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On Windows, the ringbuffer thread function must have the "Pro Audio" priority set, otherwise it sometimes doesn't get scheduled for 200-300ms, which will immediately cause an underrun unless you set a very high latency-time and buffer-time. This has no compile-time deps since it tries to load avrt.dll at runtime to set the thread priority.
634 lines
18 KiB
C
634 lines
18 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2005 Wim Taymans <wim@fluendo.com>
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*
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* gstaudiosink.c: simple audio sink base class
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstaudiosink
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* @title: GstAudioSink
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* @short_description: Simple base class for audio sinks
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* @see_also: #GstAudioBaseSink, #GstAudioRingBuffer, #GstAudioSink.
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*
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* This is the most simple base class for audio sinks that only requires
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* subclasses to implement a set of simple functions:
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*
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* * `open()` :Open the device.
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*
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* * `prepare()` :Configure the device with the specified format.
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*
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* * `write()` :Write samples to the device.
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*
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* * `reset()` :Unblock writes and flush the device.
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*
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* * `delay()` :Get the number of samples written but not yet played
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* by the device.
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*
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* * `unprepare()` :Undo operations done by prepare.
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*
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* * `close()` :Close the device.
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*
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* All scheduling of samples and timestamps is done in this base class
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* together with #GstAudioBaseSink using a default implementation of a
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* #GstAudioRingBuffer that uses threads.
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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 <gst/audio/audio.h>
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#include "gstaudiosink.h"
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#include "gstaudioutilsprivate.h"
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GST_DEBUG_CATEGORY_STATIC (gst_audio_sink_debug);
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#define GST_CAT_DEFAULT gst_audio_sink_debug
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#define GST_TYPE_AUDIO_SINK_RING_BUFFER \
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(gst_audio_sink_ring_buffer_get_type())
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#define GST_AUDIO_SINK_RING_BUFFER(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_SINK_RING_BUFFER,GstAudioSinkRingBuffer))
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#define GST_AUDIO_SINK_RING_BUFFER_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_SINK_RING_BUFFER,GstAudioSinkRingBufferClass))
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#define GST_AUDIO_SINK_RING_BUFFER_GET_CLASS(obj) \
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(G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_SINK_RING_BUFFER, GstAudioSinkRingBufferClass))
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#define GST_AUDIO_SINK_RING_BUFFER_CAST(obj) \
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((GstAudioSinkRingBuffer *)obj)
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#define GST_IS_AUDIO_SINK_RING_BUFFER(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_SINK_RING_BUFFER))
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#define GST_IS_AUDIO_SINK_RING_BUFFER_CLASS(klass)\
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_SINK_RING_BUFFER))
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typedef struct _GstAudioSinkRingBuffer GstAudioSinkRingBuffer;
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typedef struct _GstAudioSinkRingBufferClass GstAudioSinkRingBufferClass;
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#define GST_AUDIO_SINK_RING_BUFFER_GET_COND(buf) (&(((GstAudioSinkRingBuffer *)buf)->cond))
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#define GST_AUDIO_SINK_RING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIO_SINK_RING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf)))
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#define GST_AUDIO_SINK_RING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIO_SINK_RING_BUFFER_GET_COND (buf)))
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#define GST_AUDIO_SINK_RING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIO_SINK_RING_BUFFER_GET_COND (buf)))
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struct _GstAudioSinkRingBuffer
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{
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GstAudioRingBuffer object;
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gboolean running;
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gint queuedseg;
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GCond cond;
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};
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struct _GstAudioSinkRingBufferClass
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{
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GstAudioRingBufferClass parent_class;
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};
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static void gst_audio_sink_ring_buffer_class_init (GstAudioSinkRingBufferClass *
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klass);
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static void gst_audio_sink_ring_buffer_init (GstAudioSinkRingBuffer *
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ringbuffer, GstAudioSinkRingBufferClass * klass);
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static void gst_audio_sink_ring_buffer_dispose (GObject * object);
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static void gst_audio_sink_ring_buffer_finalize (GObject * object);
