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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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95d4239819
Original commit message from CVS: * ext/pulse/pulsemixer.c: (gst_pulsemixer_class_init), (gst_pulsemixer_set_property), (gst_pulsemixer_get_property): * ext/pulse/pulsemixerctrl.c: (gst_pulsemixer_ctrl_subscribe_cb), (gst_pulsemixer_ctrl_open), (gst_pulsemixer_ctrl_timeout_event), (gst_pulsemixer_ctrl_set_volume): * ext/pulse/pulsemixertrack.c: (gst_pulsemixer_track_new): * ext/pulse/pulseprobe.c: (gst_pulseprobe_open): * ext/pulse/pulsesink.c: (gst_pulsesink_class_init), (gst_pulsesink_init), (gst_pulsesink_open), (gst_pulsesink_prepare), (gst_pulsesink_write), (gst_pulsesink_delay), (gst_pulsesink_reset): * ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init), (gst_pulsesrc_init): Use GST_BOILERPLATE everywhere and fix coding style at some places. Fix a locking issue in pulsesink's prepare function. * ext/pulse/pulseutil.c: (gst_pulse_channel_map_to_gst): Check if the created channel layout is valid for GStreamer.
921 lines
25 KiB
C
921 lines
25 KiB
C
/*
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* GStreamer pulseaudio plugin
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*
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* Copyright (c) 2004-2008 Lennart Poettering
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*
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* gst-pulse is free software; you can redistribute it and/or modify
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* it under the terms of the GNU Lesser General Public License as
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* published by the Free Software Foundation; either version 2.1 of the
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* License, or (at your option) any later version.
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*
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* gst-pulse is distributed in the hope that it will be useful, but
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* 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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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with gst-pulse; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
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* USA.
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*/
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/**
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* SECTION:element-pulsesink
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* @short_description: Output audio to a PulseAudio sound server
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* @see_also: pulsesrc, pulsemixer
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*
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* <refsect2>
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* <para>
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* This element outputs audio to a PulseAudio sound server.
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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 -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! pulsesink
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* </programlisting>
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* Play an Ogg/Vorbis file.
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* </para>
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* <para>
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* <programlisting>
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* gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.4 ! pulsesink
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* </programlisting>
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* Play a 440Hz sine wave.
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* </para>
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* </refsect2>
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*
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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 <string.h>
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#include <stdio.h>
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#include <gst/base/gstbasesink.h>
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#include <gst/gsttaglist.h>
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#include "pulsesink.h"
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#include "pulseutil.h"
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GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
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#define GST_CAT_DEFAULT pulse_debug
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enum
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{
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PROP_SERVER = 1,
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PROP_DEVICE,
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PROP_DEVICE_NAME,
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PROP_VOLUME
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};
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static void gst_pulsesink_destroy_stream (GstPulseSink * pulsesink);
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static void gst_pulsesink_destroy_context (GstPulseSink * pulsesink);
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static void gst_pulsesink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_pulsesink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_pulsesink_finalize (GObject * object);
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static void gst_pulsesink_dispose (GObject * object);
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static gboolean gst_pulsesink_open (GstAudioSink * asink);
