static void
gst_directsound_src_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstDirectSoundSrc * src = GST_DIRECTSOUND_SRC (object);
switch (prop_id) {
/*case ARG_VOLUME:
sink->volume = g_value_get_double (value);
gst_directsound_sink_set_volume (sink);
break;*/
case ARG_DEVICE:
if (src->device_id) {
g_free (src->device_id);
src->device_id = NULL;
}
if (src->device_name) {
g_free (src->device_name);
src->device_name = NULL;
}
src->device_id = g_strdup (g_value_get_string (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_directsound_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstDirectSoundSrc *src = GST_DIRECTSOUND_SRC (object);
GST_DEBUG ("set property");
switch (prop_id) {
case PROP_DEVICE_NAME:
if (src->device_name) {
g_free (src->device_name);
src->device_name = NULL;
}
if (g_value_get_string (value)) {
src->device_name = g_strdup (g_value_get_string (value));
}
break;
case PROP_VOLUME:
gst_directsound_src_set_volume (src, g_value_get_double (value));
break;
case PROP_MUTE:
gst_directsound_src_set_mute (src, g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstBaseAudioSrc vmethod implementations */
static GstRingBuffer *
gst_directsound_src_create_ringbuffer (GstBaseAudioSrc * src)
{
GstDirectSoundSrc * dsoundsrc;
GstDirectSoundRingBuffer * ringbuffer;
dsoundsrc = GST_DIRECTSOUND_SRC (src);
GST_DEBUG ("creating ringbuffer");
ringbuffer = g_object_new (GST_TYPE_DIRECTSOUND_RING_BUFFER, NULL);
GST_DEBUG ("directsound src 0x%p", dsoundsrc);
/* capture */
ringbuffer->is_src = TRUE;
/* set the src element on the ringbuffer for error messages */
ringbuffer->element = GST_ELEMENT (dsoundsrc);
/* set the ringbuffer on the src */
dsoundsrc->dsoundbuffer = ringbuffer;
/* set initial volume on ringbuffer */
dsoundsrc->dsoundbuffer->volume = dsoundsrc->volume;
return GST_RING_BUFFER (ringbuffer);
}
static guint
gst_directsound_src_delay (GstAudioSrc * asrc)
{
GstDirectSoundSrc *dsoundsrc;
HRESULT hRes;
DWORD dwCurrentCaptureCursor;
DWORD dwBytesInQueue = 0;
gint nNbSamplesInQueue = 0;
GST_DEBUG_OBJECT (asrc, "Delay");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
/* evaluate the number of samples in queue in the circular buffer */
hRes =
IDirectSoundCaptureBuffer_GetCurrentPosition (dsoundsrc->pDSBSecondary,
&dwCurrentCaptureCursor, NULL);
/* FIXME: Check is this calculated right */
if (hRes == S_OK) {
if (dwCurrentCaptureCursor < dsoundsrc->current_circular_offset) {
dwBytesInQueue =
dsoundsrc->buffer_size - (dsoundsrc->current_circular_offset -
dwCurrentCaptureCursor);
} else {
dwBytesInQueue =
dwCurrentCaptureCursor - dsoundsrc->current_circular_offset;
}
nNbSamplesInQueue = dwBytesInQueue / dsoundsrc->bytes_per_sample;
}
return nNbSamplesInQueue;
}
static void
gst_directsound_src_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstDirectSoundSrc * src = GST_DIRECTSOUND_SRC (object);
switch (prop_id) {
/*case ARG_VOLUME:
g_value_set_double (value, sink->volume);
break;*/
case ARG_DEVICE:
if (!src->device_id)
device_set_default (src);
g_value_set_string (value, src->device_id);
break;
case ARG_DEVICE_NAME:
if (!src->device_name)
device_get_name (src);
g_value_set_string (value, src->device_name);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_directsound_src_finalize (GObject * object)
{
GstDirectSoundSrc *dsoundsrc = GST_DIRECTSOUND_SRC (object);
g_mutex_clear (&dsoundsrc->dsound_lock);
g_free (dsoundsrc->device_name);
g_free (dsoundsrc->device_guid);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_directsound_src_unprepare (GstAudioSrc * asrc)
{
GstDirectSoundSrc *dsoundsrc;
