--- /dev/null
+/* 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 "gstmfaudioenc.h"
+#include "gstmfaacenc.h"
+#include <wrl.h>
+#include <set>
+#include <vector>
+#include <string>
+
+using namespace Microsoft::WRL;
+
+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
+{
+ GstMFAudioEnc parent;
+
+ /* properteies */
+ guint bitrate;
+} GstMFAacEnc;
+
+typedef struct _GstMFAacEncClass
+{
+ GstMFAudioEncClass parent_class;
+
+} GstMFAacEncClass;
+
+typedef struct
+{
+ GstCaps *sink_caps;
+ GstCaps *src_caps;
+ gchar *device_name;
+ guint32 enum_flags;
+ guint device_index;
+ std::set<UINT32> bitrate_list;
+} GstMFAacEncClassData;
+
+static GstElementClass *parent_class = NULL;
+
+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 (GstMFAudioEnc * mfenc,
+ GstAudioInfo * info, IMFMediaType ** output_type);
+static gboolean gst_mf_aac_enc_get_input_type (GstMFAudioEnc * mfenc,
+ GstAudioInfo * info, IMFMediaType ** input_type);
+static gboolean gst_mf_aac_enc_set_src_caps (GstMFAudioEnc * mfenc,
+ 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);
+ GstMFAudioEncClass *mfenc_class = GST_MF_AUDIO_ENC_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";
+ 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 += " }";
+
+ 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));
+
+ mfenc_class->get_output_type =
+ GST_DEBUG_FUNCPTR (gst_mf_aac_enc_get_output_type);
+ mfenc_class->get_input_type =
+ GST_DEBUG_FUNCPTR (gst_mf_aac_enc_get_input_type);
+ mfenc_class->set_src_caps =
+ GST_DEBUG_FUNCPTR (gst_mf_aac_enc_set_src_caps);
+
+ mfenc_class->codec_id = MFAudioFormat_AAC;
+ mfenc_class->enum_flags = cdata->enum_flags;
+ mfenc_class->device_index = cdata->device_index;
+ mfenc_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 (GstMFAudioEnc * mfenc, GstAudioInfo * info,
+ IMFMediaType ** output_type)
+{
+ GstMFAacEnc *self = (GstMFAacEnc *) mfenc;
+ GstMFTransform *transform = mfenc->transform;
+ GList *output_list = NULL;
+ 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");
+ 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;
+ }
+ }
+
+ 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 (GstMFAudioEnc * mfenc, GstAudioInfo * info,
+ IMFMediaType ** input_type)
+{
+ GstMFAacEnc *self = (GstMFAacEnc *) mfenc;
+ GstMFTransform *transform = mfenc->transform;
+ GList *input_list = NULL;
+ 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 (GstMFAudioEnc * mfenc,
+ GstAudioInfo * info)
+{
+ GstMFAacEnc *self = (GstMFAacEnc *) mfenc;
+ HRESULT hr;
+ GstCaps *src_caps;
+ GstBuffer *codec_data;
+ UINT8 *blob = NULL;
+ 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 (mfenc->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, NULL);
+ 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),
+ NULL,
+ NULL,
+ (GClassInitFunc) gst_mf_aac_enc_class_init,
+ NULL,
+ NULL,
+ 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_ENC, 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 = NULL;
+ GstCaps *sink_caps = NULL;
+ gchar *device_name = NULL;
+ GList *output_list = NULL;