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static gboolean gst_pulsesink_close (GstAudioSink * asink);
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static gboolean gst_pulsesink_prepare (GstAudioSink * asink,
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GstRingBufferSpec * spec);
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static gboolean gst_pulsesink_unprepare (GstAudioSink * asink);
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static guint gst_pulsesink_write (GstAudioSink * asink, gpointer data,
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guint length);
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static guint gst_pulsesink_delay (GstAudioSink * asink);
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static void gst_pulsesink_reset (GstAudioSink * asink);
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static gboolean gst_pulsesink_event (GstBaseSink * sink, GstEvent * event);
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static GstStateChangeReturn gst_pulsesink_change_state (GstElement *
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element, GstStateChange transition);
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static void gst_pulsesink_init_interfaces (GType type);
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#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
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# define ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
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#else
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# define ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
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#endif
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GST_IMPLEMENT_PULSEPROBE_METHODS (GstPulseSink, gst_pulsesink);
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GST_BOILERPLATE_FULL (GstPulseSink, gst_pulsesink, GstAudioSink,
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GST_TYPE_AUDIO_SINK, gst_pulsesink_init_interfaces);
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static gboolean
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gst_pulsesink_interface_supported (GstImplementsInterface *
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iface, GType interface_type)
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{
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GstPulseSink *this = GST_PULSESINK (iface);
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if (interface_type == GST_TYPE_PROPERTY_PROBE && this->probe)
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return TRUE;
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return FALSE;
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}
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static void
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gst_pulsesink_implements_interface_init (GstImplementsInterfaceClass * klass)
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{
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klass->supported = gst_pulsesink_interface_supported;
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}
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static void
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gst_pulsesink_init_interfaces (GType type)
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{
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static const GInterfaceInfo implements_iface_info = {
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(GInterfaceInitFunc) gst_pulsesink_implements_interface_init,
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NULL,
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NULL,
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};
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static const GInterfaceInfo probe_iface_info = {
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(GInterfaceInitFunc) gst_pulsesink_property_probe_interface_init,
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NULL,
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NULL,
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};
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g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
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&implements_iface_info);
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g_type_add_interface_static (type, GST_TYPE_PROPERTY_PROBE,
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&probe_iface_info);
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}
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static void
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gst_pulsesink_base_init (gpointer g_class)
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{
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static GstStaticPadTemplate pad_template = 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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"endianness = (int) { " ENDIANNESS " }, "
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"signed = (boolean) TRUE, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, 16 ];"
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"audio/x-raw-float, "
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"endianness = (int) { " ENDIANNESS " }, "
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"width = (int) 32, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, 16 ];"
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"audio/x-raw-int, "
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"endianness = (int) { " ENDIANNESS " }, "
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"signed = (boolean) TRUE, "
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"width = (int) 32, "
