mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-27 02:30:35 +00:00
11bcac7c90
This reverts commit 1e2c1467ae.
The commit causes pulsesink to ignore the latency-time baseaudiosink property.
2815 lines
79 KiB
C
2815 lines
79 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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* |[
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* gst-launch -v audiotestsrc ! pulsesink stream-properties="props,media.title=test"
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* ]| Play a sine wave and set a stream property. The property can be checked
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* with "pactl list".
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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 <gst/pbutils/pbutils.h> /* only used for GST_PLUGINS_BASE_VERSION_* */
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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_CLIENT,
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PROP_STREAM_PROPERTIES,
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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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typedef struct _GstPulseContext GstPulseContext;
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/* Store the PA contexts in a hash table to allow easy sharing among
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* multiple instances of the sink. Keys are $context_name@$server_name
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* (strings) and values should be GstPulseContext pointers.
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*/
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struct _GstPulseContext
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{
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pa_context *context;
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GSList *ring_buffers;
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};
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static GHashTable *gst_pulse_shared_contexts = NULL;
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/* use one static main-loop for all instances
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* this is needed to make the context sharing work as the contexts are
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* released when releasing their parent main-loop
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*/
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static pa_threaded_mainloop *mainloop = NULL;
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static guint mainloop_ref_ct = 0;
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/* lock for access to shared resources */
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static GMutex *pa_shared_resource_mutex = NULL;
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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 *context_name;
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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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#ifdef HAVE_PULSE_0_9_16
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void *m_data;
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size_t m_towrite;
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size_t m_writable;
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gint64 m_offset;
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gint64 m_lastoffset;
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#endif
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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 GType gst_pulseringbuffer_get_type (void);
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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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G_DEFINE_TYPE (GstPulseRingBuffer, gst_pulseringbuffer, GST_TYPE_RING_BUFFER);
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static void
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gst_pulsesink_init_contexts (void)
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{
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g_assert (pa_shared_resource_mutex == NULL);
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pa_shared_resource_mutex = g_mutex_new ();
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gst_pulse_shared_contexts = g_hash_table_new_full (g_str_hash, g_str_equal,
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g_free, NULL);
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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_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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}
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static void
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gst_pulseringbuffer_init (GstPulseRingBuffer * pbuf)
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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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#ifdef 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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#ifdef HAVE_PULSE_0_9_16
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pbuf->m_data = NULL;
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pbuf->m_towrite = 0;
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pbuf->m_writable = 0;
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pbuf->m_offset = 0;
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pbuf->m_lastoffset = 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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#ifdef HAVE_PULSE_0_9_16
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if (pbuf->m_data) {
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/* drop shm memory buffer */
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pa_stream_cancel_write (pbuf->stream);
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/* reset internal variables */
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pbuf->m_data = NULL;
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pbuf->m_towrite = 0;
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pbuf->m_writable = 0;
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pbuf->m_offset = 0;
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pbuf->m_lastoffset = 0;
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}
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#endif
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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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g_mutex_lock (pa_shared_resource_mutex);
