_pcm_state(stream->handle[1]);
- if (state != SND_PCM_STATE_PREPARED) {
- err = snd_pcm_prepare(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error preparing pcm device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- }
- stream->state = STREAM_RUNNING;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-void RtAudio :: stopStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- int err;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
- err = snd_pcm_drain(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
- err = snd_pcm_drain(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- stream->state = STREAM_STOPPED;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-void RtAudio :: abortStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- int err;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
- err = snd_pcm_drop(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
- err = snd_pcm_drop(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- stream->state = STREAM_STOPPED;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-int RtAudio :: streamWillBlock(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- int err = 0, frames = 0;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
- err = snd_pcm_avail_update(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error getting available frames for device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
-
- frames = err;
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
- err = snd_pcm_avail_update(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error getting available frames for device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- if (frames > err) frames = err;
- }
-
- frames = stream->bufferSize - frames;
- if (frames < 0) frames = 0;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
- return frames;
-}
-
-void RtAudio :: tickStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- int stopStream = 0;
- if (stream->state == STREAM_STOPPED) {
- if (stream->usingCallback) usleep(50000); // sleep 50 milliseconds
- return;
- }
- else if (stream->usingCallback) {
- stopStream = stream->callback(stream->userBuffer, stream->bufferSize, stream->userData);
- }
-
- MUTEX_LOCK(&stream->mutex);
-
- // The state might change while waiting on a mutex.
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- int err;
- char *buffer;
- int channels;
- RTAUDIO_FORMAT format;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
-
- // Setup parameters and do buffer conversion if necessary.
- if (stream->doConvertBuffer[0]) {
- convertStreamBuffer(stream, PLAYBACK);
- buffer = stream->deviceBuffer;
- channels = stream->nDeviceChannels[0];
- format = stream->deviceFormat[0];
- }
- else {
- buffer = stream->userBuffer;
- channels = stream->nUserChannels[0];
- format = stream->userFormat;
- }
-
- // Do byte swapping if necessary.
- if (stream->doByteSwap[0])
- byteSwapBuffer(buffer, stream->bufferSize * channels, format);
-
- // Write samples to device in interleaved/non-interleaved format.
- if (stream->deInterleave[0]) {
- void *bufs[channels];
- size_t offset = stream->bufferSize * formatBytes(format);
- for (int i=0; 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(RtError::WARNING);
- err = snd_pcm_prepare(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error preparing handle after underrun: %s.",
- snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- else {
- sprintf(message, "RtAudio: ALSA error, current state is %s.",
- snd_pcm_state_name(state));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- goto unlock;
- }
- else {
- sprintf(message, "RtAudio: ALSA audio write error for device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- }
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
-
- // Setup parameters.
- if (stream->doConvertBuffer[1]) {
- buffer = stream->deviceBuffer;
- channels = stream->nDeviceChannels[1];
- format = stream->deviceFormat[1];
- }
- else {
- buffer = stream->userBuffer;
- channels = stream->nUserChannels[1];
- format = stream->userFormat;
- }
-
- // Read samples from device in interleaved/non-interleaved format.
- if (stream->deInterleave[1]) {
- void *bufs[channels];
- size_t offset = stream->bufferSize * formatBytes(format);
- for (int i=0; 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(RtError::WARNING);
- err = snd_pcm_prepare(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error preparing handle after overrun: %s.",
- snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- else {
- sprintf(message, "RtAudio: ALSA error, current state is %s.",
- snd_pcm_state_name(state));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- goto unlock;
- }
- else {
- sprintf(message, "RtAudio: ALSA audio read error for device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
-
- // Do byte swapping if necessary.
- if (stream->doByteSwap[1])
- byteSwapBuffer(buffer, stream->bufferSize * channels, format);
-
- // Do buffer conversion if necessary.
- if (stream->doConvertBuffer[1])
- convertStreamBuffer(stream, RECORD);
- }
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-
- if (stream->usingCallback && stopStream)
- this->stopStream(streamId);
-}
-
-extern "C" void *callbackHandler(void *ptr)
-{
- RtAudio *object = thread_info.object;
- int stream = thread_info.streamId;
- bool *usingCallback = (bool *) ptr;
-
- while ( *usingCallback ) {
- pthread_testcancel();
- try {
- object->tickStream(stream);
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nCallback thread error (%s) ... closing thread.\n\n",
- exception.getMessage());
- break;
- }
- }
-
- return 0;
-}
-
-//******************** End of __LINUX_ALSA__ *********************//
-
-#elif defined(__LINUX_OSS__)
-
-#include <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(RtError::SYSTEM_ERROR);
- }
- }
- }
- else {
- sprintf(message, "RtAudio: cannot stat %s.", DAC_NAME);
- error(RtError::SYSTEM_ERROR);
- }
-
- // The OSS API doesn't provide a routine for determining the number
- // of devices. Thus, we'll just pursue a brute force method. The
- // idea is to start with /dev/dsp(0) and continue with higher device
- // numbers until we reach MAX_DSP_DEVICES. This should tell us how
- // many devices we have ... it is not a fullproof scheme, but hopefully
- // it will work most of the time.
-
- int fd = 0;
- char names[MAX_DEVICES][16];
- for (i=-1; i<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(RtError::WARNING);
- // still count it for now
- }
- }
- }
- else {
- sprintf(message, "RtAudio: OSS playback device (%s) is busy.", device_name);
- error(RtError::WARNING);
- // still count it for now
- }
- }
-
- if (fd >= 0) close(fd);
- strncpy(names[nDevices], device_name, 16);
- nDevices++;
- }
-
- if (nDevices == 0) return;
-
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
-
- // Write device ascii identifiers to device control structure and then probe capabilities.
- for (i=0; 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(RtError::WARNING);
- goto capture_probe;
- }
-
- // We have an open device ... see how many channels it can handle
- for (i=MAX_CHANNELS; i>0; i--) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) {
- // This would normally indicate some sort of hardware error, but under ALSA's
- // OSS emulation, it sometimes indicates an invalid channel value. Further,
- // the returned channel value is not changed. So, we'll ignore the possible
- // hardware error.
- continue; // try next channel number
- }
- // Check to see whether the device supports the requested number of channels
- if (channels != i ) continue; // try next channel number
- // If here, we found the largest working channel value
- break;
- }
- info->maxOutputChannels = channels;
-
- // Now find the minimum number of channels it can handle
- for (i=1; i<=info->maxOutputChannels; i++) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
- continue; // try next channel number
- // If here, we found the smallest working channel value
- break;
- }
- info->minOutputChannels = channels;
- close(fd);
-
- capture_probe:
- // Now try for capture
- fd = open(info->name, O_RDONLY | O_NONBLOCK);
- if (fd == -1) {
- // Open device for capture failed ... either busy or doesn't exist
- if (errno == EBUSY || errno == EAGAIN)
- sprintf(message, "RtAudio: OSS capture device (%s) is busy and cannot be probed.",
- info->name);
- else
- sprintf(message, "RtAudio: OSS capture device (%s) open error.", info->name);
- error(RtError::WARNING);
- if (info->maxOutputChannels == 0)
- // didn't open for playback either ... device invalid
- return;
- goto probe_parameters;
- }
-
- // We have the device open for capture ... see how many channels it can handle
- for (i=MAX_CHANNELS; i>0; i--) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) {
- continue; // as above
- }
- // If here, we found a working channel value
- break;
- }
- info->maxInputChannels = channels;
-
- // Now find the minimum number of channels it can handle
- for (i=1; i<=info->maxInputChannels; i++) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
- continue; // try next channel number
- // If here, we found the smallest working channel value
- break;
- }
- info->minInputChannels = channels;
- close(fd);
-
- // If device opens for both playback and capture, we determine the channels.
- if (info->maxOutputChannels == 0 || info->maxInputChannels == 0)
- goto probe_parameters;
-
- fd = open(info->name, O_RDWR | O_NONBLOCK);
- if (fd == -1)
- goto probe_parameters;
-
- ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
- ioctl(fd, SNDCTL_DSP_GETCAPS, &mask);
- if (mask & DSP_CAP_DUPLEX) {
- info->hasDuplexSupport = true;
- // We have the device open for duplex ... see how many channels it can handle
- for (i=MAX_CHANNELS; i>0; i--) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
- continue; // as above
- // If here, we found a working channel value
- break;
- }
- info->maxDuplexChannels = channels;
-
- // Now find the minimum number of channels it can handle
- for (i=1; i<=info->maxDuplexChannels; i++) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
- continue; // try next channel number
- // If here, we found the smallest working channel value
- break;
- }
- info->minDuplexChannels = channels;
- }
- close(fd);
-
- probe_parameters:
- // At this point, we need to figure out the supported data formats
- // and sample rates. We'll proceed by openning the device in the
- // direction with the maximum number of channels, or playback if
- // they are equal. This might limit our sample rate options, but so
- // be it.
-
- if (info->maxOutputChannels >= info->maxInputChannels) {
- fd = open(info->name, O_WRONLY | O_NONBLOCK);
- channels = info->maxOutputChannels;
- }
- else {
- fd = open(info->name, O_RDONLY | O_NONBLOCK);
- channels = info->maxInputChannels;
- }
-
- if (fd == -1) {
- // We've got some sort of conflict ... abort
- sprintf(message, "RtAudio: OSS device (%s) won't reopen during probe.",
- info->name);
- error(RtError::WARNING);
- return;
- }
-
- // We have an open device ... set to maximum channels.
- i = channels;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) {
- // We've got some sort of conflict ... abort
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) won't revert to previous channel setting.",
- info->name);
- error(RtError::WARNING);
- return;
- }
-
- if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) {
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) can't get supported audio formats.",
- info->name);
- error(RtError::WARNING);
- return;
- }
-
- // Probe the supported data formats ... we don't care about endian-ness just yet.
- int format;
- info->nativeFormats = 0;
-#if defined (AFMT_S32_BE)
- // This format does not seem to be in the 2.4 kernel version of OSS soundcard.h
- if (mask & AFMT_S32_BE) {
- format = AFMT_S32_BE;
- info->nativeFormats |= RTAUDIO_SINT32;
- }
-#endif
-#if defined (AFMT_S32_LE)
- /* This format is not in the 2.4.4 kernel version of OSS soundcard.h */
- if (mask & AFMT_S32_LE) {
- format = AFMT_S32_LE;
- info->nativeFormats |= RTAUDIO_SINT32;
- }
-#endif
- if (mask & AFMT_S8) {
- format = AFMT_S8;
- info->nativeFormats |= RTAUDIO_SINT8;
- }
- if (mask & AFMT_S16_BE) {
- format = AFMT_S16_BE;
- info->nativeFormats |= RTAUDIO_SINT16;
- }
- if (mask & AFMT_S16_LE) {
- format = AFMT_S16_LE;
- info->nativeFormats |= RTAUDIO_SINT16;
- }
-
- // Check that we have at least one supported format
- if (info->nativeFormats == 0) {
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) data format not supported by RtAudio.",
- info->name);
- error(RtError::WARNING);
- return;
- }
-
- // Set the format
- i = format;
- if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1 || format != i) {
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) error setting data format.",
- info->name);
- error(RtError::WARNING);
- return;
- }
-
- // Probe the supported sample rates ... first get lower limit
- int speed = 1;
- if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) {
- // If we get here, we're probably using an ALSA driver with OSS-emulation,
- // which doesn't conform to the OSS specification. In this case,
- // we'll probe our predefined list of sample rates for working values.
- info->nSampleRates = 0;
- for (i=0; 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(RtError::WARNING);
- return;
- }
- info->sampleRates[1] = speed;
- info->nSampleRates = -1;
-
- finished: // That's all ... close the device and return
- close(fd);
- info->probed = true;
- return;
-}
-
-bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
- STREAM_MODE mode, int channels,
- int sampleRate, RTAUDIO_FORMAT format,
- int *bufferSize, int numberOfBuffers)
-{
- int buffers, buffer_bytes, device_channels, device_format;
- int srate, temp, fd;
-
- const char *name = devices[device].name;
-
- if (mode == PLAYBACK)
- fd = open(name, O_WRONLY | O_NONBLOCK);
- else { // mode == RECORD
- if (stream->mode == PLAYBACK && stream->device[0] == device) {
- // We just set the same device for playback ... close and reopen for duplex (OSS only).
- close(stream->handle[0]);
- stream->handle[0] = 0;
- // First check that the number previously set channels is the same.
- if (stream->nUserChannels[0] != channels) {
- sprintf(message, "RtAudio: input/output channels must be equal for OSS duplex device (%s).", name);
- goto error;
- }
- fd = open(name, O_RDWR | O_NONBLOCK);
- }
- else
- fd = open(name, O_RDONLY | O_NONBLOCK);
- }
-
- if (fd == -1) {
- if (errno == EBUSY || errno == EAGAIN)
- sprintf(message, "RtAudio: OSS device (%s) is busy and cannot be opened.",
- name);
- else
- sprintf(message, "RtAudio: OSS device (%s) cannot be opened.", name);
- goto error;
- }
-
- // Now reopen in blocking mode.
- close(fd);
- if (mode == PLAYBACK)
- fd = open(name, O_WRONLY | O_SYNC);
- else { // mode == RECORD
- if (stream->mode == PLAYBACK && stream->device[0] == device)
- fd = open(name, O_RDWR | O_SYNC);
- else
- fd = open(name, O_RDONLY | O_SYNC);
- }
-
- if (fd == -1) {
- sprintf(message, "RtAudio: OSS device (%s) cannot be opened.", name);
- goto error;
- }
-
- // Get the sample format mask
- int mask;
- if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) {
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) can't get supported audio formats.",
- name);
- goto error;
- }
-
- // Determine how to set the device format.
- stream->userFormat = format;
- device_format = -1;
- stream->doByteSwap[mode] = false;
- if (format == RTAUDIO_SINT8) {
- if (mask & AFMT_S8) {
- device_format = AFMT_S8;
- stream->deviceFormat[mode] = RTAUDIO_SINT8;
- }
- }
- else if (format == RTAUDIO_SINT16) {
- if (mask & AFMT_S16_NE) {
- device_format = AFMT_S16_NE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- }
-#if BYTE_ORDER == LITTLE_ENDIAN
- else if (mask & AFMT_S16_BE) {
- device_format = AFMT_S16_BE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->doByteSwap[mode] = true;
- }
-#else
- else if (mask & AFMT_S16_LE) {
- device_format = AFMT_S16_LE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->doByteSwap[mode] = true;
- }
-#endif
- }
-#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE)
- else if (format == RTAUDIO_SINT32) {
- if (mask & AFMT_S32_NE) {
- device_format = AFMT_S32_NE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- }
-#if BYTE_ORDER == LITTLE_ENDIAN
- else if (mask & AFMT_S32_BE) {
- device_format = AFMT_S32_BE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- stream->doByteSwap[mode] = true;
- }
-#else
- else if (mask & AFMT_S32_LE) {
- device_format = AFMT_S32_LE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- stream->doByteSwap[mode] = true;
- }
-#endif
- }
-#endif
-
- if (device_format == -1) {
- // The user requested format is not natively supported by the device.
- if (mask & AFMT_S16_NE) {
- device_format = AFMT_S16_NE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- }
-#if BYTE_ORDER == LITTLE_ENDIAN
- else if (mask & AFMT_S16_BE) {
- device_format = AFMT_S16_BE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->doByteSwap[mode] = true;
- }
-#else
- else if (mask & AFMT_S16_LE) {
- device_format = AFMT_S16_LE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->doByteSwap[mode] = true;
- }
-#endif
-#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE)
- else if (mask & AFMT_S32_NE) {
- device_format = AFMT_S32_NE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- }
-#if BYTE_ORDER == LITTLE_ENDIAN
- else if (mask & AFMT_S32_BE) {
- device_format = AFMT_S32_BE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- stream->doByteSwap[mode] = true;
- }
-#else
- else if (mask & AFMT_S32_LE) {
- device_format = AFMT_S32_LE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- stream->doByteSwap[mode] = true;
- }
-#endif
-#endif
- else if (mask & AFMT_S8) {
- device_format = AFMT_S8;
- stream->deviceFormat[mode] = RTAUDIO_SINT8;
- }
- }
-
- if (stream->deviceFormat[mode] == 0) {
- // This really shouldn't happen ...
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) data format not supported by RtAudio.",
- name);
- goto error;
- }
-
- // Determine the number of channels for this device. Note that the
- // channel value requested by the user might be < min_X_Channels.
- stream->nUserChannels[mode] = channels;
- device_channels = channels;
- if (mode == PLAYBACK) {
- if (channels < devices[device].minOutputChannels)
- device_channels = devices[device].minOutputChannels;
- }
- else { // mode == RECORD
- if (stream->mode == PLAYBACK && stream->device[0] == device) {
- // We're doing duplex setup here.
- if (channels < devices[device].minDuplexChannels)
- device_channels = devices[device].minDuplexChannels;
- }
- else {
- if (channels < devices[device].minInputChannels)
- device_channels = devices[device].minInputChannels;
- }
- }
- stream->nDeviceChannels[mode] = device_channels;
-
- // Attempt to set the buffer size. According to OSS, the minimum
- // number of buffers is two. The supposed minimum buffer size is 16
- // bytes, so that will be our lower bound. The argument to this
- // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
- // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
- // We'll check the actual value used near the end of the setup
- // procedure.
- buffer_bytes = *bufferSize * formatBytes(stream->deviceFormat[mode]) * device_channels;
- if (buffer_bytes < 16) buffer_bytes = 16;
- buffers = numberOfBuffers;
- if (buffers < 2) buffers = 2;
- temp = ((int) buffers << 16) + (int)(log10((double)buffer_bytes)/log10(2.0));
- if (ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &temp)) {
- close(fd);
- sprintf(message, "RtAudio: OSS error setting fragment size for device (%s).",
- name);
- goto error;
- }
- stream->nBuffers = buffers;
-
- // Set the data format.
- temp = device_format;
- if (ioctl(fd, SNDCTL_DSP_SETFMT, &device_format) == -1 || device_format != temp) {
- close(fd);
- sprintf(message, "RtAudio: OSS error setting data format for device (%s).",
- name);
- goto error;
- }
-
- // Set the number of channels.
- temp = device_channels;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &device_channels) == -1 || device_channels != temp) {
- close(fd);
- sprintf(message, "RtAudio: OSS error setting %d channels on device (%s).",
- temp, name);
- goto error;
- }
-
- // Set the sample rate.
- srate = sampleRate;
- temp = srate;
- if (ioctl(fd, SNDCTL_DSP_SPEED, &srate) == -1) {
- close(fd);
- sprintf(message, "RtAudio: OSS error setting sample rate = %d on device (%s).",
- temp, name);
- goto error;
- }
-
- // Verify the sample rate setup worked.
- if (abs(srate - temp) > 100) {
- close(fd);
- sprintf(message, "RtAudio: OSS error ... audio device (%s) doesn't support sample rate of %d.",
- name, temp);
- goto error;
- }
- stream->sampleRate = sampleRate;
-
- if (ioctl(fd, SNDCTL_DSP_GETBLKSIZE, &buffer_bytes) == -1) {
- close(fd);
- sprintf(message, "RtAudio: OSS error getting buffer size for device (%s).",
- name);
- goto error;
- }
-
- // Save buffer size (in sample frames).
- *bufferSize = buffer_bytes / (formatBytes(stream->deviceFormat[mode]) * device_channels);
- stream->bufferSize = *bufferSize;
-
- if (mode == RECORD && stream->mode == PLAYBACK &&
- stream->device[0] == device) {
- // We're doing duplex setup here.
- stream->deviceFormat[0] = stream->deviceFormat[1];
- stream->nDeviceChannels[0] = device_channels;
- }
-
- // Set flags for buffer conversion
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode])
- stream->doConvertBuffer[mode] = true;
-
- // Allocate necessary internal buffers
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
-
- long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
- else
- buffer_bytes = stream->nUserChannels[1];
-
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL) {
- close(fd);
- sprintf(message, "RtAudio: OSS error allocating user buffer memory (%s).",
- name);
- goto error;
- }
- }
-
- if ( stream->doConvertBuffer[mode] ) {
-
- long buffer_bytes;
- bool makeBuffer = true;
- if ( mode == PLAYBACK )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- else { // mode == RECORD
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == PLAYBACK ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- if ( buffer_bytes > bytes_out )
- buffer_bytes = (buffer_bytes > bytes_out) ? buffer_bytes : bytes_out;
- else
- makeBuffer = false;
- }
- }
-
- if ( makeBuffer ) {
- buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL) {
- close(fd);
- free(stream->userBuffer);
- sprintf(message, "RtAudio: OSS error allocating device buffer memory (%s).",
- name);
- goto error;
- }
- }
- }
-
- stream->device[mode] = device;
- stream->handle[mode] = fd;
- stream->state = STREAM_STOPPED;
- if ( stream->mode == PLAYBACK && mode == RECORD ) {
- stream->mode = DUPLEX;
- if (stream->device[0] == device)
- stream->handle[0] = fd;
- }
- else
- stream->mode = mode;
-
- return SUCCESS;
-
- error:
- if (stream->handle[0]) {
- close(stream->handle[0]);
- stream->handle[0] = 0;
- }
- error(RtError::WARNING);
- return FAILURE;
-}
-
-void RtAudio :: cancelStreamCallback(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- if (stream->usingCallback) {
- stream->usingCallback = false;
- pthread_cancel(stream->thread);
- pthread_join(stream->thread, NULL);
- stream->thread = 0;
- stream->callback = NULL;
- stream->userData = NULL;
- }
-}
-
-void RtAudio :: closeStream(int streamId)
-{
- // We don't want an exception to be thrown here because this
- // function is called by our class destructor. So, do our own
- // streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
- error(RtError::WARNING);
- return;
- }
-
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
-
- if (stream->usingCallback) {
- pthread_cancel(stream->thread);
- pthread_join(stream->thread, NULL);
- }
-
- if (stream->state == STREAM_RUNNING) {
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX)
- ioctl(stream->handle[0], SNDCTL_DSP_RESET, 0);
- if (stream->mode == RECORD || stream->mode == DUPLEX)
- ioctl(stream->handle[1], SNDCTL_DSP_RESET, 0);
- }
-
- pthread_mutex_destroy(&stream->mutex);
-
- if (stream->handle[0])
- close(stream->handle[0]);
-
- if (stream->handle[1])
- close(stream->handle[1]);
-
- if (stream->userBuffer)
- free(stream->userBuffer);
-
- if (stream->deviceBuffer)
- free(stream->deviceBuffer);
-
- free(stream);
- streams.erase(streamId);
-}
-
-void RtAudio :: startStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- stream->state = STREAM_RUNNING;
-
- // No need to do anything else here ... OSS automatically starts when fed samples.
-}
-
-void RtAudio :: stopStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- int err;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
- err = ioctl(stream->handle[0], SNDCTL_DSP_SYNC, 0);
- if (err < -1) {
- sprintf(message, "RtAudio: OSS error stopping device (%s).",
- devices[stream->device[0]].name);
- error(RtError::DRIVER_ERROR);
- }
- }
- else {
- err = ioctl(stream->handle[1], SNDCTL_DSP_SYNC, 0);
- if (err < -1) {
- sprintf(message, "RtAudio: OSS error stopping device (%s).",
- devices[stream->device[1]].name);
- error(RtError::DRIVER_ERROR);
- }
- }
- stream->state = STREAM_STOPPED;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-void RtAudio :: abortStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- int err;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
- err = ioctl(stream->handle[0], SNDCTL_DSP_RESET, 0);
- if (err < -1) {
- sprintf(message, "RtAudio: OSS error aborting device (%s).",
- devices[stream->device[0]].name);
- error(RtError::DRIVER_ERROR);
- }
- }
- else {
- err = ioctl(stream->handle[1], SNDCTL_DSP_RESET, 0);
- if (err < -1) {
- sprintf(message, "RtAudio: OSS error aborting device (%s).",
- devices[stream->device[1]].name);
- error(RtError::DRIVER_ERROR);
- }
- }
- stream->state = STREAM_STOPPED;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-int RtAudio :: streamWillBlock(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- int bytes = 0, channels = 0, frames = 0;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- audio_buf_info info;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
- ioctl(stream->handle[0], SNDCTL_DSP_GETOSPACE, &info);
- bytes = info.bytes;
- channels = stream->nDeviceChannels[0];
- }
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
- ioctl(stream->handle[1], SNDCTL_DSP_GETISPACE, &info);
- if (stream->mode == DUPLEX ) {
- bytes = (bytes < info.bytes) ? bytes : info.bytes;
- channels = stream->nDeviceChannels[0];
- }
- else {
- bytes = info.bytes;
- channels = stream->nDeviceChannels[1];
- }
- }
-
- frames = (int) (bytes / (channels * formatBytes(stream->deviceFormat[0])));
- frames -= stream->bufferSize;
- if (frames < 0) frames = 0;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
- return frames;
-}
-
-void RtAudio :: tickStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- int stopStream = 0;
- if (stream->state == STREAM_STOPPED) {
- if (stream->usingCallback) usleep(50000); // sleep 50 milliseconds
- return;
- }
- else if (stream->usingCallback) {
- stopStream = stream->callback(stream->userBuffer, stream->bufferSize, stream->userData);
- }
-
- MUTEX_LOCK(&stream->mutex);
-
- // The state might change while waiting on a mutex.
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- int result;
- char *buffer;
- int samples;
- RTAUDIO_FORMAT format;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
-
- // Setup parameters and do buffer conversion if necessary.
- if (stream->doConvertBuffer[0]) {
- convertStreamBuffer(stream, PLAYBACK);
- buffer = stream->deviceBuffer;
- samples = stream->bufferSize * stream->nDeviceChannels[0];
- format = stream->deviceFormat[0];
- }
- else {
- buffer = stream->userBuffer;
- samples = stream->bufferSize * stream->nUserChannels[0];
- format = stream->userFormat;
- }
-
- // Do byte swapping if necessary.
- if (stream->doByteSwap[0])
- byteSwapBuffer(buffer, samples, format);
-
- // Write samples to device.
- result = write(stream->handle[0], buffer, samples * formatBytes(format));
-
- if (result == -1) {
- // This could be an underrun, but the basic OSS API doesn't provide a means for determining that.
- sprintf(message, "RtAudio: OSS audio write error for device (%s).",
- devices[stream->device[0]].name);
- error(RtError::DRIVER_ERROR);
- }
- }
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
-
- // Setup parameters.
- if (stream->doConvertBuffer[1]) {
- buffer = stream->deviceBuffer;
- samples = stream->bufferSize * stream->nDeviceChannels[1];
- format = stream->deviceFormat[1];
- }
- else {
- buffer = stream->userBuffer;
- samples = stream->bufferSize * stream->nUserChannels[1];
- format = stream->userFormat;
- }
-
- // Read samples from device.
- result = read(stream->handle[1], buffer, samples * formatBytes(format));
-
- if (result == -1) {
- // This could be an overrun, but the basic OSS API doesn't provide a means for determining that.
- sprintf(message, "RtAudio: OSS audio read error for device (%s).",
- devices[stream->device[1]].name);
- error(RtError::DRIVER_ERROR);
- }
-
- // Do byte swapping if necessary.
- if (stream->doByteSwap[1])
- byteSwapBuffer(buffer, samples, format);
-
- // Do buffer conversion if necessary.
- if (stream->doConvertBuffer[1])
- convertStreamBuffer(stream, RECORD);
- }
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-
- if (stream->usingCallback && stopStream)
- this->stopStream(streamId);
-}
-
-extern "C" void *callbackHandler(void *ptr)
-{
- RtAudio *object = thread_info.object;
- int stream = thread_info.streamId;
- bool *usingCallback = (bool *) ptr;
-
- while ( *usingCallback ) {
- pthread_testcancel();
- try {
- object->tickStream(stream);
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nCallback thread error (%s) ... closing thread.\n\n",
- exception.getMessage());
- break;
- }
- }
-
- return 0;
-}
-
-//******************** End of __LINUX_OSS__ *********************//
-
-#elif defined(__WINDOWS_DS__) // Windows DirectSound API
-
-#include <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(RtError::DRIVER_ERROR);
- }
-
- // Count DirectSoundCapture devices.
- result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceCountCallback, &ins);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to enumerate through sound capture devices: %s.",
- getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- count = ins + outs;
- if (count == 0) return;
-
- std::vector<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(RtError::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(RtError::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(RtError::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(RtError::WARNING);
- return;
- }
-
- // Do capture probe first. If this is not the default device (index
- // = 0) _and_ GUID = NULL, then the capture handle is invalid.
- if ( index != 0 && info->id[1] == NULL )
- goto playback_probe;
-
- LPDIRECTSOUNDCAPTURE input;
- result = DirectSoundCaptureCreate( info->id[0], &input, NULL );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Could not create DirectSound capture object (%s): %s.",
- info->name, getErrorString(result));
- error(RtError::WARNING);
- goto playback_probe;
- }
-
- DSCCAPS in_caps;
- in_caps.dwSize = sizeof(in_caps);
- result = input->GetCaps( &in_caps );
- if ( FAILED(result) ) {
- input->Release();
- sprintf(message, "RtAudio: Could not get DirectSound capture capabilities (%s): %s.",
- info->name, getErrorString(result));
- error(RtError::WARNING);
- goto playback_probe;
- }
-
- // Get input channel information.
- info->minInputChannels = 1;
- info->maxInputChannels = in_caps.dwChannels;
-
- // Get sample rate and format information.
