gstreamer/subprojects/gst-plugins-bad/sys/mediafoundation/gstmfaacenc.cpp
Seungha Yang 0b26254a6a mediafoundation: Cosmetic changes
Rename baseclass to be consistent with other Windows plugins

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1596>
2022-02-11 04:16:22 +09:00

736 lines
21 KiB
C++

/* GStreamer
* Copyright (C) 2020 Seungha Yang <seungha@centricular.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-mfaacenc
* @title: mfaacenc
*
* This element encodes raw audio into AAC compressed data.
*
* ## Example pipelines
* |[
* gst-launch-1.0 -v audiotestsrc ! mfaacenc ! aacparse ! qtmux ! filesink location=audiotestsrc.mp4
* ]| This example pipeline will encode a test audio source to AAC using
* Media Foundation encoder, and muxes it in a mp4 container.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/pbutils/pbutils.h>
#include "gstmfaudioencoder.h"
#include "gstmfaacenc.h"
#include <wrl.h>
#include <set>
#include <vector>
#include <string>
/* *INDENT-OFF* */
using namespace Microsoft::WRL;
/* *INDENT-ON* */
GST_DEBUG_CATEGORY (gst_mf_aac_enc_debug);
#define GST_CAT_DEFAULT gst_mf_aac_enc_debug
enum
{
PROP_0,
PROP_BITRATE,
};
#define DEFAULT_BITRATE (0)
typedef struct _GstMFAacEnc
{
GstMFAudioEncoder parent;
/* properties */
guint bitrate;
} GstMFAacEnc;
typedef struct _GstMFAacEncClass
{
GstMFAudioEncoderClass parent_class;
} GstMFAacEncClass;
/* *INDENT-OFF* */
typedef struct
{
GstCaps *sink_caps;
GstCaps *src_caps;
gchar *device_name;
guint32 enum_flags;
guint device_index;
std::set<UINT32> bitrate_list;
} GstMFAacEncClassData;
/* *INDENT-ON* */
static GstElementClass *parent_class = nullptr;
static void gst_mf_aac_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_mf_aac_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static gboolean gst_mf_aac_enc_get_output_type (GstMFAudioEncoder * encoder,
GstAudioInfo * info, IMFMediaType ** output_type);
static gboolean gst_mf_aac_enc_get_input_type (GstMFAudioEncoder * encoder,
GstAudioInfo * info, IMFMediaType ** input_type);
static gboolean gst_mf_aac_enc_set_src_caps (GstMFAudioEncoder * encoder,
GstAudioInfo * info);
static void
gst_mf_aac_enc_class_init (GstMFAacEncClass * klass, gpointer data)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstMFAudioEncoderClass *encoder_class = GST_MF_AUDIO_ENCODER_CLASS (klass);
GstMFAacEncClassData *cdata = (GstMFAacEncClassData *) data;
gchar *long_name;
gchar *classification;
guint max_bitrate = 0;
std::string bitrate_blurb;
parent_class = (GstElementClass *) g_type_class_peek_parent (klass);
gobject_class->get_property = gst_mf_aac_enc_get_property;
gobject_class->set_property = gst_mf_aac_enc_set_property;
bitrate_blurb = "Bitrate in bit/sec, (0 = auto), valid values are { 0";
/* *INDENT-OFF* */
for (auto iter: cdata->bitrate_list) {
bitrate_blurb += ", " + std::to_string (iter);
/* std::set<> stores values in a sorted fashion */
max_bitrate = iter;
}
bitrate_blurb += " }";
/* *INDENT-ON* */
g_object_class_install_property (gobject_class, PROP_BITRATE,
g_param_spec_uint ("bitrate", "Bitrate", bitrate_blurb.c_str (), 0,
max_bitrate, DEFAULT_BITRATE,
(GParamFlags) (GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
G_PARAM_STATIC_NAME | G_PARAM_STATIC_NICK)));
long_name = g_strdup_printf ("Media Foundation %s", cdata->device_name);
classification = g_strdup_printf ("Codec/Encoder/Audio%s",
(cdata->enum_flags & MFT_ENUM_FLAG_HARDWARE) == MFT_ENUM_FLAG_HARDWARE ?