GST_DEBUG_OBJECT (asrc, "unpreparing directsoundsrc");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
/* Stop capturing */
IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
/* Release buffer */
IDirectSoundCaptureBuffer_Release (dsoundsrc->pDSBSecondary);
return TRUE;
}
static void
gst_directsound_src_dispose (GObject * object)
{
GstDirectSoundSrc * self = GST_DIRECTSOUND_SRC (object);
GST_DEBUG_OBJECT (object, G_STRFUNC);
if (self->device_id) {
g_free (self->device_id);
self->device_id = NULL;
}
if (self->device_name) {
g_free (self->device_name);
self->device_name = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_directsound_src_finalize (GObject * object)
{
GstDirectSoundSrc *dsoundsrc = GST_DIRECTSOUND_SRC (object);
g_mutex_clear (&dsoundsrc->dsound_lock);
gst_object_unref (dsoundsrc->system_clock);
if (dsoundsrc->read_wait_clock_id != NULL)
gst_clock_id_unref (dsoundsrc->read_wait_clock_id);
g_free (dsoundsrc->device_name);
g_free (dsoundsrc->device_id);
g_free (dsoundsrc->device_guid);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_directsound_src_reset (GstAudioSrc * asrc)
{
GstDirectSoundSrc *dsoundsrc;
LPVOID pLockedBuffer = NULL;
DWORD dwSizeBuffer = 0;
GST_DEBUG_OBJECT (asrc, "reset directsoundsrc");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DSOUND_LOCK (dsoundsrc);
dsoundsrc->reset_while_sleeping = TRUE;
/* Interrupt read sleep if required */
if (dsoundsrc->read_wait_clock_id != NULL)
gst_clock_id_unschedule (dsoundsrc->read_wait_clock_id);
if (dsoundsrc->pDSBSecondary) {
/*stop capturing */
HRESULT hRes = IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
/*reset position */
/* hRes = IDirectSoundCaptureBuffer_SetCurrentPosition (dsoundsrc->pDSBSecondary, 0); */
/*reset the buffer */
hRes = IDirectSoundCaptureBuffer_Lock (dsoundsrc->pDSBSecondary,
dsoundsrc->current_circular_offset, dsoundsrc->buffer_size,
pLockedBuffer, &dwSizeBuffer, NULL, NULL, 0L);
if (SUCCEEDED (hRes)) {
memset (pLockedBuffer, 0, dwSizeBuffer);
hRes =
IDirectSoundCaptureBuffer_Unlock (dsoundsrc->pDSBSecondary,
pLockedBuffer, dwSizeBuffer, NULL, 0);
}
dsoundsrc->current_circular_offset = 0;
}
GST_DSOUND_UNLOCK (dsoundsrc);
}
static void
gst_directsound_src_reset (GstAudioSrc * asrc)
{
GstDirectSoundSrc *dsoundsrc;
LPVOID pLockedBuffer = NULL;
DWORD dwSizeBuffer = 0;
GST_DEBUG_OBJECT (asrc, "reset directsoundsrc");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
#if 0
IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
#endif
GST_DSOUND_LOCK (dsoundsrc);
if (dsoundsrc->pDSBSecondary) {
/*stop capturing */
HRESULT hRes = IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
/*reset position */
/* hRes = IDirectSoundCaptureBuffer_SetCurrentPosition (dsoundsrc->pDSBSecondary, 0); */
/*reset the buffer */
hRes = IDirectSoundCaptureBuffer_Lock (dsoundsrc->pDSBSecondary,
dsoundsrc->current_circular_offset, dsoundsrc->buffer_size,
pLockedBuffer, &dwSizeBuffer, NULL, NULL, 0L);
if (SUCCEEDED (hRes)) {
memset (pLockedBuffer, 0, dwSizeBuffer);
hRes =
IDirectSoundCaptureBuffer_Unlock (dsoundsrc->pDSBSecondary,
pLockedBuffer, dwSizeBuffer, NULL, 0);
}
dsoundsrc->current_circular_offset = 0;
}
GST_DSOUND_UNLOCK (dsoundsrc);
}
static gboolean
gst_directsound_src_close (GstAudioSrc * asrc)
{
GstDirectSoundSrc *dsoundsrc;
GST_DEBUG_OBJECT (asrc, "closing directsoundsrc");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
/* Release capture handler */
IDirectSoundCapture_Release (dsoundsrc->pDSC);
/* Close library */
FreeLibrary (dsoundsrc->DSoundDLL);
if (dsoundsrc->mixer)
mixerClose (dsoundsrc->mixer);
return TRUE;
}
/* Enumeration callback called by DirectSoundCaptureEnumerate.