- if( in_caps.dwChannels == 2 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->nativeFormats |= RTAUDIO_SINT16;
- if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->nativeFormats |= RTAUDIO_SINT16;
- if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->nativeFormats |= RTAUDIO_SINT16;
- if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->nativeFormats |= RTAUDIO_SINT8;
- if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->nativeFormats |= RTAUDIO_SINT8;
- if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->nativeFormats |= RTAUDIO_SINT8;
-
- if ( info->nativeFormats & RTAUDIO_SINT16 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->sampleRates[info->nSampleRates++] = 11025;
- if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->sampleRates[info->nSampleRates++] = 22050;
- if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->sampleRates[info->nSampleRates++] = 44100;
- }
- else if ( info->nativeFormats & RTAUDIO_SINT8 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->sampleRates[info->nSampleRates++] = 11025;
- if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->sampleRates[info->nSampleRates++] = 22050;
- if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->sampleRates[info->nSampleRates++] = 44100;
- }
- }
- else if ( in_caps.dwChannels == 1 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1M16 ) info->nativeFormats |= RTAUDIO_SINT16;
- if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->nativeFormats |= RTAUDIO_SINT16;
- if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->nativeFormats |= RTAUDIO_SINT16;
- if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->nativeFormats |= RTAUDIO_SINT8;
- if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->nativeFormats |= RTAUDIO_SINT8;
- if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->nativeFormats |= RTAUDIO_SINT8;
-
- if ( info->nativeFormats & RTAUDIO_SINT16 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1M16 ) info->sampleRates[info->nSampleRates++] = 11025;
- if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->sampleRates[info->nSampleRates++] = 22050;
- if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->sampleRates[info->nSampleRates++] = 44100;
- }
- else if ( info->nativeFormats & RTAUDIO_SINT8 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->sampleRates[info->nSampleRates++] = 11025;
- if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->sampleRates[info->nSampleRates++] = 22050;
- if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->sampleRates[info->nSampleRates++] = 44100;
- }
- }
- else info->minInputChannels = 0; // technically, this would be an error
-
- input->Release();
-
- playback_probe:
- LPDIRECTSOUND output;
- DSCAPS out_caps;
-
- // Now do playback probe. If this is not the default device (index
- // = 0) _and_ GUID = NULL, then the playback handle is invalid.
- if ( index != 0 && info->id[0] == NULL )
- goto check_parameters;
-
- result = DirectSoundCreate( info->id[0], &output, NULL );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Could not create DirectSound playback object (%s): %s.",
- info->name, getErrorString(result));
- error(RtError::WARNING);
- goto check_parameters;
- }
-
- out_caps.dwSize = sizeof(out_caps);
- result = output->GetCaps( &out_caps );
- if ( FAILED(result) ) {
- output->Release();
- sprintf(message, "RtAudio: Could not get DirectSound playback capabilities (%s): %s.",
- info->name, getErrorString(result));
- error(RtError::WARNING);
- goto check_parameters;
- }
-
- // Get output channel information.
- info->minOutputChannels = 1;
- info->maxOutputChannels = ( out_caps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
-
- // Get sample rate information. Use capture device rate information
- // if it exists.
- if ( info->nSampleRates == 0 ) {
- info->sampleRates[0] = (int) out_caps.dwMinSecondarySampleRate;
- info->sampleRates[1] = (int) out_caps.dwMaxSecondarySampleRate;
- if ( out_caps.dwFlags & DSCAPS_CONTINUOUSRATE )
- info->nSampleRates = -1;
- else if ( out_caps.dwMinSecondarySampleRate == out_caps.dwMaxSecondarySampleRate ) {
- if ( out_caps.dwMinSecondarySampleRate == 0 ) {
- // This is a bogus driver report ... fake the range and cross
- // your fingers.
- info->sampleRates[0] = 11025;
- info->sampleRates[1] = 48000;
- info->nSampleRates = -1; /* continuous range */
- sprintf(message, "RtAudio: bogus sample rates reported by DirectSound driver ... using defaults (%s).",
- info->name);
- error(RtError::WARNING);
- }
- else {
- info->nSampleRates = 1;
- }
- }
- else if ( (out_caps.dwMinSecondarySampleRate < 1000.0) &&
- (out_caps.dwMaxSecondarySampleRate > 50000.0) ) {
- // This is a bogus driver report ... support for only two
- // distant rates. We'll assume this is a range.
- info->nSampleRates = -1;
- sprintf(message, "RtAudio: bogus sample rates reported by DirectSound driver ... using range (%s).",
- info->name);
- error(RtError::WARNING);
- }
- else info->nSampleRates = 2;
- }
- else {
- // Check input rates against output rate range
- for ( int i=info->nSampleRates-1; i>=0; i-- ) {
- if ( info->sampleRates[i] <= out_caps.dwMaxSecondarySampleRate )
- break;
- info->nSampleRates--;
- }
- while ( info->sampleRates[0] < out_caps.dwMinSecondarySampleRate ) {
- info->nSampleRates--;
- for ( int i=0; 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(RtError::WARNING);
- return FAILURE;
- }
-
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
-
- if ( mode == PLAYBACK ) {
-
- if ( devices[device].maxOutputChannels < channels )
- return FAILURE;
-
- LPGUID id = devices[device].id[0];
- LPDIRECTSOUND object;
- LPDIRECTSOUNDBUFFER buffer;
- DSBUFFERDESC bufferDescription;
-
- result = DirectSoundCreate( id, &object, NULL );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Could not create DirectSound playback object (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Set cooperative level to DSSCL_EXCLUSIVE
- result = object->SetCooperativeLevel(hWnd, DSSCL_EXCLUSIVE);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to set DirectSound cooperative level (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Even though we will write to the secondary buffer, we need to
- // access the primary buffer to set the correct output format.
- // The default is 8-bit, 22 kHz!
- // Setup the DS primary buffer description.
- ZeroMemory(&bufferDescription, sizeof(DSBUFFERDESC));
- bufferDescription.dwSize = sizeof(DSBUFFERDESC);
- bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
- // Obtain the primary buffer
- result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to access DS primary buffer (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Set the primary DS buffer sound format.
- result = buffer->SetFormat(&waveFormat);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to set DS primary buffer format (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Setup the secondary DS buffer description.
- buffer_size = channels * *bufferSize * nBuffers * waveFormat.wBitsPerSample / 8;
- ZeroMemory(&bufferDescription, sizeof(DSBUFFERDESC));
- bufferDescription.dwSize = sizeof(DSBUFFERDESC);
- bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
- DSBCAPS_GETCURRENTPOSITION2 |
- DSBCAPS_LOCHARDWARE ); // Force hardware mixing
- bufferDescription.dwBufferBytes = buffer_size;
- bufferDescription.lpwfxFormat = &waveFormat;
-
- // Try to create the secondary DS buffer. If that doesn't work,
- // try to use software mixing. Otherwise, there's a problem.
- result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL);
- if ( FAILED(result) ) {
- bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
- DSBCAPS_GETCURRENTPOSITION2 |
- DSBCAPS_LOCSOFTWARE ); // Force software mixing
- result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to create secondary DS buffer (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
- }
-
- // Get the buffer size ... might be different from what we specified.
- DSBCAPS dsbcaps;
- dsbcaps.dwSize = sizeof(DSBCAPS);
- buffer->GetCaps(&dsbcaps);
- buffer_size = dsbcaps.dwBufferBytes;
-
- // Lock the DS buffer
- result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to lock DS buffer (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Zero the DS buffer
- ZeroMemory(audioPtr, dataLen);
-
- // Unlock the DS buffer
- result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to unlock DS buffer(%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- stream->handle[0].object = (void *) object;
- stream->handle[0].buffer = (void *) buffer;
- stream->nDeviceChannels[0] = channels;
- }
-
- if ( mode == RECORD ) {
-
- if ( devices[device].maxInputChannels < channels )
- return FAILURE;
-
- LPGUID id = devices[device].id[1];
- LPDIRECTSOUNDCAPTURE object;
- LPDIRECTSOUNDCAPTUREBUFFER buffer;
- DSCBUFFERDESC bufferDescription;
-
- result = DirectSoundCaptureCreate( id, &object, NULL );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Could not create DirectSound capture object (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Setup the secondary DS buffer description.
- buffer_size = channels * *bufferSize * nBuffers * waveFormat.wBitsPerSample / 8;
- ZeroMemory(&bufferDescription, sizeof(DSCBUFFERDESC));
- bufferDescription.dwSize = sizeof(DSCBUFFERDESC);
- bufferDescription.dwFlags = 0;
- bufferDescription.dwReserved = 0;
- bufferDescription.dwBufferBytes = buffer_size;
- bufferDescription.lpwfxFormat = &waveFormat;
-
- // Create the capture buffer.
- result = object->CreateCaptureBuffer(&bufferDescription, &buffer, NULL);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to create DS capture buffer (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Lock the capture buffer
- result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Zero the buffer
- ZeroMemory(audioPtr, dataLen);
-
- // Unlock the buffer
- result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- stream->handle[1].object = (void *) object;
- stream->handle[1].buffer = (void *) buffer;
- stream->nDeviceChannels[1] = channels;
- }
-
- stream->userFormat = format;
- if ( waveFormat.wBitsPerSample == 8 )
- stream->deviceFormat[mode] = RTAUDIO_SINT8;
- else
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->nUserChannels[mode] = channels;
- *bufferSize = buffer_size / (channels * nBuffers * waveFormat.wBitsPerSample / 8);
- stream->bufferSize = *bufferSize;
-
- // Set flags for buffer conversion
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode])
- stream->doConvertBuffer[mode] = true;
-
- // Allocate necessary internal buffers
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
-
- long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
- else
- buffer_bytes = stream->nUserChannels[1];
-
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL)
- goto memory_error;
- }
-
- if ( stream->doConvertBuffer[mode] ) {
-
- long buffer_bytes;
- bool makeBuffer = true;
- if ( mode == PLAYBACK )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- else { // mode == RECORD
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == PLAYBACK ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- if ( buffer_bytes > bytes_out )
- buffer_bytes = (buffer_bytes > bytes_out) ? buffer_bytes : bytes_out;
- else
- makeBuffer = false;
- }
- }
-
- if ( makeBuffer ) {
- buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL)
- goto memory_error;
- }
- }
-
- stream->device[mode] = device;
- stream->state = STREAM_STOPPED;
- if ( stream->mode == PLAYBACK && mode == RECORD )
- // We had already set up an output stream.
- stream->mode = DUPLEX;
- else
- stream->mode = mode;
- stream->nBuffers = nBuffers;
- stream->sampleRate = sampleRate;
-
- return SUCCESS;
-
- memory_error:
- if (stream->handle[0].object) {
- LPDIRECTSOUND object = (LPDIRECTSOUND) stream->handle[0].object;
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- if (buffer) {
- buffer->Release();
- stream->handle[0].buffer = NULL;
- }
- object->Release();
- stream->handle[0].object = NULL;
- }
- if (stream->handle[1].object) {
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) stream->handle[1].object;
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- if (buffer) {
- buffer->Release();
- stream->handle[1].buffer = NULL;
- }
- object->Release();
- stream->handle[1].object = NULL;
- }
- if (stream->userBuffer) {
- free(stream->userBuffer);
- stream->userBuffer = 0;
- }
- sprintf(message, "RtAudio: error allocating buffer memory (%s).",
- devices[device].name);
- error(RtError::WARNING);
- return FAILURE;
-}
-
-void RtAudio :: cancelStreamCallback(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- if (stream->usingCallback) {
- stream->usingCallback = false;
- WaitForSingleObject( (HANDLE)stream->thread, INFINITE );
- CloseHandle( (HANDLE)stream->thread );
- stream->thread = 0;
- stream->callback = NULL;
- stream->userData = NULL;
- }
-}
-
-void RtAudio :: closeStream(int streamId)
-{
- // We don't want an exception to be thrown here because this
- // function is called by our class destructor. So, do our own
- // streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
- error(RtError::WARNING);
- return;
- }
-
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
-
- if (stream->usingCallback) {
- stream->usingCallback = false;
- WaitForSingleObject( (HANDLE)stream->thread, INFINITE );
- CloseHandle( (HANDLE)stream->thread );
- }
-
- DeleteCriticalSection(&stream->mutex);
-
- if (stream->handle[0].object) {
- LPDIRECTSOUND object = (LPDIRECTSOUND) stream->handle[0].object;
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- if (buffer) {
- buffer->Stop();
- buffer->Release();
- }
- object->Release();
- }
-
- if (stream->handle[1].object) {
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) stream->handle[1].object;
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- if (buffer) {
- buffer->Stop();
- buffer->Release();
- }
- object->Release();
- }
-
- if (stream->userBuffer)
- free(stream->userBuffer);
-
- if (stream->deviceBuffer)
- free(stream->deviceBuffer);
-
- free(stream);
- streams.erase(streamId);
-}
-
-void RtAudio :: startStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_RUNNING)
- goto unlock;
-
- HRESULT result;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- result = buffer->Play(0, 0, DSBPLAY_LOOPING );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to start DS buffer (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
- }
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- result = buffer->Start(DSCBSTART_LOOPING );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to start DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
- }
- stream->state = STREAM_RUNNING;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-void RtAudio :: stopStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED) {
- MUTEX_UNLOCK(&stream->mutex);
- return;
- }
-
- // There is no specific DirectSound API call to "drain" a buffer
- // before stopping. We can hack this for playback by writing zeroes
- // for another bufferSize * nBuffers frames. For capture, the
- // concept is less clear so we'll repeat what we do in the
- // abortStream() case.
- HRESULT result;
- DWORD dsBufferSize;
- LPVOID buffer1 = NULL;
- LPVOID buffer2 = NULL;
- DWORD bufferSize1 = 0;
- DWORD bufferSize2 = 0;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
-
- DWORD currentPos, safePos;
- long buffer_bytes = stream->bufferSize * stream->nDeviceChannels[0];
- buffer_bytes *= formatBytes(stream->deviceFormat[0]);
-
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- UINT nextWritePos = stream->handle[0].bufferPointer;
- dsBufferSize = buffer_bytes * stream->nBuffers;
-
- // Write zeroes for nBuffer counts.
- for (int i=0; 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(RtError::DRIVER_ERROR);
- }
-
- if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
- DWORD endWrite = nextWritePos + buffer_bytes;
-
- // Check whether the entire write region is behind the play pointer.
- while ( currentPos < endWrite ) {
- float millis = (endWrite - currentPos) * 900.0;
- millis /= ( formatBytes(stream->deviceFormat[0]) * stream->sampleRate);
- if ( millis < 1.0 ) millis = 1.0;
- Sleep( (DWORD) millis );
-
- // Wake up, find out where we are now
- result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
- if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
- }
-
- // Lock free space in the buffer
- result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1,
- &bufferSize1, &buffer2, &bufferSize2, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS buffer during playback (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- // Zero the free space
- ZeroMemory(buffer1, bufferSize1);
- if (buffer2 != NULL) ZeroMemory(buffer2, bufferSize2);
-
- // Update our buffer offset and unlock sound buffer
- dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS buffer during playback (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
- nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize;
- stream->handle[0].bufferPointer = nextWritePos;
- }
-
- // If we play again, start at the beginning of the buffer.
- stream->handle[0].bufferPointer = 0;
- }
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- buffer1 = NULL;
- bufferSize1 = 0;
-
- result = buffer->Stop();
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to stop DS capture buffer (%s): %s",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- dsBufferSize = stream->bufferSize * stream->nDeviceChannels[1];
- dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers;
-
- // Lock the buffer and clear it so that if we start to play again,
- // we won't have old data playing.
- result = buffer->Lock(0, dsBufferSize, &buffer1, &bufferSize1, NULL, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- // Zero the DS buffer
- ZeroMemory(buffer1, bufferSize1);
-
- // Unlock the DS buffer
- result = buffer->Unlock(buffer1, bufferSize1, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- // If we start recording again, we must begin at beginning of buffer.
- stream->handle[1].bufferPointer = 0;
- }
- stream->state = STREAM_STOPPED;
-
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-void RtAudio :: abortStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- HRESULT result;
- long dsBufferSize;
- LPVOID audioPtr;
- DWORD dataLen;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- result = buffer->Stop();
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to stop DS buffer (%s): %s",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- dsBufferSize = stream->bufferSize * stream->nDeviceChannels[0];
- dsBufferSize *= formatBytes(stream->deviceFormat[0]) * stream->nBuffers;
-
- // Lock the buffer and clear it so that if we start to play again,
- // we won't have old data playing.
- result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS buffer (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- // Zero the DS buffer
- ZeroMemory(audioPtr, dataLen);
-
- // Unlock the DS buffer
- result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS buffer (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- // If we start playing again, we must begin at beginning of buffer.
- stream->handle[0].bufferPointer = 0;
- }
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- audioPtr = NULL;
- dataLen = 0;
-
- result = buffer->Stop();
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to stop DS capture buffer (%s): %s",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- dsBufferSize = stream->bufferSize * stream->nDeviceChannels[1];
- dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers;
-
- // Lock the buffer and clear it so that if we start to play again,
- // we won't have old data playing.
- result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- // Zero the DS buffer
- ZeroMemory(audioPtr, dataLen);
-
- // Unlock the DS buffer
- result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- // If we start recording again, we must begin at beginning of buffer.
- stream->handle[1].bufferPointer = 0;
- }
- stream->state = STREAM_STOPPED;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-int RtAudio :: streamWillBlock(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- int channels;
- int frames = 0;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- HRESULT result;
- DWORD currentPos, safePos;
- channels = 1;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
-
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- UINT nextWritePos = stream->handle[0].bufferPointer;
- channels = stream->nDeviceChannels[0];
- DWORD dsBufferSize = stream->bufferSize * channels;
- dsBufferSize *= formatBytes(stream->deviceFormat[0]) * stream->nBuffers;
-
- // Find out where the read and "safe write" pointers are.
- result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
- frames = currentPos - nextWritePos;
- frames /= channels * formatBytes(stream->deviceFormat[0]);
- }
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
-
- LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- UINT nextReadPos = stream->handle[1].bufferPointer;
- channels = stream->nDeviceChannels[1];
- DWORD dsBufferSize = stream->bufferSize * channels;
- dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers;
-
- // Find out where the write and "safe read" pointers are.
- result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset
-
- if (stream->mode == DUPLEX ) {
- // Take largest value of the two.
- int temp = safePos - nextReadPos;
- temp /= channels * formatBytes(stream->deviceFormat[1]);
- frames = ( temp > frames ) ? temp : frames;
- }
- else {
- frames = safePos - nextReadPos;
- frames /= channels * formatBytes(stream->deviceFormat[1]);
- }
- }
-
- frames = stream->bufferSize - frames;
- if (frames < 0) frames = 0;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
- return frames;
-}
-
-void RtAudio :: tickStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- int stopStream = 0;
- if (stream->state == STREAM_STOPPED) {
- if (stream->usingCallback) Sleep(50); // sleep 50 milliseconds
- return;
- }
- else if (stream->usingCallback) {
- stopStream = stream->callback(stream->userBuffer, stream->bufferSize, stream->userData);
- }
-
- MUTEX_LOCK(&stream->mutex);
-
- // The state might change while waiting on a mutex.
- if (stream->state == STREAM_STOPPED) {
- MUTEX_UNLOCK(&stream->mutex);
- if (stream->usingCallback && stopStream)
- this->stopStream(streamId);
- }
-
- HRESULT result;
- DWORD currentPos, safePos;
- LPVOID buffer1 = NULL;
- LPVOID buffer2 = NULL;
- DWORD bufferSize1 = 0;
- DWORD bufferSize2 = 0;
- char *buffer;
- long buffer_bytes;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
-
- // Setup parameters and do buffer conversion if necessary.
- if (stream->doConvertBuffer[0]) {
- convertStreamBuffer(stream, PLAYBACK);
- buffer = stream->deviceBuffer;
- buffer_bytes = stream->bufferSize * stream->nDeviceChannels[0];
- buffer_bytes *= formatBytes(stream->deviceFormat[0]);
- }
- else {
- buffer = stream->userBuffer;
- buffer_bytes = stream->bufferSize * stream->nUserChannels[0];
- buffer_bytes *= formatBytes(stream->userFormat);
- }
-
- // No byte swapping necessary in DirectSound implementation.
-
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- UINT nextWritePos = stream->handle[0].bufferPointer;
- DWORD dsBufferSize = buffer_bytes * stream->nBuffers;
-
- // Find out where the read and "safe write" pointers are.
- result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
- DWORD endWrite = nextWritePos + buffer_bytes;
-
- // Check whether the entire write region is behind the play pointer.
- while ( currentPos < endWrite ) {
- // If we are here, then we must wait until the play pointer gets
- // beyond the write region. The approach here is to use the
- // Sleep() function to suspend operation until safePos catches
- // up. Calculate number of milliseconds to wait as:
- // time = distance * (milliseconds/second) * fudgefactor /
- // ((bytes/sample) * (samples/second))
- // A "fudgefactor" less than 1 is used because it was found
- // that sleeping too long was MUCH worse than sleeping for
- // several shorter periods.
- float millis = (endWrite - currentPos) * 900.0;
- millis /= ( formatBytes(stream->deviceFormat[0]) * stream->sampleRate);
- if ( millis < 1.0 ) millis = 1.0;
- Sleep( (DWORD) millis );
-
- // Wake up, find out where we are now
- result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
- if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
- }
-
- // Lock free space in the buffer
- result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1,
- &bufferSize1, &buffer2, &bufferSize2, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS buffer during playback (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- // Copy our buffer into the DS buffer
- CopyMemory(buffer1, buffer, bufferSize1);
- if (buffer2 != NULL) CopyMemory(buffer2, buffer+bufferSize1, bufferSize2);
-
- // Update our buffer offset and unlock sound buffer
- dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS buffer during playback (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
- nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize;
- stream->handle[0].bufferPointer = nextWritePos;
- }
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
-
- // Setup parameters.
- if (stream->doConvertBuffer[1]) {
- buffer = stream->deviceBuffer;
- buffer_bytes = stream->bufferSize * stream->nDeviceChannels[1];
- buffer_bytes *= formatBytes(stream->deviceFormat[1]);
- }
- else {
- buffer = stream->userBuffer;
- buffer_bytes = stream->bufferSize * stream->nUserChannels[1];
- buffer_bytes *= formatBytes(stream->userFormat);
- }
-
- LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- UINT nextReadPos = stream->handle[1].bufferPointer;
- DWORD dsBufferSize = buffer_bytes * stream->nBuffers;
-
- // Find out where the write and "safe read" pointers are.
- result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset
- DWORD endRead = nextReadPos + buffer_bytes;
-
- // Check whether the entire write region is behind the play pointer.
- while ( safePos < endRead ) {
- // See comments for playback.
- float millis = (endRead - safePos) * 900.0;
- millis /= ( formatBytes(stream->deviceFormat[1]) * stream->sampleRate);
- if ( millis < 1.0 ) millis = 1.0;
- Sleep( (DWORD) millis );
-
- // Wake up, find out where we are now
- result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset
- }
-
- // Lock free space in the buffer
- result = dsBuffer->Lock (nextReadPos, buffer_bytes, &buffer1,
- &bufferSize1, &buffer2, &bufferSize2, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS buffer during capture (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- // Copy our buffer into the DS buffer
- CopyMemory(buffer, buffer1, bufferSize1);
- if (buffer2 != NULL) CopyMemory(buffer+bufferSize1, buffer2, bufferSize2);
-
- // Update our buffer offset and unlock sound buffer
- nextReadPos = (nextReadPos + bufferSize1 + bufferSize2) % dsBufferSize;
- dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS buffer during capture (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
- stream->handle[1].bufferPointer = nextReadPos;
-
- // No byte swapping necessary in DirectSound implementation.
-
- // Do buffer conversion if necessary.
- if (stream->doConvertBuffer[1])
- convertStreamBuffer(stream, RECORD);
- }
-
- MUTEX_UNLOCK(&stream->mutex);
-
- if (stream->usingCallback && stopStream)
- this->stopStream(streamId);
-}
-
-// Definitions for utility functions and callbacks
-// specific to the DirectSound implementation.
-
-extern "C" unsigned __stdcall callbackHandler(void *ptr)
-{
- RtAudio *object = thread_info.object;
- int stream = thread_info.streamId;
- bool *usingCallback = (bool *) ptr;
-
- while ( *usingCallback ) {
- try {
- object->tickStream(stream);
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nCallback thread error (%s) ... closing thread.\n\n",
- exception.getMessage());
- break;
- }
- }
-
- _endthreadex( 0 );
- return 0;
-}
-
-static bool CALLBACK deviceCountCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext)
-{
- int *pointer = ((int *) lpContext);
- (*pointer)++;
-
- return true;
-}
-
-static bool CALLBACK deviceInfoCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext)
-{
- enum_info *info = ((enum_info *) lpContext);
- while (strlen(info->name) > 0) info++;
-
- strncpy(info->name, lpcstrDescription, 64);
- info->id = lpguid;
-
- HRESULT hr;
- info->isValid = false;
- if (info->isInput == true) {
- DSCCAPS caps;
- LPDIRECTSOUNDCAPTURE object;
-
- hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
- if( hr != DS_OK ) return true;
-
- caps.dwSize = sizeof(caps);
- hr = object->GetCaps( &caps );
- if( hr == DS_OK ) {
- if (caps.dwChannels > 0 && caps.dwFormats > 0)
- info->isValid = true;
- }
- object->Release();
- }
- else {
- DSCAPS caps;
- LPDIRECTSOUND object;
- hr = DirectSoundCreate( lpguid, &object, NULL );
- if( hr != DS_OK ) return true;
-
- caps.dwSize = sizeof(caps);
- hr = object->GetCaps( &caps );
- if( hr == DS_OK ) {
- if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
- info->isValid = true;
- }
- object->Release();
- }
-
- return true;
-}
-
-static char* getErrorString(int code)
-{
- switch (code) {
-
- case DSERR_ALLOCATED:
- return "Direct Sound already allocated";
-
- case DSERR_CONTROLUNAVAIL:
- return "Direct Sound control unavailable";
-
- case DSERR_INVALIDPARAM:
- return "Direct Sound invalid parameter";
-
- case DSERR_INVALIDCALL:
- return "Direct Sound invalid call";
-
- case DSERR_GENERIC:
- return "Direct Sound generic error";
-
- case DSERR_PRIOLEVELNEEDED:
- return "Direct Sound Priority level needed";
-
- case DSERR_OUTOFMEMORY:
- return "Direct Sound out of memory";
-
- case DSERR_BADFORMAT:
- return "Direct Sound bad format";
-
- case DSERR_UNSUPPORTED:
- return "Direct Sound unsupported error";
-
- case DSERR_NODRIVER:
- return "Direct Sound no driver error";
-
- case DSERR_ALREADYINITIALIZED:
- return "Direct Sound already initialized";
-
- case DSERR_NOAGGREGATION:
- return "Direct Sound no aggregation";
-
- case DSERR_BUFFERLOST:
- return "Direct Sound buffer lost";
-
- case DSERR_OTHERAPPHASPRIO:
- return "Direct Sound other app has priority";
-
- case DSERR_UNINITIALIZED:
- return "Direct Sound uninitialized";
-
- default:
- return "Direct Sound unknown error";
- }
-}
-
-//******************** End of __WINDOWS_DS__ *********************//
-
-#elif defined(__IRIX_AL__) // SGI's AL API for IRIX
-
-#include <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(RtError::DRIVER_ERROR);
- }
-
- if (nDevices <= 0) return;
-
- ALvalue *vls = (ALvalue *) new ALvalue[nDevices];
-
- // Add one for our default input/output devices.
- nDevices++;
-
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
-
- // Write device ascii identifiers to device info structure.
- char name[32];
- int outs, ins, i;
- ALpv pvs[1];
- pvs[0].param = AL_NAME;
- pvs[0].value.ptr = name;
- pvs[0].sizeIn = 32;
-
- strcpy(devices[0].name, "Default Input/Output Devices");
-
- outs = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, vls, nDevices-1, 0, 0);
- if (outs < 0) {
- sprintf(message, "RtAudio: AL error getting output devices: %s.",
- alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
- }
-
- for (i=0; i<outs; i++) {
- if (alGetParams(vls[i].i, pvs, 1) < 0) {
- sprintf(message, "RtAudio: AL error querying output devices: %s.",
- alGetErrorString(oserror()));
- error(RtError::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(RtError::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(RtError::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(RtError::WARNING);
- }
- else
- resource = value.i;
- }
- else
- resource = info->id[0];
-
- if (resource > 0) {
-
- // Probe output device parameters.
- result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0);
- if (result < 0) {
- sprintf(message, "RtAudio: AL error getting device (%s) channels: %s.",
- info->name, alGetErrorString(oserror()));
- error(RtError::WARNING);
- }
- else {
- info->maxOutputChannels = value.i;
- info->minOutputChannels = 1;
- }
-
- result = alGetParamInfo(resource, AL_RATE, &pinfo);
- if (result < 0) {
- sprintf(message, "RtAudio: AL error getting device (%s) rates: %s.",
- info->name, alGetErrorString(oserror()));
- error(RtError::WARNING);
- }
- else {
- info->nSampleRates = 0;
- for (i=0; i<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(RtError::WARNING);
- }
- else
- resource = value.i;
- }
- else
- resource = info->id[1];
-
- if (resource > 0) {
-
- // Probe input device parameters.
