ext/openal/gstopenalsrc.c - imx-gst-plugins-bad - Git at Google

 /*
  * 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>
  * Copyright (C) 2013 Juan Manuel Borges Caño <juanmabcmail@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:
  *
  * Copyright (C) 2013 Juan Manuel Borges Caño <juanmabcmail@gmail.com>
  *
  * 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., 51 Franklin St, Fifth Floor,
  * Boston, MA 02110-1301, USA.
  */

  /**
  * SECTION:element-openalsrc
  * @title: openalsrc
  * @see_also: openalsink
  * @short_description: capture raw audio samples through OpenAL
  *
  * This element captures raw audio samples through OpenAL.
  *
  * ## Example pipelines
  * |[
  * gst-launch-1.0 -v openalsrc ! audioconvert ! wavenc ! filesink location=stream.wav
  * ]| * will capture sound through OpenAL and encode it to a wav file.
  * |[
  * gst-launch-1.0 openalsrc ! "audio/x-raw,format=S16LE,rate=44100" ! audioconvert ! volume volume=0.25 ! openalsink
  * ]| will capture and play audio through OpenAL.
  *
  */

 /*
  * DEV:
  * To get better timing/delay information you may also be interested in this:
  *  http://kcat.strangesoft.net/openal-extensions/SOFT_source_latency.txt
  */

 #ifdef HAVE_CONFIG_H
 #include <config.h>
 #endif

 #include <gst/gst.h>
 #include <gst/gsterror.h>

 GST_DEBUG_CATEGORY_EXTERN (openal_debug);
 #define GST_CAT_DEFAULT openal_debug

 #include "gstopenalsrc.h"

 static void gst_openal_src_dispose (GObject * object);
 static void gst_openal_src_finalize (GObject * object);
 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 GstCaps *gst_openal_src_getcaps (GstBaseSrc * basesrc, GstCaps * filter);
 static gboolean gst_openal_src_open (GstAudioSrc * audiosrc);
 static gboolean gst_openal_src_prepare (GstAudioSrc * audiosrc,
     GstAudioRingBufferSpec * spec);
 static gboolean gst_openal_src_unprepare (GstAudioSrc * audiosrc);
 static gboolean gst_openal_src_close (GstAudioSrc * audiosrc);
 static guint gst_openal_src_read (GstAudioSrc * audiosrc, gpointer data,
     guint length, GstClockTime * timestamp);
 static guint gst_openal_src_delay (GstAudioSrc * audiosrc);
 static void gst_openal_src_reset (GstAudioSrc * audiosrc);

 #define OPENAL_DEFAULT_DEVICE_NAME NULL
 #define OPENAL_DEFAULT_DEVICE NULL

 #define OPENAL_MIN_RATE 8000
 #define OPENAL_MAX_RATE 192000

 enum
 {
   PROP_0,
   PROP_DEVICE,
   PROP_DEVICE_NAME
 };

 static GstStaticPadTemplate openalsrc_factory = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS (
         /* These caps do not work on my card */
         // "audio/x-adpcm, " "layout = (string) ima, "
         // "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
         // "audio/x-alaw, " "rate = (int) [ 1, MAX ], "
         // "channels = (int) 1; "
         // "audio/x-mulaw, " "rate = (int) [ 1, MAX ], "
         // "channels = (int) 1; "
         // "audio/x-raw, " "format = (string) " GST_AUDIO_NE (F64) ", "
         // "rate = (int) [ 1, MAX ], " "channels = (int) 1; "
         // "audio/x-raw, " "format = (string) " GST_AUDIO_NE (F32) ", "
         // "rate = (int) [ 1, MAX ], " "channels = (int) 1; "
         "audio/x-raw, " "format = (string) " GST_AUDIO_NE (S16) ", "
         "rate = (int) [ 1, MAX ], " "channels = (int) 1; "
         /* These caps work wrongly on my card */
         // "audio/x-raw, " "format = (string) " GST_AUDIO_NE (U16) ", "
         // "rate = (int) [ 1, MAX ], " "channels = (int) 1; "
         // "audio/x-raw, " "format = (string) " G_STRINGIFY (S8) ", "
         // "rate = (int) [ 1, MAX ], " "channels = (int) 1"));
         "audio/x-raw, " "format = (string) " G_STRINGIFY (U8) ", "
         "rate = (int) [ 1, MAX ], " "channels = (int) 1")
     );

