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296687a398
Original commit message from CVS: * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: Remove include of unused headers. * sys/waveform/gstwaveformplugin.c: * sys/waveform/gstwaveformsink.c: * sys/waveform/gstwaveformsink.h: * win32/vs6/libgstwaveform.dsp: Add a new waveform plugin which includes an audio sink element using the WaveForm win32 API. * win32/MANIFEST: Add the new project file form waveform plugin.
529 lines
17 KiB
C
529 lines
17 KiB
C
/* GStreamer
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* Copyright (C) 2005 Sebastien Moutte <sebastien@moutte.net>
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*
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* gstdirectsoundsink.c:
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-directsoundsink
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* @short_description: output sound using Directsound API
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*
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* <refsect2>
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* <para>
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* This element lets you output sound using the DirectSound API.
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* </para>
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* <para>
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* Note that you should almost always use generic audio conversion elements
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* like audioconvert and audioresample in front of an audiosink to make sure
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* your pipeline works under all circumstances (those conversion elements will
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* act in passthrough-mode if no conversion is necessary).
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* </para>
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* <title>Example pipelines</title>
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* <para>
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* <programlisting>
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* gst-launch-0.10 -v audiotestsrc ! audioconvert ! volume volume=0.1 ! directsoundsink
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* </programlisting>
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* will output a sine wave (continuous beep sound) to your sound card (with
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* a very low volume as precaution).
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* </para>
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* <para>
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* <programlisting>
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* gst-launch-0.10 -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! directsoundsink
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* </programlisting>
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* will play an Ogg/Vorbis audio file and output it.
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* </para>
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstdirectsoundsink.h"
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GST_DEBUG_CATEGORY_STATIC (directsoundsink_debug);
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/* elementfactory information */
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static const GstElementDetails gst_directsound_sink_details =
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GST_ELEMENT_DETAILS ("Direct Sound Audio Sink",
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"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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static void gst_directsound_sink_base_init (gpointer g_class);
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static void gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass);
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static void gst_directsound_sink_init (GstDirectSoundSink * dsoundsink,
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GstDirectSoundSinkClass * g_class);
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static void gst_directsound_sink_finalise (GObject * object);
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static void gst_directsound_sink_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_directsound_sink_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstCaps *gst_directsound_sink_getcaps (GstBaseSink * bsink);
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static gboolean gst_directsound_sink_prepare (GstAudioSink * asink,
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GstRingBufferSpec * 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 guint gst_directsound_sink_delay (GstAudioSink * asink);
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static void gst_directsound_sink_reset (GstAudioSink * asink);
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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, FALSE }, "
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"width = (int) 16, "
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"depth = (int) 16, "
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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) { TRUE, FALSE }, "
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"width = (int) 8, "
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"depth = (int) 8, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]"));
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enum
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{
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PROP_0,
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PROP_ATTENUATION
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};
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static void
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_do_init (GType directsoundsink_type)
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{
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GST_DEBUG_CATEGORY_INIT (directsoundsink_debug, "directsoundsink", 0,
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"DirectSound sink");
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}
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GST_BOILERPLATE_FULL (GstDirectSoundSink, gst_directsound_sink, GstAudioSink,
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GST_TYPE_AUDIO_SINK, _do_init);
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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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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 (element_class, &gst_directsound_sink_details);
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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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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_directsound_sink_finalise);
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gobject_class->get_property =
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GST_DEBUG_FUNCPTR (gst_directsound_sink_get_property);
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gobject_class->set_property =
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GST_DEBUG_FUNCPTR (gst_directsound_sink_set_property);
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gstbasesink_class->get_caps =
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GST_DEBUG_FUNCPTR (gst_directsound_sink_getcaps);
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gstaudiosink_class->prepare =
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GST_DEBUG_FUNCPTR (gst_directsound_sink_prepare);
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gstaudiosink_class->unprepare =
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GST_DEBUG_FUNCPTR (gst_directsound_sink_unprepare);
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gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_directsound_sink_open);
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gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_directsound_sink_close);
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gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_directsound_sink_write);
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gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_directsound_sink_delay);
