gstreamer/sys/opensles/openslessrc.c

111 lines
3.2 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-openslessrc
* @see_also: openslessink
*
* This element reads data from default audio input using the OpenSL ES API in Android OS.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v openslessrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=recorded.ogg
* ]| Record from default audio input and encode to Ogg/Vorbis.
* </refsect2>
*
*/
#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-int, "
"endianness = (int) {" G_STRINGIFY (G_BYTE_ORDER) " }, "
"signed = (boolean) { TRUE }, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) 16000, "
"channels = (int) 1")
);
/* *INDENT-ON* */
static void
_do_init (GType type)
{
GST_DEBUG_CATEGORY_INIT (opensles_src_debug, "opensles_src", 0,
"OpenSL ES Src");
}
GST_BOILERPLATE_FULL (GstOpenSLESSrc, gst_opensles_src, GstBaseAudioSrc,
GST_TYPE_BASE_AUDIO_SRC, _do_init);
static void
gst_opensles_src_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_static_pad_template (element_class, &src_factory);
gst_element_class_set_details_simple (element_class, "OpenSL ES Src",
"Src/Audio",
"Input sound using the OpenSL ES APIs",
"Josep Torra <support@fluendo.com>");
}
static GstRingBuffer *
gst_opensles_src_create_ringbuffer (GstBaseAudioSrc * base)
{
GstRingBuffer *rb;
rb = gst_opensles_ringbuffer_new (RB_MODE_SRC);
return rb;
}
static void
gst_opensles_src_class_init (GstOpenSLESSrcClass * klass)
{
GstBaseAudioSrcClass *gstbaseaudiosrc_class;
gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gstbaseaudiosrc_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_opensles_src_create_ringbuffer);
}
static void
gst_opensles_src_init (GstOpenSLESSrc * src, GstOpenSLESSrcClass * gclass)
{
/* Override some default values to fit on the AudioFlinger behaviour of
* processing 20ms buffers as minimum buffer size. */
GST_BASE_AUDIO_SRC (src)->buffer_time = 400000;
GST_BASE_AUDIO_SRC (src)->latency_time = 20000;
}