 G_DEFINE_TYPE (GstOpenalSrc, gst_openal_src, GST_TYPE_AUDIO_SRC);

 static void
 gst_openal_src_dispose (GObject * object)
 {
   GstOpenalSrc *openalsrc = GST_OPENAL_SRC (object);

   if (openalsrc->probed_caps)
     gst_caps_unref (openalsrc->probed_caps);
   openalsrc->probed_caps = NULL;

   G_OBJECT_CLASS (gst_openal_src_parent_class)->dispose (object);
 }

 static void
 gst_openal_src_class_init (GstOpenalSrcClass * klass)
 {
   GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
   GstElementClass *gstelement_class = (GstElementClass *) klass;
   GstBaseSrcClass *gstbasesrc_class = (GstBaseSrcClass *) klass;
   GstAudioSrcClass *gstaudiosrc_class = (GstAudioSrcClass *) (klass);

   gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_openal_src_dispose);
   gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_openal_src_finalize);
   gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_openal_src_set_property);
   gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_openal_src_get_property);

   gst_openal_src_parent_class = g_type_class_peek_parent (klass);

   gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_openal_src_getcaps);

   gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_openal_src_open);
   gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_openal_src_prepare);
   gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_openal_src_unprepare);
   gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_openal_src_close);
   gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_openal_src_read);
   gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_openal_src_delay);
   gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_openal_src_reset);

   g_object_class_install_property (gobject_class, PROP_DEVICE,
       g_param_spec_string ("device", "ALCdevice",
           "User device, default device if NULL", OPENAL_DEFAULT_DEVICE,
           G_PARAM_READWRITE));

   g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
       g_param_spec_string ("device-name", "Device name",
           "Human-readable name of the device", OPENAL_DEFAULT_DEVICE_NAME,
           G_PARAM_READABLE));

   gst_element_class_set_static_metadata (gstelement_class,
       "OpenAL Audio Source", "Source/Audio", "Input audio through OpenAL",
       "Juan Manuel Borges Caño <juanmabcmail@gmail.com>");

   gst_element_class_add_static_pad_template (gstelement_class,
       &openalsrc_factory);
 }

 static void
 gst_openal_src_init (GstOpenalSrc * openalsrc)
 {
   GST_DEBUG_OBJECT (openalsrc, "initializing");

   openalsrc->default_device_name = g_strdup (OPENAL_DEFAULT_DEVICE_NAME);
   openalsrc->default_device = OPENAL_DEFAULT_DEVICE;
   openalsrc->device = NULL;

   openalsrc->buffer_length = 0;

   openalsrc->probed_caps = NULL;
 }

 static void
 gst_openal_src_finalize (GObject * object)
 {
   GstOpenalSrc *openalsrc = GST_OPENAL_SRC (object);

   g_free (openalsrc->default_device_name);
   g_free (openalsrc->default_device);

   G_OBJECT_CLASS (gst_openal_src_parent_class)->finalize (object);
 }

 static void
 gst_openal_src_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec)
 {
   GstOpenalSrc *openalsrc = GST_OPENAL_SRC (object);

   switch (prop_id) {
     case PROP_DEVICE:
       openalsrc->default_device = g_value_dup_string (value);
       break;
     case PROP_DEVICE_NAME:
       openalsrc->default_device_name = 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 *openalsrc = GST_OPENAL_SRC (object);

   switch (prop_id) {
     case PROP_DEVICE:
       g_value_set_string (value, openalsrc->default_device);
       break;
     case PROP_DEVICE_NAME:
       g_value_set_string (value, openalsrc->default_device_name);
       break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
   }
 }

 static GstCaps *
 gst_openal_helper_probe_caps (ALCcontext * context)
 {
   GstStructure *structure;
   GstCaps *caps;
 //  ALCcontext *old;

 //  old = pushContext(context);

   caps = gst_caps_new_empty ();

   if (alIsExtensionPresent ("AL_EXT_DOUBLE")) {
     structure =
         gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
         GST_AUDIO_NE (F64), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
         OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
     gst_caps_append_structure (caps, structure);
   }

   if (alIsExtensionPresent ("AL_EXT_FLOAT32")) {
     structure =
         gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
         GST_AUDIO_NE (F32), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
         OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
     gst_caps_append_structure (caps, structure);
   }

   structure =
       gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
       GST_AUDIO_NE (S16), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
       OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
   gst_caps_append_structure (caps, structure);

   structure =
       gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
       G_STRINGIFY (U8), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
       OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
   gst_caps_append_structure (caps, structure);

