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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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direcsoundsink: Port element to 0.11
This commit is contained in:
Andoni Morales Alastruey
Sebastian Dröge
parent
9095f455d5
commit
83090f6530
2 changed files with 149 additions and 229 deletions
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@ -52,6 +52,8 @@
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#include "config.h"
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#endif
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#include <gst/base/gstbasesink.h>
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#include <gst/audio/streamvolume.h>
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#include "gstdirectsoundsink.h"
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#include <math.h>
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@ -63,229 +65,85 @@
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#endif
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#endif
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#define DEFAULT_MUTE FALSE
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GST_DEBUG_CATEGORY_STATIC (directsoundsink_debug);
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#define GST_CAT_DEFAULT directsoundsink_debug
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static void gst_directsound_sink_finalise (GObject * object);
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static void gst_directsound_sink_finalize (GObject * object);
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static void gst_directsound_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_directsound_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstCaps *gst_directsound_sink_getcaps (GstBaseSink * bsink);
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static GstCaps *gst_directsound_sink_getcaps (GstBaseSink * bsink,
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GstCaps * filter);
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static gboolean gst_directsound_sink_prepare (GstAudioSink * asink,
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GstRingBufferSpec * spec);
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GstAudioRingBufferSpec * spec);
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static gboolean gst_directsound_sink_unprepare (GstAudioSink * asink);
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static gboolean gst_directsound_sink_open (GstAudioSink * asink);
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static gboolean gst_directsound_sink_close (GstAudioSink * asink);
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static guint gst_directsound_sink_write (GstAudioSink * asink, gpointer data,
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guint length);
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static gint gst_directsound_sink_write (GstAudioSink * asink,
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gpointer data, guint length);
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static guint gst_directsound_sink_delay (GstAudioSink * asink);
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static void gst_directsound_sink_reset (GstAudioSink * asink);
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static GstCaps *gst_directsound_probe_supported_formats (GstDirectSoundSink *
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dsoundsink, const GstCaps * template_caps);
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/* interfaces */
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static void gst_directsound_sink_interfaces_init (GType type);
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static void
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gst_directsound_sink_implements_interface_init (GstImplementsInterfaceClass *
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iface);
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static void gst_directsound_sink_mixer_interface_init (GstMixerInterface *
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iface);
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static void gst_directsound_sink_set_volume (GstDirectSoundSink * sink,
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gdouble volume, gboolean store);
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static gdouble gst_directsound_sink_get_volume (GstDirectSoundSink * sink);
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static void gst_directsound_sink_set_mute (GstDirectSoundSink * sink,
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gboolean mute);
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static gboolean gst_directsound_sink_get_mute (GstDirectSoundSink * sink);
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static GstStaticPadTemplate directsoundsink_sink_factory =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"signed = (boolean) TRUE, "
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"width = (int) 16, "
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"depth = (int) 16, "
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) S16LE, "
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"layout = (string) interleaved, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
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"audio/x-raw-int, "
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"signed = (boolean) FALSE, "
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"width = (int) 8, "
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"depth = (int) 8, "
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"audio/x-raw, "
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"format = (string) S8, "
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"layout = (string) interleaved, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ];"
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"audio/x-iec958"));
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enum
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{
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PROP_0,
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PROP_VOLUME
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PROP_VOLUME,
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PROP_MUTE
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};
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GST_BOILERPLATE_FULL (GstDirectSoundSink, gst_directsound_sink, GstAudioSink,
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GST_TYPE_AUDIO_SINK, gst_directsound_sink_interfaces_init);
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/* interfaces stuff */
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static void
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gst_directsound_sink_interfaces_init (GType type)
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{
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static const GInterfaceInfo implements_interface_info = {
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(GInterfaceInitFunc) gst_directsound_sink_implements_interface_init,
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NULL,
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NULL,
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};
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static const GInterfaceInfo mixer_interface_info = {
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(GInterfaceInitFunc) gst_directsound_sink_mixer_interface_init,
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NULL,
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NULL,
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};
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g_type_add_interface_static (type,
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GST_TYPE_IMPLEMENTS_INTERFACE, &implements_interface_info);
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g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_interface_info);
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}
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static gboolean
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gst_directsound_sink_interface_supported (GstImplementsInterface * iface,
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GType iface_type)
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{
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g_return_val_if_fail (iface_type == GST_TYPE_MIXER, FALSE);
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/* for the sake of this example, we'll always support it. However, normally,
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* you would check whether the device you've opened supports mixers. */
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return TRUE;
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}
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#define gst_directsound_sink_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstDirectSoundSink, gst_directsound_sink,
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GST_TYPE_AUDIO_SINK, G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL)
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);
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static void
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gst_directsound_sink_implements_interface_init (GstImplementsInterfaceClass *
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iface)
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gst_directsound_sink_finalize (GObject * object)
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{
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iface->supported = gst_directsound_sink_interface_supported;
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}
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/*
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* This function returns the list of support tracks (inputs, outputs)
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* on this element instance. Elements usually build this list during
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* _init () or when going from NULL to READY.
