text: /* 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., 51 Franklin St, Fifth Floor,
  * Boston, MA 02110-1301, 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
   * @title: 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).
   *
   * ## Example pipelines
   * |[
   * gst-launch-1.0 -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-1.0 -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! directsoundsink
   * ]| will play an Ogg/Vorbis audio file and output it.
   *
   */
  
  #ifdef HAVE_CONFIG_H
  #include "config.h"
  #endif
  
  #include <gst/base/gstbasesink.h>
  #include "gstdirectsoundsink.h"
  #include <gst/audio/gstaudioiec61937.h>
  
  #include <math.h>
  
  #ifdef __CYGWIN__
  #include <unistd.h>
  #ifndef _swab
  #define _swab swab
  #endif
  #endif
  
  #define DEFAULT_MUTE FALSE
  
  GST_DEBUG_CATEGORY_STATIC (directsoundsink_debug);
  #define GST_CAT_DEFAULT directsoundsink_debug
  
  static void gst_directsound_sink_finalize (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,
      GstCaps * filter);
  static GstBuffer *gst_directsound_sink_payload (GstAudioBaseSink * sink,
      GstBuffer * buf);
  static gboolean gst_directsound_sink_prepare (GstAudioSink * asink,
      GstAudioRingBufferSpec * 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 gint 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);
  static gboolean gst_directsound_sink_query (GstBaseSink * pad,
      GstQuery * query);
  
  static void gst_directsound_sink_set_volume (GstDirectSoundSink * sink,
      gdouble volume, gboolean store);
  static gdouble gst_directsound_sink_get_volume (GstDirectSoundSink * sink);
  static void gst_directsound_sink_set_mute (GstDirectSoundSink * sink,
      gboolean mute);
  static gboolean gst_directsound_sink_get_mute (GstDirectSoundSink * sink);
  static const gchar *gst_directsound_sink_get_device (GstDirectSoundSink *
      dsoundsink);
  static void gst_directsound_sink_set_device (GstDirectSoundSink * dsoundsink,
      const gchar * device_id);
  
  static gboolean gst_directsound_sink_is_spdif_format (GstAudioRingBufferSpec *
      spec);
  
  static gchar *gst_hres_to_string (HRESULT hRes);
  
  static GstStaticPadTemplate directsoundsink_sink_factory =
  GST_STATIC_PAD_TEMPLATE ("sink",
      GST_PAD_SINK,
      GST_PAD_ALWAYS,
      GST_STATIC_CAPS (GST_DIRECTSOUND_SINK_CAPS));
  
  enum
  {
    PROP_0,
    PROP_VOLUME,
    PROP_MUTE,
    PROP_DEVICE
  };
  
  #define gst_directsound_sink_parent_class parent_class
  G_DEFINE_TYPE_WITH_CODE (GstDirectSoundSink, gst_directsound_sink,
      GST_TYPE_AUDIO_SINK, G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL)
      );
  
  static void
  gst_directsound_sink_finalize (GObject * object)
  {
    GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (object);
  
    g_free (dsoundsink->device_id);
    dsoundsink->device_id = NULL;
  
    g_mutex_clear (&dsoundsink->dsound_lock);
    gst_object_unref (dsoundsink->system_clock);
    if (dsoundsink->write_wait_clock_id != NULL) {
      gst_clock_id_unref (dsoundsink->write_wait_clock_id);
    }
  
    G_OBJECT_CLASS (parent_class)->finalize (object);
  }
  
  static void
  gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass)
  {
    GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
    GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
    GstAudioSinkClass *gstaudiosink_class = GST_AUDIO_SINK_CLASS (klass);
    GstAudioBaseSinkClass *gstaudiobasesink_class =
        GST_AUDIO_BASE_SINK_CLASS (klass);
    GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
  
    GST_DEBUG_CATEGORY_INIT (directsoundsink_debug, "directsoundsink", 0,
        "DirectSound sink");
  
    gobject_class->finalize = gst_directsound_sink_finalize;
    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);
  
    gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_directsound_sink_query);
  
    gstaudiobasesink_class->payload =
        GST_DEBUG_FUNCPTR (gst_directsound_sink_payload);
  
