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gstreamer/ext/openal/gstopenalsink.c

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/*
* GStreamer
* Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
* Copyright (C) 2009-2010 Chris Robinson <chris.kcat@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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-openalsink
*
* This element renders raw audio samples using the OpenAL API
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.1 ! openalsink
* ]| will output a sine wave (continuous beep sound) to your sound card (with
* a very low volume as precaution).
* |[
* gst-launch -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! openalsink
* ]| will play an Ogg/Vorbis audio file and output it using OpenAL.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstopenalsink.h"
GST_DEBUG_CATEGORY (openalsink_debug);
static void gst_openal_sink_dispose (GObject * object);
static void gst_openal_sink_finalize (GObject * object);
static void gst_openal_sink_get_property (GObject * object, guint prop_id,
GValue * val, GParamSpec * pspec);
static void gst_openal_sink_set_property (GObject * object, guint prop_id,
const GValue * val, GParamSpec * pspec);
static GstCaps *gst_openal_sink_getcaps (GstBaseSink * bsink);
static gboolean gst_openal_sink_open (GstAudioSink * asink);
static gboolean gst_openal_sink_close (GstAudioSink * asink);
static gboolean gst_openal_sink_prepare (GstAudioSink * asink,
GstRingBufferSpec * spec);
static gboolean gst_openal_sink_unprepare (GstAudioSink * asink);
static guint gst_openal_sink_write (GstAudioSink * asink, gpointer data,
guint length);
static guint gst_openal_sink_delay (GstAudioSink * asink);
static void gst_openal_sink_reset (GstAudioSink * asink);
#define DEFAULT_DEVICE NULL
enum
{
PROP_0,
PROP_DEVICE,
PROP_DEVICE_NAME,
PROP_DEVICE_HDL,
PROP_CONTEXT_HDL,
PROP_SOURCE_ID
};
static GstStaticPadTemplate openalsink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
"width = (int) 32, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ]; "
"audio/x-raw-int, "
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
"signed = (boolean) TRUE, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ]; "
"audio/x-raw-int, "
"signed = (boolean) FALSE, "
"width = (int) 8, "
"depth = (int) 8, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ]; "
"audio/x-mulaw, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
static PFNALCSETTHREADCONTEXTPROC palcSetThreadContext;
static PFNALCGETTHREADCONTEXTPROC palcGetThreadContext;
static inline ALCcontext *
pushContext (ALCcontext * ctx)
{
ALCcontext *old;
if (!palcGetThreadContext || !palcSetThreadContext)
return NULL;
old = palcGetThreadContext ();
if (old != ctx)
palcSetThreadContext (ctx);
return old;
}
static inline void
popContext (ALCcontext * old, ALCcontext * ctx)
{
if (!palcGetThreadContext || !palcSetThreadContext)
return;
if (old != ctx)
palcSetThreadContext (old);
}
static inline ALenum
checkALError (const char *fname, unsigned int fline)
{
ALenum err = alGetError ();
if (err != AL_NO_ERROR)
g_warning ("%s:%u: context error: %s", fname, fline, alGetString (err));
return err;
}
#define checkALError() checkALError(__FILE__, __LINE__)
GST_BOILERPLATE (GstOpenALSink, gst_openal_sink, GstAudioSink,
GST_TYPE_AUDIO_SINK);
static void
gst_openal_sink_dispose (GObject * object)
