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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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0694be2a82
Use the acquired field of the ringbuffer in get_time to know when we are in an invalid state. We don't clear the rate flag when releasing the ringbuffer so this values is not usable. Avoids some error messages being posted because the pulseaudio connection is down.
2260 lines
63 KiB
C
2260 lines
63 KiB
C
/*-*- Mode: C; c-basic-offset: 2 -*-*/
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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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* (c) 2009 Wim Taymans
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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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* @see_also: pulsesrc, pulsemixer
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*
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* This element outputs audio to a
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* <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink>.
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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 filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! pulsesink
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* ]| Play an Ogg/Vorbis file.
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* |[
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* gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.4 ! pulsesink
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* ]| Play a 440Hz sine wave.
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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 <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 <gst/interfaces/streamvolume.h>
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#include <gst/gst-i18n-plugin.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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/* according to
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* http://www.pulseaudio.org/ticket/314
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* we need pulse-0.9.12 to use sink volume properties
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*/
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#define DEFAULT_SERVER NULL
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#define DEFAULT_DEVICE NULL
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#define DEFAULT_DEVICE_NAME NULL
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#define DEFAULT_VOLUME 1.0
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#define DEFAULT_MUTE FALSE
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#define MAX_VOLUME 10.0
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enum
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{
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PROP_0,
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PROP_SERVER,
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PROP_DEVICE,
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PROP_DEVICE_NAME,
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PROP_VOLUME,
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PROP_MUTE,
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PROP_LAST
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};
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#define GST_TYPE_PULSERING_BUFFER \
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(gst_pulseringbuffer_get_type())
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#define GST_PULSERING_BUFFER(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_PULSERING_BUFFER,GstPulseRingBuffer))
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#define GST_PULSERING_BUFFER_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_PULSERING_BUFFER,GstPulseRingBufferClass))
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#define GST_PULSERING_BUFFER_GET_CLASS(obj) \
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(G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_PULSERING_BUFFER, GstPulseRingBufferClass))
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#define GST_PULSERING_BUFFER_CAST(obj) \
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((GstPulseRingBuffer *)obj)
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#define GST_IS_PULSERING_BUFFER(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_PULSERING_BUFFER))
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#define GST_IS_PULSERING_BUFFER_CLASS(klass)\
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_PULSERING_BUFFER))
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typedef struct _GstPulseRingBuffer GstPulseRingBuffer;
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typedef struct _GstPulseRingBufferClass GstPulseRingBufferClass;
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/* We keep a custom ringbuffer that is backed up by data allocated by
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* pulseaudio. We must also overide the commit function to write into
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* pulseaudio memory instead. */
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struct _GstPulseRingBuffer
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{
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GstRingBuffer object;
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gchar *stream_name;
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pa_context *context;
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pa_stream *stream;
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pa_sample_spec sample_spec;
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gboolean corked:1;
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gboolean in_commit:1;
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gboolean paused:1;
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};
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struct _GstPulseRingBufferClass
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{
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GstRingBufferClass parent_class;
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};
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static void gst_pulseringbuffer_class_init (GstPulseRingBufferClass * klass);
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static void gst_pulseringbuffer_init (GstPulseRingBuffer * ringbuffer,
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GstPulseRingBufferClass * klass);
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static void gst_pulseringbuffer_finalize (GObject * object);
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static GstRingBufferClass *ring_parent_class = NULL;
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static gboolean gst_pulseringbuffer_open_device (GstRingBuffer * buf);
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static gboolean gst_pulseringbuffer_close_device (GstRingBuffer * buf);
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static gboolean gst_pulseringbuffer_acquire (GstRingBuffer * buf,
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GstRingBufferSpec * spec);
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static gboolean gst_pulseringbuffer_release (GstRingBuffer * buf);
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static gboolean gst_pulseringbuffer_start (GstRingBuffer * buf);
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static gboolean gst_pulseringbuffer_pause (GstRingBuffer * buf);
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static gboolean gst_pulseringbuffer_stop (GstRingBuffer * buf);
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static void gst_pulseringbuffer_clear (GstRingBuffer * buf);
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static guint gst_pulseringbuffer_commit (GstRingBuffer * buf,
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guint64 * sample, guchar * data, gint in_samples, gint out_samples,
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gint * accum);
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/* ringbuffer abstract base class */
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static GType
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gst_pulseringbuffer_get_type (void)
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{
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static GType ringbuffer_type = 0;
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if (!ringbuffer_type) {
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static const GTypeInfo ringbuffer_info = {
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sizeof (GstPulseRingBufferClass),
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NULL,
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NULL,
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(GClassInitFunc) gst_pulseringbuffer_class_init,
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NULL,
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NULL,
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sizeof (GstPulseRingBuffer),
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0,
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(GInstanceInitFunc) gst_pulseringbuffer_init,
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NULL
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};
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ringbuffer_type =
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g_type_register_static (GST_TYPE_RING_BUFFER, "GstPulseSinkRingBuffer",
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&ringbuffer_info, 0);
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}
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return ringbuffer_type;
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}
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static void
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gst_pulseringbuffer_class_init (GstPulseRingBufferClass * klass)
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{
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GObjectClass *gobject_class;
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GstRingBufferClass *gstringbuffer_class;
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gobject_class = (GObjectClass *) klass;
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gstringbuffer_class = (GstRingBufferClass *) klass;
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ring_parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_finalize);
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gstringbuffer_class->open_device =
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GST_DEBUG_FUNCPTR (gst_pulseringbuffer_open_device);
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gstringbuffer_class->close_device =
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GST_DEBUG_FUNCPTR (gst_pulseringbuffer_close_device);
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gstringbuffer_class->acquire =
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GST_DEBUG_FUNCPTR (gst_pulseringbuffer_acquire);
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gstringbuffer_class->release =
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GST_DEBUG_FUNCPTR (gst_pulseringbuffer_release);
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gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_start);
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gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_pause);
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gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_start);
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gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_stop);
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gstringbuffer_class->clear_all =
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GST_DEBUG_FUNCPTR (gst_pulseringbuffer_clear);
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gstringbuffer_class->commit = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_commit);
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/* ref class from a thread-safe context to work around missing bit of
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* thread-safety in GObject */
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g_type_class_ref (GST_TYPE_PULSERING_BUFFER);
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}
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static void
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gst_pulseringbuffer_init (GstPulseRingBuffer * pbuf,
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GstPulseRingBufferClass * g_class)
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{
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pbuf->stream_name = NULL;
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pbuf->context = NULL;
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pbuf->stream = NULL;
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#if HAVE_PULSE_0_9_13
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pa_sample_spec_init (&pbuf->sample_spec);
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#else
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pbuf->sample_spec.format = PA_SAMPLE_INVALID;
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pbuf->sample_spec.rate = 0;
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pbuf->sample_spec.channels = 0;
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#endif
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pbuf->corked = TRUE;
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pbuf->in_commit = FALSE;
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pbuf->paused = FALSE;
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}
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static void
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gst_pulsering_destroy_stream (GstPulseRingBuffer * pbuf)
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{
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if (pbuf->stream) {
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pa_stream_disconnect (pbuf->stream);
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/* Make sure we don't get any further callbacks */
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pa_stream_set_state_callback (pbuf->stream, NULL, NULL);
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pa_stream_set_write_callback (pbuf->stream, NULL, NULL);
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pa_stream_set_underflow_callback (pbuf->stream, NULL, NULL);
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pa_stream_set_overflow_callback (pbuf->stream, NULL, NULL);
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pa_stream_unref (pbuf->stream);
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pbuf->stream = NULL;
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}
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g_free (pbuf->stream_name);
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pbuf->stream_name = NULL;
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}
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static void
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gst_pulsering_destroy_context (GstPulseRingBuffer * pbuf)
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{
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gst_pulsering_destroy_stream (pbuf);
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if (pbuf->context) {
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pa_context_disconnect (pbuf->context);
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/* Make sure we don't get any further callbacks */
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pa_context_set_state_callback (pbuf->context, NULL, NULL);
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#if HAVE_PULSE_0_9_12
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pa_context_set_subscribe_callback (pbuf->context, NULL, NULL);
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#endif
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pa_context_unref (pbuf->context);
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pbuf->context = NULL;
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}
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}
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static void
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gst_pulseringbuffer_finalize (GObject * object)
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{
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GstPulseRingBuffer *ringbuffer;
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ringbuffer = GST_PULSERING_BUFFER_CAST (object);
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gst_pulsering_destroy_context (ringbuffer);
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G_OBJECT_CLASS (ring_parent_class)->finalize (object);
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}
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static gboolean
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gst_pulsering_is_dead (GstPulseSink * psink, GstPulseRingBuffer * pbuf)
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{
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if (!pbuf->context
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|| !PA_CONTEXT_IS_GOOD (pa_context_get_state (pbuf->context))
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|| !pbuf->stream
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|| !PA_STREAM_IS_GOOD (pa_stream_get_state (pbuf->stream))) {
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const gchar *err_str = pbuf->context ?
