mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-25 03:01:03 +00:00
46a3c9ac8b
This reverts commit 0dd46accf6.
With some audiosinks, starting the ringbuffer on the first commit
causes audio glitches at startup by starting to output segments
from the ringbuffer before it has been filled / fully prerolled. This
doesn't usually happen with pulsesink because we map the pulseaudio
ringbuffer directly, but we should keep things consistent with
other sinks with regards to startup latency, plus it gives more
headway to avoid glitching, should the initial 2nd segment take
more than 10ms to generate.
https://bugzilla.gnome.org/show_bug.cgi?id=657076
3307 lines
94 KiB
C
3307 lines
94 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., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301
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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
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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-1.0 -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-1.0 -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-1.0 -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/audio/audio.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 <gst/glib-compat-private.h>
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#include "pulsesink.h"
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#include "pulseutil.h"
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GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
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#define GST_CAT_DEFAULT pulse_debug
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#define DEFAULT_SERVER NULL
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#define DEFAULT_DEVICE NULL
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#define DEFAULT_CURRENT_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_CURRENT_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_NAME,
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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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/* A note on threading.
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*
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* We use a pa_threaded_mainloop to interact with the PulseAudio server. This
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* starts up a separate thread that runs a mainloop to carry back events,
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* messages and timing updates from the PulseAudio server.
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*
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* In most cases, the PulseAudio API we use communicates with the server and
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* processes replies asynchronously. Operations on PA objects that result in
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* such communication are protected with a pa_threaded_mainloop_lock() and
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* pa_threaded_mainloop_unlock(). These guarantee mutual exclusion with the
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* mainloop thread -- when an iteration of the mainloop thread begins, it first
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* tries to acquire this lock, and cannot do so if our code also holds that
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* lock.
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*
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* When we need to complete an operation synchronously, we use
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* pa_threaded_mainloop_wait() and pa_threaded_mainloop_signal(). These work
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* much as pthread conditionals do. pa_threaded_mainloop_wait() is called with
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* the mainloop lock held. It releases the lock (thereby allowing the mainloop
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* to execute), and waits till one of our callbacks to be executed by the
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* mainloop thread calls pa_threaded_mainloop_signal(). At the end of the
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* mainloop iteration, the pa_threaded_mainloop_wait() will reacquire the
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* mainloop lock and return control to the caller.
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*/
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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;
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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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GstAudioRingBuffer 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_stream *probe_stream;
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pa_format_info *format;
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guint channels;
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gboolean is_pcm;
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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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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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GstAudioRingBufferClass 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 GstAudioRingBufferClass *ring_parent_class = NULL;
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static gboolean gst_pulseringbuffer_open_device (GstAudioRingBuffer * buf);
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static gboolean gst_pulseringbuffer_close_device (GstAudioRingBuffer * buf);
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static gboolean gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
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GstAudioRingBufferSpec * spec);
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static gboolean gst_pulseringbuffer_release (GstAudioRingBuffer * buf);
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static gboolean gst_pulseringbuffer_start (GstAudioRingBuffer * buf);
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static gboolean gst_pulseringbuffer_pause (GstAudioRingBuffer * buf);
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static gboolean gst_pulseringbuffer_stop (GstAudioRingBuffer * buf);
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static void gst_pulseringbuffer_clear (GstAudioRingBuffer * buf);
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static guint gst_pulseringbuffer_commit (GstAudioRingBuffer * 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,
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GST_TYPE_AUDIO_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_mutex_init (&pa_shared_resource_mutex);
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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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GstAudioRingBufferClass *gstringbuffer_class;
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gobject_class = (GObjectClass *) klass;
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gstringbuffer_class = (GstAudioRingBufferClass *) 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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pbuf->probe_stream = NULL;
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pbuf->format = NULL;
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pbuf->channels = 0;
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pbuf->is_pcm = FALSE;