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static GstAudioRingBufferClass *ring_parent_class = NULL;
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static gboolean gst_audio_sink_ring_buffer_open_device (GstAudioRingBuffer *
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buf);
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static gboolean gst_audio_sink_ring_buffer_close_device (GstAudioRingBuffer *
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buf);
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static gboolean gst_audio_sink_ring_buffer_acquire (GstAudioRingBuffer * buf,
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GstAudioRingBufferSpec * spec);
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static gboolean gst_audio_sink_ring_buffer_release (GstAudioRingBuffer * buf);
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static gboolean gst_audio_sink_ring_buffer_start (GstAudioRingBuffer * buf);
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static gboolean gst_audio_sink_ring_buffer_pause (GstAudioRingBuffer * buf);
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static gboolean gst_audio_sink_ring_buffer_stop (GstAudioRingBuffer * buf);
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static guint gst_audio_sink_ring_buffer_delay (GstAudioRingBuffer * buf);
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static gboolean gst_audio_sink_ring_buffer_activate (GstAudioRingBuffer * buf,
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gboolean active);
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/* ringbuffer abstract base class */
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static GType
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gst_audio_sink_ring_buffer_get_type (void)
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{
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static GType ringbuffer_type = 0;
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if (!ringbuffer_type) {
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static const GTypeInfo ringbuffer_info = {
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sizeof (GstAudioSinkRingBufferClass),
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NULL,
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NULL,
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(GClassInitFunc) gst_audio_sink_ring_buffer_class_init,
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NULL,
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NULL,
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sizeof (GstAudioSinkRingBuffer),
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0,
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(GInstanceInitFunc) gst_audio_sink_ring_buffer_init,
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NULL
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};
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ringbuffer_type =
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g_type_register_static (GST_TYPE_AUDIO_RING_BUFFER,
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"GstAudioSinkRingBuffer", &ringbuffer_info, 0);
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}
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return ringbuffer_type;
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}
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static void
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gst_audio_sink_ring_buffer_class_init (GstAudioSinkRingBufferClass * 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->dispose = gst_audio_sink_ring_buffer_dispose;
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gobject_class->finalize = gst_audio_sink_ring_buffer_finalize;
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gstringbuffer_class->open_device =
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GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_open_device);
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gstringbuffer_class->close_device =
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GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_close_device);
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gstringbuffer_class->acquire =
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GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_acquire);
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gstringbuffer_class->release =
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GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_release);
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gstringbuffer_class->start =
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GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_start);
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gstringbuffer_class->pause =
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GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_pause);
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gstringbuffer_class->resume =
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GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_start);
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gstringbuffer_class->stop =
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GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_stop);
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gstringbuffer_class->delay =
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GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_delay);
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gstringbuffer_class->activate =
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GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_activate);
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}
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typedef gint (*WriteFunc) (GstAudioSink * sink, gpointer data, guint length);
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/* this internal thread does nothing else but write samples to the audio device.
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* It will write each segment in the ringbuffer and will update the play
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* pointer.
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* The start/stop methods control the thread.
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*/
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static void
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audioringbuffer_thread_func (GstAudioRingBuffer * buf)
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{
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GstAudioSink *sink;
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GstAudioSinkClass *csink;
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GstAudioSinkRingBuffer *abuf = GST_AUDIO_SINK_RING_BUFFER_CAST (buf);
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WriteFunc writefunc;
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GstMessage *message;
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GValue val = { 0 };
