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6b728f184f
This allows us to signal what kind of audio we are expecting to record, which should tell the system to apply filters (such as echo cancellation, noise suppression, etc.) if required.
155 lines
4.4 KiB
C
155 lines
4.4 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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"layout = (string) interleaved")
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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, "openslessrc", 0, \
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"OpenSLES Source");
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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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enum
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{
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PROP_0,
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PROP_PRESET,
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};
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#define DEFAULT_PRESET GST_OPENSLES_RECORDING_PRESET_NONE
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static void
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gst_opensles_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstOpenSLESSrc *src = GST_OPENSLES_SRC (object);
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switch (prop_id) {
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case PROP_PRESET:
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src->preset = g_value_get_enum (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_opensles_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstOpenSLESSrc *src = GST_OPENSLES_SRC (object);
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switch (prop_id) {
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case PROP_PRESET:
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g_value_set_enum (value, src->preset);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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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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GST_OPENSLES_RING_BUFFER (rb)->preset = GST_OPENSLES_SRC (base)->preset;
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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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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstAudioBaseSrcClass *gstaudiobasesrc_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass;
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gobject_class->set_property = gst_opensles_src_set_property;
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gobject_class->get_property = gst_opensles_src_get_property;
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g_object_class_install_property (gobject_class, PROP_PRESET,
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g_param_spec_enum ("preset", "Preset", "Recording preset to use",
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GST_TYPE_OPENSLES_RECORDING_PRESET, DEFAULT_PRESET,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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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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src->preset = DEFAULT_PRESET;
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}
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