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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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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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/* Call with mainloop lock held if wait == TRUE) */
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static void
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gst_pulse_destroy_stream (pa_stream * stream, gboolean wait)
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{
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/* Make sure we don't get any further callbacks */
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pa_stream_set_write_callback (stream, NULL, NULL);
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pa_stream_set_underflow_callback (stream, NULL, NULL);
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pa_stream_set_overflow_callback (stream, NULL, NULL);
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pa_stream_disconnect (stream);
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if (wait)
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pa_threaded_mainloop_wait (mainloop);
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pa_stream_set_state_callback (stream, NULL, NULL);
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pa_stream_unref (stream);
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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->probe_stream) {
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gst_pulse_destroy_stream (pbuf->probe_stream, FALSE);
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pbuf->probe_stream = NULL;
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}
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if (pbuf->stream) {
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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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if (pbuf->format) {
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pa_format_info_free (pbuf->format);
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pbuf->format = NULL;
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pbuf->channels = 0;
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pbuf->is_pcm = FALSE;
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}
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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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pa_context_set_subscribe_callback (pctx->context, NULL, NULL);
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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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|
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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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|
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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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|
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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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|
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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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|
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GST_LOG_OBJECT (psink, "type %04x, idx %u", t, idx);
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|
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if (!pbuf->stream)
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continue;
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|
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if (idx != pa_stream_get_index (pbuf->stream))
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continue;
|
|
|
|
if (psink->device && pbuf->is_pcm &&
|
|
!g_str_equal (psink->device,
|
|
pa_stream_get_device_name (pbuf->stream))) {
|
|
/* Underlying sink changed. And this is not a passthrough stream. Let's
|
|
* see if someone upstream wants to try to renegotiate. */
|
|
GstEvent *renego;
|
|
|
|
g_free (psink->device);
|
|
psink->device = g_strdup (pa_stream_get_device_name (pbuf->stream));
|
|
|
|
GST_INFO_OBJECT (psink, "emitting sink-changed");
|
|
|
|
/* FIXME: send reconfigure event instead and let decodebin/playbin
|
|
* handle that. Also take care of ac3 alignment. See "pulse-format-lost" */
|
|
renego = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
|
|
gst_structure_new_empty ("pulse-sink-changed"));
|
|
|
|
if (!gst_pad_push_event (GST_BASE_SINK (psink)->sinkpad, renego))
|
|
GST_DEBUG_OBJECT (psink, "Emitted sink-changed - nobody was listening");
|
|
}
|
|
|
|
/* Actually this event is also triggered when other properties of
|
|
* the stream change that are unrelated to the volume. However it is
|
|
* probably cheaper to signal the change here and check for the
|
|
* volume when the GObject property is read instead of querying it always. */
|
|
|
|
/* inform streaming thread to notify */
|
|
g_atomic_int_compare_and_exchange (&psink->notify, 0, 1);
|
|
}
|
|
}
|
|
|
|
/* will be called when the device should be opened. In this case we will connect
|
|
* to the server. We should not try to open any streams in this state. */
|
|
static gboolean
|
|
gst_pulseringbuffer_open_device (GstAudioRingBuffer * buf)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
GstPulseContext *pctx;
|
|
pa_mainloop_api *api;
|
|
gboolean need_unlock_shared;
|
|
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
|
|
pbuf = GST_PULSERING_BUFFER_CAST (buf);
|
|
|
|
g_assert (!pbuf->stream);
|
|
g_assert (psink->client_name);
|
|
|
|
if (psink->server)
|
|
pbuf->context_name = g_strdup_printf ("%s@%s", psink->client_name,
|
|
psink->server);
|
|
else
|
|
pbuf->context_name = g_strdup (psink->client_name);
|
|
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
|
|
g_mutex_lock (&pa_shared_resource_mutex);
|
|
need_unlock_shared = TRUE;
|
|
|
|
pctx = g_hash_table_lookup (gst_pulse_shared_contexts, pbuf->context_name);
|
|
if (pctx == NULL) {
|
|
pctx = g_slice_new0 (GstPulseContext);
|
|
|
|
/* get the mainloop api and create a context */
|
|
GST_INFO_OBJECT (psink, "new context with name %s, pbuf=%p, pctx=%p",
|
|
pbuf->context_name, pbuf, pctx);
|
|
api = pa_threaded_mainloop_get_api (mainloop);
|
|
if (!(pctx->context = pa_context_new (api, pbuf->context_name)))
|
|
goto create_failed;
|
|
|
|
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);
|
|
pa_context_set_subscribe_callback (pctx->context,
|
|
gst_pulsering_context_subscribe_cb, pctx);
|
|
|
|
/* 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);
|
|
}
|
|
|
|
if (pa_context_get_server_protocol_version (pbuf->context) < 22) {
|
|
/* We need PulseAudio >= 1.0 on the server side for the extended API */
|
|
goto bad_server_version;
|
|
}
|
|
|
|
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;
|
|
}
|
|
bad_server_version:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED, ("PulseAudio server version "
|
|
"is too old."), (NULL));
|
|
goto unlock_and_fail;
|
|
}
|
|
}
|
|
|
|
/* close the device */
|
|
static gboolean
|
|
gst_pulseringbuffer_close_device (GstAudioRingBuffer * 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;
|
|
GstAudioRingBuffer *rbuf;
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
rbuf = GST_AUDIO_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;
|
|
GstAudioRingBuffer *ringbuf;
|
|
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));
|
|
ringbuf = GST_AUDIO_RING_BUFFER (pbuf);
|
|
|
|
if (!info) {
