blob: d06b3c440ccbf00e6666e0f345677a28213c444d [file] [log] [blame]
/*
* 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
* @see_also: openalsink
* @short_description: capture raw audio samples through OpenAL
*
* This element captures raw audio samples through OpenAL.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* 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.
* </refsect2>
*/
/*
* 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_pad_template (gstelement_class,
gst_static_pad_template_get (&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)
{
}