gstreamer/gst-libs/gst/audio/gstaudiosink.c
Wim Taymans fc523e047c gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
Choose to allocate one less segment but require one additional segment
as latency.
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
No need to increment the number of segments in the source.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (clock_convert_external),
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
Remove adding latency when returning the internal time while subtracting
it again when we use the value a little later.
When calculating the end timestamp, we are making a rounding error
with the current algorithm. Ensure that we don't accumulate these
rounding errors when aligning samples by not resampling at all if we
don't need to. Fixes #419351.
Make the initial calibration of the clock slaving a little more
predictable and accurate. Also handle the case where we don't do
clock slaving.
2008-05-09 16:38:10 +00:00

569 lines
15 KiB
C

/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstaudiosink.c: simple audio sink base class
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:gstaudiosink
* @short_description: Simple base class for audio sinks
* @see_also: #GstBaseAudioSink, #GstRingBuffer, #GstAudioSink.
*
* This is the most simple base class for audio sinks that only requires
* subclasses to implement a set of simple functions:
*
* <variablelist>
* <varlistentry>
* <term>open()</term>
* <listitem><para>Open the device.</para></listitem>
* </varlistentry>
* <varlistentry>
* <term>prepare()</term>
* <listitem><para>Configure the device with the specified format.</para></listitem>
* </varlistentry>
* <varlistentry>
* <term>write()</term>
* <listitem><para>Write samples to the device.</para></listitem>
* </varlistentry>
* <varlistentry>
* <term>reset()</term>
* <listitem><para>Unblock writes and flush the device.</para></listitem>
* </varlistentry>
* <varlistentry>
* <term>delay()</term>
* <listitem><para>Get the number of samples written but not yet played
* by the device.</para></listitem>
* </varlistentry>
* <varlistentry>
* <term>unprepare()</term>
* <listitem><para>Undo operations done by prepare.</para></listitem>
* </varlistentry>
* <varlistentry>
* <term>close()</term>
* <listitem><para>Close the device.</para></listitem>
* </varlistentry>
* </variablelist>
*
* All scheduling of samples and timestamps is done in this base class
* together with #GstBaseAudioSink using a default implementation of a
* #GstRingBuffer that uses threads.
*
* Last reviewed on 2006-09-27 (0.10.12)
*/
#include <string.h>
#include "gstaudiosink.h"
GST_DEBUG_CATEGORY_STATIC (gst_audio_sink_debug);
#define GST_CAT_DEFAULT gst_audio_sink_debug
#define GST_TYPE_AUDIORING_BUFFER \
(gst_audioringbuffer_get_type())
#define GST_AUDIORING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORING_BUFFER,GstAudioRingBuffer))
#define GST_AUDIORING_BUFFER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORING_BUFFER,GstAudioRingBufferClass))
#define GST_AUDIORING_BUFFER_GET_CLASS(obj) \
(G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIORING_BUFFER, GstAudioRingBufferClass))
#define GST_AUDIORING_BUFFER_CAST(obj) \
((GstAudioRingBuffer *)obj)
#define GST_IS_AUDIORING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORING_BUFFER))
#define GST_IS_AUDIORING_BUFFER_CLASS(klass)\
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORING_BUFFER))
typedef struct _GstAudioRingBuffer GstAudioRingBuffer;
typedef struct _GstAudioRingBufferClass GstAudioRingBufferClass;
#define GST_AUDIORING_BUFFER_GET_COND(buf) (((GstAudioRingBuffer *)buf)->cond)
#define GST_AUDIORING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIORING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf)))
#define GST_AUDIORING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIORING_BUFFER_GET_COND (buf)))
#define GST_AUDIORING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIORING_BUFFER_GET_COND (buf)))
struct _GstAudioRingBuffer
{
GstRingBuffer object;
gboolean running;
gint queuedseg;
GCond *cond;
};
struct _GstAudioRingBufferClass
{
GstRingBufferClass parent_class;
};
static void gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass);
static void gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer,
GstAudioRingBufferClass * klass);