+ 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, NULL);
+ 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);
+
+ 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);
+ }
+
+ 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_HARDWARE | MFT_ENUM_FLAG_ASYNCMFT |
+ MFT_ENUM_FLAG_SORTANDFILTER | MFT_ENUM_FLAG_SORTANDFILTER_APPROVED_ONLY);
+ enum_params.output_typeinfo = &output_type;
+
+ /* register hardware encoders first (likey no hardware audio encoder) */
+ 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);
+
+ /* register software encoders */
+ enum_params.enum_flags = (MFT_ENUM_FLAG_SYNCMFT |
+ MFT_ENUM_FLAG_SORTANDFILTER | MFT_ENUM_FLAG_SORTANDFILTER_APPROVED_ONLY);
+ 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);
+}
--- /dev/null
+/* 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include "gstmfaudioenc.h"
+#include <wrl.h>
+#include <string.h>
+
+using namespace Microsoft::WRL;
+
+GST_DEBUG_CATEGORY (gst_mf_audio_enc_debug);
+#define GST_CAT_DEFAULT gst_mf_audio_enc_debug
+
+#define gst_mf_audio_enc_parent_class parent_class
+G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstMFAudioEnc, gst_mf_audio_enc,
+ GST_TYPE_AUDIO_ENCODER,
+ GST_DEBUG_CATEGORY_INIT (gst_mf_audio_enc_debug, "mfaudioenc", 0,
+ "mfaudioenc"));
+
+static gboolean gst_mf_audio_enc_open (GstAudioEncoder * enc);
+static gboolean gst_mf_audio_enc_close (GstAudioEncoder * enc);
+static gboolean gst_mf_audio_enc_set_format (GstAudioEncoder * enc,
+ GstAudioInfo * info);
+static GstFlowReturn gst_mf_audio_enc_handle_frame (GstAudioEncoder * enc,
+ GstBuffer *buffer);
+static GstFlowReturn gst_mf_audio_enc_drain (GstAudioEncoder * enc);
+static void gst_mf_audio_enc_flush (GstAudioEncoder * enc);
+
+static void
+gst_mf_audio_enc_class_init (GstMFAudioEncClass * klass)
+{
+ GstAudioEncoderClass *audioenc_class = GST_AUDIO_ENCODER_CLASS (klass);
+
+ audioenc_class->open = GST_DEBUG_FUNCPTR (gst_mf_audio_enc_open);
+ audioenc_class->close = GST_DEBUG_FUNCPTR (gst_mf_audio_enc_close);
+ audioenc_class->set_format = GST_DEBUG_FUNCPTR (gst_mf_audio_enc_set_format);
+ audioenc_class->handle_frame =
+ GST_DEBUG_FUNCPTR (gst_mf_audio_enc_handle_frame);
+ audioenc_class->flush =
+ GST_DEBUG_FUNCPTR (gst_mf_audio_enc_flush);
+}
+
+static void
+gst_mf_audio_enc_init (GstMFAudioEnc * self)
+{
+ gst_audio_encoder_set_drainable (GST_AUDIO_ENCODER (self), TRUE);
+}
+
+static gboolean
+gst_mf_audio_enc_open (GstAudioEncoder * enc)
+{
+ GstMFAudioEnc *self = GST_MF_AUDIO_ENC (enc);
+ GstMFAudioEncClass *klass = GST_MF_AUDIO_ENC_GET_CLASS (enc);
+ GstMFTransformEnumParams enum_params = { 0, };
+ MFT_REGISTER_TYPE_INFO output_type;
+ gboolean ret;
+
+ output_type.guidMajorType = MFMediaType_Audio;
+ output_type.guidSubtype = klass->codec_id;
+
+ enum_params.category = MFT_CATEGORY_AUDIO_ENCODER;
+ enum_params.enum_flags = klass->enum_flags;
+ enum_params.output_typeinfo = &output_type;
+ enum_params.device_index = klass->device_index;
+
+ GST_DEBUG_OBJECT (self, "Create MFT with enum flags 0x%x, device index %d",