* Gets the GUID of request audio device
*/
static BOOL CALLBACK
gst_directsound_enum_callback (GUID * pGUID, TCHAR * strDesc,
TCHAR * strDrvName, VOID * pContext)
{
GstDirectSoundSrc *dsoundsrc = GST_DIRECTSOUND_SRC (pContext);
if (pGUID && dsoundsrc && dsoundsrc->device_name &&
!g_strcmp0 (dsoundsrc->device_name, strDesc)) {
g_free (dsoundsrc->device_guid);
dsoundsrc->device_guid = (GUID *) g_malloc0 (sizeof (GUID));
memcpy (dsoundsrc->device_guid, pGUID, sizeof (GUID));
GST_INFO_OBJECT (dsoundsrc, "found the requested audio device :%s",
dsoundsrc->device_name);
return FALSE;
}
GST_INFO_OBJECT (dsoundsrc, "sound device names: %s, %s, requested device:%s",
strDesc, strDrvName, dsoundsrc->device_name);
return TRUE;
}
/* Enumeration callback called by DirectSoundCaptureEnumerate.
* Gets the GUID of request audio device
*/
static BOOL CALLBACK
gst_directsound_enum_callback (GUID * pGUID, TCHAR * strDesc,
TCHAR * strDrvName, VOID * pContext)
{
GstDirectSoundSrc *dsoundsrc = GST_DIRECTSOUND_SRC (pContext);
gchar *driver, *description;
description = g_locale_to_utf8 (strDesc, -1, NULL, NULL, NULL);
if (!description) {
GST_ERROR_OBJECT (dsoundsrc,
"Failed to convert description from locale encoding to UTF8");
return TRUE;
}
driver = g_locale_to_utf8 (strDrvName, -1, NULL, NULL, NULL);
if (pGUID && dsoundsrc && dsoundsrc->device_name &&
!g_strcmp0 (dsoundsrc->device_name, description)) {
g_free (dsoundsrc->device_guid);
dsoundsrc->device_guid = (GUID *) g_malloc0 (sizeof (GUID));
memcpy (dsoundsrc->device_guid, pGUID, sizeof (GUID));
GST_INFO_OBJECT (dsoundsrc, "found the requested audio device :%s",
dsoundsrc->device_name);
g_free (description);
g_free (driver);
return FALSE;
}
GST_INFO_OBJECT (dsoundsrc, "sound device names: %s, %s, requested device:%s",
description, driver, dsoundsrc->device_name);
g_free (description);
g_free (driver);
return TRUE;
}
/*
return number of readed bytes */
static guint
gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length,
GstClockTime * timestamp)
{
GstDirectSoundSrc *dsoundsrc;
guint64 sleep_time_ms, sleep_until;
GstClockID clock_id;
HRESULT hRes; /* Result for windows functions */
DWORD dwCurrentCaptureCursor = 0;
DWORD dwBufferSize = 0;
LPVOID pLockedBuffer1 = NULL;
LPVOID pLockedBuffer2 = NULL;
DWORD dwSizeBuffer1 = 0;
DWORD dwSizeBuffer2 = 0;
DWORD dwStatus = 0;
GST_DEBUG_OBJECT (asrc, "reading directsoundsrc");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DSOUND_LOCK (dsoundsrc);
/* Get current buffer status */
hRes = IDirectSoundCaptureBuffer_GetStatus (dsoundsrc->pDSBSecondary,
&dwStatus);
if (FAILED (hRes)) {
GST_DSOUND_UNLOCK (dsoundsrc);
return -1;
}
/* Starting capturing if not already */
if (!(dwStatus & DSCBSTATUS_CAPTURING)) {
hRes = IDirectSoundCaptureBuffer_Start (dsoundsrc->pDSBSecondary,
DSCBSTART_LOOPING);
GST_INFO_OBJECT (asrc, "capture started");
}
/* Loop till the source has produced bytes equal to or greater than @length.
*
* DirectSound has a notification-based API that uses Windows CreateEvent()
* + WaitForSingleObject(), but it is completely useless for live streams.
*
* 1. You must schedule all events before starting capture
* 2. The events are all fired exactly once
* 3. You cannot schedule new events while a capture is running
* 4. You cannot stop/schedule/start either
*
* This means you cannot use the API while doing live looped capture and we
* must resort to this.