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pa_strerror (pa_context_errno (pbuf->context)) : NULL;
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GST_ELEMENT_ERROR (psink, RESOURCE, FAILED, ("Disconnected: %s",
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err_str), (NULL));
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return TRUE;
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}
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return FALSE;
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}
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static void
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gst_pulsering_context_state_cb (pa_context * c, void *userdata)
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{
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GstPulseSink *psink;
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GstPulseRingBuffer *pbuf;
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pa_context_state_t state;
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pbuf = GST_PULSERING_BUFFER_CAST (userdata);
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psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
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state = pa_context_get_state (c);
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GST_LOG_OBJECT (psink, "got new context state %d", state);
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switch (state) {
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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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GST_LOG_OBJECT (psink, "signaling");
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pa_threaded_mainloop_signal (psink->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;
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}
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}
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#if HAVE_PULSE_0_9_12
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static void
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gst_pulsering_context_subscribe_cb (pa_context * c,
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pa_subscription_event_type_t t, uint32_t idx, void *userdata)
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{
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GstPulseSink *psink;
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GstPulseRingBuffer *pbuf;
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pbuf = GST_PULSERING_BUFFER_CAST (userdata);
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psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
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GST_LOG_OBJECT (psink, "type %d, idx %u", t, idx);
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if (t != (PA_SUBSCRIPTION_EVENT_SINK_INPUT | PA_SUBSCRIPTION_EVENT_CHANGE) &&
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t != (PA_SUBSCRIPTION_EVENT_SINK_INPUT | PA_SUBSCRIPTION_EVENT_NEW))
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return;
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if (!pbuf->stream)
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return;
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if (idx != pa_stream_get_index (pbuf->stream))
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return;
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/* Actually this event is also triggered when other properties of
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* the stream change that are unrelated to the volume. However it is
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* probably cheaper to signal the change here and check for the
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* volume when the GObject property is read instead of querying it always. */
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/* inform streaming thread to notify */
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g_atomic_int_compare_and_exchange (&psink->notify, 0, 1);
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}
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#endif
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/* will be called when the device should be opened. In this case we will connect
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* to the server. We should not try to open any streams in this state. */
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static gboolean
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gst_pulseringbuffer_open_device (GstRingBuffer * buf)
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{
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GstPulseSink *psink;
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GstPulseRingBuffer *pbuf;
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gchar *name;
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pa_mainloop_api *api;
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psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
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pbuf = GST_PULSERING_BUFFER_CAST (buf);
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g_assert (!pbuf->context);
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g_assert (!pbuf->stream);
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name = gst_pulse_client_name ();
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pa_threaded_mainloop_lock (psink->mainloop);
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/* get the mainloop api and create a context */
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GST_LOG_OBJECT (psink, "new context with name %s", GST_STR_NULL (name));
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api = pa_threaded_mainloop_get_api (psink->mainloop);
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if (!(pbuf->context = pa_context_new (api, name)))
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goto create_failed;
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/* register some essential callbacks */
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pa_context_set_state_callback (pbuf->context,
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gst_pulsering_context_state_cb, pbuf);
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#if HAVE_PULSE_0_9_12
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pa_context_set_subscribe_callback (pbuf->context,
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gst_pulsering_context_subscribe_cb, pbuf);
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#endif
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/* try to connect to the server and wait for completioni, we don't want to
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* autospawn a deamon */
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GST_LOG_OBJECT (psink, "connect to server %s", GST_STR_NULL (psink->server));
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if (pa_context_connect (pbuf->context, psink->server, PA_CONTEXT_NOAUTOSPAWN,
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NULL) < 0)
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goto connect_failed;
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for (;;) {
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pa_context_state_t state;
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state = pa_context_get_state (pbuf->context);
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GST_LOG_OBJECT (psink, "context state is now %d", state);
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if (!PA_CONTEXT_IS_GOOD (state))
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goto connect_failed;
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|
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if (state == PA_CONTEXT_READY)
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break;
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|
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/* Wait until the context is ready */
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GST_LOG_OBJECT (psink, "waiting..");
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pa_threaded_mainloop_wait (psink->mainloop);
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}
