static gboolean
gst_directsound_sink_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_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 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 const GList *
gst_directsound_sink_mixer_list_tracks (GstMixer * mixer)
{
GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (mixer);
return dsoundsink->tracks;
}
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_finalize (GObject * object)
{
GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (object);
g_mutex_clear (&dsoundsink->dsound_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
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 gboolean
gst_directsound_sink_acceptcaps (GstPad * pad, GstCaps * caps)
{
GstDirectSoundSink *dsink =
GST_DIRECTSOUND_SINK (gst_pad_get_parent_element (pad));
GstCaps *pad_caps;
GstStructure *st;
gboolean ret = FALSE;
GstRingBufferSpec spec = { 0 };
pad_caps = gst_pad_get_caps_reffed (pad);
if (pad_caps) {
gboolean cret = gst_caps_can_intersect (pad_caps, caps);
gst_caps_unref (pad_caps);
if (!cret)
goto done;
}
/* If we've not got fixed caps, creating a stream might fail, so let's just
* return from here with default acceptcaps behaviour */
if (!gst_caps_is_fixed (caps))
goto done;
/* parse helper expects this set, so avoid nasty warning
* will be set properly later on anyway */
spec.latency_time = GST_SECOND;
if (!gst_ring_buffer_parse_caps (&spec, caps))
goto done;
/* Make sure input is framed (one frame per buffer) and can be payloaded */
switch (spec.type) {
case GST_BUFTYPE_AC3:
case GST_BUFTYPE_DTS:
{
gboolean framed = FALSE, parsed = FALSE;
st = gst_caps_get_structure (caps, 0);
gst_structure_get_boolean (st, "framed", &framed);
gst_structure_get_boolean (st, "parsed", &parsed);
if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0)
goto done;
}
default:
break;
}
ret = TRUE;
done:
gst_object_unref (dsink);
return ret;
}
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);
dsoundsink->pDSBSecondary = NULL;
}
return TRUE;
}
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 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 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);
dsoundsink->pDS = NULL;
gst_caps_replace (&dsoundsink->cached_caps, NULL);
return TRUE;
}
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:
gst_directsound_sink_set_volume (sink, g_value_get_double (value), TRUE);
break;
case PROP_MUTE:
gst_directsound_sink_set_mute (sink, g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_directsound_sink_getcaps (GstBaseSink * bsink, GstCaps * filter)
{
GstElementClass *element_class;
GstPadTemplate *pad_template;
GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (bsink);
GstCaps *caps;
if (dsoundsink->pDS == NULL) {
GST_DEBUG_OBJECT (dsoundsink, "device not open, using template caps");
return NULL; /* base class will get template caps for us */
}
if (dsoundsink->cached_caps) {
caps = gst_caps_ref (dsoundsink->cached_caps);
} else {
element_class = GST_ELEMENT_GET_CLASS (dsoundsink);
pad_template = gst_element_class_get_pad_template (element_class, "sink");
g_return_val_if_fail (pad_template != NULL, NULL);
caps = gst_directsound_probe_supported_formats (dsoundsink,
gst_pad_template_get_caps (pad_template));
if (caps)
dsoundsink->cached_caps = gst_caps_ref (caps);
}
if (caps && filter) {
GstCaps *tmp =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = tmp;
}
if (caps) {
gchar *caps_string = gst_caps_to_string (caps);
GST_DEBUG_OBJECT (dsoundsink, "returning caps %s", caps_string);
g_free (caps_string);
}
return caps;
}
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);
}
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);
/* Fix endianness */
if (dsoundsink->buffer_format == GST_IEC958)
_swab (data, data, length);
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 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 and buffer format */
dsoundsink->bytes_per_sample = spec->bytes_per_sample;
dsoundsink->buffer_format = spec->format;
/* fill the WAVEFORMATEX structure with spec params */
memset (&wfx, 0, sizeof (wfx));
if (spec->format != GST_IEC958) {
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);
/* Make sure we make those numbers multiple of our sample size in bytes */
dsoundsink->buffer_size += dsoundsink->buffer_size % spec->bytes_per_sample;
spec->segsize =
gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->latency_time,
GST_MSECOND);
spec->segsize += spec->segsize % spec->bytes_per_sample;
spec->segtotal = dsoundsink->buffer_size / spec->segsize;
} else {
wfx.cbSize = 0;
wfx.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
wfx.nChannels = 2;
wfx.nSamplesPerSec = spec->rate;
wfx.wBitsPerSample = 16;
wfx.nBlockAlign = wfx.wBitsPerSample / 8 * wfx.nChannels;
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
spec->segsize = 6144;
spec->segtotal = 10;
}
// 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 circular buffer=>%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;
if (spec->format != GST_IEC958)
descSecondary.dwFlags |= 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_acceptcaps (GstBaseSink * sink, GstQuery * query)
{
GstDirectSoundSink *dsink = GST_DIRECTSOUND_SINK (sink);
GstPad *pad;
GstCaps *caps;
GstCaps *pad_caps;
GstStructure *st;
gboolean ret = FALSE;
GstAudioRingBufferSpec spec = { 0 };
if (G_UNLIKELY (dsink == NULL))
return FALSE;
pad = sink->sinkpad;
gst_query_parse_accept_caps (query, &caps);
GST_DEBUG_OBJECT (pad, "caps %" GST_PTR_FORMAT, caps);
pad_caps = gst_pad_query_caps (pad, NULL);
if (pad_caps) {
gboolean cret = gst_caps_is_subset (caps, pad_caps);
gst_caps_unref (pad_caps);
if (!cret) {
GST_DEBUG_OBJECT (dsink,
"Caps are not a subset of the pad caps, not accepting caps");
goto done;
}
}
/* If we've not got fixed caps, creating a stream might fail, so let's just
* return from here with default acceptcaps behaviour */
if (!gst_caps_is_fixed (caps)) {
GST_DEBUG_OBJECT (dsink, "Caps are not fixed, not accepting caps");
goto done;
}
spec.latency_time = GST_SECOND;
if (!gst_audio_ring_buffer_parse_caps (&spec, caps)) {
GST_DEBUG_OBJECT (dsink, "Failed to parse caps, not accepting");
goto done;
}
/* Make sure input is framed (one frame per buffer) and can be payloaded */
switch (spec.type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
{
gboolean framed = FALSE, parsed = FALSE;
st = gst_caps_get_structure (caps, 0);
gst_structure_get_boolean (st, "framed", &framed);
gst_structure_get_boolean (st, "parsed", &parsed);
if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0) {
GST_DEBUG_OBJECT (dsink, "Wrong AC3/DTS caps, not accepting");
goto done;
}
}
default:
break;
}
ret = TRUE;
GST_DEBUG_OBJECT (dsink, "Accepting caps");
done:
gst_query_set_accept_caps_result (query, ret);
return TRUE;
}