diff options
Diffstat (limited to 'sys/directsound')
| -rw-r--r-- | sys/directsound/Makefile.am | 18 | ||||
| -rw-r--r-- | sys/directsound/Makefile.in | 828 | ||||
| -rw-r--r-- | sys/directsound/gstdirectsoundplugin.c | 49 | ||||
| -rw-r--r-- | sys/directsound/gstdirectsoundsink.c | 785 | ||||
| -rw-r--r-- | sys/directsound/gstdirectsoundsink.h | 97 |
5 files changed, 1777 insertions, 0 deletions
diff --git a/sys/directsound/Makefile.am b/sys/directsound/Makefile.am new file mode 100644 index 0000000..8227905 --- /dev/null +++ b/sys/directsound/Makefile.am @@ -0,0 +1,18 @@ +plugin_LTLIBRARIES = libgstdirectsoundsink.la
+
+libgstdirectsoundsink_la_SOURCES = gstdirectsoundsink.c gstdirectsoundplugin.c
+libgstdirectsoundsink_la_CFLAGS = \ + $(GST_PLUGINS_BASE_CFLAGS) \ + $(GST_BASE_CFLAGS) \ + $(GST_CFLAGS) \ + $(DIRECTSOUND_CFLAGS) +libgstdirectsoundsink_la_LIBADD = \ + $(GST_PLUGINS_BASE_LIBS) \ + -lgstaudio-$(GST_MAJORMINOR) -lgstinterfaces-$(GST_MAJORMINOR) \ + $(GST_BASE_LIBS) \ + $(GST_LIBS) \ + $(DIRECTSOUND_LIBS) +libgstdirectsoundsink_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(DIRECTSOUND_LDFLAGS)
+libgstdirectsoundsink_la_LIBTOOLFLAGS = --tag=disable-static
+
+noinst_HEADERS = gstdirectsoundsink.h diff --git a/sys/directsound/Makefile.in b/sys/directsound/Makefile.in new file mode 100644 index 0000000..94cbb01 --- /dev/null +++ b/sys/directsound/Makefile.in @@ -0,0 +1,828 @@ +# Makefile.in generated by automake 1.11.3 from Makefile.am. +# @configure_input@ + +# Copyright (C) 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001, 2002, +# 2003, 2004, 2005, 2006, 2007, 2008, 2009, 2010, 2011 Free Software +# Foundation, Inc. +# This Makefile.in is free software; the Free Software Foundation +# gives unlimited permission to copy and/or distribute it, +# with or without modifications, as long as this notice is preserved. + +# This program is distributed in the hope that it will be useful, +# but WITHOUT ANY WARRANTY, to the extent permitted by law; without +# even the implied warranty of MERCHANTABILITY or FITNESS FOR A +# PARTICULAR PURPOSE. + +@SET_MAKE@ + + +VPATH = @srcdir@ +pkgdatadir = $(datadir)/@PACKAGE@ +pkgincludedir = 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+* 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 +* +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "gstdirectsoundsink.h" + + +static gboolean +plugin_init (GstPlugin * plugin) +{ + if (!gst_element_register (plugin, "directsoundsink", GST_RANK_PRIMARY, + GST_TYPE_DIRECTSOUND_SINK)) + return FALSE; + + return TRUE; +} + +GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "directsound", + "Direct Sound plugin library", + plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN) diff --git a/sys/directsound/gstdirectsoundsink.c b/sys/directsound/gstdirectsoundsink.c new file mode 100644 index 0000000..2f9a04c --- /dev/null +++ b/sys/directsound/gstdirectsoundsink.c @@ -0,0 +1,785 @@ +/* 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; +} diff --git a/sys/directsound/gstdirectsoundsink.h b/sys/directsound/gstdirectsoundsink.h new file mode 100644 index 0000000..8bb10bf --- /dev/null +++ b/sys/directsound/gstdirectsoundsink.h @@ -0,0 +1,97 @@ +/* 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.h: + * + * 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 + * + * + */ + +#ifndef __GST_DIRECTSOUNDSINK_H__ +#define __GST_DIRECTSOUNDSINK_H__ + +#include <gst/gst.h> +#include <gst/audio/gstaudiosink.h> +#include <gst/interfaces/mixer.h> + +#include <windows.h> +#include <dxerr9.h> +#include <dsound.h> +#include <mmreg.h> +#include <ks.h> +#include <ksmedia.h> + +G_BEGIN_DECLS +#define GST_TYPE_DIRECTSOUND_SINK (gst_directsound_sink_get_type()) +#define GST_DIRECTSOUND_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_DIRECTSOUND_SINK,GstDirectSoundSink)) +#define GST_DIRECTSOUND_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_DIRECTSOUND_SINK,GstDirectSoundSinkClass)) +#define GST_IS_DIRECTSOUND_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_DIRECTSOUND_SINK)) +#define GST_IS_DIRECTSOUND_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_DIRECTSOUND_SINK)) +typedef struct _GstDirectSoundSink GstDirectSoundSink; +typedef struct _GstDirectSoundSinkClass GstDirectSoundSinkClass; + +#define GST_DSOUND_LOCK(obj) (g_mutex_lock (obj->dsound_lock)) +#define GST_DSOUND_UNLOCK(obj) (g_mutex_unlock (obj->dsound_lock)) + +struct _GstDirectSoundSink +{ + GstAudioSink sink; + + /* directsound object interface pointer */ + LPDIRECTSOUND pDS; + + /* directsound sound object interface pointer */ + LPDIRECTSOUNDBUFFER pDSBSecondary; + + /* directSound buffer size */ + guint buffer_size; + + /* offset of the circular buffer where we must write next */ + guint current_circular_offset; + + guint bytes_per_sample; + + /* current volume setup by mixer interface */ + glong volume; + + /* tracks list of our mixer interface implementation */ + GList *tracks; + + GstCaps *cached_caps; + + /* lock used to protect writes and resets */ + GMutex *dsound_lock; + + gboolean first_buffer_after_reset; + + GstBufferFormat buffer_format; +}; + +struct _GstDirectSoundSinkClass +{ + GstAudioSinkClass parent_class; +}; + +GType gst_directsound_sink_get_type (void); + +G_END_DECLS +#endif /* __GST_DIRECTSOUNDSINK_H__ */ |