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GST_DEBUG_OBJECT (pbuf, "destroying ringbuffer %p", pbuf);
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gst_pulsering_destroy_stream (pbuf);
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if (pbuf->context) {
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pa_context_unref (pbuf->context);
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pbuf->context = NULL;
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}
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if (pbuf->context_name) {
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GstPulseContext *pctx;
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pctx = g_hash_table_lookup (gst_pulse_shared_contexts, pbuf->context_name);
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GST_DEBUG_OBJECT (pbuf, "releasing context with name %s, pbuf=%p, pctx=%p",
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pbuf->context_name, pbuf, pctx);
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if (pctx) {
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pctx->ring_buffers = g_slist_remove (pctx->ring_buffers, pbuf);
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if (pctx->ring_buffers == NULL) {
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GST_DEBUG_OBJECT (pbuf,
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"destroying final context with name %s, pbuf=%p, pctx=%p",
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pbuf->context_name, pbuf, pctx);
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pa_context_disconnect (pctx->context);
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/* Make sure we don't get any further callbacks */
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pa_context_set_state_callback (pctx->context, NULL, NULL);
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#ifdef HAVE_PULSE_0_9_12
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pa_context_set_subscribe_callback (pctx->context, NULL, NULL);
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#endif
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g_hash_table_remove (gst_pulse_shared_contexts, pbuf->context_name);
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pa_context_unref (pctx->context);
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g_slice_free (GstPulseContext, pctx);
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}
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}
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g_free (pbuf->context_name);
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pbuf->context_name = NULL;
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}
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g_mutex_unlock (pa_shared_resource_mutex);
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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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#define CONTEXT_OK(c) ((c) && PA_CONTEXT_IS_GOOD (pa_context_get_state ((c))))
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#define STREAM_OK(s) ((s) && PA_STREAM_IS_GOOD (pa_stream_get_state ((s))))
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static gboolean
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gst_pulsering_is_dead (GstPulseSink * psink, GstPulseRingBuffer * pbuf,
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gboolean check_stream)
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{
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if (!CONTEXT_OK (pbuf->context))
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goto error;
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if (check_stream && !STREAM_OK (pbuf->stream))
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goto error;
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return FALSE;
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error:
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{
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const gchar *err_str =
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pbuf->context ? 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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}
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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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pa_context_state_t state;
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pa_threaded_mainloop *mainloop = (pa_threaded_mainloop *) userdata;
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state = pa_context_get_state (c);
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GST_LOG ("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 ("signaling");
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pa_threaded_mainloop_signal (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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#ifdef 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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GstPulseContext *pctx = (GstPulseContext *) userdata;
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GSList *walk;
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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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for (walk = pctx->ring_buffers; walk; walk = g_slist_next (walk)) {
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GstPulseRingBuffer *pbuf = (GstPulseRingBuffer *) walk->data;
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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 (!pbuf->stream)
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continue;
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if (idx != pa_stream_get_index (pbuf->stream))
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continue;
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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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}
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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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GstPulseContext *pctx;
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pa_mainloop_api *api;
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gboolean need_unlock_shared;
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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->stream);
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g_assert (psink->client_name);
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if (psink->server)
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pbuf->context_name = g_strdup_printf ("%s@%s", psink->client_name,
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psink->server);
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else