- result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0);
- if (result < 0) {
- sprintf(message, "RtAudio: AL error getting device (%s) channels: %s.",
- info->name, alGetErrorString(oserror()));
- error(RtError::WARNING);
- }
- else {
- info->maxInputChannels = value.i;
- info->minInputChannels = 1;
- }
-
- result = alGetParamInfo(resource, AL_RATE, &pinfo);
- if (result < 0) {
- sprintf(message, "RtAudio: AL error getting device (%s) rates: %s.",
- info->name, alGetErrorString(oserror()));
- error(RtError::WARNING);
- }
- else {
- // In the case of the default device, these values will
- // overwrite the rates determined for the output device. Since
- // the input device is most likely to be more limited than the
- // output device, this is ok.
- info->nSampleRates = 0;
- for (i=0; i<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(RtError::WARNING);
- return FAILURE;
- }
-
- // Set the channels.
- result = alSetChannels(al_config, channels);
- if ( result < 0 ) {
- sprintf(message,"RtAudio: can't set %d channels in AL config: %s.",
- channels, alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Set the queue (buffer) size.
- if ( numberOfBuffers < 1 )
- nBuffers = 1;
- else
- nBuffers = numberOfBuffers;
- long buffer_size = *bufferSize * nBuffers;
- result = alSetQueueSize(al_config, buffer_size); // in sample frames
- if ( result < 0 ) {
- sprintf(message,"RtAudio: can't set buffer size (%ld) in AL config: %s.",
- buffer_size, alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Set the data format.
- stream->userFormat = format;
- stream->deviceFormat[mode] = format;
- if (format == RTAUDIO_SINT8) {
- result = alSetSampFmt(al_config, AL_SAMPFMT_TWOSCOMP);
- result = alSetWidth(al_config, AL_SAMPLE_8);
- }
- else if (format == RTAUDIO_SINT16) {
- result = alSetSampFmt(al_config, AL_SAMPFMT_TWOSCOMP);
- result = alSetWidth(al_config, AL_SAMPLE_16);
- }
- else if (format == RTAUDIO_SINT24) {
- // Our 24-bit format assumes the upper 3 bytes of a 4 byte word.
- // The AL library uses the lower 3 bytes, so we'll need to do our
- // own conversion.
- result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT);
- stream->deviceFormat[mode] = RTAUDIO_FLOAT32;
- }
- else if (format == RTAUDIO_SINT32) {
- // The AL library doesn't seem to support the 32-bit integer
- // format, so we'll need to do our own conversion.
- result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT);
- stream->deviceFormat[mode] = RTAUDIO_FLOAT32;
- }
- else if (format == RTAUDIO_FLOAT32)
- result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT);
- else if (format == RTAUDIO_FLOAT64)
- result = alSetSampFmt(al_config, AL_SAMPFMT_DOUBLE);
-
- if ( result == -1 ) {
- sprintf(message,"RtAudio: AL error setting sample format in AL config: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- if (mode == PLAYBACK) {
-
- // Set our device.
- if (device == 0)
- resource = AL_DEFAULT_OUTPUT;
- else
- resource = devices[device].id[0];
- result = alSetDevice(al_config, resource);
- if ( result == -1 ) {
- sprintf(message,"RtAudio: AL error setting device (%s) in AL config: %s.",
- devices[device].name, alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Open the port.
- port = alOpenPort("RtAudio Output Port", "w", al_config);
- if( !port ) {
- sprintf(message,"RtAudio: AL error opening output port: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Set the sample rate
- pvs[0].param = AL_MASTER_CLOCK;
- pvs[0].value.i = AL_CRYSTAL_MCLK_TYPE;
- pvs[1].param = AL_RATE;
- pvs[1].value.ll = alDoubleToFixed((double)sampleRate);
- result = alSetParams(resource, pvs, 2);
- if ( result < 0 ) {
- alClosePort(port);
- sprintf(message,"RtAudio: AL error setting sample rate (%d) for device (%s): %s.",
- sampleRate, devices[device].name, alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
- }
- else { // mode == RECORD
-
- // Set our device.
- if (device == 0)
- resource = AL_DEFAULT_INPUT;
- else
- resource = devices[device].id[1];
- result = alSetDevice(al_config, resource);
- if ( result == -1 ) {
- sprintf(message,"RtAudio: AL error setting device (%s) in AL config: %s.",
- devices[device].name, alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Open the port.
- port = alOpenPort("RtAudio Output Port", "r", al_config);
- if( !port ) {
- sprintf(message,"RtAudio: AL error opening input port: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Set the sample rate
- pvs[0].param = AL_MASTER_CLOCK;
- pvs[0].value.i = AL_CRYSTAL_MCLK_TYPE;
- pvs[1].param = AL_RATE;
- pvs[1].value.ll = alDoubleToFixed((double)sampleRate);
- result = alSetParams(resource, pvs, 2);
- if ( result < 0 ) {
- alClosePort(port);
- sprintf(message,"RtAudio: AL error setting sample rate (%d) for device (%s): %s.",
- sampleRate, devices[device].name, alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
- }
-
- alFreeConfig(al_config);
-
- stream->nUserChannels[mode] = channels;
- stream->nDeviceChannels[mode] = channels;
-
- // Set handle and flags for buffer conversion
- stream->handle[mode] = port;
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
-
- // Allocate necessary internal buffers
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
-
- long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
- else
- buffer_bytes = stream->nUserChannels[1];
-
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL)
- goto memory_error;
- }
-
- if ( stream->doConvertBuffer[mode] ) {
-
- long buffer_bytes;
- bool makeBuffer = true;
- if ( mode == PLAYBACK )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- else { // mode == RECORD
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == PLAYBACK ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- if ( buffer_bytes > bytes_out )
- buffer_bytes = (buffer_bytes > bytes_out) ? buffer_bytes : bytes_out;
- else
- makeBuffer = false;
- }
- }
-
- if ( makeBuffer ) {
- buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL)
- goto memory_error;
- }
- }
-
- stream->device[mode] = device;
- stream->state = STREAM_STOPPED;
- if ( stream->mode == PLAYBACK && mode == RECORD )
- // We had already set up an output stream.
- stream->mode = DUPLEX;
- else
- stream->mode = mode;
- stream->nBuffers = nBuffers;
- stream->bufferSize = *bufferSize;
- stream->sampleRate = sampleRate;
-
- return SUCCESS;
-
- memory_error:
- if (stream->handle[0]) {
- alClosePort(stream->handle[0]);
- stream->handle[0] = 0;
- }
- if (stream->handle[1]) {
- alClosePort(stream->handle[1]);
- stream->handle[1] = 0;
- }
- if (stream->userBuffer) {
- free(stream->userBuffer);
- stream->userBuffer = 0;
- }
- sprintf(message, "RtAudio: ALSA error allocating buffer memory for device (%s).",
- devices[device].name);
- error(RtError::WARNING);
- return FAILURE;
-}
-
-void RtAudio :: cancelStreamCallback(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- if (stream->usingCallback) {
- stream->usingCallback = false;
- pthread_cancel(stream->thread);
- pthread_join(stream->thread, NULL);
- stream->thread = 0;
- stream->callback = NULL;
- stream->userData = NULL;
- }
-}
-
-void RtAudio :: closeStream(int streamId)
-{
- // We don't want an exception to be thrown here because this
- // function is called by our class destructor. So, do our own
- // streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
- error(RtError::WARNING);
- return;
- }
-
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
-
- if (stream->usingCallback) {
- pthread_cancel(stream->thread);
- pthread_join(stream->thread, NULL);
- }
-
- pthread_mutex_destroy(&stream->mutex);
-
- if (stream->handle[0])
- alClosePort(stream->handle[0]);
-
- if (stream->handle[1])
- alClosePort(stream->handle[1]);
-
- if (stream->userBuffer)
- free(stream->userBuffer);
-
- if (stream->deviceBuffer)
- free(stream->deviceBuffer);
-
- free(stream);
- streams.erase(streamId);
-}
-
-void RtAudio :: startStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- if (stream->state == STREAM_RUNNING)
- return;
-
- // The AL port is ready as soon as it is opened.
- stream->state = STREAM_RUNNING;
-}
-
-void RtAudio :: stopStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- int result;
- int buffer_size = stream->bufferSize * stream->nBuffers;
-
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX)
- alZeroFrames(stream->handle[0], buffer_size);
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
- result = alDiscardFrames(stream->handle[1], buffer_size);
- if (result == -1) {
- sprintf(message, "RtAudio: AL error draining stream device (%s): %s.",
- devices[stream->device[1]].name, alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
- }
- }
- stream->state = STREAM_STOPPED;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-void RtAudio :: abortStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
-
- int buffer_size = stream->bufferSize * stream->nBuffers;
- int result = alDiscardFrames(stream->handle[0], buffer_size);
- if (result == -1) {
- sprintf(message, "RtAudio: AL error aborting stream device (%s): %s.",
- devices[stream->device[0]].name, alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
- }
- }
-
- // There is no clear action to take on the input stream, since the
- // port will continue to run in any event.
- stream->state = STREAM_STOPPED;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-int RtAudio :: streamWillBlock(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- int frames = 0;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- int err = 0;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
- err = alGetFillable(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: AL error getting available frames for stream (%s): %s.",
- devices[stream->device[0]].name, alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
- }
- }
-
- frames = err;
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
- err = alGetFilled(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: AL error getting available frames for stream (%s): %s.",
- devices[stream->device[1]].name, alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
- }
- if (frames > err) frames = err;
- }
-
- frames = stream->bufferSize - frames;
- if (frames < 0) frames = 0;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
- return frames;
-}
-
-void RtAudio :: tickStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- int stopStream = 0;
- if (stream->state == STREAM_STOPPED) {
- if (stream->usingCallback) usleep(50000); // sleep 50 milliseconds
- return;
- }
- else if (stream->usingCallback) {
- stopStream = stream->callback(stream->userBuffer, stream->bufferSize, stream->userData);
- }
-
- MUTEX_LOCK(&stream->mutex);
-
- // The state might change while waiting on a mutex.
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- char *buffer;
- int channels;
- RTAUDIO_FORMAT format;
- if (stream->mode == PLAYBACK || stream->mode == DUPLEX) {
-
- // Setup parameters and do buffer conversion if necessary.
- if (stream->doConvertBuffer[0]) {
- convertStreamBuffer(stream, PLAYBACK);
- buffer = stream->deviceBuffer;
- channels = stream->nDeviceChannels[0];
- format = stream->deviceFormat[0];
- }
- else {
- buffer = stream->userBuffer;
- channels = stream->nUserChannels[0];
- format = stream->userFormat;
- }
-
- // Do byte swapping if necessary.
- if (stream->doByteSwap[0])
- byteSwapBuffer(buffer, stream->bufferSize * channels, format);
-
- // Write interleaved samples to device.
- alWriteFrames(stream->handle[0], buffer, stream->bufferSize);
- }
-
- if (stream->mode == RECORD || stream->mode == DUPLEX) {
-
- // Setup parameters.
- if (stream->doConvertBuffer[1]) {
- buffer = stream->deviceBuffer;
- channels = stream->nDeviceChannels[1];
- format = stream->deviceFormat[1];
- }
- else {
- buffer = stream->userBuffer;
- channels = stream->nUserChannels[1];
- format = stream->userFormat;
- }
-
- // Read interleaved samples from device.
- alReadFrames(stream->handle[1], buffer, stream->bufferSize);
-
- // Do byte swapping if necessary.
- if (stream->doByteSwap[1])
- byteSwapBuffer(buffer, stream->bufferSize * channels, format);
-
- // Do buffer conversion if necessary.
- if (stream->doConvertBuffer[1])
- convertStreamBuffer(stream, RECORD);
- }
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-
- if (stream->usingCallback && stopStream)
- this->stopStream(streamId);
-}
-
-extern "C" void *callbackHandler(void *ptr)
-{
- RtAudio *object = thread_info.object;
- int stream = thread_info.streamId;
- bool *usingCallback = (bool *) ptr;
-
- while ( *usingCallback ) {
- pthread_testcancel();
- try {
- object->tickStream(stream);
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nCallback thread error (%s) ... closing thread.\n\n",
- exception.getMessage());
- break;
- }
- }
-
- return 0;
-}
-
-//******************** End of __IRIX_AL__ *********************//
-
-#endif
-
-
-// *************************************************** //
-//
-// Private common (OS-independent) RtAudio methods.
-//
-// *************************************************** //
-
-// This method can be modified to control the behavior of error
-// message reporting and throwing.
-void RtAudio :: error(RtError::TYPE type)
-{
- if (type == RtError::WARNING) {
-#if defined(RTAUDIO_DEBUG)
- fprintf(stderr, "\n%s\n\n", message);
- else if (type == RtError::DEBUG_WARNING) {
- fprintf(stderr, "\n%s\n\n", message);
-#endif
- }
- else {
- fprintf(stderr, "\n%s\n\n", message);
- throw RtError(message, type);
- }
-}
-
-void *RtAudio :: verifyStream(int streamId)
-{
- // Verify the stream key.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
- error(RtError::INVALID_STREAM);
- }
-
- return streams[streamId];
-}
-
-void RtAudio :: clearDeviceInfo(RTAUDIO_DEVICE *info)
-{
- // Don't clear the name or DEVICE_ID fields here ... they are
- // typically set prior to a call of this function.
- info->probed = false;
- info->maxOutputChannels = 0;
- info->maxInputChannels = 0;
- info->maxDuplexChannels = 0;
- info->minOutputChannels = 0;
- info->minInputChannels = 0;
- info->minDuplexChannels = 0;
- info->hasDuplexSupport = false;
- info->nSampleRates = 0;
- for (int i=0; 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(RtError::WARNING);
-
- return 0;
-}
-
-void RtAudio :: convertStreamBuffer(RTAUDIO_STREAM *stream, STREAM_MODE mode)
-{
- // This method does format conversion, input/output channel compensation, and
- // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
- // the upper three bytes of a 32-bit integer.
-
- int j, channels_in, channels_out, channels;
- RTAUDIO_FORMAT format_in, format_out;
- char *input, *output;
-
- if (mode == RECORD) { // convert device to user buffer
- input = stream->deviceBuffer;
- output = stream->userBuffer;
- channels_in = stream->nDeviceChannels[1];
- channels_out = stream->nUserChannels[1];
- format_in = stream->deviceFormat[1];
- format_out = stream->userFormat;
- }
- else { // convert user to device buffer
- input = stream->userBuffer;
- output = stream->deviceBuffer;
- channels_in = stream->nUserChannels[0];
- channels_out = stream->nDeviceChannels[0];
- format_in = stream->userFormat;
- format_out = stream->deviceFormat[0];
-
- // clear our device buffer when in/out duplex device channels are different
- if ( stream->mode == DUPLEX &&
- stream->nDeviceChannels[0] != stream->nDeviceChannels[1] )
- memset(output, 0, stream->bufferSize * channels_out * formatBytes(format_out));
- }
-
- channels = (channels_in < channels_out) ? channels_in : channels_out;
-
- // Set up the interleave/deinterleave offsets
- std::vector<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;
- }
- }
-}
-
-
-// *************************************************** //
-//
-// RtError class definition.
-//
-// *************************************************** //
-
-RtError :: RtError(const char *p, TYPE tipe)
-{
- type = tipe;
- strncpy(error_message, p, 256);
-}
-
-RtError :: ~RtError()
-{
-}
-
-void RtError :: printMessage()
-{
- printf("\n%s\n\n", error_message);
-}
+/************************************************************************/ +/*! \class RtAudio + \brief Realtime audio i/o C++ class. + + RtAudio provides a common API (Application Programming Interface) + for realtime audio input/output across Linux (native ALSA and + OSS), SGI, Macintosh OS X (CoreAudio), and Windows (DirectSound + and ASIO) operating systems. + + RtAudio WWW site: http://www-ccrma.stanford.edu/~gary/rtaudio/ + + RtAudio: a realtime audio i/o C++ class + Copyright (c) 2001-2002 Gary P. Scavone + + Permission is hereby granted, free of charge, to any person + obtaining a copy of this software and associated documentation files + (the "Software"), to deal in the Software without restriction, + including without limitation the rights to use, copy, modify, merge, + publish, distribute, sublicense, and/or sell copies of the Software, + and to permit persons to whom the Software is furnished to do so, + subject to the following conditions: + + The above copyright notice and this permission notice shall be + included in all copies or substantial portions of the Software. + + Any person wishing to distribute modifications to the Software is + requested to send the modifications to the original developer so that + they can be incorporated into the canonical version. + + THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR + ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF + CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION + WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. +*/ +/************************************************************************/ + + +#include "RtAudio.h" +#include <vector> +#include <stdio.h> +#include <iostream.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__) || defined(__WINDOWS_ASIO__) + #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) + #define MUTEX_LOCK(A) EnterCriticalSection(A) + #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) +#else // pthread API + #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) + #define MUTEX_LOCK(A) pthread_mutex_lock(A) + #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A) +#endif + +// *************************************************** // +// +// Public common (OS-independent) methods. +// +// *************************************************** // + +RtAudio :: RtAudio() +{ + initialize(); + + if (nDevices <= 0) { + sprintf(message, "RtAudio: no audio devices found!"); + error(RtError::NO_DEVICES_FOUND); + } +} + +RtAudio :: RtAudio(int *streamId, + int outputDevice, int outputChannels, + int inputDevice, int inputChannels, + RTAUDIO_FORMAT format, int sampleRate, + int *bufferSize, int numberOfBuffers) +{ + initialize(); + + if (nDevices <= 0) { + sprintf(message, "RtAudio: no audio devices found!"); + error(RtError::NO_DEVICES_FOUND); + } + + try { + *streamId = openStream(outputDevice, outputChannels, inputDevice, inputChannels, + format, sampleRate, bufferSize, numberOfBuffers); + } + catch (RtError &exception) { + // deallocate the RTAUDIO_DEVICE structures + if (devices) free(devices); + throw exception; + } +} + +RtAudio :: ~RtAudio() +{ + // close any existing streams + while ( streams.size() ) + closeStream( streams.begin()->first ); + + // deallocate the RTAUDIO_DEVICE structures + if (devices) free(devices); +} + +int RtAudio :: openStream(int outputDevice, int outputChannels, + int inputDevice, int inputChannels, + RTAUDIO_FORMAT format, int sampleRate, + int *bufferSize, int numberOfBuffers) +{ + static int streamKey = 0; // Unique stream identifier ... OK for multiple instances. + + if (outputChannels < 1 && inputChannels < 1) { + sprintf(message,"RtAudio: one or both 'channel' parameters must be greater than zero."); + error(RtError::INVALID_PARAMETER); + } + + if ( formatBytes(format) == 0 ) { + sprintf(message,"RtAudio: 'format' parameter value is undefined."); + error(RtError::INVALID_PARAMETER); + } + + if ( outputChannels > 0 ) { + if (outputDevice > nDevices || outputDevice < 0) { + sprintf(message,"RtAudio: 'outputDevice' parameter value (%d) is invalid.", outputDevice); + error(RtError::INVALID_PARAMETER); + } + } + + if ( inputChannels > 0 ) { + if (inputDevice > nDevices || inputDevice < 0) { + sprintf(message,"RtAudio: 'inputDevice' parameter value (%d) is invalid.", inputDevice); + error(RtError::INVALID_PARAMETER); + } + } + + // Allocate a new stream structure. + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) calloc(1, sizeof(RTAUDIO_STREAM)); + if (stream == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + stream->mode = UNINITIALIZED; + MUTEX_INITIALIZE(&stream->mutex); + + bool result = FAILURE; + int device, defaultDevice = 0; + STREAM_MODE mode; + int channels; + if ( outputChannels > 0 ) { + + mode = OUTPUT; + channels = outputChannels; + + if ( outputDevice == 0 ) { // Try default device first. + defaultDevice = getDefaultOutputDevice(); + device = defaultDevice; + } + else + device = outputDevice - 1; + + for (int i=-1; i<nDevices; i++) { + if (i >= 0 ) { + if ( i == defaultDevice ) continue; + device = i; + } + if (devices[device].probed == false) { + // If the device wasn't successfully probed before, try it + // again now. + clearDeviceInfo(&devices[device]); + probeDeviceInfo(&devices[device]); + } + if ( devices[device].probed ) + result = probeDeviceOpen(device, stream, mode, channels, sampleRate, + format, bufferSize, numberOfBuffers); + if (result == SUCCESS) break; + if ( outputDevice > 0 ) break; + } + } + + if ( inputChannels > 0 && ( result == SUCCESS || outputChannels <= 0 ) ) { + + mode = INPUT; + channels = inputChannels; + + if ( inputDevice == 0 ) { // Try default device first. + defaultDevice = getDefaultInputDevice(); + device = defaultDevice; + } + else + device = inputDevice - 1; + + for (int i=-1; i<nDevices; i++) { + if (i >= 0 ) { + if ( i == defaultDevice ) continue; + device = i; + } + if (devices[device].probed == false) { + // If the device wasn't successfully probed before, try it + // again now. + clearDeviceInfo(&devices[device]); + probeDeviceInfo(&devices[device]); + } + if ( devices[device].probed ) + result = probeDeviceOpen(device, stream, mode, channels, sampleRate, + format, bufferSize, numberOfBuffers); + if (result == SUCCESS) break; + if ( outputDevice > 0 ) break; + } + } + + streams[++streamKey] = (void *) stream; + if ( result == SUCCESS ) + return streamKey; + + // If we get here, all attempted probes failed. Close any opened + // devices and delete the allocated stream. + closeStream(streamKey); + if ( ( outputDevice == 0 && outputChannels > 0 ) + || ( inputDevice == 0 && inputChannels > 0 ) ) + sprintf(message,"RtAudio: no devices found for given parameters."); + else + sprintf(message,"RtAudio: unable to open specified device(s) with given stream parameters."); + error(RtError::INVALID_PARAMETER); + + return -1; +} + +int RtAudio :: getDeviceCount(void) +{ + return nDevices; +} + +void RtAudio :: getDeviceInfo(int device, RTAUDIO_DEVICE *info) +{ + if (device > nDevices || device < 1) { + sprintf(message, "RtAudio: invalid device specifier (%d)!", device); + error(RtError::INVALID_DEVICE); + } + + int deviceIndex = device - 1; + + // If the device wasn't successfully probed before, try it now (or again). + if (devices[deviceIndex].probed == false) { + clearDeviceInfo(&devices[deviceIndex]); + probeDeviceInfo(&devices[deviceIndex]); + } + + // Clear the info structure. + memset(info, 0, sizeof(RTAUDIO_DEVICE)); + + strncpy(info->name, devices[deviceIndex].name, 128); + info->probed = devices[deviceIndex].probed; + if ( info->probed == true ) { + info->maxOutputChannels = devices[deviceIndex].maxOutputChannels; + info->maxInputChannels = devices[deviceIndex].maxInputChannels; + info->maxDuplexChannels = devices[deviceIndex].maxDuplexChannels; + info->minOutputChannels = devices[deviceIndex].minOutputChannels; + info->minInputChannels = devices[deviceIndex].minInputChannels; + info->minDuplexChannels = devices[deviceIndex].minDuplexChannels; + info->hasDuplexSupport = devices[deviceIndex].hasDuplexSupport; + info->nSampleRates = devices[deviceIndex].nSampleRates; + if (info->nSampleRates == -1) { + info->sampleRates[0] = devices[deviceIndex].sampleRates[0]; + info->sampleRates[1] = devices[deviceIndex].sampleRates[1]; + } + else { + for (int i=0; i<info->nSampleRates; i++) + info->sampleRates[i] = devices[deviceIndex].sampleRates[i]; + } + info->nativeFormats = devices[deviceIndex].nativeFormats; + if ( deviceIndex == getDefaultOutputDevice() || + deviceIndex == getDefaultInputDevice() ) + info->isDefault = true; + } + + return; +} + +char * const RtAudio :: getStreamBuffer(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + return stream->userBuffer; +} + +#if defined(__LINUX_ALSA__) || defined(__LINUX_OSS__) || defined(__IRIX_AL__) + +extern "C" void *callbackHandler(void * ptr); + +void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + CALLBACK_INFO *info = (CALLBACK_INFO *) &stream->callbackInfo; + if ( info->usingCallback ) { + sprintf(message, "RtAudio: A callback is already set for this stream!"); + error(RtError::WARNING); + return; + } + + info->callback = (void *) callback; + info->userData = userData; + info->usingCallback = true; + info->object = (void *) this; + info->streamId = streamId; + + int err = pthread_create(&info->thread, NULL, callbackHandler, &stream->callbackInfo); + + if (err) { + info->usingCallback = false; + sprintf(message, "RtAudio: error starting callback thread!"); + error(RtError::THREAD_ERROR); + } +} + +void RtAudio :: cancelStreamCallback(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + if (stream->callbackInfo.usingCallback) { + + if (stream->state == STREAM_RUNNING) + stopStream( streamId ); + + MUTEX_LOCK(&stream->mutex); + + stream->callbackInfo.usingCallback = false; + pthread_cancel(stream->callbackInfo.thread); + pthread_join(stream->callbackInfo.thread, NULL); + stream->callbackInfo.thread = 0; + stream->callbackInfo.callback = NULL; + stream->callbackInfo.userData = NULL; + + MUTEX_UNLOCK(&stream->mutex); + } +} + +#endif + +// *************************************************** // +// +// OS/API-specific methods. +// +// *************************************************** // + +#if defined(__MACOSX_CORE__) + +// The OS X CoreAudio API is designed to use a separate callback +// procedure for each of its audio devices. A single RtAudio duplex +// stream using two different devices is supported here, though it +// cannot be guaranteed to always behave correctly because we cannot +// synchronize these two callbacks. This same functionality can be +// achieved with better synchrony by opening two separate streams for +// the devices and using RtAudio blocking calls (i.e. tickStream()). +// +// The possibility of having multiple RtAudio streams accessing the +// same CoreAudio device is not currently supported. The problem +// involves the inability to install our callbackHandler function for +// the same device more than once. I experimented with a workaround +// for this, but it requires an additional buffer for mixing output +// data before filling the CoreAudio device buffer. In the end, I +// decided it wasn't worth supporting. +// +// Property listeners are currently not used. The issue is what could +// be done if a critical stream parameter (buffer size, sample rate, +// device disconnect) notification arrived. The listeners entail +// quite a bit of extra code and most likely, a user program wouldn't +// be prepared for the result anyway. Some initial listener code is +// commented out. + +void RtAudio :: initialize(void) +{ + OSStatus err = noErr; + UInt32 dataSize; + AudioDeviceID *deviceList = NULL; + nDevices = 0; + + // Find out how many audio devices there are, if any. + err = AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &dataSize, NULL); + if (err != noErr) { + sprintf(message, "RtAudio: OSX error getting device info!"); + error(RtError::SYSTEM_ERROR); + } + + nDevices = dataSize / sizeof(AudioDeviceID); + if (nDevices == 0) return; + + // Allocate the RTAUDIO_DEVICE structures. + devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); + if (devices == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + + // Make space for the devices we are about to get. + deviceList = (AudioDeviceID *) malloc( dataSize ); + if (deviceList == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + + // Get the array of AudioDeviceIDs. + err = AudioHardwareGetProperty(kAudioHardwarePropertyDevices, &dataSize, (void *) deviceList); + if (err != noErr) { + free(deviceList); + sprintf(message, "RtAudio: OSX error getting device properties!"); + error(RtError::SYSTEM_ERROR); + } + + // Write device identifiers to device structures and then + // probe the device capabilities. + for (int i=0; i<nDevices; i++) { + devices[i].id[0] = deviceList[i]; + //probeDeviceInfo(&devices[i]); + } + + free(deviceList); +} + +int RtAudio :: getDefaultInputDevice(void) +{ + AudioDeviceID id; + UInt32 dataSize = sizeof( AudioDeviceID ); + + OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultInputDevice, + &dataSize, &id ); + + if (result != noErr) { + sprintf( message, "RtAudio: OSX error getting default input device." ); + error(RtError::WARNING); + return 0; + } + + for ( int i=0; i<nDevices; i++ ) { + if ( id == devices[i].id[0] ) return i; + } + + return 0; +} + +int RtAudio :: getDefaultOutputDevice(void) +{ + AudioDeviceID id; + UInt32 dataSize = sizeof( AudioDeviceID ); + + OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultOutputDevice, + &dataSize, &id ); + + if (result != noErr) { + sprintf( message, "RtAudio: OSX error getting default output device." ); + error(RtError::WARNING); + return 0; + } + + for ( int i=0; i<nDevices; i++ ) { + if ( id == devices[i].id[0] ) return i; + } + + return 0; +} + +static bool deviceSupportsFormat( AudioDeviceID id, bool isInput, + AudioStreamBasicDescription *desc, bool isDuplex ) +{ + OSStatus result = noErr; + UInt32 dataSize = sizeof( AudioStreamBasicDescription ); + + result = AudioDeviceGetProperty( id, 0, isInput, + kAudioDevicePropertyStreamFormatSupported, + &dataSize, desc ); + + if (result == kAudioHardwareNoError) { + if ( isDuplex ) { + result = AudioDeviceGetProperty( id, 0, true, + kAudioDevicePropertyStreamFormatSupported, + &dataSize, desc ); + + + if (result != kAudioHardwareNoError) + return false; + } + return true; + } + + return false; +} + +void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info) +{ + OSStatus err = noErr; + + // Get the device manufacturer and name. + char name[256]; + char fullname[512]; + UInt32 dataSize = 256; + err = AudioDeviceGetProperty( info->id[0], 0, false, + kAudioDevicePropertyDeviceManufacturer, + &dataSize, name ); + if (err != noErr) { + sprintf( message, "RtAudio: OSX error getting device manufacturer." ); + error(RtError::DEBUG_WARNING); + return; + } + strncpy(fullname, name, 256); + strcat(fullname, ": " ); + + dataSize = 256; + err = AudioDeviceGetProperty( info->id[0], 0, false, + kAudioDevicePropertyDeviceName, + &dataSize, name ); + if (err != noErr) { + sprintf( message, "RtAudio: OSX error getting device name." ); + error(RtError::DEBUG_WARNING); + return; + } + strncat(fullname, name, 254); + strncat(info->name, fullname, 128); + + // Get output channel information. + unsigned int i, minChannels, maxChannels, nStreams = 0; + AudioBufferList *bufferList = nil; + err = AudioDeviceGetPropertyInfo( info->id[0], 0, false, + kAudioDevicePropertyStreamConfiguration, + &dataSize, NULL ); + if (err == noErr && dataSize > 0) { + bufferList = (AudioBufferList *) malloc( dataSize ); + if (bufferList == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::DEBUG_WARNING); + return; + } + + err = AudioDeviceGetProperty( info->id[0], 