{
GstOpenALSink *sink = GST_OPENAL_SINK (object);
if (sink->probed_caps)
gst_caps_unref (sink->probed_caps);
sink->probed_caps = NULL;
G_OBJECT_CLASS (parent_class)->dispose (object);
}
/* GObject vmethod implementations */
static void
gst_openal_sink_base_init (gpointer gclass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
gst_element_class_set_details_simple (element_class, "Audio sink (OpenAL)",
"Sink/Audio",
"Output to a sound device via OpenAL",
"Chris Robinson <chris.kcat@gmail.com>");
gst_element_class_add_static_pad_template (element_class,
&openalsink_sink_factory);
}
/* initialize the plugin's class */
static void
gst_openal_sink_class_init (GstOpenALSinkClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstBaseSinkClass *gstbasesink_class = (GstBaseSinkClass *) klass;
GstAudioSinkClass *gstaudiosink_class = (GstAudioSinkClass *) klass;
GParamSpec *spec;
if (alcIsExtensionPresent (NULL, "ALC_EXT_thread_local_context")) {
palcSetThreadContext = alcGetProcAddress (NULL, "alcSetThreadContext");
palcGetThreadContext = alcGetProcAddress (NULL, "alcGetThreadContext");
}
GST_DEBUG_CATEGORY_INIT (openalsink_debug, "openalsink", 0, "OpenAL sink");
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_openal_sink_dispose);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_openal_sink_finalize);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_openal_sink_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_openal_sink_get_property);
spec = g_param_spec_string ("device-name", "Device name",
"Opened OpenAL device name", "", G_PARAM_READABLE);
g_object_class_install_property (gobject_class, PROP_DEVICE_NAME, spec);
spec = g_param_spec_string ("device", "Device", "OpenAL device string",
DEFAULT_DEVICE, G_PARAM_READWRITE);
g_object_class_install_property (gobject_class, PROP_DEVICE, spec);
spec = g_param_spec_pointer ("device-handle", "ALCdevice",
"Custom playback device", G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
g_object_class_install_property (gobject_class, PROP_DEVICE_HDL, spec);
spec = g_param_spec_pointer ("context-handle", "ALCcontext",
"Custom playback context", G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
g_object_class_install_property (gobject_class, PROP_CONTEXT_HDL, spec);
spec = g_param_spec_uint ("source-id", "Source ID", "Custom playback sID",
0, UINT_MAX, 0, G_PARAM_READWRITE);
g_object_class_install_property (gobject_class, PROP_SOURCE_ID, spec);
parent_class = g_type_class_peek_parent (klass);
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_openal_sink_getcaps);
gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_openal_sink_open);
gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_openal_sink_close);
gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_openal_sink_prepare);
gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_openal_sink_unprepare);
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_openal_sink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_openal_sink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_openal_sink_reset);
}
static void
gst_openal_sink_init (GstOpenALSink * sink, GstOpenALSinkClass * klass)
{
GST_DEBUG_OBJECT (sink, "initializing openalsink");
sink->devname = g_strdup (DEFAULT_DEVICE);
sink->custom_dev = NULL;
sink->custom_ctx = NULL;
sink->custom_sID = 0;
sink->device = NULL;