    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));
  
    g_object_class_install_property (gobject_class,
        PROP_MUTE,
        g_param_spec_boolean ("mute", "Mute",
            "Mute state of this stream", DEFAULT_MUTE,
            G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
  
    g_object_class_install_property (gobject_class,
        PROP_DEVICE,
        g_param_spec_string ("device", "Device",
            "DirectSound playback device as a GUID string",
            NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
  
    gst_element_class_set_static_metadata (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_init (GstDirectSoundSink * dsoundsink)
  {
    dsoundsink->volume = 100;
    dsoundsink->mute = FALSE;
    dsoundsink->device_id = NULL;
    dsoundsink->pDS = NULL;
    dsoundsink->cached_caps = NULL;
    dsoundsink->pDSBSecondary = NULL;
    dsoundsink->current_circular_offset = 0;
    dsoundsink->buffer_size = DSBSIZE_MIN;
    g_mutex_init (&dsoundsink->dsound_lock);
    dsoundsink->system_clock = gst_system_clock_obtain ();
    dsoundsink->write_wait_clock_id = NULL;
    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:
        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;
      case PROP_DEVICE:
        gst_directsound_sink_set_device (sink, g_value_get_string (value));
        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, gst_directsound_sink_get_volume (sink));
        break;
      case PROP_MUTE:
        g_value_set_boolean (value, gst_directsound_sink_get_mute (sink));
        break;
      case PROP_DEVICE:
        g_value_set_string (value, gst_directsound_sink_get_device (sink));
        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 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;
  }
  
  static gboolean
  gst_directsound_sink_query (GstBaseSink * sink, GstQuery * query)
  {
    gboolean res = TRUE;
  
    switch (GST_QUERY_TYPE (query)) {
      case GST_QUERY_ACCEPT_CAPS:
        res = gst_directsound_sink_acceptcaps (sink, query);
        break;
      default:
        res = GST_BASE_SINK_CLASS (parent_class)->query (sink, query);
    }
  
    return res;
  }
  
  static LPGUID
  string_to_guid (const gchar * str)
  {
    HRESULT ret;
    gunichar2 *wstr;
    LPGUID out;
  
    wstr = g_utf8_to_utf16 (str, -1, NULL, NULL, NULL);
    if (!wstr)
      return NULL;
  
    out = g_new (GUID, 1);
    ret = CLSIDFromString ((LPOLESTR) wstr, out);
    g_free (wstr);
    if (ret != NOERROR) {
      g_free (out);
      return NULL;
    }
  
    return out;
  }
  
  static gboolean
  gst_directsound_sink_open (GstAudioSink * asink)
  {
    GstDirectSoundSink *dsoundsink;
    HRESULT hRes;
    LPGUID lpGuid = NULL;
  
    dsoundsink = GST_DIRECTSOUND_SINK (asink);
  
    if (dsoundsink->device_id) {
      lpGuid = string_to_guid (dsoundsink->device_id);
      if (lpGuid == NULL) {
        GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
            ("device set but guid not found: %s", dsoundsink->device_id), (NULL));
        return FALSE;
      }
    }
  
    /* create and initialize a DirectSound object */
    if (FAILED (hRes = DirectSoundCreate (lpGuid, &dsoundsink->pDS, NULL))) {
      gchar *error_text = gst_hres_to_string (hRes);
      GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
          ("DirectSoundCreate: %s", error_text), (NULL));
      g_free (lpGuid);
      g_free (error_text);
      return FALSE;
    }
  
    g_free (lpGuid);
  
    if (FAILED (hRes = IDirectSound_SetCooperativeLevel (dsoundsink->pDS,
                GetDesktopWindow (), DSSCL_PRIORITY))) {
      gchar *error_text = gst_hres_to_string (hRes);
      GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
          ("IDirectSound_SetCooperativeLevel: %s", error_text), (NULL));
      g_free (error_text);
      return FALSE;
    }
  
    return TRUE;
  }
  
  static gboolean
  gst_directsound_sink_is_spdif_format (GstAudioRingBufferSpec * spec)
  {
    return spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3 ||
        spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS;
  }
  
  static gboolean
  gst_directsound_sink_prepare (GstAudioSink * asink,
      GstAudioRingBufferSpec * spec)
  {
    GstDirectSoundSink *dsoundsink;
    HRESULT hRes;
    DSBUFFERDESC descSecondary;
    WAVEFORMATEX wfx;
  
    dsoundsink = GST_DIRECTSOUND_SINK (asink);
  
    /*save number of bytes per sample and buffer format */
    dsoundsink->bytes_per_sample = spec->info.bpf;
    dsoundsink->type = spec->type;
  
