gstreamer/sys/directsound/gstdirectsoundsink.c
Stefan Kost 8e462968cf Remove version numbers from a few gst-launch examples.
The majority of the examples doe not use -0.10 and this will also help us to maintain the docs.
2009-01-29 11:07:59 +02:00

658 lines
21 KiB
C

/* GStreamer
* Copyright (C) 2005 Sebastien Moutte <sebastien@moutte.net>
* Copyright (C) 2007 Pioneers of the Inevitable <songbird@songbirdnest.com>
*
* gstdirectsoundsink.c:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*
*
* The development of this code was made possible due to the involvement
* of Pioneers of the Inevitable, the creators of the Songbird Music player
*
*/
/**
* SECTION:element-directsoundsink
*
* This element lets you output sound using the DirectSound API.
*
* Note that you should almost always use generic audio conversion elements
* like audioconvert and audioresample in front of an audiosink to make sure
* your pipeline works under all circumstances (those conversion elements will
* act in passthrough-mode if no conversion is necessary).
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.1 ! directsoundsink
* ]| will output a sine wave (continuous beep sound) to your sound card (with
* a very low volume as precaution).
* |[
* gst-launch -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! directsoundsink
* ]| will play an Ogg/Vorbis audio file and output it.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstdirectsoundsink.h"
#include <math.h>
GST_DEBUG_CATEGORY_STATIC (directsoundsink_debug);
/* elementfactory information */
static const GstElementDetails gst_directsound_sink_details =
GST_ELEMENT_DETAILS ("Direct Sound Audio Sink",
"Sink/Audio",
"Output to a sound card via Direct Sound",
"Sebastien Moutte <sebastien@moutte.net>");
static void gst_directsound_sink_base_init (gpointer g_class);
static void gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass);
static void gst_directsound_sink_init (GstDirectSoundSink * dsoundsink,
GstDirectSoundSinkClass * g_class);
static void gst_directsound_sink_finalise (GObject * object);
static void gst_directsound_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_directsound_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_directsound_sink_getcaps (GstBaseSink * bsink);
static gboolean gst_directsound_sink_prepare (GstAudioSink * asink,
GstRingBufferSpec * spec);
static gboolean gst_directsound_sink_unprepare (GstAudioSink * asink);
static gboolean gst_directsound_sink_open (GstAudioSink * asink);
static gboolean gst_directsound_sink_close (GstAudioSink * asink);
static guint gst_directsound_sink_write (GstAudioSink * asink, gpointer data,
guint length);
static guint gst_directsound_sink_delay (GstAudioSink * asink);
static void gst_directsound_sink_reset (GstAudioSink * asink);
/* interfaces */
static void gst_directsound_sink_interfaces_init (GType type);
static void
gst_directsound_sink_implements_interface_init (GstImplementsInterfaceClass *
iface);
static void gst_directsound_sink_mixer_interface_init (GstMixerClass * iface);
static GstStaticPadTemplate directsoundsink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
"audio/x-raw-int, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) 8, "
"depth = (int) 8, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]"));
enum
{
PROP_0,
PROP_VOLUME
};
GST_BOILERPLATE_FULL (GstDirectSoundSink, gst_directsound_sink, GstAudioSink,
GST_TYPE_AUDIO_SINK, gst_directsound_sink_interfaces_init);
/* interfaces stuff */
static void
gst_directsound_sink_interfaces_init (GType type)
{
static const GInterfaceInfo implements_interface_info = {
(GInterfaceInitFunc) gst_directsound_sink_implements_interface_init,
NULL,
NULL,
};
static const GInterfaceInfo mixer_interface_info = {
(GInterfaceInitFunc) gst_directsound_sink_mixer_interface_init,
NULL,
NULL,
};
g_type_add_interface_static (type,
GST_TYPE_IMPLEMENTS_INTERFACE, &implements_interface_info);
g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_interface_info);
}
static gboolean
gst_directsound_sink_interface_supported (GstImplementsInterface * iface,
GType iface_type)
{
g_return_val_if_fail (iface_type == GST_TYPE_MIXER, FALSE);
/* for the sake of this example, we'll always support it. However, normally,
* you would check whether the device you've opened supports mixers. */
return TRUE;
}
static void
gst_directsound_sink_implements_interface_init (GstImplementsInterfaceClass *
iface)
{
iface->supported = gst_directsound_sink_interface_supported;
}
/*
* This function returns the list of support tracks (inputs, outputs)
* on this element instance. Elements usually build this list during
* _init () or when going from NULL to READY.
