gstreamer/sys/magicleap/mlaudiosink.c

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2019-04-09 19:22:19 +00:00
/*
* Copyright (C) 2019 Collabora Ltd.
* Author: Xavier Claessens <xavier.claessens@collabora.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation
* version 2.1 of the License.
*
* 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
/**
* SECTION:mlaudiosink
* @short_description: Audio sink for Magic Leap platform
* @see_also: #GstAudioSink
*
* An audio sink element for LuminOS, the Magic Leap platform. There are 2 modes
* supported: normal and spatial. By default the audio is output directly to the
* stereo speakers, but in spatial mode the audio will be localised in the 3D
* environment. The user ears the sound as coming from a point in space, from a
* given distance and direction.
*
* To enable the spatial mode, the application needs to set a sync bus
* handler, using gst_bus_set_sync_handler(), to catch messages of type
* %GST_MESSAGE_ELEMENT named "gst.mlaudiosink.need-app" and
* "gst.mlaudiosink.need-audio-node". The need-app message will be posted first,
* application must then set the #GstMLAudioSink::app property with the pointer
* to application's lumin::BaseApp C++ object. That property can also be set on
* element creation in which case the need-app message won't be posted. After
* that, and if #GstMLAudioSink::app has been set, the need-audio-node message
* is posted from lumin::BaseApp's main thread. The application must then create
* a lumin::AudioNode C++ object, using lumin::Prism::createAudioNode(), and set
* the #GstMLAudioSink::audio-node property. Note that it is important that the
* lumin::AudioNode object must be created from within that message handler,
* and in the caller's thread, this is a limitation/bug of the platform
* (atleast until version 0.97).
*
* Here is an example of bus message handler to enable spatial sound:
* ```C
* static GstBusSyncReply
* bus_sync_handler_cb (GstBus * bus, GstMessage * msg, gpointer user_data)
* {
* MyApplication * self = user_data;
*
* if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ELEMENT) {
* if (gst_message_has_name (msg, "gst.mlaudiosink.need-app")) {
* g_object_set (G_OBJECT (msg->src), "app", &self->app, NULL);
* } else if (gst_message_has_name (msg, "gst.mlaudiosink.need-audio-node")) {
* self->audio_node = self->prism->createAudioNode ();
* self->audio_node->setSpatialSoundEnable (true);
* self->ui_node->addChild(self->audio_node);
* g_object_set (G_OBJECT (msg->src), "audio-node", self->audio_node, NULL);
* }
* }
* return GST_BUS_PASS;
* }
* ```
*
* Since: 1.18
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "mlaudiosink.h"
#include "mlaudiowrapper.h"
GST_DEBUG_CATEGORY_EXTERN (mgl_debug);
#define GST_CAT_DEFAULT mgl_debug
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) { S16LE }, "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 16000, 48000 ], " "layout = (string) interleaved"));
/* HACK: After calling MLAudioStopSound() there is no way to know when it will
* actually stop calling buffer_cb(). If the sink is disposed first, it would
* crash. Keep here a set of active sinks. */
static GHashTable *active_sinks;
static GMutex active_sinks_mutex;
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struct _GstMLAudioSink
{
GstAudioSink parent;
gpointer audio_node;
gpointer app;
GstMLAudioWrapper *wrapper;
MLAudioBufferFormat format;
uint32_t recommended_buffer_size;
MLAudioBuffer buffer;
guint buffer_offset;
gboolean has_buffer;
gboolean paused;
gboolean stopped;
GMutex mutex;
GCond cond;
};
G_DEFINE_TYPE (GstMLAudioSink, gst_ml_audio_sink, GST_TYPE_AUDIO_SINK);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (mlaudiosink, "mlaudiosink",
GST_RANK_PRIMARY + 10, GST_TYPE_ML_AUDIO_SINK,
GST_DEBUG_CATEGORY_INIT (mgl_debug, "magicleap", 0, "Magic Leap elements"));
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enum
{
PROP_0,
PROP_AUDIO_NODE,
PROP_APP,
};
static void
gst_ml_audio_sink_init (GstMLAudioSink * self)
{
g_mutex_init (&self->mutex);
g_cond_init (&self->cond);
}
static void
gst_ml_audio_sink_dispose (GObject * object)
{
GstMLAudioSink *self = GST_ML_AUDIO_SINK (object);
g_mutex_clear (&self->mutex);
g_cond_clear (&self->cond);
G_OBJECT_CLASS (gst_ml_audio_sink_parent_class)->dispose (object);
}
static void
gst_ml_audio_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstMLAudioSink *self = GST_ML_AUDIO_SINK (object);
switch (prop_id) {
case PROP_AUDIO_NODE:
self->audio_node = g_value_get_pointer (value);
break;
case PROP_APP:
self->app = g_value_get_pointer (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_ml_audio_sink_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstMLAudioSink *self = GST_ML_AUDIO_SINK (object);
switch (prop_id) {
case PROP_AUDIO_NODE:
g_value_set_pointer (value, self->audio_node);
break;
case PROP_APP:
g_value_set_pointer (value, self->app);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_ml_audio_sink_getcaps (GstBaseSink * bsink, GstCaps * filter)
{
GstCaps *caps;