    /* fill the WAVEFORMATEX structure with spec params */
    memset (&wfx, 0, sizeof (wfx));
    if (!gst_directsound_sink_is_spdif_format (spec)) {
      wfx.cbSize = sizeof (wfx);
      wfx.wFormatTag = WAVE_FORMAT_PCM;
      wfx.nChannels = spec->info.channels;
      wfx.nSamplesPerSec = spec->info.rate;
      wfx.wBitsPerSample = (spec->info.bpf * 8) / wfx.nChannels;
      wfx.nBlockAlign = spec->info.bpf;
      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->info.bpf;
  
      spec->segsize =
          gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->latency_time,
          GST_MSECOND);
      spec->segsize -= spec->segsize % spec->info.bpf;
      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 = 48000;
      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, "channels: %d, rate: %d, bytes_per_sample: %d"
        " WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d,"
        " WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld\n"
        "Size of dsound circular buffer=>%d\n",
        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, dsoundsink->buffer_size);
  
    /* create a secondary directsound buffer */
    memset (&descSecondary, 0, sizeof (DSBUFFERDESC));
    descSecondary.dwSize = sizeof (DSBUFFERDESC);
    descSecondary.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS;
    if (!gst_directsound_sink_is_spdif_format (spec))
      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)) {
      gchar *error_text = gst_hres_to_string (hRes);
      GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
          ("IDirectSound_CreateSoundBuffer: %s", error_text), (NULL));
      g_free (error_text);
      return FALSE;
    }
  
    gst_directsound_sink_set_volume (dsoundsink,
        gst_directsound_sink_get_volume (dsoundsink), FALSE);
    gst_directsound_sink_set_mute (dsoundsink, dsoundsink->mute);
  
    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 gint
  gst_directsound_sink_write (GstAudioSink * asink, gpointer data, guint length)
  {
    GstDirectSoundSink *dsoundsink;
    DWORD dwStatus = 0;
    HRESULT hRes, hRes2;
    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 */
    hRes2 = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary,
        &dwCurrentPlayCursor, NULL);
  
    if (SUCCEEDED (hRes) && SUCCEEDED (hRes2) && (dwStatus & DSBSTATUS_PLAYING)) {
      DWORD dwFreeBufferSize = 0;
      GstClockTime sleep_time_ms = 0, sleep_until;
      GstClockID clock_id;
  
    calculate_freesize:
      /* Calculate the free space in 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;
  
      /* Not enough free space, wait for some samples to be played out. We could
       * write out partial data, but that will result in a tight loop in the
       * audioringbuffer write thread, and lead to high CPU usage. */
      if (length > dwFreeBufferSize) {
        gint rate = GST_AUDIO_BASE_SINK (asink)->ringbuffer->spec.info.rate;
        /* Wait for a time proportional to the space needed. In reality, the
         * directsound sink's position does not update frequently enough, so we
         * will end up waiting for much longer. Note that Sleep() has millisecond
         * resolution at best. */
        sleep_time_ms = gst_util_uint64_scale_int ((length - dwFreeBufferSize),
            1000, dsoundsink->bytes_per_sample * rate);
        /* Make sure we don't run in a tight loop unnecessarily */
        sleep_time_ms = MAX (sleep_time_ms, 10);
        sleep_until = gst_clock_get_time (dsoundsink->system_clock) +
            sleep_time_ms * GST_MSECOND;
  
        GST_DEBUG_OBJECT (dsoundsink,
            "length: %u, FreeBufSiz: %ld, sleep_time_ms: %" G_GUINT64_FORMAT
            ", bps: %i, rate: %i", length, dwFreeBufferSize, sleep_time_ms,
            dsoundsink->bytes_per_sample, rate);
  
        if (G_UNLIKELY (dsoundsink->write_wait_clock_id == NULL ||
                gst_clock_single_shot_id_reinit (dsoundsink->system_clock,
                    dsoundsink->write_wait_clock_id, sleep_until) == FALSE)) {
  
          if (dsoundsink->write_wait_clock_id != NULL) {
            gst_clock_id_unref (dsoundsink->write_wait_clock_id);
          }
  
          dsoundsink->write_wait_clock_id =
              gst_clock_new_single_shot_id (dsoundsink->system_clock,
              sleep_until);
        }
  
        clock_id = dsoundsink->write_wait_clock_id;
        dsoundsink->reset_while_sleeping = FALSE;
  
        GST_DSOUND_UNLOCK (dsoundsink);
  