sink->context = NULL;
sink->sID = 0;
sink->bID_idx = 0;
sink->bID_count = 0;
sink->bIDs = NULL;
sink->bID_length = 0;
sink->write_reset = AL_FALSE;
sink->probed_caps = NULL;
sink->openal_lock = g_mutex_new ();
}
static void
gst_openal_sink_finalize (GObject * object)
{
GstOpenALSink *sink = GST_OPENAL_SINK (object);
g_free (sink->devname);
sink->devname = NULL;
g_mutex_free (sink->openal_lock);
sink->openal_lock = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_openal_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOpenALSink *sink = GST_OPENAL_SINK (object);
switch (prop_id) {
case PROP_DEVICE:
g_free (sink->devname);
sink->devname = g_value_dup_string (value);
if (sink->probed_caps)
gst_caps_unref (sink->probed_caps);
sink->probed_caps = NULL;
break;
case PROP_DEVICE_HDL:
if (!sink->device)
sink->custom_dev = g_value_get_pointer (value);
break;
case PROP_CONTEXT_HDL:
if (!sink->device)
sink->custom_ctx = g_value_get_pointer (value);
break;
case PROP_SOURCE_ID:
if (!sink->device)
sink->custom_sID = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_openal_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstOpenALSink *sink = GST_OPENAL_SINK (object);
const ALCchar *name = sink->devname;
ALCdevice *device = sink->device;
ALCcontext *context = sink->context;
ALuint sourceID = sink->sID;
switch (prop_id) {
case PROP_DEVICE_NAME:
name = "";
if (device)
name = alcGetString (device, ALC_DEVICE_SPECIFIER);
/* fall-through */
case PROP_DEVICE:
g_value_set_string (value, name);
break;
case PROP_DEVICE_HDL:
if (!device)
device = sink->custom_dev;
g_value_set_pointer (value, device);
break;
case PROP_CONTEXT_HDL:
if (!context)
context = sink->custom_ctx;
g_value_set_pointer (value, context);
break;
case PROP_SOURCE_ID:
if (!sourceID)
sourceID = sink->custom_sID;
g_value_set_uint (value, sourceID);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_openal_helper_probe_caps (ALCcontext * ctx)
{
static const struct
{
gint count;
GstAudioChannelPosition pos[8];
} chans[] = {
{
1, {
GST_AUDIO_CHANNEL_POSITION_FRONT_MONO}}, {
2, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}}, {
4, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}}, {
6, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}}, {
7, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}}, {
8, {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}},};
GstStructure *structure;
ALCcontext *old;
GstCaps *caps;
old = pushContext (ctx);
caps = gst_caps_new_empty ();
if (alIsExtensionPresent ("AL_EXT_MCFORMATS")) {
const char *fmt32[] = {
"AL_FORMAT_MONO_FLOAT32", "AL_FORMAT_STEREO_FLOAT32",
"AL_FORMAT_QUAD32", "AL_FORMAT_51CHN32", "AL_FORMAT_61CHN32",
"AL_FORMAT_71CHN32", NULL
}, *fmt16[] = {
"AL_FORMAT_MONO16", "AL_FORMAT_STEREO16", "AL_FORMAT_QUAD16",
"AL_FORMAT_51CHN16", "AL_FORMAT_61CHN16", "AL_FORMAT_71CHN16", NULL},
*fmt8[] = {
"AL_FORMAT_MONO8", "AL_FORMAT_STEREO8", "AL_FORMAT_QUAD8",
"AL_FORMAT_51CHN8", "AL_FORMAT_61CHN8", "AL_FORMAT_71CHN8", NULL};
int i;
if (alIsExtensionPresent ("AL_EXT_FLOAT32")) {
for (i = 0; fmt32[i]; i++) {
ALenum val = alGetEnumValue (fmt32[i]);
if (checkALError () != AL_NO_ERROR || val == 0 || val == -1)
continue;
structure = gst_structure_new ("audio/x-raw-float",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE,
OPENAL_MAX_RATE, "width", G_TYPE_INT, 32, NULL);
gst_structure_set (structure, "channels", G_TYPE_INT,