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sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
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csink = GST_AUDIO_SINK_GET_CLASS (sink);
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GST_DEBUG_OBJECT (sink, "enter thread");
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GST_OBJECT_LOCK (abuf);
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GST_DEBUG_OBJECT (sink, "signal wait");
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GST_AUDIO_SINK_RING_BUFFER_SIGNAL (buf);
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GST_OBJECT_UNLOCK (abuf);
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writefunc = csink->write;
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if (writefunc == NULL)
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goto no_function;
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if (G_UNLIKELY (!__gst_audio_set_thread_priority ()))
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GST_WARNING_OBJECT (sink, "failed to set thread priority");
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message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
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GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT_CAST (sink));
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g_value_init (&val, GST_TYPE_G_THREAD);
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g_value_set_boxed (&val, g_thread_self ());
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gst_message_set_stream_status_object (message, &val);
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g_value_unset (&val);
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GST_DEBUG_OBJECT (sink, "posting ENTER stream status");
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gst_element_post_message (GST_ELEMENT_CAST (sink), message);
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while (TRUE) {
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gint left, len;
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guint8 *readptr;
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gint readseg;
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/* buffer must be started */
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if (gst_audio_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
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gint written;
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left = len;
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do {
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written = writefunc (sink, readptr, left);
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GST_LOG_OBJECT (sink, "transfered %d bytes of %d from segment %d",
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written, left, readseg);
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if (written < 0 || written > left) {
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/* might not be critical, it e.g. happens when aborting playback */
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GST_WARNING_OBJECT (sink,
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"error writing data in %s (reason: %s), skipping segment (left: %d, written: %d)",
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GST_DEBUG_FUNCPTR_NAME (writefunc),
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(errno > 1 ? g_strerror (errno) : "unknown"), left, written);
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break;
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}
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left -= written;
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readptr += written;
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} while (left > 0);
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/* clear written samples */
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gst_audio_ring_buffer_clear (buf, readseg);
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/* we wrote one segment */
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gst_audio_ring_buffer_advance (buf, 1);
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} else {
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GST_OBJECT_LOCK (abuf);
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if (!abuf->running)
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goto stop_running;
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if (G_UNLIKELY (g_atomic_int_get (&buf->state) ==
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GST_AUDIO_RING_BUFFER_STATE_STARTED)) {
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GST_OBJECT_UNLOCK (abuf);
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continue;
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}
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GST_DEBUG_OBJECT (sink, "signal wait");
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GST_AUDIO_SINK_RING_BUFFER_SIGNAL (buf);
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GST_DEBUG_OBJECT (sink, "wait for action");
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GST_AUDIO_SINK_RING_BUFFER_WAIT (buf);
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GST_DEBUG_OBJECT (sink, "got signal");
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if (!abuf->running)
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goto stop_running;
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GST_DEBUG_OBJECT (sink, "continue running");
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GST_OBJECT_UNLOCK (abuf);
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}
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}
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/* Will never be reached */
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g_assert_not_reached ();
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return;
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/* ERROR */
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no_function:
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{
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GST_DEBUG_OBJECT (sink, "no write function, exit thread");
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return;
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}
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stop_running:
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{
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GST_OBJECT_UNLOCK (abuf);
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GST_DEBUG_OBJECT (sink, "stop running, exit thread");
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message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
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GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT_CAST (sink));
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g_value_init (&val, GST_TYPE_G_THREAD);
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g_value_set_boxed (&val, g_thread_self ());