   if (alIsExtensionPresent ("AL_EXT_IMA4")) {
     structure =
         gst_structure_new ("audio/x-adpcm", "layout", G_TYPE_STRING, "ima",
         "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
         "channels", G_TYPE_INT, 1, NULL);
     gst_caps_append_structure (caps, structure);
   }

   if (alIsExtensionPresent ("AL_EXT_ALAW")) {
     structure =
         gst_structure_new ("audio/x-alaw", "rate", GST_TYPE_INT_RANGE,
         OPENAL_MIN_RATE, OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
     gst_caps_append_structure (caps, structure);
   }

   if (alIsExtensionPresent ("AL_EXT_MULAW")) {
     structure =
         gst_structure_new ("audio/x-mulaw", "rate", GST_TYPE_INT_RANGE,
         OPENAL_MIN_RATE, OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
     gst_caps_append_structure (caps, structure);
   }
 //  popContext(old, context);

   return caps;
 }

 static GstCaps *
 gst_openal_src_getcaps (GstBaseSrc * basesrc, GstCaps * filter)
 {
   GstOpenalSrc *openalsrc = GST_OPENAL_SRC (basesrc);
   GstCaps *caps;
   ALCdevice *device;

   device = alcOpenDevice (NULL);

   if (device == NULL) {
     GstPad *pad = GST_BASE_SRC_PAD (basesrc);
     GstCaps *tcaps = gst_pad_get_pad_template_caps (pad);

     GST_ELEMENT_WARNING (openalsrc, RESOURCE, OPEN_WRITE,
         ("Could not open temporary device."), GST_ALC_ERROR (device));
     caps = gst_caps_copy (tcaps);
     gst_caps_unref (tcaps);
   } else if (openalsrc->probed_caps)
     caps = gst_caps_copy (openalsrc->probed_caps);
   else {
     ALCcontext *context = alcCreateContext (device, NULL);
     if (context) {
       caps = gst_openal_helper_probe_caps (context);
       alcDestroyContext (context);
     } else {
       GST_ELEMENT_WARNING (openalsrc, RESOURCE, FAILED,
           ("Could not create temporary context."), GST_ALC_ERROR (device));
       caps = NULL;
     }

     if (caps && !gst_caps_is_empty (caps))
       openalsrc->probed_caps = gst_caps_copy (caps);
   }

   if (device != NULL) {
     if (alcCloseDevice (device) == ALC_FALSE) {
       GST_ELEMENT_WARNING (openalsrc, RESOURCE, CLOSE,
           ("Could not close temporary device."), GST_ALC_ERROR (device));
     }
   }

   if (filter) {
     GstCaps *intersection;

     intersection =
         gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
     return intersection;
   } else {
     return caps;
   }
 }


 static gboolean
 gst_openal_src_open (GstAudioSrc * audiosrc)
 {
   return TRUE;
 }

 static void
 gst_openal_src_parse_spec (GstOpenalSrc * openalsrc,
     const GstAudioRingBufferSpec * spec)
 {
   ALuint format = AL_NONE;

   GST_DEBUG_OBJECT (openalsrc,
       "looking up format for type %d, gst-format %d, and %d channels",
       spec->type, GST_AUDIO_INFO_FORMAT (&spec->info),
       GST_AUDIO_INFO_CHANNELS (&spec->info));

   switch (spec->type) {
     case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
       switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
         case GST_AUDIO_FORMAT_U8:
           switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
             case 1:
               format = AL_FORMAT_MONO8;
               break;
             default:
               break;
           }
           break;

         case GST_AUDIO_FORMAT_U16:
         case GST_AUDIO_FORMAT_S16:
           switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
             case 1:
               format = AL_FORMAT_MONO16;
               break;
             default:
               break;
           }
           break;

         case GST_AUDIO_FORMAT_F32:
           switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
             case 1:
               format = AL_FORMAT_MONO_FLOAT32;
               break;
             default:
               break;
           }
           break;

         case GST_AUDIO_FORMAT_F64:
           switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
             case 1:
               format = AL_FORMAT_MONO_DOUBLE_EXT;
               break;
             default:
               break;
           }
           break;

         default:
           break;
       }
       break;

     case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM:
       switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
         case 1:
           format = AL_FORMAT_MONO_IMA4;
           break;
         default:
           break;
       }
       break;

