gstreamer/sys/opensles/openslessrc.c

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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
* @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,
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GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"rate = (int) { 16000 }, "
"channels = (int) 1")
);
/* *INDENT-ON* */
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#define _do_init \
GST_DEBUG_CATEGORY_INIT (opensles_src_debug, "opensles_src", 0, \
"OpenSL ES Src");
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#define parent_class gst_opensles_src_parent_class
G_DEFINE_TYPE_WITH_CODE (GstOpenSLESSrc, gst_opensles_src,
GST_TYPE_AUDIO_BASE_SRC, _do_init);
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static GstAudioRingBuffer *
gst_opensles_src_create_ringbuffer (GstAudioBaseSrc * base)
{
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GstAudioRingBuffer *rb;
rb = gst_opensles_ringbuffer_new (RB_MODE_SRC);
return rb;
}
static void
gst_opensles_src_class_init (GstOpenSLESSrcClass * klass)
{
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GstElementClass *gstelement_class;
GstAudioBaseSrcClass *gstaudiobasesrc_class;
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gstelement_class = (GstElementClass *) klass;
gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass;
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gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_set_static_metadata (gstelement_class, "OpenSL ES Src",
"Src/Audio",
"Input sound using the OpenSL ES APIs",
"Josep Torra <support@fluendo.com>");
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gstaudiobasesrc_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_opensles_src_create_ringbuffer);
}
static void
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gst_opensles_src_init (GstOpenSLESSrc * src)
{
/* Override some default values to fit on the AudioFlinger behaviour of
* processing 20ms buffers as minimum buffer size. */
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GST_AUDIO_BASE_SRC (src)->buffer_time = 400000;
GST_AUDIO_BASE_SRC (src)->latency_time = 20000;
}