mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 05:38:58 +00:00
153 lines
4.4 KiB
C
153 lines
4.4 KiB
C
/* 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;
|
|
}
|