|
|
GST_LOG_OBJECT (psink, "latency update (information unknown)");
|
|
return;
|
|
}
|
|
|
|
if (!info->read_index_corrupt) {
|
|
/* Update segdone based on the read index. segdone is of segment
|
|
* granularity, while the read index is at byte granularity. We take the
|
|
* ceiling while converting the latter to the former since it is more
|
|
* conservative to report that we've read more than we have than to report
|
|
* less. One concern here is that latency updates happen every 100ms, which
|
|
* means segdone is not updated very often, but increasing the update
|
|
* frequency would mean more communication overhead. */
|
|
g_atomic_int_set (&ringbuf->segdone,
|
|
(int) gst_util_uint64_scale_ceil (info->read_index, 1,
|
|
ringbuf->spec.segsize));
|
|
}
|
|
|
|
sink_usec = info->configured_sink_usec;
|
|
|
|
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");
|
|
}
|
|
|
|
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");
|
|
}
|
|
|
|
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 if (!strcmp (name, PA_STREAM_EVENT_FORMAT_LOST)) {
|
|
GstEvent *renego;
|
|
|
|
if (g_atomic_int_get (&psink->format_lost)) {
|
|
/* Duplicate event before we're done reconfiguring, discard */
|
|
return;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (psink, "got FORMAT LOST");
|
|
g_atomic_int_set (&psink->format_lost, 1);
|
|
psink->format_lost_time = g_ascii_strtoull (pa_proplist_gets (pl,
|
|
"stream-time"), NULL, 0) * 1000;
|
|
|
|
g_free (psink->device);
|
|
psink->device = g_strdup (pa_proplist_gets (pl, "device"));
|
|
|
|
/* FIXME: send reconfigure event instead and let decodebin/playbin
|
|
* handle that. Also take care of ac3 alignment */
|
|
renego = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
|
|
gst_structure_new_empty ("pulse-format-lost"));
|
|
|
|
#if 0
|
|
if (g_str_equal (gst_structure_get_name (st), "audio/x-eac3")) {
|
|
GstStructure *event_st = gst_structure_new ("ac3parse-set-alignment",
|
|
"alignment", G_TYPE_STRING, pbin->dbin ? "frame" : "iec61937", NULL);
|
|
|
|
if (!gst_pad_push_event (pbin->sinkpad,
|
|
gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, event_st)))
|
|
GST_WARNING_OBJECT (pbin->sinkpad, "Could not update alignment");
|
|
}
|
|
#endif
|
|
|
|
if (!gst_pad_push_event (GST_BASE_SINK (psink)->sinkpad, renego)) {
|
|
/* Nobody handled the format change - emit an error */
|
|
GST_ELEMENT_ERROR (psink, STREAM, FORMAT, ("Sink format changed"),
|
|
("Sink format changed"));
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (psink, "got unknown event %s", name);
|
|
}
|
|
}
|
|
|
|
/* Called with the mainloop locked */
|
|
static gboolean
|
|
gst_pulsering_wait_for_stream_ready (GstPulseSink * psink, pa_stream * stream)
|
|
{
|
|
pa_stream_state_t state;
|
|
|
|
for (;;) {
|
|
state = pa_stream_get_state (stream);
|
|
|
|
GST_LOG_OBJECT (psink, "stream state is now %d", state);
|
|
|
|
if (!PA_STREAM_IS_GOOD (state))
|
|
return FALSE;
|
|
|
|
if (state == PA_STREAM_READY)
|
|
return TRUE;
|
|
|
|
/* Wait until the stream is ready */
|
|
pa_threaded_mainloop_wait (mainloop);
|
|
}
|
|
}
|
|
|
|
|
|
/* 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 (GstAudioRingBuffer * buf,
|
|
GstAudioRingBufferSpec * spec)
|
|
{
|
|
GstPulseSink *psink;
|
|
GstPulseRingBuffer *pbuf;
|
|
pa_buffer_attr wanted;
|
|
const pa_buffer_attr *actual;
|
|
pa_channel_map channel_map;
|
|
pa_operation *o = NULL;
|
|
pa_cvolume v;
|
|
pa_cvolume *pv = NULL;
|
|
pa_stream_flags_t flags;
|
|
const gchar *name;
|
|
GstAudioClock *clock;
|
|
pa_format_info *formats[1];
|
|
#ifndef GST_DISABLE_GST_DEBUG
|
|
gchar print_buf[PA_FORMAT_INFO_SNPRINT_MAX];
|
|
#endif
|
|
|
|
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_format_info (spec, &pbuf->format, &pbuf->channels))
|
|
goto invalid_spec;
|
|
pbuf->is_pcm = pa_format_info_is_pcm (pbuf->format);
|
|
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
|
|
/* we need a context and a no stream */
|
|
g_assert (pbuf->context);
|
|
g_assert (!pbuf->stream);
|
|
|
|
/* if we have a probe, disconnect it first so that if we're creating a
|
|
* compressed stream, it doesn't get blocked by a PCM stream */
|
|
if (pbuf->probe_stream) {
|
|
gst_pulse_destroy_stream (pbuf->probe_stream, TRUE);
|
|
pbuf->probe_stream = NULL;
|
|
}
|
|
|
|
/* 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 */
|
|
if (pbuf->is_pcm && gst_pulse_gst_to_channel_map (&channel_map, spec))
|
|
pa_format_info_set_channel_map (pbuf->format, &channel_map);
|
|
|
|
/* find a good name for the stream */
|
|
if (psink->stream_name)
|
|
name = psink->stream_name;
|
|
else
|
|
name = "Playback Stream";
|
|
|
|
/* create a stream */
|
|
formats[0] = pbuf->format;
|
|
if (!(pbuf->stream = pa_stream_new_extended (pbuf->context, name, formats, 1,
|
|
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);
|
|
pa_stream_set_started_callback (pbuf->stream,
|
|
gst_pulsering_stream_started_cb, pbuf);
|
|
pa_stream_set_event_callback (pbuf->stream,
|
|
gst_pulsering_stream_event_cb, pbuf);
|
|
|
|
/* 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);
|
|
|
|
/* 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;
|
|
if (pbuf->is_pcm)
|
|
gst_pulse_cvolume_from_linear (pv, pbuf->channels, psink->volume);
|
|
else {
|
|
GST_DEBUG_OBJECT (psink, "passthrough stream, not setting volume");
|
|
pv = NULL;
|
|
}
|
|
} else {
|
|
pv = NULL;
|
|
}
|
|
|
|
/* construct the flags */
|
|
flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
|
|
PA_STREAM_ADJUST_LATENCY | PA_STREAM_START_CORKED;
|
|
|
|
if (psink->mute_set) {
|
|
if (psink->mute)
|
|
flags |= PA_STREAM_START_MUTED;
|
|
else
|
|
flags |= PA_STREAM_START_UNMUTED;
|
|
}
|
|
|
|
/* 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_AUDIO_BASE_SINK (psink)->provided_clock);
|
|
gst_audio_clock_reset (clock, 0);
|
|
|
|
if (!gst_pulsering_wait_for_stream_ready (psink, pbuf->stream))
|
|
goto connect_failed;
|
|
|
|
g_free (psink->device);
|
|
psink->device = g_strdup (pa_stream_get_device_name (pbuf->stream));
|
|
|
|
#ifndef GST_DISABLE_GST_DEBUG
|
|
pa_format_info_snprint (print_buf, sizeof (print_buf),
|
|
pa_stream_get_format_info (pbuf->stream));
|
|
GST_INFO_OBJECT (psink, "negotiated to: %s", print_buf);
|
|
#endif
|
|
|
|
/* 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 (GstAudioRingBuffer * buf)
|
|
{
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (buf);
|
|
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
gst_pulsering_destroy_stream (pbuf);
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
|
|
{
|
|
GstPulseSink *psink;
|
|
|
|
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
|
|
g_atomic_int_set (&psink->format_lost, FALSE);
|
|
psink->format_lost_time = GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_pulsering_success_cb (pa_stream * s, int success, void *userdata)
|
|
{
|
|
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));
|
|
|
|
if (g_atomic_int_get (&psink->format_lost)) {
|
|
/* Sink format changed, stream's gone so fake being paused */
|
|
return TRUE;
|
|
}
|
|
|
|
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 (GstAudioRingBuffer * 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);
|
|
}
|
|
|
|
#if 0
|
|
/* called from pulse thread with the mainloop lock */
|
|
static void
|
|
mainloop_enter_defer_cb (pa_mainloop_api * api, void *userdata)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
|
|
GstMessage *message;
|
|
GValue val = { 0 };
|
|
|
|
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));
|
|
g_value_init (&val, GST_TYPE_G_THREAD);
|
|
g_value_set_boxed (&val, g_thread_self ());
|
|
gst_message_set_stream_status_object (message, &val);
|
|
g_value_unset (&val);
|