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"depth = (int) 32, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, 16 ];"
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"audio/x-raw-int, "
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"signed = (boolean) FALSE, "
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"width = (int) 8, "
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"depth = (int) 8, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, 16 ];"
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"audio/x-alaw, "
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"rate = (int) [ 1, MAX], "
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"channels = (int) [ 1, 16 ];"
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"audio/x-mulaw, "
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"rate = (int) [ 1, MAX], " "channels = (int) [ 1, 16 ]")
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);
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_set_details_simple (element_class,
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"PulseAudio Audio Sink",
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"Sink/Audio", "Plays audio to a PulseAudio server", "Lennart Poettering");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&pad_template));
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}
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static void
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gst_pulsesink_class_init (GstPulseSinkClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
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GstAudioSinkClass *gstaudiosink_class = GST_AUDIO_SINK_CLASS (klass);
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gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_pulsesink_dispose);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_pulsesink_finalize);
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gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_pulsesink_set_property);
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gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_pulsesink_get_property);
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_pulsesink_change_state);
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gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_pulsesink_event);
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gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_pulsesink_open);
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gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_pulsesink_close);
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gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_pulsesink_prepare);
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gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_pulsesink_unprepare);
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gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_pulsesink_write);
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gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_pulsesink_delay);
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gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_pulsesink_reset);
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/* Overwrite GObject fields */
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g_object_class_install_property (gobject_class,
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PROP_SERVER,
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g_param_spec_string ("server", "Server",
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"The PulseAudio server to connect to", NULL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_DEVICE,
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g_param_spec_string ("device", "Sink",
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"The PulseAudio sink device to connect to", NULL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_DEVICE_NAME,
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g_param_spec_string ("device-name", "Device name",
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"Human-readable name of the sound device", NULL,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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#if 0
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g_object_class_install_property (gobject_class,
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PROP_VOLUME,
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g_param_spec_double ("volume", "Volume",
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"Volume of this stream", 0.0, 10.0, 1.0,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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#endif
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}
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static void
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gst_pulsesink_init (GstPulseSink * pulsesink, GstPulseSinkClass * klass)
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{
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int e;
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pulsesink->server = pulsesink->device = pulsesink->stream_name = NULL;
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pulsesink->context = NULL;
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pulsesink->stream = NULL;
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pulsesink->mainloop = pa_threaded_mainloop_new ();
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g_assert (pulsesink->mainloop);
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e = pa_threaded_mainloop_start (pulsesink->mainloop);
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g_assert (e == 0);
