diff options
Diffstat (limited to 'gst-libs/gst/audio/gstbaseaudiosink.c')
| -rw-r--r-- | gst-libs/gst/audio/gstbaseaudiosink.c | 235 |
1 files changed, 235 insertions, 0 deletions
diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c new file mode 100644 index 000000000..d27b4a222 --- /dev/null +++ b/gst-libs/gst/audio/gstbaseaudiosink.c @@ -0,0 +1,235 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> + * 2005 Wim Taymans <wim@fluendo.com> + * + * gstbaseaudiosink.c: + * + * 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 "gstbaseaudiosink.h" + +GST_DEBUG_CATEGORY_STATIC (gst_baseaudiosink_debug); +#define GST_CAT_DEFAULT gst_baseaudiosink_debug + +/* BaseAudioSink signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + ARG_0, +}; + +#define _do_init(bla) \ + GST_DEBUG_CATEGORY_INIT (gst_baseaudiosink_debug, "baseaudiosink", 0, "baseaudiosink element"); + +GST_BOILERPLATE_FULL (GstBaseAudioSink, gst_baseaudiosink, GstBaseSink, + GST_TYPE_BASESINK, _do_init); + +static void gst_baseaudiosink_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_baseaudiosink_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); + +static GstElementStateReturn gst_baseaudiosink_change_state (GstElement * + element); + +static GstFlowReturn gst_baseaudiosink_preroll (GstBaseSink * bsink, + GstBuffer * buffer); +static GstFlowReturn gst_baseaudiosink_render (GstBaseSink * bsink, + GstBuffer * buffer); +static void gst_baseaudiosink_event (GstBaseSink * bsink, GstEvent * event); +static void gst_baseaudiosink_get_times (GstBaseSink * bsink, + GstBuffer * buffer, GstClockTime * start, GstClockTime * end); +static gboolean gst_baseaudiosink_setcaps (GstBaseSink * bsink, GstCaps * caps); + +//static guint gst_baseaudiosink_signals[LAST_SIGNAL] = { 0 }; + +static void +gst_baseaudiosink_base_init (gpointer g_class) +{ +} + +static void +gst_baseaudiosink_class_init (GstBaseAudioSinkClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + GstBaseSinkClass *gstbasesink_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + gstbasesink_class = (GstBaseSinkClass *) klass; + + gobject_class->set_property = + GST_DEBUG_FUNCPTR (gst_baseaudiosink_set_property); + gobject_class->get_property = + GST_DEBUG_FUNCPTR (gst_baseaudiosink_get_property); + + gstelement_class->change_state = + GST_DEBUG_FUNCPTR (gst_baseaudiosink_change_state); + + gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_baseaudiosink_event); + gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_baseaudiosink_preroll); + gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_baseaudiosink_render); + gstbasesink_class->get_times = + GST_DEBUG_FUNCPTR (gst_baseaudiosink_get_times); + gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_baseaudiosink_setcaps); +} + +static void +gst_baseaudiosink_init (GstBaseAudioSink * baseaudiosink) +{ +} + +static void +gst_baseaudiosink_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstBaseAudioSink *sink; + + sink = GST_BASEAUDIOSINK (object); + + switch (prop_id) { + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_baseaudiosink_get_property (GObject * object, guint prop_id, GValue * value, + GParamSpec * pspec) +{ + GstBaseAudioSink *sink; + + sink = GST_BASEAUDIOSINK (object); + + switch (prop_id) { + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static gboolean +gst_baseaudiosink_setcaps (GstBaseSink * bsink, GstCaps * caps) +{ + GstBaseAudioSink *sink = GST_BASEAUDIOSINK (bsink); + GstRingBufferSpec spec; + + spec.caps = caps; + spec.segsize = 64; + spec.segtotal = 64; + + gst_ringbuffer_release (sink->ringbuffer); + gst_ringbuffer_acquire (sink->ringbuffer, &spec); + + return TRUE; +} + +static void +gst_baseaudiosink_get_times (GstBaseSink * bsink, GstBuffer * buffer, + GstClockTime * start, GstClockTime * end) +{ + *start = GST_CLOCK_TIME_NONE; + *end = GST_CLOCK_TIME_NONE; +} + +static void +gst_baseaudiosink_event (GstBaseSink * bsink, GstEvent * event) +{ +} + +static GstFlowReturn +gst_baseaudiosink_preroll (GstBaseSink * bsink, GstBuffer * buffer) +{ + return GST_FLOW_OK; +} + +static GstFlowReturn +gst_baseaudiosink_render (GstBaseSink * bsink, GstBuffer * buf) +{ + GstBaseAudioSink *sink = GST_BASEAUDIOSINK (bsink); + + gst_ringbuffer_write (sink->ringbuffer, GST_CLOCK_TIME_NONE, + GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); + + return GST_FLOW_OK; +} + +GstRingBuffer * +gst_baseaudiosink_create_ringbuffer (GstBaseAudioSink * sink) +{ + GstBaseAudioSinkClass *bclass; + GstRingBuffer *buffer = NULL; + + bclass = GST_BASEAUDIOSINK_GET_CLASS (sink); + if (bclass->create_ringbuffer) + buffer = bclass->create_ringbuffer (sink); + + return buffer; +} + +void +gst_baseaudiosink_callback (GstRingBuffer * rbuf, guint advance, gpointer data) +{ + //GstBaseAudioSink *sink = GST_BASEAUDIOSINK (data); +} + +static GstElementStateReturn +gst_baseaudiosink_change_state (GstElement * element) +{ + GstElementStateReturn ret = GST_STATE_SUCCESS; + GstBaseAudioSink *sink = GST_BASEAUDIOSINK (element); + GstElementState transition = GST_STATE_TRANSITION (element); + + switch (transition) { + case GST_STATE_NULL_TO_READY: + break; + case GST_STATE_READY_TO_PAUSED: + sink->ringbuffer = gst_baseaudiosink_create_ringbuffer (sink); + gst_ringbuffer_set_callback (sink->ringbuffer, gst_baseaudiosink_callback, + sink); + break; + case GST_STATE_PAUSED_TO_PLAYING: + gst_ringbuffer_play (sink->ringbuffer); + break; + default: + break; + } + + ret = GST_ELEMENT_CLASS (parent_class)->change_state (element); + + switch (transition) { + case GST_STATE_PLAYING_TO_PAUSED: + gst_ringbuffer_stop (sink->ringbuffer); + break; + case GST_STATE_PAUSED_TO_READY: + gst_object_unref (GST_OBJECT (sink->ringbuffer)); + break; + case GST_STATE_READY_TO_NULL: + break; + default: + break; + } + + return ret; +} |