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100 lines
3 KiB
C
100 lines
3 KiB
C
/* GStreamer
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* Copyright (C) 2012 Fluendo S.A. <support@fluendo.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-openslessrc
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* @see_also: openslessink
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*
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* This element reads data from default audio input using the OpenSL ES API in Android OS.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch -v openslessrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=recorded.ogg
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* ]| Record from default audio input and encode to Ogg/Vorbis.
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* </refsect2>
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*
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*/
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#ifdef HAVE_CONFIG_H
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# include <config.h>
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#endif
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#include "openslessrc.h"
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GST_DEBUG_CATEGORY_STATIC (opensles_src_debug);
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#define GST_CAT_DEFAULT opensles_src_debug
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/* *INDENT-OFF* */
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static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"rate = (int) 16000, "
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"channels = (int) 1")
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);
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/* *INDENT-ON* */
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#define _do_init \
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GST_DEBUG_CATEGORY_INIT (opensles_src_debug, "opensles_src", 0, \
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"OpenSL ES Src");
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#define parent_class gst_opensles_src_parent_class
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G_DEFINE_TYPE_WITH_CODE (GstOpenSLESSrc, gst_opensles_src,
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GST_TYPE_AUDIO_BASE_SRC, _do_init);
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static GstAudioRingBuffer *
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gst_opensles_src_create_ringbuffer (GstAudioBaseSrc * base)
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{
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GstAudioRingBuffer *rb;
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rb = gst_opensles_ringbuffer_new (RB_MODE_SRC);
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return rb;
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}
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static void
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gst_opensles_src_class_init (GstOpenSLESSrcClass * klass)
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{
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GstElementClass *gstelement_class;
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GstAudioBaseSrcClass *gstaudiobasesrc_class;
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gstelement_class = (GstElementClass *) klass;
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gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&src_factory));
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gst_element_class_set_static_metadata (gstelement_class, "OpenSL ES Src",
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"Source/Audio",
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"Input sound using the OpenSL ES APIs",
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"Josep Torra <support@fluendo.com>");
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gstaudiobasesrc_class->create_ringbuffer =
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GST_DEBUG_FUNCPTR (gst_opensles_src_create_ringbuffer);
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}
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static void
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gst_opensles_src_init (GstOpenSLESSrc * src)
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{
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/* Override some default values to fit on the AudioFlinger behaviour of
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* processing 20ms buffers as minimum buffer size. */
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GST_AUDIO_BASE_SRC (src)->buffer_time = 200000;
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GST_AUDIO_BASE_SRC (src)->latency_time = 20000;
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}
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