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*/
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static const GList *
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gst_directsound_sink_mixer_list_tracks (GstMixer * mixer)
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{
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GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (mixer);
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return dsoundsink->tracks;
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}
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static void
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gst_directsound_sink_set_volume (GstDirectSoundSink * dsoundsink)
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{
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if (dsoundsink->pDSBSecondary) {
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/* DirectSound controls volume using units of 100th of a decibel,
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* ranging from -10000 to 0. We use a linear scale of 0 - 100
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* here, so remap.
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*/
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long dsVolume;
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if (dsoundsink->volume == 0)
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dsVolume = -10000;
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else
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dsVolume = 100 * (long) (20 * log10 ((double) dsoundsink->volume / 100.));
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dsVolume = CLAMP (dsVolume, -10000, 0);
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GST_DEBUG_OBJECT (dsoundsink,
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"Setting volume on secondary buffer to %d from %d", (int) dsVolume,
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(int) dsoundsink->volume);
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IDirectSoundBuffer_SetVolume (dsoundsink->pDSBSecondary, dsVolume);
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}
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}
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/*
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* Set volume. volumes is an array of size track->num_channels, and
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* each value in the array gives the wanted volume for one channel
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* on the track.
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*/
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static void
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gst_directsound_sink_mixer_set_volume (GstMixer * mixer,
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GstMixerTrack * track, gint * volumes)
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{
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GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (mixer);
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if (volumes[0] != dsoundsink->volume) {
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dsoundsink->volume = volumes[0];
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gst_directsound_sink_set_volume (dsoundsink);
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}
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}
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static void
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gst_directsound_sink_mixer_get_volume (GstMixer * mixer,
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GstMixerTrack * track, gint * volumes)
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{
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GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (mixer);
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volumes[0] = dsoundsink->volume;
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}
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static void
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gst_directsound_sink_mixer_interface_init (GstMixerInterface * iface)
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{
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/* the mixer interface requires a definition of the mixer type:
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* hardware or software? */
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GST_MIXER_TYPE (iface) = GST_MIXER_SOFTWARE;
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/* virtual function pointers */
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iface->list_tracks = gst_directsound_sink_mixer_list_tracks;
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iface->set_volume = gst_directsound_sink_mixer_set_volume;
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iface->get_volume = gst_directsound_sink_mixer_get_volume;
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}
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static void
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gst_directsound_sink_finalise (GObject * object)
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{
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GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (object);
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g_mutex_free (dsoundsink->dsound_lock);
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if (dsoundsink->tracks) {
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g_list_foreach (dsoundsink->tracks, (GFunc) g_object_unref, NULL);
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g_list_free (dsoundsink->tracks);
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dsoundsink->tracks = NULL;
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}
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_directsound_sink_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_set_details_simple (element_class,
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"Direct Sound Audio Sink", "Sink/Audio",
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"Output to a sound card via Direct Sound",
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"Sebastien Moutte <sebastien@moutte.net>");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&directsoundsink_sink_factory));
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}
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static void
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gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseSinkClass *gstbasesink_class;
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GstBaseAudioSinkClass *gstbaseaudiosink_class;
