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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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582 lines
18 KiB
C
582 lines
18 KiB
C
/*
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* GStreamer
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*
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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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* Copyright (C) 2013 Juan Manuel Borges Caño <juanmabcmail@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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* Copyright (C) 2013 Juan Manuel Borges Caño <juanmabcmail@gmail.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-openalsrc
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* @see_also: openalsink
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* @short_description: capture raw audio samples through OpenAL
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*
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* This element captures raw audio samples through OpenAL.
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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 openalsrc ! audioconvert ! wavenc ! filesink location=stream.wav
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* ]| * will capture sound through OpenAL and encode it to a wav file.
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* |[
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* gst-launch openalsrc ! "audio/x-raw,format=S16LE,rate=44100" ! audioconvert ! volume volume=0.25 ! openalsink
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* ]| will capture and play audio through OpenAL.
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* </refsect2>
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*/
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/*
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* DEV:
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* To get better timing/delay information you may also be interested in this:
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* http://kcat.strangesoft.net/openal-extensions/SOFT_source_latency.txt
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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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GST_DEBUG_CATEGORY_EXTERN (openal_debug);
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#define GST_CAT_DEFAULT openal_debug
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#include "gstopenalsrc.h"
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static void gst_openal_src_dispose (GObject * object);
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static void gst_openal_src_finalize (GObject * object);
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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 GstCaps *gst_openal_src_getcaps (GstBaseSrc * basesrc, GstCaps * filter);
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static gboolean gst_openal_src_open (GstAudioSrc * audiosrc);
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static gboolean gst_openal_src_prepare (GstAudioSrc * audiosrc,
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GstAudioRingBufferSpec * spec);
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static gboolean gst_openal_src_unprepare (GstAudioSrc * audiosrc);
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static gboolean gst_openal_src_close (GstAudioSrc * audiosrc);
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static guint gst_openal_src_read (GstAudioSrc * audiosrc, gpointer data,
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guint length, GstClockTime * timestamp);
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static guint gst_openal_src_delay (GstAudioSrc * audiosrc);
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static void gst_openal_src_reset (GstAudioSrc * audiosrc);
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#define OPENAL_DEFAULT_DEVICE_NAME NULL
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#define OPENAL_DEFAULT_DEVICE NULL
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#define OPENAL_MIN_RATE 8000
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#define OPENAL_MAX_RATE 192000
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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 openalsrc_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 (
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/* These caps do not work on my card */
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// "audio/x-adpcm, " "layout = (string) ima, "
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// "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
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// "audio/x-alaw, " "rate = (int) [ 1, MAX ], "
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// "channels = (int) 1; "
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// "audio/x-mulaw, " "rate = (int) [ 1, MAX ], "
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// "channels = (int) 1; "
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// "audio/x-raw, " "format = (string) " GST_AUDIO_NE (F64) ", "
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// "rate = (int) [ 1, MAX ], " "channels = (int) 1; "
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// "audio/x-raw, " "format = (string) " GST_AUDIO_NE (F32) ", "
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// "rate = (int) [ 1, MAX ], " "channels = (int) 1; "
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"audio/x-raw, " "format = (string) " GST_AUDIO_NE (S16) ", "
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"rate = (int) [ 1, MAX ], " "channels = (int) 1; "
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/* These caps work wrongly on my card */
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// "audio/x-raw, " "format = (string) " GST_AUDIO_NE (U16) ", "
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// "rate = (int) [ 1, MAX ], " "channels = (int) 1; "
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// "audio/x-raw, " "format = (string) " G_STRINGIFY (S8) ", "
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// "rate = (int) [ 1, MAX ], " "channels = (int) 1"));
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"audio/x-raw, " "format = (string) " G_STRINGIFY (U8) ", "
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"rate = (int) [ 1, MAX ], " "channels = (int) 1")
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);
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G_DEFINE_TYPE (GstOpenalSrc, gst_openal_src, GST_TYPE_AUDIO_SRC);
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static void
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gst_openal_src_dispose (GObject * object)
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{
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GstOpenalSrc *openalsrc = GST_OPENAL_SRC (object);
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if (openalsrc->probed_caps)
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gst_caps_unref (openalsrc->probed_caps);
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openalsrc->probed_caps = NULL;
