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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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0e596670ef
Based on patches by Victor Lin <bornstub@gmail.com> Fixes bug #550230.
359 lines
10 KiB
C
359 lines
10 KiB
C
/*
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* GStreamer
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* Copyright (C) 2005 Thomas Vander Stichele <thomas@apestaart.org>
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* Copyright (C) 2005 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
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* Copyright (C) 2008 Victor Lin <bornstub@gmail.com>
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*
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* Permission is hereby granted, free of charge, to any person obtaining a
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* copy of this software and associated documentation files (the "Software"),
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* to deal in the Software without restriction, including without limitation
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* the rights to use, copy, modify, merge, publish, distribute, sublicense,
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* and/or sell copies of the Software, and to permit persons to whom the
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* Software is furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
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* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
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* DEALINGS IN THE SOFTWARE.
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*
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* Alternatively, the contents of this file may be used under the
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* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
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* which case the following provisions apply instead of the ones
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* mentioned above:
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-openalsrc
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* @short_description: record sound from your sound card using OpenAL
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*
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* <refsect2>
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* <para>
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* This element lets you record sound using the OpenAL
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* </para>
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* <title>Example pipelines</title>
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* <para>
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* <programlisting>
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* gst-launch -v openalsrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg
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* </programlisting>
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* will record sound from your sound card using OpenAL and encode it to an Ogg/Vorbis file
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* </para>
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* </refsect2>
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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 <gst/gst.h>
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#include <gst/gsterror.h>
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#include "gstopenalsrc.h"
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GST_DEBUG_CATEGORY_STATIC (openalsrc_debug);
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#define GST_CAT_DEFAULT openalsrc_debug
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#define DEFAULT_DEVICE NULL
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#define DEFAULT_DEVICE_NAME NULL
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/**
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Filter signals and args
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**/
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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/**
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Properties
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**/
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enum
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{
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PROP_0,
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PROP_DEVICE,
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PROP_DEVICE_NAME
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};
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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-int, "
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"endianness = (int) BYTE_ORDER, "
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"signed = (boolean) TRUE, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
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"audio/x-raw-int, "
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"signed = (boolean) TRUE, "
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"width = (int) 8, "
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"depth = (int) 8, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
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);
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GST_BOILERPLATE (GstOpenalSrc, gst_openal_src, GstAudioSrc, GST_TYPE_AUDIO_SRC);
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static void gst_openal_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_openal_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_openal_src_open (GstAudioSrc * src);
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static gboolean
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gst_openal_src_prepare (GstAudioSrc * src, GstRingBufferSpec * spec);
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static gboolean gst_openal_src_unprepare (GstAudioSrc * src);
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static gboolean gst_openal_src_close (GstAudioSrc * src);
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static guint
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gst_openal_src_read (GstAudioSrc * src, gpointer data, guint length);
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static guint gst_openal_src_delay (GstAudioSrc * src);
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static void gst_openal_src_reset (GstAudioSrc * src);
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static void gst_openal_src_finalize (GObject * object);
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static void
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gst_openal_src_base_init (gpointer gclass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
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gst_element_class_set_details_simple (element_class, "OpenAL src",
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"Source/Audio",
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"OpenAL source capture audio from device",
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"Victor Lin <bornstub@gmail.com>");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_factory)
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);
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}
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static void
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gst_openal_src_class_init (GstOpenalSrcClass * klass)
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{
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GObjectClass *gobject_class;
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GstAudioSrcClass *gstaudio_src_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gstaudio_src_class = GST_AUDIO_SRC_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (openalsrc_debug, "openalsrc",
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0, "OpenAL source capture audio from device");
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gobject_class->set_property = gst_openal_src_set_property;
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gobject_class->get_property = gst_openal_src_get_property;
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gobject_class->finalize = gst_openal_src_finalize;
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gstaudio_src_class->open = GST_DEBUG_FUNCPTR (gst_openal_src_open);
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gstaudio_src_class->prepare = GST_DEBUG_FUNCPTR (gst_openal_src_prepare);
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gstaudio_src_class->unprepare = GST_DEBUG_FUNCPTR (gst_openal_src_unprepare);
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gstaudio_src_class->close = GST_DEBUG_FUNCPTR (gst_openal_src_close);
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gstaudio_src_class->read = GST_DEBUG_FUNCPTR (gst_openal_src_read);
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gstaudio_src_class->delay = GST_DEBUG_FUNCPTR (gst_openal_src_delay);
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gstaudio_src_class->reset = GST_DEBUG_FUNCPTR (gst_openal_src_reset);
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g_object_class_install_property (gobject_class,
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PROP_DEVICE,
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g_param_spec_string ("device",
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"Device",
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"Specific capture device to open, NULL indicate default device",
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DEFAULT_DEVICE, G_PARAM_READWRITE)
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);
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g_object_class_install_property (gobject_class,
