--- /dev/null
+/*
+ * 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"));
+
+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);
+
+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;
+
+ gst_ml_audio_wrapper_set_handle (self->wrapper, handle);
+
+ g_mutex_lock (&self->mutex);
+ g_cond_signal (&self->cond);
+ g_mutex_unlock (&self->mutex);
+}
+
+/* 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;
+
+ /* 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);
+
+ release_current_buffer (self);
+ g_clear_pointer (&self->wrapper, gst_ml_audio_wrapper_free);
+
+ 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);
+
+ 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);
+}
--- /dev/null
+/*
+ * 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
+ *
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "mlaudiowrapper.h"
+
+#include <ml_audio.h>
+
+#include <lumin/node/AudioNode.h>
+#include <lumin/BaseApp.h>
+#include <lumin/Prism.h>
+
+GST_DEBUG_CATEGORY_EXTERN (mgl_debug);
+#define GST_CAT_DEFAULT mgl_debug
+
+using lumin::BaseApp;
+using lumin::AudioNode;
+using lumin::AudioBuffer;
+using lumin::AudioBufferFormat;
+using lumin::AudioSampleFormat;
+
+struct _GstMLAudioWrapper
+{
+ BaseApp *app;
+ AudioNode *node;
+ MLHandle handle;
+};
+
+AudioBufferFormat
+convert_buffer_format(const MLAudioBufferFormat *format)
+{
+ AudioBufferFormat ret;
+ ret.channel_count = format->channel_count;
+ ret.samples_per_second = format->samples_per_second;
+ ret.bits_per_sample = format->bits_per_sample;
+ ret.valid_bits_per_sample = format->valid_bits_per_sample;
+ switch (format->sample_format) {
+ case MLAudioSampleFormat_Int:
+ ret.sample_format = AudioSampleFormat::Integer;
+ break;
+ case MLAudioSampleFormat_Float:
+ ret.sample_format = AudioSampleFormat::Float;
+ break;
+ default:
+ g_warn_if_reached ();
+ ret.sample_format = (AudioSampleFormat)format->sample_format;
+ };
+ ret.reserved = format->reserved;
+
+ return ret;
+}
+
+GstMLAudioWrapper *
+gst_ml_audio_wrapper_new (gpointer app)
+{
+ GstMLAudioWrapper *self;
+
+ self = g_new0 (GstMLAudioWrapper, 1);
+ self->app = reinterpret_cast<BaseApp *>(app);
+ self->node = nullptr;
+ self->handle = ML_INVALID_HANDLE;
+
+ return self;
+}
+
+void
+gst_ml_audio_wrapper_free (GstMLAudioWrapper *self)
+{
+ if (self->node) {
+ self->app->RunOnMainThreadSync ([self] {
+ /* Stop playing sound, but user is responsible to destroy the node */
+ self->node->stopSound ();
+ });
+ } else {
+ MLAudioDestroySound (self->handle);
+ }
+
+ g_free (self);
+}
+
+MLResult
+gst_ml_audio_wrapper_create_sound (GstMLAudioWrapper *self,
+ const MLAudioBufferFormat *format,
+ uint32_t buffer_size,
+ MLAudioBufferCallback callback,
+ gpointer user_data)
+{
+ if (self->node) {
+ auto format2 = convert_buffer_format (format);
+ bool success = FALSE;
+ success = self->node->createSoundWithOutputStream (&format2,
+ buffer_size, callback, user_data);
+ if (success)
+ self->node->startSound ();
+ return success ? MLResult_Ok : MLResult_UnspecifiedFailure;
+ }
+
+ MLResult result = MLAudioCreateSoundWithOutputStream (format, buffer_size,
+ callback, user_data, &self->handle);
+ if (result == MLResult_Ok)
+ result = MLAudioStartSound (self->handle);
+
+ return result;
+}
+
+MLResult
+gst_ml_audio_wrapper_pause_sound (GstMLAudioWrapper *self)
+{
+ g_return_val_if_fail (self->handle != ML_INVALID_HANDLE,
+ MLResult_UnspecifiedFailure);
+ return MLAudioPauseSound (self->handle);
+}
+
+MLResult
+gst_ml_audio_wrapper_resume_sound (GstMLAudioWrapper *self)
+{
+ g_return_val_if_fail (self->handle != ML_INVALID_HANDLE,
+ MLResult_UnspecifiedFailure);
+ return MLAudioResumeSound (self->handle);
+}
+
+MLResult
+gst_ml_audio_wrapper_stop_sound (GstMLAudioWrapper *self)
+{
+ g_return_val_if_fail (self->handle != ML_INVALID_HANDLE,
+ MLResult_UnspecifiedFailure);
+ return MLAudioStopSound (self->handle);
+}
+
+MLResult
+gst_ml_audio_wrapper_get_latency (GstMLAudioWrapper *self,
+ float *out_latency_in_msec)
+{
+ if (self->handle == ML_INVALID_HANDLE) {
+ *out_latency_in_msec = 0;
+ return MLResult_Ok;
+ }
+
+ return MLAudioGetOutputStreamLatency (self->handle, out_latency_in_msec);
+}
+
+MLResult
+gst_ml_audio_wrapper_get_buffer (GstMLAudioWrapper *self,
+ MLAudioBuffer *out_buffer)
+{
+ return MLAudioGetOutputStreamBuffer (self->handle, out_buffer);
+}
+
+MLResult
+gst_ml_audio_wrapper_release_buffer (GstMLAudioWrapper *self)
+{
+ return MLAudioReleaseOutputStreamBuffer (self->handle);
+}
+
+void
+gst_ml_audio_wrapper_set_handle (GstMLAudioWrapper *self, MLHandle handle)
+{
+ g_return_if_fail (self->handle == ML_INVALID_HANDLE || self->handle == handle);
+ self->handle = handle;
+}
+
+void
+gst_ml_audio_wrapper_set_node (GstMLAudioWrapper *self,
+ gpointer node)
+{
+ g_return_if_fail (self->node == nullptr);
+ self->node = reinterpret_cast<AudioNode *>(node);
+}
+
+gboolean
+gst_ml_audio_wrapper_invoke_sync (GstMLAudioWrapper *self,
+ GstMLAudioWrapperCallback callback, gpointer user_data)
+{
+ gboolean ret;
+
+ if (self->app) {
+ self->app->RunOnMainThreadSync ([self, callback, user_data, &ret] {
+ ret = callback (self, user_data);
+ });
+ } else {
+ ret = callback (self, user_data);
+ }
+
+ return ret;
+}
projects / platform / upstream / gstreamer.git / commitdiff