     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)
 {
 }
	/*
	* 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>
	* Copyright (C) 2013 Juan Manuel Borges Caño <juanmabcmail@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:
	*
	* Copyright (C) 2013 Juan Manuel Borges Caño <juanmabcmail@gmail.com>
	*
	* 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., 51 Franklin St, Fifth Floor,
	* Boston, MA 02110-1301, USA.
	*/

	/**
	* SECTION:element-openalsrc
	* @title: openalsrc
	* @see_also: openalsink
	* @short_description: capture raw audio samples through OpenAL
	*
	* This element captures raw audio samples through OpenAL.
	*
	* ## Example pipelines
	* \|[
	* gst-launch-1.0 -v openalsrc ! audioconvert ! wavenc ! filesink location=stream.wav
	* ]\| * will capture sound through OpenAL and encode it to a wav file.
	* \|[
	* gst-launch-1.0 openalsrc ! "audio/x-raw,format=S16LE,rate=44100" ! audioconvert ! volume volume=0.25 ! openalsink
	* ]\| will capture and play audio through OpenAL.
	*
	*/

	/*
	* DEV:
	* To get better timing/delay information you may also be interested in this:
	* http://kcat.strangesoft.net/openal-extensions/SOFT_source_latency.txt
	*/

	#ifdef HAVE_CONFIG_H
	#include <config.h>
	#endif

	#include <gst/gst.h>
	#include <gst/gsterror.h>

	GST_DEBUG_CATEGORY_EXTERN (openal_debug);
	#define GST_CAT_DEFAULT openal_debug

	#include "gstopenalsrc.h"

	static void gst_openal_src_dispose (GObject * object);
	static void gst_openal_src_finalize (GObject * object);
	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 GstCaps gst_openal_src_getcaps (GstBaseSrc basesrc, GstCaps * filter);
	static gboolean gst_openal_src_open (GstAudioSrc * audiosrc);
	static gboolean gst_openal_src_prepare (GstAudioSrc * audiosrc,
	GstAudioRingBufferSpec * spec);
	static gboolean gst_openal_src_unprepare (GstAudioSrc * audiosrc);
	static gboolean gst_openal_src_close (GstAudioSrc * audiosrc);
	static guint gst_openal_src_read (GstAudioSrc * audiosrc, gpointer data,
	guint length, GstClockTime * timestamp);
	static guint gst_openal_src_delay (GstAudioSrc * audiosrc);
	static void gst_openal_src_reset (GstAudioSrc * audiosrc);

	#define OPENAL_DEFAULT_DEVICE_NAME NULL
	#define OPENAL_DEFAULT_DEVICE NULL

	#define OPENAL_MIN_RATE 8000
	#define OPENAL_MAX_RATE 192000

	enum
	{
	PROP_0,
	PROP_DEVICE,
	PROP_DEVICE_NAME
	};

	static GstStaticPadTemplate openalsrc_factory = GST_STATIC_PAD_TEMPLATE ("src",
	GST_PAD_SRC,
	GST_PAD_ALWAYS,
	GST_STATIC_CAPS (
	/* These caps do not work on my card */
	// "audio/x-adpcm, " "layout = (string) ima, "
	// "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
	// "audio/x-alaw, " "rate = (int) [ 1, MAX ], "
	// "channels = (int) 1; "
	// "audio/x-mulaw, " "rate = (int) [ 1, MAX ], "
	// "channels = (int) 1; "
	// "audio/x-raw, " "format = (string) " GST_AUDIO_NE (F64) ", "
	// "rate = (int) [ 1, MAX ], " "channels = (int) 1; "
	// "audio/x-raw, " "format = (string) " GST_AUDIO_NE (F32) ", "
	// "rate = (int) [ 1, MAX ], " "channels = (int) 1; "
	"audio/x-raw, " "format = (string) " GST_AUDIO_NE (S16) ", "
	"rate = (int) [ 1, MAX ], " "channels = (int) 1; "
	/* These caps work wrongly on my card */
	// "audio/x-raw, " "format = (string) " GST_AUDIO_NE (U16) ", "
	// "rate = (int) [ 1, MAX ], " "channels = (int) 1; "
	// "audio/x-raw, " "format = (string) " G_STRINGIFY (S8) ", "
	// "rate = (int) [ 1, MAX ], " "channels = (int) 1"));
	"audio/x-raw, " "format = (string) " G_STRINGIFY (U8) ", "
	"rate = (int) [ 1, MAX ], " "channels = (int) 1")
	);