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PROP_DEVICE_NAME,
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g_param_spec_string ("device-name",
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"Device name",
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"Readable name of device", DEFAULT_DEVICE_NAME, G_PARAM_READABLE)
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);
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}
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static void
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gst_openal_src_init (GstOpenalSrc * osrc, GstOpenalSrcClass * gclass)
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{
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osrc->deviceName = g_strdup (DEFAULT_DEVICE_NAME);
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osrc->device = DEFAULT_DEVICE;
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osrc->deviceHandle = NULL;
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}
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static void
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gst_openal_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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GstOpenalSrc *osrc = GST_OPENAL_SRC (object);
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switch (prop_id) {
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case PROP_DEVICE:
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osrc->device = g_value_dup_string (value);
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break;
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case PROP_DEVICE_NAME:
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osrc->deviceName = g_value_dup_string (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_openal_src_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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GstOpenalSrc *osrc = GST_OPENAL_SRC (object);
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switch (prop_id) {
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case PROP_DEVICE:
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g_value_set_string (value, osrc->device);
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break;
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case PROP_DEVICE_NAME:
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g_value_set_string (value, osrc->deviceName);
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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 gboolean
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gst_openal_src_open (GstAudioSrc * asrc)
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{
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/* We don't do anything here */
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return TRUE;
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}
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static gboolean
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gst_openal_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
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{
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GstOpenalSrc *osrc = GST_OPENAL_SRC (asrc);
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ALenum format;
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guint64 bufferSize;
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switch (spec->width) {
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case 8:
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format = AL_FORMAT_STEREO8;
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break;
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case 16:
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format = AL_FORMAT_STEREO16;
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break;
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default:
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g_assert_not_reached ();
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}
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bufferSize =
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spec->buffer_time * spec->rate * spec->bytes_per_sample / 1000000;
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GST_INFO_OBJECT (osrc, "Open device : %s", osrc->deviceName);
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osrc->deviceHandle =
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alcCaptureOpenDevice (osrc->device, spec->rate, format, bufferSize);
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if (!osrc->deviceHandle) {
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GST_ELEMENT_ERROR (osrc,
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RESOURCE,
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FAILED,
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("Can't open device \"%s\"", osrc->device),
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("Can't open device \"%s\"", osrc->device)
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);
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return FALSE;
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}
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osrc->deviceName =
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g_strdup (alcGetString (osrc->deviceHandle, ALC_DEVICE_SPECIFIER));
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osrc->bytes_per_sample = spec->bytes_per_sample;
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GST_INFO_OBJECT (osrc, "Start capture");
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alcCaptureStart (osrc->deviceHandle);
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return TRUE;
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}
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static gboolean
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gst_openal_src_unprepare (GstAudioSrc * asrc)
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{
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GstOpenalSrc *osrc = GST_OPENAL_SRC (asrc);
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GST_INFO_OBJECT (osrc, "Close device : %s", osrc->deviceName);
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if (osrc->deviceHandle) {
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alcCaptureStop (osrc->deviceHandle);
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alcCaptureCloseDevice (osrc->deviceHandle);
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}
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return TRUE;
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}
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static gboolean
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gst_openal_src_close (GstAudioSrc * asrc)
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{
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/* We don't do anything here */
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return TRUE;
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}
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static guint
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gst_openal_src_read (GstAudioSrc * asrc, gpointer data, guint length)
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{
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GstOpenalSrc *osrc = GST_OPENAL_SRC (asrc);
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gint samples;
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alcGetIntegerv (osrc->deviceHandle, ALC_CAPTURE_SAMPLES, sizeof (samples),
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&samples);
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if (samples * osrc->bytes_per_sample > length) {
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samples = length / osrc->bytes_per_sample;
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}
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if (samples) {
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GST_DEBUG_OBJECT (osrc, "Read samples : %d", samples);
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alcCaptureSamples (osrc->deviceHandle, data, samples);
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}
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return samples * osrc->bytes_per_sample;
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}
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static guint
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gst_openal_src_delay (GstAudioSrc * asrc)
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{
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GstOpenalSrc *osrc = GST_OPENAL_SRC (asrc);
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gint samples;
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alcGetIntegerv (osrc->deviceHandle, ALC_CAPTURE_SAMPLES, sizeof (samples),
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&samples);
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return samples;
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}
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static void
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gst_openal_src_reset (GstAudioSrc * asrc)
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{
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/* We don't do anything here */
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}
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static void
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gst_openal_src_finalize (GObject * object)
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{
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GstOpenalSrc *osrc = GST_OPENAL_SRC (object);
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g_free (osrc->deviceName);
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g_free (osrc->device);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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