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G_OBJECT_CLASS (gst_openal_src_parent_class)->dispose (object);
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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 = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = (GstElementClass *) klass;
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GstBaseSrcClass *gstbasesrc_class = (GstBaseSrcClass *) klass;
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GstAudioSrcClass *gstaudiosrc_class = (GstAudioSrcClass *) (klass);
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gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_openal_src_dispose);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_openal_src_finalize);
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gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_openal_src_set_property);
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gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_openal_src_get_property);
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gst_openal_src_parent_class = g_type_class_peek_parent (klass);
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gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_openal_src_getcaps);
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gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_openal_src_open);
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gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_openal_src_prepare);
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gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_openal_src_unprepare);
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gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_openal_src_close);
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gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_openal_src_read);
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gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_openal_src_delay);
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gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_openal_src_reset);
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g_object_class_install_property (gobject_class, PROP_DEVICE,
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g_param_spec_string ("device", "ALCdevice",
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"User device, default device if NULL", OPENAL_DEFAULT_DEVICE,
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G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
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g_param_spec_string ("device-name", "Device name",
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"Human-readable name of the device", OPENAL_DEFAULT_DEVICE_NAME,
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G_PARAM_READABLE));
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gst_element_class_set_static_metadata (gstelement_class,
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"OpenAL Audio Source", "Source/Audio", "Input audio through OpenAL",
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"Juan Manuel Borges Caño <juanmabcmail@gmail.com>");
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&openalsrc_factory));
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}
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static void
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gst_openal_src_init (GstOpenalSrc * openalsrc)
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{
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GST_DEBUG_OBJECT (openalsrc, "initializing");
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openalsrc->default_device_name = g_strdup (OPENAL_DEFAULT_DEVICE_NAME);
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openalsrc->default_device = OPENAL_DEFAULT_DEVICE;
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openalsrc->device = NULL;
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openalsrc->buffer_length = 0;
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openalsrc->probed_caps = NULL;
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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 *openalsrc = GST_OPENAL_SRC (object);
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g_free (openalsrc->default_device_name);
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g_free (openalsrc->default_device);
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G_OBJECT_CLASS (gst_openal_src_parent_class)->finalize (object);
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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 *openalsrc = GST_OPENAL_SRC (object);
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switch (prop_id) {
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case PROP_DEVICE:
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openalsrc->default_device = g_value_dup_string (value);
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break;
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case PROP_DEVICE_NAME:
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openalsrc->default_device_name = 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 *openalsrc = 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, openalsrc->default_device);
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break;
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case PROP_DEVICE_NAME:
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g_value_set_string (value, openalsrc->default_device_name);
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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 GstCaps *
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gst_openal_helper_probe_caps (ALCcontext * context)
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{
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GstStructure *structure;
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GstCaps *caps;
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// ALCcontext *old;
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// old = pushContext(context);
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caps = gst_caps_new_empty ();
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if (alIsExtensionPresent ("AL_EXT_DOUBLE")) {
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structure =
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gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
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GST_AUDIO_NE (F64), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
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OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
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gst_caps_append_structure (caps, structure);
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}
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if (alIsExtensionPresent ("AL_EXT_FLOAT32")) {
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structure =
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gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
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GST_AUDIO_NE (F32), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
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OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