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GstAudioSinkClass *gstaudiosink_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasesink_class = (GstBaseSinkClass *) klass;
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gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
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gstaudiosink_class = (GstAudioSinkClass *) klass;
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
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GstAudioSinkClass *gstaudiosink_class = GST_AUDIO_SINK_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (directsoundsink_debug, "directsoundsink", 0,
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"DirectSound sink");
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_directsound_sink_finalise;
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gobject_class->finalize = gst_directsound_sink_finalize;
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gobject_class->set_property = gst_directsound_sink_set_property;
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gobject_class->get_property = gst_directsound_sink_get_property;
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@ -307,23 +165,27 @@ gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass)
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g_param_spec_double ("volume", "Volume",
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"Volume of this stream", 0.0, 1.0, 1.0,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_MUTE,
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g_param_spec_boolean ("mute", "Mute",
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"Mute state of this stream", DEFAULT_MUTE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_set_details_simple (element_class,
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"Direct Sound Audio Sink", "Sink/Audio",
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"Output to a sound card via Direct Sound",
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"Sebastien Moutte <sebastien@moutte.net>");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&directsoundsink_sink_factory));
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}
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static void
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gst_directsound_sink_init (GstDirectSoundSink * dsoundsink,
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GstDirectSoundSinkClass * g_class)
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gst_directsound_sink_init (GstDirectSoundSink * dsoundsink)
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{
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GstMixerTrack *track = NULL;
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dsoundsink->tracks = NULL;
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track = g_object_new (GST_TYPE_MIXER_TRACK, NULL);
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track->label = g_strdup ("DSoundTrack");
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track->num_channels = 2;
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track->min_volume = 0;
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track->max_volume = 100;
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track->flags = GST_MIXER_TRACK_OUTPUT;
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dsoundsink->tracks = g_list_append (dsoundsink->tracks, track);
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dsoundsink->volume = 100;
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dsoundsink->mute = FALSE;
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dsoundsink->pDS = NULL;
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dsoundsink->cached_caps = NULL;
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dsoundsink->pDSBSecondary = NULL;
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@ -342,8 +204,10 @@ gst_directsound_sink_set_property (GObject * object,
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switch (prop_id) {
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case PROP_VOLUME:
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sink->volume = (int) (g_value_get_double (value) * 100);
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gst_directsound_sink_set_volume (sink);
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gst_directsound_sink_set_volume (sink, g_value_get_double (value), TRUE);
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break;
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case PROP_MUTE:
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gst_directsound_sink_set_mute (sink, g_value_get_boolean (value));
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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@ -359,7 +223,10 @@ gst_directsound_sink_get_property (GObject * object,
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switch (prop_id) {
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case PROP_VOLUME:
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g_value_set_double (value, (double) sink->volume / 100.);
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g_value_set_double (value, gst_directsound_sink_get_volume (sink));
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break;
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case PROP_MUTE:
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g_value_set_boolean (value, gst_directsound_sink_get_mute (sink));
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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@ -368,7 +235,7 @@ gst_directsound_sink_get_property (GObject * object,
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}
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static GstCaps *
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gst_directsound_sink_getcaps (GstBaseSink * bsink)
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gst_directsound_sink_getcaps (GstBaseSink * bsink, GstCaps * filter)
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{
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GstElementClass *element_class;
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GstPadTemplate *pad_template;
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@ -410,9 +277,11 @@ gst_directsound_sink_getcaps (GstBaseSink * bsink)
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static gboolean
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gst_directsound_sink_open (GstAudioSink * asink)
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{
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GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (asink);
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GstDirectSoundSink *dsoundsink;
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HRESULT hRes;