*/
static const GList *
gst_directsound_sink_mixer_list_tracks (GstMixer * mixer)
{
GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (mixer);
return dsoundsink->tracks;
}
static void
gst_directsound_sink_set_volume (GstDirectSoundSink * dsoundsink)
{
if (dsoundsink->pDSBSecondary) {
/* DirectSound controls volume using units of 100th of a decibel,
* ranging from -10000 to 0. We use a linear scale of 0 - 100
* here, so remap.
*/
long dsVolume;
if (dsoundsink->volume == 0)
dsVolume = -10000;
else
dsVolume = 100 * (long) (20 * log10 ((double) dsoundsink->volume / 100.));
dsVolume = CLAMP (dsVolume, -10000, 0);
GST_DEBUG_OBJECT (dsoundsink,
"Setting volume on secondary buffer to %d from %d", (int) dsVolume,
(int) dsoundsink->volume);
IDirectSoundBuffer_SetVolume (dsoundsink->pDSBSecondary, dsVolume);
}
}
/*
* Set volume. volumes is an array of size track->num_channels, and
* each value in the array gives the wanted volume for one channel
* on the track.
*/
static void
gst_directsound_sink_mixer_set_volume (GstMixer * mixer,
GstMixerTrack * track, gint * volumes)
{
GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (mixer);
if (volumes[0] != dsoundsink->volume) {
dsoundsink->volume = volumes[0];
gst_directsound_sink_set_volume (dsoundsink);
}
}
static void
gst_directsound_sink_mixer_get_volume (GstMixer * mixer,
GstMixerTrack * track, gint * volumes)
{
GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (mixer);
volumes[0] = dsoundsink->volume;
}
static void
gst_directsound_sink_mixer_interface_init (GstMixerClass * iface)
{
/* the mixer interface requires a definition of the mixer type:
* hardware or software? */
GST_MIXER_TYPE (iface) = GST_MIXER_SOFTWARE;
/* virtual function pointers */
iface->list_tracks = gst_directsound_sink_mixer_list_tracks;
iface->set_volume = gst_directsound_sink_mixer_set_volume;
iface->get_volume = gst_directsound_sink_mixer_get_volume;
}
static void
gst_directsound_sink_finalise (GObject * object)
{
GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (object);
g_mutex_free (dsoundsink->dsound_lock);
if (dsoundsink->tracks) {
g_list_foreach (dsoundsink->tracks, (GFunc) g_object_unref, NULL);
g_list_free (dsoundsink->tracks);
dsoundsink->tracks = NULL;
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_directsound_sink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &gst_directsound_sink_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&directsoundsink_sink_factory));
}
static void
gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstBaseAudioSinkClass *gstbaseaudiosink_class;
GstAudioSinkClass *gstaudiosink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
gstaudiosink_class = (GstAudioSinkClass *) klass;
GST_DEBUG_CATEGORY_INIT (directsoundsink_debug, "directsoundsink", 0,
"DirectSound sink");
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_directsound_sink_finalise);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_directsound_sink_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_directsound_sink_get_property);
gstbasesink_class->get_caps =
GST_DEBUG_FUNCPTR (gst_directsound_sink_getcaps);
gstaudiosink_class->prepare =
GST_DEBUG_FUNCPTR (gst_directsound_sink_prepare);
gstaudiosink_class->unprepare =
GST_DEBUG_FUNCPTR (gst_directsound_sink_unprepare);
gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_directsound_sink_open);
gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_directsound_sink_close);
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_directsound_sink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_directsound_sink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_directsound_sink_reset);