|
|
|
gst_element_post_message (GST_ELEMENT (pulsesink), message);
|
|
|
|
g_return_if_fail (pulsesink->defer_pending);
|
|
pulsesink->defer_pending--;
|
|
pa_threaded_mainloop_signal (mainloop, 0);
|
|
}
|
|
#endif
|
|
|
|
/* start/resume playback ASAP, we don't uncork here but in the commit method */
|
|
static gboolean
|
|
gst_pulseringbuffer_start (GstAudioRingBuffer * 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, "starting");
|
|
pbuf->paused = FALSE;
|
|
|
|
/* EOS needs running clock */
|
|
if (GST_BASE_SINK_CAST (psink)->eos ||
|
|
g_atomic_int_get (&GST_AUDIO_BASE_SINK (psink)->eos_rendering))
|
|
gst_pulsering_set_corked (pbuf, FALSE, FALSE);
|
|
|
|
#if 0
|
|
GST_DEBUG_OBJECT (psink, "scheduling stream status");
|
|
psink->defer_pending++;
|
|
pa_mainloop_api_once (pa_threaded_mainloop_get_api (mainloop),
|
|
mainloop_enter_defer_cb, psink);
|
|
|
|
/* Wait for the stream status message to be posted. This needs to be done
|
|
* synchronously because the callback will take the mainloop lock
|
|
* (implicitly) and then take the GST_OBJECT_LOCK. Everywhere else, we take
|
|
* the locks in the reverse order, so not doing this synchronously could
|
|
* cause a deadlock. */
|
|
GST_DEBUG_OBJECT (psink, "waiting for stream status (ENTER) to be posted");
|
|
pa_threaded_mainloop_wait (mainloop);
|
|
#endif
|
|
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* pause/stop playback ASAP */
|
|
static gboolean
|
|
gst_pulseringbuffer_pause (GstAudioRingBuffer * 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;
|
|
}
|
|
|
|
#if 0
|
|
/* called from pulse thread with the mainloop lock */
|
|
static void
|
|
mainloop_leave_defer_cb (pa_mainloop_api * api, void *userdata)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
|
|
GstMessage *message;
|
|
GValue val = { 0 };
|
|
|
|
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));
|
|
g_value_init (&val, GST_TYPE_G_THREAD);
|
|
g_value_set_boxed (&val, g_thread_self ());
|
|
gst_message_set_stream_status_object (message, &val);
|
|
g_value_unset (&val);
|
|
|
|
gst_element_post_message (GST_ELEMENT (pulsesink), message);
|
|
|
|
g_return_if_fail (pulsesink->defer_pending);
|
|
pulsesink->defer_pending--;
|
|
pa_threaded_mainloop_signal (mainloop, 0);
|
|
}
|
|
#endif
|
|
|
|
/* stop playback, we flush everything. */
|
|
static gboolean
|
|
gst_pulseringbuffer_stop (GstAudioRingBuffer * 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 (g_atomic_int_get (&psink->format_lost)) {
|
|
/* Don't try to flush, the stream's probably gone by now */
|
|
res = TRUE;
|
|
goto cleanup;
|
|
}
|
|
|
|
/* 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);
|
|
}
|
|
#if 0
|
|
GST_DEBUG_OBJECT (psink, "scheduling stream status");
|
|
psink->defer_pending++;
|
|
pa_mainloop_api_once (pa_threaded_mainloop_get_api (mainloop),
|
|
mainloop_leave_defer_cb, psink);
|
|
|
|
/* Wait for the stream status message to be posted. This needs to be done
|
|
* synchronously because the callback will take the mainloop lock
|
|
* (implicitly) and then take the GST_OBJECT_LOCK. Everywhere else, we take
|
|
* the locks in the reverse order, so not doing this synchronously could
|
|
* cause a deadlock. */
|
|
GST_DEBUG_OBJECT (psink, "waiting for stream status (LEAVE) to be posted");
|
|
pa_threaded_mainloop_wait (mainloop);
|
|
#endif
|
|
|
|
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, bpf); \
|
|
s += bpf; \
|
|
*accum += outr; \
|
|
if ((*accum << 1) >= inr) { \
|
|
*accum -= inr; \
|
|
d += bpf; \
|
|
} \
|
|
} \
|
|
in_samples -= (s - sb)/bpf; \
|
|
out_samples -= (d - db)/bpf; \
|
|
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, bpf); \
|
|
d += bpf; \
|
|
*accum += inr; \
|
|
if ((*accum << 1) >= outr) { \
|
|
*accum -= outr; \
|
|
s += bpf; \
|
|
} \
|
|
} \
|
|
in_samples -= (s - sb)/bpf; \
|
|
out_samples -= (d - db)/bpf; \
|
|
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, bpf); \
|
|
se -= bpf; \
|
|
*accum += outr; \
|
|
while (d < de && (*accum << 1) >= inr) { \
|
|
*accum -= inr; \
|
|
d += bpf; \
|
|
} \
|
|
} \
|
|
in_samples -= (sb - se)/bpf; \
|
|
out_samples -= (d - db)/bpf; \
|
|
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, bpf); \
|
|
d += bpf; \
|
|
*accum += inr; \
|
|
while (s <= se && (*accum << 1) >= outr) { \
|
|
*accum -= outr; \
|
|
se -= bpf; \
|
|
} \
|
|
} \
|
|
in_samples -= (sb - se)/bpf; \
|
|
out_samples -= (d - db)/bpf; \
|
|
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 (GstAudioRingBuffer * 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, bpf;
|
|
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");
|
|
g_object_notify (G_OBJECT (psink), "current-device");
|
|
}
|
|
|
|
/* make sure the ringbuffer is started */
|
|
if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
|
|
GST_AUDIO_RING_BUFFER_STATE_STARTED)) {
|
|
/* see if we are allowed to start it */
|
|
if (G_UNLIKELY (g_atomic_int_get (&buf->may_start) == FALSE))
|
|
goto no_start;
|
|
|
|
GST_DEBUG_OBJECT (buf, "start!");
|
|
if (!gst_audio_ring_buffer_start (buf))
|
|
goto start_failed;
|
|
}
|
|
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
|
|
GST_DEBUG_OBJECT (psink, "entering commit");
|
|
pbuf->in_commit = TRUE;
|
|
|
|
bpf = GST_AUDIO_INFO_BPF (&buf->spec.info);
|
|
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 + (bpf * inr);
|
|
|
|
if (g_atomic_int_get (&psink->format_lost)) {
|
|
/* Sink format changed, drop the data and hope upstream renegotiates */
|
|
goto fake_done;
|
|
}
|
|
|
|
if (pbuf->paused)
|
|
goto was_paused;
|
|
|
|
/* offset is in bytes */
|
|
offset = *sample * bpf;
|
|
|
|
while (*toprocess > 0) {
|
|
size_t avail;
|
|
guint towrite;
|
|
|
|
GST_LOG_OBJECT (psink,
|
|
"need to write %d samples at offset %" G_GINT64_FORMAT, *toprocess,
|
|
offset);
|
|
|
|
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 * bpf;
|
|
|
|
/* Wait for at least segsize bytes to become available */
|
|
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 / bpf, 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 (g_atomic_int_get (&psink->format_lost)) {
|
|
/* Sink format changed, give up and hope upstream renegotiates */
|
|
goto fake_done;
|
|
}
|
|
|
|
if (pbuf->m_writable == (size_t) - 1)
|
|
goto writable_size_failed;
|
|
|
|
pbuf->m_writable /= bpf;
|
|
pbuf->m_writable *= bpf; /* 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 segsize bytes, wait a bit */
|
|
GST_LOG_OBJECT (psink, "waiting for free space");
|
|
pa_threaded_mainloop_wait (mainloop);
|
|
|
|
if (pbuf->paused)
|
|
goto was_paused;
|
|
}
|
|
|
|
/* Recalculate what we can write in the next chunk */
|
|
towrite = out_samples * bpf;
|
|
if (pbuf->m_writable > towrite)
|
|
pbuf->m_writable = towrite;
|
|
|
|
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);
|
|
|
|
}
|
|
|
|
if (towrite > pbuf->m_writable)
|
|
towrite = pbuf->m_writable;
|
|
avail = towrite / bpf;
|
|
|
|
GST_LOG_OBJECT (psink, "writing %u samples at offset %" G_GUINT64_FORMAT,
|
|
(guint) avail, offset);
|
|
|
|
/* No trick modes for passthrough streams */
|
|
if (G_UNLIKELY (!pbuf->is_pcm && (inr != outr || reverse))) {
|
|
GST_WARNING_OBJECT (psink, "Passthrough stream can't run in trick mode");
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
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 / bpf;
|
|
}
|
|
|
|
/* 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 / bpf, 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 */
|
|
}
|
|
|
|
*sample += avail;
|
|
offset += avail * bpf;
|
|
pbuf->m_lastoffset = offset;
|
|
|
|
/* 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");
|
|
}
|
|
}
|
|
}
|
|
|
|
fake_done:
|
|
/* we consumed all samples here */
|
|
data = data_end + bpf;