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GST_LOG_OBJECT (psink, "opened the device");
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|
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pa_threaded_mainloop_unlock (psink->mainloop);
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g_free (name);
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return TRUE;
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/* ERRORS */
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unlock_and_fail:
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{
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gst_pulsering_destroy_context (pbuf);
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pa_threaded_mainloop_unlock (psink->mainloop);
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g_free (name);
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return FALSE;
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}
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create_failed:
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{
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GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
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("Failed to create context"), (NULL));
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goto unlock_and_fail;
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}
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connect_failed:
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{
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GST_ELEMENT_ERROR (psink, RESOURCE, FAILED, ("Failed to connect: %s",
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pa_strerror (pa_context_errno (pbuf->context))), (NULL));
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goto unlock_and_fail;
|
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}
|
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}
|
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|
|
/* close the device */
|
|
static gboolean
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gst_pulseringbuffer_close_device (GstRingBuffer * buf)
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{
|
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GstPulseSink *psink;
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GstPulseRingBuffer *pbuf;
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pbuf = GST_PULSERING_BUFFER_CAST (buf);
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psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
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GST_LOG_OBJECT (psink, "closing device");
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pa_threaded_mainloop_lock (psink->mainloop);
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gst_pulsering_destroy_context (pbuf);
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pa_threaded_mainloop_unlock (psink->mainloop);
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GST_LOG_OBJECT (psink, "closed device");
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return TRUE;
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}
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|
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static void
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gst_pulsering_stream_state_cb (pa_stream * s, void *userdata)
|
|
{
|
|
GstPulseSink *psink;
|
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GstPulseRingBuffer *pbuf;
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pa_stream_state_t state;
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|
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pbuf = GST_PULSERING_BUFFER_CAST (userdata);
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psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
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|
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state = pa_stream_get_state (s);
|
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GST_LOG_OBJECT (psink, "got new stream state %d", state);
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|
|
switch (state) {
|
|
case PA_STREAM_READY:
|
|
case PA_STREAM_FAILED:
|
|
case PA_STREAM_TERMINATED:
|
|
GST_LOG_OBJECT (psink, "signaling");
|
|
pa_threaded_mainloop_signal (psink->mainloop, 0);
|
|
break;
|
|
case PA_STREAM_UNCONNECTED:
|
|
case PA_STREAM_CREATING:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsering_stream_request_cb (pa_stream * s, size_t length, void *userdata)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstRingBuffer *rbuf;
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
rbuf = GST_RING_BUFFER_CAST (userdata);
|
|
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
GST_LOG_OBJECT (psink, "got request for length %" G_GSIZE_FORMAT, length);
|
|
|
|
if (pbuf->in_commit && (length >= rbuf->spec.segsize)) {
|
|
/* only signal when we are waiting in the commit thread
|
|
* and got request for atleast a segment */
|
|
pa_threaded_mainloop_signal (psink->mainloop, 0);
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsering_stream_underflow_cb (pa_stream * s, void *userdata)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
GST_WARNING_OBJECT (psink, "Got underflow");
|
|
}
|
|
|
|
static void
|
|
gst_pulsering_stream_overflow_cb (pa_stream * s, void *userdata)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
GST_WARNING_OBJECT (psink, "Got overflow");
|
|
}
|
|
|
|
static void
|
|
gst_pulsering_stream_latency_cb (pa_stream * s, void *userdata)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
const pa_timing_info *info;
|
|
pa_usec_t sink_usec;
|
|
|
|
info = pa_stream_get_timing_info (s);
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
if (!info) {
|
|
GST_LOG_OBJECT (psink, "latency update (information unknown)");
|
|
return;
|
|
}
|
|
#if HAVE_PULSE_0_9_11
|
|
sink_usec = info->configured_sink_usec;
|
|
#else
|
|
sink_usec = 0;
|
|
#endif
|
|
|
|
GST_LOG_OBJECT (psink,
|
|
"latency_update, %" G_GUINT64_FORMAT ", %d:%" G_GINT64_FORMAT ", %d:%"
|
|
G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT,
|
|
GST_TIMEVAL_TO_TIME (info->timestamp), info->write_index_corrupt,
|
|
info->write_index, info->read_index_corrupt, info->read_index,
|
|
info->sink_usec, sink_usec);
|
|
}
|
|
|
|
static void
|
|
gst_pulsering_stream_suspended_cb (pa_stream * p, void *userdata)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
if (pa_stream_is_suspended (p))
|
|
GST_DEBUG_OBJECT (psink, "stream suspended");
|
|
else
|
|
GST_DEBUG_OBJECT (psink, "stream resumed");
|
|
}
|
|
|
|
#if HAVE_PULSE_0_9_11
|
|
static void
|
|
gst_pulsering_stream_started_cb (pa_stream * p, void *userdata)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
GST_DEBUG_OBJECT (psink, "stream started");
|
|
}
|
|
#endif
|
|
|
|
#if HAVE_PULSE_0_9_15
|
|
static void
|
|
gst_pulsering_stream_event_cb (pa_stream * p, const char *name,
|
|
pa_proplist * pl, void *userdata)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
if (!strcmp (name, PA_STREAM_EVENT_REQUEST_CORK)) {
|
|
/* the stream wants to PAUSE, post a message for the application. */
|
|
GST_DEBUG_OBJECT (psink, "got request for CORK");
|
|
gst_element_post_message (GST_ELEMENT_CAST (psink),
|
|
gst_message_new_request_state (GST_OBJECT_CAST (psink),
|
|
GST_STATE_PAUSED));
|
|
|
|
} else if (!strcmp (name, PA_STREAM_EVENT_REQUEST_UNCORK)) {
|
|
GST_DEBUG_OBJECT (psink, "got request for UNCORK");
|
|
gst_element_post_message (GST_ELEMENT_CAST (psink),
|
|
gst_message_new_request_state (GST_OBJECT_CAST (psink),
|
|
GST_STATE_PLAYING));
|
|
} else {
|
|
GST_DEBUG_OBJECT (psink, "got unknown event %s", name);
|
|
}
|
|
}
|
|
#endif
|
|
|
|
/* This method should create a new stream of the given @spec. No playback should
|
|
* start yet so we start in the corked state. */
|
|
static gboolean
|
|
gst_pulseringbuffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
pa_buffer_attr wanted;
|
|
const pa_buffer_attr *actual;
|
|
pa_channel_map channel_map;
|
|
pa_operation *o = NULL;
|
|
#if HAVE_PULSE_0_9_20
|
|
pa_cvolume v;
|
|
#endif
|
|
pa_cvolume *pv = NULL;
|
|
pa_stream_flags_t flags;
|
|
const gchar *name;
|
|
GstAudioClock *clock;
|
|
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
|
|
pbuf = GST_PULSERING_BUFFER_CAST (buf);
|
|
|
|
