gstreamer/ext/hal/gsthalaudiosrc.c

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/* GStreamer
* (c) 2005 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* (c) 2005 Tim-Philipp Müller <tim centricular net>
* (c) 2006 Jürg Billeter <j@bitron.ch>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-halaudiosrc
*
* <refsect2>
* <para>
* HalAudioSrc allows access to input of sound devices by specifying the
* corresponding persistent Unique Device Id (UDI) from the Hardware Abstraction
* Layer (HAL) in the <link linkend="GstHalAudioSrc--udi">udi</link> property.
* It currently always embeds alsasrc as HAL doesn't support other sound systems
* yet.
* </para>
* <title>Examples</title>
* <para>
* To list the UDIs of all your ALSA input devices :
* <programlisting>
* hal-find-by-property --key alsa.type --string capture
* </programlisting>
* Here is a pipeline to test your sound input :
* <programlisting>
* gst-launch -v halaudiosrc udi=/org/freedesktop/Hal/devices/pci_8086_27d8_alsa_capture_0 ! autoaudiosink
* </programlisting>
* You should now hear yourself with a small delay if you have a microphone
* connected to the specified sound device.
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gsthalelements.h"
#include "gsthalaudiosrc.h"
static void gst_hal_audio_src_dispose (GObject * object);
static GstStateChangeReturn
gst_hal_audio_src_change_state (GstElement * element,
GstStateChange transition);
enum
{
PROP_0,
PROP_UDI
};
GST_BOILERPLATE (GstHalAudioSrc, gst_hal_audio_src, GstBin, GST_TYPE_BIN);
static void gst_hal_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_hal_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void
gst_hal_audio_src_base_init (gpointer klass)
{
GstElementClass *eklass = GST_ELEMENT_CLASS (klass);
better/unified long descriptions Original commit message from CVS: * ext/aalib/gstaasink.c: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttimeoverlay.c: * ext/cdio/gstcdiocddasrc.c: * ext/dv/gstdvdec.c: * ext/esd/esdmon.c: * ext/esd/esdsink.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init): * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init): * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init): * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init): * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libmng/gstmngdec.c: * ext/libmng/gstmngenc.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/mikmod/gstmikmod.c: * ext/raw1394/gstdv1394src.c: * ext/shout2/gstshout2.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_base_init): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_base_init): * gst/avi/gstavimux.c: (gst_avimux_base_init): * gst/cutter/gstcutter.c: * gst/debug/breakmydata.c: * gst/debug/efence.c: * gst/debug/gstnavigationtest.c: * gst/debug/negotiation.c: * gst/debug/progressreport.c: * gst/debug/testplugin.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstquark.c: * gst/effectv/gstrev.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/law/alaw-decode.c: (gst_alawdec_base_init): * gst/law/alaw-encode.c: (gst_alawenc_base_init): * gst/law/mulaw-decode.c: (gst_mulawdec_base_init): * gst/law/mulaw-encode.c: (gst_mulawenc_base_init): * gst/level/gstlevel.c: * gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init): * gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init): * gst/median/gstmedian.c: * gst/monoscope/gstmonoscope.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/oldcore/gstmd5sink.c: * gst/oldcore/gstmultifilesrc.c: * gst/oldcore/gstpipefilter.c: * gst/oldcore/gstshaper.c: * gst/oldcore/gststatistics.c: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtsp/gstrtpdec.c: * gst/smpte/gstsmpte.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstgamma.c: (gst_gamma_base_init): * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideotemplate.c: (gst_videotemplate_base_init): * gst/videomixer/videomixer.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init): better/unified long descriptions Fixed #336602 Some cleanups to auparse, don't send multiple newsegments.