|
|
|
|
pbuf->in_commit = FALSE;
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
|
|
done:
|
|
result = inr - ((data_end - data) / bpf);
|
|
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)
|
|
{
|
|
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 bpf;
|
|
|
|
bpf = (GST_AUDIO_RING_BUFFER_CAST (pbuf))->spec.info.bpf;
|
|
GST_LOG_OBJECT (psink,
|
|
"flushing %u samples at offset %" G_GINT64_FORMAT,
|
|
(guint) pbuf->m_towrite / bpf, 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;
|
|
}
|
|
}
|
|
|
|
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 gboolean gst_pulsesink_query (GstBaseSink * sink, GstQuery * query);
|
|
|
|
static GstStateChangeReturn gst_pulsesink_change_state (GstElement * element,
|
|
GstStateChange transition);
|
|
|
|
static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS (PULSE_SINK_TEMPLATE_CAPS));
|
|
|
|
#define gst_pulsesink_parent_class parent_class
|
|
G_DEFINE_TYPE_WITH_CODE (GstPulseSink, gst_pulsesink, GST_TYPE_AUDIO_BASE_SINK,
|
|
gst_pulsesink_init_contexts ();
|
|
G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL)
|
|
);
|
|
|
|
static GstAudioRingBuffer *
|
|
gst_pulsesink_create_ringbuffer (GstAudioBaseSink * sink)
|
|
{
|
|
GstAudioRingBuffer *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 GstBuffer *
|
|
gst_pulsesink_payload (GstAudioBaseSink * sink, GstBuffer * buf)
|
|
{
|
|
switch (sink->ringbuffer->spec.type) {
|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:
|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC:
|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC:
|
|
{
|
|
/* FIXME: alloc memory from PA if possible */
|
|
gint framesize = gst_audio_iec61937_frame_size (&sink->ringbuffer->spec);
|
|
GstBuffer *out;
|
|
GstMapInfo inmap, outmap;
|
|
gboolean res;
|
|
|
|
if (framesize <= 0)
|
|
return NULL;
|
|
|
|
out = gst_buffer_new_and_alloc (framesize);
|
|
|
|
gst_buffer_map (buf, &inmap, GST_MAP_READ);
|
|
gst_buffer_map (out, &outmap, GST_MAP_WRITE);
|
|
|
|
res = gst_audio_iec61937_payload (inmap.data, inmap.size,
|
|
outmap.data, outmap.size, &sink->ringbuffer->spec, G_BIG_ENDIAN);
|
|
|
|
gst_buffer_unmap (buf, &inmap);
|
|
gst_buffer_unmap (out, &outmap);
|
|
|
|
if (!res) {
|
|
gst_buffer_unref (out);
|
|
return NULL;
|
|
}
|
|
|
|
gst_buffer_copy_into (out, buf, GST_BUFFER_COPY_METADATA, 0, -1);
|
|
return out;
|
|
}
|
|
|
|
default:
|
|
return gst_buffer_ref (buf);
|
|
}
|
|
}
|
|
|
|
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;
|
|
GstAudioBaseSinkClass *gstaudiosink_class = GST_AUDIO_BASE_SINK_CLASS (klass);
|
|
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
|
|
gchar *clientname;
|
|
|
|
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);
|
|
gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_pulsesink_query);
|
|
|
|
/* 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);
|
|
gstaudiosink_class->payload = GST_DEBUG_FUNCPTR (gst_pulsesink_payload);
|
|
|
|
/* 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_CURRENT_DEVICE,
|
|
g_param_spec_string ("current-device", "Current Device",
|
|
"The current PulseAudio sink device", DEFAULT_CURRENT_DEVICE,
|
|
G_PARAM_READABLE | 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));
|
|
|
|
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));
|
|
|
|
/**
|
|
* GstPulseSink:client-name:
|
|
*
|
|
* The PulseAudio client name to use.
|
|
*/
|
|
clientname = gst_pulse_client_name ();
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_CLIENT_NAME,
|
|
g_param_spec_string ("client-name", "Client Name",
|
|
"The PulseAudio client name to use", clientname,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
|
|
GST_PARAM_MUTABLE_READY));
|
|
g_free (clientname);
|
|
|
|
/**
|
|
* 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
|
|
* ]|
|
|
*/
|
|
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));
|
|
|
|
gst_element_class_set_static_metadata (gstelement_class,
|
|
"PulseAudio Audio Sink",
|
|
"Sink/Audio", "Plays audio to a PulseAudio server", "Lennart Poettering");
|
|
gst_element_class_add_static_pad_template (gstelement_class, &pad_template);
|
|
}
|
|
|
|
static void
|
|
free_device_info (GstPulseDeviceInfo * device_info)
|
|
{
|
|
GList *l;
|
|
|
|
g_free (device_info->description);
|
|
|
|
for (l = g_list_first (device_info->formats); l; l = g_list_next (l))
|
|
pa_format_info_free ((pa_format_info *) l->data);
|
|
|
|
g_list_free (device_info->formats);
|
|
}
|
|
|
|
/* Returns the current time of the sink ringbuffer. The timing_info is updated
|
|
* on every data write/flush and every 100ms (PA_STREAM_AUTO_TIMING_UPDATE).
|
|
*/
|
|
static GstClockTime
|
|
gst_pulsesink_get_time (GstClock * clock, GstAudioBaseSink * 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));
|
|
|
|
if (g_atomic_int_get (&psink->format_lost)) {
|
|
/* Stream was lost in a format change, it'll get set up again once
|
|
* upstream renegotiates */
|
|
return psink->format_lost_time;
|
|
}
|
|
|
|
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_sink_info_cb (pa_context * c, const pa_sink_info * i, int eol,
|
|
void *userdata)
|
|
{
|
|
GstPulseDeviceInfo *device_info = (GstPulseDeviceInfo *) userdata;
|
|
guint8 j;
|
|
|
|
if (!i)
|
|
goto done;
|
|
|
|
device_info->description = g_strdup (i->description);
|
|
|
|
device_info->formats = NULL;
|
|
for (j = 0; j < i->n_formats; j++)
|
|
device_info->formats = g_list_prepend (device_info->formats,
|
|
pa_format_info_copy (i->formats[j]));
|
|
|
|
done:
|
|
pa_threaded_mainloop_signal (mainloop, 0);
|
|
}
|
|
|
|
/* Call with mainloop lock held */
|
|
static pa_stream *
|
|
gst_pulsesink_create_probe_stream (GstPulseSink * psink,
|
|
GstPulseRingBuffer * pbuf, pa_format_info * format)
|
|
{
|
|
pa_format_info *formats[1] = { format };
|
|
pa_stream *stream;
|
|
pa_stream_flags_t flags;
|
|
|
|
GST_LOG_OBJECT (psink, "Creating probe stream");
|
|
|
|
if (!(stream = pa_stream_new_extended (pbuf->context, "pulsesink probe",
|
|
formats, 1, psink->proplist)))
|
|
goto error;
|
|
|
|
/* construct the flags */
|
|
flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
|
|
PA_STREAM_ADJUST_LATENCY | PA_STREAM_START_CORKED;
|
|
|
|
pa_stream_set_state_callback (stream, gst_pulsering_stream_state_cb, pbuf);
|
|
|
|
if (pa_stream_connect_playback (stream, psink->device, NULL, flags, NULL,
|
|
NULL) < 0)
|
|
goto error;
|
|
|
|
if (!gst_pulsering_wait_for_stream_ready (psink, stream))
|
|
goto error;
|
|
|
|
return stream;
|
|
|
|
error:
|
|
if (stream)
|
|
pa_stream_unref (stream);
|
|
return NULL;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_pulsesink_query_getcaps (GstPulseSink * psink, GstCaps * filter)
|
|
{
|
|
GstPulseRingBuffer *pbuf = NULL;
|
|
GstPulseDeviceInfo device_info = { NULL, NULL };
|
|
GstCaps *ret = NULL;
|
|
GList *i;
|
|
pa_operation *o = NULL;
|
|
pa_stream *stream;
|
|
|
|
GST_OBJECT_LOCK (psink);
|
|
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
|
|
if (pbuf != NULL)
|
|
gst_object_ref (pbuf);
|
|
GST_OBJECT_UNLOCK (psink);
|
|
|
|
if (!pbuf) {
|
|
ret = gst_pad_get_pad_template_caps (GST_AUDIO_BASE_SINK_PAD (psink));
|
|
goto out;
|
|
}
|
|
|
|
GST_OBJECT_LOCK (pbuf);
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
|
|
if (!pbuf->context) {
|
|
ret = gst_pad_get_pad_template_caps (GST_AUDIO_BASE_SINK_PAD (psink));
|
|
goto unlock;
|
|
}
|
|
|
|
ret = gst_caps_new_empty ();
|
|
|
|
if (pbuf->stream) {
|
|
/* We're in PAUSED or higher */
|
|
stream = pbuf->stream;
|
|
|
|
} else if (pbuf->probe_stream) {
|
|
/* We're not paused, but have a cached probe stream */
|
|
stream = pbuf->probe_stream;
|
|
|
|
} else {
|
|
/* We're not yet in PAUSED and still need to create a probe stream.