static void gst_audioringbuffer_dispose (GObject * object);
static void gst_audioringbuffer_finalize (GObject * object);
static GstRingBufferClass *ring_parent_class = NULL;
static gboolean gst_audioringbuffer_open_device (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_close_device (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_acquire (GstRingBuffer * buf,
GstRingBufferSpec * spec);
static gboolean gst_audioringbuffer_release (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_start (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_pause (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_stop (GstRingBuffer * buf);
static guint gst_audioringbuffer_delay (GstRingBuffer * buf);
/* ringbuffer abstract base class */
static GType
gst_audioringbuffer_get_type (void)
{
static GType ringbuffer_type = 0;
if (!ringbuffer_type) {
static const GTypeInfo ringbuffer_info = {
sizeof (GstAudioRingBufferClass),
NULL,
NULL,
(GClassInitFunc) gst_audioringbuffer_class_init,
NULL,
NULL,
sizeof (GstAudioRingBuffer),
0,
(GInstanceInitFunc) gst_audioringbuffer_init,
NULL
};
ringbuffer_type =
g_type_register_static (GST_TYPE_RING_BUFFER, "GstAudioSinkRingBuffer",
&ringbuffer_info, 0);
}
return ringbuffer_type;
}
static void
gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstObjectClass *gstobject_class;
GstRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstobject_class = (GstObjectClass *) klass;
gstringbuffer_class = (GstRingBufferClass *) klass;
ring_parent_class = g_type_class_peek_parent (klass);
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_audioringbuffer_dispose);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audioringbuffer_finalize);
gstringbuffer_class->open_device =
GST_DEBUG_FUNCPTR (gst_audioringbuffer_open_device);
gstringbuffer_class->close_device =
GST_DEBUG_FUNCPTR (gst_audioringbuffer_close_device);
gstringbuffer_class->acquire =
GST_DEBUG_FUNCPTR (gst_audioringbuffer_acquire);
gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_audioringbuffer_release);
gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start);
gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_audioringbuffer_pause);
gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start);
gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_audioringbuffer_stop);
gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_audioringbuffer_delay);
}
typedef guint (*WriteFunc) (GstAudioSink * sink, gpointer data, guint length);
/* this internal thread does nothing else but write samples to the audio device.
* It will write each segment in the ringbuffer and will update the play
* pointer.
* The start/stop methods control the thread.
*/
static void
audioringbuffer_thread_func (GstRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
GstAudioRingBuffer *abuf = GST_AUDIORING_BUFFER_CAST (buf);
WriteFunc writefunc;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
GST_DEBUG_OBJECT (sink, "enter thread");
writefunc = csink->write;
if (writefunc == NULL)
goto no_function;
while (TRUE) {
gint left, len;
guint8 *readptr;
gint readseg;
if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
gint written = 0;
left = len;
do {
written = writefunc (sink, readptr + written, left);
GST_LOG_OBJECT (sink, "transfered %d bytes of %d from segment %d",
written, left, readseg);
if (written < 0 || written > left) {
GST_WARNING_OBJECT (sink,
"error writing data (reason: %s), skipping segment",
g_strerror (errno));
break;
}
left -= written;
} while (left > 0);
/* clear written samples */
gst_ring_buffer_clear (buf, readseg);
/* we wrote one segment */
gst_ring_buffer_advance (buf, 1);
} else {
GST_OBJECT_LOCK (abuf);
if (!abuf->running)
goto stop_running;
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIORING_BUFFER_SIGNAL (buf);
GST_DEBUG_OBJECT (sink, "wait for action");
GST_AUDIORING_BUFFER_WAIT (buf);
GST_DEBUG_OBJECT (sink, "got signal");
if (!abuf->running)
goto stop_running;
GST_DEBUG_OBJECT (sink, "continue running");
GST_OBJECT_UNLOCK (abuf);
}
}
/* Will never be reached */
return;
/* ERROR */
no_function:
{
GST_DEBUG_OBJECT (sink, "no write function, exit thread");
return;
}
stop_running:
{
GST_OBJECT_UNLOCK (abuf);
GST_DEBUG_OBJECT (sink, "stop running, exit thread");
return;
}
}
static void
gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer,
GstAudioRingBufferClass * g_class)
{
