gstreamer/gst-libs/gst/audio/gstaudiosink.c
Tim-Philipp Müller bcb8068e27 docs: remove outdated and pointless 'Last reviewed' lines from docs
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
2014-04-26 23:28:57 +01:00

632 lines
18 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:gstaudiosink
* @short_description: Simple base class for audio sinks
* @see_also: #GstAudioBaseSink, #GstAudioRingBuffer, #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 #GstAudioBaseSink using a default implementation of a
* #GstAudioRingBuffer that uses threads.
*/
#include <string.h>
#include <gst/audio/audio.h>
#include "gstaudiosink.h"
GST_DEBUG_CATEGORY_STATIC (gst_audio_sink_debug);
#define GST_CAT_DEFAULT gst_audio_sink_debug
#define GST_TYPE_AUDIO_SINK_RING_BUFFER \
(gst_audio_sink_ring_buffer_get_type())
#define GST_AUDIO_SINK_RING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_SINK_RING_BUFFER,GstAudioSinkRingBuffer))
#define GST_AUDIO_SINK_RING_BUFFER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_SINK_RING_BUFFER,GstAudioSinkRingBufferClass))
#define GST_AUDIO_SINK_RING_BUFFER_GET_CLASS(obj) \
(G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_SINK_RING_BUFFER, GstAudioSinkRingBufferClass))
#define GST_AUDIO_SINK_RING_BUFFER_CAST(obj) \
((GstAudioSinkRingBuffer *)obj)
#define GST_IS_AUDIO_SINK_RING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_SINK_RING_BUFFER))
#define GST_IS_AUDIO_SINK_RING_BUFFER_CLASS(klass)\
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_SINK_RING_BUFFER))
typedef struct _GstAudioSinkRingBuffer GstAudioSinkRingBuffer;
typedef struct _GstAudioSinkRingBufferClass GstAudioSinkRingBufferClass;
#define GST_AUDIO_SINK_RING_BUFFER_GET_COND(buf) (&(((GstAudioSinkRingBuffer *)buf)->cond))
#define GST_AUDIO_SINK_RING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIO_SINK_RING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf)))
#define GST_AUDIO_SINK_RING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIO_SINK_RING_BUFFER_GET_COND (buf)))
#define GST_AUDIO_SINK_RING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIO_SINK_RING_BUFFER_GET_COND (buf)))
struct _GstAudioSinkRingBuffer
{
GstAudioRingBuffer object;
gboolean running;
gint queuedseg;
GCond cond;
};
struct _GstAudioSinkRingBufferClass
{
GstAudioRingBufferClass parent_class;
};
static void gst_audio_sink_ring_buffer_class_init (GstAudioSinkRingBufferClass *
klass);
static void gst_audio_sink_ring_buffer_init (GstAudioSinkRingBuffer *
ringbuffer, GstAudioSinkRingBufferClass * klass);
static void gst_audio_sink_ring_buffer_dispose (GObject * object);
static void gst_audio_sink_ring_buffer_finalize (GObject * object);
static GstAudioRingBufferClass *ring_parent_class = NULL;
static gboolean gst_audio_sink_ring_buffer_open_device (GstAudioRingBuffer *
buf);
static gboolean gst_audio_sink_ring_buffer_close_device (GstAudioRingBuffer *
buf);
static gboolean gst_audio_sink_ring_buffer_acquire (GstAudioRingBuffer * buf,
GstAudioRingBufferSpec * spec);
static gboolean gst_audio_sink_ring_buffer_release (GstAudioRingBuffer * buf);
static gboolean gst_audio_sink_ring_buffer_start (GstAudioRingBuffer * buf);
static gboolean gst_audio_sink_ring_buffer_pause (GstAudioRingBuffer * buf);
static gboolean gst_audio_sink_ring_buffer_stop (GstAudioRingBuffer * buf);
static guint gst_audio_sink_ring_buffer_delay (GstAudioRingBuffer * buf);
static gboolean gst_audio_sink_ring_buffer_activate (GstAudioRingBuffer * buf,
gboolean active);
/* ringbuffer abstract base class */
static GType
gst_audio_sink_ring_buffer_get_type (void)
{
static GType ringbuffer_type = 0;
if (!ringbuffer_type) {
static const GTypeInfo ringbuffer_info = {
sizeof (GstAudioSinkRingBufferClass),
NULL,
NULL,
(GClassInitFunc) gst_audio_sink_ring_buffer_class_init,
NULL,
NULL,
sizeof (GstAudioSinkRingBuffer),
0,
(GInstanceInitFunc) gst_audio_sink_ring_buffer_init,
NULL
};
ringbuffer_type =
g_type_register_static (GST_TYPE_AUDIO_RING_BUFFER,
"GstAudioSinkRingBuffer", &ringbuffer_info, 0);
}
return ringbuffer_type;
}
static void
gst_audio_sink_ring_buffer_class_init (GstAudioSinkRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstAudioRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstringbuffer_class = (GstAudioRingBufferClass *) klass;
ring_parent_class = g_type_class_peek_parent (klass);
gobject_class->dispose = gst_audio_sink_ring_buffer_dispose;
gobject_class->finalize = gst_audio_sink_ring_buffer_finalize;
gstringbuffer_class->open_device =
GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_open_device);
gstringbuffer_class->close_device =
GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_close_device);
gstringbuffer_class->acquire =
GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_acquire);
gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_release);
gstringbuffer_class->start =
GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_start);
gstringbuffer_class->pause =
GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_pause);
gstringbuffer_class->resume =
GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_start);
