/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2005 Wim Taymans * * 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. */ #include #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 ("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 ("transfered %d bytes of %d from segment %d", written, left, readseg); if (written < 0 || written > left) { GST_WARNING ("error writing data (reason: %s), skipping segment\n", 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 ("signal wait"); GST_AUDIORING_BUFFER_SIGNAL (buf); GST_DEBUG ("wait for action"); GST_AUDIORING_BUFFER_WAIT (buf); GST_DEBUG ("got signal"); if (!abuf->running) goto stop_running; GST_DEBUG ("continue running"); GST_OBJECT_UNLOCK (abuf); } } GST_DEBUG ("exit thread"); return; /* ERROR */ no_function: { GST_DEBUG ("no write function, exit thread"); return; } stop_running: { GST_OBJECT_UNLOCK (abuf); GST_DEBUG ("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 ("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 ("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; /* allocate one more segment as we need some headroom */ spec->segtotal++; 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 ("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 ("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; return result; could_not_unprepare: { GST_DEBUG ("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 ("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 ("reset..."); csink->reset (sink); GST_DEBUG ("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 ("reset..."); csink->reset (sink); GST_DEBUG ("reset done"); } if (abuf->running) { GST_DEBUG ("stop, waiting..."); GST_AUDIORING_BUFFER_WAIT (buf); GST_DEBUG ("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 ("creating ringbuffer"); buffer = g_object_new (GST_TYPE_AUDIORING_BUFFER, NULL); GST_DEBUG ("created ringbuffer @%p", buffer); return buffer; }