caps = gst_static_caps_get (&sink_template.static_caps);
if (filter) {
gst_caps_replace (&caps,
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST));
}
return caps;
}
static gboolean
gst_ml_audio_sink_open (GstAudioSink * sink)
{
/* Nothing to do in open/close */
return TRUE;
}
static void
buffer_cb (MLHandle handle, gpointer user_data)
{
GstMLAudioSink *self = user_data;
g_mutex_lock (&active_sinks_mutex);
if (!g_hash_table_contains (active_sinks, self))
goto out;
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gst_ml_audio_wrapper_set_handle (self->wrapper, handle);
g_mutex_lock (&self->mutex);
g_cond_signal (&self->cond);
g_mutex_unlock (&self->mutex);
out:
g_mutex_unlock (&active_sinks_mutex);
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}
/* Must be called with self->mutex locked */
static gboolean
wait_for_buffer (GstMLAudioSink * self)
{
gboolean ret = TRUE;
while (!self->has_buffer && !self->stopped) {
MLResult result;
result = gst_ml_audio_wrapper_get_buffer (self->wrapper, &self->buffer);
if (result == MLResult_Ok) {
self->has_buffer = TRUE;
self->buffer_offset = 0;
} else if (result == MLAudioResult_BufferNotReady) {
g_cond_wait (&self->cond, &self->mutex);
} else {
GST_ERROR_OBJECT (self, "Failed to get output buffer: %d", result);
ret = FALSE;
break;
}
}
return ret;
}
static gboolean
create_sound_cb (GstMLAudioWrapper * wrapper, gpointer user_data)
{
GstMLAudioSink *self = user_data;
MLResult result;
if (self->app) {
gst_element_post_message (GST_ELEMENT (self),
gst_message_new_element (GST_OBJECT (self),
gst_structure_new_empty ("gst.mlaudiosink.need-audio-node")));
}
gst_ml_audio_wrapper_set_node (self->wrapper, self->audio_node);
result = gst_ml_audio_wrapper_create_sound (self->wrapper, &self->format,
self->recommended_buffer_size, buffer_cb, self);
if (result != MLResult_Ok) {
GST_ERROR_OBJECT (self, "Failed to create output stream: %d", result);
return FALSE;
}
return TRUE;
}
static gboolean
gst_ml_audio_sink_prepare (GstAudioSink * sink, GstAudioRingBufferSpec * spec)
{
GstMLAudioSink *self = GST_ML_AUDIO_SINK (sink);
float max_pitch = 1.0f;
uint32_t min_size;
MLResult result;
result =
MLAudioGetOutputStreamDefaults (GST_AUDIO_INFO_CHANNELS (&spec->info),
GST_AUDIO_INFO_RATE (&spec->info), max_pitch, &self->format,
&self->recommended_buffer_size, &min_size);
if (result != MLResult_Ok) {
GST_ERROR_OBJECT (self, "Failed to get output stream defaults: %d", result);
return FALSE;
}
if (!self->app) {
gst_element_post_message (GST_ELEMENT (self),
gst_message_new_element (GST_OBJECT (self),
gst_structure_new_empty ("gst.mlaudiosink.need-app")));
}
self->wrapper = gst_ml_audio_wrapper_new (self->app);
self->has_buffer = FALSE;
self->stopped = FALSE;
self->paused = FALSE;
g_mutex_lock (&active_sinks_mutex);
g_hash_table_add (active_sinks, self);
g_mutex_unlock (&active_sinks_mutex);
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/* createAudioNode() and createSoundWithOutputStream() must both be called in
* application's main thread, and in a single main loop iteration. */
if (!gst_ml_audio_wrapper_invoke_sync (self->wrapper, create_sound_cb, self))
return FALSE;
return TRUE;
}
static void
release_current_buffer (GstMLAudioSink * self)
{
if (self->has_buffer) {
memset (self->buffer.ptr + self->buffer_offset, 0,
self->buffer.size - self->buffer_offset);
gst_ml_audio_wrapper_release_buffer (self->wrapper);
self->has_buffer = false;
}
}
static gboolean
gst_ml_audio_sink_unprepare (GstAudioSink * sink)
{
GstMLAudioSink *self = GST_ML_AUDIO_SINK (sink);
g_mutex_lock (&active_sinks_mutex);
g_hash_table_remove (active_sinks, self);
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release_current_buffer (self);
g_clear_pointer (&self->wrapper, gst_ml_audio_wrapper_free);
g_mutex_unlock (&active_sinks_mutex);
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return TRUE;
}
static gboolean
gst_ml_audio_sink_close (GstAudioSink * sink)
{
/* Nothing to do in open/close */
return TRUE;
}
static gint
gst_ml_audio_sink_write (GstAudioSink * sink, gpointer data, guint length)
{
GstMLAudioSink *self = GST_ML_AUDIO_SINK (sink);
guint8 *input = data;
gint written = 0;
g_mutex_lock (&self->mutex);
while (length > 0) {
MLResult result;
guint to_write;
if (!wait_for_buffer (self)) {
written = -1;
break;
}
if (self->stopped) {
/* Pretend we have written the full buffer (drop data) and return
* immediately. */
release_current_buffer (self);
gst_ml_audio_wrapper_stop_sound (self->wrapper);
written = length;
break;
}
to_write = MIN (length, self->buffer.size - self->buffer_offset);
memcpy (self->buffer.ptr + self->buffer_offset, input + written, to_write);
self->buffer_offset += to_write;
if (self->buffer_offset == self->buffer.size) {
result = gst_ml_audio_wrapper_release_buffer (self->wrapper);
if (result != MLResult_Ok) {
GST_ERROR_OBJECT (self, "Failed to release buffer: %d", result);
written = -1;
break;
}
self->has_buffer = FALSE;
}
length -= to_write;
written += to_write;
}
if (self->paused) {
/* Pause was requested and we finished writing current buffer.