|
|
*
|
|
* FIXME: PA doesn't accept "any" format. We fix something reasonable since
|
|
* this is merely a probe. This should eventually be fixed in PA and
|
|
* hard-coding the format should be dropped. */
|
|
pa_format_info *format = pa_format_info_new ();
|
|
format->encoding = PA_ENCODING_PCM;
|
|
pa_format_info_set_sample_format (format, PA_SAMPLE_S16LE);
|
|
pa_format_info_set_rate (format, GST_AUDIO_DEF_RATE);
|
|
pa_format_info_set_channels (format, GST_AUDIO_DEF_CHANNELS);
|
|
|
|
pbuf->probe_stream = gst_pulsesink_create_probe_stream (psink, pbuf,
|
|
format);
|
|
|
|
pa_format_info_free (format);
|
|
|
|
if (!pbuf->probe_stream) {
|
|
GST_WARNING_OBJECT (psink, "Could not create probe stream");
|
|
goto unlock;
|
|
}
|
|
|
|
stream = pbuf->probe_stream;
|
|
}
|
|
|
|
if (!(o = pa_context_get_sink_info_by_name (pbuf->context,
|
|
pa_stream_get_device_name (stream), gst_pulsesink_sink_info_cb,
|
|
&device_info)))
|
|
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;
|
|
}
|
|
|
|
for (i = g_list_first (device_info.formats); i; i = g_list_next (i)) {
|
|
gst_caps_append (ret,
|
|
gst_pulse_format_info_to_caps ((pa_format_info *) i->data));
|
|
}
|
|
|
|
unlock:
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
/* FIXME: this could be freed after device_name is got */
|
|
GST_OBJECT_UNLOCK (pbuf);
|
|
|
|
if (filter) {
|
|
GstCaps *tmp = gst_caps_intersect_full (filter, ret,
|
|
GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (ret);
|
|
ret = tmp;
|
|
}
|
|
|
|
out:
|
|
free_device_info (&device_info);
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
if (pbuf)
|
|
gst_object_unref (pbuf);
|
|
|
|
GST_DEBUG_OBJECT (psink, "caps %" GST_PTR_FORMAT, ret);
|
|
|
|
return ret;
|
|
|
|
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_query_acceptcaps (GstPulseSink * psink, GstCaps * caps)
|
|
{
|
|
GstPulseRingBuffer *pbuf = NULL;
|
|
GstPulseDeviceInfo device_info = { NULL, NULL };
|
|
GstCaps *pad_caps;
|
|
GstStructure *st;
|
|
gboolean ret = FALSE;
|
|
|
|
GstAudioRingBufferSpec spec = { 0 };
|
|
pa_operation *o = NULL;
|
|
pa_channel_map channel_map;
|
|
pa_format_info *format = NULL;
|
|
guint channels;
|
|
|
|
pad_caps = gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD (psink));
|
|
ret = gst_caps_is_subset (caps, pad_caps);
|
|
gst_caps_unref (pad_caps);
|
|
|
|
GST_DEBUG_OBJECT (psink, "caps %" GST_PTR_FORMAT, caps);
|
|
|
|
/* Template caps didn't match */
|
|
if (!ret)
|
|
goto done;
|
|
|
|
/* If we've not got fixed caps, creating a stream might fail, so let's just
|
|
* return from here with default acceptcaps behaviour */
|
|
if (!gst_caps_is_fixed (caps))
|
|
goto done;
|
|
|
|
GST_OBJECT_LOCK (psink);
|
|
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
|
|
if (pbuf != NULL)
|
|
gst_object_ref (pbuf);
|
|
GST_OBJECT_UNLOCK (psink);
|
|
|
|
/* We're still in NULL state */
|
|
if (pbuf == NULL)
|
|
goto done;
|
|
|
|
GST_OBJECT_LOCK (pbuf);
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
|
|
if (pbuf->context == NULL)
|
|
goto out;
|
|
|
|
ret = FALSE;
|
|
|
|
spec.latency_time = GST_AUDIO_BASE_SINK (psink)->latency_time;
|
|
if (!gst_audio_ring_buffer_parse_caps (&spec, caps))
|
|
goto out;
|
|
|
|
if (!gst_pulse_fill_format_info (&spec, &format, &channels))
|
|
goto out;
|
|
|
|
/* Make sure input is framed (one frame per buffer) and can be payloaded */
|
|
if (!pa_format_info_is_pcm (format)) {
|
|
gboolean framed = FALSE, parsed = FALSE;
|
|
st = gst_caps_get_structure (caps, 0);
|
|
|
|
gst_structure_get_boolean (st, "framed", &framed);
|
|
gst_structure_get_boolean (st, "parsed", &parsed);
|
|
if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0)
|
|
goto out;
|
|
}
|
|
|
|
/* initialize the channel map */
|
|
if (pa_format_info_is_pcm (format) &&
|
|
gst_pulse_gst_to_channel_map (&channel_map, &spec))
|
|
pa_format_info_set_channel_map (format, &channel_map);
|
|
|
|
if (pbuf->stream || pbuf->probe_stream) {
|
|
/* We're already in PAUSED or above, so just reuse this stream to query
|
|
* sink formats and use those. */
|
|
GList *i;
|
|
const char *device_name = pa_stream_get_device_name (pbuf->stream ?
