mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-20 23:36:38 +00:00
205 lines
5.1 KiB
C++
205 lines
5.1 KiB
C++
/*
|
|
* 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;
|
|
}
|