g_object_class_install_property (gobject_class,
PROP_VOLUME,
g_param_spec_double ("volume", "Volume",
"Volume of this stream", 0.0, 1.0, 1.0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_directsound_sink_init (GstDirectSoundSink * dsoundsink,
GstDirectSoundSinkClass * g_class)
{
GstMixerTrack *track = NULL;
dsoundsink->tracks = NULL;
track = g_object_new (GST_TYPE_MIXER_TRACK, NULL);
track->label = g_strdup ("DSoundTrack");
track->num_channels = 2;
track->min_volume = 0;
track->max_volume = 100;
track->flags = GST_MIXER_TRACK_OUTPUT;
dsoundsink->tracks = g_list_append (dsoundsink->tracks, track);
dsoundsink->pDS = NULL;
dsoundsink->pDSBSecondary = NULL;
dsoundsink->current_circular_offset = 0;
dsoundsink->buffer_size = DSBSIZE_MIN;
dsoundsink->volume = 100;
dsoundsink->dsound_lock = g_mutex_new ();
dsoundsink->first_buffer_after_reset = FALSE;
}
static void
gst_directsound_sink_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstDirectSoundSink *sink = GST_DIRECTSOUND_SINK (object);
switch (prop_id) {
case PROP_VOLUME:
sink->volume = (int) (g_value_get_double (value) * 100);
gst_directsound_sink_set_volume (sink);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_directsound_sink_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstDirectSoundSink *sink = GST_DIRECTSOUND_SINK (object);
switch (prop_id) {
case PROP_VOLUME:
g_value_set_double (value, (double) sink->volume / 100.);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_directsound_sink_getcaps (GstBaseSink * bsink)
{
GstDirectSoundSink *dsoundsink;
dsoundsink = GST_DIRECTSOUND_SINK (bsink);
return
gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
(dsoundsink)));
}
static gboolean
gst_directsound_sink_open (GstAudioSink * asink)
{
GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (asink);
HRESULT hRes;
/* create and initialize a DirecSound object */
if (FAILED (hRes = DirectSoundCreate (NULL, &dsoundsink->pDS, NULL))) {
GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
("gst_directsound_sink_open: DirectSoundCreate: %s",
DXGetErrorString9 (hRes)), (NULL));
return FALSE;
}
if (FAILED (hRes = IDirectSound_SetCooperativeLevel (dsoundsink->pDS,
GetDesktopWindow (), DSSCL_PRIORITY))) {
GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
("gst_directsound_sink_open: IDirectSound_SetCooperativeLevel: %s",
DXGetErrorString9 (hRes)), (NULL));
return FALSE;
}
return TRUE;
}
static gboolean
gst_directsound_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
{
GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (asink);
HRESULT hRes;
DSBUFFERDESC descSecondary;
WAVEFORMATEX wfx;
/*save number of bytes per sample */
dsoundsink->bytes_per_sample = spec->bytes_per_sample;
/* fill the WAVEFORMATEX struture with spec params */
memset (&wfx, 0, sizeof (wfx));
wfx.cbSize = sizeof (wfx);
wfx.wFormatTag = WAVE_FORMAT_PCM;
wfx.nChannels = spec->channels;
wfx.nSamplesPerSec = spec->rate;
wfx.wBitsPerSample = (spec->bytes_per_sample * 8) / wfx.nChannels;
wfx.nBlockAlign = spec->bytes_per_sample;
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
/* Create directsound buffer with size based on our configured
* buffer_size (which is 200 ms by default) */
dsoundsink->buffer_size =
gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->buffer_time,
GST_MSECOND);
spec->segsize =
gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->latency_time,
GST_MSECOND);
spec->segtotal = dsoundsink->buffer_size / spec->segsize;
// Make the final buffer size be an integer number of segments