	G_DEFINE_TYPE (GstOpenalSrc, gst_openal_src, GST_TYPE_AUDIO_SRC);

	static void
	gst_openal_src_dispose (GObject * object)
	{
	GstOpenalSrc *openalsrc = GST_OPENAL_SRC (object);

	if (openalsrc->probed_caps)
	gst_caps_unref (openalsrc->probed_caps);
	openalsrc->probed_caps = NULL;

	G_OBJECT_CLASS (gst_openal_src_parent_class)->dispose (object);
	}

	static void
	gst_openal_src_class_init (GstOpenalSrcClass * klass)
	{
	GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
	GstElementClass gstelement_class = (GstElementClass ) klass;
	GstBaseSrcClass gstbasesrc_class = (GstBaseSrcClass ) klass;
	GstAudioSrcClass gstaudiosrc_class = (GstAudioSrcClass ) (klass);

	gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_openal_src_dispose);
	gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_openal_src_finalize);
	gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_openal_src_set_property);
	gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_openal_src_get_property);

	gst_openal_src_parent_class = g_type_class_peek_parent (klass);

	gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_openal_src_getcaps);

	gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_openal_src_open);
	gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_openal_src_prepare);
	gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_openal_src_unprepare);
	gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_openal_src_close);
	gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_openal_src_read);
	gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_openal_src_delay);
	gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_openal_src_reset);

	g_object_class_install_property (gobject_class, PROP_DEVICE,
	g_param_spec_string ("device", "ALCdevice",
	"User device, default device if NULL", OPENAL_DEFAULT_DEVICE,
	G_PARAM_READWRITE));

	g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
	g_param_spec_string ("device-name", "Device name",
	"Human-readable name of the device", OPENAL_DEFAULT_DEVICE_NAME,
	G_PARAM_READABLE));

	gst_element_class_set_static_metadata (gstelement_class,
	"OpenAL Audio Source", "Source/Audio", "Input audio through OpenAL",
	"Juan Manuel Borges Caño <juanmabcmail@gmail.com>");

	gst_element_class_add_static_pad_template (gstelement_class,
	&openalsrc_factory);
	}

	static void
	gst_openal_src_init (GstOpenalSrc * openalsrc)
	{
	GST_DEBUG_OBJECT (openalsrc, "initializing");

	openalsrc->default_device_name = g_strdup (OPENAL_DEFAULT_DEVICE_NAME);
	openalsrc->default_device = OPENAL_DEFAULT_DEVICE;
	openalsrc->device = NULL;

	openalsrc->buffer_length = 0;

	openalsrc->probed_caps = NULL;
	}

	static void
	gst_openal_src_finalize (GObject * object)
	{
	GstOpenalSrc *openalsrc = GST_OPENAL_SRC (object);

	g_free (openalsrc->default_device_name);
	g_free (openalsrc->default_device);

	G_OBJECT_CLASS (gst_openal_src_parent_class)->finalize (object);
	}

	static void
	gst_openal_src_set_property (GObject * object, guint prop_id,
	const GValue * value, GParamSpec * pspec)
	{
	GstOpenalSrc *openalsrc = GST_OPENAL_SRC (object);

	switch (prop_id) {
	case PROP_DEVICE:
	openalsrc->default_device = g_value_dup_string (value);
	break;
	case PROP_DEVICE_NAME:
	openalsrc->default_device_name = 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 *openalsrc = GST_OPENAL_SRC (object);

	switch (prop_id) {
	case PROP_DEVICE:
	g_value_set_string (value, openalsrc->default_device);
	break;
	case PROP_DEVICE_NAME:
	g_value_set_string (value, openalsrc->default_device_name);
	break;
	default:
	G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
	break;
	}
	}

	static GstCaps *
	gst_openal_helper_probe_caps (ALCcontext * context)
	{
	GstStructure *structure;
	GstCaps *caps;
	// ALCcontext *old;

	// old = pushContext(context);

	caps = gst_caps_new_empty ();

	if (alIsExtensionPresent ("AL_EXT_DOUBLE")) {
	structure =
	gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
	GST_AUDIO_NE (F64), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
	OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
	gst_caps_append_structure (caps, structure);
	}

	if (alIsExtensionPresent ("AL_EXT_FLOAT32")) {
	structure =
	gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
	GST_AUDIO_NE (F32), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
	OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
	gst_caps_append_structure (caps, structure);
	}

	structure =
	gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
	GST_AUDIO_NE (S16), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
	OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
	gst_caps_append_structure (caps, structure);

	structure =
	gst_structure_new ("audio/x-raw", "format", G_TYPE_STRING,
	G_STRINGIFY (U8), "rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
	OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
	gst_caps_append_structure (caps, structure);