*
* However, this is almost as efficient as event-based capture since it's ok
* to consistently overwait by a fixed amount; the extra bytes will just end
* up being used in the next call, and the extra latency will be constant. */
while (TRUE) {
hRes =
IDirectSoundCaptureBuffer_GetCurrentPosition (dsoundsrc->pDSBSecondary,
&dwCurrentCaptureCursor, NULL);
if (FAILED (hRes)) {
GST_DSOUND_UNLOCK (dsoundsrc);
return -1;
}
/* calculate the size of the buffer that's been captured while accounting
* for wrap-arounds */
if (dwCurrentCaptureCursor < dsoundsrc->current_circular_offset) {
dwBufferSize = dsoundsrc->buffer_size -
(dsoundsrc->current_circular_offset - dwCurrentCaptureCursor);
} else {
dwBufferSize =
dwCurrentCaptureCursor - dsoundsrc->current_circular_offset;
}
if (dwBufferSize >= length) {
/* Yay, we got all the data we need */
break;
} else {
GST_DEBUG_OBJECT (asrc, "not enough data, got %lu (want at least %u)",
dwBufferSize, length);
/* If we didn't get enough data, sleep for a proportionate time */
sleep_time_ms = gst_util_uint64_scale (dsoundsrc->latency_time,
length - dwBufferSize, length * 1000);
/* Make sure we don't run in a tight loop unnecessarily */
sleep_time_ms = MAX (sleep_time_ms, 10);
/* Sleep using gst_clock_id_wait() so that we can be interrupted */
sleep_until = gst_clock_get_time (dsoundsrc->system_clock) +
sleep_time_ms * GST_MSECOND;
/* Setup the clock id wait */
if (G_UNLIKELY (dsoundsrc->read_wait_clock_id == NULL ||
gst_clock_single_shot_id_reinit (dsoundsrc->system_clock,
dsoundsrc->read_wait_clock_id, sleep_until) == FALSE)) {
if (dsoundsrc->read_wait_clock_id != NULL)
gst_clock_id_unref (dsoundsrc->read_wait_clock_id);
dsoundsrc->read_wait_clock_id =
gst_clock_new_single_shot_id (dsoundsrc->system_clock, sleep_until);
}
clock_id = dsoundsrc->read_wait_clock_id;
dsoundsrc->reset_while_sleeping = FALSE;
GST_DEBUG_OBJECT (asrc, "waiting %" G_GUINT64_FORMAT "ms for more data",
sleep_time_ms);
GST_DSOUND_UNLOCK (dsoundsrc);
gst_clock_id_wait (clock_id, NULL);
GST_DSOUND_LOCK (dsoundsrc);
if (dsoundsrc->reset_while_sleeping == TRUE) {
GST_DEBUG_OBJECT (asrc, "reset while sleeping, cancelled read");
GST_DSOUND_UNLOCK (dsoundsrc);
return -1;
}
}
}
GST_DEBUG_OBJECT (asrc, "Got enough data: %lu bytes (wanted at least %u)",
dwBufferSize, length);
/* Lock the buffer and read only the first @length bytes. Keep the rest in
* the capture buffer for the next read. */
hRes = IDirectSoundCaptureBuffer_Lock (dsoundsrc->pDSBSecondary,
dsoundsrc->current_circular_offset,
length,
&pLockedBuffer1, &dwSizeBuffer1, &pLockedBuffer2, &dwSizeBuffer2, 0L);
/* NOTE: We now assume that dwSizeBuffer1 + dwSizeBuffer2 == length since the
* API is supposed to guarantee that */
/* Copy buffer data to another buffer */
if (hRes == DS_OK) {
memcpy (data, pLockedBuffer1, dwSizeBuffer1);
}
/* ...and if something is in another buffer */
if (pLockedBuffer2 != NULL) {
memcpy (((guchar *) data + dwSizeBuffer1), pLockedBuffer2, dwSizeBuffer2);
}
dsoundsrc->current_circular_offset += dwSizeBuffer1 + dwSizeBuffer2;
dsoundsrc->current_circular_offset %= dsoundsrc->buffer_size;
IDirectSoundCaptureBuffer_Unlock (dsoundsrc->pDSBSecondary,
pLockedBuffer1, dwSizeBuffer1, pLockedBuffer2, dwSizeBuffer2);