"/Hardware" : "");
gst_element_class_set_metadata (element_class, long_name,
classification,
"Microsoft Media Foundation AAC Encoder",
"Seungha Yang <seungha@centricular.com>");
g_free (long_name);
g_free (classification);
gst_element_class_add_pad_template (element_class,
gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
cdata->sink_caps));
gst_element_class_add_pad_template (element_class,
gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
cdata->src_caps));
encoder_class->get_output_type =
GST_DEBUG_FUNCPTR (gst_mf_aac_enc_get_output_type);
encoder_class->get_input_type =
GST_DEBUG_FUNCPTR (gst_mf_aac_enc_get_input_type);
encoder_class->set_src_caps = GST_DEBUG_FUNCPTR (gst_mf_aac_enc_set_src_caps);
encoder_class->codec_id = MFAudioFormat_AAC;
encoder_class->enum_flags = cdata->enum_flags;
encoder_class->device_index = cdata->device_index;
encoder_class->frame_samples = 1024;
g_free (cdata->device_name);
gst_caps_unref (cdata->sink_caps);
gst_caps_unref (cdata->src_caps);
delete cdata;
}
static void
gst_mf_aac_enc_init (GstMFAacEnc * self)
{
self->bitrate = DEFAULT_BITRATE;
}
static void
gst_mf_aac_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstMFAacEnc *self = (GstMFAacEnc *) (object);
switch (prop_id) {
case PROP_BITRATE:
g_value_set_uint (value, self->bitrate);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_mf_aac_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstMFAacEnc *self = (GstMFAacEnc *) (object);
switch (prop_id) {
case PROP_BITRATE:
self->bitrate = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_mf_aac_enc_get_output_type (GstMFAudioEncoder * encoder,
GstAudioInfo * info, IMFMediaType ** output_type)
{
GstMFAacEnc *self = (GstMFAacEnc *) encoder;
GstMFTransform *transform = encoder->transform;
GList *output_list = nullptr;
GList *iter;
ComPtr < IMFMediaType > target_output;
std::vector < ComPtr < IMFMediaType >> filtered_types;
std::set < UINT32 > bitrate_list;
UINT32 bitrate;
UINT32 target_bitrate = 0;
HRESULT hr;
if (!gst_mf_transform_get_output_available_types (transform, &output_list)) {
GST_ERROR_OBJECT (self, "Couldn't get available output type");
return FALSE;
}
/* 1. Filtering based on channels and sample rate */
for (iter = output_list; iter; iter = g_list_next (iter)) {
IMFMediaType *type = (IMFMediaType *) iter->data;
GUID guid = GUID_NULL;
UINT32 value;
hr = type->GetGUID (MF_MT_MAJOR_TYPE, &guid);
if (!gst_mf_result (hr))
continue;
if (!IsEqualGUID (guid, MFMediaType_Audio)) {
GST_WARNING_OBJECT (self, "Major type is not audio");
continue;
}
hr = type->GetGUID (MF_MT_SUBTYPE, &guid);
if (!gst_mf_result (hr))
continue;
if (!IsEqualGUID (guid, MFAudioFormat_AAC)) {
GST_WARNING_OBJECT (self, "Sub type is not AAC");
continue;
}
hr = type->GetUINT32 (MF_MT_AUDIO_NUM_CHANNELS, &value);
if (!gst_mf_result (hr))
continue;
if (value != GST_AUDIO_INFO_CHANNELS (info))
continue;
hr = type->GetUINT32 (MF_MT_AUDIO_SAMPLES_PER_SECOND, &value);
if (!gst_mf_result (hr))
continue;
if (value != GST_AUDIO_INFO_RATE (info))
continue;
hr = type->GetUINT32 (MF_MT_AUDIO_AVG_BYTES_PER_SECOND, &value);