chans[i].count, NULL);
if (chans[i].count > 2)
gst_audio_set_channel_positions (structure, chans[i].pos);
gst_caps_append_structure (caps, structure);
}
}
for (i = 0; fmt16[i]; i++) {
ALenum val = alGetEnumValue (fmt16[i]);
if (checkALError () != AL_NO_ERROR || val == 0 || val == -1)
continue;
structure = gst_structure_new ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
gst_structure_set (structure, "channels", G_TYPE_INT,
chans[i].count, NULL);
if (chans[i].count > 2)
gst_audio_set_channel_positions (structure, chans[i].pos);
gst_caps_append_structure (caps, structure);
}
for (i = 0; fmt8[i]; i++) {
ALenum val = alGetEnumValue (fmt8[i]);
if (checkALError () != AL_NO_ERROR || val == 0 || val == -1)
continue;
structure = gst_structure_new ("audio/x-raw-int",
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
"width", G_TYPE_INT, 8,
"depth", G_TYPE_INT, 8, "signed", G_TYPE_BOOLEAN, FALSE, NULL);
gst_structure_set (structure, "channels", G_TYPE_INT,
chans[i].count, NULL);
if (chans[i].count > 2)
gst_audio_set_channel_positions (structure, chans[i].pos);
gst_caps_append_structure (caps, structure);
}
} else {
if (alIsExtensionPresent ("AL_EXT_FLOAT32")) {
structure = gst_structure_new ("audio/x-raw-float",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
"width", G_TYPE_INT, 32, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
gst_caps_append_structure (caps, structure);
}
structure = gst_structure_new ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"signed", G_TYPE_BOOLEAN, TRUE,
"channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
gst_caps_append_structure (caps, structure);
structure = gst_structure_new ("audio/x-raw-int",
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
"width", G_TYPE_INT, 8,
"depth", G_TYPE_INT, 8,
"signed", G_TYPE_BOOLEAN, FALSE,
"channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
gst_caps_append_structure (caps, structure);
}
if (alIsExtensionPresent ("AL_EXT_MULAW_MCFORMATS")) {
const char *fmtmulaw[] = {
"AL_FORMAT_MONO_MULAW", "AL_FORMAT_STEREO_MULAW",
"AL_FORMAT_QUAD_MULAW", "AL_FORMAT_51CHN_MULAW",
"AL_FORMAT_61CHN_MULAW", "AL_FORMAT_71CHN_MULAW", NULL
};
int i;
for (i = 0; fmtmulaw[i]; i++) {
ALenum val = alGetEnumValue (fmtmulaw[i]);
if (checkALError () != AL_NO_ERROR || val == 0 || val == -1)
continue;
structure = gst_structure_new ("audio/x-mulaw",
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE, NULL);
gst_structure_set (structure, "channels", G_TYPE_INT,
chans[i].count, NULL);
if (chans[i].count > 2)
gst_audio_set_channel_positions (structure, chans[i].pos);
gst_caps_append_structure (caps, structure);
}
} else if (alIsExtensionPresent ("AL_EXT_MULAW")) {
structure = gst_structure_new ("audio/x-mulaw",
"rate", GST_TYPE_INT_RANGE, OPENAL_MIN_RATE, OPENAL_MAX_RATE,
"channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
gst_caps_append_structure (caps, structure);
}
popContext (old, ctx);
return caps;
}
static GstCaps *
gst_openal_sink_getcaps (GstBaseSink * bsink)
{
GstOpenALSink *sink = GST_OPENAL_SINK (bsink);
GstCaps *caps;
if (sink->device == NULL) {
GstPad *pad = GST_BASE_SINK_PAD (bsink);
caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
} else if (sink->probed_caps)
caps = gst_caps_copy (sink->probed_caps);
else {
if (sink->context)
caps = gst_openal_helper_probe_caps (sink->context);