GST_DSOUND_UNLOCK (dsoundsrc);
/* We always read exactly @length data */
return length;
}
/*
return number of readed bytes */
static guint
gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length,
GstClockTime * timestamp)
{
GstDirectSoundSrc *dsoundsrc;
HRESULT hRes; /* Result for windows functions */
DWORD dwCurrentCaptureCursor = 0;
DWORD dwBufferSize = 0;
LPVOID pLockedBuffer1 = NULL;
LPVOID pLockedBuffer2 = NULL;
DWORD dwSizeBuffer1 = 0;
DWORD dwSizeBuffer2 = 0;
DWORD dwStatus = 0;
GST_DEBUG_OBJECT (asrc, "reading directsoundsrc");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DSOUND_LOCK (dsoundsrc);
/* Get current buffer status */
hRes = IDirectSoundCaptureBuffer_GetStatus (dsoundsrc->pDSBSecondary,
&dwStatus);
/* Starting capturing if not already */
if (!(dwStatus & DSCBSTATUS_CAPTURING)) {
hRes = IDirectSoundCaptureBuffer_Start (dsoundsrc->pDSBSecondary,
DSCBSTART_LOOPING);
// Sleep (dsoundsrc->latency_time/1000);
GST_DEBUG_OBJECT (asrc, "capture started");
}
// calculate_buffersize:
while (length > dwBufferSize) {
Sleep (dsoundsrc->latency_time / 1000);
hRes =
IDirectSoundCaptureBuffer_GetCurrentPosition (dsoundsrc->pDSBSecondary,
&dwCurrentCaptureCursor, NULL);
/* calculate the buffer */
if (dwCurrentCaptureCursor < dsoundsrc->current_circular_offset) {
dwBufferSize = dsoundsrc->buffer_size -
(dsoundsrc->current_circular_offset - dwCurrentCaptureCursor);
} else {
dwBufferSize =
dwCurrentCaptureCursor - dsoundsrc->current_circular_offset;
}
} // while (...
/* Lock the buffer */
hRes = IDirectSoundCaptureBuffer_Lock (dsoundsrc->pDSBSecondary,
dsoundsrc->current_circular_offset,
length,
&pLockedBuffer1, &dwSizeBuffer1, &pLockedBuffer2, &dwSizeBuffer2, 0L);
/* Copy buffer data to another buffer */
if (hRes == DS_OK) {
memcpy (data, pLockedBuffer1, dwSizeBuffer1);
}
/* ...and if something is in another buffer */
if (pLockedBuffer2 != NULL) {
memcpy (((guchar *) data + dwSizeBuffer1), pLockedBuffer2, dwSizeBuffer2);
}
dsoundsrc->current_circular_offset += dwSizeBuffer1 + dwSizeBuffer2;
dsoundsrc->current_circular_offset %= dsoundsrc->buffer_size;
IDirectSoundCaptureBuffer_Unlock (dsoundsrc->pDSBSecondary,
pLockedBuffer1, dwSizeBuffer1, pLockedBuffer2, dwSizeBuffer2);
GST_DSOUND_UNLOCK (dsoundsrc);
/* return length (readed data size in bytes) */
return length;
}
static gboolean
gst_directsound_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstDirectSoundSrc *dsoundsrc;
WAVEFORMATEX wfx; /* Wave format structure */
HRESULT hRes; /* Result for windows functions */
DSCBUFFERDESC descSecondary; /* Capturebuffer description */
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DEBUG_OBJECT (asrc, "preparing directsoundsrc");
/* Define buffer */
memset (&wfx, 0, sizeof (WAVEFORMATEX));
wfx.wFormatTag = WAVE_FORMAT_PCM;
wfx.nChannels = GST_AUDIO_INFO_CHANNELS (&spec->info);
wfx.nSamplesPerSec = GST_AUDIO_INFO_RATE (&spec->info);
wfx.wBitsPerSample = GST_AUDIO_INFO_BPF (&spec->info) * 8 / wfx.nChannels;
wfx.nBlockAlign = GST_AUDIO_INFO_BPF (&spec->info);
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
/* Ignored for WAVE_FORMAT_PCM. */
wfx.cbSize = 0;
if (wfx.wBitsPerSample != 16 && wfx.wBitsPerSample != 8)
goto dodgy_width;
/* Set the buffer size to two seconds.
This should never reached.