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dsoundsink = GST_DIRECTSOUND_SINK (asink);
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/* create and initialize a DirecSound object */
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if (FAILED (hRes = DirectSoundCreate (NULL, &dsoundsink->pDS, NULL))) {
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GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
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@ -433,47 +302,50 @@ gst_directsound_sink_open (GstAudioSink * asink)
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}
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static gboolean
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gst_directsound_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
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gst_directsound_sink_prepare (GstAudioSink * asink,
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GstAudioRingBufferSpec * spec)
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{
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GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (asink);
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GstDirectSoundSink *dsoundsink;
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HRESULT hRes;
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DSBUFFERDESC descSecondary;
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WAVEFORMATEX wfx;
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dsoundsink = GST_DIRECTSOUND_SINK (asink);
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/*save number of bytes per sample and buffer format */
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dsoundsink->bytes_per_sample = spec->bytes_per_sample;
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dsoundsink->buffer_format = spec->format;
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dsoundsink->bytes_per_sample = spec->info.bpf;
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dsoundsink->type = spec->type;
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/* fill the WAVEFORMATEX structure with spec params */
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memset (&wfx, 0, sizeof (wfx));
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if (spec->format != GST_IEC958) {
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if (spec->type != GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958) {
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wfx.cbSize = sizeof (wfx);
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wfx.wFormatTag = WAVE_FORMAT_PCM;
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wfx.nChannels = spec->channels;
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wfx.nSamplesPerSec = spec->rate;
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wfx.wBitsPerSample = (spec->bytes_per_sample * 8) / wfx.nChannels;
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wfx.nBlockAlign = spec->bytes_per_sample;
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wfx.nChannels = spec->info.channels;
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wfx.nSamplesPerSec = spec->info.rate;
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wfx.wBitsPerSample = (spec->info.bpf * 8) / wfx.nChannels;
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wfx.nBlockAlign = spec->info.bpf;
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wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
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/* Create directsound buffer with size based on our configured
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/* Create directsound buffer with size based on our configured
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* buffer_size (which is 200 ms by default) */
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dsoundsink->buffer_size =
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gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->buffer_time,
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GST_MSECOND);
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/* Make sure we make those numbers multiple of our sample size in bytes */
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dsoundsink->buffer_size += dsoundsink->buffer_size % spec->bytes_per_sample;
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dsoundsink->buffer_size += dsoundsink->buffer_size % spec->info.bpf;
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spec->segsize =
|
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gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->latency_time,
|
||||
GST_MSECOND);
|
||||
spec->segsize += spec->segsize % spec->bytes_per_sample;
|
||||
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 = spec->rate;
|
||||
wfx.nSamplesPerSec = spec->info.rate;
|
||||
wfx.wBitsPerSample = 16;
|
||||
wfx.nBlockAlign = wfx.wBitsPerSample / 8 * wfx.nChannels;
|
||||
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
|
||||
|
|
@ -491,15 +363,15 @@ gst_directsound_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
|
|||
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,
|
||||
"Size of dsound circular buffer=>%d\n", spec->info.channels,
|
||||
spec->info.rate, spec->info.bpf, 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)
|
||||
if (spec->type != GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958)
|
||||
descSecondary.dwFlags |= DSBCAPS_CTRLVOLUME;
|
||||
|
||||
descSecondary.dwBufferBytes = dsoundsink->buffer_size;
|
||||
|
|
@ -514,7 +386,7 @@ gst_directsound_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
|
|||
return FALSE;
|
||||
}
|
||||
|
||||
gst_directsound_sink_set_volume (dsoundsink);
|
||||
gst_directsound_sink_set_volume (dsoundsink, dsoundsink->volume, FALSE);
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
|
@ -552,7 +424,7 @@ gst_directsound_sink_close (GstAudioSink * asink)
|
|||
return TRUE;
|
||||
}
|
||||
|
||||
static guint
|
||||
static gint
|
||||
gst_directsound_sink_write (GstAudioSink * asink, gpointer data, guint length)
|
||||
{
|
||||
GstDirectSoundSink *dsoundsink;
|
||||
|
|
@ -565,7 +437,7 @@ gst_directsound_sink_write (GstAudioSink * asink, gpointer data, guint length)
|
|||
dsoundsink = GST_DIRECTSOUND_SINK (asink);
|
||||
|
||||
/* Fix endianness */
|
||||
if (dsoundsink->buffer_format == GST_IEC958)
|
||||
if (dsoundsink->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958)
|
||||
_swab (data, data, length);
|
||||
|
||||
GST_DSOUND_LOCK (dsoundsink);
|
||||
|
|
@ -720,12 +592,12 @@ gst_directsound_sink_reset (GstAudioSink * asink)
|
|||
GST_DSOUND_UNLOCK (dsoundsink);
|
||||
}
|
||||
|
||||
/*
|
||||
* gst_directsound_probe_supported_formats:
|
||||
*
|
||||
* Takes the template caps and returns the subset which is actually
|
||||
* supported by this device.