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pbuf->context_name = g_strdup (psink->client_name);
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pa_threaded_mainloop_lock (mainloop);
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g_mutex_lock (pa_shared_resource_mutex);
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need_unlock_shared = TRUE;
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pctx = g_hash_table_lookup (gst_pulse_shared_contexts, pbuf->context_name);
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if (pctx == NULL) {
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pctx = g_slice_new0 (GstPulseContext);
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/* get the mainloop api and create a context */
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GST_INFO_OBJECT (psink, "new context with name %s, pbuf=%p, pctx=%p",
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pbuf->context_name, pbuf, pctx);
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api = pa_threaded_mainloop_get_api (mainloop);
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if (!(pctx->context = pa_context_new (api, pbuf->context_name)))
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goto create_failed;
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pctx->ring_buffers = g_slist_prepend (pctx->ring_buffers, pbuf);
|
|
g_hash_table_insert (gst_pulse_shared_contexts,
|
|
g_strdup (pbuf->context_name), (gpointer) pctx);
|
|
/* register some essential callbacks */
|
|
pa_context_set_state_callback (pctx->context,
|
|
gst_pulsering_context_state_cb, mainloop);
|
|
#ifdef HAVE_PULSE_0_9_12
|
|
pa_context_set_subscribe_callback (pctx->context,
|
|
gst_pulsering_context_subscribe_cb, pctx);
|
|
#endif
|
|
|
|
/* try to connect to the server and wait for completion, we don't want to
|
|
* autospawn a deamon */
|
|
GST_LOG_OBJECT (psink, "connect to server %s",
|
|
GST_STR_NULL (psink->server));
|
|
if (pa_context_connect (pctx->context, psink->server,
|
|
PA_CONTEXT_NOAUTOSPAWN, NULL) < 0)
|
|
goto connect_failed;
|
|
} else {
|
|
GST_INFO_OBJECT (psink,
|
|
"reusing shared context with name %s, pbuf=%p, pctx=%p",
|
|
pbuf->context_name, pbuf, pctx);
|
|
pctx->ring_buffers = g_slist_prepend (pctx->ring_buffers, pbuf);
|
|
}
|
|
|
|
g_mutex_unlock (pa_shared_resource_mutex);
|
|
need_unlock_shared = FALSE;
|
|
|
|
/* context created or shared okay */
|
|
pbuf->context = pa_context_ref (pctx->context);
|
|
|
|
for (;;) {
|
|
pa_context_state_t state;
|
|
|
|
state = pa_context_get_state (pbuf->context);
|
|
|
|
GST_LOG_OBJECT (psink, "context state is now %d", state);
|
|
|
|
if (!PA_CONTEXT_IS_GOOD (state))
|
|
goto connect_failed;
|
|
|
|
if (state == PA_CONTEXT_READY)
|
|
break;
|
|
|
|
/* Wait until the context is ready */
|
|
GST_LOG_OBJECT (psink, "waiting..");
|
|
pa_threaded_mainloop_wait (mainloop);
|
|
}
|
|
|
|
GST_LOG_OBJECT (psink, "opened the device");
|
|
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
unlock_and_fail:
|
|
{
|
|
if (need_unlock_shared)
|
|
g_mutex_unlock (pa_shared_resource_mutex);
|
|
gst_pulsering_destroy_context (pbuf);
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
return FALSE;
|
|
}
|
|
create_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("Failed to create context"), (NULL));
|
|
g_slice_free (GstPulseContext, pctx);
|
|
goto unlock_and_fail;
|
|
}
|
|
connect_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED, ("Failed to connect: %s",
|
|
pa_strerror (pa_context_errno (pctx->context))), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
}
|
|
|
|
/* close the device */
|
|
static gboolean
|
|
gst_pulseringbuffer_close_device (GstRingBuffer * buf)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (buf);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
|
|
|
|
GST_LOG_OBJECT (psink, "closing device");
|
|
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
gst_pulsering_destroy_context (pbuf);
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
|
|
GST_LOG_OBJECT (psink, "closed device");
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_pulsering_stream_state_cb (pa_stream * s, void *userdata)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
pa_stream_state_t state;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
|
|
state = pa_stream_get_state (s);
|
|
GST_LOG_OBJECT (psink, "got new stream state %d", state);
|
|
|
|
switch (state) {
|
|
case PA_STREAM_READY:
|
|
case PA_STREAM_FAILED:
|
|
case PA_STREAM_TERMINATED:
|
|
GST_LOG_OBJECT (psink, "signaling");
|
|
pa_threaded_mainloop_signal (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 (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;
|
|
}
|
|
#ifdef 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");
|
|
}
|
|
|
|
#ifdef 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
|
|
|
|
#ifdef 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;
|
|
#ifdef 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 (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_with_proplist (pbuf->context, name,
|
|
&pbuf->sample_spec, &channel_map, psink->proplist)))
|
|
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);
|
|
#ifdef HAVE_PULSE_0_9_11
|
|
pa_stream_set_started_callback (pbuf->stream,
|
|
gst_pulsering_stream_started_cb, pbuf);
|
|
#endif
|
|
#ifdef 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);
|
|
|
|
#ifdef 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 |
|
|
#ifdef HAVE_PULSE_0_9_11
|
|
PA_STREAM_ADJUST_LATENCY |
|
|
#endif
|
|
PA_STREAM_START_CORKED;
|
|
|
|
#ifdef 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 (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 (mainloop);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
unlock_and_fail:
|
|
{
|
|
gst_pulsering_destroy_stream (pbuf);
|
|
pa_threaded_mainloop_unlock (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 (mainloop);
|
|
gst_pulsering_destroy_stream (pbuf);
|
|
pa_threaded_mainloop_unlock (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 (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 (mainloop);
|
|
if (gst_pulsering_is_dead (psink, pbuf, TRUE))
|
|
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 (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 (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 (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 (mainloop);
|
|
|
|
GST_DEBUG_OBJECT (psink, "scheduling stream status");
|
|
psink->pa_defer_ran = FALSE;
|
|
pa_mainloop_api_once (pa_threaded_mainloop_get_api (mainloop),
|
|
mainloop_enter_defer_cb, psink);
|
|
|
|
GST_DEBUG_OBJECT (psink, "starting");
|
|
pbuf->paused = FALSE;
|
|
|
|
/* EOS needs running clock */
|
|
if (GST_BASE_SINK_CAST (psink)->eos ||
|
|
g_atomic_int_get (&GST_BASE_AUDIO_SINK (psink)->abidata.