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gst_caps_append_structure (caps, structure);
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}
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structure =
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gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
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GST_AUDIO_NE (S16), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
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OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
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gst_caps_append_structure (caps, structure);
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structure =
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gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
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G_STRINGIFY (U8), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
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OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
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gst_caps_append_structure (caps, structure);
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if (alIsExtensionPresent ("AL_EXT_IMA4")) {
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structure =
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gst_structure_new ("audio/x-adpcm", "layout", G_TYPE_STRING, "ima",
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"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
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"channels", G_TYPE_INT, 1, NULL);
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gst_caps_append_structure (caps, structure);
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}
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if (alIsExtensionPresent ("AL_EXT_ALAW")) {
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structure =
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gst_structure_new ("audio/x-alaw", "rate", GST_TYPE_INT_RANGE,
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OPENAL_MIN_RATE, OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
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gst_caps_append_structure (caps, structure);
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}
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if (alIsExtensionPresent ("AL_EXT_MULAW")) {
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structure =
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gst_structure_new ("audio/x-mulaw", "rate", GST_TYPE_INT_RANGE,
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OPENAL_MIN_RATE, OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
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gst_caps_append_structure (caps, structure);
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}
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// popContext(old, context);
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return caps;
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}
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static GstCaps *
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gst_openal_src_getcaps (GstBaseSrc * basesrc, GstCaps * filter)
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{
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GstOpenalSrc *openalsrc = GST_OPENAL_SRC (basesrc);
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GstCaps *caps;
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ALCdevice *device;
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device = alcOpenDevice (NULL);
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if (device == NULL) {
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GstPad *pad = GST_BASE_SRC_PAD (basesrc);
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GST_ELEMENT_WARNING (openalsrc, RESOURCE, OPEN_WRITE,
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("Could not open temporary device."), GST_ALC_ERROR (device));
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caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
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} else if (openalsrc->probed_caps)
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caps = gst_caps_copy (openalsrc->probed_caps);
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else {
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ALCcontext *context = alcCreateContext (device, NULL);
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if (context) {
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caps = gst_openal_helper_probe_caps (context);
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alcDestroyContext (context);
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} else {
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GST_ELEMENT_WARNING (openalsrc, RESOURCE, FAILED,
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("Could not create temporary context."), GST_ALC_ERROR (device));
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caps = NULL;
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}
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if (caps && !gst_caps_is_empty (caps))
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openalsrc->probed_caps = gst_caps_copy (caps);
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}
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if (device != NULL) {
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if (alcCloseDevice (device) == ALC_FALSE) {
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GST_ELEMENT_WARNING (openalsrc, RESOURCE, CLOSE,
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("Could not close temporary device."), GST_ALC_ERROR (device));
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}
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}
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if (filter) {
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GstCaps *intersection;
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intersection =
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gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
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return intersection;
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} else {
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return caps;
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}
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}
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static gboolean
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gst_openal_src_open (GstAudioSrc * audiosrc)
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{
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return TRUE;
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}
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static void
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gst_openal_src_parse_spec (GstOpenalSrc * openalsrc,
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const GstAudioRingBufferSpec * spec)
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{
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ALuint format = AL_NONE;
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GST_DEBUG_OBJECT (openalsrc,
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"looking up format for type %d, gst-format %d, and %d channels",