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gst_message_set_stream_status_object (message, &val);
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g_value_unset (&val);
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GST_DEBUG_OBJECT (sink, "posting LEAVE stream status");
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gst_element_post_message (GST_ELEMENT_CAST (sink), message);
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return;
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}
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}
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static void
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gst_audio_sink_ring_buffer_init (GstAudioSinkRingBuffer * ringbuffer,
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GstAudioSinkRingBufferClass * g_class)
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{
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ringbuffer->running = FALSE;
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ringbuffer->queuedseg = 0;
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g_cond_init (&ringbuffer->cond);
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}
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static void
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gst_audio_sink_ring_buffer_dispose (GObject * object)
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{
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G_OBJECT_CLASS (ring_parent_class)->dispose (object);
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}
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static void
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gst_audio_sink_ring_buffer_finalize (GObject * object)
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{
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GstAudioSinkRingBuffer *ringbuffer = GST_AUDIO_SINK_RING_BUFFER_CAST (object);
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g_cond_clear (&ringbuffer->cond);
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G_OBJECT_CLASS (ring_parent_class)->finalize (object);
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}
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static gboolean
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gst_audio_sink_ring_buffer_open_device (GstAudioRingBuffer * buf)
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{
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GstAudioSink *sink;
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GstAudioSinkClass *csink;
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gboolean result = TRUE;
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sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
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csink = GST_AUDIO_SINK_GET_CLASS (sink);
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if (csink->open)
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result = csink->open (sink);
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if (!result)
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goto could_not_open;
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return result;
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could_not_open:
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{
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GST_DEBUG_OBJECT (sink, "could not open device");
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return FALSE;
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}
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}
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static gboolean
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gst_audio_sink_ring_buffer_close_device (GstAudioRingBuffer * buf)
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{
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GstAudioSink *sink;
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GstAudioSinkClass *csink;
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gboolean result = TRUE;
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sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
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csink = GST_AUDIO_SINK_GET_CLASS (sink);
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if (csink->close)
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result = csink->close (sink);
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if (!result)
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goto could_not_close;
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return result;
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could_not_close:
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{
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GST_DEBUG_OBJECT (sink, "could not close device");
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return FALSE;
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}
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}
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static gboolean
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gst_audio_sink_ring_buffer_acquire (GstAudioRingBuffer * buf,
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GstAudioRingBufferSpec * spec)
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{
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GstAudioSink *sink;
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GstAudioSinkClass *csink;
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gboolean result = FALSE;
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sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
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csink = GST_AUDIO_SINK_GET_CLASS (sink);
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if (csink->prepare)
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result = csink->prepare (sink, spec);
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if (!result)
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goto could_not_prepare;
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/* set latency to one more segment as we need some headroom */
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spec->seglatency = spec->segtotal + 1;
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buf->size = spec->segtotal * spec->segsize;
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buf->memory = g_malloc (buf->size);
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if (buf->spec.type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW) {
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gst_audio_format_fill_silence (buf->spec.info.finfo, buf->memory,
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buf->size);
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} else {
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/* FIXME, non-raw formats get 0 as the empty sample */
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memset (buf->memory, 0, buf->size);
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}