*/
dsoundsrc->buffer_size = wfx.nAvgBytesPerSec * 2;
GST_DEBUG_OBJECT (asrc, "Buffer size: %d", dsoundsrc->buffer_size);
/* Init secondary buffer desciption */
memset (&descSecondary, 0, sizeof (DSCBUFFERDESC));
descSecondary.dwSize = sizeof (DSCBUFFERDESC);
descSecondary.dwFlags = 0;
descSecondary.dwReserved = 0;
/* This is not primary buffer so have to set size */
descSecondary.dwBufferBytes = dsoundsrc->buffer_size;
descSecondary.lpwfxFormat = &wfx;
/* Create buffer */
hRes = IDirectSoundCapture_CreateCaptureBuffer (dsoundsrc->pDSC,
&descSecondary, &dsoundsrc->pDSBSecondary, NULL);
if (hRes != DS_OK)
goto capture_buffer;
dsoundsrc->bytes_per_sample = GST_AUDIO_INFO_BPF (&spec->info);
GST_DEBUG ("latency time: %" G_GUINT64_FORMAT " - buffer time: %"
G_GUINT64_FORMAT, spec->latency_time, spec->buffer_time);
/* Buffer-time should be always more than 2*latency */
if (spec->buffer_time < spec->latency_time * 2) {
spec->buffer_time = spec->latency_time * 2;
GST_WARNING ("buffer-time was less than latency");
}
/* Save the times */
dsoundsrc->buffer_time = spec->buffer_time;
dsoundsrc->latency_time = spec->latency_time;
dsoundsrc->latency_size = (gint) wfx.nAvgBytesPerSec *
dsoundsrc->latency_time / 1000000.0;
spec->segsize = (guint) (((double) spec->buffer_time / 1000000.0) *
wfx.nAvgBytesPerSec);
/* just in case */
if (spec->segsize < 1)
spec->segsize = 1;
spec->segtotal = GST_AUDIO_INFO_BPF (&spec->info) * 8 *
(wfx.nAvgBytesPerSec / spec->segsize);
GST_DEBUG_OBJECT (asrc,
"bytes/sec: %lu, buffer size: %d, segsize: %d, segtotal: %d",
wfx.nAvgBytesPerSec, dsoundsrc->buffer_size, spec->segsize,
spec->segtotal);
/* Not read anything yet */
dsoundsrc->current_circular_offset = 0;
GST_DEBUG_OBJECT (asrc, "channels: %d, rate: %d, bytes_per_sample: %d"
" WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d,"
" WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld",
GST_AUDIO_INFO_CHANNELS (&spec->info), GST_AUDIO_INFO_RATE (&spec->info),
GST_AUDIO_INFO_BPF (&spec->info), wfx.nSamplesPerSec, wfx.wBitsPerSample,
wfx.nBlockAlign, wfx.nAvgBytesPerSec);
return TRUE;
capture_buffer:
{
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to create capturebuffer"), (NULL));
return FALSE;
}
dodgy_width:
{
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unexpected width %d", wfx.wBitsPerSample), (NULL));
return FALSE;
}
}
static gboolean
gst_directsound_src_open (GstAudioSrc * asrc)
{
GstDirectSoundSrc *dsoundsrc;
HRESULT hRes; /* Result for windows functions */
GST_DEBUG_OBJECT (asrc, "opening directsoundsrc");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
/* Open dsound.dll */
dsoundsrc->DSoundDLL = LoadLibrary ("dsound.dll");
if (!dsoundsrc->DSoundDLL) {
goto dsound_open;
}
/* Building the DLL Calls */
pDSoundCaptureCreate =
(void *) GetProcAddress (dsoundsrc->DSoundDLL,
TEXT ("DirectSoundCaptureCreate"));
/* If everything is not ok */
if (!pDSoundCaptureCreate) {
goto capture_function;
}
hRes = DirectSoundCaptureEnumerate ((LPDSENUMCALLBACK)
gst_directsound_enum_callback, (VOID *) dsoundsrc);
if (FAILED (hRes)) {
goto capture_enumerate;
}
/* Create capture object */
hRes = pDSoundCaptureCreate (dsoundsrc->device_guid, &dsoundsrc->pDSC, NULL);
if (FAILED (hRes)) {
goto capture_object;
}
gst_directsound_src_mixer_init (dsoundsrc);
return TRUE;
capture_function:
{