|
|
ABI.eos_rendering))
|
|
gst_pulsering_set_corked (pbuf, FALSE, FALSE);
|
|
|
|
pa_threaded_mainloop_unlock (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 (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, TRUE);
|
|
if (pbuf->in_commit) {
|
|
/* we are waiting in a commit, signal */
|
|
GST_DEBUG_OBJECT (psink, "signal commit");
|
|
pa_threaded_mainloop_signal (mainloop, 0);
|
|
}
|
|
pa_threaded_mainloop_unlock (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 (mainloop, 0);
|
|
gst_object_unref (pulsesink);
|
|
}
|
|
|
|
/* 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 (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 (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 (mainloop);
|
|
if (gst_pulsering_is_dead (psink, pbuf, TRUE))
|
|
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;
|
|
gst_object_ref (psink);
|
|
pa_mainloop_api_once (pa_threaded_mainloop_get_api (mainloop),
|
|
mainloop_leave_defer_cb, psink);
|
|
|
|
GST_DEBUG_OBJECT (psink, "waiting for stream status");
|
|
pa_threaded_mainloop_unlock (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 (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);
|
|
|
|
#ifdef HAVE_PULSE_0_9_16
|
|
if (offset != pbuf->m_lastoffset)
|
|
GST_LOG_OBJECT (psink, "discontinuity, offset is %" G_GINT64_FORMAT ", "
|
|
"last offset was %" G_GINT64_FORMAT, offset, pbuf->m_lastoffset);
|
|
|
|
towrite = out_samples * bps;
|
|
|
|
/* Only ever write segsize bytes at once. This will
|
|
* also limit the PA shm buffer to segsize
|
|
*/
|
|
if (towrite > buf->spec.segsize)
|
|
towrite = buf->spec.segsize;
|
|
|
|
if ((pbuf->m_writable < towrite) || (offset != pbuf->m_lastoffset)) {
|
|
/* if no room left or discontinuity in offset,
|
|
we need to flush data and get a new buffer */
|
|
|
|
/* flush the buffer if possible */
|
|
if ((pbuf->m_data != NULL) && (pbuf->m_towrite > 0)) {
|
|
|
|
GST_LOG_OBJECT (psink,
|
|
"flushing %u samples at offset %" G_GINT64_FORMAT,
|
|
(guint) pbuf->m_towrite / bps, pbuf->m_offset);
|
|
|
|
if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
|
|
pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) {
|
|
goto write_failed;
|
|
}
|
|
}
|
|
pbuf->m_towrite = 0;
|
|
pbuf->m_offset = offset; /* keep track of current offset */
|
|
|
|
/* get a buffer to write in for now on */
|
|
for (;;) {
|
|
pbuf->m_writable = pa_stream_writable_size (pbuf->stream);
|
|
|
|
if (pbuf->m_writable == (size_t) - 1)
|
|
goto writable_size_failed;
|
|
|
|
pbuf->m_writable /= bps;
|
|
pbuf->m_writable *= bps; /* handle only complete samples */
|
|
|
|
if (pbuf->m_writable >= towrite)
|
|
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 (mainloop);
|
|
|
|
if (pbuf->paused)
|
|
goto was_paused;
|
|
}
|
|
|
|
/* make sure we only buffer up latency-time samples */
|
|
if (pbuf->m_writable > buf->spec.segsize) {
|
|
/* limit buffering to latency-time value */
|
|
pbuf->m_writable = buf->spec.segsize;
|
|
|
|
GST_LOG_OBJECT (psink, "Limiting buffering to %" G_GSIZE_FORMAT,
|
|
pbuf->m_writable);
|
|
}
|
|
|
|
GST_LOG_OBJECT (psink, "requesting %" G_GSIZE_FORMAT " bytes of "
|
|
"shared memory", pbuf->m_writable);
|
|
|
|
if (pa_stream_begin_write (pbuf->stream, &pbuf->m_data,
|
|
&pbuf->m_writable) < 0) {
|
|
GST_LOG_OBJECT (psink, "pa_stream_begin_write() failed");
|
|
goto writable_size_failed;
|
|
}
|
|
|
|
GST_LOG_OBJECT (psink, "got %" G_GSIZE_FORMAT " bytes of shared memory",
|
|
pbuf->m_writable);
|
|
|
|
/* Just to make sure that we didn't get more than requested */
|
|
if (pbuf->m_writable > buf->spec.segsize) {
|
|
/* limit buffering to latency-time value */
|
|
pbuf->m_writable = buf->spec.segsize;
|
|
}
|
|
}
|
|
|
|
if (pbuf->m_writable < towrite)
|
|
towrite = pbuf->m_writable;
|
|
avail = towrite / 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 */
|
|
/* copy the data into internal buffer */
|
|
|
|
memcpy ((guint8 *) pbuf->m_data + pbuf->m_towrite, data, towrite);
|
|
pbuf->m_towrite += towrite;
|
|
pbuf->m_writable -= towrite;
|