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gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_directsound_sink_reset);
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_ATTENUATION,
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g_param_spec_long ("attenuation", "Attenuation of the sound",
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"The attenuation for the directsound buffer (default is 0 so the directsound buffer will not be attenuated)",
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-10000, 0, 0, G_PARAM_READWRITE));
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}
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static void
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gst_directsound_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstDirectSoundSink *dsoundsink;
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dsoundsink = GST_DIRECTSOUND_SINK (object);
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switch (prop_id) {
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case PROP_ATTENUATION:
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{
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glong attenuation = g_value_get_long (value);
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if (attenuation != dsoundsink->attenuation) {
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dsoundsink->attenuation = attenuation;
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if (dsoundsink->pDSBSecondary)
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IDirectSoundBuffer_SetVolume (dsoundsink->pDSBSecondary,
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dsoundsink->attenuation);
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}
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break;
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}
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_directsound_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstDirectSoundSink *dsoundsink;
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dsoundsink = GST_DIRECTSOUND_SINK (object);
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switch (prop_id) {
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case PROP_ATTENUATION:
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g_value_set_long (value, dsoundsink->attenuation);
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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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break;
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}
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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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{
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dsoundsink->pDS = NULL;
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dsoundsink->pDSBSecondary = NULL;
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dsoundsink->current_circular_offset = 0;
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dsoundsink->buffer_size = DSBSIZE_MIN;
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dsoundsink->attenuation = 0;
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dsoundsink->dsound_lock = g_mutex_new ();
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dsoundsink->first_buffer_after_reset = FALSE;
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}
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static GstCaps *
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gst_directsound_sink_getcaps (GstBaseSink * bsink)
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{
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GstDirectSoundSink *dsoundsink;
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dsoundsink = GST_DIRECTSOUND_SINK (bsink);
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return
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gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
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(dsoundsink)));
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}
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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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HRESULT hRes;
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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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("gst_directsound_sink_open: DirectSoundCreate: %s",
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DXGetErrorString9 (hRes)), (NULL));
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return FALSE;
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}
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if (FAILED (hRes = IDirectSound_SetCooperativeLevel (dsoundsink->pDS,
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GetDesktopWindow (), DSSCL_PRIORITY))) {
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GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
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("gst_directsound_sink_open: IDirectSound_SetCooperativeLevel: %s",
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DXGetErrorString9 (hRes)), (NULL));
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return FALSE;
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}
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return TRUE;
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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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{
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GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (asink);
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HRESULT hRes;
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DSBUFFERDESC descSecondary;
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WAVEFORMATEX wfx;
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/*save number of bytes per sample */
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dsoundsink->bytes_per_sample = spec->bytes_per_sample;
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/* fill the WAVEFORMATEX struture with spec params */
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memset (&wfx, 0, sizeof (wfx));
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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.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
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/* directsound buffer size can handle 1/2 sec of the stream */
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dsoundsink->buffer_size = wfx.nAvgBytesPerSec / 2;
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GST_INFO_OBJECT (dsoundsink,
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"GstRingBufferSpec->channels: %d, GstRingBufferSpec->rate: %d, GstRingBufferSpec->bytes_per_sample: %d\n"
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"WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d, WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld\n"
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"Size of dsound cirucular buffe=>%d\n", spec->channels, spec->rate,
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spec->bytes_per_sample, wfx.nSamplesPerSec, wfx.wBitsPerSample,
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wfx.nBlockAlign, wfx.nAvgBytesPerSec, dsoundsink->buffer_size);
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/* create a secondary directsound buffer */
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memset (&descSecondary, 0, sizeof (DSBUFFERDESC));
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descSecondary.dwSize = sizeof (DSBUFFERDESC);
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descSecondary.dwFlags = DSBCAPS_GETCURRENTPOSITION2 |
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DSBCAPS_GLOBALFOCUS | DSBCAPS_CTRLVOLUME;
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descSecondary.dwBufferBytes = dsoundsink->buffer_size;
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descSecondary.lpwfxFormat = (WAVEFORMATEX *) & wfx;
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hRes = IDirectSound_CreateSoundBuffer (dsoundsink->pDS, &descSecondary,
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&dsoundsink->pDSBSecondary, NULL);
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if (FAILED (hRes)) {
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GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ,
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("gst_directsound_sink_prepare: IDirectSound_CreateSoundBuffer: %s",