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pulsesink->probe = gst_pulseprobe_new (G_OBJECT (pulsesink), G_OBJECT_GET_CLASS (pulsesink), PROP_DEVICE, pulsesink->device, TRUE, FALSE); /* TRUE for sinks, FALSE for sources */
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pulsesink->mixer = NULL;
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}
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static void
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gst_pulsesink_destroy_stream (GstPulseSink * pulsesink)
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{
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if (pulsesink->stream) {
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pa_stream_disconnect (pulsesink->stream);
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pa_stream_unref (pulsesink->stream);
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pulsesink->stream = NULL;
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}
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g_free (pulsesink->stream_name);
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pulsesink->stream_name = NULL;
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}
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static void
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gst_pulsesink_destroy_context (GstPulseSink * pulsesink)
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{
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gst_pulsesink_destroy_stream (pulsesink);
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if (pulsesink->context) {
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pa_context_disconnect (pulsesink->context);
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pa_context_unref (pulsesink->context);
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pulsesink->context = NULL;
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}
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}
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static void
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gst_pulsesink_finalize (GObject * object)
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{
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GstPulseSink *pulsesink = GST_PULSESINK (object);
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pa_threaded_mainloop_stop (pulsesink->mainloop);
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gst_pulsesink_destroy_context (pulsesink);
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g_free (pulsesink->server);
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g_free (pulsesink->device);
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g_free (pulsesink->stream_name);
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pa_threaded_mainloop_free (pulsesink->mainloop);
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if (pulsesink->probe) {
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gst_pulseprobe_free (pulsesink->probe);
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pulsesink->probe = NULL;
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}
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if (pulsesink->mixer) {
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gst_pulsemixer_ctrl_free (pulsesink->mixer);
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pulsesink->mixer = NULL;
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}
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_pulsesink_dispose (GObject * object)
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{
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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#if 0
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static void
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gst_pulsesink_set_volume (GstPulseSink * pulsesink, gdouble volume)
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{
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if (pulsesink->mixer && pulsesink->mixer->track->num_channels > 0) {
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gint *volumes = g_new0 (gint, pulsesink->mixer->track->num_channels);
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gint i;
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g_print ("setting volume for real\n");
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for (i = 0; i < pulsesink->mixer->track->num_channels; i++)
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volumes[i] = volume;
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gst_pulsemixer_ctrl_set_volume (pulsesink->mixer, pulsesink->mixer->track,
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volumes);
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pulsesink->volume = volume;
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g_free (volumes);
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} else {
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pulsesink->volume = volume;
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}
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}
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static gdouble
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gst_pulsesink_get_volume (GstPulseSink * pulsesink)
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{
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if (pulsesink->mixer && pulsesink->mixer->track->num_channels > 0) {
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gint *volumes = g_new0 (gint, pulsesink->mixer->track->num_channels);
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gdouble volume = 0.0;
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gint i;
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gst_pulsemixer_ctrl_get_volume (pulsesink->mixer, pulsesink->mixer->track,
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volumes);
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for (i = 0; i < pulsesink->mixer->track->num_channels; i++)