dsoundsink->buffer_size = spec->segsize * spec->segtotal;
GST_INFO_OBJECT (dsoundsink,
"GstRingBufferSpec->channels: %d, GstRingBufferSpec->rate: %d, GstRingBufferSpec->bytes_per_sample: %d\n"
"WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d, WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld\n"
"Size of dsound cirucular buffe=>%d\n", spec->channels, spec->rate,
spec->bytes_per_sample, wfx.nSamplesPerSec, wfx.wBitsPerSample,
wfx.nBlockAlign, wfx.nAvgBytesPerSec, dsoundsink->buffer_size);
/* create a secondary directsound buffer */
memset (&descSecondary, 0, sizeof (DSBUFFERDESC));
descSecondary.dwSize = sizeof (DSBUFFERDESC);
descSecondary.dwFlags = DSBCAPS_GETCURRENTPOSITION2 |
DSBCAPS_GLOBALFOCUS | DSBCAPS_CTRLVOLUME;
descSecondary.dwBufferBytes = dsoundsink->buffer_size;
descSecondary.lpwfxFormat = (WAVEFORMATEX *) & wfx;
hRes = IDirectSound_CreateSoundBuffer (dsoundsink->pDS, &descSecondary,
&dsoundsink->pDSBSecondary, NULL);
if (FAILED (hRes)) {
GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
("gst_directsound_sink_prepare: IDirectSound_CreateSoundBuffer: %s",
DXGetErrorString9 (hRes)), (NULL));
return FALSE;
}
gst_directsound_sink_set_volume (dsoundsink);
return TRUE;
}
static gboolean
gst_directsound_sink_unprepare (GstAudioSink * asink)
{
GstDirectSoundSink *dsoundsink;
dsoundsink = GST_DIRECTSOUND_SINK (asink);
/* release secondary DirectSound buffer */
if (dsoundsink->pDSBSecondary)
IDirectSoundBuffer_Release (dsoundsink->pDSBSecondary);
return TRUE;
}
static gboolean
gst_directsound_sink_close (GstAudioSink * asink)
{
GstDirectSoundSink *dsoundsink = NULL;
dsoundsink = GST_DIRECTSOUND_SINK (asink);
/* release DirectSound object */
g_return_val_if_fail (dsoundsink->pDS != NULL, FALSE);
IDirectSound_Release (dsoundsink->pDS);
return TRUE;
}
static guint
gst_directsound_sink_write (GstAudioSink * asink, gpointer data, guint length)
{
GstDirectSoundSink *dsoundsink;
DWORD dwStatus;
HRESULT hRes;
LPVOID pLockedBuffer1 = NULL, pLockedBuffer2 = NULL;
DWORD dwSizeBuffer1, dwSizeBuffer2;
DWORD dwCurrentPlayCursor;
dsoundsink = GST_DIRECTSOUND_SINK (asink);
GST_DSOUND_LOCK (dsoundsink);
/* get current buffer status */
hRes = IDirectSoundBuffer_GetStatus (dsoundsink->pDSBSecondary, &dwStatus);
/* get current play cursor position */
hRes = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary,
&dwCurrentPlayCursor, NULL);
if (SUCCEEDED (hRes) && (dwStatus & DSBSTATUS_PLAYING)) {
DWORD dwFreeBufferSize;
calculate_freesize:
/* calculate the free size of the circular buffer */
if (dwCurrentPlayCursor < dsoundsink->current_circular_offset)
dwFreeBufferSize =
dsoundsink->buffer_size - (dsoundsink->current_circular_offset -
dwCurrentPlayCursor);
else
dwFreeBufferSize =
dwCurrentPlayCursor - dsoundsink->current_circular_offset;
if (length >= dwFreeBufferSize) {
Sleep (100);
hRes = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary,
&dwCurrentPlayCursor, NULL);
hRes =
IDirectSoundBuffer_GetStatus (dsoundsink->pDSBSecondary, &dwStatus);
if (SUCCEEDED (hRes) && (dwStatus & DSBSTATUS_PLAYING))
goto calculate_freesize;
else {
dsoundsink->first_buffer_after_reset = FALSE;
GST_DSOUND_UNLOCK (dsoundsink);
return 0;
}
}
}
if (dwStatus & DSBSTATUS_BUFFERLOST) {
hRes = IDirectSoundBuffer_Restore (dsoundsink->pDSBSecondary); /*need a loop waiting the buffer is restored?? */
dsoundsink->current_circular_offset = 0;
}
hRes = IDirectSoundBuffer_Lock (dsoundsink->pDSBSecondary,
dsoundsink->current_circular_offset, length, &pLockedBuffer1,
&dwSizeBuffer1, &pLockedBuffer2, &dwSizeBuffer2, 0L);
if (SUCCEEDED (hRes)) {
// Write to pointers without reordering.