|
|
|
data += towrite;
|
|
in_samples -= avail;
|
|
out_samples -= avail;
|
|
} else {
|
|
guint8 *dest, *d, *d_end;
|
|
|
|
/* write into the PulseAudio shm buffer */
|
|
dest = d = (guint8 *) pbuf->m_data + pbuf->m_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);
|
|
pbuf->m_towrite += towrite;
|
|
pbuf->m_writable -= towrite;
|
|
|
|
avail = towrite / bps;
|
|
}
|
|
|
|
/* flush the buffer if it's full */
|
|
if ((pbuf->m_data != NULL) && (pbuf->m_towrite > 0)
|
|
&& (pbuf->m_writable == 0)) {
|
|
GST_LOG_OBJECT (psink, "flushing %u samples at offset %" G_GINT64_FORMAT,
|
|
(guint) pbuf->m_towrite / bps, pbuf->m_offset);
|
|
|
|
if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
|
|
pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) {
|
|
goto write_failed;
|
|
}
|
|
pbuf->m_towrite = 0;
|
|
pbuf->m_offset = offset + towrite; /* keep track of current offset */
|
|
}
|
|
#else
|
|
|
|
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 (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;
|
|
}
|
|
#endif /* HAVE_PULSE_0_9_16 */
|
|
|
|
*sample += avail;
|
|
offset += avail * bps;
|
|
|
|
#ifdef HAVE_PULSE_0_9_16
|
|
pbuf->m_lastoffset = offset;
|
|
#endif
|
|
|
|
/* 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 (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 (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 (mainloop);
|
|
goto done;
|
|
}
|
|
was_paused:
|
|
{
|
|
pbuf->in_commit = FALSE;
|
|
GST_LOG_OBJECT (psink, "we are paused");
|
|
pa_threaded_mainloop_unlock (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;
|
|
}
|
|
}
|
|
|
|
/* write pending local samples, must be called with the mainloop lock */
|
|
static void
|
|
gst_pulsering_flush (GstPulseRingBuffer * pbuf)
|
|
{
|
|
#ifdef HAVE_PULSE_0_9_16
|
|
GstPulseSink *psink;
|
|
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
GST_DEBUG_OBJECT (psink, "entering flush");
|
|
|
|
/* flush the buffer if possible */
|
|
if (pbuf->stream && (pbuf->m_data != NULL) && (pbuf->m_towrite > 0)) {
|
|
#ifndef GST_DISABLE_GST_DEBUG
|
|
gint bps;
|
|
|
|
bps = (GST_RING_BUFFER_CAST (pbuf))->spec.bytes_per_sample;
|
|
GST_LOG_OBJECT (psink,
|
|
"flushing %u samples at offset %" G_GINT64_FORMAT,
|
|
(guint) pbuf->m_towrite / bps, pbuf->m_offset);
|
|
#endif
|
|
|
|
if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
|
|
pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) {
|
|
goto write_failed;
|
|
}
|
|
|
|
pbuf->m_towrite = 0;
|
|
pbuf->m_offset += pbuf->m_towrite; /* keep track of current offset */
|
|
}
|
|
|
|
done:
|
|
return;
|
|
|
|
/* ERRORS */
|
|
write_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("pa_stream_write() failed: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto done;
|
|
}
|
|
#endif
|
|
}
|
|
|
|
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 GstStateChangeReturn gst_pulsesink_change_state (GstElement * element,
|
|
GstStateChange transition);
|
|
|
|
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);
|
|
|
|
#define _do_init(type) \
|
|
gst_pulsesink_init_contexts (); \
|
|
gst_pulsesink_init_interfaces (type);
|
|
|
|
GST_BOILERPLATE_FULL (GstPulseSink, gst_pulsesink, GstBaseAudioSink,
|
|
GST_TYPE_BASE_AUDIO_SINK, _do_init);
|
|
|
|
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,
|
|
};
|
|
#ifdef 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 ];"
|
|
#ifdef 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);
|
|
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
|
|
|
|
gobject_class->finalize = gst_pulsesink_finalize;
|
|
gobject_class->set_property = gst_pulsesink_set_property;
|
|
gobject_class->get_property = 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);
|
|
|
|
gstelement_class->change_state =
|
|
GST_DEBUG_FUNCPTR (gst_pulsesink_change_state);
|
|
|
|
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));
|
|
|
|
#ifdef 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
|
|
|
|
/**
|
|
* GstPulseSink:client
|
|
*
|
|
* The PulseAudio client name to use.