else if (sink->custom_ctx)
caps = gst_openal_helper_probe_caps (sink->custom_ctx);
else {
ALCcontext *ctx = alcCreateContext (sink->device, NULL);
if (ctx) {
caps = gst_openal_helper_probe_caps (ctx);
alcDestroyContext (ctx);
} else {
GST_ELEMENT_WARNING (sink, RESOURCE, FAILED,
("Could not create temporary context."),
GST_ALC_ERROR (sink->device));
caps = NULL;
}
}
if (caps && !gst_caps_is_empty (caps))
sink->probed_caps = gst_caps_copy (caps);
}
return caps;
}
static gboolean
gst_openal_sink_open (GstAudioSink * asink)
{
GstOpenALSink *openal = GST_OPENAL_SINK (asink);
if (openal->custom_dev) {
ALCint val = -1;
alcGetIntegerv (openal->custom_dev, ALC_ATTRIBUTES_SIZE, 1, &val);
if (val > 0) {
if (!openal->custom_ctx ||
alcGetContextsDevice (openal->custom_ctx) == openal->custom_dev)
openal->device = openal->custom_dev;
}
} else if (openal->custom_ctx)
openal->device = alcGetContextsDevice (openal->custom_ctx);
else
openal->device = alcOpenDevice (openal->devname);
if (!openal->device) {
GST_ELEMENT_ERROR (openal, RESOURCE, OPEN_WRITE,
("Could not open audio device for playback."),
GST_ALC_ERROR (openal->device));
return FALSE;
}
return TRUE;
}
static gboolean
gst_openal_sink_close (GstAudioSink * asink)
{
GstOpenALSink *openal = GST_OPENAL_SINK (asink);
if (!openal->custom_dev && !openal->custom_ctx) {
if (alcCloseDevice (openal->device) == ALC_FALSE) {
GST_ELEMENT_ERROR (openal, RESOURCE, CLOSE,
("Could not close audio device."), GST_ALC_ERROR (openal->device));
return FALSE;
}
}
openal->device = NULL;
if (openal->probed_caps)
gst_caps_unref (openal->probed_caps);
openal->probed_caps = NULL;
return TRUE;
}
static void
gst_openal_sink_parse_spec (GstOpenALSink * openal,
const GstRingBufferSpec * spec)
{
ALuint format = AL_NONE;
GST_DEBUG_OBJECT (openal, "Looking up format for type %d, gst-format %d, "
"and %d channels", spec->type, spec->format, spec->channels);
/* Don't need to verify supported formats, since the probed caps will only
* report what was detected and we shouldn't get anything different */
switch (spec->type) {
case GST_BUFTYPE_LINEAR:
switch (spec->format) {
case GST_U8:
if (spec->channels == 1)
format = AL_FORMAT_MONO8;
if (spec->channels == 2)
format = AL_FORMAT_STEREO8;
if (spec->channels == 4)
format = AL_FORMAT_QUAD8;
if (spec->channels == 6)
format = AL_FORMAT_51CHN8;
if (spec->channels == 7)
format = AL_FORMAT_61CHN8;
if (spec->channels == 8)
format = AL_FORMAT_71CHN8;
break;
case GST_S16_NE:
if (spec->channels == 1)
format = AL_FORMAT_MONO16;
if (spec->channels == 2)
format = AL_FORMAT_STEREO16;
if (spec->channels == 4)
format = AL_FORMAT_QUAD16;
if (spec->channels == 6)
format = AL_FORMAT_51CHN16;
if (spec->channels == 7)
format = AL_FORMAT_61CHN16;
if (spec->channels == 8)
format = AL_FORMAT_71CHN16;
break;
default:
break;
}
break;
case GST_BUFTYPE_FLOAT:
switch (spec->format) {
case GST_FLOAT32_NE:
if (spec->channels == 1)
format = AL_FORMAT_MONO_FLOAT32;
if (spec->channels == 2)
format = AL_FORMAT_STEREO_FLOAT32;
if (spec->channels == 4)
format = AL_FORMAT_QUAD32;
if (spec->channels == 6)
format = AL_FORMAT_51CHN32;
if (spec->channels == 7)
format = AL_FORMAT_61CHN32;
if (spec->channels == 8)
format = AL_FORMAT_71CHN32;
break;
default:
break;
}
break;
case GST_BUFTYPE_MU_LAW:
switch (spec->format) {
case GST_MU_LAW:
if (spec->channels == 1)
format = AL_FORMAT_MONO_MULAW;