        /* don't bother with the return value as we'll detect reset separately,
           as reset could happen between when this returns and we obtain the lock
           again -- so we can't use UNSCHEDULED here */
        gst_clock_id_wait (clock_id, NULL);
  
        GST_DSOUND_LOCK (dsoundsink);
  
        /* if a reset occurs, exit now */
        if (dsoundsink->reset_while_sleeping == TRUE) {
          GST_DSOUND_UNLOCK (dsoundsink);
          return -1;
        }
  
        /* May we send out? */
        hRes = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary,
            &dwCurrentPlayCursor, NULL);
        hRes2 =
            IDirectSoundBuffer_GetStatus (dsoundsink->pDSBSecondary, &dwStatus);
        if (SUCCEEDED (hRes) && SUCCEEDED (hRes2)
            && (dwStatus & DSBSTATUS_PLAYING))
          goto calculate_freesize;
        else {
          gchar *err1, *err2;
  
          dsoundsink->first_buffer_after_reset = FALSE;
          GST_DSOUND_UNLOCK (dsoundsink);
  
          err1 = gst_hres_to_string (hRes);
          err2 = gst_hres_to_string (hRes2);
          GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_WRITE,
              ("IDirectSoundBuffer_GetStatus %s, "
                  "IDirectSoundBuffer_GetCurrentPosition: %s, dwStatus: %lu",
                  err2, err1, dwStatus), (NULL));
          g_free (err1);
          g_free (err2);
          return -1;
        }
      }
    }
  
    if (dwStatus & DSBSTATUS_BUFFERLOST) {
      hRes = IDirectSoundBuffer_Restore (dsoundsink->pDSBSecondary);      /*need a loop waiting the buffer is restored?? */
      dsoundsink->current_circular_offset = 0;
    }
  
    /* Lock a buffer of length @length for writing */
    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);
  
      hRes = IDirectSoundBuffer_Unlock (dsoundsink->pDSBSecondary, pLockedBuffer1,
          dwSizeBuffer1, pLockedBuffer2, 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 */
    }
  
    /* 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 (SUCCEEDED (hRes) && (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,
          0, 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->reset_while_sleeping = TRUE;
    dsoundsink->first_buffer_after_reset = TRUE;
    if (dsoundsink->write_wait_clock_id != NULL) {
      gst_clock_id_unschedule (dsoundsink->write_wait_clock_id);
    }
  
    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;
    GstCaps *tmp, *tmp2;
    LPDIRECTSOUNDBUFFER tmpBuffer;
  
    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,
        &tmpBuffer, NULL);
    if (FAILED (hRes)) {
      gchar *error_text = gst_hres_to_string (hRes);
      GST_INFO_OBJECT (dsoundsink, "AC3 passthrough not supported "
          "(IDirectSound_CreateSoundBuffer returned: %s)\n", error_text);
      g_free (error_text);
      tmp = gst_caps_new_empty_simple ("audio/x-ac3");
      tmp2 = gst_caps_subtract (caps, tmp);
      gst_caps_unref (tmp);
      gst_caps_unref (caps);
      caps = tmp2;
      tmp = gst_caps_new_empty_simple ("audio/x-dts");
      tmp2 = gst_caps_subtract (caps, tmp);
      gst_caps_unref (tmp);
      gst_caps_unref (caps);
      caps = tmp2;
    } else {
      GST_INFO_OBJECT (dsoundsink, "AC3 passthrough supported");
      hRes = IDirectSoundBuffer_Release (tmpBuffer);
      if (FAILED (hRes)) {
        gchar *error_text = gst_hres_to_string (hRes);
        GST_DEBUG_OBJECT (dsoundsink,
            "(IDirectSoundBuffer_Release returned: %s)\n", error_text);
        g_free (error_text);
      }
    }
  #else
    tmp = gst_caps_new_empty_simple ("audio/x-ac3");
    tmp2 = gst_caps_subtract (caps, tmp);
    gst_caps_unref (tmp);
    gst_caps_unref (caps);
    caps = tmp2;
    tmp = gst_caps_new_empty_simple ("audio/x-dts");
    tmp2 = gst_caps_subtract (caps, tmp);
    gst_caps_unref (tmp);
    gst_caps_unref (caps);
    caps = tmp2;
  #endif
  
    return caps;
  }
  
  static GstBuffer *
  gst_directsound_sink_payload (GstAudioBaseSink * sink, GstBuffer * buf)
  {
    if (gst_directsound_sink_is_spdif_format (&sink->ringbuffer->spec)) {
      gint framesize = gst_audio_iec61937_frame_size (&sink->ringbuffer->spec);
      GstBuffer *out;
      GstMapInfo infobuf, infoout;
      gboolean success;
  
