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directsoundsrc: port to 1.0
https://bugzilla.gnome.org/show_bug.cgi?id=673414
This commit is contained in:
Raimo Järvi
Tim-Philipp Müller
parent
5e6783f5af
commit
a7258842ab
2 changed files with 79 additions and 128 deletions
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@ -324,7 +324,7 @@ GST_PLUGINS_NONPORTED=" androidmedia aiff \
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apexsink cdaudio cog dc1394 dirac directfb \
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gsettings jasper ladspa \
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musepack musicbrainz nas neon ofa openal rsvg sdl sndfile timidity \
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directsound directdraw direct3d9 acm wininet \
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directdraw direct3d9 acm wininet \
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wildmidi xvid lv2 teletextdec sndio uvch264"
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AC_SUBST(GST_PLUGINS_NONPORTED)
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@ -48,12 +48,25 @@
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TODO: add device selection and check rate etc.
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*/
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/**
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* SECTION:element-directsoundsrc
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*
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* Reads audio data using the DirectSound API.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch -v directsoundsrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=dsound.ogg
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* ]| Record from DirectSound and encode to Ogg/Vorbis.
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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 <gst/gst.h>
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#include <gst/audio/gstbaseaudiosrc.h>
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#include <gst/audio/gstaudiobasesrc.h>
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#include "gstdirectsoundsrc.h"
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@ -73,7 +86,6 @@ enum
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PROP_DEVICE
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};
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static HRESULT (WINAPI * pDSoundCaptureCreate) (LPGUID,
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LPDIRECTSOUNDCAPTURE *, LPUNKNOWN);
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@ -88,47 +100,30 @@ static void gst_directsound_src_get_property (GObject * object,
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static gboolean gst_directsound_src_open (GstAudioSrc * asrc);
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static gboolean gst_directsound_src_close (GstAudioSrc * asrc);
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static gboolean gst_directsound_src_prepare (GstAudioSrc * asrc,
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GstRingBufferSpec * spec);
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GstAudioRingBufferSpec * spec);
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static gboolean gst_directsound_src_unprepare (GstAudioSrc * asrc);
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static void gst_directsound_src_reset (GstAudioSrc * asrc);
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static GstCaps *gst_directsound_src_getcaps (GstBaseSrc * bsrc);
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static GstCaps *gst_directsound_src_getcaps (GstBaseSrc * bsrc,
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GstCaps * filter);
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static guint gst_directsound_src_read (GstAudioSrc * asrc,
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gpointer data, guint length);
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static void gst_directsound_src_dispose (GObject * object);
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static void gst_directsound_src_do_init (GType type);
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static guint gst_directsound_src_delay (GstAudioSrc * asrc);
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static GstStaticPadTemplate directsound_src_src_factory =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
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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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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) { S16LE, S8 }, "
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"layout = (string) interleaved, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]"));
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static void
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gst_directsound_src_do_init (GType type)
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{
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GST_DEBUG_CATEGORY_INIT (directsoundsrc_debug, "directsoundsrc", 0,
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"DirectSound Src");
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}
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GST_BOILERPLATE_FULL (GstDirectSoundSrc, gst_directsound_src, GstAudioSrc,
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GST_TYPE_AUDIO_SRC, gst_directsound_src_do_init);
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#define gst_directsound_src_parent_class parent_class
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G_DEFINE_TYPE (GstDirectSoundSrc, gst_directsound_src, GST_TYPE_AUDIO_SRC);
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static void
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gst_directsound_src_dispose (GObject * object)
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@ -146,40 +141,20 @@ gst_directsound_src_finalize (GObject * object)
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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_src_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_DEBUG ("initializing directsoundsrc base\n");
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gst_element_class_set_metadata (element_class, "Direct Sound Audio Src",
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"Source/Audio",
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"Capture from a soundcard via DIRECTSOUND",
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"Joni Valtanen <joni.valtanen@movial.fi>");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&directsound_src_src_factory));
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}
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/* initialize the plugin's class */
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static void
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gst_directsound_src_class_init (GstDirectSoundSrcClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseSrcClass *gstbasesrc_class;
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GstBaseAudioSrcClass *gstbaseaudiosrc_class;
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GstAudioSrcClass *gstaudiosrc_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasesrc_class = (GstBaseSrcClass *) klass;
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gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
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gstaudiosrc_class = (GstAudioSrcClass *) klass;
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GST_DEBUG ("initializing directsoundsrc class\n");
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GST_DEBUG ("initializing directsoundsrc class");
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_directsound_src_finalize);
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gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_directsound_src_dispose);
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@ -199,23 +174,30 @@ gst_directsound_src_class_init (GstDirectSoundSrcClass * klass)
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gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_directsound_src_delay);
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gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_directsound_src_reset);