* See https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/665
*/
gst_ml_audio_wrapper_pause_sound (self->wrapper);
}
g_mutex_unlock (&self->mutex);
return written;
}
static guint
gst_ml_audio_sink_delay (GstAudioSink * sink)
{
GstMLAudioSink *self = GST_ML_AUDIO_SINK (sink);
MLResult result;
float latency_ms;
result = gst_ml_audio_wrapper_get_latency (self->wrapper, &latency_ms);
if (result != MLResult_Ok) {
GST_ERROR_OBJECT (self, "Failed to get latency: %d", result);
return 0;
}
return latency_ms * self->format.samples_per_second / 1000;
}
static void
gst_ml_audio_sink_pause (GstAudioSink * sink)
{
GstMLAudioSink *self = GST_ML_AUDIO_SINK (sink);
g_mutex_lock (&self->mutex);
self->paused = TRUE;
g_cond_signal (&self->cond);
g_mutex_unlock (&self->mutex);
}
static void
gst_ml_audio_sink_resume (GstAudioSink * sink)
{
GstMLAudioSink *self = GST_ML_AUDIO_SINK (sink);
g_mutex_lock (&self->mutex);
self->paused = FALSE;
gst_ml_audio_wrapper_resume_sound (self->wrapper);
g_mutex_unlock (&self->mutex);
}
static void
gst_ml_audio_sink_stop (GstAudioSink * sink)
{
GstMLAudioSink *self = GST_ML_AUDIO_SINK (sink);
g_mutex_lock (&self->mutex);
self->stopped = TRUE;
g_cond_signal (&self->cond);
g_mutex_unlock (&self->mutex);
}
static void
gst_ml_audio_sink_class_init (GstMLAudioSinkClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseSinkClass *basesink_class = GST_BASE_SINK_CLASS (klass);
GstAudioSinkClass *audiosink_class = GST_AUDIO_SINK_CLASS (klass);
active_sinks = g_hash_table_new (NULL, NULL);
g_mutex_init (&active_sinks_mutex);
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gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_ml_audio_sink_dispose);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_ml_audio_sink_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_ml_audio_sink_get_property);
g_object_class_install_property (gobject_class,
PROP_AUDIO_NODE, g_param_spec_pointer ("audio-node",
"A pointer to a lumin::AudioNode object",
"Enable spatial sound", G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_APP, g_param_spec_pointer ("app",
"A pointer to a lumin::BaseApp object",
"Enable spatial sound", G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_set_static_metadata (element_class,
"Magic Leap Audio Sink",
"Sink/Audio", "Plays audio on a Magic Leap device",
"Xavier Claessens <xavier.claessens@collabora.com>");
gst_element_class_add_static_pad_template (element_class, &sink_template);
basesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_ml_audio_sink_getcaps);
audiosink_class->open = GST_DEBUG_FUNCPTR (gst_ml_audio_sink_open);
audiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_ml_audio_sink_prepare);
audiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_ml_audio_sink_unprepare);
audiosink_class->close = GST_DEBUG_FUNCPTR (gst_ml_audio_sink_close);
audiosink_class->write = GST_DEBUG_FUNCPTR (gst_ml_audio_sink_write);
audiosink_class->delay = GST_DEBUG_FUNCPTR (gst_ml_audio_sink_delay);
audiosink_class->pause = GST_DEBUG_FUNCPTR (gst_ml_audio_sink_pause);
audiosink_class->resume = GST_DEBUG_FUNCPTR (gst_ml_audio_sink_resume);
audiosink_class->stop = GST_DEBUG_FUNCPTR (gst_ml_audio_sink_stop);
}