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DXGetErrorString9 (hRes)), (NULL));
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return FALSE;
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}
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if (dsoundsink->attenuation != 0)
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IDirectSoundBuffer_SetVolume (dsoundsink->pDSBSecondary,
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dsoundsink->attenuation);
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return TRUE;
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}
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static gboolean
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gst_directsound_sink_unprepare (GstAudioSink * asink)
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{
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GstDirectSoundSink *dsoundsink;
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dsoundsink = GST_DIRECTSOUND_SINK (asink);
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/* release secondary DirectSound buffer */
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if (dsoundsink->pDSBSecondary)
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IDirectSoundBuffer_Release (dsoundsink->pDSBSecondary);
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return TRUE;
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}
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static gboolean
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gst_directsound_sink_close (GstAudioSink * asink)
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{
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GstDirectSoundSink *dsoundsink = NULL;
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dsoundsink = GST_DIRECTSOUND_SINK (asink);
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/* release DirectSound object */
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g_return_val_if_fail (dsoundsink->pDS != NULL, FALSE);
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IDirectSound_Release (dsoundsink->pDS);
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return TRUE;
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}
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static guint
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gst_directsound_sink_write (GstAudioSink * asink, gpointer data, guint length)
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{
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GstDirectSoundSink *dsoundsink;
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DWORD dwStatus;
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HRESULT hRes;
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LPVOID pLockedBuffer1 = NULL, pLockedBuffer2 = NULL;
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DWORD dwSizeBuffer1, dwSizeBuffer2;
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DWORD dwCurrentPlayCursor;
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dsoundsink = GST_DIRECTSOUND_SINK (asink);
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GST_DSOUND_LOCK (dsoundsink);
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/* get current buffer status */
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hRes = IDirectSoundBuffer_GetStatus (dsoundsink->pDSBSecondary, &dwStatus);
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/* get current play cursor position */
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hRes = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary,
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&dwCurrentPlayCursor, NULL);
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if (SUCCEEDED (hRes) && (dwStatus & DSBSTATUS_PLAYING)) {
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DWORD dwFreeBufferSize;
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calculate_freesize:
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/* calculate the free size of the circular buffer */
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if (dwCurrentPlayCursor < dsoundsink->current_circular_offset)
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dwFreeBufferSize =
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dsoundsink->buffer_size - (dsoundsink->current_circular_offset -
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dwCurrentPlayCursor);
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else
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dwFreeBufferSize =
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dwCurrentPlayCursor - dsoundsink->current_circular_offset;
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if (length >= dwFreeBufferSize) {
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Sleep (100);
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hRes = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary,
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&dwCurrentPlayCursor, NULL);
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hRes =
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IDirectSoundBuffer_GetStatus (dsoundsink->pDSBSecondary, &dwStatus);
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if (SUCCEEDED (hRes) && (dwStatus & DSBSTATUS_PLAYING))
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goto calculate_freesize;
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else {
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dsoundsink->first_buffer_after_reset = FALSE;
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GST_DSOUND_UNLOCK (dsoundsink);
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return 0;
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}
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}
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}
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if (dwStatus & DSBSTATUS_BUFFERLOST) {
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hRes = IDirectSoundBuffer_Restore (dsoundsink->pDSBSecondary); /*need a loop waiting the buffer is restored?? */
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dsoundsink->current_circular_offset = 0;
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}
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hRes = IDirectSoundBuffer_Lock (dsoundsink->pDSBSecondary,
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dsoundsink->current_circular_offset, length, &pLockedBuffer1,
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&dwSizeBuffer1, &pLockedBuffer2, &dwSizeBuffer2, 0L);
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if (SUCCEEDED (hRes)) {
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// Write to pointers without reordering.
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memcpy (pLockedBuffer1, data, dwSizeBuffer1);
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if (pLockedBuffer2 != NULL)
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memcpy (pLockedBuffer2, (LPBYTE) data + dwSizeBuffer1, dwSizeBuffer2);
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// Update where the buffer will lock (for next time)
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dsoundsink->current_circular_offset += dwSizeBuffer1 + dwSizeBuffer2;
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dsoundsink->current_circular_offset %= dsoundsink->buffer_size; /* Circular buffer */
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hRes = IDirectSoundBuffer_Unlock (dsoundsink->pDSBSecondary, pLockedBuffer1,
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dwSizeBuffer1, pLockedBuffer2, dwSizeBuffer2);
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}
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/* if the buffer was not in playing state yet, call play on the buffer
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except if this buffer is the fist after a reset (base class call reset and write a buffer when setting the sink to pause) */
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if (!(dwStatus & DSBSTATUS_PLAYING) &&
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dsoundsink->first_buffer_after_reset == FALSE) {
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hRes = IDirectSoundBuffer_Play (dsoundsink->pDSBSecondary, 0, 0,
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DSBPLAY_LOOPING);
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}
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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);
|
|
}
|