|
|
*
|
|
* Since: 0.10.25
|
|
*/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_CLIENT,
|
|
g_param_spec_string ("client", "Client",
|
|
"The PulseAudio client name to use", gst_pulse_client_name (),
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
|
|
GST_PARAM_MUTABLE_READY));
|
|
|
|
/**
|
|
* GstPulseSink:stream-properties
|
|
*
|
|
* List of pulseaudio stream properties. A list of defined properties can be
|
|
* found in the <ulink url="http://0pointer.de/lennart/projects/pulseaudio/doxygen/proplist_8h.html">pulseaudio api docs</ulink>.
|
|
*
|
|
* Below is an example for registering as a music application to pulseaudio.
|
|
* |[
|
|
* GstStructure *props;
|
|
*
|
|
* props = gst_structure_from_string ("props,media.role=music", NULL);
|
|
* g_object_set (pulse, "stream-properties", props, NULL);
|
|
* gst_structure_free
|
|
* ]|
|
|
*
|
|
* Since: 0.10.26
|
|
*/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_STREAM_PROPERTIES,
|
|
g_param_spec_boxed ("stream-properties", "stream properties",
|
|
"list of pulseaudio stream properties",
|
|
GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
}
|
|
|
|
/* 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 (mainloop);
|
|
if (gst_pulsering_is_dead (psink, pbuf, TRUE))
|
|
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 (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 (mainloop);
|
|
|
|
return GST_CLOCK_TIME_NONE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_init (GstPulseSink * pulsesink, GstPulseSinkClass * klass)
|
|
{
|
|
pulsesink->server = NULL;
|
|
pulsesink->device = NULL;
|
|
pulsesink->device_description = NULL;
|
|
pulsesink->client_name = gst_pulse_client_name ();
|
|
|
|
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 ();
|
|
|
|
pulsesink->properties = NULL;
|
|
pulsesink->proplist = NULL;
|
|
|
|
GST_DEBUG_OBJECT (pulsesink, "using pulseaudio version %s",
|
|
pulsesink->pa_version);
|
|
|
|
/* 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);
|
|
|
|
/* 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);
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_finalize (GObject * object)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
|
|
|
|
g_free (pulsesink->server);
|
|
g_free (pulsesink->device);
|
|
g_free (pulsesink->device_description);
|
|
g_free (pulsesink->client_name);
|
|
|
|
if (pulsesink->properties)
|
|
gst_structure_free (pulsesink->properties);
|
|
if (pulsesink->proplist)
|
|
pa_proplist_free (pulsesink->proplist);
|
|
|
|
if (pulsesink->probe) {
|
|
gst_pulseprobe_free (pulsesink->probe);
|
|
pulsesink->probe = NULL;
|
|
}
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
#ifdef 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;
|
|
|
|
if (!mainloop)
|
|
goto no_mainloop;
|
|
|
|
pa_threaded_mainloop_lock (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 (mainloop);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_mainloop:
|
|
{
|
|
psink->volume = volume;
|
|
psink->volume_set = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (psink, "we have no mainloop");
|
|
return;
|
|
}
|
|
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;
|
|
|
|
if (!mainloop)
|
|
goto no_mainloop;
|
|
|
|
pa_threaded_mainloop_lock (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 (mainloop);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_mainloop:
|
|
{
|
|
psink->mute = mute;
|
|
psink->mute_set = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (psink, "we have no mainloop");
|
|
return;
|
|
}
|
|
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 (mainloop, 0);
|
|
}
|
|
|
|
static gdouble
|
|
gst_pulsesink_get_volume (GstPulseSink * psink)
