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-rw-r--r--sys/directsound/gstdirectsoundsink.c785
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diff --git a/sys/directsound/gstdirectsoundsink.c b/sys/directsound/gstdirectsoundsink.c
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+++ b/sys/directsound/gstdirectsoundsink.c
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+/* GStreamer
+* Copyright (C) 2005 Sebastien Moutte <sebastien@moutte.net>
+* Copyright (C) 2007 Pioneers of the Inevitable <songbird@songbirdnest.com>
+* Copyright (C) 2010 Fluendo S.A. <support@fluendo.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>
+
+#ifdef __CYGWIN__
+#include <unistd.h>
+#ifndef _swab
+#define _swab swab
+#endif
+#endif
+
+GST_DEBUG_CATEGORY_STATIC (directsoundsink_debug);
+#define GST_CAT_DEFAULT directsoundsink_debug
+
+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);
+static GstCaps *gst_directsound_probe_supported_formats (GstDirectSoundSink *
+ dsoundsink, const GstCaps * template_caps);
+
+/* 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, "
+ "width = (int) 16, "
+ "depth = (int) 16, "
+ "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
+ "audio/x-raw-int, "
+ "signed = (boolean) FALSE, "
+ "width = (int) 8, "
+ "depth = (int) 8, "
+ "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ];"
+ "audio/x-iec958"));
+
+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_simple (element_class,
+ "Direct Sound Audio Sink", "Sink/Audio",
+ "Output to a sound card via Direct Sound",
+ "Sebastien Moutte <sebastien@moutte.net>");
+ gst_element_class_add_static_pad_template (element_class,
+ &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_directsound_sink_finalise;
+ gobject_class->set_property = gst_directsound_sink_set_property;
+ gobject_class->get_property = 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->cached_caps = 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)
+{
+ GstElementClass *element_class;
+ GstPadTemplate *pad_template;
+ GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (bsink);
+ GstCaps *caps;
+ gchar *caps_string = NULL;
+
+ 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_string = gst_caps_to_string (dsoundsink->cached_caps);
+ GST_DEBUG_OBJECT (dsoundsink, "Returning cached caps: %s", caps_string);
+ g_free (caps_string);
+ return gst_caps_ref (dsoundsink->cached_caps);
+ }
+
+ 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) {
+ gchar *caps_string = gst_caps_to_string (caps);
+ GST_DEBUG_OBJECT (dsoundsink, "returning caps %s", caps_string);
+ g_free (caps_string);
+ }
+
+ return caps;
+}
+
+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 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 {
+#ifdef WAVE_FORMAT_DOLBY_AC3_SPDIF
+ 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;
+#else
+ g_assert_not_reached ();
+#endif
+ }
+
+ // 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_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 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 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 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);
+}
+
+/*
+ * gst_directsound_probe_supported_formats:
+ *
+ * Takes the template caps and returns the subset which is actually
+ * supported by this device.
+ *
+ */
+
+static GstCaps *
+gst_directsound_probe_supported_formats (GstDirectSoundSink * dsoundsink,
+ const GstCaps * template_caps)
+{
+ HRESULT hRes;
+ DSBUFFERDESC descSecondary;
+ WAVEFORMATEX wfx;
+ GstCaps *caps;
+
+ caps = gst_caps_copy (template_caps);
+
+ /*
+ * Check availability of digital output by trying to create an SPDIF buffer
+ */
+
+#ifdef WAVE_FORMAT_DOLBY_AC3_SPDIF
+ /* fill the WAVEFORMATEX structure with some standard AC3 over SPDIF params */
+ memset (&wfx, 0, sizeof (wfx));
+ wfx.cbSize = 0;
+ wfx.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
+ wfx.nChannels = 2;
+ wfx.nSamplesPerSec = 48000;
+ wfx.wBitsPerSample = 16;
+ wfx.nBlockAlign = 4;
+ wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
+
+ // create a secondary directsound buffer
+ memset (&descSecondary, 0, sizeof (DSBUFFERDESC));
+ descSecondary.dwSize = sizeof (DSBUFFERDESC);
+ descSecondary.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS;
+ descSecondary.dwBufferBytes = 6144;
+ descSecondary.lpwfxFormat = &wfx;
+
+ hRes = IDirectSound_CreateSoundBuffer (dsoundsink->pDS, &descSecondary,
+ &dsoundsink->pDSBSecondary, NULL);
+ if (FAILED (hRes)) {
+ GST_INFO_OBJECT (dsoundsink, "AC3 passthrough not supported "
+ "(IDirectSound_CreateSoundBuffer returned: %s)\n",
+ DXGetErrorString9 (hRes));
+ caps =
+ gst_caps_subtract (caps, gst_caps_new_simple ("audio/x-iec958", NULL));
+ } else {
+ GST_INFO_OBJECT (dsoundsink, "AC3 passthrough supported");
+ hRes = IDirectSoundBuffer_Release (dsoundsink->pDSBSecondary);
+ if (FAILED (hRes)) {
+ GST_DEBUG_OBJECT (dsoundsink,
+ "(IDirectSoundBuffer_Release returned: %s)\n",
+ DXGetErrorString9 (hRes));
+ }
+ }
+#else
+ caps = gst_caps_subtract (caps, gst_caps_new_simple ("audio/x-iec958", NULL));
+#endif
+
+ return caps;
+}
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-21 16:17:12 +0000