|
||||
*
|
||||
/*
|
||||
* gst_directsound_probe_supported_formats:
|
||||
*
|
||||
* Takes the template caps and returns the subset which is actually
|
||||
* supported by this device.
|
||||
*
|
||||
*/
|
||||
|
||||
static GstCaps *
|
||||
|
|
@ -739,8 +611,8 @@ gst_directsound_probe_supported_formats (GstDirectSoundSink * dsoundsink,
|
|||
|
||||
caps = gst_caps_copy (template_caps);
|
||||
|
||||
/*
|
||||
* Check availability of digital output by trying to create an SPDIF buffer
|
||||
/*
|
||||
* Check availability of digital output by trying to create an SPDIF buffer
|
||||
*/
|
||||
|
||||
#ifdef WAVE_FORMAT_DOLBY_AC3_SPDIF
|
||||
|
|
@ -754,7 +626,7 @@ gst_directsound_probe_supported_formats (GstDirectSoundSink * dsoundsink,
|
|||
wfx.nBlockAlign = 4;
|
||||
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
|
||||
|
||||
// create a secondary directsound buffer
|
||||
// create a secondary directsound buffer
|
||||
memset (&descSecondary, 0, sizeof (DSBUFFERDESC));
|
||||
descSecondary.dwSize = sizeof (DSBUFFERDESC);
|
||||
descSecondary.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS;
|
||||
|
|
@ -768,7 +640,7 @@ gst_directsound_probe_supported_formats (GstDirectSoundSink * dsoundsink,
|
|||
"(IDirectSound_CreateSoundBuffer returned: %s)\n",
|
||||
DXGetErrorString9 (hRes));
|
||||
caps =
|
||||
gst_caps_subtract (caps, gst_caps_new_simple ("audio/x-iec958", NULL));
|
||||
gst_caps_subtract (caps, gst_caps_new_empty_simple ("audio/x-iec958"));
|
||||
} else {
|
||||
GST_INFO_OBJECT (dsoundsink, "AC3 passthrough supported");
|
||||
hRes = IDirectSoundBuffer_Release (dsoundsink->pDSBSecondary);
|
||||
|
|
@ -784,3 +656,53 @@ gst_directsound_probe_supported_formats (GstDirectSoundSink * dsoundsink,
|
|||
|
||||
return caps;
|
||||
}
|
||||
|
||||
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 (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);
|
||||
}
|
||||
}
|
||||
|
||||
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);
|
||||
else
|
||||
gst_directsound_sink_set_volume (dsoundsink, dsoundsink->volume, FALSE);
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_directsound_sink_get_mute (GstDirectSoundSink * dsoundsink)
|
||||
{
|
||||
return FALSE;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -31,7 +31,7 @@
|
|||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/audio/gstaudiosink.h>
|
||||
#include <gst/interfaces/mixer.h>
|
||||
#include "gstdirectsoundringbuffer.h"
|
||||
|
||||
#include <windows.h>
|
||||
#include <dxerr9.h>
|
||||
|
|
@ -56,6 +56,7 @@ struct _GstDirectSoundSink
|
|||
{
|
||||
GstAudioSink sink;
|
||||
|
||||
|
||||
/* directsound object interface pointer */
|
||||
LPDIRECTSOUND pDS;
|
||||
|
||||
|
|
@ -72,18 +73,15 @@ struct _GstDirectSoundSink
|
|||
|
||||
/* current volume setup by mixer interface */
|
||||
glong volume;
|
||||
|
||||
/* tracks list of our mixer interface implementation */
|
||||
GList *tracks;
|
||||
gboolean mute;
|
||||
|
||||
GstCaps *cached_caps;
|
||||
|
||||
/* lock used to protect writes and resets */
|
||||
GMutex *dsound_lock;
|
||||
|
||||
gboolean first_buffer_after_reset;
|
||||
|
||||
GstBufferFormat buffer_format;
|
||||
GstAudioRingBufferFormatType type;
|
||||
};
|
||||
|
||||
struct _GstDirectSoundSinkClass
|
||||
|
|
|
|||
Loading…
Reference in a new issue