+ klass->enum_flags, klass->device_index);
+
+ self->transform = gst_mf_transform_new (&enum_params);
+ ret = !!self->transform;
+
+ if (!ret)
+ GST_ERROR_OBJECT (self, "Cannot create MFT object");
+
+ return ret;
+}
+
+static gboolean
+gst_mf_audio_enc_close (GstAudioEncoder * enc)
+{
+ GstMFAudioEnc *self = GST_MF_AUDIO_ENC (enc);
+
+ gst_clear_object (&self->transform);
+
+ return TRUE;
+}
+
+static gboolean
+gst_mf_audio_enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
+{
+ GstMFAudioEnc *self = GST_MF_AUDIO_ENC (enc);
+ GstMFAudioEncClass *klass = GST_MF_AUDIO_ENC_GET_CLASS (enc);
+ ComPtr<IMFMediaType> in_type;
+ ComPtr<IMFMediaType> out_type;
+
+ GST_DEBUG_OBJECT (self, "Set format");
+
+ gst_mf_audio_enc_drain (enc);
+
+ if (!gst_mf_transform_open (self->transform)) {
+ GST_ERROR_OBJECT (self, "Failed to open MFT");
+ return FALSE;
+ }
+
+ g_assert (klass->get_output_type != NULL);
+ if (!klass->get_output_type (self, info, &out_type)) {
+ GST_ERROR_OBJECT (self, "subclass failed to set output type");
+ return FALSE;
+ }
+
+ gst_mf_dump_attributes (out_type.Get(), "Set output type", GST_LEVEL_DEBUG);
+
+ if (!gst_mf_transform_set_output_type (self->transform, out_type.Get ())) {
+ GST_ERROR_OBJECT (self, "Couldn't set output type");
+ return FALSE;
+ }
+
+ g_assert (klass->get_input_type != NULL);
+ if (!klass->get_input_type (self, info, &in_type)) {
+ GST_ERROR_OBJECT (self, "subclass didn't provide input type");
+ return FALSE;
+ }
+
+ gst_mf_dump_attributes (in_type.Get(), "Set input type", GST_LEVEL_DEBUG);
+
+ if (!gst_mf_transform_set_input_type (self->transform, in_type.Get ())) {
+ GST_ERROR_OBJECT (self, "Couldn't set input media type");
+ return FALSE;
+ }
+
+ g_assert (klass->set_src_caps != NULL);
+ if (!klass->set_src_caps (self, info))
+ return FALSE;
+
+ g_assert (klass->frame_samples > 0);
+ gst_audio_encoder_set_frame_samples_min (enc, klass->frame_samples);
+ gst_audio_encoder_set_frame_samples_max (enc, klass->frame_samples);
+ gst_audio_encoder_set_frame_max (enc, 1);
+
+ /* mediafoundation encoder needs timestamp and duration */
+ self->sample_count = 0;
+ self->sample_duration_in_mf = gst_util_uint64_scale (klass->frame_samples,
+ 10000000, GST_AUDIO_INFO_RATE (info));
+
+ GST_DEBUG_OBJECT (self,
+ "Calculated sample duration %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (self->sample_duration_in_mf * 100));
+
+ return TRUE;
+}
+
+static gboolean
+gst_mf_audio_enc_process_input (GstMFAudioEnc * self, GstBuffer * buffer)
+{
+ HRESULT hr;
+ ComPtr<IMFSample> sample;
+ ComPtr<IMFMediaBuffer> media_buffer;
+ BYTE *data;
+ gboolean res = FALSE;
+ GstMapInfo info;
+ guint64 timestamp;
+
+ if (!gst_buffer_map (buffer, &info, GST_MAP_READ)) {
+ GST_ELEMENT_ERROR (self,
+ RESOURCE, READ, ("Couldn't map input buffer"), (NULL));
+ return FALSE;
+ }
+
+ GST_TRACE_OBJECT (self, "Process buffer %" GST_PTR_FORMAT, buffer);
+
+ timestamp = self->sample_count * self->sample_duration_in_mf;
+
+ hr = MFCreateSample (sample.GetAddressOf ());
+ if (!gst_mf_result (hr))
+ goto done;
+