FreeLibrary (dsoundsrc->DSoundDLL);
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to get capturecreate function"), (NULL));
return FALSE;
}
capture_enumerate:
{
FreeLibrary (dsoundsrc->DSoundDLL);
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to enumerate audio capture devices"), (NULL));
return FALSE;
}
capture_object:
{
FreeLibrary (dsoundsrc->DSoundDLL);
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to create capture object"), (NULL));
return FALSE;
}
dsound_open:
{
DWORD err = GetLastError ();
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to open dsound.dll"), (NULL));
g_print ("0x%lx\n", HRESULT_FROM_WIN32 (err));
return FALSE;
}
}
static gboolean
gst_directsound_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstDirectSoundSrc *dsoundsrc;
WAVEFORMATEX wfx; /* Wave format structure */
HRESULT hRes; /* Result for windows functions */
DSCBUFFERDESC descSecondary; /* Capturebuffer description */
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DEBUG_OBJECT (asrc, "preparing directsoundsrc");
/* Define buffer */
memset (&wfx, 0, sizeof (WAVEFORMATEX));
wfx.wFormatTag = WAVE_FORMAT_PCM;
wfx.nChannels = GST_AUDIO_INFO_CHANNELS (&spec->info);
wfx.nSamplesPerSec = GST_AUDIO_INFO_RATE (&spec->info);
wfx.wBitsPerSample = GST_AUDIO_INFO_BPF (&spec->info) * 8 / wfx.nChannels;
wfx.nBlockAlign = GST_AUDIO_INFO_BPF (&spec->info);
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
/* Ignored for WAVE_FORMAT_PCM. */
wfx.cbSize = 0;
if (wfx.wBitsPerSample != 16 && wfx.wBitsPerSample != 8)
goto dodgy_width;
GST_INFO_OBJECT (asrc, "latency time: %" G_GUINT64_FORMAT " - buffer time: %"
G_GUINT64_FORMAT, spec->latency_time, spec->buffer_time);
/* Buffer-time should always be >= 2*latency */
if (spec->buffer_time < spec->latency_time * 2) {
spec->buffer_time = spec->latency_time * 2;
GST_WARNING ("buffer-time was less than 2*latency-time, clamping");
}
/* Set the buffer size from our configured buffer time (in microsecs) */
dsoundsrc->buffer_size =
gst_util_uint64_scale_int (spec->buffer_time, wfx.nAvgBytesPerSec,
GST_SECOND / GST_USECOND);
GST_INFO_OBJECT (asrc, "Buffer size: %d", dsoundsrc->buffer_size);
spec->segsize =
gst_util_uint64_scale (spec->latency_time, wfx.nAvgBytesPerSec,
GST_SECOND / GST_USECOND);
/* Sanitized segsize */
if (spec->segsize < GST_AUDIO_INFO_BPF (&spec->info))
spec->segsize = GST_AUDIO_INFO_BPF (&spec->info);
else if (spec->segsize % GST_AUDIO_INFO_BPF (&spec->info) != 0)
spec->segsize =
((spec->segsize + GST_AUDIO_INFO_BPF (&spec->info) -
1) / GST_AUDIO_INFO_BPF (&spec->info)) *
GST_AUDIO_INFO_BPF (&spec->info);
spec->segtotal = dsoundsrc->buffer_size / spec->segsize;
/* The device usually takes time = 1-2 segments to start producing buffers */
spec->seglatency = spec->segtotal + 2;
/* Fetch and set the actual latency time that will be used */
dsoundsrc->latency_time =
gst_util_uint64_scale (spec->segsize, GST_SECOND / GST_USECOND,
GST_AUDIO_INFO_BPF (&spec->info) * GST_AUDIO_INFO_RATE (&spec->info));
GST_INFO_OBJECT (asrc, "actual latency time: %" G_GUINT64_FORMAT,
spec->latency_time);
/* Init secondary buffer desciption */
memset (&descSecondary, 0, sizeof (DSCBUFFERDESC));
descSecondary.dwSize = sizeof (DSCBUFFERDESC);
descSecondary.dwFlags = 0;
descSecondary.dwReserved = 0;
/* This is not primary buffer so have to set size */