if (spec->channels == 2)
format = AL_FORMAT_STEREO_MULAW;
if (spec->channels == 4)
format = AL_FORMAT_QUAD_MULAW;
if (spec->channels == 6)
format = AL_FORMAT_51CHN_MULAW;
if (spec->channels == 7)
format = AL_FORMAT_61CHN_MULAW;
if (spec->channels == 8)
format = AL_FORMAT_71CHN_MULAW;
break;
default:
break;
}
break;
default:
break;
}
openal->bytes_per_sample = spec->bytes_per_sample;
openal->srate = spec->rate;
openal->bID_count = spec->segtotal;
openal->bID_length = spec->segsize;
openal->format = format;
}
static gboolean
gst_openal_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
{
GstOpenALSink *openal = GST_OPENAL_SINK (asink);
ALCcontext *ctx, *old;
if (openal->context && !gst_openal_sink_unprepare (asink))
return FALSE;
if (openal->custom_ctx)
ctx = openal->custom_ctx;
else {
ALCint attribs[3] = { 0, 0, 0 };
/* Don't try to change the playback frequency of an app's device */
if (!openal->custom_dev) {
attribs[0] = ALC_FREQUENCY;
attribs[1] = spec->rate;
attribs[2] = 0;
}
ctx = alcCreateContext (openal->device, attribs);
if (!ctx) {
GST_ELEMENT_ERROR (openal, RESOURCE, FAILED,
("Unable to prepare device."), GST_ALC_ERROR (openal->device));
return FALSE;
}
}
old = pushContext (ctx);
if (openal->custom_sID) {
if (!openal->custom_ctx || !alIsSource (openal->custom_sID)) {
GST_ELEMENT_ERROR (openal, RESOURCE, NOT_FOUND, (NULL),
("Invalid source ID specified for context"));
goto fail;
}
openal->sID = openal->custom_sID;
} else {
ALuint sourceID;
alGenSources (1, &sourceID);
if (checkALError () != AL_NO_ERROR) {
GST_ELEMENT_ERROR (openal, RESOURCE, NO_SPACE_LEFT, (NULL),
("Unable to generate source"));
goto fail;
}
openal->sID = sourceID;
}
gst_openal_sink_parse_spec (openal, spec);
if (openal->format == AL_NONE) {
GST_ELEMENT_ERROR (openal, RESOURCE, SETTINGS, (NULL),
("Unable to get type %d, format %d, and %d channels",
spec->type, spec->format, spec->channels));
goto fail;
}
openal->bIDs = g_malloc (openal->bID_count * sizeof (*openal->bIDs));
if (!openal->bIDs) {
GST_ELEMENT_ERROR (openal, RESOURCE, FAILED, ("Out of memory."),
("Unable to allocate buffer IDs"));
goto fail;
}
alGenBuffers (openal->bID_count, openal->bIDs);
if (checkALError () != AL_NO_ERROR) {
GST_ELEMENT_ERROR (openal, RESOURCE, NO_SPACE_LEFT, (NULL),
("Unable to generate %d buffers", openal->bID_count));
goto fail;
}
openal->bID_idx = 0;
popContext (old, ctx);
openal->context = ctx;
return TRUE;
fail:
if (!openal->custom_sID && openal->sID)
alDeleteSources (1, &openal->sID);
openal->sID = 0;
g_free (openal->bIDs);
openal->bIDs = NULL;
openal->bID_count = 0;
openal->bID_length = 0;
popContext (old, ctx);
if (!openal->custom_ctx)
alcDestroyContext (ctx);
return FALSE;
}
static gboolean
gst_openal_sink_unprepare (GstAudioSink * asink)
{
GstOpenALSink *openal = GST_OPENAL_SINK (asink);
ALCcontext *old;
if (!openal->context)
return TRUE;
old = pushContext (openal->context);
alSourceStop (openal->sID);
alSourcei (openal->sID, AL_BUFFER, 0);
if (!openal->custom_sID)
alDeleteSources (1, &openal->sID);
openal->sID = 0;
alDeleteBuffers (openal->bID_count, openal->bIDs);
g_free (openal->bIDs);
openal->bIDs = NULL;
openal->bID_idx = 0;
openal->bID_count = 0;
openal->bID_length = 0;
checkALError ();
popContext (old, openal->context);
if (!openal->custom_ctx)
alcDestroyContext (openal->context);
openal->context = NULL;
return TRUE;
}
static guint