GST_LOG_OBJECT (psink, "creating sample spec");
|
|
/* convert the gstreamer sample spec to the pulseaudio format */
|
|
if (!gst_pulse_fill_sample_spec (spec, &pbuf->sample_spec))
|
|
goto invalid_spec;
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
|
|
/* we need a context and a no stream */
|
|
g_assert (pbuf->context);
|
|
g_assert (!pbuf->stream);
|
|
|
|
/* enable event notifications */
|
|
GST_LOG_OBJECT (psink, "subscribing to context events");
|
|
if (!(o = pa_context_subscribe (pbuf->context,
|
|
PA_SUBSCRIPTION_MASK_SINK_INPUT, NULL, NULL)))
|
|
goto subscribe_failed;
|
|
|
|
pa_operation_unref (o);
|
|
|
|
/* initialize the channel map */
|
|
gst_pulse_gst_to_channel_map (&channel_map, spec);
|
|
|
|
/* find a good name for the stream */
|
|
if (psink->stream_name)
|
|
name = psink->stream_name;
|
|
else
|
|
name = "Playback Stream";
|
|
|
|
/* create a stream */
|
|
GST_LOG_OBJECT (psink, "creating stream with name %s", name);
|
|
if (!(pbuf->stream = pa_stream_new (pbuf->context,
|
|
name, &pbuf->sample_spec, &channel_map)))
|
|
goto stream_failed;
|
|
|
|
/* install essential callbacks */
|
|
pa_stream_set_state_callback (pbuf->stream,
|
|
gst_pulsering_stream_state_cb, pbuf);
|
|
pa_stream_set_write_callback (pbuf->stream,
|
|
gst_pulsering_stream_request_cb, pbuf);
|
|
pa_stream_set_underflow_callback (pbuf->stream,
|
|
gst_pulsering_stream_underflow_cb, pbuf);
|
|
pa_stream_set_overflow_callback (pbuf->stream,
|
|
gst_pulsering_stream_overflow_cb, pbuf);
|
|
pa_stream_set_latency_update_callback (pbuf->stream,
|
|
gst_pulsering_stream_latency_cb, pbuf);
|
|
pa_stream_set_suspended_callback (pbuf->stream,
|
|
gst_pulsering_stream_suspended_cb, pbuf);
|
|
#if HAVE_PULSE_0_9_11
|
|
pa_stream_set_started_callback (pbuf->stream,
|
|
gst_pulsering_stream_started_cb, pbuf);
|
|
#endif
|
|
#if HAVE_PULSE_0_9_15
|
|
pa_stream_set_event_callback (pbuf->stream,
|
|
gst_pulsering_stream_event_cb, pbuf);
|
|
#endif
|
|
|
|
/* buffering requirements. When setting prebuf to 0, the stream will not pause
|
|
* when we cause an underrun, which causes time to continue. */
|
|
memset (&wanted, 0, sizeof (wanted));
|
|
wanted.tlength = spec->segtotal * spec->segsize;
|
|
wanted.maxlength = -1;
|
|
wanted.prebuf = 0;
|
|
wanted.minreq = spec->segsize;
|
|
|
|
GST_INFO_OBJECT (psink, "tlength: %d", wanted.tlength);
|
|
GST_INFO_OBJECT (psink, "maxlength: %d", wanted.maxlength);
|
|
GST_INFO_OBJECT (psink, "prebuf: %d", wanted.prebuf);
|
|
GST_INFO_OBJECT (psink, "minreq: %d", wanted.minreq);
|
|
|
|
#if HAVE_PULSE_0_9_20
|
|
/* configure volume when we changed it, else we leave the default */
|
|
if (psink->volume_set) {
|
|
GST_LOG_OBJECT (psink, "have volume of %f", psink->volume);
|
|
pv = &v;
|
|
gst_pulse_cvolume_from_linear (pv, pbuf->sample_spec.channels,
|
|
psink->volume);
|
|
} else {
|
|
pv = NULL;
|
|
}
|
|
#endif
|
|
|
|
/* construct the flags */
|
|
flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
|
|
#if HAVE_PULSE_0_9_11
|
|
PA_STREAM_ADJUST_LATENCY |
|
|
#endif
|
|
PA_STREAM_START_CORKED;
|
|
|
|
#if HAVE_PULSE_0_9_12
|
|
if (psink->mute_set && psink->mute)
|
|
flags |= PA_STREAM_START_MUTED;
|
|
#endif
|
|
|
|
/* we always start corked (see flags above) */
|
|
pbuf->corked = TRUE;
|
|
|
|
/* try to connect now */
|
|
GST_LOG_OBJECT (psink, "connect for playback to device %s",
|
|
GST_STR_NULL (psink->device));
|
|
if (pa_stream_connect_playback (pbuf->stream, psink->device,
|
|
&wanted, flags, pv, NULL) < 0)
|
|
goto connect_failed;
|
|
|
|
/* our clock will now start from 0 again */
|
|
clock = GST_AUDIO_CLOCK (GST_BASE_AUDIO_SINK (psink)->provided_clock);
|
|
gst_audio_clock_reset (clock, 0);
|
|
|
|
for (;;) {
|
|
pa_stream_state_t state;
|
|
|
|
state = pa_stream_get_state (pbuf->stream);
|
|
|
|
GST_LOG_OBJECT (psink, "stream state is now %d", state);
|
|
|
|
if (!PA_STREAM_IS_GOOD (state))
|
|
goto connect_failed;
|
|
|
|
if (state == PA_STREAM_READY)
|
|
break;
|
|
|
|
/* Wait until the stream is ready */
|
|
pa_threaded_mainloop_wait (psink->mainloop);
|
|
}
|
|
|
|
/* After we passed the volume off of to PA we never want to set it
|
|
again, since it is PA's job to save/restore volumes. */
|
|
psink->volume_set = psink->mute_set = FALSE;
|
|
|
|
GST_LOG_OBJECT (psink, "stream is acquired now");
|
|
|
|
/* get the actual buffering properties now */
|
|
actual = pa_stream_get_buffer_attr (pbuf->stream);
|
|
|
|
GST_INFO_OBJECT (psink, "tlength: %d (wanted: %d)", actual->tlength,
|
|
wanted.tlength);
|
|
GST_INFO_OBJECT (psink, "maxlength: %d", actual->maxlength);
|
|
GST_INFO_OBJECT (psink, "prebuf: %d", actual->prebuf);
|
|
GST_INFO_OBJECT (psink, "minreq: %d (wanted %d)", actual->minreq,
|
|
wanted.minreq);
|
|
|
|
spec->segsize = actual->minreq;
|
|
spec->segtotal = actual->tlength / spec->segsize;
|
|
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
unlock_and_fail:
|
|
{
|
|
gst_pulsering_destroy_stream (pbuf);
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
return FALSE;
|
|
}
|
|
invalid_spec:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, SETTINGS,
|
|
("Invalid sample specification."), (NULL));
|
|
return FALSE;
|
|
}
|
|
subscribe_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("pa_context_subscribe() failed: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
stream_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("Failed to create stream: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
connect_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("Failed to connect stream: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
}
|
|
|
|
/* free the stream that we acquired before */
|
|
static gboolean
|
|
gst_pulseringbuffer_release (GstRingBuffer * buf)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
|
|
pbuf = GST_PULSERING_BUFFER_CAST (buf);
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
gst_pulsering_destroy_stream (pbuf);
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_pulsering_success_cb (pa_stream * s, int success, void *userdata)
|
|
{
|
|
GstPulseRingBuffer *pbuf;
|
|
GstPulseSink *psink;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
pa_threaded_mainloop_signal (psink->mainloop, 0);
|
|
}
|
|
|
|
/* update the corked state of a stream, must be called with the mainloop
|
|
* lock */
|
|
static gboolean
|
|
gst_pulsering_set_corked (GstPulseRingBuffer * pbuf, gboolean corked,
|
|
gboolean wait)
|
|
{
|
|
pa_operation *o = NULL;
|
|
GstPulseSink *psink;
|
|
gboolean res = FALSE;
|
|
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
GST_DEBUG_OBJECT (psink, "setting corked state to %d", corked);
|
|
if (pbuf->corked != corked) {
|
|
if (!(o = pa_stream_cork (pbuf->stream, corked,
|
|
gst_pulsering_success_cb, pbuf)))
|
|
goto cork_failed;
|
|
|
|
while (wait && pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
|
|
pa_threaded_mainloop_wait (psink->mainloop);
|
|
if (gst_pulsering_is_dead (psink, pbuf))
|
|
goto server_dead;
|
|
}
|
|
pbuf->corked = corked;
|
|
} else {
|
|
GST_DEBUG_OBJECT (psink, "skipping, already in requested state");
|
|
}
|
|
res = TRUE;
|
|
|
|
cleanup:
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
server_dead:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "the server is dead");
|
|
goto cleanup;
|
|
}
|
|
cork_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("pa_stream_cork() failed: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto cleanup;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulseringbuffer_clear (GstRingBuffer * buf)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
pa_operation *o = NULL;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (buf);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
GST_DEBUG_OBJECT (psink, "clearing");
|
|
if (pbuf->stream) {
|
|
/* don't wait for the flush to complete */
|
|
if ((o = pa_stream_flush (pbuf->stream, NULL, pbuf)))
|
|
pa_operation_unref (o);
|
|
}
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
}
|
|
|
|
static void
|
|
mainloop_enter_defer_cb (pa_mainloop_api * api, void *userdata)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
|
|
GstMessage *message;
|
|
GValue val = { 0 };
|
|
|
|
g_value_init (&val, G_TYPE_POINTER);
|
|
g_value_set_pointer (&val, g_thread_self ());
|
|
|
|
GST_DEBUG_OBJECT (pulsesink, "posting ENTER stream status");
|
|
message = gst_message_new_stream_status (GST_OBJECT (pulsesink),
