| /* |
| * 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); |
| } |