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27f2c9b255
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/dv/gstdvdemux.c: * ext/esd/esdmon.c: * ext/esd/esdsink.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/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: (gst_shout2send_init): * ext/shout2/gstshout2.h: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.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/gstavidemux.c: (gst_avi_demux_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/gstnavseek.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/gstshagadelic.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/id3demux/gstid3demux.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/gstaggregator.c: * gst/oldcore/gstfdsink.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/gstrtpilbcdepay.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/rtsp/gstrtspsrc.c: * gst/smpte/gstsmpte.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.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/wavparse/gstwavparse.c: (gst_wavparse_base_init), (gst_wavparse_class_init), (gst_wavparse_dispose), (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info), (gst_wavparse_peek_chunk), (gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init), (gst_wavparse_send_event), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data), (gst_wavparse_chain), (gst_wavparse_srcpad_event), (gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull), (gst_wavparse_change_state): * gst/wavparse/gstwavparse.h: * sys/oss/gstossmixerelement.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/osxaudio/gstosxaudioelement.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/sunaudio/gstsunaudiomixer.c: * sys/sunaudio/gstsunaudiosink.c: Define GstElementDetails as const and also static (when defined as global)
254 lines
7 KiB
C
254 lines
7 KiB
C
/* GStreamer
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* (c) 2005 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
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* (c) 2005 Tim-Philipp Müller <tim centricular net>
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* (c) 2006 Jürg Billeter <j@bitron.ch>
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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-halaudiosrc
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*
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* <refsect2>
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* <para>
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* HalAudioSrc allows access to input of sound devices by specifying the
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* corresponding persistent Unique Device Id (UDI) from the Hardware Abstraction
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* Layer (HAL) in the <link linkend="GstHalAudioSrc--udi">udi</link> property.
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* It currently always embeds alsasrc as HAL doesn't support other sound systems
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* yet.
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* </para>
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* <title>Examples</title>
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* <para>
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* To list the UDIs of all your ALSA input devices :
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* <programlisting>
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* hal-find-by-property --key alsa.type --string capture
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* </programlisting>
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* Here is a pipeline to test your sound input :
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* <programlisting>
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* gst-launch -v halaudiosrc udi=/org/freedesktop/Hal/devices/pci_8086_27d8_alsa_capture_0 ! autoaudiosink
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* </programlisting>
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* You should now hear yourself with a small delay if you have a microphone
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* connected to the specified sound device.
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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 "gsthalelements.h"
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#include "gsthalaudiosrc.h"
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static void gst_hal_audio_src_dispose (GObject * object);
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static GstStateChangeReturn
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gst_hal_audio_src_change_state (GstElement * element,
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GstStateChange transition);
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enum
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{
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PROP_0,
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PROP_UDI
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};
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GST_BOILERPLATE (GstHalAudioSrc, gst_hal_audio_src, GstBin, GST_TYPE_BIN);
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static void gst_hal_audio_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_hal_audio_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void
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gst_hal_audio_src_base_init (gpointer klass)
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{
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GstElementClass *eklass = GST_ELEMENT_CLASS (klass);
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static const GstElementDetails gst_hal_audio_src_details =
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GST_ELEMENT_DETAILS ("HAL audio source",
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"Source/Audio",
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"Audio source for sound device access via HAL",
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"Jürg Billeter <j@bitron.ch>");
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GstStaticPadTemplate src_template = 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_ANY);
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gst_element_class_add_pad_template (eklass,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_details (eklass, &gst_hal_audio_src_details);
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}
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static void
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gst_hal_audio_src_class_init (GstHalAudioSrcClass * klass)
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{
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GObjectClass *oklass = G_OBJECT_CLASS (klass);
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GstElementClass *eklass = GST_ELEMENT_CLASS (klass);
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oklass->set_property = gst_hal_audio_src_set_property;
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oklass->get_property = gst_hal_audio_src_get_property;
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oklass->dispose = gst_hal_audio_src_dispose;
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eklass->change_state = gst_hal_audio_src_change_state;
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g_object_class_install_property (oklass, PROP_UDI,
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g_param_spec_string ("udi",
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"UDI", "Unique Device Id", NULL, G_PARAM_READWRITE));
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}
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/*
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* Hack to make negotiation work.