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volume += volumes[i];
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volume /= pulsesink->mixer->track->num_channels;
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pulsesink->volume = volume;
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g_free (volumes);
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g_print ("real volume: %lf\n", volume);
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return volume;
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} else {
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return pulsesink->volume;
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}
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}
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#endif
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static void
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gst_pulsesink_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec)
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{
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GstPulseSink *pulsesink = GST_PULSESINK (object);
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switch (prop_id) {
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case PROP_SERVER:
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g_free (pulsesink->server);
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pulsesink->server = g_value_dup_string (value);
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if (pulsesink->probe)
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gst_pulseprobe_set_server (pulsesink->probe, pulsesink->server);
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break;
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case PROP_DEVICE:
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g_free (pulsesink->device);
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pulsesink->device = g_value_dup_string (value);
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break;
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#if 0
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case PROP_VOLUME:
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gst_pulsesink_set_volume (pulsesink, g_value_get_double (value));
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break;
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#endif
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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_pulsesink_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec)
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{
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GstPulseSink *pulsesink = GST_PULSESINK (object);
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switch (prop_id) {
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case PROP_SERVER:
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g_value_set_string (value, pulsesink->server);
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break;
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case PROP_DEVICE:
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g_value_set_string (value, pulsesink->device);
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break;
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case PROP_DEVICE_NAME:
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if (pulsesink->mixer)
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g_value_set_string (value, pulsesink->mixer->description);
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else
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g_value_set_string (value, NULL);
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break;
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#if 0
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case PROP_VOLUME:
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g_value_set_double (value, gst_pulsesink_get_volume (pulsesink));
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break;
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#endif
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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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static void
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gst_pulsesink_context_state_cb (pa_context * c, void *userdata)
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{
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GstPulseSink *pulsesink = GST_PULSESINK (userdata);
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switch (pa_context_get_state (c)) {
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case PA_CONTEXT_READY:
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case PA_CONTEXT_TERMINATED:
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case PA_CONTEXT_FAILED:
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pa_threaded_mainloop_signal (pulsesink->mainloop, 0);
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break;
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case PA_CONTEXT_UNCONNECTED:
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case PA_CONTEXT_CONNECTING:
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case PA_CONTEXT_AUTHORIZING:
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case PA_CONTEXT_SETTING_NAME:
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|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_stream_state_cb (pa_stream * s, void *userdata)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
|
|
|
|
switch (pa_stream_get_state (s)) {
|
|
|
|
case PA_STREAM_READY:
|
|
case PA_STREAM_FAILED:
|
|
case PA_STREAM_TERMINATED:
|
|
pa_threaded_mainloop_signal (pulsesink->mainloop, 0);
|
|
break;
|
|
|
|
case PA_STREAM_UNCONNECTED:
|
|
