mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-24 18:51:11 +00:00
be7d42b548
Original commit message from CVS: Add semicolons after GST_BOILERPLATE[_FULL] so that indent doesn't mess up following lines.
345 lines
11 KiB
C
345 lines
11 KiB
C
/*
|
|
* GStreamer
|
|
* Copyright 2005,2006 Zaheer Abbas Merali <zaheerabbas at merali dot org>
|
|
*
|
|
* Permission is hereby granted, free of charge, to any person obtaining a
|
|
* copy of this software and associated documentation files (the "Software"),
|
|
* to deal in the Software without restriction, including without limitation
|
|
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
|
|
* and/or sell copies of the Software, and to permit persons to whom the
|
|
* Software is furnished to do so, subject to the following conditions:
|
|
*
|
|
* The above copyright notice and this permission notice shall be included in
|
|
* all copies or substantial portions of the Software.
|
|
*
|
|
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
|
|
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
|
|
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
|
|
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
|
|
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
|
|
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
|
|
* DEALINGS IN THE SOFTWARE.
|
|
*
|
|
* Alternatively, the contents of this file may be used under the
|
|
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
|
|
* which case the following provisions apply instead of the ones
|
|
* mentioned above:
|
|
*
|
|
* 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:element-plugin
|
|
*
|
|
* <refsect2>
|
|
* <title>Example launch line</title>
|
|
* <para>
|
|
* <programlisting>
|
|
* gst-launch -v -m audiotestsrc ! audioconvert ! osxaudiosink
|
|
* </programlisting>
|
|
* </para>
|
|
* </refsect2>
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
# include <config.h>
|
|
#endif
|
|
|
|
#include <gst/gst.h>
|
|
#include <CoreAudio/CoreAudio.h>
|
|
#include "gstosxaudiosink.h"
|
|
#include "gstosxaudiosrc.h"
|
|
|
|
#include "gstosxaudioelement.h"
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (osx_audiosink_debug);
|
|
#define GST_CAT_DEFAULT osx_audiosink_debug
|
|
|
|
static GstElementDetails gst_osx_audio_sink_details =
|
|
GST_ELEMENT_DETAILS ("Audio Sink (OSX)",
|
|
"Sink/Audio",
|
|
"Output to a sound card in OS X",
|
|
"Zaheer Abbas Merali <zaheerabbas at merali dot org>");
|
|
|
|
/* Filter signals and args */
|
|
enum
|
|
{
|
|
/* FILL ME */
|
|
LAST_SIGNAL
|
|
};
|
|
|
|
enum
|
|
{
|
|
ARG_0,
|
|
ARG_DEVICE
|
|
};
|
|
|
|
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-raw-float, "
|
|
"endianness = (int) {" G_STRINGIFY (G_BYTE_ORDER) " }, "
|
|
"signed = (boolean) { TRUE }, "
|
|
"width = (int) 32, "
|
|
"depth = (int) 32, " "rate = (int) 44100, " "channels = (int) 2")
|
|
);
|
|
|
|
static void gst_osx_audio_sink_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec);
|
|
static void gst_osx_audio_sink_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec);
|
|
static GstCaps *gst_osx_audio_sink_getcaps (GstBaseSink * sink);
|
|
|
|
|
|
static GstRingBuffer *gst_osx_audio_sink_create_ringbuffer (GstBaseAudioSink *
|
|
sink);
|
|
/*static GstCaps* gst_osx_audio_sink_getcaps (GstBaseSink * bsink);*/
|
|
static void gst_osx_audio_sink_osxelement_init (gpointer g_iface,
|
|
gpointer iface_data);
|
|
OSStatus gst_osx_audio_sink_io_proc (AudioDeviceID inDevice,
|
|
const AudioTimeStamp * inNow, const AudioBufferList * inInputData,
|
|
const AudioTimeStamp * inInputTime, AudioBufferList * outOutputData,
|
|
const AudioTimeStamp * inOutputTime, void *inClientData);
|
|
static void
|
|
gst_osx_audio_sink_osxelement_do_init (GType type)
|
|
{
|
|
static const GInterfaceInfo osxelement_info = {
|
|
gst_osx_audio_sink_osxelement_init,
|
|
NULL,
|
|
NULL
|
|
};
|
|
|
|
GST_DEBUG_CATEGORY_INIT (osx_audiosink_debug, "osxaudiosink", 0,
|
|
"OSX Audio Sink");
|
|
GST_DEBUG ("Adding static interface\n");
|
|
g_type_add_interface_static (type, GST_OSX_AUDIO_ELEMENT_TYPE,
|
|
&osxelement_info);
|
|
}
|
|
|
|
GST_BOILERPLATE_FULL (GstOsxAudioSink, gst_osx_audio_sink, GstBaseAudioSink,
|
|
GST_TYPE_BASE_AUDIO_SINK, gst_osx_audio_sink_osxelement_do_init);
|
|
|
|
|
|
static void
|
|
gst_osx_audio_sink_base_init (gpointer g_class)
|
|
{
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
|
|
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&sink_factory));
|
|
|
|
gst_element_class_set_details (element_class, &gst_osx_audio_sink_details);
|
|
}
|
|
|
|
/* initialize the plugin's class */
|
|
static void
|
|
gst_osx_audio_sink_class_init (GstOsxAudioSinkClass * 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;
|
|
|
|
parent_class = g_type_class_peek_parent (klass);
|
|
|
|
gobject_class->set_property =
|
|
GST_DEBUG_FUNCPTR (gst_osx_audio_sink_set_property);
|
|
gobject_class->get_property =
|
|
GST_DEBUG_FUNCPTR (gst_osx_audio_sink_get_property);
|
|
|
|
g_object_class_install_property (gobject_class, ARG_DEVICE,
|
|
g_param_spec_int ("device", "Device ID", "Device ID of output device",
|
|
0, G_MAXINT, 0, G_PARAM_READWRITE));
|
|
|
|
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_osx_audio_sink_getcaps);
|
|
gstbaseaudiosink_class->create_ringbuffer =
|
|
GST_DEBUG_FUNCPTR (gst_osx_audio_sink_create_ringbuffer);
|
|
|
|
}
|
|
|
|
/* initialize the new element
|
|
* instantiate pads and add them to element
|
|
* set functions
|
|
* initialize structure
|
|
*/
|
|
static void
|
|
gst_osx_audio_sink_init (GstOsxAudioSink * sink, GstOsxAudioSinkClass * gclass)
|
|
{
|
|
/* GstElementClass *klass = GST_ELEMENT_GET_CLASS (sink); */
|
|
sink->ringbuffer = NULL;
|
|
GST_DEBUG ("Initialising object\n");
|
|
gst_osx_audio_sink_create_ringbuffer (sink);
|
|
|
|
}
|
|
|
|
static void
|
|
gst_osx_audio_sink_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_DEVICE:
|
|
if (sink->ringbuffer)
|
|
sink->ringbuffer->device_id = g_value_get_int (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_osx_audio_sink_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (object);
|
|
int val = 0;
|
|
|
|
switch (prop_id) {
|
|
case ARG_DEVICE:
|
|
if (sink->ringbuffer)
|
|
val = sink->ringbuffer->device_id;
|
|
|
|
g_value_set_int (value, val);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* GstElement vmethod implementations */
|
|
|
|
/* GstBaseSink vmethod implementations */
|
|
static GstCaps *
|
|
gst_osx_audio_sink_getcaps (GstBaseSink * sink)
|
|
{
|
|
GstCaps *caps;
|
|
GstOsxAudioSink *osxsink;
|
|
OSStatus status;
|
|
AudioValueRange rates[10];
|
|
UInt32 propertySize;
|
|
int i;
|
|
|
|
propertySize = sizeof (AudioValueRange) * 9;
|
|
osxsink = GST_OSX_AUDIO_SINK (sink);
|
|
|
|
caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
|
|
(sink)));
|
|
|
|
|
|
status = AudioDeviceGetProperty (osxsink->ringbuffer->device_id, 0, FALSE,
|
|
kAudioDevicePropertyAvailableNominalSampleRates, &propertySize, &rates);
|
|
|
|
GST_DEBUG
|
|
("Getting available sample rates: Status: %d number of ranges: %d\n",
|
|
status, propertySize / sizeof (AudioValueRange));
|
|
|
|
for (i = 0; i < propertySize / sizeof (AudioValueRange); i++) {
|
|
g_print ("Range from %f to %f\n", rates[i].mMinimum, rates[i].mMaximum);
|
|
}
|
|
|
|
return caps;
|
|
}
|
|
|
|
/* GstBaseAudioSink vmethod implementations */
|
|
static GstRingBuffer *
|
|
gst_osx_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
|
|
{
|
|
GstOsxAudioSink *osxsink;
|
|
|
|
osxsink = GST_OSX_AUDIO_SINK (sink);
|
|
if (!osxsink->ringbuffer) {
|
|
GST_DEBUG ("Creating ringbuffer\n");
|
|
osxsink->ringbuffer = g_object_new (GST_TYPE_OSX_RING_BUFFER, NULL);
|
|
GST_DEBUG ("osx sink 0x%x element 0x%x ioproc 0x%x\n", osxsink,
|
|
GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsink),
|
|
(void *) gst_osx_audio_sink_io_proc);
|
|
osxsink->ringbuffer->element =
|
|
GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsink);
|
|
}
|
|
|
|
return GST_RING_BUFFER (osxsink->ringbuffer);
|
|
}
|
|
|
|
OSStatus
|
|
gst_osx_audio_sink_io_proc (AudioDeviceID inDevice,
|
|
const AudioTimeStamp * inNow, const AudioBufferList * inInputData,
|
|
const AudioTimeStamp * inInputTime, AudioBufferList * outOutputData,
|
|
const AudioTimeStamp * inOutputTime, void *inClientData)
|
|
{
|
|
GstOsxRingBuffer *buf = GST_OSX_RING_BUFFER (inClientData);
|
|
|
|
guint8 *readptr;
|
|
gint readseg;
|
|
gint len;
|
|
|
|
if (gst_ring_buffer_prepare_read (GST_RING_BUFFER (buf), &readseg, &readptr,
|
|
&len)) {
|
|
outOutputData->mBuffers[0].mDataByteSize = len;
|
|
memcpy ((char *) outOutputData->mBuffers[0].mData, readptr, len);
|
|
|
|
/* clear written samples */
|
|
gst_ring_buffer_clear (GST_RING_BUFFER (buf), readseg);
|
|
|
|
/* we wrote one segment */
|
|
gst_ring_buffer_advance (GST_RING_BUFFER (buf), 1);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static void
|
|
gst_osx_audio_sink_osxelement_init (gpointer g_iface, gpointer iface_data)
|
|
{
|
|
GstOsxAudioElementInterface *iface = (GstOsxAudioElementInterface *) g_iface;
|
|
|
|
iface->io_proc = gst_osx_audio_sink_io_proc;
|
|
}
|
|
|
|
/* entry point to initialize the plug-in
|
|
* initialize the plug-in itself
|
|
* register the element factories and pad templates
|
|
* register the features
|
|
*
|
|
* exchange the string 'plugin' with your elemnt name
|
|
*/
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
gboolean ret;
|
|
|
|
ret = gst_element_register (plugin, "osxaudiosink",
|
|
GST_RANK_NONE, GST_TYPE_OSX_AUDIO_SINK);
|
|
return ret && gst_element_register (plugin, "osxaudiosrc",
|
|
GST_RANK_NONE, GST_TYPE_OSX_AUDIO_SRC);
|
|
}
|
|
|
|
/* this is the structure that gstreamer looks for to register plugins
|
|
*
|
|
* exchange the strings 'plugin' and 'Template plugin' with you plugin name and
|
|
* description
|
|
*/
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
"osxaudio",
|
|
"OSX Audio plugin",
|
|
plugin_init, VERSION, "LGPL", "GStreamer", "http://gstreamer.net/")
|