|
|
GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT (pulsesink));
|
|
gst_message_set_stream_status_object (message, &val);
|
|
|
|
gst_element_post_message (GST_ELEMENT (pulsesink), message);
|
|
|
|
/* signal the waiter */
|
|
pulsesink->pa_defer_ran = TRUE;
|
|
pa_threaded_mainloop_signal (pulsesink->mainloop, 0);
|
|
}
|
|
|
|
/* start/resume playback ASAP, we don't uncork here but in the commit method */
|
|
static gboolean
|
|
gst_pulseringbuffer_start (GstRingBuffer * buf)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (buf);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
|
|
GST_DEBUG_OBJECT (psink, "scheduling stream status");
|
|
psink->pa_defer_ran = FALSE;
|
|
pa_mainloop_api_once (pa_threaded_mainloop_get_api (psink->mainloop),
|
|
mainloop_enter_defer_cb, psink);
|
|
|
|
GST_DEBUG_OBJECT (psink, "starting");
|
|
pbuf->paused = FALSE;
|
|
gst_pulsering_set_corked (pbuf, FALSE, FALSE);
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* pause/stop playback ASAP */
|
|
static gboolean
|
|
gst_pulseringbuffer_pause (GstRingBuffer * buf)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
gboolean res;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (buf);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
GST_DEBUG_OBJECT (psink, "pausing and corking");
|
|
/* make sure the commit method stops writing */
|
|
pbuf->paused = TRUE;
|
|
res = gst_pulsering_set_corked (pbuf, TRUE, FALSE);
|
|
if (pbuf->in_commit) {
|
|
/* we are waiting in a commit, signal */
|
|
GST_DEBUG_OBJECT (psink, "signal commit");
|
|
pa_threaded_mainloop_signal (psink->mainloop, 0);
|
|
}
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
mainloop_leave_defer_cb (pa_mainloop_api * api, void *userdata)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
|
|
GstMessage *message;
|
|
GValue val = { 0 };
|
|
|
|
g_value_init (&val, G_TYPE_POINTER);
|
|
g_value_set_pointer (&val, g_thread_self ());
|
|
|
|
GST_DEBUG_OBJECT (pulsesink, "posting LEAVE stream status");
|
|
message = gst_message_new_stream_status (GST_OBJECT (pulsesink),
|
|
GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT (pulsesink));
|
|
gst_message_set_stream_status_object (message, &val);
|
|
gst_element_post_message (GST_ELEMENT (pulsesink), message);
|
|
|
|
pulsesink->pa_defer_ran = TRUE;
|
|
pa_threaded_mainloop_signal (pulsesink->mainloop, 0);
|
|
}
|
|
|
|
/* stop playback, we flush everything. */
|
|
static gboolean
|
|
gst_pulseringbuffer_stop (GstRingBuffer * buf)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
gboolean res = FALSE;
|
|
pa_operation *o = NULL;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (buf);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
pbuf->paused = TRUE;
|
|
res = gst_pulsering_set_corked (pbuf, TRUE, TRUE);
|
|
/* Inform anyone waiting in _commit() call that it shall wakeup */
|
|
if (pbuf->in_commit) {
|
|
GST_DEBUG_OBJECT (psink, "signal commit thread");
|
|
pa_threaded_mainloop_signal (psink->mainloop, 0);
|
|
}
|
|
|
|
if (strcmp (psink->pa_version, "0.9.12")) {
|
|
/* then try to flush, it's not fatal when this fails */
|
|
GST_DEBUG_OBJECT (psink, "flushing");
|
|
if ((o = pa_stream_flush (pbuf->stream, gst_pulsering_success_cb, pbuf))) {
|
|
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
|
|
GST_DEBUG_OBJECT (psink, "wait for completion");
|
|
pa_threaded_mainloop_wait (psink->mainloop);
|
|
if (gst_pulsering_is_dead (psink, pbuf))
|
|
goto server_dead;
|
|
}
|
|
GST_DEBUG_OBJECT (psink, "flush completed");
|
|
}
|
|
}
|
|
res = TRUE;
|
|
|
|
cleanup:
|
|
if (o) {
|
|
pa_operation_cancel (o);
|
|
pa_operation_unref (o);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (psink, "scheduling stream status");
|
|
psink->pa_defer_ran = FALSE;
|
|
pa_mainloop_api_once (pa_threaded_mainloop_get_api (psink->mainloop),
|
|
mainloop_leave_defer_cb, psink);
|
|
|
|
GST_DEBUG_OBJECT (psink, "waiting for stream status");
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
server_dead:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "the server is dead");
|
|
goto cleanup;
|
|
}
|
|
}
|
|
|
|
/* in_samples >= out_samples, rate > 1.0 */
|
|
#define FWD_UP_SAMPLES(s,se,d,de) \
|
|
G_STMT_START { \
|
|
guint8 *sb = s, *db = d; \
|
|
while (s <= se && d < de) { \
|
|
memcpy (d, s, bps); \
|
|
s += bps; \
|
|
*accum += outr; \
|
|
if ((*accum << 1) >= inr) { \
|
|
*accum -= inr; \
|
|
d += bps; \
|
|
} \
|
|
} \
|
|
in_samples -= (s - sb)/bps; \
|
|
out_samples -= (d - db)/bps; \
|
|
GST_DEBUG ("fwd_up end %d/%d",*accum,*toprocess); \
|
|
} G_STMT_END
|
|
|
|
/* out_samples > in_samples, for rates smaller than 1.0 */
|
|
#define FWD_DOWN_SAMPLES(s,se,d,de) \
|
|
G_STMT_START { \
|
|
guint8 *sb = s, *db = d; \
|
|
while (s <= se && d < de) { \
|
|
memcpy (d, s, bps); \
|
|
d += bps; \
|
|
*accum += inr; \
|
|
if ((*accum << 1) >= outr) { \
|
|
*accum -= outr; \
|
|
s += bps; \
|
|
} \
|
|
} \
|
|
in_samples -= (s - sb)/bps; \
|
|
out_samples -= (d - db)/bps; \
|
|
GST_DEBUG ("fwd_down end %d/%d",*accum,*toprocess); \
|
|
} G_STMT_END
|
|
|
|
#define REV_UP_SAMPLES(s,se,d,de) \
|
|
G_STMT_START { \
|
|
guint8 *sb = se, *db = d; \
|
|
while (s <= se && d < de) { \
|
|
memcpy (d, se, bps); \
|
|
se -= bps; \
|
|
*accum += outr; \
|
|
while (d < de && (*accum << 1) >= inr) { \
|
|
*accum -= inr; \
|
|
d += bps; \
|
|
} \
|
|
} \
|
|
in_samples -= (sb - se)/bps; \
|
|
out_samples -= (d - db)/bps; \
|
|
GST_DEBUG ("rev_up end %d/%d",*accum,*toprocess); \
|
|
} G_STMT_END
|
|
|
|
#define REV_DOWN_SAMPLES(s,se,d,de) \
|
|
G_STMT_START { \
|
|
guint8 *sb = se, *db = d; \
|
|
while (s <= se && d < de) { \
|
|
memcpy (d, se, bps); \
|
|
d += bps; \
|
|
*accum += inr; \
|
|
while (s <= se && (*accum << 1) >= outr) { \
|
|
*accum -= outr; \
|
|
se -= bps; \
|
|
} \
|
|
} \
|
|
in_samples -= (sb - se)/bps; \
|
|
out_samples -= (d - db)/bps; \
|
|
GST_DEBUG ("rev_down end %d/%d",*accum,*toprocess); \
|
|
} G_STMT_END
|
|
|
|
|
|
/* our custom commit function because we write into the buffer of pulseaudio
|
|
* instead of keeping our own buffer */
|
|
static guint
|
|
gst_pulseringbuffer_commit (GstRingBuffer * buf, guint64 * sample,
|
|
guchar * data, gint in_samples, gint out_samples, gint * accum)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
guint result;
|
|
guint8 *data_end;
|
|
gboolean reverse;
|
|
gint *toprocess;
|
|
gint inr, outr, bps;
|
|
gint64 offset;
|
|
guint bufsize;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (buf);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
/* FIXME post message rather than using a signal (as mixer interface) */
|
|
if (g_atomic_int_compare_and_exchange (&psink->notify, 1, 0)) {
|
|
g_object_notify (G_OBJECT (psink), "volume");
|
|
g_object_notify (G_OBJECT (psink), "mute");
|
|
}
|
|
|
|
/* make sure the ringbuffer is started */
|
|
if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
|
|
GST_RING_BUFFER_STATE_STARTED)) {
|
|
/* see if we are allowed to start it */
|
|
if (G_UNLIKELY (g_atomic_int_get (&buf->abidata.ABI.may_start) == FALSE))
|
|
goto no_start;
|
|
|
|
GST_DEBUG_OBJECT (buf, "start!");
|
|
if (!gst_ring_buffer_start (buf))
|
|
goto start_failed;
|
|
}
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
GST_DEBUG_OBJECT (psink, "entering commit");
|
|
pbuf->in_commit = TRUE;
|
|
|
|
bps = buf->spec.bytes_per_sample;
|
|
bufsize = buf->spec.segsize * buf->spec.segtotal;
|
|
|
|
/* our toy resampler for trick modes */
|
|
reverse = out_samples < 0;
|
|
out_samples = ABS (out_samples);
|
|
|
|
if (in_samples >= out_samples)
|
|
toprocess = &in_samples;
|
|
else
|
|
toprocess = &out_samples;
|
|
|
|
inr = in_samples - 1;
|
|
outr = out_samples - 1;
|
|
|
|
GST_DEBUG_OBJECT (psink, "in %d, out %d", inr, outr);
|
|
|
|
/* data_end points to the last sample we have to write, not past it. This is
|
|
* needed to properly handle reverse playback: it points to the last sample. */
|
|
data_end = data + (bps * inr);
|
|
|
|
if (pbuf->paused)
|
|
goto was_paused;
|
|
|
|
/* offset is in bytes */
|
|
offset = *sample * bps;
|
|
|
|
while (*toprocess > 0) {
|
|
size_t avail;
|
|
guint towrite;
|
|
|
|
GST_LOG_OBJECT (psink,
|
|
"need to write %d samples at offset %" G_GINT64_FORMAT, *toprocess,
|
|
offset);
|
|
|
|
for (;;) {
|
|
/* FIXME, this is not quite right */
|
|
if ((avail = pa_stream_writable_size (pbuf->stream)) == (size_t) - 1)