case PA_STREAM_CREATING:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_stream_request_cb (pa_stream * s, size_t length, void *userdata)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
|
|
|
|
pa_threaded_mainloop_signal (pulsesink->mainloop, 0);
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_stream_latency_update_cb (pa_stream * s, void *userdata)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
|
|
|
|
pa_threaded_mainloop_signal (pulsesink->mainloop, 0);
|
|
}
|
|
|
|
static gboolean
|
|
gst_pulsesink_open (GstAudioSink * asink)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (asink);
|
|
gchar *name = gst_pulse_client_name ();
|
|
|
|
pa_threaded_mainloop_lock (pulsesink->mainloop);
|
|
|
|
if (!(pulsesink->context =
|
|
pa_context_new (pa_threaded_mainloop_get_api (pulsesink->mainloop),
|
|
name))) {
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
|
|
("Failed to create context"), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
pa_context_set_state_callback (pulsesink->context,
|
|
gst_pulsesink_context_state_cb, pulsesink);
|
|
|
|
if (pa_context_connect (pulsesink->context, pulsesink->server, 0, NULL) < 0) {
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED, ("Failed to connect: %s",
|
|
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
/* Wait until the context is ready */
|
|
pa_threaded_mainloop_wait (pulsesink->mainloop);
|
|
|
|
if (pa_context_get_state (pulsesink->context) != PA_CONTEXT_READY) {
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED, ("Failed to connect: %s",
|
|
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
pa_threaded_mainloop_unlock (pulsesink->mainloop);
|
|
g_free (name);
|
|
return TRUE;
|
|
|
|
unlock_and_fail:
|
|
|
|
pa_threaded_mainloop_unlock (pulsesink->mainloop);
|
|
g_free (name);
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_pulsesink_close (GstAudioSink * asink)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (asink);
|
|
|
|
pa_threaded_mainloop_lock (pulsesink->mainloop);
|
|
gst_pulsesink_destroy_context (pulsesink);
|
|
pa_threaded_mainloop_unlock (pulsesink->mainloop);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_pulsesink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
|
|
{
|
|
pa_buffer_attr buf_attr;
|
|
pa_channel_map channel_map;
|
|
GstPulseSink *pulsesink = GST_PULSESINK (asink);
|
|
|
|
if (!gst_pulse_fill_sample_spec (spec, &pulsesink->sample_spec)) {
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, SETTINGS,
|
|
("Invalid sample specification."), (NULL));
|
|
goto fail;
|
|
}
|
|
|
|
pa_threaded_mainloop_lock (pulsesink->mainloop);
|
|
|
|
if (!pulsesink->context
|
|
|| pa_context_get_state (pulsesink->context) != PA_CONTEXT_READY) {
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED, ("Bad context state: %s",
|
|
pulsesink->
|
|
context ? pa_strerror (pa_context_errno (pulsesink->context)) :
|
|
NULL), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
if (!(pulsesink->stream = pa_stream_new (pulsesink->context,
|
|
pulsesink->
|
|
stream_name ? pulsesink->stream_name : "Playback Stream",
|
|
&pulsesink->sample_spec,
|
|
gst_pulse_gst_to_channel_map (&channel_map, spec)))) {
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
|
|
("Failed to create stream: %s",
|
|
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
pa_stream_set_state_callback (pulsesink->stream,
|
|
gst_pulsesink_stream_state_cb, pulsesink);
|
|
pa_stream_set_write_callback (pulsesink->stream,
|
|
gst_pulsesink_stream_request_cb, pulsesink);
|
|
pa_stream_set_latency_update_callback (pulsesink->stream,
|
|
gst_pulsesink_stream_latency_update_cb, pulsesink);
|
|
|
|
memset (&buf_attr, 0, sizeof (buf_attr));
|
|
buf_attr.tlength = spec->segtotal * spec->segsize;
|
|
buf_attr.maxlength = buf_attr.tlength * 2;
|
|
buf_attr.prebuf = buf_attr.tlength - spec->segsize;
|
|
buf_attr.minreq = spec->segsize;
|
|
|
|
if (pa_stream_connect_playback (pulsesink->stream, pulsesink->device,
|
|
&buf_attr,
|
|
PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
|
|
PA_STREAM_NOT_MONOTONOUS, NULL, NULL) < 0) {
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
|
|
("Failed to connect stream: %s",
|
|
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
/* Wait until the stream is ready */
|
|
pa_threaded_mainloop_wait (pulsesink->mainloop);
|
|
|
|
if (pa_stream_get_state (pulsesink->stream) != PA_STREAM_READY) {
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
|
|
("Failed to connect stream: %s",
|
|
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
pa_threaded_mainloop_unlock (pulsesink->mainloop);
|
|
|
|
#if 0
|
|
gst_pulsesink_set_volume (pulsesink, pulsesink->volume);
|
|
#endif
|
|
|
|
return TRUE;
|
|
|
|
unlock_and_fail:
|
|
pa_threaded_mainloop_unlock (pulsesink->mainloop);
|
|
|
|
fail:
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_pulsesink_unprepare (GstAudioSink * asink)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (asink);
|
|
|
|
pa_threaded_mainloop_lock (pulsesink->mainloop);
|
|
gst_pulsesink_destroy_stream (pulsesink);
|
|
pa_threaded_mainloop_unlock (pulsesink->mainloop);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
#define CHECK_DEAD_GOTO(pulsesink, label) \
|
|
if (!(pulsesink)->context || pa_context_get_state((pulsesink)->context) != PA_CONTEXT_READY || \
|
|
!(pulsesink)->stream || pa_stream_get_state((pulsesink)->stream) != PA_STREAM_READY) { \
|
|
GST_ELEMENT_ERROR((pulsesink), RESOURCE, FAILED, ("Disconnected: %s", (pulsesink)->context ? pa_strerror(pa_context_errno((pulsesink)->context)) : NULL), (NULL)); \
|
|
goto label; \
|
|
}
|
|
|
|
static guint
|
|
gst_pulsesink_write (GstAudioSink * asink, gpointer data, guint length)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (asink);
|
|
size_t sum = 0;
|
|
|
|
pa_threaded_mainloop_lock (pulsesink->mainloop);
|
|
|
|
while (length > 0) {
|
|
size_t l;
|
|
|
|
for (;;) {
|
|
CHECK_DEAD_GOTO (pulsesink, unlock_and_fail);
|
|
|