if (!gst_mf_result (hr))
continue;
filtered_types.push_back (type);
/* convert bytes to bit */
bitrate_list.insert (value * 8);
}
g_list_free_full (output_list, (GDestroyNotify) gst_mf_media_type_release);
if (filtered_types.empty ()) {
GST_ERROR_OBJECT (self, "Couldn't find target output type");
return FALSE;
}
GST_DEBUG_OBJECT (self, "have %d candidate output", filtered_types.size ());
/* 2. Find the best matching bitrate */
bitrate = self->bitrate;
/* Media Foundation AAC encoder supports sample-rate 44100 or 48000 */
if (bitrate == 0) {
/* http://wiki.hydrogenaud.io/index.php?title=Fraunhofer_FDK_AAC#Recommended_Sampling_Rate_and_Bitrate_Combinations
* was referenced but the supported range by MediaFoudation is much limited
* than it */
if (GST_AUDIO_INFO_CHANNELS (info) == 1) {
if (GST_AUDIO_INFO_RATE (info) <= 44100) {
bitrate = 96000;
} else {
bitrate = 160000;
}
} else if (GST_AUDIO_INFO_CHANNELS (info) == 2) {
if (GST_AUDIO_INFO_RATE (info) <= 44100) {
bitrate = 112000;
} else {
bitrate = 320000;
}
} else {
/* 5.1 */
if (GST_AUDIO_INFO_RATE (info) <= 44100) {
bitrate = 240000;
} else {
bitrate = 320000;
}
}
GST_DEBUG_OBJECT (self, "Calculated bitrate %d", bitrate);
} else {
GST_DEBUG_OBJECT (self, "Requested bitrate %d", bitrate);
}
GST_DEBUG_OBJECT (self, "Available bitrates");
/* *INDENT-OFF* */
for (auto it: bitrate_list)
GST_DEBUG_OBJECT (self, "\t%d", it);
/* Based on calculated or requested bitrate, find the closest supported
* bitrate */
{
const auto it = bitrate_list.lower_bound (bitrate);
if (it == bitrate_list.end()) {
target_bitrate = *std::prev (it);
} else {
target_bitrate = *it;
}
}
GST_DEBUG_OBJECT (self, "Selected target bitrate %d", target_bitrate);
for (auto it: filtered_types) {
UINT32 value = 0;
it->GetUINT32 (MF_MT_AUDIO_AVG_BYTES_PER_SECOND, &value);
if (value * 8 == target_bitrate) {
target_output = it;
break;
}
}
/* *INDENT-ON* */
if (!target_output) {
GST_ERROR_OBJECT (self, "Failed to decide final output type");
return FALSE;
}
*output_type = target_output.Detach ();
return TRUE;
}
static gboolean
gst_mf_aac_enc_get_input_type (GstMFAudioEncoder * encoder, GstAudioInfo * info,
IMFMediaType ** input_type)
{
GstMFAacEnc *self = (GstMFAacEnc *) encoder;
GstMFTransform *transform = encoder->transform;
GList *input_list = nullptr;
GList *iter;
ComPtr < IMFMediaType > target_input;
std::vector < ComPtr < IMFMediaType >> filtered_types;
std::set < UINT32 > bitrate_list;
HRESULT hr;
if (!gst_mf_transform_get_input_available_types (transform, &input_list)) {
GST_ERROR_OBJECT (self, "Couldn't get available output type");
return FALSE;
}
/* 1. Filtering based on channels and sample rate */
for (iter = input_list; iter; iter = g_list_next (iter)) {
IMFMediaType *type = (IMFMediaType *) iter->data;
GUID guid = GUID_NULL;
UINT32 value;
hr = type->GetGUID (MF_MT_MAJOR_TYPE, &guid);
if (!gst_mf_result (hr))
continue;
if (!IsEqualGUID (guid, MFMediaType_Audio)) {
GST_WARNING_OBJECT (self, "Major type is not audio");
continue;
}
hr = type->GetGUID (MF_MT_SUBTYPE, &guid);
if (!gst_mf_result (hr))
continue;
if (!IsEqualGUID (guid, MFAudioFormat_PCM)) {
GST_WARNING_OBJECT (self, "Sub type is not PCM");
continue;
}
hr = type->GetUINT32 (MF_MT_AUDIO_NUM_CHANNELS, &value);
if (!gst_mf_result (hr))
continue;
if (value != GST_AUDIO_INFO_CHANNELS (info))
continue;
hr = type->GetUINT32 (MF_MT_AUDIO_SAMPLES_PER_SECOND, &value);
if (!gst_mf_result (hr))
continue;
if (value != GST_AUDIO_INFO_RATE (info))
continue;
filtered_types.push_back (type);
}
g_list_free_full (input_list, (GDestroyNotify) gst_mf_media_type_release);
if (filtered_types.empty ()) {
GST_ERROR_OBJECT (self, "Couldn't find target input type");
return FALSE;
}
GST_DEBUG_OBJECT (self, "Total %d input types are available",
filtered_types.size ());
/* Just select the first one */
target_input = *filtered_types.begin ();
*input_type = target_input.Detach ();
return TRUE;
}
static gboolean
gst_mf_aac_enc_set_src_caps (GstMFAudioEncoder * encoder, GstAudioInfo * info)
{
GstMFAacEnc *self = (GstMFAacEnc *) encoder;
HRESULT hr;
GstCaps *src_caps;
GstBuffer *codec_data;
UINT8 *blob = nullptr;
UINT32 blob_size = 0;
gboolean ret;
ComPtr < IMFMediaType > output_type;
static const guint config_data_offset = 12;
if (!gst_mf_transform_get_output_current_type (encoder->transform,
&output_type)) {
GST_ERROR_OBJECT (self, "Couldn't get current output type");
return FALSE;
}
/* user data contains the portion of the HEAACWAVEINFO structure that appears
* after the WAVEFORMATEX structure (that is, after the wfx member).
* This is followed by the AudioSpecificConfig() data,
* as defined by ISO/IEC 14496-3.
* https://docs.microsoft.com/en-us/windows/win32/medfound/aac-encoder
*
* The offset AudioSpecificConfig() data is 12 in this case
*/
hr = output_type->GetBlobSize (MF_MT_USER_DATA, &blob_size);
if (!gst_mf_result (hr) || blob_size <= config_data_offset) {
GST_ERROR_OBJECT (self,
"Couldn't get size of MF_MT_USER_DATA, size %d, %d", blob_size);
return FALSE;
}
hr = output_type->GetAllocatedBlob (MF_MT_USER_DATA, &blob, &blob_size);
if (!gst_mf_result (hr)) {
GST_ERROR_OBJECT (self, "Couldn't get user data blob");
return FALSE;
}
codec_data = gst_buffer_new_and_alloc (blob_size - config_data_offset);
gst_buffer_fill (codec_data, 0, blob + config_data_offset,
blob_size - config_data_offset);
src_caps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 4,
"stream-format", G_TYPE_STRING, "raw",
"channels", G_TYPE_INT, GST_AUDIO_INFO_CHANNELS (info),
"rate", G_TYPE_INT, GST_AUDIO_INFO_RATE (info),
"framed", G_TYPE_BOOLEAN, TRUE,
"codec_data", GST_TYPE_BUFFER, codec_data, nullptr);
gst_buffer_unref (codec_data);
gst_codec_utils_aac_caps_set_level_and_profile (src_caps,
blob + config_data_offset, blob_size - config_data_offset);
CoTaskMemFree (blob);
ret =
gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (self), src_caps);
if (!ret) {
GST_WARNING_OBJECT (self,
"Couldn't set output format %" GST_PTR_FORMAT, src_caps);
}
gst_caps_unref (src_caps);
return ret;
}
static void
gst_mf_aac_enc_register (GstPlugin * plugin, guint rank,
const gchar * device_name, guint32 enum_flags, guint device_index,