ringbuffer->running = FALSE;
ringbuffer->queuedseg = 0;
ringbuffer->cond = g_cond_new ();
}
static void
gst_audioringbuffer_dispose (GObject * object)
{
G_OBJECT_CLASS (ring_parent_class)->dispose (object);
}
static void
gst_audioringbuffer_finalize (GObject * object)
{
GstAudioRingBuffer *ringbuffer = GST_AUDIORING_BUFFER_CAST (object);
g_cond_free (ringbuffer->cond);
G_OBJECT_CLASS (ring_parent_class)->finalize (object);
}
static gboolean
gst_audioringbuffer_open_device (GstRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
gboolean result = TRUE;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
if (csink->open)
result = csink->open (sink);
if (!result)
goto could_not_open;
return result;
could_not_open:
{
GST_DEBUG_OBJECT (sink, "could not open device");
return FALSE;
}
}
static gboolean
gst_audioringbuffer_close_device (GstRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
gboolean result = TRUE;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
if (csink->close)
result = csink->close (sink);
if (!result)
goto could_not_close;
return result;
could_not_close:
{
GST_DEBUG_OBJECT (sink, "could not close device");
return FALSE;
}
}
static gboolean
gst_audioringbuffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
GstAudioRingBuffer *abuf;
gboolean result = FALSE;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
if (csink->prepare)
result = csink->prepare (sink, spec);
if (!result)
goto could_not_prepare;
/* set latency to one more segment as we need some headroom */
spec->seglatency = spec->segtotal + 1;
buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
abuf = GST_AUDIORING_BUFFER_CAST (buf);
abuf->running = TRUE;
sink->thread =
g_thread_create ((GThreadFunc) audioringbuffer_thread_func, buf, TRUE,
NULL);
GST_AUDIORING_BUFFER_WAIT (buf);
return result;
could_not_prepare:
{
GST_DEBUG_OBJECT (sink, "could not prepare device");
return FALSE;
}
}
/* function is called with LOCK */
static gboolean
gst_audioringbuffer_release (GstRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
GstAudioRingBuffer *abuf;
gboolean result = FALSE;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
abuf = GST_AUDIORING_BUFFER_CAST (buf);
abuf->running = FALSE;
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIORING_BUFFER_SIGNAL (buf);
GST_OBJECT_UNLOCK (buf);
/* join the thread */
g_thread_join (sink->thread);
GST_OBJECT_LOCK (buf);
/* free the buffer */
gst_buffer_unref (buf->data);
buf->data = 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_audioringbuffer_start (GstRingBuffer * buf)
{
GstAudioSink *sink;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "start, sending signal");
GST_AUDIORING_BUFFER_SIGNAL (buf);
return TRUE;
}
static gboolean
gst_audioringbuffer_pause (GstRingBuffer * 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_audioringbuffer_stop (GstRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
GstAudioRingBuffer *abuf;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
abuf = GST_AUDIORING_BUFFER_CAST (buf);
/* 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 (abuf->running) {
GST_DEBUG_OBJECT (sink, "stop, waiting...");
GST_AUDIORING_BUFFER_WAIT (buf);
GST_DEBUG_OBJECT (sink, "stopped");
}
return TRUE;
}
static guint
gst_audioringbuffer_delay (GstRingBuffer * 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(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_sink_debug, "audiosink", 0, "audiosink element");
GST_BOILERPLATE_FULL (GstAudioSink, gst_audio_sink, GstBaseAudioSink,
GST_TYPE_BASE_AUDIO_SINK, _do_init);
static GstRingBuffer *gst_audio_sink_create_ringbuffer (GstBaseAudioSink *
sink);
static void
gst_audio_sink_base_init (gpointer g_class)
{
}
static void
gst_audio_sink_class_init (GstAudioSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstBaseAudioSinkClass *gstbaseaudiosink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
gstbaseaudiosink_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_audio_sink_create_ringbuffer);
}
static void
gst_audio_sink_init (GstAudioSink * audiosink, GstAudioSinkClass * g_class)
{
}
static GstRingBuffer *
gst_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
{
GstRingBuffer *buffer;
GST_DEBUG_OBJECT (sink, "creating ringbuffer");
buffer = g_object_new (GST_TYPE_AUDIORING_BUFFER, NULL);
GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
return buffer;
}