|
|
{
|
|
GstPulseRingBuffer *pbuf;
|
|
pa_operation *o = NULL;
|
|
gdouble v = DEFAULT_VOLUME;
|
|
uint32_t idx;
|
|
|
|
if (!mainloop)
|
|
goto no_mainloop;
|
|
|
|
pa_threaded_mainloop_lock (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 (mainloop);
|
|
if (gst_pulsering_is_dead (psink, pbuf, TRUE))
|
|
goto unlock;
|
|
}
|
|
|
|
unlock:
|
|
v = psink->volume;
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (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_mainloop:
|
|
{
|
|
v = psink->volume;
|
|
GST_DEBUG_OBJECT (psink, "we have no mainloop");
|
|
return v;
|
|
}
|
|
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;
|
|
|
|
if (!mainloop)
|
|
goto no_mainloop;
|
|
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
mute = psink->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_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 (mainloop);
|
|
if (gst_pulsering_is_dead (psink, pbuf, TRUE))
|
|
goto unlock;
|
|
}
|
|
|
|
unlock:
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
|
|
return mute;
|
|
|
|
/* ERRORS */
|
|
no_mainloop:
|
|
{
|
|
mute = psink->mute;
|
|
GST_DEBUG_OBJECT (psink, "we have no mainloop");
|
|
return mute;
|
|
}
|
|
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;
|
|
|
|
g_free (psink->device_description);
|
|
psink->device_description = g_strdup (i->description);
|
|
|
|
done:
|
|
pa_threaded_mainloop_signal (mainloop, 0);
|
|
}
|
|
|
|
static gchar *
|
|
gst_pulsesink_device_description (GstPulseSink * psink)
|
|
{
|
|
GstPulseRingBuffer *pbuf;
|
|
pa_operation *o = NULL;
|
|
gchar *t;
|
|
|
|
if (!mainloop)
|
|
goto no_mainloop;
|
|
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
|
|
if (pbuf == NULL)
|
|
goto no_buffer;
|
|
|
|
if (!(o = pa_context_get_sink_info_by_name (pbuf->context,
|
|
psink->device, gst_pulsesink_sink_info_cb, pbuf)))
|
|
goto info_failed;
|
|
|
|
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
|
|
pa_threaded_mainloop_wait (mainloop);
|
|
if (gst_pulsering_is_dead (psink, pbuf, FALSE))
|
|
goto unlock;
|
|
}
|
|
|
|
unlock:
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
t = g_strdup (psink->device_description);
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
|
|
return t;
|
|
|
|
/* ERRORS */
|
|
no_mainloop:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "we have no mainloop");
|
|
return NULL;
|
|
}
|
|
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;
|
|
#ifdef 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
|
|
case PROP_CLIENT:
|
|
g_free (pulsesink->client_name);
|
|
if (!g_value_get_string (value)) {
|
|
GST_WARNING_OBJECT (pulsesink,
|
|
"Empty PulseAudio client name not allowed. Resetting to default value");
|
|
pulsesink->client_name = gst_pulse_client_name ();
|
|
} else
|
|
pulsesink->client_name = g_value_dup_string (value);
|
|
break;
|
|
case PROP_STREAM_PROPERTIES:
|
|
if (pulsesink->properties)
|
|
gst_structure_free (pulsesink->properties);
|
|
pulsesink->properties =
|
|
gst_structure_copy (gst_value_get_structure (value));
|
|
if (pulsesink->proplist)
|
|
pa_proplist_free (pulsesink->proplist);
|
|
pulsesink->proplist = gst_pulse_make_proplist (pulsesink->properties);
|
|
break;
|
|
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;
|
|
#ifdef 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
|
|
case PROP_CLIENT:
|
|
g_value_set_string (value, pulsesink->client_name);
|
|
break;
|
|
case PROP_STREAM_PROPERTIES:
|
|
gst_value_set_structure (value, pulsesink->properties);
|
|
break;
|
|
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 (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 (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;