|
|
goto writable_size_failed;
|
|
|
|
/* We always try to satisfy a request for data */
|
|
GST_LOG_OBJECT (psink, "writable bytes %" G_GSIZE_FORMAT, avail);
|
|
|
|
/* convert to samples, we can only deal with multiples of the
|
|
* sample size */
|
|
avail /= bps;
|
|
|
|
if (avail > 0)
|
|
break;
|
|
|
|
/* see if we need to uncork because we have no free space */
|
|
if (pbuf->corked) {
|
|
if (!gst_pulsering_set_corked (pbuf, FALSE, FALSE))
|
|
goto uncork_failed;
|
|
}
|
|
|
|
/* we can't write a single byte, wait a bit */
|
|
GST_LOG_OBJECT (psink, "waiting for free space");
|
|
pa_threaded_mainloop_wait (psink->mainloop);
|
|
|
|
if (pbuf->paused)
|
|
goto was_paused;
|
|
}
|
|
|
|
if (avail > out_samples)
|
|
avail = out_samples;
|
|
|
|
towrite = avail * bps;
|
|
|
|
GST_LOG_OBJECT (psink, "writing %u samples at offset %" G_GUINT64_FORMAT,
|
|
(guint) avail, offset);
|
|
|
|
if (G_LIKELY (inr == outr && !reverse)) {
|
|
/* no rate conversion, simply write out the samples */
|
|
if (pa_stream_write (pbuf->stream, data, towrite, NULL, offset,
|
|
PA_SEEK_ABSOLUTE) < 0)
|
|
goto write_failed;
|
|
|
|
data += towrite;
|
|
in_samples -= avail;
|
|
out_samples -= avail;
|
|
} else {
|
|
guint8 *dest, *d, *d_end;
|
|
|
|
/* we need to allocate a temporary buffer to resample the data into,
|
|
* FIXME, we should have a pulseaudio API to allocate this buffer for us
|
|
* from the shared memory. */
|
|
dest = d = g_malloc (towrite);
|
|
d_end = d + towrite;
|
|
|
|
if (!reverse) {
|
|
if (inr >= outr)
|
|
/* forward speed up */
|
|
FWD_UP_SAMPLES (data, data_end, d, d_end);
|
|
else
|
|
/* forward slow down */
|
|
FWD_DOWN_SAMPLES (data, data_end, d, d_end);
|
|
} else {
|
|
if (inr >= outr)
|
|
/* reverse speed up */
|
|
REV_UP_SAMPLES (data, data_end, d, d_end);
|
|
else
|
|
/* reverse slow down */
|
|
REV_DOWN_SAMPLES (data, data_end, d, d_end);
|
|
}
|
|
/* see what we have left to write */
|
|
towrite = (d - dest);
|
|
if (pa_stream_write (pbuf->stream, dest, towrite,
|
|
g_free, offset, PA_SEEK_ABSOLUTE) < 0)
|
|
goto write_failed;
|
|
|
|
avail = towrite / bps;
|
|
}
|
|
*sample += avail;
|
|
offset += avail * bps;
|
|
|
|
/* check if we need to uncork after writing the samples */
|
|
if (pbuf->corked) {
|
|
const pa_timing_info *info;
|
|
|
|
if ((info = pa_stream_get_timing_info (pbuf->stream))) {
|
|
GST_LOG_OBJECT (psink,
|
|
"read_index at %" G_GUINT64_FORMAT ", offset %" G_GINT64_FORMAT,
|
|
info->read_index, offset);
|
|
|
|
/* we uncork when the read_index is too far behind the offset we need
|
|
* to write to. */
|
|
if (info->read_index + bufsize <= offset) {
|
|
if (!gst_pulsering_set_corked (pbuf, FALSE, FALSE))
|
|
goto uncork_failed;
|
|
}
|
|
} else {
|
|
GST_LOG_OBJECT (psink, "no timing info available yet");
|
|
}
|
|
}
|
|
}
|
|
/* we consumed all samples here */
|
|
data = data_end + bps;
|
|
|
|
pbuf->in_commit = FALSE;
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
done:
|
|
result = inr - ((data_end - data) / bps);
|
|
GST_LOG_OBJECT (psink, "wrote %d samples", result);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
unlock_and_fail:
|
|
{
|
|
pbuf->in_commit = FALSE;
|
|
GST_LOG_OBJECT (psink, "we are reset");
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
goto done;
|
|
}
|
|
no_start:
|
|
{
|
|
GST_LOG_OBJECT (psink, "we can not start");
|
|
return 0;
|
|
}
|
|
start_failed:
|
|
{
|
|
GST_LOG_OBJECT (psink, "failed to start the ringbuffer");
|
|
return 0;
|
|
}
|
|
uncork_failed:
|
|
{
|
|
pbuf->in_commit = FALSE;
|
|
GST_ERROR_OBJECT (psink, "uncork failed");
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
goto done;
|
|
}
|
|
was_paused:
|
|
{
|
|
pbuf->in_commit = FALSE;
|
|
GST_LOG_OBJECT (psink, "we are paused");
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
goto done;
|
|
}
|
|
writable_size_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("pa_stream_writable_size() failed: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
write_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("pa_stream_write() failed: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
}
|
|
|
|
static void gst_pulsesink_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec);
|
|
static void gst_pulsesink_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec);
|
|
static void gst_pulsesink_finalize (GObject * object);
|
|
|
|
static gboolean gst_pulsesink_event (GstBaseSink * sink, GstEvent * event);
|
|
|
|
static void gst_pulsesink_init_interfaces (GType type);
|
|
|
|
#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
|
|
# define ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
|
|
#else
|
|
# define ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
|
|
#endif
|
|
|
|
GST_IMPLEMENT_PULSEPROBE_METHODS (GstPulseSink, gst_pulsesink);
|
|
GST_BOILERPLATE_FULL (GstPulseSink, gst_pulsesink, GstBaseAudioSink,
|
|
GST_TYPE_BASE_AUDIO_SINK, gst_pulsesink_init_interfaces);
|
|
|
|
static gboolean
|
|
gst_pulsesink_interface_supported (GstImplementsInterface *
|
|
iface, GType interface_type)
|
|
{
|
|
GstPulseSink *this = GST_PULSESINK_CAST (iface);
|
|
|
|
if (interface_type == GST_TYPE_PROPERTY_PROBE && this->probe)
|
|
return TRUE;
|
|
if (interface_type == GST_TYPE_STREAM_VOLUME)
|
|
return TRUE;
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_implements_interface_init (GstImplementsInterfaceClass * klass)
|
|
{
|
|
klass->supported = gst_pulsesink_interface_supported;
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_init_interfaces (GType type)
|
|
{
|
|
static const GInterfaceInfo implements_iface_info = {
|
|
(GInterfaceInitFunc) gst_pulsesink_implements_interface_init,
|
|
NULL,
|
|
NULL,
|
|
};
|
|
static const GInterfaceInfo probe_iface_info = {
|
|
(GInterfaceInitFunc) gst_pulsesink_property_probe_interface_init,
|
|
NULL,
|
|
NULL,
|
|
};
|
|
#if HAVE_PULSE_0_9_12
|
|
static const GInterfaceInfo svol_iface_info = {
|
|
NULL, NULL, NULL
|
|
};
|
|
|
|
g_type_add_interface_static (type, GST_TYPE_STREAM_VOLUME, &svol_iface_info);
|
|
#endif
|
|
|
|
g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
|
|
&implements_iface_info);
|
|
g_type_add_interface_static (type, GST_TYPE_PROPERTY_PROBE,
|
|
&probe_iface_info);
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_base_init (gpointer g_class)
|
|
{
|
|
static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-raw-int, "
|
|
"endianness = (int) { " ENDIANNESS " }, "
|
|
"signed = (boolean) TRUE, "
|
|
"width = (int) 16, "
|
|
"depth = (int) 16, "
|
|
"rate = (int) [ 1, MAX ], "
|
|
"channels = (int) [ 1, 32 ];"
|
|
"audio/x-raw-float, "
|
|
"endianness = (int) { " ENDIANNESS " }, "
|
|
"width = (int) 32, "
|
|
"rate = (int) [ 1, MAX ], "
|
|
"channels = (int) [ 1, 32 ];"
|
|
"audio/x-raw-int, "
|
|
"endianness = (int) { " ENDIANNESS " }, "
|
|
"signed = (boolean) TRUE, "
|
|
"width = (int) 32, "
|
|
"depth = (int) 32, "
|
|
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 32 ];"
|
|
#if HAVE_PULSE_0_9_15
|
|
"audio/x-raw-int, "
|
|
"endianness = (int) { " ENDIANNESS " }, "
|
|
"signed = (boolean) TRUE, "
|
|
"width = (int) 24, "
|
|
"depth = (int) 24, "
|
|
"rate = (int) [ 1, MAX ], "
|
|
"channels = (int) [ 1, 32 ];"
|
|
"audio/x-raw-int, "
|
|
"endianness = (int) { " ENDIANNESS " }, "
|
|
"signed = (boolean) TRUE, "
|
|
"width = (int) 32, "
|
|
"depth = (int) 24, "
|
|
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 32 ];"
|
|
#endif
|
|
"audio/x-raw-int, "
|
|
"signed = (boolean) FALSE, "
|
|
"width = (int) 8, "
|
|
"depth = (int) 8, "
|
|
"rate = (int) [ 1, MAX ], "
|
|
"channels = (int) [ 1, 32 ];"
|
|
"audio/x-alaw, "
|
|
"rate = (int) [ 1, MAX], "
|
|
"channels = (int) [ 1, 32 ];"
|
|
"audio/x-mulaw, "
|
|
"rate = (int) [ 1, MAX], " "channels = (int) [ 1, 32 ]")
|
|
);
|
|
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
|
|
|
|
gst_element_class_set_details_simple (element_class,
|
|
"PulseAudio Audio Sink",
|
|
"Sink/Audio", "Plays audio to a PulseAudio server", "Lennart Poettering");
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&pad_template));
|
|
}
|
|
|
|
static GstRingBuffer *
|
|
gst_pulsesink_create_ringbuffer (GstBaseAudioSink * sink)
|
|
{
|
|
GstRingBuffer *buffer;
|
|
|
|
GST_DEBUG_OBJECT (sink, "creating ringbuffer");
|
|
buffer = g_object_new (GST_TYPE_PULSERING_BUFFER, NULL);
|
|
GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
