mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 20:21:24 +00:00
0e596670ef
Based on patches by Victor Lin <bornstub@gmail.com> Fixes bug #550230.
359 lines
10 KiB
C
359 lines
10 KiB
C
/*
|
|
* GStreamer
|
|
* Copyright (C) 2005 Thomas Vander Stichele <thomas@apestaart.org>
|
|
* Copyright (C) 2005 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
|
|
* Copyright (C) 2008 Victor Lin <bornstub@gmail.com>
|
|
*
|
|
* Permission is hereby granted, free of charge, to any person obtaining a
|
|
* copy of this software and associated documentation files (the "Software"),
|
|
* to deal in the Software without restriction, including without limitation
|
|
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
|
|
* and/or sell copies of the Software, and to permit persons to whom the
|
|
* Software is furnished to do so, subject to the following conditions:
|
|
*
|
|
* The above copyright notice and this permission notice shall be included in
|
|
* all copies or substantial portions of the Software.
|
|
*
|
|
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
|
|
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
|
|
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
|
|
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
|
|
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
|
|
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
|
|
* DEALINGS IN THE SOFTWARE.
|
|
*
|
|
* Alternatively, the contents of this file may be used under the
|
|
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
|
|
* which case the following provisions apply instead of the ones
|
|
* mentioned above:
|
|
*
|
|
* 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., 59 Temple Place - Suite 330,
|
|
* Boston, MA 02111-1307, USA.
|
|
*/
|
|
|
|
/**
|
|
* SECTION:element-openalsrc
|
|
* @short_description: record sound from your sound card using OpenAL
|
|
*
|
|
* <refsect2>
|
|
* <para>
|
|
* This element lets you record sound using the OpenAL
|
|
* </para>
|
|
* <title>Example pipelines</title>
|
|
* <para>
|
|
* <programlisting>
|
|
* gst-launch -v openalsrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg
|
|
* </programlisting>
|
|
* will record sound from your sound card using OpenAL and encode it to an Ogg/Vorbis file
|
|
* </para>
|
|
* </refsect2>
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
# include <config.h>
|
|
#endif
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/gsterror.h>
|
|
|
|
#include "gstopenalsrc.h"
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (openalsrc_debug);
|
|
|
|
#define GST_CAT_DEFAULT openalsrc_debug
|
|
|
|
#define DEFAULT_DEVICE NULL
|
|
#define DEFAULT_DEVICE_NAME NULL
|
|
|
|
/**
|
|
Filter signals and args
|
|
**/
|
|
enum
|
|
{
|
|
/* FILL ME */
|
|
LAST_SIGNAL
|
|
};
|
|
|
|
|
|
/**
|
|
Properties
|
|
**/
|
|
enum
|
|
{
|
|
PROP_0,
|
|
PROP_DEVICE,
|
|
PROP_DEVICE_NAME
|
|
};
|
|
|
|
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
|
|
GST_PAD_SRC,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-raw-int, "
|
|
"endianness = (int) BYTE_ORDER, "
|
|
"signed = (boolean) TRUE, "
|
|
"width = (int) 16, "
|
|
"depth = (int) 16, "
|
|
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
|
|
"audio/x-raw-int, "
|
|
"signed = (boolean) TRUE, "
|
|
"width = (int) 8, "
|
|
"depth = (int) 8, "
|
|