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*/
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static void
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gst_hal_audio_src_reset (GstHalAudioSrc * src)
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{
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GstPad *targetpad;
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/* fakesrc */
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if (src->kid) {
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gst_element_set_state (src->kid, GST_STATE_NULL);
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gst_bin_remove (GST_BIN (src), src->kid);
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}
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src->kid = gst_element_factory_make ("fakesrc", "testsrc");
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gst_bin_add (GST_BIN (src), src->kid);
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targetpad = gst_element_get_pad (src->kid, "src");
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gst_ghost_pad_set_target (GST_GHOST_PAD (src->pad), targetpad);
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gst_object_unref (targetpad);
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}
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static void
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gst_hal_audio_src_init (GstHalAudioSrc * src, GstHalAudioSrcClass * g_class)
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{
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src->pad = gst_ghost_pad_new_no_target ("src", GST_PAD_SRC);
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gst_element_add_pad (GST_ELEMENT (src), src->pad);
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gst_hal_audio_src_reset (src);
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}
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static void
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gst_hal_audio_src_dispose (GObject * object)
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{
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GstHalAudioSrc *src = GST_HAL_AUDIO_SRC (object);
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if (src->udi) {
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g_free (src->udi);
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src->udi = NULL;
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}
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GST_CALL_PARENT (G_OBJECT_CLASS, dispose, (object));
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}
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static gboolean
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do_toggle_element (GstHalAudioSrc * src)
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{
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GstPad *targetpad;
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/* kill old element */
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if (src->kid) {
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GST_DEBUG_OBJECT (src, "Removing old kid");
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gst_element_set_state (src->kid, GST_STATE_NULL);
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gst_bin_remove (GST_BIN (src), src->kid);
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src->kid = NULL;
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}
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GST_DEBUG_OBJECT (src, "Creating new kid");
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if (!(src->kid = gst_hal_get_audio_src (src->udi))) {
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GST_ELEMENT_ERROR (src, LIBRARY, SETTINGS, (NULL),
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("Failed to render audio source from Hal"));
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return FALSE;
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}
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gst_element_set_state (src->kid, GST_STATE (src));
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gst_bin_add (GST_BIN (src), src->kid);
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/* re-attach ghostpad */
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GST_DEBUG_OBJECT (src, "Creating new ghostpad");
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targetpad = gst_element_get_pad (src->kid, "src");
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gst_ghost_pad_set_target (GST_GHOST_PAD (src->pad), targetpad);
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gst_object_unref (targetpad);
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GST_DEBUG_OBJECT (src, "done changing hal audio source");
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return TRUE;
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}
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static void
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gst_hal_audio_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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GstHalAudioSrc *this = GST_HAL_AUDIO_SRC (object);
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GST_OBJECT_LOCK (this);
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switch (prop_id) {
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case PROP_UDI:
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if (this->udi)
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g_free (this->udi);
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this->udi = g_value_dup_string (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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break;
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}
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GST_OBJECT_UNLOCK (this);
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}
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static void
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gst_hal_audio_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstHalAudioSrc *this = GST_HAL_AUDIO_SRC (object);
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GST_OBJECT_LOCK (this);
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switch (prop_id) {
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case PROP_UDI:
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g_value_set_string (value, this->udi);
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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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GST_OBJECT_UNLOCK (this);
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}
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static GstStateChangeReturn
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gst_hal_audio_src_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
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GstHalAudioSrc *src = GST_HAL_AUDIO_SRC (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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if (!do_toggle_element (src))
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return GST_STATE_CHANGE_FAILURE;
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break;
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default:
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break;
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}
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ret = GST_CALL_PARENT_WITH_DEFAULT (GST_ELEMENT_CLASS, change_state,
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(element, transition), GST_STATE_CHANGE_SUCCESS);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_NULL:
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gst_hal_audio_src_reset (src);
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break;
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default:
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break;
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}
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return ret;
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}
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