+ hr = MFCreateMemoryBuffer (info.size, media_buffer.GetAddressOf ());
+ if (!gst_mf_result (hr))
+ goto done;
+
+ hr = media_buffer->Lock (&data, NULL, NULL);
+ if (!gst_mf_result (hr))
+ goto done;
+
+ memcpy (data, info.data, info.size);
+ media_buffer->Unlock ();
+
+ hr = media_buffer->SetCurrentLength (info.size);
+ if (!gst_mf_result (hr))
+ goto done;
+
+ hr = sample->AddBuffer (media_buffer.Get ());
+ if (!gst_mf_result (hr))
+ goto done;
+
+ hr = sample->SetSampleTime (timestamp);
+ if (!gst_mf_result (hr))
+ goto done;
+
+ hr = sample->SetSampleDuration (self->sample_duration_in_mf);
+ if (!gst_mf_result (hr))
+ goto done;
+
+ if (!gst_mf_transform_process_input (self->transform, sample.Get ())) {
+ GST_ERROR_OBJECT (self, "Failed to process input");
+ goto done;
+ }
+
+ self->sample_count++;
+
+ res = TRUE;
+
+done:
+ gst_buffer_unmap (buffer, &info);
+
+ return res;
+}
+
+static GstFlowReturn
+gst_mf_audio_enc_process_output (GstMFAudioEnc * self)
+{
+ GstMFAudioEncClass *klass = GST_MF_AUDIO_ENC_GET_CLASS (self);
+ HRESULT hr;
+ BYTE *data;
+ ComPtr<IMFMediaBuffer> media_buffer;
+ ComPtr<IMFSample> sample;
+ GstBuffer *buffer;
+ GstFlowReturn res = GST_FLOW_ERROR;
+ DWORD buffer_len;
+
+ res = gst_mf_transform_get_output (self->transform, sample.GetAddressOf ());
+
+ if (res != GST_FLOW_OK)
+ return res;
+
+ hr = sample->GetBufferByIndex (0, media_buffer.GetAddressOf ());
+ if (!gst_mf_result (hr))
+ return GST_FLOW_ERROR;
+
+ hr = media_buffer->Lock (&data, NULL, &buffer_len);
+ if (!gst_mf_result (hr))
+ return GST_FLOW_ERROR;
+
+ buffer = gst_audio_encoder_allocate_output_buffer (GST_AUDIO_ENCODER (self),
+ buffer_len);
+ gst_buffer_fill (buffer, 0, data, buffer_len);
+ media_buffer->Unlock ();
+
+ return gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (self), buffer,
+ klass->frame_samples);
+}
+
+static GstFlowReturn
+gst_mf_audio_enc_handle_frame (GstAudioEncoder * enc,
+ GstBuffer *buffer)
+{
+ GstMFAudioEnc *self = GST_MF_AUDIO_ENC (enc);
+ GstFlowReturn ret;
+
+ if (!buffer)
+ return gst_mf_audio_enc_drain (enc);
+
+ if (!gst_mf_audio_enc_process_input (self, buffer)) {
+ GST_ERROR_OBJECT (self, "Failed to process input");
+ return GST_FLOW_ERROR;
+ }
+
+ do {
+ ret = gst_mf_audio_enc_process_output (self);
+ } while (ret == GST_FLOW_OK);
+
+ if (ret == GST_MF_TRANSFORM_FLOW_NEED_DATA)
+ ret = GST_FLOW_OK;
+
+ return ret;
+}
+
+static GstFlowReturn
+gst_mf_audio_enc_drain (GstAudioEncoder * enc)
+{
+ GstMFAudioEnc *self = GST_MF_AUDIO_ENC (enc);
+ GstFlowReturn ret = GST_FLOW_OK;
+
+ if (!self->transform)
+ return GST_FLOW_OK;
+
+ gst_mf_transform_drain (self->transform);
+
+ do {
+ ret = gst_mf_audio_enc_process_output (self);
+ } while (ret == GST_FLOW_OK);
+
+ if (ret == GST_MF_TRANSFORM_FLOW_NEED_DATA)
+ ret = GST_FLOW_OK;
+
+ return ret;
+}
+
+static void
+gst_mf_audio_enc_flush (GstAudioEncoder * enc)
+{
+ GstMFAudioEnc *self = GST_MF_AUDIO_ENC (enc);
+
+ if (!self->transform)
+ return;
+
+ gst_mf_transform_flush (self->transform);
+}
projects / platform / upstream / gstreamer.git / commitdiff