	if (alIsExtensionPresent ("AL_EXT_IMA4")) {
	structure =
	gst_structure_new ("audio/x-adpcm", "layout", G_TYPE_STRING, "ima",
	"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
	"channels", G_TYPE_INT, 1, NULL);
	gst_caps_append_structure (caps, structure);
	}

	if (alIsExtensionPresent ("AL_EXT_ALAW")) {
	structure =
	gst_structure_new ("audio/x-alaw", "rate", GST_TYPE_INT_RANGE,
	OPENAL_MIN_RATE, OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
	gst_caps_append_structure (caps, structure);
	}

	if (alIsExtensionPresent ("AL_EXT_MULAW")) {
	structure =
	gst_structure_new ("audio/x-mulaw", "rate", GST_TYPE_INT_RANGE,
	OPENAL_MIN_RATE, OPENAL_MAX_RATE, "channels", G_TYPE_INT, 1, NULL);
	gst_caps_append_structure (caps, structure);
	}
	// popContext(old, context);

	return caps;
	}

	static GstCaps *
	gst_openal_src_getcaps (GstBaseSrc * basesrc, GstCaps * filter)
	{
	GstOpenalSrc *openalsrc = GST_OPENAL_SRC (basesrc);
	GstCaps *caps;
	ALCdevice *device;

	device = alcOpenDevice (NULL);

	if (device == NULL) {
	GstPad *pad = GST_BASE_SRC_PAD (basesrc);
	GstCaps *tcaps = gst_pad_get_pad_template_caps (pad);

	GST_ELEMENT_WARNING (openalsrc, RESOURCE, OPEN_WRITE,
	("Could not open temporary device."), GST_ALC_ERROR (device));
	caps = gst_caps_copy (tcaps);
	gst_caps_unref (tcaps);
	} else if (openalsrc->probed_caps)
	caps = gst_caps_copy (openalsrc->probed_caps);
	else {
	ALCcontext *context = alcCreateContext (device, NULL);
	if (context) {
	caps = gst_openal_helper_probe_caps (context);
	alcDestroyContext (context);
	} else {
	GST_ELEMENT_WARNING (openalsrc, RESOURCE, FAILED,
	("Could not create temporary context."), GST_ALC_ERROR (device));
	caps = NULL;
	}

	if (caps && !gst_caps_is_empty (caps))
	openalsrc->probed_caps = gst_caps_copy (caps);
	}

	if (device != NULL) {
	if (alcCloseDevice (device) == ALC_FALSE) {
	GST_ELEMENT_WARNING (openalsrc, RESOURCE, CLOSE,
	("Could not close temporary device."), GST_ALC_ERROR (device));
	}
	}

	if (filter) {
	GstCaps *intersection;

	intersection =
	gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
	return intersection;
	} else {
	return caps;
	}
	}


	static gboolean
	gst_openal_src_open (GstAudioSrc * audiosrc)
	{
	return TRUE;
	}

	static void
	gst_openal_src_parse_spec (GstOpenalSrc * openalsrc,
	const GstAudioRingBufferSpec * spec)
	{
	ALuint format = AL_NONE;

	GST_DEBUG_OBJECT (openalsrc,
	"looking up format for type %d, gst-format %d, and %d channels",
	spec->type, GST_AUDIO_INFO_FORMAT (&spec->info),
	GST_AUDIO_INFO_CHANNELS (&spec->info));

	switch (spec->type) {
	case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
	switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
	case GST_AUDIO_FORMAT_U8:
	switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
	case 1:
	format = AL_FORMAT_MONO8;
	break;
	default:
	break;
	}
	break;

	case GST_AUDIO_FORMAT_U16:
	case GST_AUDIO_FORMAT_S16:
	switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
	case 1:
	format = AL_FORMAT_MONO16;
	break;
	default:
	break;
	}
	break;

	case GST_AUDIO_FORMAT_F32:
	switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
	case 1:
	format = AL_FORMAT_MONO_FLOAT32;
	break;
	default:
	break;
	}
	break;

	case GST_AUDIO_FORMAT_F64:
	switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
	case 1:
	format = AL_FORMAT_MONO_DOUBLE_EXT;
	break;
	default:
	break;
	}
	break;

	default:
	break;
	}
	break;

	case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM:
	switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
	case 1:
	format = AL_FORMAT_MONO_IMA4;
	break;
	default:
	break;
	}
	break;

	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)
	{
	}