gstreamer/sys/directsound/gstdirectsoundsink.c
Sébastien Moutte 454c252335 sys/directsound/gstdirectsoundsink.*: Add an attenuation property that will directly attenuate the directsound buffer.
Original commit message from CVS:
* sys/directsound/gstdirectsoundsink.h:
* sys/directsound/gstdirectsoundsink.c:
Add an attenuation property that will directly attenuate the
directsound buffer.
Change the size of the directsound secondary buffer to a half second.
Add more debug logs.
Add a lock to protect dsound buffer write access.
Fix a bad implementation of reset.
* sys/directsound/gstdirectdrawsink.c:
* sys/directsound/gstdirectdrawsink.h:
Add a keep_aspect_ratio property.
Do not use overlay if not supported.
Add more debug logs.
Remove overwrite of WM_ERASEBKGND message handling. It was not
redrawing border when keep_aspect_ratio was enabled.
* win32/common/config.h:
update version waiting an auto-generated config.h
2006-07-24 21:43:06 +00:00

518 lines
16 KiB
C

/* GStreamer
* Copyright (C) 2005 Sebastien Moutte <sebastien@moutte.net>
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstdirectsoundsink.h"
#include <fcntl.h>
#include <errno.h>
#include <unistd.h>
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (directsoundsink_debug);
/* elementfactory information */
static const GstElementDetails gst_directsoundsink_details =
GST_ELEMENT_DETAILS ("Audio Sink (DIRECTSOUND)",
"Sink/Audio",
"Output to a sound card via DIRECTSOUND",
"Sebastien Moutte <sebastien@moutte.net>");
static void gst_directsoundsink_base_init (gpointer g_class);
static void gst_directsoundsink_class_init (GstDirectSoundSinkClass * klass);
static void gst_directsoundsink_init (GstDirectSoundSink * dsoundsink,
GstDirectSoundSinkClass * g_class);
static void gst_directsoundsink_dispose (GObject * object);
static void gst_directsoundsink_finalise (GObject * object);
static void gst_directsoundsink_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_directsoundsink_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstCaps *gst_directsoundsink_getcaps (GstBaseSink * bsink);
static gboolean gst_directsoundsink_prepare (GstAudioSink * asink,
GstRingBufferSpec * spec);
static gboolean gst_directsoundsink_unprepare (GstAudioSink * asink);
static gboolean gst_directsoundsink_open (GstAudioSink * asink);
static gboolean gst_directsoundsink_close (GstAudioSink * asink);
static guint gst_directsoundsink_write (GstAudioSink * asink, gpointer data,
guint length);
static guint gst_directsoundsink_delay (GstAudioSink * asink);
static void gst_directsoundsink_reset (GstAudioSink * asink);
static GstStaticPadTemplate directsoundsink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
"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_ATTENUATION
};
static void
_do_init (GType directsoundsink_type)
{
GST_DEBUG_CATEGORY_INIT (directsoundsink_debug, "directsoundsink", 0,
"DirectSound sink");
}
GST_BOILERPLATE_FULL (GstDirectSoundSink, gst_directsoundsink, GstAudioSink,
GST_TYPE_AUDIO_SINK, _do_init);
static void
gst_directsoundsink_dispose (GObject * object)
{
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_directsoundsink_finalise (GObject * object)
{
GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (object);
g_mutex_free (dsoundsink->dsound_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_directsoundsink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &gst_directsoundsink_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&directsoundsink_sink_factory));
}
static void
gst_directsoundsink_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;
parent_class = g_type_class_peek_parent (klass);
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_directsoundsink_dispose);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_directsoundsink_finalise);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_directsoundsink_get_property);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_directsoundsink_set_property);
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_directsoundsink_getcaps);
gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_directsoundsink_prepare);
gstaudiosink_class->unprepare =
GST_DEBUG_FUNCPTR (gst_directsoundsink_unprepare);
gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_directsoundsink_open);
gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_directsoundsink_close);
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_directsoundsink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_directsoundsink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_directsoundsink_reset);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_ATTENUATION,
g_param_spec_long ("attenuation", "Attenuation of the sound",
"The attenuation for the directsound buffer (default is 0 so the directsound buffer will not be attenuated)",
-10000, 0, 0, G_PARAM_READWRITE));
}
static void
gst_directsoundsink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstDirectSoundSink *dsoundsink;
dsoundsink = GST_DIRECTSOUND_SINK (object);
switch (prop_id) {
case PROP_ATTENUATION:
{
glong attenuation = g_value_get_long (value);
if (attenuation != dsoundsink->attenuation) {
dsoundsink->attenuation = attenuation;
if (dsoundsink->pDSBSecondary)
IDirectSoundBuffer_SetVolume (dsoundsink->pDSBSecondary,
dsoundsink->attenuation);
}
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_directsoundsink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstDirectSoundSink *dsoundsink;
dsoundsink = GST_DIRECTSOUND_SINK (object);
switch (prop_id) {
case PROP_ATTENUATION:
g_value_set_long (value, dsoundsink->attenuation);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_directsoundsink_init (GstDirectSoundSink * dsoundsink,
GstDirectSoundSinkClass * g_class)
{
dsoundsink->pDS = NULL;
dsoundsink->pDSBSecondary = NULL;
dsoundsink->current_circular_offset = 0;
dsoundsink->buffer_size = DSBSIZE_MIN;
dsoundsink->attenuation = 0;
dsoundsink->dsound_lock = g_mutex_new ();
dsoundsink->first_buffer_after_reset = FALSE;
}
static GstCaps *
gst_directsoundsink_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_directsoundsink_open (GstAudioSink * asink)
{
GstDirectSoundSink *dsoundsink;
HRESULT hRes;
dsoundsink = GST_DIRECTSOUND_SINK (asink);
/* create and initialize a DirecSound object */
if (FAILED (hRes = DirectSoundCreate (NULL, &dsoundsink->pDS, NULL))) {
GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
("gst_directsoundsink_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_directsoundsink_open: IDirectSound_SetCooperativeLevel: %s",
DXGetErrorString9 (hRes)), (NULL));
return FALSE;
}
return TRUE;
}
static gboolean
gst_directsoundsink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
{
GstDirectSoundSink *dsoundsink;
HRESULT hRes;
DSBUFFERDESC descSecondary;
WAVEFORMATEX wfx;
dsoundsink = GST_DIRECTSOUND_SINK (asink);
/*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;
/* directsound buffer size can handle 1/2 sec of the stream */
dsoundsink->buffer_size = wfx.nAvgBytesPerSec / 2;
GST_CAT_INFO (directsoundsink_debug,
"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_directsoundsink_prepare: IDirectSound_CreateSoundBuffer: %s",
DXGetErrorString9 (hRes)), (NULL));
return FALSE;
}
if (dsoundsink->attenuation != 0)
IDirectSoundBuffer_SetVolume (dsoundsink->pDSBSecondary,
dsoundsink->attenuation);
return TRUE;
}
static gboolean
gst_directsoundsink_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_directsoundsink_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_directsoundsink_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_directsoundsink_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_directsoundsink_reset (GstAudioSink * asink)
{
/*not tested for seeking */
GstDirectSoundSink *dsoundsink;
LPBYTE 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);
}