|
|
}
|
|
}
|
|
|
|
#ifdef 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 (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 (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 void
|
|
gst_pulsesink_flush_ringbuffer (GstPulseSink * psink)
|
|
{
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
|
|
|
|
if (pbuf == NULL || pbuf->stream == NULL)
|
|
goto no_buffer;
|
|
|
|
gst_pulsering_flush (pbuf);
|
|
|
|
/* Uncork if we haven't already (happens when waiting to get enough data
|
|
* to send out the first time) */
|
|
if (pbuf->corked)
|
|
gst_pulsering_set_corked (pbuf, FALSE, FALSE);
|
|
|
|
/* We're not interested if this operation failed or not */
|
|
unlock:
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_buffer:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
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)
|
|
/* TRANSLATORS: 'song title' by 'artist name' */
|
|
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);
|
|
|
|
#ifdef HAVE_PULSE_0_9_11
|
|
gst_pulsesink_change_props (pulsesink, l);
|
|
#endif
|
|
|
|
break;
|
|
}
|
|
case GST_EVENT_EOS:
|
|
gst_pulsesink_flush_ringbuffer (pulsesink);
|
|
break;
|
|
default:
|
|
;
|
|
}
|
|
|
|
return GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_pulsesink_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (element);
|
|
GstStateChangeReturn ret;
|
|
guint res;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
g_mutex_lock (pa_shared_resource_mutex);
|
|
if (!mainloop_ref_ct) {
|
|
GST_INFO_OBJECT (element, "new pa main loop thread");
|
|
if (!(mainloop = pa_threaded_mainloop_new ()))
|
|
goto mainloop_failed;
|
|
mainloop_ref_ct = 1;
|
|
res = pa_threaded_mainloop_start (mainloop);
|
|
g_assert (res == 0);
|
|
g_mutex_unlock (pa_shared_resource_mutex);
|
|
} else {
|
|
GST_INFO_OBJECT (element, "reusing pa main loop thread");
|
|
mainloop_ref_ct++;
|
|
g_mutex_unlock (pa_shared_resource_mutex);
|
|
}
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
gst_element_post_message (element,
|
|
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
|
|
GST_BASE_AUDIO_SINK (pulsesink)->provided_clock, TRUE));
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
goto state_failure;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_element_post_message (element,
|
|
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
|
|
GST_BASE_AUDIO_SINK (pulsesink)->provided_clock));
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
if (mainloop) {
|
|
g_mutex_lock (pa_shared_resource_mutex);
|
|
mainloop_ref_ct--;
|
|
if (!mainloop_ref_ct) {
|
|
GST_INFO_OBJECT (element, "terminating pa main loop thread");
|
|
pa_threaded_mainloop_stop (mainloop);
|
|
pa_threaded_mainloop_free (mainloop);
|
|
mainloop = NULL;
|
|
}
|
|
g_mutex_unlock (pa_shared_resource_mutex);
|
|
}
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
mainloop_failed:
|
|
{
|
|
g_mutex_unlock (pa_shared_resource_mutex);
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
|
|
("pa_threaded_mainloop_new() failed"), (NULL));
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
state_failure:
|
|
{
|
|
if (transition == GST_STATE_CHANGE_NULL_TO_READY) {
|
|
/* Clear the PA mainloop if baseaudiosink failed to open the ring_buffer */
|
|
g_assert (mainloop);
|
|
g_mutex_lock (pa_shared_resource_mutex);
|
|
mainloop_ref_ct--;
|
|
if (!mainloop_ref_ct) {
|
|
GST_INFO_OBJECT (element, "terminating pa main loop thread");
|
|
pa_threaded_mainloop_stop (mainloop);
|
|
pa_threaded_mainloop_free (mainloop);
|
|
mainloop = NULL;
|
|
}
|
|
g_mutex_unlock (pa_shared_resource_mutex);
|
|
}
|
|
return ret;
|
|
}
|
|
}
|