0, false, + kAudioDevicePropertyStreamConfiguration, + &dataSize, bufferList ); + if (err == noErr) { + maxChannels = 0; + minChannels = 1000; + nStreams = bufferList->mNumberBuffers; + for ( i=0; i<nStreams; i++ ) { + maxChannels += bufferList->mBuffers[i].mNumberChannels; + if ( bufferList->mBuffers[i].mNumberChannels < minChannels ) + minChannels = bufferList->mBuffers[i].mNumberChannels; + } + } + } + if (err != noErr || dataSize <= 0) { + sprintf( message, "RtAudio: OSX error getting output channels for device (%s).", info->name ); + error(RtError::DEBUG_WARNING); + return; + } + + free (bufferList); + if ( nStreams ) { + if ( maxChannels > 0 ) + info->maxOutputChannels = maxChannels; + if ( minChannels > 0 ) + info->minOutputChannels = minChannels; + } + + // Get input channel information. + bufferList = nil; + err = AudioDeviceGetPropertyInfo( info->id[0], 0, true, + kAudioDevicePropertyStreamConfiguration, + &dataSize, NULL ); + if (err == noErr && dataSize > 0) { + bufferList = (AudioBufferList *) malloc( dataSize ); + if (bufferList == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::DEBUG_WARNING); + return; + } + err = AudioDeviceGetProperty( info->id[0], 0, true, + kAudioDevicePropertyStreamConfiguration, + &dataSize, bufferList ); + if (err == noErr) { + maxChannels = 0; + minChannels = 1000; + nStreams = bufferList->mNumberBuffers; + for ( i=0; i<nStreams; i++ ) { + if ( bufferList->mBuffers[i].mNumberChannels < minChannels ) + minChannels = bufferList->mBuffers[i].mNumberChannels; + maxChannels += bufferList->mBuffers[i].mNumberChannels; + } + } + } + if (err != noErr || dataSize <= 0) { + sprintf( message, "RtAudio: OSX error getting input channels for device (%s).", info->name ); + error(RtError::DEBUG_WARNING); + return; + } + + free (bufferList); + if ( nStreams ) { + if ( maxChannels > 0 ) + info->maxInputChannels = maxChannels; + if ( minChannels > 0 ) + info->minInputChannels = minChannels; + } + + // If device opens for both playback and capture, we determine the channels. + if (info->maxOutputChannels > 0 && info->maxInputChannels > 0) { + info->hasDuplexSupport = true; + info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ? + info->maxInputChannels : info->maxOutputChannels; + info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ? + info->minInputChannels : info->minOutputChannels; + } + + // Probe the device sample rate and data format parameters. The + // core audio query mechanism is performed on a "stream" + // description, which can have a variable number of channels and + // apply to input or output only. + + // Create a stream description structure. + AudioStreamBasicDescription description; + dataSize = sizeof( AudioStreamBasicDescription ); + memset(&description, 0, sizeof(AudioStreamBasicDescription)); + bool isInput = false; + if ( info->maxOutputChannels == 0 ) isInput = true; + bool isDuplex = false; + if ( info->maxDuplexChannels > 0 ) isDuplex = true; + + // Determine the supported sample rates. + info->nSampleRates = 0; + for (i=0; i<MAX_SAMPLE_RATES; i++) { + description.mSampleRate = (double) SAMPLE_RATES[i]; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->sampleRates[info->nSampleRates++] = SAMPLE_RATES[i]; + } + + if (info->nSampleRates == 0) { + sprintf( message, "RtAudio: No supported sample rates found for OSX device (%s).", info->name ); + error(RtError::DEBUG_WARNING); + return; + } + + // Check for continuous sample rate support. + description.mSampleRate = kAudioStreamAnyRate; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) { + info->sampleRates[1] = info->sampleRates[info->nSampleRates-1]; + info->nSampleRates = -1; + } + + // Determine the supported data formats. + info->nativeFormats = 0; + description.mFormatID = kAudioFormatLinearPCM; + description.mBitsPerChannel = 8; + description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT8; + else { + description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT8; + } + + description.mBitsPerChannel = 16; + description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT16; + else { + description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT16; + } + + description.mBitsPerChannel = 32; + description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT32; + else { + description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT32; + } + + description.mBitsPerChannel = 24; + description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsAlignedHigh | kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT24; + else { + description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_SINT24; + } + + description.mBitsPerChannel = 32; + description.mFormatFlags = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_FLOAT32; + else { + description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_FLOAT32; + } + + description.mBitsPerChannel = 64; + description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_FLOAT64; + else { + description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; + if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) + info->nativeFormats |= RTAUDIO_FLOAT64; + } + + // Check that we have at least one supported format. + if (info->nativeFormats == 0) { + sprintf(message, "RtAudio: OSX PCM device (%s) data format not supported by RtAudio.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + info->probed = true; +} + +OSStatus callbackHandler(AudioDeviceID inDevice, + const AudioTimeStamp* inNow, + const AudioBufferList* inInputData, + const AudioTimeStamp* inInputTime, + AudioBufferList* outOutputData, + const AudioTimeStamp* inOutputTime, + void* infoPointer) +{ + CALLBACK_INFO *info = (CALLBACK_INFO *) infoPointer; + + RtAudio *object = (RtAudio *) info->object; + try { + object->callbackEvent( info->streamId, inDevice, (void *)inInputData, (void *)outOutputData ); + } + catch (RtError &exception) { + fprintf(stderr, "\nCallback handler error (%s)!\n\n", exception.getMessage()); + return kAudioHardwareUnspecifiedError; + } + + return kAudioHardwareNoError; +} + +/* +OSStatus deviceListener(AudioDeviceID inDevice, + UInt32 channel, + Boolean isInput, + AudioDevicePropertyID propertyID, + void* infoPointer) +{ + CALLBACK_INFO *info = (CALLBACK_INFO *) infoPointer; + + RtAudio *object = (RtAudio *) info->object; + try { + object->settingChange( info->streamId ); + } + catch (RtError &exception) { + fprintf(stderr, "\nDevice listener error (%s)!\n\n", exception.getMessage()); + return kAudioHardwareUnspecifiedError; + } + + return kAudioHardwareNoError; +} +*/ + +bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, + STREAM_MODE mode, int channels, + int sampleRate, RTAUDIO_FORMAT format, + int *bufferSize, int numberOfBuffers) +{ + // Check to make sure we don't already have a stream accessing this device. + RTAUDIO_STREAM *streamPtr; + std::map<int, void *>::const_iterator i; + for ( i=streams.begin(); i!=streams.end(); ++i ) { + streamPtr = (RTAUDIO_STREAM *) i->second; + if ( streamPtr->device[0] == device || streamPtr->device[1] == device ) { + sprintf(message, "RtAudio: no current OS X support for multiple streams accessing the same device!"); + error(RtError::WARNING); + return FAILURE; + } + } + + // Setup for stream mode. + bool isInput = false; + AudioDeviceID id = devices[device].id[0]; + if ( mode == INPUT ) isInput = true; + + // Search for a stream which contains the desired number of channels. + OSStatus err = noErr; + UInt32 dataSize; + unsigned int deviceChannels, nStreams; + UInt32 iChannel = 0, iStream = 0; + AudioBufferList *bufferList = nil; + err = AudioDeviceGetPropertyInfo( id, 0, isInput, + kAudioDevicePropertyStreamConfiguration, + &dataSize, NULL ); + + if (err == noErr && dataSize > 0) { + bufferList = (AudioBufferList *) malloc( dataSize ); + if (bufferList == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + err = AudioDeviceGetProperty( id, 0, isInput, + kAudioDevicePropertyStreamConfiguration, + &dataSize, bufferList ); + + if (err == noErr) { + stream->deInterleave[mode] = false; + nStreams = bufferList->mNumberBuffers; + for ( iStream=0; iStream<nStreams; iStream++ ) { + if ( bufferList->mBuffers[iStream].mNumberChannels >= (unsigned int) channels ) break; + iChannel += bufferList->mBuffers[iStream].mNumberChannels; + } + // If we didn't find a single stream above, see if we can meet + // the channel specification in mono mode (i.e. using separate + // non-interleaved buffers). This can only work if there are N + // consecutive one-channel streams, where N is the number of + // desired channels. + iChannel = 0; + if ( iStream >= nStreams && nStreams >= (unsigned int) channels ) { + int counter = 0; + for ( iStream=0; iStream<nStreams; iStream++ ) { + if ( bufferList->mBuffers[iStream].mNumberChannels == 1 ) + counter++; + else + counter = 0; + if ( counter == channels ) { + iStream -= channels - 1; + iChannel -= channels - 1; + stream->deInterleave[mode] = true; + break; + } + iChannel += bufferList->mBuffers[iStream].mNumberChannels; + } + } + } + } + if (err != noErr || dataSize <= 0) { + if ( bufferList ) free( bufferList ); + sprintf( message, "RtAudio: OSX error getting channels for device (%s).", devices[device].name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + if (iStream >= nStreams) { + free (bufferList); + sprintf( message, "RtAudio: unable to find OSX audio stream on device (%s) for requested channels (%d).", + devices[device].name, channels ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + // This is ok even for mono mode ... it gets updated later. + deviceChannels = bufferList->mBuffers[iStream].mNumberChannels; + free (bufferList); + + // Determine the buffer size. + AudioValueRange bufferRange; + dataSize = sizeof(AudioValueRange); + err = AudioDeviceGetProperty( id, 0, isInput, + kAudioDevicePropertyBufferSizeRange, + &dataSize, &bufferRange); + if (err != noErr) { + sprintf( message, "RtAudio: OSX error getting buffer size range for device (%s).", + devices[device].name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + long bufferBytes = *bufferSize * deviceChannels * formatBytes(RTAUDIO_FLOAT32); + if (bufferRange.mMinimum > bufferBytes) bufferBytes = (int) bufferRange.mMinimum; + else if (bufferRange.mMaximum < bufferBytes) bufferBytes = (int) bufferRange.mMaximum; + + // Set the buffer size. For mono mode, I'm assuming we only need to + // make this setting for the first channel. + UInt32 theSize = (UInt32) bufferBytes; + dataSize = sizeof( UInt32); + err = AudioDeviceSetProperty(id, NULL, 0, isInput, + kAudioDevicePropertyBufferSize, + dataSize, &theSize); + if (err != noErr) { + sprintf( message, "RtAudio: OSX error setting the buffer size for device (%s).", + devices[device].name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + *bufferSize = bufferBytes / ( deviceChannels * formatBytes(RTAUDIO_FLOAT32) ); + if ( stream->mode == OUTPUT && mode == INPUT && *bufferSize != stream->bufferSize ) { + sprintf( message, "RtAudio: OSX error setting buffer size for duplex stream on device (%s).", + devices[device].name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + stream->bufferSize = *bufferSize; + stream->nBuffers = 1; + + // Set the stream format description. Do for each channel in mono mode. + AudioStreamBasicDescription description; + dataSize = sizeof( AudioStreamBasicDescription ); + if ( stream->deInterleave[mode] ) nStreams = channels; + else nStreams = 1; + for ( unsigned int i=0; i<nStreams; i++, iChannel++ ) { + + err = AudioDeviceGetProperty( id, iChannel, isInput, + kAudioDevicePropertyStreamFormat, + &dataSize, &description ); + if (err != noErr) { + sprintf( message, "RtAudio: OSX error getting stream format for device (%s).", devices[device].name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + // Set the sample rate and data format id. + description.mSampleRate = (double) sampleRate; + description.mFormatID = kAudioFormatLinearPCM; + err = AudioDeviceSetProperty( id, NULL, iChannel, isInput, + kAudioDevicePropertyStreamFormat, + dataSize, &description ); + if (err != noErr) { + sprintf( message, "RtAudio: OSX error setting sample rate or data format for device (%s).", devices[device].name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + } + + // Check whether we need byte-swapping (assuming OS X host is big-endian). + iChannel -= nStreams; + err = AudioDeviceGetProperty( id, iChannel, isInput, + kAudioDevicePropertyStreamFormat, + &dataSize, &description ); + if (err != noErr) { + sprintf( message, "RtAudio: OSX error getting stream format for device (%s).", devices[device].name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + stream->doByteSwap[mode] = false; + if ( !description.mFormatFlags & kLinearPCMFormatFlagIsBigEndian ) + stream->doByteSwap[mode] = true; + + // From the CoreAudio documentation, PCM data must be supplied as + // 32-bit floats. + stream->userFormat = format; + stream->deviceFormat[mode] = RTAUDIO_FLOAT32; + + if ( stream->deInterleave[mode] ) + stream->nDeviceChannels[mode] = channels; + else + stream->nDeviceChannels[mode] = description.mChannelsPerFrame; + stream->nUserChannels[mode] = channels; + + // Set handle and flags for buffer conversion. + stream->handle[mode] = iStream; + stream->doConvertBuffer[mode] = false; + if (stream->userFormat != stream->deviceFormat[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] > 1 && stream->deInterleave[mode]) + stream->doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers. + if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { + + long buffer_bytes; + if (stream->nUserChannels[0] >= stream->nUserChannels[1]) + buffer_bytes = stream->nUserChannels[0]; + else + buffer_bytes = stream->nUserChannels[1]; + + buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); + if (stream->userBuffer) free(stream->userBuffer); + stream->userBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->userBuffer == NULL) + goto memory_error; + } + + if ( stream->deInterleave[mode] ) { + + long buffer_bytes; + bool makeBuffer = true; + if ( mode == OUTPUT ) + buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + else { // mode == INPUT + buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); + if ( stream->mode == OUTPUT && stream->deviceBuffer ) { + long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + if ( buffer_bytes < bytes_out ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + buffer_bytes *= *bufferSize; + if (stream->deviceBuffer) free(stream->deviceBuffer); + stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->deviceBuffer == NULL) + goto memory_error; + + // If not de-interleaving, we point stream->deviceBuffer to the + // OS X supplied device buffer before doing any necessary data + // conversions. This presents a problem if we have a duplex + // stream using one device which needs de-interleaving and + // another device which doesn't. So, save a pointer to our own + // device buffer in the CALLBACK_INFO structure. + stream->callbackInfo.buffers = stream->deviceBuffer; + } + } + + stream->sampleRate = sampleRate; + stream->device[mode] = device; + stream->state = STREAM_STOPPED; + stream->callbackInfo.object = (void *) this; + stream->callbackInfo.waitTime = (unsigned long) (200000.0 * stream->bufferSize / stream->sampleRate); + stream->callbackInfo.device[mode] = id; + if ( stream->mode == OUTPUT && mode == INPUT && stream->device[0] == device ) + // Only one callback procedure per device. + stream->mode = DUPLEX; + else { + err = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream->callbackInfo ); + if (err != noErr) { + sprintf( message, "RtAudio: OSX error setting callback for device (%s).", devices[device].name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + if ( stream->mode == OUTPUT && mode == INPUT ) + stream->mode = DUPLEX; + else + stream->mode = mode; + } + + // If we wanted to use property listeners, they would be setup here. + + return SUCCESS; + + memory_error: + if (stream->userBuffer) { + free(stream->userBuffer); + stream->userBuffer = 0; + } + sprintf(message, "RtAudio: OSX error allocating buffer memory (%s).", devices[device].name); + error(RtError::WARNING); + return FAILURE; +} + +void RtAudio :: cancelStreamCallback(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + if (stream->callbackInfo.usingCallback) { + + if (stream->state == STREAM_RUNNING) + stopStream( streamId ); + + MUTEX_LOCK(&stream->mutex); + + stream->callbackInfo.usingCallback = false; + stream->callbackInfo.userData = NULL; + stream->state = STREAM_STOPPED; + stream->callbackInfo.callback = NULL; + + MUTEX_UNLOCK(&stream->mutex); + } +} + +void RtAudio :: closeStream(int streamId) +{ + // We don't want an exception to be thrown here because this + // function is called by our class destructor. So, do our own + // streamId check. + if ( streams.find( streamId ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtError::WARNING); + return; + } + + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + + AudioDeviceID id; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + id = devices[stream->device[0]].id[0]; + if (stream->state == STREAM_RUNNING) + AudioDeviceStop( id, callbackHandler ); + AudioDeviceRemoveIOProc( id, callbackHandler ); + } + + if (stream->mode == INPUT || ( stream->mode == DUPLEX && stream->device[0] != stream->device[1]) ) { + id = devices[stream->device[1]].id[0]; + if (stream->state == STREAM_RUNNING) + AudioDeviceStop( id, callbackHandler ); + AudioDeviceRemoveIOProc( id, callbackHandler ); + } + + pthread_mutex_destroy(&stream->mutex); + + if (stream->userBuffer) + free(stream->userBuffer); + + if ( stream->deInterleave[0] || stream->deInterleave[1] ) + free(stream->callbackInfo.buffers); + + free(stream); + streams.erase(streamId); +} + +void RtAudio :: startStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_RUNNING) + goto unlock; + + OSStatus err; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + err = AudioDeviceStart(devices[stream->device[0]].id[0], callbackHandler); + if (err != noErr) { + sprintf(message, "RtAudio: OSX error starting callback procedure on device (%s).", + devices[stream->device[0]].name); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + if (stream->mode == INPUT || ( stream->mode == DUPLEX && stream->device[0] != stream->device[1]) ) { + + err = AudioDeviceStart(devices[stream->device[1]].id[0], callbackHandler); + if (err != noErr) { + sprintf(message, "RtAudio: OSX error starting input callback procedure on device (%s).", + devices[stream->device[0]].name); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + stream->callbackInfo.streamId = streamId; + stream->state = STREAM_RUNNING; + stream->callbackInfo.blockTick = true; + stream->callbackInfo.stopStream = false; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: stopStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + OSStatus err; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + err = AudioDeviceStop(devices[stream->device[0]].id[0], callbackHandler); + if (err != noErr) { + sprintf(message, "RtAudio: OSX error stopping callback procedure on device (%s).", + devices[stream->device[0]].name); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + if (stream->mode == INPUT || ( stream->mode == DUPLEX && stream->device[0] != stream->device[1]) ) { + + err = AudioDeviceStop(devices[stream->device[1]].id[0], callbackHandler); + if (err != noErr) { + sprintf(message, "RtAudio: OSX error stopping input callback procedure on device (%s).", + devices[stream->device[0]].name); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: abortStream(int streamId) +{ + stopStream( streamId ); +} + +// I don't know how this function can be implemented. +int RtAudio :: streamWillBlock(int streamId) +{ + sprintf(message, "RtAudio: streamWillBlock() cannot be implemented for OS X."); + error(RtError::WARNING); + return 0; +} + +void RtAudio :: tickStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + if (stream->state == STREAM_STOPPED) + return; + + if (stream->callbackInfo.usingCallback) { + sprintf(message, "RtAudio: tickStream() should not be used when a callback function is set!"); + error(RtError::WARNING); + return; + } + + // Block waiting here until the user data is processed in callbackEvent(). + while ( stream->callbackInfo.blockTick ) + usleep(stream->callbackInfo.waitTime); + + MUTEX_LOCK(&stream->mutex); + + stream->callbackInfo.blockTick = true; + + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: callbackEvent( int streamId, DEVICE_ID deviceId, void *inData, void *outData ) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + CALLBACK_INFO *info; + AudioBufferList *inBufferList = (AudioBufferList *) inData; + AudioBufferList *outBufferList = (AudioBufferList *) outData; + + if (stream->state == STREAM_STOPPED) return; + + info = (CALLBACK_INFO *) &stream->callbackInfo; + if ( !info->usingCallback ) { + // Block waiting here until we get new user data in tickStream(). + while ( !info->blockTick ) + usleep(info->waitTime); + } + else if ( info->stopStream ) { + // Check if the stream should be stopped (via the previous user + // callback return value). We stop the stream here, rather than + // after the function call, so that output data can first be + // processed. + this->stopStream(info->streamId); + return; + } + + MUTEX_LOCK(&stream->mutex); + + if ( stream->mode == INPUT || ( stream->mode == DUPLEX && deviceId == info->device[1] ) ) { + + if (stream->doConvertBuffer[1]) { + + if ( stream->deInterleave[1] ) { + stream->deviceBuffer = (char *) stream->callbackInfo.buffers; + int bufferBytes = inBufferList->mBuffers[stream->handle[1]].mDataByteSize; + for ( int i=0; i<stream->nDeviceChannels[1]; i++ ) { + memcpy(&stream->deviceBuffer[i*bufferBytes], + inBufferList->mBuffers[stream->handle[1]+i].mData, bufferBytes ); + } + } + else + stream->deviceBuffer = (char *) inBufferList->mBuffers[stream->handle[1]].mData; + + if ( stream->doByteSwap[1] ) + byteSwapBuffer(stream->deviceBuffer, + stream->bufferSize * stream->nDeviceChannels[1], + stream->deviceFormat[1]); + convertStreamBuffer(stream, INPUT); + + } + else { + memcpy(stream->userBuffer, + inBufferList->mBuffers[stream->handle[1]].mData, + inBufferList->mBuffers[stream->handle[1]].mDataByteSize ); + + if (stream->doByteSwap[1]) + byteSwapBuffer(stream->userBuffer, + stream->bufferSize * stream->nUserChannels[1], + stream->userFormat); + } + } + + // Don't invoke the user callback if duplex mode, the input/output + // devices are different, and this function is called for the output + // device. + if ( info->usingCallback && (stream->mode != DUPLEX || deviceId == info->device[1] ) ) { + RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) info->callback; + info->stopStream = callback(stream->userBuffer, stream->bufferSize, info->userData); + } + + if ( stream->mode == OUTPUT || ( stream->mode == DUPLEX && deviceId == info->device[0] ) ) { + + if (stream->doConvertBuffer[0]) { + + if ( !stream->deInterleave[0] ) + stream->deviceBuffer = (char *) outBufferList->mBuffers[stream->handle[0]].mData; + else + stream->deviceBuffer = (char *) stream->callbackInfo.buffers; + + convertStreamBuffer(stream, OUTPUT); + if ( stream->doByteSwap[0] ) + byteSwapBuffer(stream->deviceBuffer, + stream->bufferSize * stream->nDeviceChannels[0], + stream->deviceFormat[0]); + + if ( stream->deInterleave[0] ) { + int bufferBytes = outBufferList->mBuffers[stream->handle[0]].mDataByteSize; + for ( int i=0; i<stream->nDeviceChannels[0]; i++ ) { + memcpy(outBufferList->mBuffers[stream->handle[0]+i].mData, + &stream->deviceBuffer[i*bufferBytes], bufferBytes ); + } + } + + } + else { + if (stream->doByteSwap[0]) + byteSwapBuffer(stream->userBuffer, + stream->bufferSize * stream->nUserChannels[0], + stream->userFormat); + + memcpy(outBufferList->mBuffers[stream->handle[0]].mData, + stream->userBuffer, + outBufferList->mBuffers[stream->handle[0]].mDataByteSize ); + } + } + + if ( !info->usingCallback && (stream->mode != DUPLEX || deviceId == info->device[1] ) ) + info->blockTick = false; + + MUTEX_UNLOCK(&stream->mutex); + +} + +void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + stream->callbackInfo.callback = (void *) callback; + stream->callbackInfo.userData = userData; + stream->callbackInfo.usingCallback = true; +} + +//******************** End of __MACOSX_CORE__ *********************// + +#elif defined(__LINUX_ALSA__) + +#define MAX_DEVICES 16 + +void RtAudio :: initialize(void) +{ + int card, result, device; + char name[32]; + const char *cardId; + char deviceNames[MAX_DEVICES][32]; + snd_ctl_t *handle; + snd_ctl_card_info_t *info; + snd_ctl_card_info_alloca(&info); + + // Count cards and devices + nDevices = 0; + card = -1; + snd_card_next(&card); + while ( card >= 0 ) { + sprintf(name, "hw:%d", card); + result = snd_ctl_open(&handle, name, 0); + if (result < 0) { + sprintf(message, "RtAudio: ALSA control open (%i): %s.", card, snd_strerror(result)); + error(RtError::DEBUG_WARNING); + goto next_card; + } + result = snd_ctl_card_info(handle, info); + if (result < 0) { + sprintf(message, "RtAudio: ALSA control hardware info (%i): %s.", card, snd_strerror(result)); + error(RtError::DEBUG_WARNING); + goto next_card; + } + cardId = snd_ctl_card_info_get_id(info); + device = -1; + while (1) { + result = snd_ctl_pcm_next_device(handle, &device); + if (result < 0) { + sprintf(message, "RtAudio: ALSA control next device (%i): %s.", card, snd_strerror(result)); + error(RtError::DEBUG_WARNING); + break; + } + if (device < 0) + break; + if ( strlen(cardId) ) + sprintf( deviceNames[nDevices++], "hw:%s,%d", cardId, device ); + else + sprintf( deviceNames[nDevices++], "hw:%d,%d", card, device ); + if ( nDevices > MAX_DEVICES ) break; + } + if ( nDevices > MAX_DEVICES ) break; + next_card: + snd_ctl_close(handle); + snd_card_next(&card); + } + + if (nDevices == 0) return; + + // Allocate the RTAUDIO_DEVICE structures. + devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); + if (devices == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + + // Write device ascii identifiers to device structures and then + // probe the device capabilities. + for (int i=0; i<nDevices; i++) { + strncpy(devices[i].name, deviceNames[i], 32); + //probeDeviceInfo(&devices[i]); + } +} + +int RtAudio :: getDefaultInputDevice(void) +{ + // No ALSA API functions for default devices. + return 0; +} + +int RtAudio :: getDefaultOutputDevice(void) +{ + // No ALSA API functions for default devices. + return 0; +} + +void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info) +{ + int err; + int open_mode = SND_PCM_ASYNC; + snd_pcm_t *handle; + snd_ctl_t *chandle; + snd_pcm_stream_t stream; + snd_pcm_info_t *pcminfo; + snd_pcm_info_alloca(&pcminfo); + snd_pcm_hw_params_t *params; + snd_pcm_hw_params_alloca(¶ms); + char name[32]; + char *card; + + // Open the control interface for this card. + strncpy( name, info->name, 32 ); + card = strtok(name, ","); + err = snd_ctl_open(&chandle, card, 0); + if (err < 0) { + sprintf(message, "RtAudio: ALSA control open (%s): %s.", card, snd_strerror(err)); + error(RtError::DEBUG_WARNING); + return; + } + unsigned int dev = (unsigned int) atoi( strtok(NULL, ",") ); + + // First try for playback + stream = SND_PCM_STREAM_PLAYBACK; + snd_pcm_info_set_device(pcminfo, dev); + snd_pcm_info_set_subdevice(pcminfo, 0); + snd_pcm_info_set_stream(pcminfo, stream); + + if ((err = snd_ctl_pcm_info(chandle, pcminfo)) < 0) { + if (err == -ENOENT) { + sprintf(message, "RtAudio: ALSA pcm device (%s) doesn't handle output!", info->name); + error(RtError::DEBUG_WARNING); + } + else { + sprintf(message, "RtAudio: ALSA snd_ctl_pcm_info error for device (%s) output: %s", + info->name, snd_strerror(err)); + error(RtError::DEBUG_WARNING); + } + goto capture_probe; + } + + err = snd_pcm_open(&handle, info->name, stream, open_mode | SND_PCM_NONBLOCK ); + if (err < 0) { + if ( err == EBUSY ) + sprintf(message, "RtAudio: ALSA pcm playback device (%s) is busy: %s.", + info->name, snd_strerror(err)); + else + sprintf(message, "RtAudio: ALSA pcm playback open (%s) error: %s.", + info->name, snd_strerror(err)); + error(RtError::DEBUG_WARNING); + goto capture_probe; + } + + // We have an open device ... allocate the parameter structure. + err = snd_pcm_hw_params_any(handle, params); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA hardware probe error (%s): %s.", + info->name, snd_strerror(err)); + error(RtError::WARNING); + goto capture_probe; + } + + // Get output channel information. + info->minOutputChannels = snd_pcm_hw_params_get_channels_min(params); + info->maxOutputChannels = snd_pcm_hw_params_get_channels_max(params); + + snd_pcm_close(handle); + + capture_probe: + // Now try for capture + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream(pcminfo, stream); + + err = snd_ctl_pcm_info(chandle, pcminfo); + snd_ctl_close(chandle); + if ( err < 0 ) { + if (err == -ENOENT) { + sprintf(message, "RtAudio: ALSA pcm device (%s) doesn't handle input!", info->name); + error(RtError::DEBUG_WARNING); + } + else { + sprintf(message, "RtAudio: ALSA snd_ctl_pcm_info error for device (%s) input: %s", + info->name, snd_strerror(err)); + error(RtError::DEBUG_WARNING); + } + if (info->maxOutputChannels == 0) + // didn't open for playback either ... device invalid + return; + goto probe_parameters; + } + + err = snd_pcm_open(&handle, info->name, stream, open_mode | SND_PCM_NONBLOCK); + if (err < 0) { + if ( err == EBUSY ) + sprintf(message, "RtAudio: ALSA pcm capture device (%s) is busy: %s.", + info->name, snd_strerror(err)); + else + sprintf(message, "RtAudio: ALSA pcm capture open (%s) error: %s.", + info->name, snd_strerror(err)); + error(RtError::DEBUG_WARNING); + if (info->maxOutputChannels == 0) + // didn't open for playback either ... device invalid + return; + goto probe_parameters; + } + + // We have an open capture device ... allocate the parameter structure. + err = snd_pcm_hw_params_any(handle, params); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA hardware probe error (%s): %s.", + info->name, snd_strerror(err)); + error(RtError::WARNING); + if (info->maxOutputChannels > 0) + goto probe_parameters; + else + return; + } + + // Get input channel information. + info->minInputChannels = snd_pcm_hw_params_get_channels_min(params); + info->maxInputChannels = snd_pcm_hw_params_get_channels_max(params); + + snd_pcm_close(handle); + + // If device opens for both playback and capture, we determine the channels. + if (info->maxOutputChannels == 0 || info->maxInputChannels == 0) + goto probe_parameters; + + info->hasDuplexSupport = true; + info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ? + info->maxInputChannels : info->maxOutputChannels; + info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ? + info->minInputChannels : info->minOutputChannels; + + probe_parameters: + // At this point, we just need to figure out the supported data + // formats and sample rates. We'll proceed by opening the device in + // the direction with the maximum number of channels, or playback if + // they are equal. This might limit our sample rate options, but so + // be it. + + if (info->maxOutputChannels >= info->maxInputChannels) + stream = SND_PCM_STREAM_PLAYBACK; + else + stream = SND_PCM_STREAM_CAPTURE; + + err = snd_pcm_open(&handle, info->name, stream, open_mode); + if (err < 0) { + sprintf(message, "RtAudio: ALSA pcm (%s) won't reopen during probe: %s.", + info->name, snd_strerror(err)); + error(RtError::WARNING); + return; + } + + // We have an open device ... allocate the parameter structure. + err = snd_pcm_hw_params_any(handle, params); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA hardware reopen probe error (%s): %s.", + info->name, snd_strerror(err)); + error(RtError::WARNING); + return; + } + + // Test a non-standard sample rate to see if continuous rate is supported. + int dir = 0; + if (snd_pcm_hw_params_test_rate(handle, params, 35500, dir) == 0) { + // It appears that continuous sample rate support is available. + info->nSampleRates = -1; + info->sampleRates[0] = snd_pcm_hw_params_get_rate_min(params, &dir); + info->sampleRates[1] = snd_pcm_hw_params_get_rate_max(params, &dir); + } + else { + // No continuous rate support ... test our discrete set of sample rate values. + info->nSampleRates = 0; + for (int i=0; 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(RtError::WARNING); + return; + } + + // That's all ... close the device and return + snd_pcm_close(handle); + info->probed = true; + return; +} + +bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, + STREAM_MODE mode, int channels, + int sampleRate, RTAUDIO_FORMAT format, + int *bufferSize, int numberOfBuffers) +{ +#if defined(__RTAUDIO_DEBUG__) + snd_output_t *out; + snd_output_stdio_attach(&out, stderr, 0); +#endif + + // I'm not using the "plug" interface ... too much inconsistent behavior. + const char *name = devices[device].name; + + snd_pcm_stream_t alsa_stream; + if (mode == OUTPUT) + alsa_stream = SND_PCM_STREAM_PLAYBACK; + else + alsa_stream = SND_PCM_STREAM_CAPTURE; + + int err; + snd_pcm_t *handle; + int alsa_open_mode = SND_PCM_ASYNC; + err = snd_pcm_open(&handle, name, alsa_stream, alsa_open_mode); + if (err < 0) { + sprintf(message,"RtAudio: ALSA pcm device (%s) won't open: %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + + // Fill the parameter structure. + snd_pcm_hw_params_t *hw_params; + snd_pcm_hw_params_alloca(&hw_params); + err = snd_pcm_hw_params_any(handle, hw_params); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error getting parameter handle (%s): %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf(stderr, "\nRtAudio: ALSA dump hardware params just after device open:\n\n"); + snd_pcm_hw_params_dump(hw_params, out); +#endif + + + // Set access ... try interleaved access first, then non-interleaved + if ( !snd_pcm_hw_params_test_access( handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED) ) { + err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); + } + else if ( !snd_pcm_hw_params_test_access( handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED) ) { + err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED); + stream->deInterleave[mode] = true; + } + else { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA device (%s) access not supported by RtAudio.", name); + error(RtError::WARNING); + return FAILURE; + } + + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error setting access ( (%s): %s.", name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + + // Determine how to set the device format. + stream->userFormat = format; + snd_pcm_format_t device_format; + + if (format == RTAUDIO_SINT8) + device_format = SND_PCM_FORMAT_S8; + else if (format == RTAUDIO_SINT16) + device_format = SND_PCM_FORMAT_S16; + else if (format == RTAUDIO_SINT24) + device_format = SND_PCM_FORMAT_S24; + else if (format == RTAUDIO_SINT32) + device_format = SND_PCM_FORMAT_S32; + else if (format == RTAUDIO_FLOAT32) + device_format = SND_PCM_FORMAT_FLOAT; + else if (format == RTAUDIO_FLOAT64) + device_format = SND_PCM_FORMAT_FLOAT64; + + if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { + stream->deviceFormat[mode] = format; + goto set_format; + } + + // The user requested format is not natively supported by the device. + device_format = SND_PCM_FORMAT_FLOAT64; + if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { + stream->deviceFormat[mode] = RTAUDIO_FLOAT64; + goto set_format; + } + + device_format = SND_PCM_FORMAT_FLOAT; + if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { + stream->deviceFormat[mode] = RTAUDIO_FLOAT32; + goto set_format; + } + + device_format = SND_PCM_FORMAT_S32; + if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { + stream->deviceFormat[mode] = RTAUDIO_SINT32; + goto set_format; + } + + device_format = SND_PCM_FORMAT_S24; + if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { + stream->deviceFormat[mode] = RTAUDIO_SINT24; + goto set_format; + } + + device_format = SND_PCM_FORMAT_S16; + if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { + stream->deviceFormat[mode] = RTAUDIO_SINT16; + goto set_format; + } + + device_format = SND_PCM_FORMAT_S8; + if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { + stream->deviceFormat[mode] = RTAUDIO_SINT8; + goto set_format; + } + + // If we get here, no supported format was found. + sprintf(message,"RtAudio: ALSA pcm device (%s) data format not supported by RtAudio.", name); + snd_pcm_close(handle); + error(RtError::WARNING); + return FAILURE; + + set_format: + err = snd_pcm_hw_params_set_format(handle, hw_params, device_format); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error setting format (%s): %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + + // Determine whether byte-swaping is necessary. + stream->doByteSwap[mode] = false; + if (device_format != SND_PCM_FORMAT_S8) { + err = snd_pcm_format_cpu_endian(device_format); + if (err == 0) + stream->doByteSwap[mode] = true; + else if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error getting format endian-ness (%s): %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + } + + // Set the sample rate. + err = snd_pcm_hw_params_set_rate(handle, hw_params, (unsigned int)sampleRate, 0); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error setting sample rate (%d) on device (%s): %s.", + sampleRate, name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + + // Determine the number of channels for this device. We support a possible + // minimum device channel number > than the value requested by the user. + stream->nUserChannels[mode] = channels; + int device_channels = snd_pcm_hw_params_get_channels_max(hw_params); + if (device_channels < channels) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: channels (%d) not supported by device (%s).", + channels, name); + error(RtError::WARNING); + return FAILURE; + } + + device_channels = snd_pcm_hw_params_get_channels_min(hw_params); + if (device_channels < channels) device_channels = channels; + stream->nDeviceChannels[mode] = device_channels; + + // Set the device channels. + err = snd_pcm_hw_params_set_channels(handle, hw_params, device_channels); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error setting channels (%d) on device (%s): %s.", + device_channels, name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + + // Set the buffer number, which in ALSA is referred to as the "period". + int dir; + int periods = numberOfBuffers; + // Even though the hardware might allow 1 buffer, it won't work reliably. + if (periods < 2) periods = 2; + err = snd_pcm_hw_params_get_periods_min(hw_params, &dir); + if (err > periods) periods = err; + err = snd_pcm_hw_params_get_periods_max(hw_params, &dir); + if (err < periods) periods = err; + + err = snd_pcm_hw_params_set_periods(handle, hw_params, periods, 0); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error setting periods (%s): %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + + // Set the buffer (or period) size. + err = snd_pcm_hw_params_get_period_size_min(hw_params, &dir); + if (err > *bufferSize) *bufferSize = err; + + err = snd_pcm_hw_params_set_period_size(handle, hw_params, *bufferSize, 0); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error setting period size (%s): %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + if ( stream->mode == OUTPUT && mode == INPUT && *bufferSize != stream->bufferSize ) { + sprintf( message, "RtAudio: ALSA error setting buffer size for duplex stream on device (%s).", + name ); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + stream->bufferSize = *bufferSize; + + // Install the hardware configuration + err = snd_pcm_hw_params(handle, hw_params); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error installing hardware configuration (%s): %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf(stderr, "\nRtAudio: ALSA dump hardware params after installation:\n\n"); + snd_pcm_hw_params_dump(hw_params, out); +#endif + + /* + // Install the software configuration + snd_pcm_sw_params_t *sw_params = NULL; + snd_pcm_sw_params_alloca(&sw_params); + snd_pcm_sw_params_current(handle, sw_params); + err = snd_pcm_sw_params(handle, sw_params); + if (err < 0) { + snd_pcm_close(handle); + sprintf(message, "RtAudio: ALSA error installing software configuration (%s): %s.", + name, snd_strerror(err)); + error(RtError::WARNING); + return FAILURE; + } + */ + + // Set handle and flags for buffer conversion + stream->handle[mode] = handle; + stream->doConvertBuffer[mode] = false; + if (stream->userFormat != stream->deviceFormat[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] > 1 && stream->deInterleave[mode]) + stream->doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { + + long buffer_bytes; + if (stream->nUserChannels[0] >= stream->nUserChannels[1]) + buffer_bytes = stream->nUserChannels[0]; + else + buffer_bytes = stream->nUserChannels[1]; + + buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); + if (stream->userBuffer) free(stream->userBuffer); + stream->userBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->userBuffer == NULL) + goto memory_error; + } + + if ( stream->doConvertBuffer[mode] ) { + + long buffer_bytes; + bool makeBuffer = true; + if ( mode == OUTPUT ) + buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + else { // mode == INPUT + buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); + if ( stream->mode == OUTPUT && stream->deviceBuffer ) { + long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + if ( buffer_bytes < bytes_out ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + buffer_bytes *= *bufferSize; + if (stream->deviceBuffer) free(stream->deviceBuffer); + stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->deviceBuffer == NULL) + goto memory_error; + } + } + + stream->device[mode] = device; + stream->state = STREAM_STOPPED; + if ( stream->mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream->mode = DUPLEX; + else + stream->mode = mode; + stream->nBuffers = periods; + stream->sampleRate = sampleRate; + + return SUCCESS; + + memory_error: + if (stream->handle[0]) { + snd_pcm_close(stream->handle[0]); + stream->handle[0] = 0; + } + if (stream->handle[1]) { + snd_pcm_close(stream->handle[1]); + stream->handle[1] = 0; + } + if (stream->userBuffer) { + free(stream->userBuffer); + stream->userBuffer = 0; + } + sprintf(message, "RtAudio: ALSA error allocating buffer memory (%s).", name); + error(RtError::WARNING); + return FAILURE; +} + +void RtAudio :: closeStream(int streamId) +{ + // We don't want an exception to be thrown here because this + // function is called by our class destructor. So, do our own + // streamId check. + if ( streams.find( streamId ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtError::WARNING); + return; + } + + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + + if (stream->callbackInfo.usingCallback) { + pthread_cancel(stream->callbackInfo.thread); + pthread_join(stream->callbackInfo.thread, NULL); + } + + if (stream->state == STREAM_RUNNING) { + if (stream->mode == OUTPUT || stream->mode == DUPLEX) + snd_pcm_drop(stream->handle[0]); + if (stream->mode == INPUT || stream->mode == DUPLEX) + snd_pcm_drop(stream->handle[1]); + } + + pthread_mutex_destroy(&stream->mutex); + + if (stream->handle[0]) + snd_pcm_close(stream->handle[0]); + + if (stream->handle[1]) + snd_pcm_close(stream->handle[1]); + + if (stream->userBuffer) + free(stream->userBuffer); + + if (stream->deviceBuffer) + free(stream->deviceBuffer); + + free(stream); + streams.erase(streamId); +} + +void RtAudio :: startStream(int streamId) +{ + // This method calls snd_pcm_prepare if the device isn't already in that state. + + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_RUNNING) + goto unlock; + + int err; + snd_pcm_state_t state; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + state = snd_pcm_state(stream->handle[0]); + if (state != SND_PCM_STATE_PREPARED) { + err = snd_pcm_prepare(stream->handle[0]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error preparing pcm device (%s): %s.", + devices[stream->device[0]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + state = snd_pcm_state(stream->handle[1]); + if (state != SND_PCM_STATE_PREPARED) { + err = snd_pcm_prepare(stream->handle[1]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error preparing pcm device (%s): %s.", + devices[stream->device[1]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + } + stream->state = STREAM_RUNNING; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: stopStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + int err; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + err = snd_pcm_drain(stream->handle[0]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", + devices[stream->device[0]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + err = snd_pcm_drain(stream->handle[1]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", + devices[stream->device[1]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: abortStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + int err; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + err = snd_pcm_drop(stream->handle[0]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", + devices[stream->device[0]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + err = snd_pcm_drop(stream->handle[1]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", + devices[stream->device[1]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +int RtAudio :: streamWillBlock(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + int err = 0, frames = 0; + if (stream->state == STREAM_STOPPED) + goto unlock; + + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + err = snd_pcm_avail_update(stream->handle[0]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error getting available frames for device (%s): %s.", + devices[stream->device[0]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + frames = err; + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + err = snd_pcm_avail_update(stream->handle[1]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error getting available frames for device (%s): %s.", + devices[stream->device[1]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + if (frames > err) frames = err; + } + + frames = stream->bufferSize - frames; + if (frames < 0) frames = 0; + + unlock: + MUTEX_UNLOCK(&stream->mutex); + return frames; +} + +void RtAudio :: tickStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + int stopStream = 0; + if (stream->state == STREAM_STOPPED) { + if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds + return; + } + else if (stream->callbackInfo.usingCallback) { + RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback; + stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData); + } + + MUTEX_LOCK(&stream->mutex); + + // The state might change while waiting on a mutex. + if (stream->state == STREAM_STOPPED) + goto unlock; + + int err; + char *buffer; + int channels; + RTAUDIO_FORMAT format; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + // Setup parameters and do buffer conversion if necessary. + if (stream->doConvertBuffer[0]) { + convertStreamBuffer(stream, OUTPUT); + buffer = stream->deviceBuffer; + channels = stream->nDeviceChannels[0]; + format = stream->deviceFormat[0]; + } + else { + buffer = stream->userBuffer; + channels = stream->nUserChannels[0]; + format = stream->userFormat; + } + + // Do byte swapping if necessary. + if (stream->doByteSwap[0]) + byteSwapBuffer(buffer, stream->bufferSize * channels, format); + + // Write samples to device in interleaved/non-interleaved format. + if (stream->deInterleave[0]) { + void *bufs[channels]; + size_t offset = stream->bufferSize * formatBytes(format); + for (int i=0; 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(RtError::WARNING); + err = snd_pcm_prepare(stream->handle[0]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error preparing handle after underrun: %s.", + snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + else { + sprintf(message, "RtAudio: ALSA error, current state is %s.", + snd_pcm_state_name(state)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + goto unlock; + } + else { + sprintf(message, "RtAudio: ALSA audio write error for device (%s): %s.", + devices[stream->device[0]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + + // Setup parameters. + if (stream->doConvertBuffer[1]) { + buffer = stream->deviceBuffer; + channels = stream->nDeviceChannels[1]; + format = stream->deviceFormat[1]; + } + else { + buffer = stream->userBuffer; + channels = stream->nUserChannels[1]; + format = stream->userFormat; + } + + // Read samples from device in interleaved/non-interleaved format. + if (stream->deInterleave[1]) { + void *bufs[channels]; + size_t offset = stream->bufferSize * formatBytes(format); + for (int i=0; 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(RtError::WARNING); + err = snd_pcm_prepare(stream->handle[1]); + if (err < 0) { + sprintf(message, "RtAudio: ALSA error preparing handle after overrun: %s.", + snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + else { + sprintf(message, "RtAudio: ALSA error, current state is %s.", + snd_pcm_state_name(state)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + goto unlock; + } + else { + sprintf(message, "RtAudio: ALSA audio read error for device (%s): %s.", + devices[stream->device[1]].name, snd_strerror(err)); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + } + + // Do byte swapping if necessary. + if (stream->doByteSwap[1]) + byteSwapBuffer(buffer, stream->bufferSize * channels, format); + + // Do buffer conversion if necessary. + if (stream->doConvertBuffer[1]) + convertStreamBuffer(stream, INPUT); + } + + unlock: + MUTEX_UNLOCK(&stream->mutex); + + if (stream->callbackInfo.usingCallback && stopStream) + this->stopStream(streamId); +} + +extern "C" void *callbackHandler(void *ptr) +{ + CALLBACK_INFO *info = (CALLBACK_INFO *) ptr; + RtAudio *object = (RtAudio *) info->object; + int stream = info->streamId; + bool *usingCallback = &info->usingCallback; + + while ( *usingCallback ) { + pthread_testcancel(); + try { + object->tickStream(stream); + } + catch (RtError &exception) { + fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n", + exception.getMessage()); + break; + } + } + + return 0; +} + +//******************** End of __LINUX_ALSA__ *********************// + +#elif defined(__LINUX_OSS__) + +#include <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 many cases where /dev/dsp was a + // real device, so we need to check for that. Also, sometimes the + // link is to /dev/dspx and other times just dspx. I'm not sure how + // the latter works, but it does. + char device_name[16]; + struct stat dspstat; + int dsplink = -1; + int i = 0; + if (lstat(DAC_NAME, &dspstat) == 0) { + if (S_ISLNK(dspstat.st_mode)) { + i = readlink(DAC_NAME, device_name, sizeof(device_name)); + if (i > 0) { + device_name[i] = '\0'; + if (i > 8) { // check for "/dev/dspx" + if (!strncmp(DAC_NAME, device_name, 8)) + dsplink = atoi(&device_name[8]); + } + else if (i > 3) { // check for "dspx" + if (!strncmp("dsp", device_name, 3)) + dsplink = atoi(&device_name[3]); + } + } + else { + sprintf(message, "RtAudio: cannot read value of symbolic link %s.", DAC_NAME); + error(RtError::SYSTEM_ERROR); + } + } + } + else { + sprintf(message, "RtAudio: cannot stat %s.", DAC_NAME); + error(RtError::SYSTEM_ERROR); + } + + // The OSS API doesn't provide a routine for determining the number + // of devices. Thus, we'll just pursue a brute force method. The + // idea is to start with /dev/dsp(0) and continue with higher device + // numbers until we reach MAX_DSP_DEVICES. This should tell us how + // many devices we have ... it is not a fullproof scheme, but hopefully + // it will work most of the time. + + int fd = 0; + char names[MAX_DEVICES][16]; + for (i=-1; i<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(RtError::WARNING); + // still count it for now + } + } + } + else { + sprintf(message, "RtAudio: OSS playback device (%s) is busy.", device_name); + error(RtError::WARNING); + // still count it for now + } + } + + if (fd >= 0) close(fd); + strncpy(names[nDevices], device_name, 16); + nDevices++; + } + + if (nDevices == 0) return; + + // Allocate the RTAUDIO_DEVICE structures. + devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); + if (devices == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + + // Write device ascii identifiers to device control structure and then probe capabilities. + for (i=0; i<nDevices; i++) { + strncpy(devices[i].name, names[i], 16); + //probeDeviceInfo(&devices[i]); + } + + return; +} + +int RtAudio :: getDefaultInputDevice(void) +{ + // No OSS API functions for default devices. + return 0; +} + +int RtAudio :: getDefaultOutputDevice(void) +{ + // No OSS API functions for default devices. + return 0; +} + +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(RtError::DEBUG_WARNING); + goto capture_probe; + } + + // We have an open device ... see how many channels it can handle + for (i=MAX_CHANNELS; i>0; i--) { + channels = i; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) { + // This would normally indicate some sort of hardware error, but under ALSA's + // OSS emulation, it sometimes indicates an invalid channel value. Further, + // the returned channel value is not changed. So, we'll ignore the possible + // hardware error. + continue; // try next channel number + } + // Check to see whether the device supports the requested number of channels + if (channels != i ) continue; // try next channel number + // If here, we found the largest working channel value + break; + } + info->maxOutputChannels = i; + + // Now find the minimum number of channels it can handle + for (i=1; i<=info->maxOutputChannels; i++) { + channels = i; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) + continue; // try next channel number + // If here, we found the smallest working channel value + break; + } + info->minOutputChannels = i; + close(fd); + + capture_probe: + // Now try for capture + fd = open(info->name, O_RDONLY | O_NONBLOCK); + if (fd == -1) { + // Open device for capture failed ... either busy or doesn't exist + if (errno == EBUSY || errno == EAGAIN) + sprintf(message, "RtAudio: OSS capture device (%s) is busy and cannot be probed.", + info->name); + else + sprintf(message, "RtAudio: OSS capture device (%s) open error.", info->name); + error(RtError::DEBUG_WARNING); + if (info->maxOutputChannels == 0) + // didn't open for playback either ... device invalid + return; + goto probe_parameters; + } + + // We have the device open for capture ... see how many channels it can handle + for (i=MAX_CHANNELS; i>0; i--) { + channels = i; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) { + continue; // as above + } + // If here, we found a working channel value + break; + } + info->maxInputChannels = i; + + // Now find the minimum number of channels it can handle + for (i=1; i<=info->maxInputChannels; i++) { + channels = i; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) + continue; // try next channel number + // If here, we found the smallest working channel value + break; + } + info->minInputChannels = i; + close(fd); + + if (info->maxOutputChannels == 0 && info->maxInputChannels == 0) { + sprintf(message, "RtAudio: OSS device (%s) reports zero channels for input and output.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + // If device opens for both playback and capture, we determine the channels. + if (info->maxOutputChannels == 0 || info->maxInputChannels == 0) + goto probe_parameters; + + fd = open(info->name, O_RDWR | O_NONBLOCK); + if (fd == -1) + goto probe_parameters; + + ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0); + ioctl(fd, SNDCTL_DSP_GETCAPS, &mask); + if (mask & DSP_CAP_DUPLEX) { + info->hasDuplexSupport = true; + // We have the device open for duplex ... see how many channels it can handle + for (i=MAX_CHANNELS; i>0; i--) { + channels = i; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) + continue; // as above + // If here, we found a working channel value + break; + } + info->maxDuplexChannels = i; + + // Now find the minimum number of channels it can handle + for (i=1; i<=info->maxDuplexChannels; i++) { + channels = i; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) + continue; // try next channel number + // If here, we found the smallest working channel value + break; + } + info->minDuplexChannels = i; + } + close(fd); + + probe_parameters: + // At this point, we need to figure out the supported data formats + // and sample rates. We'll proceed by openning the device in the + // direction with the maximum number of channels, or playback if + // they are equal. This might limit our sample rate options, but so + // be it. + + if (info->maxOutputChannels >= info->maxInputChannels) { + fd = open(info->name, O_WRONLY | O_NONBLOCK); + channels = info->maxOutputChannels; + } + else { + fd = open(info->name, O_RDONLY | O_NONBLOCK); + channels = info->maxInputChannels; + } + + if (fd == -1) { + // We've got some sort of conflict ... abort + sprintf(message, "RtAudio: OSS device (%s) won't reopen during probe.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + // We have an open device ... set to maximum channels. + i = channels; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) { + // We've got some sort of conflict ... abort + close(fd); + sprintf(message, "RtAudio: OSS device (%s) won't revert to previous channel setting.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) { + close(fd); + sprintf(message, "RtAudio: OSS device (%s) can't get supported audio formats.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + // Probe the supported data formats ... we don't care about endian-ness just yet. + int format; + info->nativeFormats = 0; +#if defined (AFMT_S32_BE) + // This format does not seem to be in the 2.4 kernel version of OSS soundcard.h + if (mask & AFMT_S32_BE) { + format = AFMT_S32_BE; + info->nativeFormats |= RTAUDIO_SINT32; + } +#endif +#if defined (AFMT_S32_LE) + /* This format is not in the 2.4.4 kernel version of OSS soundcard.h */ + if (mask & AFMT_S32_LE) { + format = AFMT_S32_LE; + info->nativeFormats |= RTAUDIO_SINT32; + } +#endif + if (mask & AFMT_S8) { + format = AFMT_S8; + info->nativeFormats |= RTAUDIO_SINT8; + } + if (mask & AFMT_S16_BE) { + format = AFMT_S16_BE; + info->nativeFormats |= RTAUDIO_SINT16; + } + if (mask & AFMT_S16_LE) { + format = AFMT_S16_LE; + info->nativeFormats |= RTAUDIO_SINT16; + } + + // Check that we have at least one supported format + if (info->nativeFormats == 0) { + close(fd); + sprintf(message, "RtAudio: OSS device (%s) data format not supported by RtAudio.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + // Set the format + i = format; + if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1 || format != i) { + close(fd); + sprintf(message, "RtAudio: OSS device (%s) error setting data format.