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return TRUE;
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/* ERRORS */
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could_not_prepare:
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{
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GST_DEBUG_OBJECT (sink, "could not prepare device");
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return FALSE;
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}
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}
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static gboolean
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gst_audio_sink_ring_buffer_activate (GstAudioRingBuffer * buf, gboolean active)
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{
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GstAudioSink *sink;
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GstAudioSinkRingBuffer *abuf;
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GError *error = NULL;
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sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
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abuf = GST_AUDIO_SINK_RING_BUFFER_CAST (buf);
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if (active) {
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abuf->running = TRUE;
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GST_DEBUG_OBJECT (sink, "starting thread");
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sink->thread = g_thread_try_new ("audiosink-ringbuffer",
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(GThreadFunc) audioringbuffer_thread_func, buf, &error);
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if (!sink->thread || error != NULL)
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goto thread_failed;
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GST_DEBUG_OBJECT (sink, "waiting for thread");
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/* the object lock is taken */
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GST_AUDIO_SINK_RING_BUFFER_WAIT (buf);
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GST_DEBUG_OBJECT (sink, "thread is started");
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} else {
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abuf->running = FALSE;
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GST_DEBUG_OBJECT (sink, "signal wait");
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GST_AUDIO_SINK_RING_BUFFER_SIGNAL (buf);
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GST_OBJECT_UNLOCK (buf);
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/* join the thread */
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g_thread_join (sink->thread);
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GST_OBJECT_LOCK (buf);
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}
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return TRUE;
|
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|
|
/* ERRORS */
|
|
thread_failed:
|
|
{
|
|
if (error)
|
|
GST_ERROR_OBJECT (sink, "could not create thread %s", error->message);
|
|
else
|
|
GST_ERROR_OBJECT (sink, "could not create thread for unknown reason");
|
|
g_clear_error (&error);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* function is called with LOCK */
|
|
static gboolean
|
|
gst_audio_sink_ring_buffer_release (GstAudioRingBuffer * buf)
|
|
{
|
|
GstAudioSink *sink;
|
|
GstAudioSinkClass *csink;
|
|
gboolean result = FALSE;
|
|
|
|
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
|
|
csink = GST_AUDIO_SINK_GET_CLASS (sink);
|
|
|
|
/* free the buffer */
|
|
g_free (buf->memory);
|
|
buf->memory = NULL;
|
|
|
|
if (csink->unprepare)
|
|
result = csink->unprepare (sink);
|
|
|
|
if (!result)
|
|
goto could_not_unprepare;
|
|
|
|
GST_DEBUG_OBJECT (sink, "unprepared");
|
|
|
|
return result;
|
|
|
|
could_not_unprepare:
|
|
{
|
|
GST_DEBUG_OBJECT (sink, "could not unprepare device");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_audio_sink_ring_buffer_start (GstAudioRingBuffer * buf)
|
|
{
|
|
GstAudioSink *sink;
|
|
|
|
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
|
|
|
|
GST_DEBUG_OBJECT (sink, "start, sending signal");
|
|
GST_AUDIO_SINK_RING_BUFFER_SIGNAL (buf);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_audio_sink_ring_buffer_pause (GstAudioRingBuffer * buf)
|
|
{
|
|
GstAudioSink *sink;
|
|
GstAudioSinkClass *csink;
|
|
|
|
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
|
|
csink = GST_AUDIO_SINK_GET_CLASS (sink);
|
|
|
|
/* unblock any pending writes to the audio device */
|
|
if (csink->reset) {
|
|
GST_DEBUG_OBJECT (sink, "reset...");
|
|
csink->reset (sink);
|
|
GST_DEBUG_OBJECT (sink, "reset done");
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_audio_sink_ring_buffer_stop (GstAudioRingBuffer * buf)
|
|
{
|
|
GstAudioSink *sink;
|
|
GstAudioSinkClass *csink;
|
|
|
|
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
|
|
csink = GST_AUDIO_SINK_GET_CLASS (sink);
|
|
|
|
/* unblock any pending writes to the audio device */
|
|
if (csink->reset) {
|
|
GST_DEBUG_OBJECT (sink, "reset...");
|
|
csink->reset (sink);
|
|
GST_DEBUG_OBJECT (sink, "reset done");
|
|
}
|
|
#if 0
|
|
if (abuf->running) {
|
|
GST_DEBUG_OBJECT (sink, "stop, waiting...");
|
|
GST_AUDIO_SINK_RING_BUFFER_WAIT (buf);
|
|
GST_DEBUG_OBJECT (sink, "stopped");
|
|
}
|
|
#endif
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static guint
|
|
gst_audio_sink_ring_buffer_delay (GstAudioRingBuffer * buf)
|
|
{
|
|
GstAudioSink *sink;
|
|
GstAudioSinkClass *csink;
|
|
guint res = 0;
|
|
|
|
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
|
|
csink = GST_AUDIO_SINK_GET_CLASS (sink);
|
|
|
|
if (csink->delay)
|
|
res = csink->delay (sink);
|
|
|
|
return res;
|
|
}
|
|
|
|
/* AudioSink signals and args */
|
|
enum
|
|
{
|
|
/* FILL ME */
|
|
LAST_SIGNAL
|
|
};
|
|
|
|
enum
|
|
{
|
|
ARG_0,
|
|
};
|
|
|
|
#define _do_init \
|
|
GST_DEBUG_CATEGORY_INIT (gst_audio_sink_debug, "audiosink", 0, "audiosink element");
|
|
#define gst_audio_sink_parent_class parent_class
|
|
G_DEFINE_TYPE_WITH_CODE (GstAudioSink, gst_audio_sink,
|
|
GST_TYPE_AUDIO_BASE_SINK, _do_init);
|
|
|
|
static GstAudioRingBuffer *gst_audio_sink_create_ringbuffer (GstAudioBaseSink *
|
|
sink);
|
|
|
|
static void
|
|
gst_audio_sink_class_init (GstAudioSinkClass * klass)
|
|
{
|
|
GstAudioBaseSinkClass *gstaudiobasesink_class;
|
|
|
|
gstaudiobasesink_class = (GstAudioBaseSinkClass *) klass;
|
|
|
|
gstaudiobasesink_class->create_ringbuffer =
|
|
GST_DEBUG_FUNCPTR (gst_audio_sink_create_ringbuffer);
|
|
|
|
g_type_class_ref (GST_TYPE_AUDIO_SINK_RING_BUFFER);
|
|
}
|
|
|
|
static void
|
|
gst_audio_sink_init (GstAudioSink * audiosink)
|
|
{
|
|
}
|
|
|
|
static GstAudioRingBuffer *
|
|
gst_audio_sink_create_ringbuffer (GstAudioBaseSink * sink)
|
|
{
|
|
GstAudioRingBuffer *buffer;
|
|
|
|
GST_DEBUG_OBJECT (sink, "creating ringbuffer");
|
|
buffer = g_object_new (GST_TYPE_AUDIO_SINK_RING_BUFFER, NULL);
|
|
GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
|
|
|
|
return buffer;
|
|
}
|