memcpy (pLockedBuffer1, data, dwSizeBuffer1);
if (pLockedBuffer2 != NULL)
memcpy (pLockedBuffer2, (LPBYTE) data + dwSizeBuffer1, dwSizeBuffer2);
// Update where the buffer will lock (for next time)
dsoundsink->current_circular_offset += dwSizeBuffer1 + dwSizeBuffer2;
dsoundsink->current_circular_offset %= dsoundsink->buffer_size; /* Circular buffer */
hRes = IDirectSoundBuffer_Unlock (dsoundsink->pDSBSecondary, pLockedBuffer1,
dwSizeBuffer1, pLockedBuffer2, dwSizeBuffer2);
}
/* if the buffer was not in playing state yet, call play on the buffer
except if this buffer is the fist after a reset (base class call reset and write a buffer when setting the sink to pause) */
if (!(dwStatus & DSBSTATUS_PLAYING) &&
dsoundsink->first_buffer_after_reset == FALSE) {
hRes = IDirectSoundBuffer_Play (dsoundsink->pDSBSecondary, 0, 0,
DSBPLAY_LOOPING);
}
dsoundsink->first_buffer_after_reset = FALSE;
GST_DSOUND_UNLOCK (dsoundsink);
return length;
}
static guint
gst_directsound_sink_delay (GstAudioSink * asink)
{
GstDirectSoundSink *dsoundsink;
HRESULT hRes;
DWORD dwCurrentPlayCursor;
DWORD dwBytesInQueue = 0;
gint nNbSamplesInQueue = 0;
DWORD dwStatus;
dsoundsink = GST_DIRECTSOUND_SINK (asink);
/* get current buffer status */
hRes = IDirectSoundBuffer_GetStatus (dsoundsink->pDSBSecondary, &dwStatus);
if (dwStatus & DSBSTATUS_PLAYING) {
/*evaluate the number of samples in queue in the circular buffer */
hRes = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary,
&dwCurrentPlayCursor, NULL);
if (hRes == S_OK) {
if (dwCurrentPlayCursor < dsoundsink->current_circular_offset)
dwBytesInQueue =
dsoundsink->current_circular_offset - dwCurrentPlayCursor;
else
dwBytesInQueue =
dsoundsink->current_circular_offset + (dsoundsink->buffer_size -
dwCurrentPlayCursor);
nNbSamplesInQueue = dwBytesInQueue / dsoundsink->bytes_per_sample;
}
}
return nNbSamplesInQueue;
}
static void
gst_directsound_sink_reset (GstAudioSink * asink)
{
GstDirectSoundSink *dsoundsink;
LPVOID pLockedBuffer = NULL;
DWORD dwSizeBuffer = 0;
dsoundsink = GST_DIRECTSOUND_SINK (asink);
GST_DSOUND_LOCK (dsoundsink);
if (dsoundsink->pDSBSecondary) {
/*stop playing */
HRESULT hRes = IDirectSoundBuffer_Stop (dsoundsink->pDSBSecondary);
/*reset position */
hRes = IDirectSoundBuffer_SetCurrentPosition (dsoundsink->pDSBSecondary, 0);
dsoundsink->current_circular_offset = 0;
/*reset the buffer */
hRes = IDirectSoundBuffer_Lock (dsoundsink->pDSBSecondary,
dsoundsink->current_circular_offset, dsoundsink->buffer_size,
&pLockedBuffer, &dwSizeBuffer, NULL, NULL, 0L);
if (SUCCEEDED (hRes)) {
memset (pLockedBuffer, 0, dwSizeBuffer);
hRes =
IDirectSoundBuffer_Unlock (dsoundsink->pDSBSecondary, pLockedBuffer,
dwSizeBuffer, NULL, 0);
}
}
dsoundsink->first_buffer_after_reset = TRUE;
GST_DSOUND_UNLOCK (dsoundsink);
}