mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-30 13:41:48 +00:00
100 lines
3 KiB
C
100 lines
3 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
|
|
* @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, "
|
|
"format = (string) " GST_AUDIO_NE (S16) ", "
|
|
"rate = (int) { 16000 }, "
|
|
"channels = (int) 1")
|
|
);
|
|
/* *INDENT-ON* */
|
|
|
|
#define _do_init \
|
|
GST_DEBUG_CATEGORY_INIT (opensles_src_debug, "opensles_src", 0, \
|
|
"OpenSL ES Src");
|
|
#define parent_class gst_opensles_src_parent_class
|
|
G_DEFINE_TYPE_WITH_CODE (GstOpenSLESSrc, gst_opensles_src,
|
|
GST_TYPE_AUDIO_BASE_SRC, _do_init);
|
|
|
|
static GstAudioRingBuffer *
|
|
gst_opensles_src_create_ringbuffer (GstAudioBaseSrc * base)
|
|
{
|
|
GstAudioRingBuffer *rb;
|
|
|
|
rb = gst_opensles_ringbuffer_new (RB_MODE_SRC);
|
|
|
|
return rb;
|
|
}
|
|
|
|
static void
|
|
gst_opensles_src_class_init (GstOpenSLESSrcClass * klass)
|
|
{
|
|
GstElementClass *gstelement_class;
|
|
GstAudioBaseSrcClass *gstaudiobasesrc_class;
|
|
|
|
gstelement_class = (GstElementClass *) klass;
|
|
gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass;
|
|
|
|
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>");
|
|
|
|
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 = 400000;
|
|
GST_AUDIO_BASE_SRC (src)->latency_time = 20000;
|
|
}
|