2006-03-30 15:37:05 +00:00
GstElementDetails gst_hal_audio_src_details =
GST_ELEMENT_DETAILS ("HAL audio source",
"Source/Audio",
"Audio source for sound device access via HAL",
"Jürg Billeter <j@bitron.ch>");
GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS_ANY);
gst_element_class_add_pad_template (eklass,
gst_static_pad_template_get (&src_template));
gst_element_class_set_details (eklass, &gst_hal_audio_src_details);
}
static void
gst_hal_audio_src_class_init (GstHalAudioSrcClass * klass)
{
GObjectClass *oklass = G_OBJECT_CLASS (klass);
GstElementClass *eklass = GST_ELEMENT_CLASS (klass);
oklass->set_property = gst_hal_audio_src_set_property;
oklass->get_property = gst_hal_audio_src_get_property;
oklass->dispose = gst_hal_audio_src_dispose;
eklass->change_state = gst_hal_audio_src_change_state;
g_object_class_install_property (oklass, PROP_UDI,
g_param_spec_string ("udi",
"UDI", "Unique Device Id", NULL, G_PARAM_READWRITE));
}
/*
* Hack to make negotiation work.
*/
static void
gst_hal_audio_src_reset (GstHalAudioSrc * src)
{
GstPad *targetpad;
/* fakesrc */
if (src->kid) {
gst_element_set_state (src->kid, GST_STATE_NULL);
gst_bin_remove (GST_BIN (src), src->kid);
}
src->kid = gst_element_factory_make ("fakesrc", "testsrc");
gst_bin_add (GST_BIN (src), src->kid);
targetpad = gst_element_get_pad (src->kid, "src");
gst_ghost_pad_set_target (GST_GHOST_PAD (src->pad), targetpad);
gst_object_unref (targetpad);
}
static void
gst_hal_audio_src_init (GstHalAudioSrc * src, GstHalAudioSrcClass * g_class)
{
src->pad = gst_ghost_pad_new_no_target ("src", GST_PAD_SRC);
gst_element_add_pad (GST_ELEMENT (src), src->pad);
gst_hal_audio_src_reset (src);
}
static void
gst_hal_audio_src_dispose (GObject * object)
{
GstHalAudioSrc *src = GST_HAL_AUDIO_SRC (object);
if (src->udi) {
g_free (src->udi);
src->udi = NULL;
}
GST_CALL_PARENT (G_OBJECT_CLASS, dispose, (object));
}
static gboolean
do_toggle_element (GstHalAudioSrc * src)
{
GstPad *targetpad;
/* kill old element */
if (src->kid) {
GST_DEBUG_OBJECT (src, "Removing old kid");
gst_element_set_state (src->kid, GST_STATE_NULL);
gst_bin_remove (GST_BIN (src), src->kid);
src->kid = NULL;
}
GST_DEBUG_OBJECT (src, "Creating new kid");
if (!(src->kid = gst_hal_get_audio_src (src->udi))) {
GST_ELEMENT_ERROR (src, LIBRARY, SETTINGS, (NULL),
("Failed to render audio source from Hal"));
return FALSE;
}
gst_element_set_state (src->kid, GST_STATE (src));
gst_bin_add (GST_BIN (src), src->kid);
/* re-attach ghostpad */
GST_DEBUG_OBJECT (src, "Creating new ghostpad");
targetpad = gst_element_get_pad (src->kid, "src");
gst_ghost_pad_set_target (GST_GHOST_PAD (src->pad), targetpad);
gst_object_unref (targetpad);
GST_DEBUG_OBJECT (src, "done changing hal audio source");
return TRUE;
}
static void
gst_hal_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstHalAudioSrc *this = GST_HAL_AUDIO_SRC (object);
GST_OBJECT_LOCK (this);
switch (prop_id) {
case PROP_UDI:
if (this->udi)
g_free (this->udi);
this->udi = g_value_dup_string (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (this);
}
static void
gst_hal_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstHalAudioSrc *this = GST_HAL_AUDIO_SRC (object);
GST_OBJECT_LOCK (this);
switch (prop_id) {
case PROP_UDI:
g_value_set_string (value, this->udi);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (this);
}
static GstStateChangeReturn
gst_hal_audio_src_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstHalAudioSrc *src = GST_HAL_AUDIO_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!do_toggle_element (src))
return GST_STATE_CHANGE_FAILURE;
break;
default:
break;
}
ret = GST_CALL_PARENT_WITH_DEFAULT (GST_ELEMENT_CLASS, change_state,
(element, transition), GST_STATE_CHANGE_SUCCESS);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
gst_hal_audio_src_reset (src);
break;
default:
break;
}
return ret;
}