GstCaps * sink_caps, GstCaps * src_caps,
const std::set < UINT32 > &bitrate_list)
{
GType type;
gchar *type_name;
gchar *feature_name;
gint i;
GstMFAacEncClassData *cdata;
gboolean is_default = TRUE;
GTypeInfo type_info = {
sizeof (GstMFAacEncClass),
nullptr,
nullptr,
(GClassInitFunc) gst_mf_aac_enc_class_init,
nullptr,
nullptr,
sizeof (GstMFAacEnc),
0,
(GInstanceInitFunc) gst_mf_aac_enc_init,
};
cdata = new GstMFAacEncClassData;
cdata->sink_caps = sink_caps;
cdata->src_caps = src_caps;
cdata->device_name = g_strdup (device_name);
cdata->enum_flags = enum_flags;
cdata->device_index = device_index;
cdata->bitrate_list = bitrate_list;
type_info.class_data = cdata;
type_name = g_strdup ("GstMFAacEnc");
feature_name = g_strdup ("mfaacenc");
i = 1;
while (g_type_from_name (type_name) != 0) {
g_free (type_name);
g_free (feature_name);
type_name = g_strdup_printf ("GstMFAacDevice%dEnc", i);
feature_name = g_strdup_printf ("mfaacdevice%denc", i);
is_default = FALSE;
i++;
}
type =
g_type_register_static (GST_TYPE_MF_AUDIO_ENCODER, type_name, &type_info,
(GTypeFlags) 0);
/* make lower rank than default device */
if (rank > 0 && !is_default)
rank--;
if (!gst_element_register (plugin, feature_name, rank, type))
GST_WARNING ("Failed to register plugin '%s'", type_name);
g_free (type_name);
g_free (feature_name);
}
static void
gst_mf_aac_enc_plugin_init_internal (GstPlugin * plugin, guint rank,
GstMFTransform * transform, guint device_index, guint32 enum_flags)
{
HRESULT hr;
gint i;
GstCaps *src_caps = nullptr;
GstCaps *sink_caps = nullptr;
gchar *device_name = nullptr;
GList *output_list = nullptr;
GList *iter;
std::set < UINT32 > channels_list;
std::set < UINT32 > rate_list;
std::set < UINT32 > bitrate_list;
gboolean config_found = FALSE;
GValue channles_value = G_VALUE_INIT;
GValue rate_value = G_VALUE_INIT;
if (!gst_mf_transform_open (transform))
return;
g_object_get (transform, "device-name", &device_name, nullptr);
if (!device_name) {
GST_WARNING_OBJECT (transform, "Unknown device name");
return;
}
if (!gst_mf_transform_get_output_available_types (transform, &output_list)) {
GST_WARNING_OBJECT (transform, "Couldn't get output types");
goto done;
}
GST_INFO_OBJECT (transform, "Have %d output type",
g_list_length (output_list));
for (iter = output_list, i = 0; iter; iter = g_list_next (iter), i++) {
UINT32 channels, rate, bitrate;
GUID guid = GUID_NULL;
IMFMediaType *type = (IMFMediaType *) iter->data;
#ifndef GST_DISABLE_GST_DEBUG
gchar *msg = g_strdup_printf ("Output IMFMediaType %d", i);
gst_mf_dump_attributes ((IMFAttributes *) type, msg, GST_LEVEL_TRACE);
g_free (msg);
#endif
hr = type->GetGUID (MF_MT_MAJOR_TYPE, &guid);
if (!gst_mf_result (hr))
continue;
/* shouldn't happen */
if (!IsEqualGUID (guid, MFMediaType_Audio))
continue;
hr = type->GetGUID (MF_MT_SUBTYPE, &guid);
if (!gst_mf_result (hr))
continue;
/* shouldn't happen */
if (!IsEqualGUID (guid, MFAudioFormat_AAC))
continue;
/* Windows 10 channels 6 (5.1) channels so we cannot hard code it */
hr = type->GetUINT32 (MF_MT_AUDIO_NUM_CHANNELS, &channels);