|
|
pbuf->stream : pbuf->probe_stream);
|
|
|
|
if (!(o = pa_context_get_sink_info_by_name (pbuf->context, device_name,
|
|
gst_pulsesink_sink_info_cb, &device_info)))
|
|
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 out;
|
|
}
|
|
|
|
for (i = g_list_first (device_info.formats); i; i = g_list_next (i)) {
|
|
if (pa_format_info_is_compatible ((pa_format_info *) i->data, format)) {
|
|
ret = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
} else {
|
|
/* We're in READY, let's connect a stream to see if the format is
|
|
* accepted by whatever sink we're routed to */
|
|
pbuf->probe_stream = gst_pulsesink_create_probe_stream (psink, pbuf,
|
|
format);
|
|
if (pbuf->probe_stream)
|
|
ret = TRUE;
|
|
}
|
|
|
|
out:
|
|
if (format)
|
|
pa_format_info_free (format);
|
|
|
|
free_device_info (&device_info);
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
GST_OBJECT_UNLOCK (pbuf);
|
|
|
|
gst_caps_replace (&spec.caps, NULL);
|
|
gst_object_unref (pbuf);
|
|
|
|
done:
|
|
|
|
return ret;
|
|
|
|
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 out;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_init (GstPulseSink * pulsesink)
|
|
{
|
|
pulsesink->server = NULL;
|
|
pulsesink->device = NULL;
|
|
pulsesink->device_info.description = NULL;
|
|
pulsesink->client_name = gst_pulse_client_name ();
|
|
|
|
pulsesink->device_info.formats = NULL;
|
|
|
|
pulsesink->volume = DEFAULT_VOLUME;
|
|
pulsesink->volume_set = FALSE;
|
|
|
|
pulsesink->mute = DEFAULT_MUTE;
|
|
pulsesink->mute_set = FALSE;
|
|
|
|
pulsesink->notify = 0;
|
|
|
|
g_atomic_int_set (&pulsesink->format_lost, FALSE);
|
|
pulsesink->format_lost_time = GST_CLOCK_TIME_NONE;
|
|
|
|
pulsesink->properties = NULL;
|
|
pulsesink->proplist = NULL;
|
|
|
|
/* override with a custom clock */
|
|
if (GST_AUDIO_BASE_SINK (pulsesink)->provided_clock)
|
|
gst_object_unref (GST_AUDIO_BASE_SINK (pulsesink)->provided_clock);
|
|
|
|
GST_AUDIO_BASE_SINK (pulsesink)->provided_clock =
|
|
gst_audio_clock_new ("GstPulseSinkClock",
|
|
(GstAudioClockGetTimeFunc) gst_pulsesink_get_time, pulsesink, NULL);
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_finalize (GObject * object)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
|
|
|
|
g_free (pulsesink->server);
|
|
g_free (pulsesink->device);
|
|
g_free (pulsesink->client_name);
|
|
g_free (pulsesink->current_sink_name);
|
|
|
|
free_device_info (&pulsesink->device_info);
|
|
|
|
if (pulsesink->properties)
|
|
gst_structure_free (pulsesink->properties);
|
|
if (pulsesink->proplist)
|
|
pa_proplist_free (pulsesink->proplist);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
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_AUDIO_BASE_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 (pbuf->is_pcm)
|
|
gst_pulse_cvolume_from_linear (&v, pbuf->channels, volume);
|
|
else
|
|
/* FIXME: this will eventually be superceded by checks to see if the volume
|
|
* is readable/writable */
|
|
goto unlock;
|
|
|
|
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_AUDIO_BASE_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;
|
|
psink->current_sink_idx = i->sink;
|
|
|
|
if (psink->volume > MAX_VOLUME) {
|
|
GST_WARNING_OBJECT (psink, "Clipped volume from %f to %f", psink->volume,
|
|
MAX_VOLUME);
|
|
psink->volume = MAX_VOLUME;
|
|
}
|
|
}
|
|
|
|
done:
|
|
pa_threaded_mainloop_signal (mainloop, 0);
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_get_sink_input_info (GstPulseSink * psink, gdouble * volume,
|
|
gboolean * mute)
|
|
{
|
|
GstPulseRingBuffer *pbuf;
|
|
pa_operation *o = NULL;
|
|
uint32_t idx;
|
|
|
|
if (!mainloop)
|
|
goto no_mainloop;
|
|
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_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 (volume)
|
|
*volume = psink->volume;
|
|
if (mute)
|
|
*mute = psink->mute;
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_mainloop:
|
|
{
|
|
if (volume)
|
|
*volume = psink->volume;
|
|
if (mute)
|
|
*mute = psink->mute;
|
|
|
|
GST_DEBUG_OBJECT (psink, "we have no mainloop");
|
|
return;
|
|
}
|
|
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 void
|
|
gst_pulsesink_current_sink_info_cb (pa_context * c, const pa_sink_info * i,
|
|
int eol, void *userdata)
|
|
{
|
|
GstPulseSink *psink;
|
|
|
|
psink = GST_PULSESINK_CAST (userdata);
|
|
|
|
if (!i)
|
|
goto done;
|
|
|
|
/* If the index doesn't match our current stream,
|
|
* it implies we just recreated the stream (caps change)
|
|
*/
|
|
if (i->index == psink->current_sink_idx) {
|
|
g_free (psink->current_sink_name);
|
|
psink->current_sink_name = g_strdup (i->name);
|
|
}
|
|
|
|
done:
|
|
pa_threaded_mainloop_signal (mainloop, 0);
|
|
}
|
|
|
|
static gchar *
|
|
gst_pulsesink_get_current_device (GstPulseSink * pulsesink)
|
|
{
|
|
pa_operation *o = NULL;
|
|
GstPulseRingBuffer *pbuf;
|
|
gchar *current_sink;
|
|
|
|
if (!mainloop)
|
|
goto no_mainloop;
|
|
|
|
pbuf =
|
|
GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (pulsesink)->ringbuffer);
|
|
if (pbuf == NULL || pbuf->stream == NULL)
|
|
goto no_buffer;
|
|
|
|
gst_pulsesink_get_sink_input_info (pulsesink, NULL, NULL);
|
|
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
|
|
if (!(o = pa_context_get_sink_info_by_index (pbuf->context,
|
|
pulsesink->current_sink_idx, gst_pulsesink_current_sink_info_cb,
|
|
pulsesink)))
|
|
goto info_failed;
|
|
|
|
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
|
|
pa_threaded_mainloop_wait (mainloop);
|
|
if (gst_pulsering_is_dead (pulsesink, pbuf, TRUE))
|
|
goto unlock;
|
|
}
|
|
|
|
unlock:
|
|
|
|
current_sink = g_strdup (pulsesink->current_sink_name);
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
|
|
return current_sink;
|
|
|
|
/* ERRORS */
|
|
no_mainloop:
|
|
{
|
|
GST_DEBUG_OBJECT (pulsesink, "we have no mainloop");
|
|
return NULL;
|
|
}
|
|
no_buffer:
|
|
{
|
|
GST_DEBUG_OBJECT (pulsesink, "we have no ringbuffer");
|
|
return NULL;
|
|
}
|
|
info_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
|
|
("pa_context_get_sink_input_info() failed: %s",
|
|
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
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_AUDIO_BASE_SINK (psink)->ringbuffer);
|
|
if (pbuf == NULL)
|
|
goto no_buffer;
|
|
|
|
free_device_info (&psink->device_info);
|
|
if (!(o = pa_context_get_sink_info_by_name (pbuf->context,
|
|
psink->device, gst_pulsesink_sink_info_cb, &psink->device_info)))
|
|