descSecondary.dwBufferBytes = dsoundsrc->buffer_size;
descSecondary.lpwfxFormat = &wfx;
/* Create buffer */
hRes = IDirectSoundCapture_CreateCaptureBuffer (dsoundsrc->pDSC,
&descSecondary, &dsoundsrc->pDSBSecondary, NULL);
if (hRes != DS_OK)
goto capture_buffer;
dsoundsrc->bytes_per_sample = GST_AUDIO_INFO_BPF (&spec->info);
GST_INFO_OBJECT (asrc,
"bytes/sec: %lu, buffer size: %d, segsize: %d, segtotal: %d",
wfx.nAvgBytesPerSec, dsoundsrc->buffer_size, spec->segsize,
spec->segtotal);
/* Not read anything yet */
dsoundsrc->current_circular_offset = 0;
GST_INFO_OBJECT (asrc, "channels: %d, rate: %d, bytes_per_sample: %d"
" WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d,"
" WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld",
GST_AUDIO_INFO_CHANNELS (&spec->info), GST_AUDIO_INFO_RATE (&spec->info),
GST_AUDIO_INFO_BPF (&spec->info), wfx.nSamplesPerSec, wfx.wBitsPerSample,
wfx.nBlockAlign, wfx.nAvgBytesPerSec);
return TRUE;
capture_buffer:
{
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to create capturebuffer"), (NULL));
return FALSE;
}
dodgy_width:
{
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unexpected width %d", wfx.wBitsPerSample), (NULL));
return FALSE;
}
}
static gboolean
gst_directsound_src_event (GstBaseSrc * bsrc, GstEvent * event)
{
HRESULT hr;
DWORD dwStatus;
//DWORD dwSizeBuffer = 0;
//LPVOID pLockedBuffer = NULL;
GstDirectSoundSrc * dsoundsrc;
dsoundsrc = GST_DIRECTSOUND_SRC (bsrc);
GST_BASE_SRC_CLASS (parent_class)->event (bsrc, event);
/* no buffer, no event to process */
if (!dsoundsrc->dsoundbuffer)
return TRUE;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
GST_DSOUND_LOCK (dsoundsrc->dsoundbuffer);
dsoundsrc->dsoundbuffer->flushing = TRUE;
GST_DSOUND_UNLOCK (dsoundsrc->dsoundbuffer);
break;
case GST_EVENT_FLUSH_STOP:
GST_DSOUND_LOCK (dsoundsrc->dsoundbuffer);
dsoundsrc->dsoundbuffer->flushing = FALSE;
if (dsoundsrc->dsoundbuffer->pDSCB8) {
hr = IDirectSoundCaptureBuffer8_GetStatus (dsoundsrc->dsoundbuffer->pDSCB8, &dwStatus);
if (FAILED(hr)) {
GST_DSOUND_UNLOCK (dsoundsrc->dsoundbuffer);
GST_WARNING("gst_directsound_src_event: IDirectSoundCaptureBuffer8_GetStatus, hr = %X", (unsigned int) hr);
return FALSE;
}
if (!(dwStatus & DSCBSTATUS_CAPTURING)) {
// ###: capture api doesn't support _SetCurrentPosition. commenting
// out for now.
#if 0
/* reset position */
hr = IDirectSoundBuffer8_SetCurrentPosition (dsoundsink->dsoundbuffer->pDSB8, 0);
dsoundsink->dsoundbuffer->buffer_circular_offset = 0;
/* reset the buffer */
hr = IDirectSoundBuffer8_Lock (dsoundsink->dsoundbuffer->pDSB8,
dsoundsink->dsoundbuffer->buffer_circular_offset, 0L,
&pLockedBuffer, &dwSizeBuffer, NULL, NULL, DSBLOCK_ENTIREBUFFER);
if (SUCCEEDED (hr)) {
memset (pLockedBuffer, 0, dwSizeBuffer);
hr =
IDirectSoundBuffer8_Unlock (dsoundsink->dsoundbuffer->pDSB8, pLockedBuffer,
dwSizeBuffer, NULL, 0);
if (FAILED(hr)) {
GST_DSOUND_UNLOCK (dsoundsink->dsoundbuffer);
GST_WARNING("gst_directsound_sink_event: IDirectSoundBuffer8_Unlock, hr = %X", (unsigned int) hr);
return FALSE;
}
}
else {
GST_DSOUND_UNLOCK (dsoundsink->dsoundbuffer);
GST_WARNING ( "gst_directsound_sink_event: IDirectSoundBuffer8_Lock, hr = %X", (unsigned int) hr);
return FALSE;
}
#endif
}
}
GST_DSOUND_UNLOCK (dsoundsrc->dsoundbuffer);
break;
default:
break;
}
return TRUE;
}