      if (framesize <= 0)
        return NULL;
  
      out = gst_buffer_new_and_alloc (framesize);
  
      if (!gst_buffer_map (buf, &infobuf, GST_MAP_READWRITE)) {
        gst_buffer_unref (out);
        return NULL;
      }
      if (!gst_buffer_map (out, &infoout, GST_MAP_READWRITE)) {
        gst_buffer_unmap (buf, &infobuf);
        gst_buffer_unref (out);
        return NULL;
      }
      success = gst_audio_iec61937_payload (infobuf.data, infobuf.size,
          infoout.data, infoout.size, &sink->ringbuffer->spec, G_BYTE_ORDER);
      if (!success) {
        gst_buffer_unmap (out, &infoout);
        gst_buffer_unmap (buf, &infobuf);
        gst_buffer_unref (out);
        return NULL;
      }
  
      gst_buffer_copy_into (out, buf, GST_BUFFER_COPY_ALL, 0, -1);
      /* Fix endianness */
      _swab ((gchar *) infoout.data, (gchar *) infoout.data, infobuf.size);
      gst_buffer_unmap (out, &infoout);
      gst_buffer_unmap (buf, &infobuf);
      return out;
    } else
      return gst_buffer_ref (buf);
  }
  
  static void
  gst_directsound_sink_set_volume (GstDirectSoundSink * dsoundsink,
      gdouble dvolume, gboolean store)
  {
    glong volume;
  
    volume = dvolume * 100;
    if (store)
      dsoundsink->volume = volume;
  
    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 (volume == 0 || dsoundsink->mute)
        dsVolume = -10000;
      else
        dsVolume = 100 * (long) (20 * log10 ((double) volume / 100.));
      dsVolume = CLAMP (dsVolume, -10000, 0);
  
      GST_DEBUG_OBJECT (dsoundsink,
          "Setting volume on secondary buffer to %d from %d", (int) dsVolume,
          (int) volume);
      IDirectSoundBuffer_SetVolume (dsoundsink->pDSBSecondary, dsVolume);
    }
  }
  
  gdouble
  gst_directsound_sink_get_volume (GstDirectSoundSink * dsoundsink)
  {
    return (gdouble) dsoundsink->volume / 100;
  }
  
  static void
  gst_directsound_sink_set_mute (GstDirectSoundSink * dsoundsink, gboolean mute)
  {
    if (mute) {
      gst_directsound_sink_set_volume (dsoundsink, 0, FALSE);
      dsoundsink->mute = TRUE;
    } else {
      gst_directsound_sink_set_volume (dsoundsink,
          gst_directsound_sink_get_volume (dsoundsink), FALSE);
      dsoundsink->mute = FALSE;
    }
  
  }
  
  static gboolean
  gst_directsound_sink_get_mute (GstDirectSoundSink * dsoundsink)
  {
    return dsoundsink->mute;
  }
  
  static const gchar *
  gst_directsound_sink_get_device (GstDirectSoundSink * dsoundsink)
  {
    return dsoundsink->device_id;
  }
  
  static void
  gst_directsound_sink_set_device (GstDirectSoundSink * dsoundsink,
      const gchar * device_id)
  {
    g_free (dsoundsink->device_id);
    dsoundsink->device_id = g_strdup (device_id);
  }
  
  /* Converts a HRESULT error to a text string
   * LPTSTR is either a */
  static gchar *
  gst_hres_to_string (HRESULT hRes)
  {
    DWORD flags;
    gchar *ret_text;
    LPTSTR error_text = NULL;
  
    flags = FORMAT_MESSAGE_FROM_SYSTEM | FORMAT_MESSAGE_ALLOCATE_BUFFER
        | FORMAT_MESSAGE_IGNORE_INSERTS;
    FormatMessage (flags, NULL, hRes, MAKELANGID (LANG_NEUTRAL, SUBLANG_DEFAULT),
        (LPTSTR) & error_text, 0, NULL);
  
  #ifdef UNICODE
    /* If UNICODE is defined, LPTSTR is LPWSTR which is UTF-16 */
    ret_text = g_utf16_to_utf8 (error_text, 0, NULL, NULL, NULL);
  #else
    ret_text = g_strdup (error_text);
  #endif
  
    LocalFree (error_text);
    return ret_text;
  }
  