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spec->type, GST_AUDIO_INFO_FORMAT (&spec->info),
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GST_AUDIO_INFO_CHANNELS (&spec->info));
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switch (spec->type) {
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
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switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
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case GST_AUDIO_FORMAT_U8:
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switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
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case 1:
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format = AL_FORMAT_MONO8;
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break;
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default:
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break;
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}
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break;
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case GST_AUDIO_FORMAT_U16:
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case GST_AUDIO_FORMAT_S16:
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switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
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case 1:
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format = AL_FORMAT_MONO16;
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break;
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default:
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break;
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}
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break;
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case GST_AUDIO_FORMAT_F32:
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switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
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case 1:
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format = AL_FORMAT_MONO_FLOAT32;
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break;
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default:
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break;
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}
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break;
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case GST_AUDIO_FORMAT_F64:
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switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
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case 1:
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format = AL_FORMAT_MONO_DOUBLE_EXT;
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break;
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default:
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break;
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}
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break;
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default:
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break;
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}
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break;
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM:
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switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
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case 1:
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format = AL_FORMAT_MONO_IMA4;
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break;
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default:
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break;
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}
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break;
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|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:
|
|
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
|
|
case 1:
|
|
format = AL_FORMAT_MONO_ALAW_EXT;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
break;
|
|
|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:
|
|
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
|
|
case 1:
|
|
format = AL_FORMAT_MONO_MULAW;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
break;
|
|
|
|
default:
|
|
break;
|
|
}
|
|
|
|
openalsrc->bytes_per_sample = GST_AUDIO_INFO_BPS (&spec->info);
|
|
openalsrc->rate = GST_AUDIO_INFO_RATE (&spec->info);
|
|
openalsrc->buffer_length = spec->segsize;
|
|
openalsrc->format = format;
|
|
}
|
|
|
|
static gboolean
|
|
gst_openal_src_prepare (GstAudioSrc * audiosrc, GstAudioRingBufferSpec * spec)
|
|
{
|
|
GstOpenalSrc *openalsrc = GST_OPENAL_SRC (audiosrc);
|
|
|
|
gst_openal_src_parse_spec (openalsrc, spec);
|
|
if (openalsrc->format == AL_NONE) {
|
|
GST_ELEMENT_ERROR (openalsrc, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to get type %d, format %d, and %d channels", spec->type,
|
|
GST_AUDIO_INFO_FORMAT (&spec->info),
|
|
GST_AUDIO_INFO_CHANNELS (&spec->info)));
|
|
return FALSE;
|
|
}
|
|
|
|
openalsrc->device =
|
|
alcCaptureOpenDevice (openalsrc->default_device, openalsrc->rate,
|
|
openalsrc->format, openalsrc->buffer_length);
|
|
|
|
if (!openalsrc->device) {
|
|
GST_ELEMENT_ERROR (openalsrc, RESOURCE, OPEN_READ,
|
|
("Could not open device."), GST_ALC_ERROR (openalsrc->device));
|
|
return FALSE;
|
|
}
|
|
|
|
openalsrc->default_device_name =
|
|
g_strdup (alcGetString (openalsrc->device, ALC_DEVICE_SPECIFIER));
|
|
|
|
alcCaptureStart (openalsrc->device);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_openal_src_unprepare (GstAudioSrc * audiosrc)
|
|
{
|
|
GstOpenalSrc *openalsrc = GST_OPENAL_SRC (audiosrc);
|
|
|
|
if (openalsrc->device) {
|
|
alcCaptureStop (openalsrc->device);
|
|
|
|
if (alcCaptureCloseDevice (openalsrc->device) == ALC_FALSE) {
|
|
GST_ELEMENT_ERROR (openalsrc, RESOURCE, CLOSE,
|
|
("Could not close device."), GST_ALC_ERROR (openalsrc->device));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_openal_src_close (GstAudioSrc * audiosrc)
|
|
{
|
|
return TRUE;
|
|
}
|
|
|
|
static guint
|
|
gst_openal_src_read (GstAudioSrc * audiosrc, gpointer data, guint length,
|
|
GstClockTime * timestamp)
|
|
{
|
|
GstOpenalSrc *openalsrc = GST_OPENAL_SRC (audiosrc);
|
|
gint samples;
|
|
|
|
alcGetIntegerv (openalsrc->device, ALC_CAPTURE_SAMPLES, sizeof (samples),
|
|
&samples);
|
|
|
|
if (samples * openalsrc->bytes_per_sample > length) {
|
|
samples = length / openalsrc->bytes_per_sample;
|
|
}
|
|
|
|
if (samples) {
|
|
GST_DEBUG_OBJECT (openalsrc, "read samples : %d", samples);
|
|
alcCaptureSamples (openalsrc->device, data, samples);
|
|
}
|
|
|
|
return samples * openalsrc->bytes_per_sample;
|
|
}
|
|
|
|
static guint
|
|
gst_openal_src_delay (GstAudioSrc * audiosrc)
|
|
{
|
|
GstOpenalSrc *openalsrc = GST_OPENAL_SRC (audiosrc);
|
|
ALint samples;
|
|
|
|
alcGetIntegerv (openalsrc->device, ALC_CAPTURE_SAMPLES, sizeof (samples),
|
|
&samples);
|
|
|
|
if (G_UNLIKELY (samples < 0)) {
|
|
/* make sure we never return a negative delay */
|
|
GST_WARNING_OBJECT (openal_debug, "negative delay");
|
|
samples = 0;
|
|
}
|
|
|
|
return samples;
|
|
}
|
|
|
|
static void
|
|
gst_openal_src_reset (GstAudioSrc * audiosrc)
|
|
{
|
|
}
|