--- /dev/null
+/* 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);
+static gboolean gst_audioringbuffer_activate (GstRingBuffer * buf,
+ gboolean active);
+
+/* 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;
+ GstRingBufferClass *gstringbuffer_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstringbuffer_class = (GstRingBufferClass *) klass;
+
+ ring_parent_class = g_type_class_peek_parent (klass);
+
+ gobject_class->dispose = gst_audioringbuffer_dispose;
+ gobject_class->finalize = 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);
+ gstringbuffer_class->activate =
+ GST_DEBUG_FUNCPTR (gst_audioringbuffer_activate);
+}
+
+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;
+ 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_AUDIORING_BUFFER_SIGNAL (buf);
+ GST_OBJECT_UNLOCK (abuf);
+
+ writefunc = csink->write;
+ if (writefunc == NULL)
+ goto no_function;
+
+ g_value_init (&val, G_TYPE_POINTER);
+ g_value_set_pointer (&val, sink->thread);
+ message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
+ GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT_CAST (sink));
+ gst_message_set_stream_status_object (message, &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_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_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 */
+ 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));
+ gst_message_set_stream_status_object (message, &val);
+ GST_DEBUG_OBJECT (sink, "posting LEAVE stream status");
+ gst_element_post_message (GST_ELEMENT_CAST (sink), message);
+ 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;
+ 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));
+
+ return TRUE;
+
+ /* ERRORS */
+could_not_prepare:
+ {
+ GST_DEBUG_OBJECT (sink, "could not prepare device");
+ return FALSE;
+ }
+}
+
+static gboolean
+gst_audioringbuffer_activate (GstRingBuffer * buf, gboolean active)
+{
+ GstAudioSink *sink;
+ GstAudioRingBuffer *abuf;
+ GError *error = NULL;
+
+ sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+ abuf = GST_AUDIORING_BUFFER_CAST (buf);
+
+ if (active) {
+ abuf->running = TRUE;
+
+ GST_DEBUG_OBJECT (sink, "starting thread");
+ sink->thread =
+ g_thread_create ((GThreadFunc) audioringbuffer_thread_func, buf, TRUE,
+ &error);
+ if (!sink->thread || error != NULL)
+ goto thread_failed;
+
+ GST_DEBUG_OBJECT (sink, "waiting for thread");
+ /* the object lock is taken */
+ GST_AUDIORING_BUFFER_WAIT (buf);
+ GST_DEBUG_OBJECT (sink, "thread is started");
+ } else {
+ 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);
+ }
+ 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_audioringbuffer_release (GstRingBuffer * 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 */
+ 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;
+
+ 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_AUDIORING_BUFFER_WAIT (buf);
+ GST_DEBUG_OBJECT (sink, "stopped");
+ }
+#endif
+
+ 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)
+{
+ GstBaseAudioSinkClass *gstbaseaudiosink_class;
+
+ gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
+
+ gstbaseaudiosink_class->create_ringbuffer =
+ GST_DEBUG_FUNCPTR (gst_audio_sink_create_ringbuffer);
+
+ g_type_class_ref (GST_TYPE_AUDIORING_BUFFER);
+}
+
+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;
+}
projects / mafwsubrenderer / blobdiff