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
|
|
);
|
|
|
|
GST_BOILERPLATE (GstOpenalSrc, gst_openal_src, GstAudioSrc, GST_TYPE_AUDIO_SRC);
|
|
|
|
static void gst_openal_src_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec);
|
|
static void gst_openal_src_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec);
|
|
static gboolean gst_openal_src_open (GstAudioSrc * src);
|
|
static gboolean
|
|
gst_openal_src_prepare (GstAudioSrc * src, GstRingBufferSpec * spec);
|
|
static gboolean gst_openal_src_unprepare (GstAudioSrc * src);
|
|
static gboolean gst_openal_src_close (GstAudioSrc * src);
|
|
static guint
|
|
gst_openal_src_read (GstAudioSrc * src, gpointer data, guint length);
|
|
static guint gst_openal_src_delay (GstAudioSrc * src);
|
|
static void gst_openal_src_reset (GstAudioSrc * src);
|
|
|
|
static void gst_openal_src_finalize (GObject * object);
|
|
|
|
static void
|
|
gst_openal_src_base_init (gpointer gclass)
|
|
{
|
|
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
|
|
|
|
gst_element_class_set_details_simple (element_class, "OpenAL src",
|
|
"Source/Audio",
|
|
"OpenAL source capture audio from device",
|
|
"Victor Lin <bornstub@gmail.com>");
|
|
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&src_factory)
|
|
);
|
|
}
|
|
|
|
static void
|
|
gst_openal_src_class_init (GstOpenalSrcClass * klass)
|
|
{
|
|
GObjectClass *gobject_class;
|
|
GstAudioSrcClass *gstaudio_src_class;
|
|
|
|
gobject_class = G_OBJECT_CLASS (klass);
|
|
gstaudio_src_class = GST_AUDIO_SRC_CLASS (klass);
|
|
|
|
GST_DEBUG_CATEGORY_INIT (openalsrc_debug, "openalsrc",
|
|
0, "OpenAL source capture audio from device");
|
|
|
|
gobject_class->set_property = gst_openal_src_set_property;
|
|
gobject_class->get_property = gst_openal_src_get_property;
|
|
gobject_class->finalize = gst_openal_src_finalize;
|
|
|
|
gstaudio_src_class->open = GST_DEBUG_FUNCPTR (gst_openal_src_open);
|
|
gstaudio_src_class->prepare = GST_DEBUG_FUNCPTR (gst_openal_src_prepare);
|
|
gstaudio_src_class->unprepare = GST_DEBUG_FUNCPTR (gst_openal_src_unprepare);
|
|
gstaudio_src_class->close = GST_DEBUG_FUNCPTR (gst_openal_src_close);
|
|
gstaudio_src_class->read = GST_DEBUG_FUNCPTR (gst_openal_src_read);
|
|
gstaudio_src_class->delay = GST_DEBUG_FUNCPTR (gst_openal_src_delay);
|
|
gstaudio_src_class->reset = GST_DEBUG_FUNCPTR (gst_openal_src_reset);
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_DEVICE,
|
|
g_param_spec_string ("device",
|
|
"Device",
|
|
"Specific capture device to open, NULL indicate default device",
|
|
DEFAULT_DEVICE, G_PARAM_READWRITE)
|
|
);
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_DEVICE_NAME,
|
|
g_param_spec_string ("device-name",
|
|
"Device name",
|
|
"Readable name of device", DEFAULT_DEVICE_NAME, G_PARAM_READABLE)
|
|
);
|
|
}
|
|
|
|
static void
|
|
gst_openal_src_init (GstOpenalSrc * osrc, GstOpenalSrcClass * gclass)
|
|
{
|
|
osrc->deviceName = g_strdup (DEFAULT_DEVICE_NAME);
|
|
osrc->device = DEFAULT_DEVICE;
|
|
osrc->deviceHandle = NULL;
|
|
}
|
|
|
|
static void
|
|
gst_openal_src_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstOpenalSrc *osrc = GST_OPENAL_SRC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_DEVICE:
|
|
osrc->device = g_value_dup_string (value);
|
|
break;
|
|
case PROP_DEVICE_NAME:
|
|