gst_openal_sink_write (GstAudioSink * asink, gpointer data, guint length)
{
GstOpenALSink *openal = GST_OPENAL_SINK (asink);
ALint processed, queued, state;
ALCcontext *old;
gulong rest_us;
g_assert (length == openal->bID_length);
old = pushContext (openal->context);
rest_us = (guint64) (openal->bID_length / openal->bytes_per_sample) *
G_USEC_PER_SEC / openal->srate / 2;
do {
alGetSourcei (openal->sID, AL_SOURCE_STATE, &state);
alGetSourcei (openal->sID, AL_BUFFERS_QUEUED, &queued);
alGetSourcei (openal->sID, AL_BUFFERS_PROCESSED, &processed);
if (checkALError () != AL_NO_ERROR) {
GST_ELEMENT_ERROR (openal, RESOURCE, WRITE, (NULL),
("Source state error detected"));
length = 0;
goto out_nolock;
}
if (processed > 0 || queued < openal->bID_count)
break;
if (state != AL_PLAYING)
alSourcePlay (openal->sID);
g_usleep (rest_us);
} while (1);
GST_OPENAL_SINK_LOCK (openal);
if (openal->write_reset != AL_FALSE) {
openal->write_reset = AL_FALSE;
length = 0;
goto out;
}
queued -= processed;
while (processed-- > 0) {
ALuint bid;
alSourceUnqueueBuffers (openal->sID, 1, &bid);
}
if (state == AL_STOPPED) {
/* "Restore" from underruns (not actually needed, but it keeps delay
* calculations correct while rebuffering) */
alSourceRewind (openal->sID);
}
alBufferData (openal->bIDs[openal->bID_idx], openal->format,
data, openal->bID_length, openal->srate);
alSourceQueueBuffers (openal->sID, 1, &openal->bIDs[openal->bID_idx]);
openal->bID_idx = (openal->bID_idx + 1) % openal->bID_count;
queued++;
if (state != AL_PLAYING && queued == openal->bID_count)
alSourcePlay (openal->sID);
if (checkALError () != ALC_NO_ERROR) {
GST_ELEMENT_ERROR (openal, RESOURCE, WRITE, (NULL),
("Source queue error detected"));
goto out;
}
out:
GST_OPENAL_SINK_UNLOCK (openal);
out_nolock:
popContext (old, openal->context);
return length;
}
static guint
gst_openal_sink_delay (GstAudioSink * asink)
{
GstOpenALSink *openal = GST_OPENAL_SINK (asink);
ALint queued, state, offset, delay;
ALCcontext *old;
if (!openal->context)
return 0;
GST_OPENAL_SINK_LOCK (openal);
old = pushContext (openal->context);
delay = 0;
alGetSourcei (openal->sID, AL_BUFFERS_QUEUED, &queued);
/* Order here is important. If the offset is queried after the state and an
* underrun occurs in between the two calls, it can end up with a 0 offset
* in a playing state, incorrectly reporting a len*queued/bps delay. */
alGetSourcei (openal->sID, AL_BYTE_OFFSET, &offset);
alGetSourcei (openal->sID, AL_SOURCE_STATE, &state);
/* Note: state=stopped is an underrun, meaning all buffers are processed
* and there's no delay when writing the next buffer. Pre-buffering is
* state=initial, which will introduce a delay while writing. */
if (checkALError () == AL_NO_ERROR && state != AL_STOPPED)
delay = ((queued * openal->bID_length) - offset) / openal->bytes_per_sample;
popContext (old, openal->context);
GST_OPENAL_SINK_UNLOCK (openal);
return delay;
}
static void
gst_openal_sink_reset (GstAudioSink * asink)
{
GstOpenALSink *openal = GST_OPENAL_SINK (asink);
ALCcontext *old;
GST_OPENAL_SINK_LOCK (openal);
old = pushContext (openal->context);
openal->write_reset = AL_TRUE;
alSourceStop (openal->sID);
alSourceRewind (openal->sID);
alSourcei (openal->sID, AL_BUFFER, 0);
checkALError ();
popContext (old, openal->context);
GST_OPENAL_SINK_UNLOCK (openal);
}
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