|
if ((l = pa_stream_writable_size (pulsesink->stream)) == (size_t) - 1) {
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
|
|
("pa_stream_writable_size() failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
if (l > 0)
|
|
break;
|
|
|
|
pa_threaded_mainloop_wait (pulsesink->mainloop);
|
|
}
|
|
|
|
if (l > length)
|
|
l = length;
|
|
|
|
if (pa_stream_write (pulsesink->stream, data, l, NULL, 0,
|
|
PA_SEEK_RELATIVE) < 0) {
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
|
|
("pa_stream_write() failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
data = (guint8 *) data + l;
|
|
length -= l;
|
|
|
|
sum += l;
|
|
}
|
|
|
|
pa_threaded_mainloop_unlock (pulsesink->mainloop);
|
|
|
|
return sum;
|
|
|
|
unlock_and_fail:
|
|
|
|
pa_threaded_mainloop_unlock (pulsesink->mainloop);
|
|
return 0;
|
|
}
|
|
|
|
static guint
|
|
gst_pulsesink_delay (GstAudioSink * asink)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (asink);
|
|
pa_usec_t t;
|
|
|
|
pa_threaded_mainloop_lock (pulsesink->mainloop);
|
|
|
|
for (;;) {
|
|
CHECK_DEAD_GOTO (pulsesink, unlock_and_fail);
|
|
|
|
if (pa_stream_get_latency (pulsesink->stream, &t, NULL) >= 0)
|
|
break;
|
|
|
|
if (pa_context_errno (pulsesink->context) != PA_ERR_NODATA) {
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
|
|
("pa_stream_get_latency() failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
pa_threaded_mainloop_wait (pulsesink->mainloop);
|
|
}
|
|
|
|
pa_threaded_mainloop_unlock (pulsesink->mainloop);
|
|
|
|
return gst_util_uint64_scale_int (t, pulsesink->sample_spec.rate, 1000000LL);
|
|
|
|
unlock_and_fail:
|
|
|
|
pa_threaded_mainloop_unlock (pulsesink->mainloop);
|
|
return 0;
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_success_cb (pa_stream * s, int success, void *userdata)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
|
|
|
|
pulsesink->operation_success = success;
|
|
pa_threaded_mainloop_signal (pulsesink->mainloop, 0);
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_reset (GstAudioSink * asink)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (asink);
|
|
pa_operation *o = NULL;
|
|
|
|
pa_threaded_mainloop_lock (pulsesink->mainloop);
|
|
|
|
CHECK_DEAD_GOTO (pulsesink, unlock_and_fail);
|
|
|
|
if (!(o =
|
|
pa_stream_flush (pulsesink->stream, gst_pulsesink_success_cb,
|
|
pulsesink))) {
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
|
|
("pa_stream_flush() failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
pulsesink->operation_success = 0;
|
|
while (pa_operation_get_state (o) != PA_OPERATION_DONE) {
|
|
CHECK_DEAD_GOTO (pulsesink, unlock_and_fail);
|
|
|
|
pa_threaded_mainloop_wait (pulsesink->mainloop);
|
|
}
|
|
|
|
if (!pulsesink->operation_success) {
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED, ("Flush failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
unlock_and_fail:
|
|
|
|
if (o) {
|
|
pa_operation_cancel (o);
|
|
pa_operation_unref (o);
|
|
}
|
|
|
|
pa_threaded_mainloop_unlock (pulsesink->mainloop);
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_change_title (GstPulseSink * pulsesink, const gchar * t)
|
|
{
|
|
pa_operation *o = NULL;
|
|
|
|
pa_threaded_mainloop_lock (pulsesink->mainloop);
|
|
|
|
g_free (pulsesink->stream_name);
|
|
pulsesink->stream_name = g_strdup (t);
|
|
|
|
if (!(pulsesink)->context
|
|
|| pa_context_get_state ((pulsesink)->context) != PA_CONTEXT_READY
|
|
|| !(pulsesink)->stream
|
|
|| pa_stream_get_state ((pulsesink)->stream) != PA_STREAM_READY) {
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
if (!(o =
|
|
pa_stream_set_name (pulsesink->stream, pulsesink->stream_name, NULL,
|
|
pulsesink))) {
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
|
|
("pa_stream_set_name() failed: %s",
|
|
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
/* We're not interested if this operation failed or not */
|
|
|
|
unlock_and_fail:
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (pulsesink->mainloop);
|
|
}
|
|
|
|
static gboolean
|
|
gst_pulsesink_event (GstBaseSink * sink, GstEvent * event)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (sink);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_TAG:{
|
|
gchar *title = NULL, *artist = NULL, *location = NULL, *description =
|
|
NULL, *t = NULL, *buf = NULL;
|
|
GstTagList *l;
|
|
|
|
gst_event_parse_tag (event, &l);
|
|
|
|
gst_tag_list_get_string (l, GST_TAG_TITLE, &title);
|
|
gst_tag_list_get_string (l, GST_TAG_ARTIST, &artist);
|
|
gst_tag_list_get_string (l, GST_TAG_LOCATION, &location);
|
|
gst_tag_list_get_string (l, GST_TAG_DESCRIPTION, &description);
|
|
|
|
if (title && artist)
|
|
t = buf =
|
|
g_strdup_printf ("'%s' by '%s'", g_strstrip (title),
|
|
g_strstrip (artist));
|
|
else if (title)
|
|
t = g_strstrip (title);
|
|
else if (description)
|
|
t = g_strstrip (description);
|
|
else if (location)
|
|
t = g_strstrip (location);
|
|
|
|
if (t)
|
|
gst_pulsesink_change_title (pulsesink, t);
|
|
|
|
g_free (title);
|
|
g_free (artist);
|
|
g_free (location);
|
|
g_free (description);
|
|
g_free (buf);
|
|
|
|
break;
|
|
}
|
|
default:
|
|
;
|
|
}
|
|
|
|
return GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_pulsesink_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstPulseSink *this = GST_PULSESINK (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
|
|
if (!this->mixer) {
|
|
this->mixer =
|
|
gst_pulsemixer_ctrl_new (G_OBJECT (this), this->server,
|
|
this->device, GST_PULSEMIXER_SINK);
|
|
}
|
|
break;
|
|
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
|
|
if (this->mixer) {
|
|
#if 0
|
|
this->volume = gst_pulsesink_get_volume (this);
|
|
#endif
|
|
gst_pulsemixer_ctrl_free (this->mixer);
|
|
this->mixer = NULL;
|
|
}
|
|
|
|
break;
|
|
|
|
default:
|
|
;
|
|
}
|
|
|
|
if (GST_ELEMENT_CLASS (parent_class)->change_state)
|
|
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
return GST_STATE_CHANGE_SUCCESS;
|
|
}
|