osrc->deviceName = g_value_dup_string (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_openal_src_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstOpenalSrc *osrc = GST_OPENAL_SRC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_DEVICE:
|
|
g_value_set_string (value, osrc->device);
|
|
break;
|
|
case PROP_DEVICE_NAME:
|
|
g_value_set_string (value, osrc->deviceName);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_openal_src_open (GstAudioSrc * asrc)
|
|
{
|
|
/* We don't do anything here */
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_openal_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
|
|
{
|
|
|
|
GstOpenalSrc *osrc = GST_OPENAL_SRC (asrc);
|
|
ALenum format;
|
|
guint64 bufferSize;
|
|
|
|
switch (spec->width) {
|
|
case 8:
|
|
format = AL_FORMAT_STEREO8;
|
|
break;
|
|
case 16:
|
|
format = AL_FORMAT_STEREO16;
|
|
break;
|
|
default:
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
bufferSize =
|
|
spec->buffer_time * spec->rate * spec->bytes_per_sample / 1000000;
|
|
|
|
GST_INFO_OBJECT (osrc, "Open device : %s", osrc->deviceName);
|
|
osrc->deviceHandle =
|
|
alcCaptureOpenDevice (osrc->device, spec->rate, format, bufferSize);
|
|
|
|
if (!osrc->deviceHandle) {
|
|
GST_ELEMENT_ERROR (osrc,
|
|
RESOURCE,
|
|
FAILED,
|
|
("Can't open device \"%s\"", osrc->device),
|
|
("Can't open device \"%s\"", osrc->device)
|
|
);
|
|
return FALSE;
|
|
}
|
|
|
|
osrc->deviceName =
|
|
g_strdup (alcGetString (osrc->deviceHandle, ALC_DEVICE_SPECIFIER));
|
|
osrc->bytes_per_sample = spec->bytes_per_sample;
|
|
|
|
GST_INFO_OBJECT (osrc, "Start capture");
|
|
alcCaptureStart (osrc->deviceHandle);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_openal_src_unprepare (GstAudioSrc * asrc)
|
|
{
|
|
|
|
GstOpenalSrc *osrc = GST_OPENAL_SRC (asrc);
|
|
|
|
GST_INFO_OBJECT (osrc, "Close device : %s", osrc->deviceName);
|
|
if (osrc->deviceHandle) {
|
|
alcCaptureStop (osrc->deviceHandle);
|
|
alcCaptureCloseDevice (osrc->deviceHandle);
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_openal_src_close (GstAudioSrc * asrc)
|
|
{
|
|
/* We don't do anything here */
|
|
return TRUE;
|
|
}
|
|
|
|
static guint
|
|
gst_openal_src_read (GstAudioSrc * asrc, gpointer data, guint length)
|
|
{
|
|
GstOpenalSrc *osrc = GST_OPENAL_SRC (asrc);
|
|
gint samples;
|
|
|
|
alcGetIntegerv (osrc->deviceHandle, ALC_CAPTURE_SAMPLES, sizeof (samples),
|
|
&samples);
|
|
|
|
if (samples * osrc->bytes_per_sample > length) {
|
|
samples = length / osrc->bytes_per_sample;
|
|
}
|
|
|
|
if (samples) {
|
|
GST_DEBUG_OBJECT (osrc, "Read samples : %d", samples);
|
|
alcCaptureSamples (osrc->deviceHandle, data, samples);
|
|
}
|
|
|
|
return samples * osrc->bytes_per_sample;
|
|
}
|
|
|
|
static guint
|
|
gst_openal_src_delay (GstAudioSrc * asrc)
|
|
{
|
|
GstOpenalSrc *osrc = GST_OPENAL_SRC (asrc);
|
|
gint samples;
|
|
|
|
alcGetIntegerv (osrc->deviceHandle, ALC_CAPTURE_SAMPLES, sizeof (samples),
|
|
&samples);
|
|
|
|
return samples;
|
|
}
|
|
|
|
static void
|
|
gst_openal_src_reset (GstAudioSrc * asrc)
|
|
{
|
|
/* We don't do anything here */
|
|
}
|
|
|
|
static void
|
|
gst_openal_src_finalize (GObject * object)
|
|
{
|
|
GstOpenalSrc *osrc = GST_OPENAL_SRC (object);
|
|
|
|
g_free (osrc->deviceName);
|
|
g_free (osrc->device);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|