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Original commit message from CVS: * ext/alsa/gstalsasink.c: (gst_alsasink_reset): * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_stop): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_event), (gst_base_audio_sink_render), (gst_base_audio_sink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_stop), (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all), (gst_ring_buffer_commit), (gst_ring_buffer_read): * gst-libs/gst/audio/gstringbuffer.h: Fix sync again. Moved sample alignment to basesink.
476 lines
12 KiB
C
476 lines
12 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <string.h>
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#include "gstaudiosink.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_AUDIORING_BUFFER \
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(gst_audioringbuffer_get_type())
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#define GST_AUDIORING_BUFFER(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORING_BUFFER,GstAudioRingBuffer))
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#define GST_AUDIORING_BUFFER_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORING_BUFFER,GstAudioRingBufferClass))
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#define GST_AUDIORING_BUFFER_GET_CLASS(obj) \
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(G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIORING_BUFFER, GstAudioRingBufferClass))
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#define GST_IS_AUDIORING_BUFFER(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORING_BUFFER))
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#define GST_IS_AUDIORING_BUFFER_CLASS(obj)\
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORING_BUFFER))
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typedef struct _GstAudioRingBuffer GstAudioRingBuffer;
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typedef struct _GstAudioRingBufferClass GstAudioRingBufferClass;
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#define GST_AUDIORING_BUFFER_GET_COND(buf) (((GstAudioRingBuffer *)buf)->cond)
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#define GST_AUDIORING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIORING_BUFFER_GET_COND (buf), GST_GET_LOCK (buf)))
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#define GST_AUDIORING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIORING_BUFFER_GET_COND (buf)))
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#define GST_AUDIORING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIORING_BUFFER_GET_COND (buf)))
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struct _GstAudioRingBuffer
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{
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GstRingBuffer 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 _GstAudioRingBufferClass
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{
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GstRingBufferClass parent_class;
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};
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static void gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass);
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static void gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer,
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GstAudioRingBufferClass * klass);
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static void gst_audioringbuffer_dispose (GObject * object);
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static void gst_audioringbuffer_finalize (GObject * object);
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static GstRingBufferClass *ring_parent_class = NULL;
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static gboolean gst_audioringbuffer_open_device (GstRingBuffer * buf);
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static gboolean gst_audioringbuffer_close_device (GstRingBuffer * buf);
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static gboolean gst_audioringbuffer_acquire (GstRingBuffer * buf,
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GstRingBufferSpec * spec);
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static gboolean gst_audioringbuffer_release (GstRingBuffer * buf);
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static gboolean gst_audioringbuffer_start (GstRingBuffer * buf);
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static gboolean gst_audioringbuffer_stop (GstRingBuffer * buf);
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static guint gst_audioringbuffer_delay (GstRingBuffer * buf);
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/* ringbuffer abstract base class */
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static GType
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gst_audioringbuffer_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 (GstAudioRingBufferClass),
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NULL,
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NULL,
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(GClassInitFunc) gst_audioringbuffer_class_init,
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NULL,
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NULL,
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sizeof (GstAudioRingBuffer),
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0,
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(GInstanceInitFunc) gst_audioringbuffer_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_RING_BUFFER, "GstAudioSinkRingBuffer",
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&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_audioringbuffer_class_init (GstAudioRingBufferClass * klass)
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{
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GObjectClass *gobject_class;
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GstObjectClass *gstobject_class;
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GstRingBufferClass *gstringbuffer_class;
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gobject_class = (GObjectClass *) klass;
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gstobject_class = (GstObjectClass *) klass;
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gstringbuffer_class = (GstRingBufferClass *) klass;
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ring_parent_class = g_type_class_ref (GST_TYPE_RING_BUFFER);
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gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_audioringbuffer_dispose);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audioringbuffer_finalize);
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gstringbuffer_class->open_device =
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GST_DEBUG_FUNCPTR (gst_audioringbuffer_open_device);
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gstringbuffer_class->close_device =
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GST_DEBUG_FUNCPTR (gst_audioringbuffer_close_device);
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gstringbuffer_class->acquire =
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GST_DEBUG_FUNCPTR (gst_audioringbuffer_acquire);
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gstringbuffer_class->release =
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GST_DEBUG_FUNCPTR (gst_audioringbuffer_release);
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gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start);
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gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_audioringbuffer_stop);
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gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start);
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gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_audioringbuffer_stop);
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gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_audioringbuffer_delay);
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}
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typedef guint (*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 (GstRingBuffer * buf)
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{
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GstAudioSink *sink;
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GstAudioSinkClass *csink;
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GstAudioRingBuffer *abuf = GST_AUDIORING_BUFFER (buf);
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WriteFunc writefunc;
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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 ("enter thread");
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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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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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if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
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gint written = 0;
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left = len;
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do {
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written = writefunc (sink, readptr + written, left);
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GST_DEBUG ("transfered %d bytes of %d from segment %d", written, left,
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readseg);
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if (written < 0 || written > left) {
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GST_WARNING ("error writing data (reason: %s), skipping segment\n",
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strerror (errno));
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break;
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}
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left -= written;
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} while (left > 0);
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/* clear written samples */
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gst_ring_buffer_clear (buf, readseg);
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/* we wrote one segment */
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gst_ring_buffer_advance (buf, 1);
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} else {
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GST_LOCK (abuf);
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if (!abuf->running)
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goto stop_running;
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GST_DEBUG ("signal wait");
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GST_AUDIORING_BUFFER_SIGNAL (buf);
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GST_DEBUG ("wait for action");
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GST_AUDIORING_BUFFER_WAIT (buf);
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GST_DEBUG ("got signal");
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if (!abuf->running)
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goto stop_running;
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GST_DEBUG ("continue running");
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GST_UNLOCK (abuf);
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}