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GST_DEBUG_CATEGORY_INIT (directsoundsrc_debug, "directsoundsrc", 0,
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"DirectSound Src");
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gst_element_class_set_static_metadata (gstelement_class,
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"DirectSound audio source", "Source/Audio",
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"Capture from a soundcard via DirectSound",
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"Joni Valtanen <joni.valtanen@movial.fi>");
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&directsound_src_src_factory));
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}
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static GstCaps *
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gst_directsound_src_getcaps (GstBaseSrc * bsrc)
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gst_directsound_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
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{
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GstDirectSoundSrc *dsoundsrc;
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GstCaps *caps = NULL;
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GST_DEBUG ("get caps\n");
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GST_DEBUG_OBJECT (bsrc, "get caps");
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dsoundsrc = GST_DIRECTSOUND_SRC (bsrc);
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caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD
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(bsrc)));
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return caps;
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}
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static void
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@ -223,7 +205,7 @@ gst_directsound_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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// GstDirectSoundSrc *src = GST_DIRECTSOUND_SRC (object);
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GST_DEBUG ("set property\n");
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GST_DEBUG ("set property");
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switch (prop_id) {
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#if 0
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@ -246,7 +228,7 @@ gst_directsound_src_get_property (GObject * object, guint prop_id,
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GstDirectSoundSrc *src = GST_DIRECTSOUND_SRC (object);
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#endif
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GST_DEBUG ("get property\n");
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GST_DEBUG ("get property");
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switch (prop_id) {
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#if 0
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@ -268,22 +250,19 @@ gst_directsound_src_get_property (GObject * object, guint prop_id,
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* initialize structure
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*/
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static void
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gst_directsound_src_init (GstDirectSoundSrc * src,
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GstDirectSoundSrcClass * gclass)
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gst_directsound_src_init (GstDirectSoundSrc * src)
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{
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GST_DEBUG ("initializing directsoundsrc\n");
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GST_DEBUG_OBJECT (src, "initializing directsoundsrc");
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src->dsound_lock = g_mutex_new ();
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}
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static gboolean
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gst_directsound_src_open (GstAudioSrc * asrc)
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{
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GstDirectSoundSrc *dsoundsrc;
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HRESULT hRes; /* Result for windows functions */
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GST_DEBUG ("initializing directsoundsrc\n");
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GST_DEBUG_OBJECT (asrc, "opening directsoundsrc");
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dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
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@ -342,7 +321,7 @@ gst_directsound_src_close (GstAudioSrc * asrc)
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GstDirectSoundSrc *dsoundsrc;
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HRESULT hRes; /* Result for windows functions */
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GST_DEBUG ("initializing directsoundsrc\n");
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GST_DEBUG_OBJECT (asrc, "closing directsoundsrc");
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dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
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@ -356,49 +335,37 @@ gst_directsound_src_close (GstAudioSrc * asrc)
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}
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static gboolean
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gst_directsound_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
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gst_directsound_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
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{
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GstDirectSoundSrc *dsoundsrc;
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WAVEFORMATEX wfx; /* Wave format structure */
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HRESULT hRes; /* Result for windows functions */
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DSCBUFFERDESC descSecondary; /* Capturebuffer decsiption */
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DSCBUFFERDESC descSecondary; /* Capturebuffer description */
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dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
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GST_DEBUG ("initializing directsoundsrc\n");
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GST_DEBUG_OBJECT (asrc, "preparing directsoundsrc");
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/* Define buffer */
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memset (&wfx, 0, sizeof (WAVEFORMATEX));
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wfx.wFormatTag = WAVE_FORMAT_PCM; /* should be WAVE_FORMAT_PCM */
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wfx.nChannels = spec->channels;
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wfx.nSamplesPerSec = spec->rate; /* 8000|11025|22050|44100 */
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wfx.wBitsPerSample = spec->width; // 8|16;
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wfx.nBlockAlign = wfx.nChannels * (wfx.wBitsPerSample / 8);
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wfx.wFormatTag = WAVE_FORMAT_PCM;
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wfx.nChannels = GST_AUDIO_INFO_CHANNELS (&spec->info);
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wfx.nSamplesPerSec = GST_AUDIO_INFO_RATE (&spec->info);
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wfx.wBitsPerSample = GST_AUDIO_INFO_BPF (&spec->info) * 8 / wfx.nChannels;
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wfx.nBlockAlign = GST_AUDIO_INFO_BPF (&spec->info);
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wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
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wfx.cbSize = 0; /* This size is allways for PCM-format */
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/* Ignored for WAVE_FORMAT_PCM. */
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wfx.cbSize = 0;
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/* 1 or 2 Channels etc...