if (!gst_mf_result (hr))
continue;
hr = type->GetUINT32 (MF_MT_AUDIO_SAMPLES_PER_SECOND, &rate);
if (!gst_mf_result (hr))
continue;
/* NOTE: MFT AAC encoder seems to support more bitrate than it's documented
* at https://docs.microsoft.com/en-us/windows/win32/medfound/aac-encoder
* We will pass supported bitrate values to class init
*/
hr = type->GetUINT32 (MF_MT_AUDIO_AVG_BYTES_PER_SECOND, &bitrate);
if (!gst_mf_result (hr))
continue;
channels_list.insert (channels);
rate_list.insert (rate);
/* convert bytes to bit */
bitrate_list.insert (bitrate * 8);
config_found = TRUE;
}
if (!config_found) {
GST_WARNING_OBJECT (transform, "Couldn't find available configuration");
goto done;
}
src_caps =
gst_caps_from_string ("audio/mpeg, mpegversion = (int) 4, "
"stream-format = (string) raw, framed = (boolean) true, "
"base-profile = (string) lc");
sink_caps =
gst_caps_from_string ("audio/x-raw, layout = (string) interleaved, "
"format = (string) " GST_AUDIO_NE (S16));
g_value_init (&channles_value, GST_TYPE_LIST);
g_value_init (&rate_value, GST_TYPE_LIST);
/* *INDENT-OFF* */
for (auto it: channels_list) {
GValue channles = G_VALUE_INIT;
g_value_init (&channles, G_TYPE_INT);
g_value_set_int (&channles, (gint) it);
gst_value_list_append_and_take_value (&channles_value, &channles);
}
for (auto it: rate_list) {
GValue rate = G_VALUE_INIT;
g_value_init (&rate, G_TYPE_INT);
g_value_set_int (&rate, (gint) it);
gst_value_list_append_and_take_value (&rate_value, &rate);
}
/* *INDENT-ON* */
gst_caps_set_value (src_caps, "channels", &channles_value);
gst_caps_set_value (sink_caps, "channels", &channles_value);
gst_caps_set_value (src_caps, "rate", &rate_value);
gst_caps_set_value (sink_caps, "rate", &rate_value);
GST_MINI_OBJECT_FLAG_SET (sink_caps, GST_MINI_OBJECT_FLAG_MAY_BE_LEAKED);
GST_MINI_OBJECT_FLAG_SET (src_caps, GST_MINI_OBJECT_FLAG_MAY_BE_LEAKED);
gst_mf_aac_enc_register (plugin, rank, device_name, enum_flags, device_index,
sink_caps, src_caps, bitrate_list);
done:
if (output_list)
g_list_free_full (output_list, (GDestroyNotify) gst_mf_media_type_release);
g_free (device_name);
g_value_unset (&channles_value);
g_value_unset (&rate_value);
}
void
gst_mf_aac_enc_plugin_init (GstPlugin * plugin, guint rank)
{
GstMFTransformEnumParams enum_params = { 0, };
MFT_REGISTER_TYPE_INFO output_type;
GstMFTransform *transform;
gint i;
gboolean do_next;
GST_DEBUG_CATEGORY_INIT (gst_mf_aac_enc_debug, "mfaacenc", 0, "mfaacenc");
output_type.guidMajorType = MFMediaType_Audio;
output_type.guidSubtype = MFAudioFormat_AAC;
enum_params.category = MFT_CATEGORY_AUDIO_ENCODER;
enum_params.enum_flags = (MFT_ENUM_FLAG_SYNCMFT |
MFT_ENUM_FLAG_SORTANDFILTER | MFT_ENUM_FLAG_SORTANDFILTER_APPROVED_ONLY);
enum_params.output_typeinfo = &output_type;
i = 0;
do {
enum_params.device_index = i++;
transform = gst_mf_transform_new (&enum_params);
do_next = TRUE;
if (!transform) {
do_next = FALSE;
} else {
gst_mf_aac_enc_plugin_init_internal (plugin, rank, transform,
enum_params.device_index, enum_params.enum_flags);
gst_clear_object (&transform);
}
} while (do_next);
}