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}
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GST_DEBUG ("exit thread");
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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 ("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_UNLOCK (abuf);
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GST_DEBUG ("stop running, exit thread");
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return;
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}
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}
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static void
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gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer,
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GstAudioRingBufferClass * g_class)
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{
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ringbuffer->running = FALSE;
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ringbuffer->queuedseg = 0;
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ringbuffer->cond = g_cond_new ();
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}
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static void
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gst_audioringbuffer_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_audioringbuffer_finalize (GObject * object)
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{
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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_audioringbuffer_open_device (GstRingBuffer * 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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return FALSE;
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}
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}
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static gboolean
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gst_audioringbuffer_close_device (GstRingBuffer * 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_open;
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return result;
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could_not_open:
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{
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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_audioringbuffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
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{
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GstAudioSink *sink;
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GstAudioSinkClass *csink;
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GstAudioRingBuffer *abuf;
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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_open;
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/* allocate one more segment as we need some headroom */
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spec->segtotal++;
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buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
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memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
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abuf = GST_AUDIORING_BUFFER (buf);
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abuf->running = TRUE;
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sink->thread =
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g_thread_create ((GThreadFunc) audioringbuffer_thread_func, buf, TRUE,
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NULL);
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GST_AUDIORING_BUFFER_WAIT (buf);
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return result;
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could_not_open:
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{
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return FALSE;
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}
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}
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/* function is called with LOCK */
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static gboolean
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gst_audioringbuffer_release (GstRingBuffer * buf)
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{
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GstAudioSink *sink;
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GstAudioSinkClass *csink;
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GstAudioRingBuffer *abuf;
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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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abuf = GST_AUDIORING_BUFFER (buf);
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abuf->running = FALSE;
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GST_AUDIORING_BUFFER_SIGNAL (buf);
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GST_UNLOCK (buf);
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/* join the thread */
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g_thread_join (sink->thread);
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GST_LOCK (buf);
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/* free the buffer */
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gst_buffer_unref (buf->data);
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buf->data = NULL;
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if (csink->unprepare)
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result = csink->unprepare (sink);
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return result;
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}
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static gboolean
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gst_audioringbuffer_start (GstRingBuffer * buf)
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{
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GstAudioSink *sink;
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sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
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GST_DEBUG ("start, sending signal");
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GST_AUDIORING_BUFFER_SIGNAL (buf);
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return TRUE;
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}
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static gboolean
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gst_audioringbuffer_stop (GstRingBuffer * buf)
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{
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GstAudioSink *sink;
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GstAudioSinkClass *csink;
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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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/* unblock any pending writes to the audio device */
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if (csink->reset) {
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GST_DEBUG ("reset...");
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csink->reset (sink);
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GST_DEBUG ("reset done");
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}
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GST_DEBUG ("stop, waiting...");
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GST_AUDIORING_BUFFER_WAIT (buf);
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GST_DEBUG ("stopped");
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return TRUE;
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}
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static guint
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gst_audioringbuffer_delay (GstRingBuffer * buf)
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{
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GstAudioSink *sink;
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GstAudioSinkClass *csink;
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guint res = 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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if (csink->delay)
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res = csink->delay (sink);
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return res;
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}
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/* AudioSink signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0,
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};
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#define _do_init(bla) \
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GST_DEBUG_CATEGORY_INIT (gst_audio_sink_debug, "audiosink", 0, "audiosink element");
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GST_BOILERPLATE_FULL (GstAudioSink, gst_audio_sink, GstBaseAudioSink,
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GST_TYPE_BASE_AUDIO_SINK, _do_init);
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static GstRingBuffer *gst_audio_sink_create_ringbuffer (GstBaseAudioSink *
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sink);
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static void
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gst_audio_sink_base_init (gpointer g_class)
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{
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}
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static void
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gst_audio_sink_class_init (GstAudioSinkClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseSinkClass *gstbasesink_class;
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GstBaseAudioSinkClass *gstbaseaudiosink_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasesink_class = (GstBaseSinkClass *) klass;
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gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
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gstbaseaudiosink_class->create_ringbuffer =
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GST_DEBUG_FUNCPTR (gst_audio_sink_create_ringbuffer);
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}
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static void
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gst_audio_sink_init (GstAudioSink * audiosink, GstAudioSinkClass * g_class)
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{
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}
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static GstRingBuffer *
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gst_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
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{
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GstRingBuffer *buffer;
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GST_DEBUG ("creating ringbuffer");
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buffer = g_object_new (GST_TYPE_AUDIORING_BUFFER, NULL);
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GST_DEBUG ("created ringbuffer @%p", buffer);
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return buffer;
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}
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