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_info.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_stream_device (GstPulseSink * psink, const gchar * device)
|
|
{
|
|
pa_operation *o = NULL;
|
|
GstPulseRingBuffer *pbuf;
|
|
uint32_t idx;
|
|
|
|
if (!mainloop)
|
|
goto no_mainloop;
|
|
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_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_DEBUG_OBJECT (psink, "setting stream device to %s", device);
|
|
|
|
if (!(o = pa_context_move_sink_input_by_name (pbuf->context, idx, device,
|
|
NULL, NULL)))
|
|
goto move_failed;
|
|
|
|
unlock:
|
|
|
|
if (o)
|
|
pa_operation_unref (o);
|
|
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_mainloop:
|
|
{
|
|
GST_DEBUG_OBJECT (psink, "we have no mainloop");
|
|
return;
|
|
}
|
|
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");
|
|
return;
|
|
}
|
|
move_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
|
|
("pa_context_move_sink_input_by_name(%s) failed: %s", device,
|
|
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);
|
|
break;
|
|
case PROP_DEVICE:
|
|
g_free (pulsesink->device);
|
|
pulsesink->device = g_value_dup_string (value);
|
|
gst_pulsesink_set_stream_device (pulsesink, pulsesink->device);
|
|
break;
|
|
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;
|
|
case PROP_CLIENT_NAME:
|
|
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_CURRENT_DEVICE:
|
|
{
|
|
gchar *current_device = gst_pulsesink_get_current_device (pulsesink);
|
|
if (current_device)
|
|
g_value_take_string (value, current_device);
|
|
else
|
|
g_value_set_string (value, "");
|
|
break;
|
|
}
|
|
case PROP_DEVICE_NAME:
|
|
g_value_take_string (value, gst_pulsesink_device_description (pulsesink));
|
|
break;
|
|
case PROP_VOLUME:
|
|
{
|
|
gdouble volume;
|
|
|
|
gst_pulsesink_get_sink_input_info (pulsesink, &volume, NULL);
|
|
g_value_set_double (value, volume);
|
|
break;
|
|
}
|
|
case PROP_MUTE:
|
|
{
|
|
gboolean mute;
|
|
|
|
gst_pulsesink_get_sink_input_info (pulsesink, NULL, &mute);
|
|
g_value_set_boolean (value, mute);
|
|
break;
|
|
}
|
|
case PROP_CLIENT_NAME:
|
|
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_AUDIO_BASE_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;
|
|
}
|
|
}
|
|
|
|
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_AUDIO_BASE_SINK (psink)->ringbuffer);
|
|
if (pbuf == NULL || pbuf->stream == NULL)
|
|
goto no_buffer;
|
|
|
|
/* We're not interested if this operation failed or not */
|
|
if (!(o = pa_stream_proplist_update (pbuf->stream, PA_UPDATE_REPLACE,
|
|
pl, NULL, NULL))) {
|
|
GST_DEBUG_OBJECT (psink, "pa_stream_proplist_update() failed");
|
|
}
|
|
|
|
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;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_flush_ringbuffer (GstPulseSink * psink)
|
|
{
|
|
GstPulseRingBuffer *pbuf;
|
|
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
|
|
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_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);
|
|
|
|
gst_pulsesink_change_props (pulsesink, l);
|
|
|
|
break;
|
|
}
|
|
case GST_EVENT_GAP:{
|
|
GstClockTime timestamp, duration;
|
|
|
|
gst_event_parse_gap (event, ×tamp, &duration);
|
|
if (duration == GST_CLOCK_TIME_NONE)
|
|
gst_pulsesink_flush_ringbuffer (pulsesink);
|
|
break;
|
|
}
|
|
case GST_EVENT_EOS:
|
|
gst_pulsesink_flush_ringbuffer (pulsesink);
|
|
break;
|
|
default:
|
|
;
|
|
}
|
|
|
|
return GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
|
|
}
|
|
|
|
static gboolean
|
|
gst_pulsesink_query (GstBaseSink * sink, GstQuery * query)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK_CAST (sink);
|
|
gboolean ret = FALSE;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_CAPS:
|
|
{
|
|
GstCaps *caps, *filter;
|
|
|
|
gst_query_parse_caps (query, &filter);
|
|
caps = gst_pulsesink_query_getcaps (pulsesink, filter);
|
|
|
|
if (caps) {
|
|
gst_query_set_caps_result (query, caps);
|
|
gst_caps_unref (caps);
|
|
ret = TRUE;
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_ACCEPT_CAPS:
|
|
{
|
|
GstCaps *caps;
|
|
|
|
gst_query_parse_accept_caps (query, &caps);
|
|
ret = gst_pulsesink_query_acceptcaps (pulsesink, caps);
|
|
gst_query_set_accept_caps_result (query, ret);
|
|
ret = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
ret = GST_BASE_SINK_CLASS (parent_class)->query (sink, query);
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_pulsesink_release_mainloop (GstPulseSink * psink)
|
|
{
|
|
if (!mainloop)
|
|
return;
|
|
|
|
pa_threaded_mainloop_lock (mainloop);
|
|
while (psink->defer_pending) {
|
|
GST_DEBUG_OBJECT (psink, "waiting for stream status message emission");
|
|
pa_threaded_mainloop_wait (mainloop);
|
|
}
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
|
|
g_mutex_lock (&pa_shared_resource_mutex);
|
|
mainloop_ref_ct--;
|
|
if (!mainloop_ref_ct) {
|
|
GST_INFO_OBJECT (psink, "terminating pa main loop thread");
|
|
pa_threaded_mainloop_stop (mainloop);
|
|
pa_threaded_mainloop_free (mainloop);
|
|
mainloop = NULL;
|
|
}
|
|
g_mutex_unlock (&pa_shared_resource_mutex);
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_pulsesink_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstPulseSink *pulsesink = GST_PULSESINK (element);
|
|
GstStateChangeReturn ret;
|
|
|
|
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;
|
|
if (pa_threaded_mainloop_start (mainloop) < 0) {
|
|
pa_threaded_mainloop_free (mainloop);
|
|
goto mainloop_start_failed;
|
|
}
|
|
mainloop_ref_ct = 1;
|
|
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_AUDIO_BASE_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:
|
|
/* format_lost is reset in release() in audiobasesink */
|
|
gst_element_post_message (element,
|
|
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
|
|
GST_AUDIO_BASE_SINK (pulsesink)->provided_clock));
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
gst_pulsesink_release_mainloop (pulsesink);
|
|
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;
|
|
}
|
|
mainloop_start_failed:
|
|
{
|
|
g_mutex_unlock (&pa_shared_resource_mutex);
|
|
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
|
|
("pa_threaded_mainloop_start() failed"), (NULL));
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
state_failure:
|
|
{
|
|
if (transition == GST_STATE_CHANGE_NULL_TO_READY) {
|
|
/* Clear the PA mainloop if audiobasesink failed to open the ring_buffer */
|
|
g_assert (mainloop);
|
|
gst_pulsesink_release_mainloop (pulsesink);
|
|
}
|
|
return ret;
|
|
}
|
|
}
|