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FIXME: Never really tested. Is this ok?
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*/
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if (spec->width == 16 && spec->channels == 1) {
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spec->format = GST_S16_LE;
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} else if (spec->width == 16 && spec->channels == 2) {
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spec->format = GST_U16_LE;
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} else if (spec->width == 8 && spec->channels == 1) {
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spec->format = GST_S8;
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} else if (spec->width == 8 && spec->channels == 2) {
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spec->format = GST_U8;
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}
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if (wfx.wBitsPerSample != 16 && wfx.wBitsPerSample != 8)
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goto dodgy_width;
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/* Set the buffer size to two seconds.
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This should never reached.
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*/
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dsoundsrc->buffer_size = wfx.nAvgBytesPerSec * 2;
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//notifysize * 16; //spec->width; /*original 16*/
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GST_DEBUG ("Buffer size: %d", dsoundsrc->buffer_size);
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GST_DEBUG_OBJECT (asrc, "Buffer size: %d", dsoundsrc->buffer_size);
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/* Init secondary buffer desciption */
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memset (&descSecondary, 0, sizeof (DSCBUFFERDESC));
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@ -413,19 +380,15 @@ gst_directsound_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
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/* Create buffer */
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hRes = IDirectSoundCapture_CreateCaptureBuffer (dsoundsrc->pDSC,
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&descSecondary, &dsoundsrc->pDSBSecondary, NULL);
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if (hRes != DS_OK) {
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if (hRes != DS_OK)
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goto capture_buffer;
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}
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spec->channels = wfx.nChannels;
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spec->rate = wfx.nSamplesPerSec;
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spec->bytes_per_sample = (spec->width / 8) * spec->channels;
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dsoundsrc->bytes_per_sample = spec->bytes_per_sample;
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dsoundsrc->bytes_per_sample = GST_AUDIO_INFO_BPF (&spec->info);
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GST_DEBUG ("latency time: %" G_GUINT64_FORMAT " - buffer time: %"
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G_GUINT64_FORMAT, spec->latency_time, spec->buffer_time);
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/* Buffer-time should be allways more than 2*latency */
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/* Buffer-time should be always more than 2*latency */
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if (spec->buffer_time < spec->latency_time * 2) {
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spec->buffer_time = spec->latency_time * 2;
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GST_WARNING ("buffer-time was less than latency");
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@ -438,7 +401,6 @@ gst_directsound_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
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dsoundsrc->latency_size = (gint) wfx.nAvgBytesPerSec *
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dsoundsrc->latency_time / 1000000.0;
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spec->segsize = (guint) (((double) spec->buffer_time / 1000000.0) *
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wfx.nAvgBytesPerSec);
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@ -446,27 +408,23 @@ gst_directsound_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
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if (spec->segsize < 1)
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spec->segsize = 1;
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spec->segtotal = spec->width * (wfx.nAvgBytesPerSec / spec->segsize);
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spec->segtotal = GST_AUDIO_INFO_BPF (&spec->info) * 8 *
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(wfx.nAvgBytesPerSec / spec->segsize);
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GST_DEBUG ("bytes/sec: %lu, buffer size: %d, segsize: %d, segtotal: %d",
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wfx.nAvgBytesPerSec,
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dsoundsrc->buffer_size, spec->segsize, spec->segtotal);
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GST_DEBUG_OBJECT (asrc,
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"bytes/sec: %lu, buffer size: %d, segsize: %d, segtotal: %d",
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wfx.nAvgBytesPerSec, dsoundsrc->buffer_size, spec->segsize,
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spec->segtotal);
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spec->silence_sample[0] = 0;
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spec->silence_sample[1] = 0;
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spec->silence_sample[2] = 0;