|
|
|
|
return buffer;
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_class_init (GstPulseSinkClass * klass)
|
|
{
|
|
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
|
|
GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
|
|
GstBaseSinkClass *bc;
|
|
GstBaseAudioSinkClass *gstaudiosink_class = GST_BASE_AUDIO_SINK_CLASS (klass);
|
|
|
|
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_pulsesink_finalize);
|
|
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_pulsesink_set_property);
|
|
gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_pulsesink_get_property);
|
|
|
|
gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_pulsesink_event);
|
|
|
|
/* restore the original basesink pull methods */
|
|
bc = g_type_class_peek (GST_TYPE_BASE_SINK);
|
|
gstbasesink_class->activate_pull = GST_DEBUG_FUNCPTR (bc->activate_pull);
|
|
|
|
gstaudiosink_class->create_ringbuffer =
|
|
GST_DEBUG_FUNCPTR (gst_pulsesink_create_ringbuffer);
|
|
|
|
/* Overwrite GObject fields */
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_SERVER,
|
|
g_param_spec_string ("server", "Server",
|
|
"The PulseAudio server to connect to", DEFAULT_SERVER,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_DEVICE,
|
|
g_param_spec_string ("device", "Device",
|
|
"The PulseAudio sink device to connect to", DEFAULT_DEVICE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_DEVICE_NAME,
|
|
g_param_spec_string ("device-name", "Device name",
|
|
"Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
#if HAVE_PULSE_0_9_12
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_VOLUME,
|
|
g_param_spec_double ("volume", "Volume",
|
|
"Linear volume of this stream, 1.0=100%", 0.0, MAX_VOLUME,
|
|
DEFAULT_VOLUME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_MUTE,
|
|
g_param_spec_boolean ("mute", "Mute",
|
|
"Mute state of this stream", DEFAULT_MUTE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
#endif
|
|
}
|
|
|
|
/* returns the current time of the sink ringbuffer */
|
|
static GstClockTime
|
|
gst_pulsesink_get_time (GstClock * clock, GstBaseAudioSink * sink)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
pa_usec_t time;
|
|
|
|
if (!sink->ringbuffer || !sink->ringbuffer->acquired)
|
|
return GST_CLOCK_TIME_NONE;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (sink->ringbuffer);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
if (gst_pulsering_is_dead (psink, pbuf))
|
|
goto server_dead;
|
|
|
|
/* if we don't have enough data to get a timestamp, just return NONE, which
|
|
* will return the last reported time */
|
|
if (pa_stream_get_time (pbuf->stream, &time) < 0) {
|
|
GST_DEBUG_OBJECT (psink, "could not get time");
|
|
time = GST_CLOCK_TIME_NONE;
|
|
} else
|
|
time *= 1000;
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
GST_LOG_OBJECT (psink, "current time is %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (time));
|
|
|
|
return time;
|
|
|
|
/* ERRORS */
|
|
server_dead:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "the server is dead");
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
return GST_CLOCK_TIME_NONE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_init (GstPulseSink * pulsesink, GstPulseSinkClass * klass)
|
|
{
|
|
guint res;
|
|
|
|
pulsesink->server = NULL;
|
|
pulsesink->device = NULL;
|
|
pulsesink->device_description = NULL;
|
|
|
|
pulsesink->volume = DEFAULT_VOLUME;
|
|
pulsesink->volume_set = FALSE;
|
|
|
|
pulsesink->mute = DEFAULT_MUTE;
|
|
pulsesink->mute_set = FALSE;
|
|
|
|
pulsesink->notify = 0;
|
|
|
|
/* needed for conditional execution */
|
|
pulsesink->pa_version = pa_get_library_version ();
|
|
|
|
GST_DEBUG_OBJECT (pulsesink, "using pulseaudio version %s",
|
|
pulsesink->pa_version);
|
|
|
|
pulsesink->mainloop = pa_threaded_mainloop_new ();
|
|
g_assert (pulsesink->mainloop != NULL);
|
|
res = pa_threaded_mainloop_start (pulsesink->mainloop);
|
|
g_assert (res == 0);
|
|
|
|
/* TRUE for sinks, FALSE for sources */
|
|
pulsesink->probe = gst_pulseprobe_new (G_OBJECT (pulsesink),
|
|
G_OBJECT_GET_CLASS (pulsesink), PROP_DEVICE, pulsesink->device,
|
|
TRUE, FALSE);
|
|
|
|
/* override with a custom clock */
|
|
if (GST_BASE_AUDIO_SINK (pulsesink)->provided_clock)
|
|
gst_object_unref (GST_BASE_AUDIO_SINK (pulsesink)->provided_clock);
|
|
GST_BASE_AUDIO_SINK (pulsesink)->provided_clock =
|
|
gst_audio_clock_new ("GstPulseSinkClock",
|
|
(GstAudioClockGetTimeFunc) gst_pulsesink_get_time, pulsesink);
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_finalize (GObject * object)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
|
|
|
|
pa_threaded_mainloop_stop (pulsesink->mainloop);
|
|
|
|
g_free (pulsesink->server);
|
|
g_free (pulsesink->device);
|
|
g_free (pulsesink->device_description);
|
|
|
|
pa_threaded_mainloop_free (pulsesink->mainloop);
|
|
|
|
if (pulsesink->probe) {
|
|
gst_pulseprobe_free (pulsesink->probe);
|
|
pulsesink->probe = NULL;
|
|
}
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
#if HAVE_PULSE_0_9_12
|
|
static void
|
|
gst_pulsesink_set_volume (GstPulseSink * psink, gdouble volume)
|
|
{
|
|
pa_cvolume v;
|
|
pa_operation *o = NULL;
|
|
GstPulseRingBuffer *pbuf;
|
|
uint32_t idx;
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
|
|
GST_DEBUG_OBJECT (psink, "setting volume to %f", volume);
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
|
|
if (pbuf == NULL || pbuf->stream == NULL)
|
|
goto no_buffer;
|
|
|
|
if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
|
|
goto no_index;
|
|
|
|
gst_pulse_cvolume_from_linear (&v, pbuf->sample_spec.channels, volume);
|
|
|
|
if (!(o = pa_context_set_sink_input_volume (pbuf->context, idx,
|
|
&v, NULL, NULL)))
|
|
goto volume_failed;
|
|
|
|
/* We don't really care about the result of this call */
|
|
unlock:
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_buffer:
|
|
{
|
|
psink->volume = volume;
|
|
psink->volume_set = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
|
|
goto unlock;
|
|
}
|
|
no_index:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "we don't have a stream index");
|
|
goto unlock;
|
|
}
|
|
volume_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("pa_stream_set_sink_input_volume() failed: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_set_mute (GstPulseSink * psink, gboolean mute)
|
|
{
|
|
pa_operation *o = NULL;
|
|
GstPulseRingBuffer *pbuf;
|
|
uint32_t idx;
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
|
|
GST_DEBUG_OBJECT (psink, "setting mute state to %d", mute);
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
|
|
if (pbuf == NULL || pbuf->stream == NULL)
|
|
goto no_buffer;
|
|
|
|
if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
|
|
goto no_index;
|
|
|
|
if (!(o = pa_context_set_sink_input_mute (pbuf->context, idx,
|
|
mute, NULL, NULL)))
|
|
goto mute_failed;
|
|
|
|
/* We don't really care about the result of this call */
|
|
unlock:
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_buffer:
|
|
{
|
|
psink->mute = mute;
|
|
psink->mute_set = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
|
|
goto unlock;
|
|
}
|
|
no_index:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "we don't have a stream index");
|
|
goto unlock;
|
|
}
|
|
mute_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("pa_stream_set_sink_input_mute() failed: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_sink_input_info_cb (pa_context * c, const pa_sink_input_info * i,
|
|
int eol, void *userdata)
|
|
{
|
|
GstPulseRingBuffer *pbuf;
|
|
GstPulseSink *psink;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
if (!i)
|
|
goto done;
|
|
|
|
if (!pbuf->stream)
|
|
goto done;
|
|
|
|
/* If the index doesn't match our current stream,
|
|
* it implies we just recreated the stream (caps change)
|
|
*/
|
|
if (i->index == pa_stream_get_index (pbuf->stream)) {
|
|
psink->volume = pa_sw_volume_to_linear (pa_cvolume_max (&i->volume));
|
|
psink->mute = i->mute;
|
|
}
|
|
|
|
done:
|
|
pa_threaded_mainloop_signal (psink->mainloop, 0);
|
|
}
|
|
|
|
static gdouble
|
|
gst_pulsesink_get_volume (GstPulseSink * psink)
|
|
{
|
|
GstPulseRingBuffer *pbuf;
|
|
pa_operation *o = NULL;
|
|
gdouble v = DEFAULT_VOLUME;
|
|
uint32_t idx;
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
|
|
if (pbuf == NULL || pbuf->stream == NULL)
|
|
goto no_buffer;
|
|
|
|
if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