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + // Probe the supported sample rates ... first get lower limit + int speed = 1; + if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { + // If we get here, we're probably using an ALSA driver with OSS-emulation, + // which doesn't conform to the OSS specification. In this case, + // we'll probe our predefined list of sample rates for working values. + info->nSampleRates = 0; + for (i=0; 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(RtError::DEBUG_WARNING); + return; + } + info->sampleRates[1] = speed; + info->nSampleRates = -1; + + finished: // That's all ... close the device and return + close(fd); + info->probed = true; + return; +} + +bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, + STREAM_MODE mode, int channels, + int sampleRate, RTAUDIO_FORMAT format, + int *bufferSize, int numberOfBuffers) +{ + int buffers, buffer_bytes, device_channels, device_format; + int srate, temp, fd; + + const char *name = devices[device].name; + + if (mode == OUTPUT) + fd = open(name, O_WRONLY | O_NONBLOCK); + else { // mode == INPUT + if (stream->mode == OUTPUT && stream->device[0] == device) { + // We just set the same device for playback ... close and reopen for duplex (OSS only). + close(stream->handle[0]); + stream->handle[0] = 0; + // First check that the number previously set channels is the same. + if (stream->nUserChannels[0] != channels) { + sprintf(message, "RtAudio: input/output channels must be equal for OSS duplex device (%s).", name); + goto error; + } + fd = open(name, O_RDWR | O_NONBLOCK); + } + else + fd = open(name, O_RDONLY | O_NONBLOCK); + } + + if (fd == -1) { + if (errno == EBUSY || errno == EAGAIN) + sprintf(message, "RtAudio: OSS device (%s) is busy and cannot be opened.", + name); + else + sprintf(message, "RtAudio: OSS device (%s) cannot be opened.", name); + goto error; + } + + // Now reopen in blocking mode. + close(fd); + if (mode == OUTPUT) + fd = open(name, O_WRONLY | O_SYNC); + else { // mode == INPUT + if (stream->mode == OUTPUT && stream->device[0] == device) + fd = open(name, O_RDWR | O_SYNC); + else + fd = open(name, O_RDONLY | O_SYNC); + } + + if (fd == -1) { + sprintf(message, "RtAudio: OSS device (%s) cannot be opened.", name); + goto error; + } + + // Get the sample format mask + int mask; + if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) { + close(fd); + sprintf(message, "RtAudio: OSS device (%s) can't get supported audio formats.", + name); + goto error; + } + + // Determine how to set the device format. + stream->userFormat = format; + device_format = -1; + stream->doByteSwap[mode] = false; + if (format == RTAUDIO_SINT8) { + if (mask & AFMT_S8) { + device_format = AFMT_S8; + stream->deviceFormat[mode] = RTAUDIO_SINT8; + } + } + else if (format == RTAUDIO_SINT16) { + if (mask & AFMT_S16_NE) { + device_format = AFMT_S16_NE; + stream->deviceFormat[mode] = RTAUDIO_SINT16; + } +#if BYTE_ORDER == LITTLE_ENDIAN + else if (mask & AFMT_S16_BE) { + device_format = AFMT_S16_BE; + stream->deviceFormat[mode] = RTAUDIO_SINT16; + stream->doByteSwap[mode] = true; + } +#else + else if (mask & AFMT_S16_LE) { + device_format = AFMT_S16_LE; + stream->deviceFormat[mode] = RTAUDIO_SINT16; + stream->doByteSwap[mode] = true; + } +#endif + } +#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE) + else if (format == RTAUDIO_SINT32) { + if (mask & AFMT_S32_NE) { + device_format = AFMT_S32_NE; + stream->deviceFormat[mode] = RTAUDIO_SINT32; + } +#if BYTE_ORDER == LITTLE_ENDIAN + else if (mask & AFMT_S32_BE) { + device_format = AFMT_S32_BE; + stream->deviceFormat[mode] = RTAUDIO_SINT32; + stream->doByteSwap[mode] = true; + } +#else + else if (mask & AFMT_S32_LE) { + device_format = AFMT_S32_LE; + stream->deviceFormat[mode] = RTAUDIO_SINT32; + stream->doByteSwap[mode] = true; + } +#endif + } +#endif + + if (device_format == -1) { + // The user requested format is not natively supported by the device. + if (mask & AFMT_S16_NE) { + device_format = AFMT_S16_NE; + stream->deviceFormat[mode] = RTAUDIO_SINT16; + } +#if BYTE_ORDER == LITTLE_ENDIAN + else if (mask & AFMT_S16_BE) { + device_format = AFMT_S16_BE; + stream->deviceFormat[mode] = RTAUDIO_SINT16; + stream->doByteSwap[mode] = true; + } +#else + else if (mask & AFMT_S16_LE) { + device_format = AFMT_S16_LE; + stream->deviceFormat[mode] = RTAUDIO_SINT16; + stream->doByteSwap[mode] = true; + } +#endif +#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE) + else if (mask & AFMT_S32_NE) { + device_format = AFMT_S32_NE; + stream->deviceFormat[mode] = RTAUDIO_SINT32; + } +#if BYTE_ORDER == LITTLE_ENDIAN + else if (mask & AFMT_S32_BE) { + device_format = AFMT_S32_BE; + stream->deviceFormat[mode] = RTAUDIO_SINT32; + stream->doByteSwap[mode] = true; + } +#else + else if (mask & AFMT_S32_LE) { + device_format = AFMT_S32_LE; + stream->deviceFormat[mode] = RTAUDIO_SINT32; + stream->doByteSwap[mode] = true; + } +#endif +#endif + else if (mask & AFMT_S8) { + device_format = AFMT_S8; + stream->deviceFormat[mode] = RTAUDIO_SINT8; + } + } + + if (stream->deviceFormat[mode] == 0) { + // This really shouldn't happen ... + close(fd); + sprintf(message, "RtAudio: OSS device (%s) data format not supported by RtAudio.", + name); + goto error; + } + + // Determine the number of channels for this device. Note that the + // channel value requested by the user might be < min_X_Channels. + stream->nUserChannels[mode] = channels; + device_channels = channels; + if (mode == OUTPUT) { + if (channels < devices[device].minOutputChannels) + device_channels = devices[device].minOutputChannels; + } + else { // mode == INPUT + if (stream->mode == OUTPUT && stream->device[0] == device) { + // We're doing duplex setup here. + if (channels < devices[device].minDuplexChannels) + device_channels = devices[device].minDuplexChannels; + } + else { + if (channels < devices[device].minInputChannels) + device_channels = devices[device].minInputChannels; + } + } + stream->nDeviceChannels[mode] = device_channels; + + // Attempt to set the buffer size. According to OSS, the minimum + // number of buffers is two. The supposed minimum buffer size is 16 + // bytes, so that will be our lower bound. The argument to this + // call is in the form 0xMMMMSSSS (hex), where the buffer size (in + // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. + // We'll check the actual value used near the end of the setup + // procedure. + buffer_bytes = *bufferSize * formatBytes(stream->deviceFormat[mode]) * device_channels; + if (buffer_bytes < 16) buffer_bytes = 16; + buffers = numberOfBuffers; + if (buffers < 2) buffers = 2; + temp = ((int) buffers << 16) + (int)(log10((double)buffer_bytes)/log10(2.0)); + if (ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &temp)) { + close(fd); + sprintf(message, "RtAudio: OSS error setting fragment size for device (%s).", + name); + goto error; + } + stream->nBuffers = buffers; + + // Set the data format. + temp = device_format; + if (ioctl(fd, SNDCTL_DSP_SETFMT, &device_format) == -1 || device_format != temp) { + close(fd); + sprintf(message, "RtAudio: OSS error setting data format for device (%s).", + name); + goto error; + } + + // Set the number of channels. + temp = device_channels; + if (ioctl(fd, SNDCTL_DSP_CHANNELS, &device_channels) == -1 || device_channels != temp) { + close(fd); + sprintf(message, "RtAudio: OSS error setting %d channels on device (%s).", + temp, name); + goto error; + } + + // Set the sample rate. + srate = sampleRate; + temp = srate; + if (ioctl(fd, SNDCTL_DSP_SPEED, &srate) == -1) { + close(fd); + sprintf(message, "RtAudio: OSS error setting sample rate = %d on device (%s).", + temp, name); + goto error; + } + + // Verify the sample rate setup worked. + if (abs(srate - temp) > 100) { + close(fd); + sprintf(message, "RtAudio: OSS error ... audio device (%s) doesn't support sample rate of %d.", + name, temp); + goto error; + } + stream->sampleRate = sampleRate; + + if (ioctl(fd, SNDCTL_DSP_GETBLKSIZE, &buffer_bytes) == -1) { + close(fd); + sprintf(message, "RtAudio: OSS error getting buffer size for device (%s).", + name); + goto error; + } + + // Save buffer size (in sample frames). + *bufferSize = buffer_bytes / (formatBytes(stream->deviceFormat[mode]) * device_channels); + stream->bufferSize = *bufferSize; + + if (mode == INPUT && stream->mode == OUTPUT && + stream->device[0] == device) { + // We're doing duplex setup here. + stream->deviceFormat[0] = stream->deviceFormat[1]; + stream->nDeviceChannels[0] = device_channels; + } + + // Set flags for buffer conversion + stream->doConvertBuffer[mode] = false; + if (stream->userFormat != stream->deviceFormat[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode]) + stream->doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { + + long buffer_bytes; + if (stream->nUserChannels[0] >= stream->nUserChannels[1]) + buffer_bytes = stream->nUserChannels[0]; + else + buffer_bytes = stream->nUserChannels[1]; + + buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); + if (stream->userBuffer) free(stream->userBuffer); + stream->userBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->userBuffer == NULL) { + close(fd); + sprintf(message, "RtAudio: OSS error allocating user buffer memory (%s).", + name); + goto error; + } + } + + if ( stream->doConvertBuffer[mode] ) { + + long buffer_bytes; + bool makeBuffer = true; + if ( mode == OUTPUT ) + buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + else { // mode == INPUT + buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); + if ( stream->mode == OUTPUT && stream->deviceBuffer ) { + long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + if ( buffer_bytes < bytes_out ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + buffer_bytes *= *bufferSize; + if (stream->deviceBuffer) free(stream->deviceBuffer); + stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->deviceBuffer == NULL) { + close(fd); + free(stream->userBuffer); + sprintf(message, "RtAudio: OSS error allocating device buffer memory (%s).", + name); + goto error; + } + } + } + + stream->device[mode] = device; + stream->handle[mode] = fd; + stream->state = STREAM_STOPPED; + if ( stream->mode == OUTPUT && mode == INPUT ) { + stream->mode = DUPLEX; + if (stream->device[0] == device) + stream->handle[0] = fd; + } + else + stream->mode = mode; + + return SUCCESS; + + error: + if (stream->handle[0]) { + close(stream->handle[0]); + stream->handle[0] = 0; + } + error(RtError::WARNING); + return FAILURE; +} + +void RtAudio :: closeStream(int streamId) +{ + // We don't want an exception to be thrown here because this + // function is called by our class destructor. So, do our own + // streamId check. + if ( streams.find( streamId ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtError::WARNING); + return; + } + + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + + if (stream->callbackInfo.usingCallback) { + pthread_cancel(stream->callbackInfo.thread); + pthread_join(stream->callbackInfo.thread, NULL); + } + + if (stream->state == STREAM_RUNNING) { + if (stream->mode == OUTPUT || stream->mode == DUPLEX) + ioctl(stream->handle[0], SNDCTL_DSP_RESET, 0); + if (stream->mode == INPUT || stream->mode == DUPLEX) + ioctl(stream->handle[1], SNDCTL_DSP_RESET, 0); + } + + pthread_mutex_destroy(&stream->mutex); + + if (stream->handle[0]) + close(stream->handle[0]); + + if (stream->handle[1]) + close(stream->handle[1]); + + if (stream->userBuffer) + free(stream->userBuffer); + + if (stream->deviceBuffer) + free(stream->deviceBuffer); + + free(stream); + streams.erase(streamId); +} + +void RtAudio :: startStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + stream->state = STREAM_RUNNING; + + // No need to do anything else here ... OSS automatically starts + // when fed samples. + + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: stopStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + int err; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + err = ioctl(stream->handle[0], SNDCTL_DSP_SYNC, 0); + if (err < -1) { + sprintf(message, "RtAudio: OSS error stopping device (%s).", + devices[stream->device[0]].name); + error(RtError::DRIVER_ERROR); + } + } + else { + err = ioctl(stream->handle[1], SNDCTL_DSP_SYNC, 0); + if (err < -1) { + sprintf(message, "RtAudio: OSS error stopping device (%s).", + devices[stream->device[1]].name); + error(RtError::DRIVER_ERROR); + } + } + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: abortStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + int err; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + err = ioctl(stream->handle[0], SNDCTL_DSP_RESET, 0); + if (err < -1) { + sprintf(message, "RtAudio: OSS error aborting device (%s).", + devices[stream->device[0]].name); + error(RtError::DRIVER_ERROR); + } + } + else { + err = ioctl(stream->handle[1], SNDCTL_DSP_RESET, 0); + if (err < -1) { + sprintf(message, "RtAudio: OSS error aborting device (%s).", + devices[stream->device[1]].name); + error(RtError::DRIVER_ERROR); + } + } + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +int RtAudio :: streamWillBlock(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + int bytes = 0, channels = 0, frames = 0; + if (stream->state == STREAM_STOPPED) + goto unlock; + + audio_buf_info info; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + ioctl(stream->handle[0], SNDCTL_DSP_GETOSPACE, &info); + bytes = info.bytes; + channels = stream->nDeviceChannels[0]; + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + ioctl(stream->handle[1], SNDCTL_DSP_GETISPACE, &info); + if (stream->mode == DUPLEX ) { + bytes = (bytes < info.bytes) ? bytes : info.bytes; + channels = stream->nDeviceChannels[0]; + } + else { + bytes = info.bytes; + channels = stream->nDeviceChannels[1]; + } + } + + frames = (int) (bytes / (channels * formatBytes(stream->deviceFormat[0]))); + frames -= stream->bufferSize; + if (frames < 0) frames = 0; + + unlock: + MUTEX_UNLOCK(&stream->mutex); + return frames; +} + +void RtAudio :: tickStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + int stopStream = 0; + if (stream->state == STREAM_STOPPED) { + if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds + return; + } + else if (stream->callbackInfo.usingCallback) { + RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback; + stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData); + } + + MUTEX_LOCK(&stream->mutex); + + // The state might change while waiting on a mutex. + if (stream->state == STREAM_STOPPED) + goto unlock; + + int result; + char *buffer; + int samples; + RTAUDIO_FORMAT format; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + // Setup parameters and do buffer conversion if necessary. + if (stream->doConvertBuffer[0]) { + convertStreamBuffer(stream, OUTPUT); + buffer = stream->deviceBuffer; + samples = stream->bufferSize * stream->nDeviceChannels[0]; + format = stream->deviceFormat[0]; + } + else { + buffer = stream->userBuffer; + samples = stream->bufferSize * stream->nUserChannels[0]; + format = stream->userFormat; + } + + // Do byte swapping if necessary. + if (stream->doByteSwap[0]) + byteSwapBuffer(buffer, samples, format); + + // Write samples to device. + result = write(stream->handle[0], buffer, samples * formatBytes(format)); + + if (result == -1) { + // This could be an underrun, but the basic OSS API doesn't provide a means for determining that. + sprintf(message, "RtAudio: OSS audio write error for device (%s).", + devices[stream->device[0]].name); + error(RtError::DRIVER_ERROR); + } + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + + // Setup parameters. + if (stream->doConvertBuffer[1]) { + buffer = stream->deviceBuffer; + samples = stream->bufferSize * stream->nDeviceChannels[1]; + format = stream->deviceFormat[1]; + } + else { + buffer = stream->userBuffer; + samples = stream->bufferSize * stream->nUserChannels[1]; + format = stream->userFormat; + } + + // Read samples from device. + result = read(stream->handle[1], buffer, samples * formatBytes(format)); + + if (result == -1) { + // This could be an overrun, but the basic OSS API doesn't provide a means for determining that. + sprintf(message, "RtAudio: OSS audio read error for device (%s).", + devices[stream->device[1]].name); + error(RtError::DRIVER_ERROR); + } + + // Do byte swapping if necessary. + if (stream->doByteSwap[1]) + byteSwapBuffer(buffer, samples, format); + + // Do buffer conversion if necessary. + if (stream->doConvertBuffer[1]) + convertStreamBuffer(stream, INPUT); + } + + unlock: + MUTEX_UNLOCK(&stream->mutex); + + if (stream->callbackInfo.usingCallback && stopStream) + this->stopStream(streamId); +} + +extern "C" void *callbackHandler(void *ptr) +{ + CALLBACK_INFO *info = (CALLBACK_INFO *) ptr; + RtAudio *object = (RtAudio *) info->object; + int stream = info->streamId; + bool *usingCallback = &info->usingCallback; + + while ( *usingCallback ) { + pthread_testcancel(); + try { + object->tickStream(stream); + } + catch (RtError &exception) { + fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n", + exception.getMessage()); + break; + } + } + + return 0; +} + + +//******************** End of __LINUX_OSS__ *********************// + +#elif defined(__WINDOWS_ASIO__) // ASIO API on Windows + +// The ASIO API is designed around a callback scheme, so this +// implementation is similar to that used for OS X CoreAudio. The +// primary constraint with ASIO is that it only allows access to a +// single driver at a time. Thus, it is not possible to have more +// than one simultaneous RtAudio stream. +// +// This implementation also requires a number of external ASIO files +// and a few global variables. The ASIO callback scheme does not +// allow for the passing of user data, so we must create a global +// pointer to our callbackInfo structure. + +#include "asio/asiosys.h" +#include "asio/asio.h" +#include "asio/asiodrivers.h" +#include <math.h> + +AsioDrivers drivers; +ASIOCallbacks asioCallbacks; +CALLBACK_INFO *asioCallbackInfo; +ASIODriverInfo driverInfo; + +void RtAudio :: initialize(void) +{ + nDevices = drivers.asioGetNumDev(); + if (nDevices <= 0) return; + + // Allocate the RTAUDIO_DEVICE structures. + devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); + if (devices == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + + // Write device driver names to device structures and then probe the + // device capabilities. + for (int i=0; i<nDevices; i++) { + if ( drivers.asioGetDriverName( i, devices[i].name, 128 ) == 0 ) + //probeDeviceInfo(&devices[i]); + ; + else { + sprintf(message, "RtAudio: error getting ASIO driver name for device index %d!", i); + error(RtError::WARNING); + } + } + + drivers.removeCurrentDriver(); + driverInfo.asioVersion = 2; + // See note in DirectSound implementation about GetDesktopWindow(). + driverInfo.sysRef = GetForegroundWindow(); +} + +int RtAudio :: getDefaultInputDevice(void) +{ + return 0; +} + +int RtAudio :: getDefaultOutputDevice(void) +{ + return 0; +} + +void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info) +{ + // Don't probe if a stream is already open. + if ( streams.size() > 0 ) { + sprintf(message, "RtAudio: unable to probe ASIO driver while a stream is open."); + error(RtError::DEBUG_WARNING); + return; + } + + if ( !drivers.loadDriver( info->name ) ) { + sprintf(message, "RtAudio: ASIO error loading driver (%s).", info->name); + error(RtError::DEBUG_WARNING); + return; + } + + ASIOError result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + char details[32]; + if ( result == ASE_HWMalfunction ) + sprintf(details, "hardware malfunction"); + else if ( result == ASE_NoMemory ) + sprintf(details, "no memory"); + else if ( result == ASE_NotPresent ) + sprintf(details, "driver/hardware not present"); + else + sprintf(details, "unspecified"); + sprintf(message, "RtAudio: ASIO error (%s) initializing driver (%s).", details, info->name); + error(RtError::DEBUG_WARNING); + return; + } + + // Determine the device channel information. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO error getting input/output channel count (%s).", info->name); + error(RtError::DEBUG_WARNING); + return; + } + + info->maxOutputChannels = outputChannels; + if ( outputChannels > 0 ) info->minOutputChannels = 1; + + info->maxInputChannels = inputChannels; + if ( inputChannels > 0 ) info->minInputChannels = 1; + + // If device opens for both playback and capture, we determine the channels. + if (info->maxOutputChannels > 0 && info->maxInputChannels > 0) { + info->hasDuplexSupport = true; + info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ? + info->maxInputChannels : info->maxOutputChannels; + info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ? + info->minInputChannels : info->minOutputChannels; + } + + // Determine the supported sample rates. + info->nSampleRates = 0; + for (int i=0; i<MAX_SAMPLE_RATES; i++) { + result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] ); + if ( result == ASE_OK ) + info->sampleRates[info->nSampleRates++] = SAMPLE_RATES[i]; + } + + if (info->nSampleRates == 0) { + drivers.removeCurrentDriver(); + sprintf( message, "RtAudio: No supported sample rates found for ASIO driver (%s).", info->name ); + error(RtError::DEBUG_WARNING); + return; + } + + // Determine supported data types ... just check first channel and assume rest are the same. + ASIOChannelInfo channelInfo; + channelInfo.channel = 0; + channelInfo.isInput = true; + if ( info->maxInputChannels <= 0 ) channelInfo.isInput = false; + result = ASIOGetChannelInfo( &channelInfo ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO error getting driver (%s) channel information.", info->name); + error(RtError::DEBUG_WARNING); + return; + } + + if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) + info->nativeFormats |= RTAUDIO_SINT16; + else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) + info->nativeFormats |= RTAUDIO_SINT32; + else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) + info->nativeFormats |= RTAUDIO_FLOAT32; + else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) + info->nativeFormats |= RTAUDIO_FLOAT64; + + // Check that we have at least one supported format. + if (info->nativeFormats == 0) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) data format not supported by RtAudio.", + info->name); + error(RtError::DEBUG_WARNING); + return; + } + + info->probed = true; + drivers.removeCurrentDriver(); +} + +void bufferSwitch(long index, ASIOBool processNow) +{ + RtAudio *object = (RtAudio *) asioCallbackInfo->object; + try { + object->callbackEvent( asioCallbackInfo->streamId, index, (void *)NULL, (void *)NULL ); + } + catch (RtError &exception) { + fprintf(stderr, "\nCallback handler error (%s)!\n\n", exception.getMessage()); + return; + } + + return; +} + +void sampleRateChanged(ASIOSampleRate sRate) +{ + // The ASIO documentation says that this usually only happens during + // external sync. Audio processing is not stopped by the driver, + // actual sample rate might not have even changed, maybe only the + // sample rate status of an AES/EBU or S/PDIF digital input at the + // audio device. + + RtAudio *object = (RtAudio *) asioCallbackInfo->object; + try { + object->stopStream( asioCallbackInfo->streamId ); + } + catch (RtError &exception) { + fprintf(stderr, "\nRtAudio: sampleRateChanged() error (%s)!\n\n", exception.getMessage()); + return; + } + + fprintf(stderr, "\nRtAudio: ASIO driver reports sample rate changed to %d ... stream stopped!!!", (int) sRate); +} + +long asioMessages(long selector, long value, void* message, double* opt) +{ + long ret = 0; + switch(selector) { + case kAsioSelectorSupported: + if(value == kAsioResetRequest + || value == kAsioEngineVersion + || value == kAsioResyncRequest + || value == kAsioLatenciesChanged + // The following three were added for ASIO 2.0, you don't + // necessarily have to support them. + || value == kAsioSupportsTimeInfo + || value == kAsioSupportsTimeCode + || value == kAsioSupportsInputMonitor) + ret = 1L; + break; + case kAsioResetRequest: + // Defer the task and perform the reset of the driver during the + // next "safe" situation. You cannot reset the driver right now, + // as this code is called from the driver. Reset the driver is + // done by completely destruct is. I.e. ASIOStop(), + // ASIODisposeBuffers(), Destruction Afterwards you initialize the + // driver again. + fprintf(stderr, "\nRtAudio: ASIO driver reset requested!!!"); + ret = 1L; + break; + case kAsioResyncRequest: + // This informs the application that the driver encountered some + // non-fatal data loss. It is used for synchronization purposes + // of different media. Added mainly to work around the Win16Mutex + // problems in Windows 95/98 with the Windows Multimedia system, + // which could lose data because the Mutex was held too long by + // another thread. However a driver can issue it in other + // situations, too. + fprintf(stderr, "\nRtAudio: ASIO driver resync requested!!!"); + ret = 1L; + break; + case kAsioLatenciesChanged: + // This will inform the host application that the drivers were + // latencies changed. Beware, it this does not mean that the + // buffer sizes have changed! You might need to update internal + // delay data. + fprintf(stderr, "\nRtAudio: ASIO driver latency may have changed!!!"); + ret = 1L; + break; + case kAsioEngineVersion: + // Return the supported ASIO version of the host application. If + // a host application does not implement this selector, ASIO 1.0 + // is assumed by the driver. + ret = 2L; + break; + case kAsioSupportsTimeInfo: + // Informs the driver whether the + // asioCallbacks.bufferSwitchTimeInfo() callback is supported. + // For compatibility with ASIO 1.0 drivers the host application + // should always support the "old" bufferSwitch method, too. + ret = 0; + break; + case kAsioSupportsTimeCode: + // Informs the driver wether application is interested in time + // code info. If an application does not need to know about time + // code, the driver has less work to do. + ret = 0; + break; + } + return ret; +} + +bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, + STREAM_MODE mode, int channels, + int sampleRate, RTAUDIO_FORMAT format, + int *bufferSize, int numberOfBuffers) +{ + // Don't attempt to load another driver if a stream is already open. + if ( streams.size() > 0 ) { + sprintf(message, "RtAudio: unable to load ASIO driver while a stream is open."); + error(RtError::WARNING); + return FAILURE; + } + + // For ASIO, a duplex stream MUST use the same driver. + if ( mode == INPUT && stream->mode == OUTPUT && stream->device[0] != device ) { + sprintf(message, "RtAudio: ASIO duplex stream must use the same device for input and output."); + error(RtError::WARNING); + return FAILURE; + } + + // Only load the driver once for duplex stream. + ASIOError result; + if ( mode != INPUT || stream->mode != OUTPUT ) { + if ( !drivers.loadDriver( devices[device].name ) ) { + sprintf(message, "RtAudio: ASIO error loading driver (%s).", devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + char details[32]; + if ( result == ASE_HWMalfunction ) + sprintf(details, "hardware malfunction"); + else if ( result == ASE_NoMemory ) + sprintf(details, "no memory"); + else if ( result == ASE_NotPresent ) + sprintf(details, "driver/hardware not present"); + else + sprintf(details, "unspecified"); + sprintf(message, "RtAudio: ASIO error (%s) initializing driver (%s).", details, devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + } + + // Check the device channel count. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO error getting input/output channel count (%s).", + devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + if ( ( mode == OUTPUT && channels > outputChannels) || + ( mode == INPUT && channels > inputChannels) ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) does not support requested channel count (%d).", + devices[device].name, channels); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + stream->nDeviceChannels[mode] = channels; + stream->nUserChannels[mode] = channels; + + // Verify the sample rate is supported. + result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) does not support requested sample rate (%d).", + devices[device].name, sampleRate); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + // Set the sample rate. + result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) error setting sample rate (%d).", + devices[device].name, sampleRate); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + // Determine the driver data type. + ASIOChannelInfo channelInfo; + channelInfo.channel = 0; + if ( mode == OUTPUT ) channelInfo.isInput = false; + else channelInfo.isInput = true; + result = ASIOGetChannelInfo( &channelInfo ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) error getting data format.", + devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + // Assuming WINDOWS host is always little-endian. + stream->doByteSwap[mode] = false; + stream->userFormat = format; + stream->deviceFormat[mode] = 0; + if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) { + stream->deviceFormat[mode] = RTAUDIO_SINT16; + if ( channelInfo.type == ASIOSTInt16MSB ) stream->doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) { + stream->deviceFormat[mode] = RTAUDIO_SINT32; + if ( channelInfo.type == ASIOSTInt32MSB ) stream->doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) { + stream->deviceFormat[mode] = RTAUDIO_FLOAT32; + if ( channelInfo.type == ASIOSTFloat32MSB ) stream->doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) { + stream->deviceFormat[mode] = RTAUDIO_FLOAT64; + if ( channelInfo.type == ASIOSTFloat64MSB ) stream->doByteSwap[mode] = true; + } + + if ( stream->deviceFormat[mode] == 0 ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) data format not supported by RtAudio.", + devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + // Set the buffer size. For a duplex stream, this will end up + // setting the buffer size based on the input constraints, which + // should be ok. + long minSize, maxSize, preferSize, granularity; + result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) error getting buffer size.", + devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + if ( *bufferSize < minSize ) *bufferSize = minSize; + else if ( *bufferSize > maxSize ) *bufferSize = maxSize; + else if ( granularity == -1 ) { + // Make sure bufferSize is a power of two. + double power = log10( *bufferSize ) / log10( 2.0 ); + *bufferSize = pow( 2.0, floor(power+0.5) ); + if ( *bufferSize < minSize ) *bufferSize = minSize; + else if ( *bufferSize > maxSize ) *bufferSize = maxSize; + else *bufferSize = preferSize; + } + + if ( mode == INPUT && stream->mode == OUTPUT && stream->bufferSize != *bufferSize ) + cout << "possible input/output buffersize discrepancy" << endl; + + stream->bufferSize = *bufferSize; + stream->nBuffers = 2; + + // ASIO always uses deinterleaved channels. + stream->deInterleave[mode] = true; + + // Create the ASIO internal buffers. Since RtAudio sets up input + // and output separately, we'll have to dispose of previously + // created output buffers for a duplex stream. + if ( mode == INPUT && stream->mode == OUTPUT ) { + free(stream->callbackInfo.buffers); + result = ASIODisposeBuffers(); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) error disposing previously allocated buffers.", + devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + } + + // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. + int i, nChannels = stream->nDeviceChannels[0] + stream->nDeviceChannels[1]; + stream->callbackInfo.buffers = 0; + ASIOBufferInfo *bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); + stream->callbackInfo.buffers = (void *) bufferInfos; + ASIOBufferInfo *infos = bufferInfos; + for ( i=0; i<stream->nDeviceChannels[1]; i++, infos++ ) { + infos->isInput = ASIOTrue; + infos->channelNum = i; + infos->buffers[0] = infos->buffers[1] = 0; + } + + for ( i=0; i<stream->nDeviceChannels[0]; i++, infos++ ) { + infos->isInput = ASIOFalse; + infos->channelNum = i; + infos->buffers[0] = infos->buffers[1] = 0; + } + + // Set up the ASIO callback structure and create the ASIO data buffers. + asioCallbacks.bufferSwitch = &bufferSwitch; + asioCallbacks.sampleRateDidChange = &sampleRateChanged; + asioCallbacks.asioMessage = &asioMessages; + asioCallbacks.bufferSwitchTimeInfo = NULL; + result = ASIOCreateBuffers( bufferInfos, nChannels, stream->bufferSize, &asioCallbacks); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + sprintf(message, "RtAudio: ASIO driver (%s) error creating buffers.", + devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + // Set flags for buffer conversion. + stream->doConvertBuffer[mode] = false; + if (stream->userFormat != stream->deviceFormat[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] > 1 && stream->deInterleave[mode]) + stream->doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { + + long buffer_bytes; + if (stream->nUserChannels[0] >= stream->nUserChannels[1]) + buffer_bytes = stream->nUserChannels[0]; + else + buffer_bytes = stream->nUserChannels[1]; + + buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); + if (stream->userBuffer) free(stream->userBuffer); + stream->userBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->userBuffer == NULL) + goto memory_error; + } + + if ( stream->doConvertBuffer[mode] ) { + + long buffer_bytes; + bool makeBuffer = true; + if ( mode == OUTPUT ) + buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + else { // mode == INPUT + buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); + if ( stream->mode == OUTPUT && stream->deviceBuffer ) { + long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + if ( buffer_bytes < bytes_out ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + buffer_bytes *= *bufferSize; + if (stream->deviceBuffer) free(stream->deviceBuffer); + stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->deviceBuffer == NULL) + goto memory_error; + } + } + + stream->device[mode] = device; + stream->state = STREAM_STOPPED; + if ( stream->mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream->mode = DUPLEX; + else + stream->mode = mode; + stream->sampleRate = sampleRate; + asioCallbackInfo = &stream->callbackInfo; + stream->callbackInfo.object = (void *) this; + stream->callbackInfo.waitTime = (unsigned long) (200.0 * stream->bufferSize / stream->sampleRate); + + return SUCCESS; + + memory_error: + ASIODisposeBuffers(); + drivers.removeCurrentDriver(); + + if (stream->callbackInfo.buffers) + free(stream->callbackInfo.buffers); + stream->callbackInfo.buffers = 0; + + if (stream->userBuffer) { + free(stream->userBuffer); + stream->userBuffer = 0; + } + sprintf(message, "RtAudio: error allocating buffer memory (%s).", + devices[device].name); + error(RtError::WARNING); + return FAILURE; +} + +void RtAudio :: cancelStreamCallback(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + if (stream->callbackInfo.usingCallback) { + + if (stream->state == STREAM_RUNNING) + stopStream( streamId ); + + MUTEX_LOCK(&stream->mutex); + + stream->callbackInfo.usingCallback = false; + stream->callbackInfo.userData = NULL; + stream->state = STREAM_STOPPED; + stream->callbackInfo.callback = NULL; + + MUTEX_UNLOCK(&stream->mutex); + } +} + +void RtAudio :: closeStream(int streamId) +{ + // We don't want an exception to be thrown here because this + // function is called by our class destructor. So, do our own + // streamId check. + if ( streams.find( streamId ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtError::WARNING); + return; + } + + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + + if (stream->state == STREAM_RUNNING) + ASIOStop(); + + ASIODisposeBuffers(); + //ASIOExit(); + drivers.removeCurrentDriver(); + + DeleteCriticalSection(&stream->mutex); + + if (stream->callbackInfo.buffers) + free(stream->callbackInfo.buffers); + + if (stream->userBuffer) + free(stream->userBuffer); + + if (stream->deviceBuffer) + free(stream->deviceBuffer); + + free(stream); + streams.erase(streamId); +} + +void RtAudio :: startStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_RUNNING) { + MUTEX_UNLOCK(&stream->mutex); + return; + } + + stream->callbackInfo.blockTick = true; + stream->callbackInfo.stopStream = false; + stream->callbackInfo.streamId = streamId; + ASIOError result = ASIOStart(); + if ( result != ASE_OK ) { + sprintf(message, "RtAudio: ASIO error starting device (%s).", + devices[stream->device[0]].name); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + stream->state = STREAM_RUNNING; + + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: stopStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) { + MUTEX_UNLOCK(&stream->mutex); + return; + } + + ASIOError result = ASIOStop(); + if ( result != ASE_OK ) { + sprintf(message, "RtAudio: ASIO error stopping device (%s).", + devices[stream->device[0]].name); + MUTEX_UNLOCK(&stream->mutex); + error(RtError::DRIVER_ERROR); + } + stream->state = STREAM_STOPPED; + + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: abortStream(int streamId) +{ + stopStream( streamId ); +} + +// I don't know how this function can be implemented. +int RtAudio :: streamWillBlock(int streamId) +{ + sprintf(message, "RtAudio: streamWillBlock() cannot be implemented for ASIO."); + error(RtError::WARNING); + return 0; +} + +void RtAudio :: tickStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + if (stream->state == STREAM_STOPPED) + return; + + if (stream->callbackInfo.usingCallback) { + sprintf(message, "RtAudio: tickStream() should not be used when a callback function is set!"); + error(RtError::WARNING); + return; + } + + // Block waiting here until the user data is processed in callbackEvent(). + while ( stream->callbackInfo.blockTick ) + Sleep(stream->callbackInfo.waitTime); + + MUTEX_LOCK(&stream->mutex); + + stream->callbackInfo.blockTick = true; + + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: callbackEvent(int streamId, int bufferIndex, void *inData, void *outData) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + CALLBACK_INFO *info = asioCallbackInfo; + if ( !info->usingCallback ) { + // Block waiting here until we get new user data in tickStream(). + while ( !info->blockTick ) + Sleep(info->waitTime); + } + else if ( info->stopStream ) { + // Check if the stream should be stopped (via the previous user + // callback return value). We stop the stream here, rather than + // after the function call, so that output data can first be + // processed. + this->stopStream(asioCallbackInfo->streamId); + return; + } + + MUTEX_LOCK(&stream->mutex); + int nChannels = stream->nDeviceChannels[0] + stream->nDeviceChannels[1]; + int bufferBytes; + ASIOBufferInfo *bufferInfos = (ASIOBufferInfo *) info->buffers; + if ( stream->mode == INPUT || stream->mode == DUPLEX ) { + + bufferBytes = stream->bufferSize * formatBytes(stream->deviceFormat[1]); + if (stream->doConvertBuffer[1]) { + + // Always interleave ASIO input data. + for ( int i=0; i<stream->nDeviceChannels[1]; i++, bufferInfos++ ) + memcpy(&stream->deviceBuffer[i*bufferBytes], bufferInfos->buffers[bufferIndex], bufferBytes ); + + if ( stream->doByteSwap[1] ) + byteSwapBuffer(stream->deviceBuffer, + stream->bufferSize * stream->nDeviceChannels[1], + stream->deviceFormat[1]); + convertStreamBuffer(stream, INPUT); + + } + else { // single channel only + memcpy(stream->userBuffer, bufferInfos->buffers[bufferIndex], bufferBytes ); + + if (stream->doByteSwap[1]) + byteSwapBuffer(stream->userBuffer, + stream->bufferSize * stream->nUserChannels[1], + stream->userFormat); + } + } + + if ( info->usingCallback ) { + RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) info->callback; + if ( callback(stream->userBuffer, stream->bufferSize, info->userData) ) + info->stopStream = true; + } + + if ( stream->mode == OUTPUT || stream->mode == DUPLEX ) { + + bufferBytes = stream->bufferSize * formatBytes(stream->deviceFormat[0]); + if (stream->doConvertBuffer[0]) { + + convertStreamBuffer(stream, OUTPUT); + if ( stream->doByteSwap[0] ) + byteSwapBuffer(stream->deviceBuffer, + stream->bufferSize * stream->nDeviceChannels[0], + stream->deviceFormat[0]); + + // Always de-interleave ASIO output data. + for ( int i=0; i<stream->nDeviceChannels[0]; i++, bufferInfos++ ) { + memcpy(bufferInfos->buffers[bufferIndex], + &stream->deviceBuffer[i*bufferBytes], bufferBytes ); + } + } + else { // single channel only + + if (stream->doByteSwap[0]) + byteSwapBuffer(stream->userBuffer, + stream->bufferSize * stream->nUserChannels[0], + stream->userFormat); + + memcpy(bufferInfos->buffers[bufferIndex], stream->userBuffer, bufferBytes ); + } + } + + if ( !info->usingCallback ) + info->blockTick = false; + + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + stream->callbackInfo.callback = (void *) callback; + stream->callbackInfo.userData = userData; + stream->callbackInfo.usingCallback = true; +} + +//******************** End of __WINDOWS_ASIO__ *********************// + +#elif defined(__WINDOWS_DS__) // Windows DirectSound API + +#include <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 bool CALLBACK defaultDeviceCallback(LPGUID lpguid, + LPCSTR lpcstrDescription, + LPCSTR lpcstrModule, + LPVOID lpContext); + +static bool CALLBACK deviceIdCallback(LPGUID lpguid, + LPCSTR lpcstrDescription, + LPCSTR lpcstrModule, + LPVOID lpContext); + +static char* getErrorString(int code); + +struct enum_info { + char name[64]; + LPGUID id; + bool isInput; + bool isValid; +}; + +int RtAudio :: getDefaultInputDevice(void) +{ + enum_info info; + info.name[0] = '\0'; + + // Enumerate through devices to find the default output. + HRESULT result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)defaultDeviceCallback, &info); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Error performing default input device enumeration: %s.", + getErrorString(result)); + error(RtError::WARNING); + return 0; + } + + for ( int i=0; i<nDevices; i++ ) + if ( strncmp( devices[i].name, info.name, 64 ) == 0 ) return i; + + return 0; +} + +int RtAudio :: getDefaultOutputDevice(void) +{ + enum_info info; + info.name[0] = '\0'; + + // Enumerate through devices to find the default output. + HRESULT result = DirectSoundEnumerate((LPDSENUMCALLBACK)defaultDeviceCallback, &info); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Error performing default output device enumeration: %s.", + getErrorString(result)); + error(RtError::WARNING); + return 0; + } + + for ( int i=0; i<nDevices; i++ ) + if ( strncmp(devices[i].name, info.name, 64 ) == 0 ) return i; + + return 0; +} + +void RtAudio :: initialize(void) +{ + int i, ins = 0, outs = 0, count = 0; + HRESULT result; + nDevices = 0; + + // Count DirectSound devices. + result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceCountCallback, &outs); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to enumerate through sound playback devices: %s.", + getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // Count DirectSoundCapture devices. + result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceCountCallback, &ins); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to enumerate through sound capture devices: %s.", + getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + count = ins + outs; + if (count == 0) return; + + std::vector<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(RtError::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(RtError::DRIVER_ERROR); + } + + // Parse the devices and check validity. Devices are considered + // invalid if they cannot be opened, they report < 1 supported + // channels, or they report no supported data (capture only). + for (i=0; i<count; i++) + if ( info[i].isValid ) nDevices++; + + if (nDevices == 0) return; + + // Allocate the RTAUDIO_DEVICE structures. + devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); + if (devices == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + + // Copy the names to our devices structures. + int index = 0; + for (i=0; i<count; i++) { + if ( info[i].isValid ) + strncpy(devices[index++].name, info[i].name, 64); + } + + //for (i=0;i<nDevices; i++) + //probeDeviceInfo(&devices[i]); + + return; +} + +void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info) +{ + enum_info dsinfo; + strncpy( dsinfo.name, info->name, 64 ); + dsinfo.isValid = false; + + // Enumerate through input devices to find the id (if it exists). + HRESULT result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Error performing input device id enumeration: %s.", + getErrorString(result)); + error(RtError::WARNING); + return; + } + + // Do capture probe first. + if ( dsinfo.isValid == false ) + goto playback_probe; + + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Could not create DirectSound capture object (%s): %s.", + info->name, getErrorString(result)); + error(RtError::WARNING); + goto playback_probe; + } + + DSCCAPS in_caps; + in_caps.dwSize = sizeof(in_caps); + result = input->GetCaps( &in_caps ); + if ( FAILED(result) ) { + input->Release(); + sprintf(message, "RtAudio: Could not get DirectSound capture capabilities (%s): %s.", + info->name, getErrorString(result)); + error(RtError::WARNING); + goto playback_probe; + } + + // Get input channel information. + info->minInputChannels = 1; + info->maxInputChannels = in_caps.dwChannels; + + // Get sample rate and format information. + if( in_caps.dwChannels == 2 ) { + if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->nativeFormats |= RTAUDIO_SINT16; + if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->nativeFormats |= RTAUDIO_SINT16; + if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->nativeFormats |= RTAUDIO_SINT16; + if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->nativeFormats |= RTAUDIO_SINT8; + if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->nativeFormats |= RTAUDIO_SINT8; + if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->nativeFormats |= RTAUDIO_SINT8; + + if ( info->nativeFormats & RTAUDIO_SINT16 ) { + if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->sampleRates[info->nSampleRates++] = 11025; + if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->sampleRates[info->nSampleRates++] = 22050; + if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->sampleRates[info->nSampleRates++] = 44100; + } + else if ( info->nativeFormats & RTAUDIO_SINT8 ) { + if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->sampleRates[info->nSampleRates++] = 11025; + if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->sampleRates[info->nSampleRates++] = 22050; + if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->sampleRates[info->nSampleRates++] = 44100; + } + } + else if ( in_caps.dwChannels == 1 ) { + if( in_caps.dwFormats & WAVE_FORMAT_1M16 ) info->nativeFormats |= RTAUDIO_SINT16; + if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->nativeFormats |= RTAUDIO_SINT16; + if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->nativeFormats |= RTAUDIO_SINT16; + if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->nativeFormats |= RTAUDIO_SINT8; + if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->nativeFormats |= RTAUDIO_SINT8; + if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->nativeFormats |= RTAUDIO_SINT8; + + if ( info->nativeFormats & RTAUDIO_SINT16 ) { + if( in_caps.dwFormats & WAVE_FORMAT_1M16 ) info->sampleRates[info->nSampleRates++] = 11025; + if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->sampleRates[info->nSampleRates++] = 22050; + if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->sampleRates[info->nSampleRates++] = 44100; + } + else if ( info->nativeFormats & RTAUDIO_SINT8 ) { + if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->sampleRates[info->nSampleRates++] = 11025; + if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->sampleRates[info->nSampleRates++] = 22050; + if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->sampleRates[info->nSampleRates++] = 44100; + } + } + else info->minInputChannels = 0; // technically, this would be an error + + input->Release(); + + playback_probe: + + dsinfo.isValid = false; + + // Enumerate through output devices to find the id (if it exists). + result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Error performing output device id enumeration: %s.", + getErrorString(result)); + error(RtError::WARNING); + return; + } + + // Now do playback probe. + if ( dsinfo.isValid == false ) + goto check_parameters; + + LPDIRECTSOUND output; + DSCAPS out_caps; + result = DirectSoundCreate( dsinfo.id, &output, NULL ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Could not create DirectSound playback object (%s): %s.", + info->name, getErrorString(result)); + error(RtError::WARNING); + goto check_parameters; + } + + out_caps.dwSize = sizeof(out_caps); + result = output->GetCaps( &out_caps ); + if ( FAILED(result) ) { + output->Release(); + sprintf(message, "RtAudio: Could not get DirectSound playback capabilities (%s): %s.", + info->name, getErrorString(result)); + error(RtError::WARNING); + goto check_parameters; + } + + // Get output channel information. + info->minOutputChannels = 1; + info->maxOutputChannels = ( out_caps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; + + // Get sample rate information. Use capture device rate information + // if it exists. + if ( info->nSampleRates == 0 ) { + info->sampleRates[0] = (int) out_caps.dwMinSecondarySampleRate; + info->sampleRates[1] = (int) out_caps.dwMaxSecondarySampleRate; + if ( out_caps.dwFlags & DSCAPS_CONTINUOUSRATE ) + info->nSampleRates = -1; + else if ( out_caps.dwMinSecondarySampleRate == out_caps.dwMaxSecondarySampleRate ) { + if ( out_caps.dwMinSecondarySampleRate == 0 ) { + // This is a bogus driver report ... fake the range and cross + // your fingers. + info->sampleRates[0] = 11025; + info->sampleRates[1] = 48000; + info->nSampleRates = -1; /* continuous range */ + sprintf(message, "RtAudio: bogus sample rates reported by DirectSound driver ... using defaults (%s).", + info->name); + error(RtError::DEBUG_WARNING); + } + else { + info->nSampleRates = 1; + } + } + else if ( (out_caps.dwMinSecondarySampleRate < 1000.0) && + (out_caps.dwMaxSecondarySampleRate > 50000.0) ) { + // This is a bogus driver report ... support for only two + // distant rates. We'll assume this is a range. + info->nSampleRates = -1; + sprintf(message, "RtAudio: bogus sample rates reported by DirectSound driver ... using range (%s).", + info->name); + error(RtError::WARNING); + } + else info->nSampleRates = 2; + } + else { + // Check input rates against output rate range + for ( int i=info->nSampleRates-1; i>=0; i-- ) { + if ( info->sampleRates[i] <= out_caps.dwMaxSecondarySampleRate ) + break; + info->nSampleRates--; + } + while ( info->sampleRates[0] < out_caps.dwMinSecondarySampleRate ) { + info->nSampleRates--; + for ( int i=0; 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(RtError::DEBUG_WARNING); + return FAILURE; + } + + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + + enum_info dsinfo; + strncpy( dsinfo.name, devices[device].name, 64 ); + dsinfo.isValid = false; + if ( mode == OUTPUT ) { + + if ( devices[device].maxOutputChannels < channels ) + return FAILURE; + + // Enumerate through output devices to find the id (if it exists). + result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Error performing output device id enumeration: %s.", + getErrorString(result)); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + if ( dsinfo.isValid == false ) { + sprintf(message, "RtAudio: DS output device (%s) id not found!", devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + LPGUID id = dsinfo.id; + LPDIRECTSOUND object; + LPDIRECTSOUNDBUFFER buffer; + DSBUFFERDESC bufferDescription; + + result = DirectSoundCreate( id, &object, NULL ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Could not create DirectSound playback object (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + // Set cooperative level to DSSCL_EXCLUSIVE + result = object->SetCooperativeLevel(hWnd, DSSCL_EXCLUSIVE); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to set DirectSound cooperative level (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + // Even though we will write to the secondary buffer, we need to + // access the primary buffer to set the correct output format. + // The default is 8-bit, 22 kHz! + // Setup the DS primary buffer description. + ZeroMemory(&bufferDescription, sizeof(DSBUFFERDESC)); + bufferDescription.dwSize = sizeof(DSBUFFERDESC); + bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; + // Obtain the primary buffer + result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to access DS primary buffer (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + // Set the primary DS buffer sound format. + result = buffer->SetFormat(&waveFormat); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to set DS primary buffer format (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + // Setup the secondary DS buffer description. + buffer_size = channels * *bufferSize * nBuffers * waveFormat.wBitsPerSample / 8; + ZeroMemory(&bufferDescription, sizeof(DSBUFFERDESC)); + bufferDescription.dwSize = sizeof(DSBUFFERDESC); + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCHARDWARE ); // Force hardware mixing + bufferDescription.dwBufferBytes = buffer_size; + bufferDescription.lpwfxFormat = &waveFormat; + + // Try to create the secondary DS buffer. If that doesn't work, + // try to use software mixing. Otherwise, there's a problem. + result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL); + if ( FAILED(result) ) { + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCSOFTWARE ); // Force software mixing + result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to create secondary DS buffer (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + } + + // Get the buffer size ... might be different from what we specified. + DSBCAPS dsbcaps; + dsbcaps.dwSize = sizeof(DSBCAPS); + buffer->GetCaps(&dsbcaps); + buffer_size = dsbcaps.dwBufferBytes; + + // Lock the DS buffer + result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to lock DS buffer (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + // Zero the DS buffer + ZeroMemory(audioPtr, dataLen); + + // Unlock the DS buffer + result = buffer->Unlock(audioPtr, dataLen, NULL, 0); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to unlock DS buffer(%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + stream->handle[0].object = (void *) object; + stream->handle[0].buffer = (void *) buffer; + stream->nDeviceChannels[0] = channels; + } + + if ( mode == INPUT ) { + + if ( devices[device].maxInputChannels < channels ) + return FAILURE; + + // Enumerate through input devices to find the id (if it exists). + result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Error performing input device id enumeration: %s.", + getErrorString(result)); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + if ( dsinfo.isValid == false ) { + sprintf(message, "RtAudio: DS input device (%s) id not found!", devices[device].name); + error(RtError::DEBUG_WARNING); + return FAILURE; + } + + LPGUID id = dsinfo.id; + LPDIRECTSOUNDCAPTURE object; + LPDIRECTSOUNDCAPTUREBUFFER buffer; + DSCBUFFERDESC bufferDescription; + + result = DirectSoundCaptureCreate( id, &object, NULL ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Could not create DirectSound capture object (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + // Setup the secondary DS buffer description. + buffer_size = channels * *bufferSize * nBuffers * waveFormat.wBitsPerSample / 8; + ZeroMemory(&bufferDescription, sizeof(DSCBUFFERDESC)); + bufferDescription.dwSize = sizeof(DSCBUFFERDESC); + bufferDescription.dwFlags = 0; + bufferDescription.dwReserved = 0; + bufferDescription.dwBufferBytes = buffer_size; + bufferDescription.lpwfxFormat = &waveFormat; + + // Create the capture buffer. + result = object->CreateCaptureBuffer(&bufferDescription, &buffer, NULL); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to create DS capture buffer (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + // Lock the capture buffer + result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + // Zero the buffer + ZeroMemory(audioPtr, dataLen); + + // Unlock the buffer + result = buffer->Unlock(audioPtr, dataLen, NULL, 0); + if ( FAILED(result) ) { + object->Release(); + sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.", + devices[device].name, getErrorString(result)); + error(RtError::WARNING); + return FAILURE; + } + + stream->handle[1].object = (void *) object; + stream->handle[1].buffer = (void *) buffer; + stream->nDeviceChannels[1] = channels; + } + + stream->userFormat = format; + if ( waveFormat.wBitsPerSample == 8 ) + stream->deviceFormat[mode] = RTAUDIO_SINT8; + else + stream->deviceFormat[mode] = RTAUDIO_SINT16; + stream->nUserChannels[mode] = channels; + *bufferSize = buffer_size / (channels * nBuffers * waveFormat.wBitsPerSample / 8); + stream->bufferSize = *bufferSize; + + // Set flags for buffer conversion + stream->doConvertBuffer[mode] = false; + if (stream->userFormat != stream->deviceFormat[mode]) + stream->doConvertBuffer[mode] = true; + if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode]) + stream->doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { + + long buffer_bytes; + if (stream->nUserChannels[0] >= stream->nUserChannels[1]) + buffer_bytes = stream->nUserChannels[0]; + else + buffer_bytes = stream->nUserChannels[1]; + + buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); + if (stream->userBuffer) free(stream->userBuffer); + stream->userBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->userBuffer == NULL) + goto memory_error; + } + + if ( stream->doConvertBuffer[mode] ) { + + long buffer_bytes; + bool makeBuffer = true; + if ( mode == OUTPUT ) + buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + else { // mode == INPUT + buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); + if ( stream->mode == OUTPUT && stream->deviceBuffer ) { + long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + if ( buffer_bytes < bytes_out ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + buffer_bytes *= *bufferSize; + if (stream->deviceBuffer) free(stream->deviceBuffer); + stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->deviceBuffer == NULL) + goto memory_error; + } + } + + stream->device[mode] = device; + stream->state = STREAM_STOPPED; + if ( stream->mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream->mode = DUPLEX; + else + stream->mode = mode; + stream->nBuffers = nBuffers; + stream->sampleRate = sampleRate; + + return SUCCESS; + + memory_error: + if (stream->handle[0].object) { + LPDIRECTSOUND object = (LPDIRECTSOUND) stream->handle[0].object; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; + if (buffer) { + buffer->Release(); + stream->handle[0].buffer = NULL; + } + object->Release(); + stream->handle[0].object = NULL; + } + if (stream->handle[1].object) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) stream->handle[1].object; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; + if (buffer) { + buffer->Release(); + stream->handle[1].buffer = NULL; + } + object->Release(); + stream->handle[1].object = NULL; + } + if (stream->userBuffer) { + free(stream->userBuffer); + stream->userBuffer = 0; + } + sprintf(message, "RtAudio: error allocating buffer memory (%s).", + devices[device].name); + error(RtError::WARNING); + return FAILURE; +} + +void RtAudio :: cancelStreamCallback(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + if (stream->callbackInfo.usingCallback) { + + if (stream->state == STREAM_RUNNING) + stopStream( streamId ); + + MUTEX_LOCK(&stream->mutex); + + stream->callbackInfo.usingCallback = false; + WaitForSingleObject( (HANDLE)stream->callbackInfo.thread, INFINITE ); + CloseHandle( (HANDLE)stream->callbackInfo.thread ); + stream->callbackInfo.thread = 0; + stream->callbackInfo.callback = NULL; + stream->callbackInfo.userData = NULL; + + MUTEX_UNLOCK(&stream->mutex); + } +} + +void RtAudio :: closeStream(int streamId) +{ + // We don't want an exception to be thrown here because this + // function is called by our class destructor. So, do our own + // streamId check. + if ( streams.find( streamId ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtError::WARNING); + return; + } + + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + + if (stream->callbackInfo.usingCallback) { + stream->callbackInfo.usingCallback = false; + WaitForSingleObject( (HANDLE)stream->callbackInfo.thread, INFINITE ); + CloseHandle( (HANDLE)stream->callbackInfo.thread ); + } + + DeleteCriticalSection(&stream->mutex); + + if (stream->handle[0].object) { + LPDIRECTSOUND object = (LPDIRECTSOUND) stream->handle[0].object; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; + if (buffer) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + + if (stream->handle[1].object) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) stream->handle[1].object; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; + if (buffer) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + + if (stream->userBuffer) + free(stream->userBuffer); + + if (stream->deviceBuffer) + free(stream->deviceBuffer); + + free(stream); + streams.erase(streamId); +} + +void RtAudio :: startStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_RUNNING) + goto unlock; + + HRESULT result; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; + result = buffer->Play(0, 0, DSBPLAY_LOOPING ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to start DS buffer (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; + result = buffer->Start(DSCBSTART_LOOPING ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to start DS capture buffer (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + } + stream->state = STREAM_RUNNING; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: stopStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) { + MUTEX_UNLOCK(&stream->mutex); + return; + } + + // There is no specific DirectSound API call to "drain" a buffer + // before stopping. We can hack this for playback by writing zeroes + // for another bufferSize * nBuffers frames. For capture, the + // concept is less clear so we'll repeat what we do in the + // abortStream() case. + HRESULT result; + DWORD dsBufferSize; + LPVOID buffer1 = NULL; + LPVOID buffer2 = NULL; + DWORD bufferSize1 = 0; + DWORD bufferSize2 = 0; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + DWORD currentPos, safePos; + long buffer_bytes = stream->bufferSize * stream->nDeviceChannels[0]; + buffer_bytes *= formatBytes(stream->deviceFormat[0]); + + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; + UINT nextWritePos = stream->handle[0].bufferPointer; + dsBufferSize = buffer_bytes * stream->nBuffers; + + // Write zeroes for nBuffer counts. + for (int i=0; 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(RtError::DRIVER_ERROR); + } + + if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset + DWORD endWrite = nextWritePos + buffer_bytes; + + // Check whether the entire write region is behind the play pointer. + while ( currentPos < endWrite ) { + float millis = (endWrite - currentPos) * 900.0; + millis /= ( formatBytes(stream->deviceFormat[0]) * stream->sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + + // Wake up, find out where we are now + result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset + } + + // Lock free space in the buffer + result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to lock DS buffer during playback (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // Zero the free space + ZeroMemory(buffer1, bufferSize1); + if (buffer2 != NULL) ZeroMemory(buffer2, bufferSize2); + + // Update our buffer offset and unlock sound buffer + dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to unlock DS buffer during playback (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize; + stream->handle[0].bufferPointer = nextWritePos; + } + + // If we play again, start at the beginning of the buffer. + stream->handle[0].bufferPointer = 0; + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; + buffer1 = NULL; + bufferSize1 = 0; + + result = buffer->Stop(); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to stop DS capture buffer (%s): %s", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + dsBufferSize = stream->bufferSize * stream->nDeviceChannels[1]; + dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers; + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock(0, dsBufferSize, &buffer1, &bufferSize1, NULL, NULL, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // Zero the DS buffer + ZeroMemory(buffer1, bufferSize1); + + // Unlock the DS buffer + result = buffer->Unlock(buffer1, bufferSize1, NULL, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // If we start recording again, we must begin at beginning of buffer. + stream->handle[1].bufferPointer = 0; + } + stream->state = STREAM_STOPPED; + + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: abortStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + HRESULT result; + long dsBufferSize; + LPVOID audioPtr; + DWORD dataLen; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; + result = buffer->Stop(); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to stop DS buffer (%s): %s", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + dsBufferSize = stream->bufferSize * stream->nDeviceChannels[0]; + dsBufferSize *= formatBytes(stream->deviceFormat[0]) * stream->nBuffers; + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to lock DS buffer (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // Zero the DS buffer + ZeroMemory(audioPtr, dataLen); + + // Unlock the DS buffer + result = buffer->Unlock(audioPtr, dataLen, NULL, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to unlock DS buffer (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // If we start playing again, we must begin at beginning of buffer. + stream->handle[0].bufferPointer = 0; + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; + audioPtr = NULL; + dataLen = 0; + + result = buffer->Stop(); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to stop DS capture buffer (%s): %s", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + dsBufferSize = stream->bufferSize * stream->nDeviceChannels[1]; + dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers; + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // Zero the DS buffer + ZeroMemory(audioPtr, dataLen); + + // Unlock the DS buffer + result = buffer->Unlock(audioPtr, dataLen, NULL, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // If we start recording again, we must begin at beginning of buffer. + stream->handle[1].bufferPointer = 0; + } + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +int RtAudio :: streamWillBlock(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + int channels; + int frames = 0; + if (stream->state == STREAM_STOPPED) + goto unlock; + + HRESULT result; + DWORD currentPos, safePos; + channels = 1; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; + UINT nextWritePos = stream->handle[0].bufferPointer; + channels = stream->nDeviceChannels[0]; + DWORD dsBufferSize = stream->bufferSize * channels; + dsBufferSize *= formatBytes(stream->deviceFormat[0]) * stream->nBuffers; + + // Find out where the read and "safe write" pointers are. + result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset + frames = currentPos - nextWritePos; + frames /= channels * formatBytes(stream->deviceFormat[0]); + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + + LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; + UINT nextReadPos = stream->handle[1].bufferPointer; + channels = stream->nDeviceChannels[1]; + DWORD dsBufferSize = stream->bufferSize * channels; + dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers; + + // Find out where the write and "safe read" pointers are. + result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset + + if (stream->mode == DUPLEX ) { + // Take largest value of the two. + int temp = safePos - nextReadPos; + temp /= channels * formatBytes(stream->deviceFormat[1]); + frames = ( temp > frames ) ? temp : frames; + } + else { + frames = safePos - nextReadPos; + frames /= channels * formatBytes(stream->deviceFormat[1]); + } + } + + frames = stream->bufferSize - frames; + if (frames < 0) frames = 0; + + unlock: + MUTEX_UNLOCK(&stream->mutex); + return frames; +} + +void RtAudio :: tickStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + int stopStream = 0; + if (stream->state == STREAM_STOPPED) { + if (stream->callbackInfo.usingCallback) Sleep(50); // sleep 50 milliseconds + return; + } + else if (stream->callbackInfo.usingCallback) { + RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback; + stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData); + } + + MUTEX_LOCK(&stream->mutex); + + // The state might change while waiting on a mutex. + if (stream->state == STREAM_STOPPED) { + MUTEX_UNLOCK(&stream->mutex); + return; + } + + HRESULT result; + DWORD currentPos, safePos; + LPVOID buffer1 = NULL; + LPVOID buffer2 = NULL; + DWORD bufferSize1 = 0; + DWORD bufferSize2 = 0; + char *buffer; + long buffer_bytes; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + // Setup parameters and do buffer conversion if necessary. + if (stream->doConvertBuffer[0]) { + convertStreamBuffer(stream, OUTPUT); + buffer = stream->deviceBuffer; + buffer_bytes = stream->bufferSize * stream->nDeviceChannels[0]; + buffer_bytes *= formatBytes(stream->deviceFormat[0]); + } + else { + buffer = stream->userBuffer; + buffer_bytes = stream->bufferSize * stream->nUserChannels[0]; + buffer_bytes *= formatBytes(stream->userFormat); + } + + // No byte swapping necessary in DirectSound implementation. + + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; + UINT nextWritePos = stream->handle[0].bufferPointer; + DWORD dsBufferSize = buffer_bytes * stream->nBuffers; + + // Find out where the read and "safe write" pointers are. + result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset + DWORD endWrite = nextWritePos + buffer_bytes; + + // Check whether the entire write region is behind the play pointer. + while ( currentPos < endWrite ) { + // If we are here, then we must wait until the play pointer gets + // beyond the write region. The approach here is to use the + // Sleep() function to suspend operation until safePos catches + // up. Calculate number of milliseconds to wait as: + // time = distance * (milliseconds/second) * fudgefactor / + // ((bytes/sample) * (samples/second)) + // A "fudgefactor" less than 1 is used because it was found + // that sleeping too long was MUCH worse than sleeping for + // several shorter periods. + float millis = (endWrite - currentPos) * 900.0; + millis /= ( formatBytes(stream->deviceFormat[0]) * stream->sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + + // Wake up, find out where we are now + result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset + } + + // Lock free space in the buffer + result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to lock DS buffer during playback (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // Copy our buffer into the DS buffer + CopyMemory(buffer1, buffer, bufferSize1); + if (buffer2 != NULL) CopyMemory(buffer2, buffer+bufferSize1, bufferSize2); + + // Update our buffer offset and unlock sound buffer + dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to unlock DS buffer during playback (%s): %s.", + devices[stream->device[0]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize; + stream->handle[0].bufferPointer = nextWritePos; + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + + // Setup parameters. + if (stream->doConvertBuffer[1]) { + buffer = stream->deviceBuffer; + buffer_bytes = stream->bufferSize * stream->nDeviceChannels[1]; + buffer_bytes *= formatBytes(stream->deviceFormat[1]); + } + else { + buffer = stream->userBuffer; + buffer_bytes = stream->bufferSize * stream->nUserChannels[1]; + buffer_bytes *= formatBytes(stream->userFormat); + } + + LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; + UINT nextReadPos = stream->handle[1].bufferPointer; + DWORD dsBufferSize = buffer_bytes * stream->nBuffers; + + // Find out where the write and "safe read" pointers are. + result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset + DWORD endRead = nextReadPos + buffer_bytes; + + // Check whether the entire write region is behind the play pointer. + while ( safePos < endRead ) { + // See comments for playback. + float millis = (endRead - safePos) * 900.0; + millis /= ( formatBytes(stream->deviceFormat[1]) * stream->sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + + // Wake up, find out where we are now + result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos ); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset + } + + // Lock free space in the buffer + result = dsBuffer->Lock (nextReadPos, buffer_bytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to lock DS buffer during capture (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + + // Copy our buffer into the DS buffer + CopyMemory(buffer, buffer1, bufferSize1); + if (buffer2 != NULL) CopyMemory(buffer+bufferSize1, buffer2, bufferSize2); + + // Update our buffer offset and unlock sound buffer + nextReadPos = (nextReadPos + bufferSize1 + bufferSize2) % dsBufferSize; + dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2); + if ( FAILED(result) ) { + sprintf(message, "RtAudio: Unable to unlock DS buffer during capture (%s): %s.", + devices[stream->device[1]].name, getErrorString(result)); + error(RtError::DRIVER_ERROR); + } + stream->handle[1].bufferPointer = nextReadPos; + + // No byte swapping necessary in DirectSound implementation. + + // Do buffer conversion if necessary. + if (stream->doConvertBuffer[1]) + convertStreamBuffer(stream, INPUT); + } + + MUTEX_UNLOCK(&stream->mutex); + + if (stream->callbackInfo.usingCallback && stopStream) + this->stopStream(streamId); +} + +// Definitions for utility functions and callbacks +// specific to the DirectSound implementation. + +extern "C" unsigned __stdcall callbackHandler(void *ptr) +{ + CALLBACK_INFO *info = (CALLBACK_INFO *) ptr; + RtAudio *object = (RtAudio *) info->object; + int stream = info->streamId; + bool *usingCallback = &info->usingCallback; + + while ( *usingCallback ) { + try { + object->tickStream(stream); + } + catch (RtError &exception) { + fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n", + exception.getMessage()); + break; + } + } + + _endthreadex( 0 ); + return 0; +} + +void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + CALLBACK_INFO *info = (CALLBACK_INFO *) &stream->callbackInfo; + if ( info->usingCallback ) { + sprintf(message, "RtAudio: A callback is already set for this stream!"); + error(RtError::WARNING); + return; + } + + info->callback = (void *) callback; + info->userData = userData; + info->usingCallback = true; + info->object = (void *) this; + info->streamId = streamId; + + unsigned thread_id; + info->thread = _beginthreadex(NULL, 0, &callbackHandler, + &stream->callbackInfo, 0, &thread_id); + if (info->thread == 0) { + info->usingCallback = false; + sprintf(message, "RtAudio: error starting callback thread!"); + error(RtError::THREAD_ERROR); + } + + // When spawning multiple threads in quick succession, it appears to be + // necessary to wait a bit for each to initialize ... another windoism! + Sleep(1); +} + +static bool CALLBACK deviceCountCallback(LPGUID lpguid, + LPCSTR lpcstrDescription, + LPCSTR lpcstrModule, + LPVOID lpContext) +{ + int *pointer = ((int *) lpContext); + (*pointer)++; + + return true; +} + +static bool CALLBACK deviceInfoCallback(LPGUID lpguid, + LPCSTR lpcstrDescription, + LPCSTR lpcstrModule, + LPVOID lpContext) +{ + enum_info *info = ((enum_info *) lpContext); + while (strlen(info->name) > 0) info++; + + strncpy(info->name, lpcstrDescription, 64); + info->id = lpguid; + + HRESULT hr; + info->isValid = false; + if (info->isInput == true) { + DSCCAPS caps; + LPDIRECTSOUNDCAPTURE object; + + hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); + if( hr != DS_OK ) return true; + + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if( hr == DS_OK ) { + if (caps.dwChannels > 0 && caps.dwFormats > 0) + info->isValid = true; + } + object->Release(); + } + else { + DSCAPS caps; + LPDIRECTSOUND object; + hr = DirectSoundCreate( lpguid, &object, NULL ); + if( hr != DS_OK ) return true; + + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if( hr == DS_OK ) { + if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) + info->isValid = true; + } + object->Release(); + } + + return true; +} + +static bool CALLBACK defaultDeviceCallback(LPGUID lpguid, + LPCSTR lpcstrDescription, + LPCSTR lpcstrModule, + LPVOID lpContext) +{ + enum_info *info = ((enum_info *) lpContext); + + if ( lpguid == NULL ) { + strncpy(info->name, lpcstrDescription, 64); + return false; + } + + return true; +} + +static bool CALLBACK deviceIdCallback(LPGUID lpguid, + LPCSTR lpcstrDescription, + LPCSTR lpcstrModule, + LPVOID lpContext) +{ + enum_info *info = ((enum_info *) lpContext); + + if ( strncmp( info->name, lpcstrDescription, 64 ) == 0 ) { + info->id = lpguid; + info->isValid = true; + return false; + } + + return true; +} + +static char* getErrorString(int code) +{ + switch (code) { + + case DSERR_ALLOCATED: + return "Direct Sound already allocated"; + + case DSERR_CONTROLUNAVAIL: + return "Direct Sound control unavailable"; + + case DSERR_INVALIDPARAM: + return "Direct Sound invalid parameter"; + + case DSERR_INVALIDCALL: + return "Direct Sound invalid call"; + + case DSERR_GENERIC: + return "Direct Sound generic error"; + + case DSERR_PRIOLEVELNEEDED: + return "Direct Sound Priority level needed"; + + case DSERR_OUTOFMEMORY: + return "Direct Sound out of memory"; + + case DSERR_BADFORMAT: + return "Direct Sound bad format"; + + case DSERR_UNSUPPORTED: + return "Direct Sound unsupported error"; + + case DSERR_NODRIVER: + return "Direct Sound no driver error"; + + case DSERR_ALREADYINITIALIZED: + return "Direct Sound already initialized"; + + case DSERR_NOAGGREGATION: + return "Direct Sound no aggregation"; + + case DSERR_BUFFERLOST: + return "Direct Sound buffer lost"; + + case DSERR_OTHERAPPHASPRIO: + return "Direct Sound other app has priority"; + + case DSERR_UNINITIALIZED: + return "Direct Sound uninitialized"; + + default: + return "Direct Sound unknown error"; + } +} + +//******************** End of __WINDOWS_DS__ *********************// + +#elif defined(__IRIX_AL__) // SGI's AL API for IRIX + +#include <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(RtError::DRIVER_ERROR); + } + + if (nDevices <= 0) return; + + ALvalue *vls = (ALvalue *) new ALvalue[nDevices]; + + // Allocate the RTAUDIO_DEVICE structures. + devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); + if (devices == NULL) { + sprintf(message, "RtAudio: memory allocation error!"); + error(RtError::MEMORY_ERROR); + } + + // Write device ascii identifiers and resource ids to device info + // structure. + char name[32]; + int outs, ins, i; + ALpv pvs[1]; + pvs[0].param = AL_NAME; + pvs[0].value.ptr = name; + pvs[0].sizeIn = 32; + + outs = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, vls, nDevices, 0, 0); + if (outs < 0) { + sprintf(message, "RtAudio: AL error getting output devices: %s.", + alGetErrorString(oserror())); + error(RtError::DRIVER_ERROR); + } + + for (i=0; i<outs; i++) { + if (alGetParams(vls[i].i, pvs, 1) < 0) { + sprintf(message, "RtAudio: AL error querying output devices: %s.", + alGetErrorString(oserror())); + error(RtError::DRIVER_ERROR); + } + strncpy(devices[i].name, name, 32); + devices[i].id[0] = vls[i].i; + } + + ins = alQueryValues(AL_SYSTEM, AL_DEFAULT_INPUT, &vls[outs], nDevices-outs, 0, 0); + if (ins < 0) { + sprintf(message, "RtAudio: AL error getting input devices: %s.", + alGetErrorString(oserror())); + error(RtError::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(RtError::DRIVER_ERROR); + } + strncpy(devices[i].name, name, 32); + devices[i].id[1] = vls[i].i; + } + + delete [] vls; + + return; +} + +int RtAudio :: getDefaultInputDevice(void) +{ + ALvalue value; + int result = alQueryValues(AL_SYSTEM, AL_DEFAULT_INPUT, &value, 1, 0, 0); + if (result < 0) { + sprintf(message, "RtAudio: AL error getting default input device id: %s.", + alGetErrorString(oserror())); + error(RtError::WARNING); + } + else { + for ( int i=0; i<nDevices; i++ ) + if ( devices[i].id[1] == value.i ) return i; + } + + return 0; +} + +int RtAudio :: getDefaultOutputDevice(void) +{ + ALvalue value; + int result = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, &value, 1, 0, 0); + if (result < 0) { + sprintf(message, "RtAudio: AL error getting default output device id: %s.", + alGetErrorString(oserror())); + error(RtError::WARNING); + } + else { + for ( int i=0; i<nDevices; i++ ) + if ( devices[i].id[0] == value.i ) return i; + } + + return 0; +} + +void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info) +{ + int resource, result, i; + ALvalue value; + ALparamInfo pinfo; + + // Get output resource ID if it exists. + resource = info->id[0]; + if (resource > 0) { + + // Probe output device parameters. + result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0); + if (result < 0) { + sprintf(message, "RtAudio: AL error getting device (%s) channels: %s.", + info->name, alGetErrorString(oserror())); + error(RtError::WARNING); + } + else { + info->maxOutputChannels = value.i; + info->minOutputChannels = 1; + } + + result = alGetParamInfo(resource, AL_RATE, &pinfo); + if (result < 0) { + sprintf(message, "RtAudio: AL error getting device (%s) rates: %s.", + info->name, alGetErrorString(oserror())); + error(RtError::WARNING); + } + else { + info->nSampleRates = 0; + for (i=0; i<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. + resource = info->id[1]; + if (resource > 0) { + + // Probe input device parameters. + result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0); + if (result < 0) { + sprintf(message, "RtAudio: AL error getting device (%s) channels: %s.", + info->name, alGetErrorString(oserror())); + error(RtError::WARNING); + } + else { + info->maxInputChannels = value.i; + info->minInputChannels = 1; + } + + result = alGetParamInfo(resource, AL_RATE, &pinfo); + if (result < 0) { + sprintf(message, "RtAudio: AL error getting device (%s) rates: %s.", + info->name, alGetErrorString(oserror())); + error(RtError::WARNING); + } + else { + // In the case of the default device, these values will + // overwrite the rates determined for the output device. Since + // the input device is most likely to be more limited than the + // output device, this is ok. + info->nSampleRates = 0; + for (i=0; i<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(RtError::WARNING); + return FAILURE; + } + + // Set the channels. + result = alSetChannels(al_config, channels); + if ( result < 0 ) { + sprintf(message,"RtAudio: can't set %d channels in AL config: %s.", + channels, alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + + // Attempt to set the queue size. The al API doesn't provide a + // means for querying the minimum/maximum buffer size of a device, + // so if the specified size doesn't work, take whatever the + // al_config structure returns. + if ( numberOfBuffers < 1 ) + nBuffers = 1; + else + nBuffers = numberOfBuffers; + long buffer_size = *bufferSize * nBuffers; + result = alSetQueueSize(al_config, buffer_size); // in sample frames + if ( result < 0 ) { + // Get the buffer size specified by the al_config and try that. + buffer_size = alGetQueueSize(al_config); + result = alSetQueueSize(al_config, buffer_size); + if ( result < 0 ) { + sprintf(message,"RtAudio: can't set buffer size (%ld) in AL config: %s.", + buffer_size, alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + *bufferSize = buffer_size / nBuffers; + } + + // Set the data format. + stream->userFormat = format; + stream->deviceFormat[mode] = format; + if (format == RTAUDIO_SINT8) { + result = alSetSampFmt(al_config, AL_SAMPFMT_TWOSCOMP); + result = alSetWidth(al_config, AL_SAMPLE_8); + } + else if (format == RTAUDIO_SINT16) { + result = alSetSampFmt(al_config, AL_SAMPFMT_TWOSCOMP); + result = alSetWidth(al_config, AL_SAMPLE_16); + } + else if (format == RTAUDIO_SINT24) { + // Our 24-bit format assumes the upper 3 bytes of a 4 byte word. + // The AL library uses the lower 3 bytes, so we'll need to do our + // own conversion. + result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT); + stream->deviceFormat[mode] = RTAUDIO_FLOAT32; + } + else if (format == RTAUDIO_SINT32) { + // The AL library doesn't seem to support the 32-bit integer + // format, so we'll need to do our own conversion. + result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT); + stream->deviceFormat[mode] = RTAUDIO_FLOAT32; + } + else if (format == RTAUDIO_FLOAT32) + result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT); + else if (format == RTAUDIO_FLOAT64) + result = alSetSampFmt(al_config, AL_SAMPFMT_DOUBLE); + + if ( result == -1 ) { + sprintf(message,"RtAudio: AL error setting sample format in AL config: %s.", + alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + + if (mode == OUTPUT) { + + // Set our device. + if (device == 0) + resource = AL_DEFAULT_OUTPUT; + else + resource = devices[device].id[0]; + result = alSetDevice(al_config, resource); + if ( result == -1 ) { + sprintf(message,"RtAudio: AL error setting device (%s) in AL config: %s.", + devices[device].name, alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + + // Open the port. + port = alOpenPort("RtAudio Output Port", "w", al_config); + if( !port ) { + sprintf(message,"RtAudio: AL error opening output port: %s.", + alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + + // Set the sample rate + pvs[0].param = AL_MASTER_CLOCK; + pvs[0].value.i = AL_CRYSTAL_MCLK_TYPE; + pvs[1].param = AL_RATE; + pvs[1].value.ll = alDoubleToFixed((double)sampleRate); + result = alSetParams(resource, pvs, 2); + if ( result < 0 ) { + alClosePort(port); + sprintf(message,"RtAudio: AL error setting sample rate (%d) for device (%s): %s.", + sampleRate, devices[device].name, alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + } + else { // mode == INPUT + + // Set our device. + if (device == 0) + resource = AL_DEFAULT_INPUT; + else + resource = devices[device].id[1]; + result = alSetDevice(al_config, resource); + if ( result == -1 ) { + sprintf(message,"RtAudio: AL error setting device (%s) in AL config: %s.", + devices[device].name, alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + + // Open the port. + port = alOpenPort("RtAudio Output Port", "r", al_config); + if( !port ) { + sprintf(message,"RtAudio: AL error opening input port: %s.", + alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + + // Set the sample rate + pvs[0].param = AL_MASTER_CLOCK; + pvs[0].value.i = AL_CRYSTAL_MCLK_TYPE; + pvs[1].param = AL_RATE; + pvs[1].value.ll = alDoubleToFixed((double)sampleRate); + result = alSetParams(resource, pvs, 2); + if ( result < 0 ) { + alClosePort(port); + sprintf(message,"RtAudio: AL error setting sample rate (%d) for device (%s): %s.", + sampleRate, devices[device].name, alGetErrorString(oserror())); + error(RtError::WARNING); + return FAILURE; + } + } + + alFreeConfig(al_config); + + stream->nUserChannels[mode] = channels; + stream->nDeviceChannels[mode] = channels; + + // Set handle and flags for buffer conversion + stream->handle[mode] = port; + stream->doConvertBuffer[mode] = false; + if (stream->userFormat != stream->deviceFormat[mode]) + stream->doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { + + long buffer_bytes; + if (stream->nUserChannels[0] >= stream->nUserChannels[1]) + buffer_bytes = stream->nUserChannels[0]; + else + buffer_bytes = stream->nUserChannels[1]; + + buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); + if (stream->userBuffer) free(stream->userBuffer); + stream->userBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->userBuffer == NULL) + goto memory_error; + } + + if ( stream->doConvertBuffer[mode] ) { + + long buffer_bytes; + bool makeBuffer = true; + if ( mode == OUTPUT ) + buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + else { // mode == INPUT + buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); + if ( stream->mode == OUTPUT && stream->deviceBuffer ) { + long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); + if ( buffer_bytes < bytes_out ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + buffer_bytes *= *bufferSize; + if (stream->deviceBuffer) free(stream->deviceBuffer); + stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); + if (stream->deviceBuffer == NULL) + goto memory_error; + } + } + + stream->device[mode] = device; + stream->state = STREAM_STOPPED; + if ( stream->mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream->mode = DUPLEX; + else + stream->mode = mode; + stream->nBuffers = nBuffers; + stream->bufferSize = *bufferSize; + stream->sampleRate = sampleRate; + + return SUCCESS; + + memory_error: + if (stream->handle[0]) { + alClosePort(stream->handle[0]); + stream->handle[0] = 0; + } + if (stream->handle[1]) { + alClosePort(stream->handle[1]); + stream->handle[1] = 0; + } + if (stream->userBuffer) { + free(stream->userBuffer); + stream->userBuffer = 0; + } + sprintf(message, "RtAudio: ALSA error allocating buffer memory for device (%s).", + devices[device].name); + error(RtError::WARNING); + return FAILURE; +} + +void RtAudio :: closeStream(int streamId) +{ + // We don't want an exception to be thrown here because this + // function is called by our class destructor. So, do our own + // streamId check. + if ( streams.find( streamId ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtError::WARNING); + return; + } + + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + + if (stream->callbackInfo.usingCallback) { + pthread_cancel(stream->callbackInfo.thread); + pthread_join(stream->callbackInfo.thread, NULL); + } + + pthread_mutex_destroy(&stream->mutex); + + if (stream->handle[0]) + alClosePort(stream->handle[0]); + + if (stream->handle[1]) + alClosePort(stream->handle[1]); + + if (stream->userBuffer) + free(stream->userBuffer); + + if (stream->deviceBuffer) + free(stream->deviceBuffer); + + free(stream); + streams.erase(streamId); +} + +void RtAudio :: startStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + if (stream->state == STREAM_RUNNING) + return; + + // The AL port is ready as soon as it is opened. + stream->state = STREAM_RUNNING; +} + +void RtAudio :: stopStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + int result; + int buffer_size = stream->bufferSize * stream->nBuffers; + + if (stream->mode == OUTPUT || stream->mode == DUPLEX) + alZeroFrames(stream->handle[0], buffer_size); + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + result = alDiscardFrames(stream->handle[1], buffer_size); + if (result == -1) { + sprintf(message, "RtAudio: AL error draining stream device (%s): %s.", + devices[stream->device[1]].name, alGetErrorString(oserror())); + error(RtError::DRIVER_ERROR); + } + } + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +void RtAudio :: abortStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + if (stream->state == STREAM_STOPPED) + goto unlock; + + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + int buffer_size = stream->bufferSize * stream->nBuffers; + int result = alDiscardFrames(stream->handle[0], buffer_size); + if (result == -1) { + sprintf(message, "RtAudio: AL error aborting stream device (%s): %s.", + devices[stream->device[0]].name, alGetErrorString(oserror())); + error(RtError::DRIVER_ERROR); + } + } + + // There is no clear action to take on the input stream, since the + // port will continue to run in any event. + stream->state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK(&stream->mutex); +} + +int RtAudio :: streamWillBlock(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + MUTEX_LOCK(&stream->mutex); + + int frames = 0; + if (stream->state == STREAM_STOPPED) + goto unlock; + + int err = 0; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + err = alGetFillable(stream->handle[0]); + if (err < 0) { + sprintf(message, "RtAudio: AL error getting available frames for stream (%s): %s.", + devices[stream->device[0]].name, alGetErrorString(oserror())); + error(RtError::DRIVER_ERROR); + } + } + + frames = err; + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + err = alGetFilled(stream->handle[1]); + if (err < 0) { + sprintf(message, "RtAudio: AL error getting available frames for stream (%s): %s.", + devices[stream->device[1]].name, alGetErrorString(oserror())); + error(RtError::DRIVER_ERROR); + } + if (frames > err) frames = err; + } + + frames = stream->bufferSize - frames; + if (frames < 0) frames = 0; + + unlock: + MUTEX_UNLOCK(&stream->mutex); + return frames; +} + +void RtAudio :: tickStream(int streamId) +{ + RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + + int stopStream = 0; + if (stream->state == STREAM_STOPPED) { + if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds + return; + } + else if (stream->callbackInfo.usingCallback) { + RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback; + stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData); + } + + MUTEX_LOCK(&stream->mutex); + + // The state might change while waiting on a mutex. + if (stream->state == STREAM_STOPPED) + goto unlock; + + char *buffer; + int channels; + RTAUDIO_FORMAT format; + if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + + // Setup parameters and do buffer conversion if necessary. + if (stream->doConvertBuffer[0]) { + convertStreamBuffer(stream, OUTPUT); + buffer = stream->deviceBuffer; + channels = stream->nDeviceChannels[0]; + format = stream->deviceFormat[0]; + } + else { + buffer = stream->userBuffer; + channels = stream->nUserChannels[0]; + format = stream->userFormat; + } + + // Do byte swapping if necessary. + if (stream->doByteSwap[0]) + byteSwapBuffer(buffer, stream->bufferSize * channels, format); + + // Write interleaved samples to device. + alWriteFrames(stream->handle[0], buffer, stream->bufferSize); + } + + if (stream->mode == INPUT || stream->mode == DUPLEX) { + + // Setup parameters. + if (stream->doConvertBuffer[1]) { + buffer = stream->deviceBuffer; + channels = stream->nDeviceChannels[1]; + format = stream->deviceFormat[1]; + } + else { + buffer = stream->userBuffer; + channels = stream->nUserChannels[1]; + format = stream->userFormat; + } + + // Read interleaved samples from device. + alReadFrames(stream->handle[1], buffer, stream->bufferSize); + + // Do byte swapping if necessary. + if (stream->doByteSwap[1]) + byteSwapBuffer(buffer, stream->bufferSize * channels, format); + + // Do buffer conversion if necessary. + if (stream->doConvertBuffer[1]) + convertStreamBuffer(stream, INPUT); + } + + unlock: + MUTEX_UNLOCK(&stream->mutex); + + if (stream->callbackInfo.usingCallback && stopStream) + this->stopStream(streamId); +} + +extern "C" void *callbackHandler(void *ptr) +{ + CALLBACK_INFO *info = (CALLBACK_INFO *) ptr; + RtAudio *object = (RtAudio *) info->object; + int stream = info->streamId; + bool *usingCallback = &info->usingCallback; + + while ( *usingCallback ) { + pthread_testcancel(); + try { + object->tickStream(stream); + } + catch (RtError &exception) { + fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n", + exception.getMessage()); + break; + } + } + + return 0; +} + +//******************** End of __IRIX_AL__ *********************// + +#endif + + +// *************************************************** // +// +// Private common (OS-independent) RtAudio methods. +// +// *************************************************** // + +// This method can be modified to control the behavior of error +// message reporting and throwing. +void RtAudio :: error(RtError::TYPE type) +{ + if (type == RtError::WARNING) { + fprintf(stderr, "\n%s\n\n", message); + } + else if (type == RtError::DEBUG_WARNING) { +#if defined(__RTAUDIO_DEBUG__) + fprintf(stderr, "\n%s\n\n", message); +#endif + } + else { + fprintf(stderr, "\n%s\n\n", message); + throw RtError(message, type); + } +} + +void *RtAudio :: verifyStream(int streamId) +{ + // Verify the stream key. + if ( streams.find( streamId ) == streams.end() ) { + sprintf(message, "RtAudio: invalid stream identifier!"); + error(RtError::INVALID_STREAM); + } + + return streams[streamId]; +} + +void RtAudio :: clearDeviceInfo(RTAUDIO_DEVICE *info) +{ + // Don't clear the name or DEVICE_ID fields here ... they are + // typically set prior to a call of this function. + info->probed = false; + info->maxOutputChannels = 0; + info->maxInputChannels = 0; + info->maxDuplexChannels = 0; + info->minOutputChannels = 0; + info->minInputChannels = 0; + info->minDuplexChannels = 0; + info->hasDuplexSupport = false; + info->nSampleRates = 0; + for (int i=0; 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(RtError::WARNING); + + return 0; +} + +void RtAudio :: convertStreamBuffer(RTAUDIO_STREAM *stream, STREAM_MODE mode) +{ + // This method does format conversion, input/output channel compensation, and + // data interleaving/deinterleaving. 24-bit integers are assumed to occupy + // the upper three bytes of a 32-bit integer. + + int j, jump_in, jump_out, channels; + RTAUDIO_FORMAT format_in, format_out; + char *input, *output; + + if (mode == INPUT) { // convert device to user buffer + input = stream->deviceBuffer; + output = stream->userBuffer; + jump_in = stream->nDeviceChannels[1]; + jump_out = stream->nUserChannels[1]; + format_in = stream->deviceFormat[1]; + format_out = stream->userFormat; + } + else { // convert user to device buffer + input = stream->userBuffer; + output = stream->deviceBuffer; + jump_in = stream->nUserChannels[0]; + jump_out = stream->nDeviceChannels[0]; + format_in = stream->userFormat; + format_out = stream->deviceFormat[0]; + + // clear our device buffer when in/out duplex device channels are different + if ( stream->mode == DUPLEX && + stream->nDeviceChannels[0] != stream->nDeviceChannels[1] ) + memset(output, 0, stream->bufferSize * jump_out * formatBytes(format_out)); + } + + channels = (jump_in < jump_out) ? jump_in : jump_out; + + // Set up the interleave/deinterleave offsets + std::vector<int> offset_in(channels); + std::vector<int> offset_out(channels); + if (mode == INPUT && stream->deInterleave[1]) { + for (int k=0; k<channels; k++) { + offset_in[k] = k * stream->bufferSize; + offset_out[k] = k; + jump_in = 1; + } + } + else if (mode == OUTPUT && stream->deInterleave[0]) { + for (int k=0; k<channels; k++) { + offset_in[k] = k; + offset_out[k] = k * stream->bufferSize; + jump_out = 1; + } + } + 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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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 += jump_in; + out += jump_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; + } + } +} + + +// *************************************************** // +// +// RtError class definition. +// +// *************************************************** // + +RtError :: RtError(const char *p, TYPE tipe) +{ + type = tipe; + strncpy(error_message, p, 256); +} + +RtError :: ~RtError() +{ +} + +void RtError :: printMessage() +{ + printf("\n%s\n\n", error_message); +} |