gstringbuffer_class->stop =
GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_stop);
gstringbuffer_class->delay =
GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_delay);
gstringbuffer_class->activate =
GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_activate);
}
typedef gint (*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 (GstAudioRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
GstAudioSinkRingBuffer *abuf = GST_AUDIO_SINK_RING_BUFFER_CAST (buf);
WriteFunc writefunc;
GstMessage *message;
GValue val = { 0 };
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
GST_DEBUG_OBJECT (sink, "enter thread");
GST_OBJECT_LOCK (abuf);
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIO_SINK_RING_BUFFER_SIGNAL (buf);
GST_OBJECT_UNLOCK (abuf);
writefunc = csink->write;
if (writefunc == NULL)
goto no_function;
message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT_CAST (sink));
g_value_init (&val, GST_TYPE_G_THREAD);
g_value_set_boxed (&val, sink->thread);
gst_message_set_stream_status_object (message, &val);
g_value_unset (&val);
GST_DEBUG_OBJECT (sink, "posting ENTER stream status");
gst_element_post_message (GST_ELEMENT_CAST (sink), message);
while (TRUE) {
gint left, len;
guint8 *readptr;
gint readseg;
/* buffer must be started */
if (gst_audio_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
gint written;
left = len;
do {
written = writefunc (sink, readptr, left);
GST_LOG_OBJECT (sink, "transfered %d bytes of %d from segment %d",
written, left, readseg);
if (written < 0 || written > left) {
/* might not be critical, it e.g. happens when aborting playback */
GST_WARNING_OBJECT (sink,
"error writing data in %s (reason: %s), skipping segment (left: %d, written: %d)",
GST_DEBUG_FUNCPTR_NAME (writefunc),
(errno > 1 ? g_strerror (errno) : "unknown"), left, written);
break;
}
left -= written;
readptr += written;
} while (left > 0);
/* clear written samples */
gst_audio_ring_buffer_clear (buf, readseg);
/* we wrote one segment */
gst_audio_ring_buffer_advance (buf, 1);
} else {
GST_OBJECT_LOCK (abuf);
if (!abuf->running)
goto stop_running;
if (G_UNLIKELY (g_atomic_int_get (&buf->state) ==
GST_AUDIO_RING_BUFFER_STATE_STARTED)) {
GST_OBJECT_UNLOCK (abuf);
continue;
}
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIO_SINK_RING_BUFFER_SIGNAL (buf);
GST_DEBUG_OBJECT (sink, "wait for action");
GST_AUDIO_SINK_RING_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 */
g_assert_not_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");
message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT_CAST (sink));
g_value_init (&val, GST_TYPE_G_THREAD);
g_value_set_boxed (&val, sink->thread);
gst_message_set_stream_status_object (message, &val);
g_value_unset (&val);
GST_DEBUG_OBJECT (sink, "posting LEAVE stream status");
gst_element_post_message (GST_ELEMENT_CAST (sink), message);
return;
}
}
static void
gst_audio_sink_ring_buffer_init (GstAudioSinkRingBuffer * ringbuffer,
GstAudioSinkRingBufferClass * g_class)
{
ringbuffer->running = FALSE;
ringbuffer->queuedseg = 0;
g_cond_init (&ringbuffer->cond);
}
static void
gst_audio_sink_ring_buffer_dispose (GObject * object)
{
G_OBJECT_CLASS (ring_parent_class)->dispose (object);
}
static void
gst_audio_sink_ring_buffer_finalize (GObject * object)
{
GstAudioSinkRingBuffer *ringbuffer = GST_AUDIO_SINK_RING_BUFFER_CAST (object);
g_cond_clear (&ringbuffer->cond);
G_OBJECT_CLASS (ring_parent_class)->finalize (object);
}
static gboolean
gst_audio_sink_ring_buffer_open_device (GstAudioRingBuffer * 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_audio_sink_ring_buffer_close_device (GstAudioRingBuffer * 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_audio_sink_ring_buffer_acquire (GstAudioRingBuffer * buf,
GstAudioRingBufferSpec * spec)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
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->size = spec->segtotal * spec->segsize;
buf->memory = g_malloc0 (buf->size);
return TRUE;
/* ERRORS */
could_not_prepare:
{
GST_DEBUG_OBJECT (sink, "could not prepare device");
return FALSE;
}
}
static gboolean
gst_audio_sink_ring_buffer_activate (GstAudioRingBuffer * buf, gboolean active)
{
GstAudioSink *sink;
GstAudioSinkRingBuffer *abuf;
GError *error = NULL;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
abuf = GST_AUDIO_SINK_RING_BUFFER_CAST (buf);
if (active) {
abuf->running = TRUE;
GST_DEBUG_OBJECT (sink, "starting thread");
sink->thread = g_thread_try_new ("audiosink-ringbuffer",
(GThreadFunc) audioringbuffer_thread_func, buf, &error);
if (!sink->thread || error != NULL)
goto thread_failed;
GST_DEBUG_OBJECT (sink, "waiting for thread");
/* the object lock is taken */
GST_AUDIO_SINK_RING_BUFFER_WAIT (buf);
GST_DEBUG_OBJECT (sink, "thread is started");
} else {
abuf->running = FALSE;
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIO_SINK_RING_BUFFER_SIGNAL (buf);
GST_OBJECT_UNLOCK (buf);
/* join the thread */
g_thread_join (sink->thread);
GST_OBJECT_LOCK (buf);
}
return TRUE;
/* 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");
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;
}