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spec->silence_sample[3] = 0;
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if (spec->width != 16 && spec->width != 8)
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goto dodgy_width;
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/* Not readed anything yet */
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/* Not read anything yet */
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dsoundsrc->current_circular_offset = 0;
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GST_DEBUG ("GstRingBufferSpec->channels: %d, GstRingBufferSpec->rate: %d, \
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GstRingBufferSpec->bytes_per_sample: %d\n\
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WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d, \
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WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld\n", spec->channels, spec->rate, spec->bytes_per_sample, wfx.nSamplesPerSec, wfx.wBitsPerSample, wfx.nBlockAlign, wfx.nAvgBytesPerSec);
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GST_DEBUG_OBJECT (asrc, "channels: %d, rate: %d, bytes_per_sample: %d"
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" WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d,"
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" WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld",
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GST_AUDIO_INFO_CHANNELS (&spec->info), GST_AUDIO_INFO_RATE (&spec->info),
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GST_AUDIO_INFO_BPF (&spec->info), wfx.nSamplesPerSec, wfx.wBitsPerSample,
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wfx.nBlockAlign, wfx.nAvgBytesPerSec);
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return TRUE;
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@ -479,21 +437,18 @@ capture_buffer:
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dodgy_width:
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{
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GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
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("Unexpected width %d", spec->width), (NULL));
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("Unexpected width %d", wfx.wBitsPerSample), (NULL));
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return FALSE;
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}
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}
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static gboolean
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gst_directsound_src_unprepare (GstAudioSrc * asrc)
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{
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GstDirectSoundSrc *dsoundsrc;
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HRESULT hRes;
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HRESULT hRes; /* Result for windows functions */
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/* Resets */
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GST_DEBUG ("unpreparing directsoundsrc");
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GST_DEBUG_OBJECT (asrc, "unpreparing directsoundsrc");
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dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
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@ -504,7 +459,6 @@ gst_directsound_src_unprepare (GstAudioSrc * asrc)
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hRes = IDirectSoundCaptureBuffer_Release (dsoundsrc->pDSBSecondary);
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return TRUE;
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}
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/*
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@ -525,7 +479,7 @@ gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length)
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DWORD dwStatus = 0;
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||||
|
||||
GST_DEBUG ("reading directsoundsrc\n");
|
||||
GST_DEBUG_OBJECT (asrc, "reading directsoundsrc");
|
||||
|
||||
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
|
||||
|
||||
|
|
@ -535,12 +489,12 @@ gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length)
|
|||
hRes = IDirectSoundCaptureBuffer_GetStatus (dsoundsrc->pDSBSecondary,
|
||||
&dwStatus);
|
||||
|
||||
/* Starting capturing if not allready */
|
||||
/* Starting capturing if not already */
|
||||
if (!(dwStatus & DSCBSTATUS_CAPTURING)) {
|
||||
hRes = IDirectSoundCaptureBuffer_Start (dsoundsrc->pDSBSecondary,
|
||||
DSCBSTART_LOOPING);
|
||||
// Sleep (dsoundsrc->latency_time/1000);
|
||||
GST_DEBUG ("capture started");
|
||||
GST_DEBUG_OBJECT (asrc, "capture started");
|
||||
}
|
||||
// calculate_buffersize:
|
||||
while (length > dwBufferSize) {
|
||||
|
|
@ -558,8 +512,6 @@ gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length)
|
|||
dwBufferSize =
|
||||
dwCurrentCaptureCursor - dsoundsrc->current_circular_offset;
|
||||
}
|
||||
|
||||
|
||||
} // while (...
|
||||
|
||||
/* Lock the buffer */
|
||||
|
|
@ -588,7 +540,6 @@ gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length)
|
|||
|
||||
/* return length (readed data size in bytes) */
|
||||
return length;
|
||||
|
||||
}
|
||||
|
||||
static guint
|
||||
|
|
@ -600,7 +551,7 @@ gst_directsound_src_delay (GstAudioSrc * asrc)
|
|||
DWORD dwBytesInQueue = 0;
|
||||
gint nNbSamplesInQueue = 0;
|
||||
|
||||
GST_DEBUG ("Delay\n");
|
||||
GST_DEBUG_OBJECT (asrc, "Delay");
|
||||
|
||||
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
|
||||
|
||||
|
|
@ -632,7 +583,7 @@ gst_directsound_src_reset (GstAudioSrc * asrc)
|
|||
LPVOID pLockedBuffer = NULL;
|
||||
DWORD dwSizeBuffer = 0;
|
||||
|
||||
GST_DEBUG ("reset directsoundsrc\n");
|
||||
GST_DEBUG_OBJECT (asrc, "reset directsoundsrc");
|
||||
|
||||
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue