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/* GStreamer
* Copyright (C) 2012 Fluendo S.A. <support@fluendo.com>
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-openslessrc
* @title: openslessrc
* @see_also: openslessink
*
* This element reads data from default audio input using the OpenSL ES API in Android OS.
*
* ## Example pipelines
* |[
* gst-launch-1.0 -v openslessrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=recorded.ogg
* ]| Record from default audio input and encode to Ogg/Vorbis.
*
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include "openslessrc.h"
GST_DEBUG_CATEGORY_STATIC (opensles_src_debug);
#define GST_CAT_DEFAULT opensles_src_debug
/* *INDENT-OFF* */
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"rate = (int) 16000, "
"channels = (int) 1, "
"layout = (string) interleaved")
);
/* *INDENT-ON* */
#define _do_init \
GST_DEBUG_CATEGORY_INIT (opensles_src_debug, "openslessrc", 0, \
"OpenSLES Source");
#define parent_class gst_opensles_src_parent_class
G_DEFINE_TYPE_WITH_CODE (GstOpenSLESSrc, gst_opensles_src,
GST_TYPE_AUDIO_BASE_SRC, _do_init);
enum
{
PROP_0,
PROP_PRESET,
};
#define DEFAULT_PRESET GST_OPENSLES_RECORDING_PRESET_NONE
static void
gst_opensles_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOpenSLESSrc *src = GST_OPENSLES_SRC (object);
switch (prop_id) {
case PROP_PRESET:
src->preset = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_opensles_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstOpenSLESSrc *src = GST_OPENSLES_SRC (object);
switch (prop_id) {
case PROP_PRESET:
g_value_set_enum (value, src->preset);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstAudioRingBuffer *
gst_opensles_src_create_ringbuffer (GstAudioBaseSrc * base)
{
GstAudioRingBuffer *rb;
rb = gst_opensles_ringbuffer_new (RB_MODE_SRC);
GST_OPENSLES_RING_BUFFER (rb)->preset = GST_OPENSLES_SRC (base)->preset;
return rb;
}
static void
gst_opensles_src_class_init (GstOpenSLESSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstAudioBaseSrcClass *gstaudiobasesrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass;
gobject_class->set_property = gst_opensles_src_set_property;
gobject_class->get_property = gst_opensles_src_get_property;
g_object_class_install_property (gobject_class, PROP_PRESET,
g_param_spec_enum ("preset", "Preset", "Recording preset to use",
GST_TYPE_OPENSLES_RECORDING_PRESET, DEFAULT_PRESET,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
gst_element_class_set_static_metadata (gstelement_class, "OpenSL ES Src",
"Source/Audio",
"Input sound using the OpenSL ES APIs",
"Josep Torra <support@fluendo.com>");
gstaudiobasesrc_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_opensles_src_create_ringbuffer);
}
static void
gst_opensles_src_init (GstOpenSLESSrc * src)
{
/* Override some default values to fit on the AudioFlinger behaviour of
* processing 20ms buffers as minimum buffer size. */
GST_AUDIO_BASE_SRC (src)->buffer_time = 200000;
GST_AUDIO_BASE_SRC (src)->latency_time = 20000;
src->preset = DEFAULT_PRESET;
}
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