|
|
goto no_index;
|
|
|
|
if (!(o = pa_context_get_sink_input_info (pbuf->context, idx,
|
|
gst_pulsesink_sink_input_info_cb, pbuf)))
|
|
goto info_failed;
|
|
|
|
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
|
|
pa_threaded_mainloop_wait (psink->mainloop);
|
|
if (gst_pulsering_is_dead (psink, pbuf))
|
|
goto unlock;
|
|
}
|
|
|
|
v = psink->volume;
|
|
|
|
unlock:
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
if (v > MAX_VOLUME) {
|
|
GST_WARNING_OBJECT (psink, "Clipped volume from %f to %f", v, MAX_VOLUME);
|
|
v = MAX_VOLUME;
|
|
}
|
|
|
|
return v;
|
|
|
|
/* ERRORS */
|
|
no_buffer:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
|
|
goto unlock;
|
|
}
|
|
no_index:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "we don't have a stream index");
|
|
goto unlock;
|
|
}
|
|
info_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("pa_context_get_sink_input_info() failed: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_pulsesink_get_mute (GstPulseSink * psink)
|
|
{
|
|
GstPulseRingBuffer *pbuf;
|
|
pa_operation *o = NULL;
|
|
uint32_t idx;
|
|
gboolean mute = FALSE;
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
|
|
if (pbuf == NULL || pbuf->stream == NULL)
|
|
goto no_buffer;
|
|
|
|
if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
|
|
goto no_index;
|
|
|
|
if (!(o = pa_context_get_sink_input_info (pbuf->context, idx,
|
|
gst_pulsesink_sink_input_info_cb, pbuf)))
|
|
goto info_failed;
|
|
|
|
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
|
|
pa_threaded_mainloop_wait (psink->mainloop);
|
|
if (gst_pulsering_is_dead (psink, pbuf))
|
|
goto unlock;
|
|
}
|
|
|
|
mute = psink->mute;
|
|
|
|
unlock:
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
return mute;
|
|
|
|
/* ERRORS */
|
|
no_buffer:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
|
|
goto unlock;
|
|
}
|
|
no_index:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "we don't have a stream index");
|
|
goto unlock;
|
|
}
|
|
info_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("pa_context_get_sink_input_info() failed: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto unlock;
|
|
}
|
|
}
|
|
#endif
|
|
|
|
static void
|
|
gst_pulsesink_sink_info_cb (pa_context * c, const pa_sink_info * i, int eol,
|
|
void *userdata)
|
|
{
|
|
GstPulseRingBuffer *pbuf;
|
|
GstPulseSink *psink;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
if (!i)
|
|
goto done;
|
|
|
|
if (!pbuf->stream)
|
|
goto done;
|
|
|
|
g_assert (i->index == pa_stream_get_device_index (pbuf->stream));
|
|
|
|
g_free (psink->device_description);
|
|
psink->device_description = g_strdup (i->description);
|
|
|
|
done:
|
|
pa_threaded_mainloop_signal (psink->mainloop, 0);
|
|
}
|
|
|
|
static gchar *
|
|
gst_pulsesink_device_description (GstPulseSink * psink)
|
|
{
|
|
GstPulseRingBuffer *pbuf;
|
|
pa_operation *o = NULL;
|
|
gchar *t;
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
|
|
if (pbuf == NULL || pbuf->stream == NULL)
|
|
goto no_buffer;
|
|
|
|
if (!(o = pa_context_get_sink_info_by_index (pbuf->context,
|
|
pa_stream_get_device_index (pbuf->stream),
|
|
gst_pulsesink_sink_info_cb, pbuf)))
|
|
goto info_failed;
|
|
|
|
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
|
|
pa_threaded_mainloop_wait (psink->mainloop);
|
|
if (gst_pulsering_is_dead (psink, pbuf))
|
|
goto unlock;
|
|
}
|
|
|
|
unlock:
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
t = g_strdup (psink->device_description);
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
return t;
|
|
|
|
/* ERRORS */
|
|
no_buffer:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
|
|
goto unlock;
|
|
}
|
|
info_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("pa_context_get_sink_info_by_index() failed: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_set_property (GObject * object,
|
|
guint prop_id, const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_SERVER:
|
|
g_free (pulsesink->server);
|
|
pulsesink->server = g_value_dup_string (value);
|
|
if (pulsesink->probe)
|
|
gst_pulseprobe_set_server (pulsesink->probe, pulsesink->server);
|
|
break;
|
|
case PROP_DEVICE:
|
|
g_free (pulsesink->device);
|
|
pulsesink->device = g_value_dup_string (value);
|
|
break;
|
|
#if HAVE_PULSE_0_9_12
|
|
case PROP_VOLUME:
|
|
gst_pulsesink_set_volume (pulsesink, g_value_get_double (value));
|
|
break;
|
|
case PROP_MUTE:
|
|
gst_pulsesink_set_mute (pulsesink, g_value_get_boolean (value));
|
|
break;
|
|
#endif
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_get_property (GObject * object,
|
|
guint prop_id, GValue * value, GParamSpec * pspec)
|
|
{
|
|
|
|
GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_SERVER:
|
|
g_value_set_string (value, pulsesink->server);
|
|
break;
|
|
case PROP_DEVICE:
|
|
g_value_set_string (value, pulsesink->device);
|
|
break;
|
|
case PROP_DEVICE_NAME:
|
|
g_value_take_string (value, gst_pulsesink_device_description (pulsesink));
|
|
break;
|
|
#if HAVE_PULSE_0_9_12
|
|
case PROP_VOLUME:
|
|
g_value_set_double (value, gst_pulsesink_get_volume (pulsesink));
|
|
break;
|
|
case PROP_MUTE:
|
|
g_value_set_boolean (value, gst_pulsesink_get_mute (pulsesink));
|
|
break;
|
|
#endif
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_change_title (GstPulseSink * psink, const gchar * t)
|
|
{
|
|
pa_operation *o = NULL;
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
|
|
|
|
if (pbuf == NULL || pbuf->stream == NULL)
|
|
goto no_buffer;
|
|
|
|
g_free (pbuf->stream_name);
|
|
pbuf->stream_name = g_strdup (t);
|
|
|
|
if (!(o = pa_stream_set_name (pbuf->stream, pbuf->stream_name, NULL, NULL)))
|
|
goto name_failed;
|
|
|
|
/* We're not interested if this operation failed or not */
|
|
unlock:
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_buffer:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
|
|
goto unlock;
|
|
}
|
|
name_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("pa_stream_set_name() failed: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
#if HAVE_PULSE_0_9_11
|
|
static void
|
|
gst_pulsesink_change_props (GstPulseSink * psink, GstTagList * l)
|
|
{
|
|
static const gchar *const map[] = {
|
|
GST_TAG_TITLE, PA_PROP_MEDIA_TITLE,
|
|
|
|
/* might get overriden in the next iteration by GST_TAG_ARTIST */
|
|
GST_TAG_PERFORMER, PA_PROP_MEDIA_ARTIST,
|
|
|
|
GST_TAG_ARTIST, PA_PROP_MEDIA_ARTIST,
|
|
GST_TAG_LANGUAGE_CODE, PA_PROP_MEDIA_LANGUAGE,
|
|
GST_TAG_LOCATION, PA_PROP_MEDIA_FILENAME,
|
|
/* We might add more here later on ... */
|
|
NULL
|
|
};
|
|
pa_proplist *pl = NULL;
|
|
const gchar *const *t;
|
|
gboolean empty = TRUE;
|
|
pa_operation *o = NULL;
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
pl = pa_proplist_new ();
|
|
|
|
for (t = map; *t; t += 2) {
|
|
gchar *n = NULL;
|
|
|
|
if (gst_tag_list_get_string (l, *t, &n)) {
|
|
|
|
if (n && *n) {
|
|
pa_proplist_sets (pl, *(t + 1), n);
|
|
empty = FALSE;
|
|
}
|
|
|
|
g_free (n);
|
|
}
|
|
}
|
|
if (empty)
|
|
goto finish;
|
|
|
|
pa_threaded_mainloop_lock (psink->mainloop);
|
|
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
|
|
if (pbuf == NULL || pbuf->stream == NULL)
|
|
goto no_buffer;
|
|
|
|
if (!(o = pa_stream_proplist_update (pbuf->stream, PA_UPDATE_REPLACE,
|
|
pl, NULL, NULL)))
|
|
goto update_failed;
|
|
|
|
/* We're not interested if this operation failed or not */
|
|
unlock:
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (psink->mainloop);
|
|
|
|
finish:
|
|
|
|
if (pl)
|
|
pa_proplist_free (pl);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_buffer:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
|
|
goto unlock;
|
|
}
|
|
update_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("pa_stream_proplist_update() failed: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto unlock;
|
|
}
|
|
}
|
|
#endif
|
|
|
|
static gboolean
|
|
gst_pulsesink_event (GstBaseSink * sink, GstEvent * event)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK_CAST (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 (!artist)
|
|
gst_tag_list_get_string (l, GST_TAG_PERFORMER, &artist);
|
|
|
|
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);
|
|
|
|
#if HAVE_PULSE_0_9_11
|
|
gst_pulsesink_change_props (pulsesink, l);
|
|
#endif
|
|
|
|
break;
|
|
}
|
|
default:
|
|
;
|
|
}
|
|
|
|
return GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
|
|
}
|