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611 lines
19 KiB
C
611 lines
19 KiB
C
/*
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* Copyright (C) 2016 Sebastian Dröge <sebastian@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstfdkaacenc.h"
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#include <gst/pbutils/pbutils.h>
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#include <string.h>
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/* TODO:
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* - Add support for other AOT / profiles
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* - Expose more properties, e.g. afterburner and vbr
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* - Signal encoder delay
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* - LOAS / LATM support
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*/
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enum
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{
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PROP_0,
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PROP_BITRATE
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};
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#define DEFAULT_BITRATE (0)
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#define SAMPLE_RATES " 8000, " \
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"11025, " \
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"12000, " \
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"16000, " \
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"22050, " \
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"24000, " \
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"32000, " \
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"44100, " \
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"48000, " \
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"64000, " \
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"88200, " \
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"96000"
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static const struct
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{
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gint channels;
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CHANNEL_MODE mode;
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GstAudioChannelPosition positions[8];
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} channel_layouts[] = {
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{
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1, MODE_1, {
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GST_AUDIO_CHANNEL_POSITION_MONO}}, {
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2, MODE_2, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}}, {
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3, MODE_1_2, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}}, {
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3, MODE_2_1, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE1}}, {
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4, MODE_1_2_1, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}}, {
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5, MODE_1_2_2, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}}, {
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6, MODE_1_2_2_1, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE1}}, {
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8, MODE_7_1_REAR_SURROUND, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE1}}, {
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8, MODE_7_1_FRONT_CENTER, {
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER,
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE1}}
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};
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"rate = (int) { " SAMPLE_RATES " }, "
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"channels = (int) {1, 2, 3, 4, 5, 6, 8}")
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);
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/mpeg, "
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"mpegversion = (int) 4, "
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"rate = (int) { " SAMPLE_RATES " }, "
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"channels = (int) {1, 2, 3, 4, 5, 6, 8}, "
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"stream-format = (string) { adts, adif, raw }, "
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"base-profile = (string) lc, " "framed = (boolean) true")
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);
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GST_DEBUG_CATEGORY_STATIC (gst_fdkaacenc_debug);
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#define GST_CAT_DEFAULT gst_fdkaacenc_debug
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static void gst_fdkaacenc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_fdkaacenc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_fdkaacenc_start (GstAudioEncoder * enc);
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static gboolean gst_fdkaacenc_stop (GstAudioEncoder * enc);
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static gboolean gst_fdkaacenc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_fdkaacenc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static GstCaps *gst_fdkaacenc_get_caps (GstAudioEncoder * enc,
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GstCaps * filter);
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G_DEFINE_TYPE (GstFdkAacEnc, gst_fdkaacenc, GST_TYPE_AUDIO_ENCODER);
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static void
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gst_fdkaacenc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstFdkAacEnc *self = GST_FDKAACENC (object);
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switch (prop_id) {
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case PROP_BITRATE:
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self->bitrate = g_value_get_int (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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return;
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}
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static void
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gst_fdkaacenc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstFdkAacEnc *self = GST_FDKAACENC (object);
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switch (prop_id) {
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case PROP_BITRATE:
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g_value_set_int (value, self->bitrate);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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return;
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}
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static gboolean
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gst_fdkaacenc_start (GstAudioEncoder * enc)
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{
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GstFdkAacEnc *self = GST_FDKAACENC (enc);
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GST_DEBUG_OBJECT (self, "start");
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return TRUE;
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}
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static gboolean
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gst_fdkaacenc_stop (GstAudioEncoder * enc)
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{
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GstFdkAacEnc *self = GST_FDKAACENC (enc);
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GST_DEBUG_OBJECT (self, "stop");
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if (self->enc)
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aacEncClose (&self->enc);
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return TRUE;
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}
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static GstCaps *
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gst_fdkaacenc_get_caps (GstAudioEncoder * enc, GstCaps * filter)
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{
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GstCaps *res, *caps;
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gint i;
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caps = gst_caps_new_empty ();
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for (i = 0; i < G_N_ELEMENTS (channel_layouts); i++) {
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guint64 channel_mask;
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GstCaps *tmp =
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gst_caps_make_writable (gst_pad_get_pad_template_caps
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(GST_AUDIO_ENCODER_SINK_PAD (enc)));
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if (channel_layouts[i].channels == 1) {
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gst_caps_set_simple (tmp, "channels", G_TYPE_INT,
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channel_layouts[i].channels, NULL);
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} else {
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gst_audio_channel_positions_to_mask (channel_layouts[i].positions,
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channel_layouts[i].channels, FALSE, &channel_mask);
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gst_caps_set_simple (tmp, "channels", G_TYPE_INT,
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channel_layouts[i].channels, "channel-mask", GST_TYPE_BITMASK,
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channel_mask, NULL);
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}
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gst_caps_append (caps, tmp);
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}
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res = gst_audio_encoder_proxy_getcaps (enc, caps, filter);
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gst_caps_unref (caps);
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return res;
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}
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static gboolean
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gst_fdkaacenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
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{
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GstFdkAacEnc *self = GST_FDKAACENC (enc);
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gboolean ret = FALSE;
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GstCaps *allowed_caps;
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GstCaps *src_caps;
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AACENC_ERROR err;
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gint transmux = 0, aot = AOT_AAC_LC;
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gint mpegversion = 4;
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CHANNEL_MODE channel_mode;
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AACENC_InfoStruct enc_info = { 0 };
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gint bitrate;
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if (self->enc) {
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/* drain */
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gst_fdkaacenc_handle_frame (enc, NULL);
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aacEncClose (&self->enc);
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}
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allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (self));
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GST_DEBUG_OBJECT (self, "allowed caps: %" GST_PTR_FORMAT, allowed_caps);
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if (allowed_caps && gst_caps_get_size (allowed_caps) > 0) {
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GstStructure *s = gst_caps_get_structure (allowed_caps, 0);
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const gchar *str = NULL;
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if ((str = gst_structure_get_string (s, "stream-format"))) {
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if (strcmp (str, "adts") == 0) {
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GST_DEBUG_OBJECT (self, "use ADTS format for output");
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transmux = 2;
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} else if (strcmp (str, "adif") == 0) {
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GST_DEBUG_OBJECT (self, "use ADIF format for output");
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transmux = 1;
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} else if (strcmp (str, "raw") == 0) {
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GST_DEBUG_OBJECT (self, "use RAW format for output");
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transmux = 0;
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}
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}
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gst_structure_get_int (s, "mpegversion", &mpegversion);
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}
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if (allowed_caps)
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gst_caps_unref (allowed_caps);
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if ((err =
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aacEncOpen (&self->enc, 0,
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GST_AUDIO_INFO_CHANNELS (info))) != AACENC_OK) {
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GST_ERROR_OBJECT (self, "Unable to open encoder: %d\n", err);
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return FALSE;
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}
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aot = AOT_AAC_LC;
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if ((err = aacEncoder_SetParam (self->enc, AACENC_AOT, aot)) != AACENC_OK) {
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GST_ERROR_OBJECT (self, "Unable to set AOT %d: %d\n", aot, err);
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return FALSE;
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}
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if ((err = aacEncoder_SetParam (self->enc, AACENC_SAMPLERATE,
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GST_AUDIO_INFO_RATE (info))) != AACENC_OK) {
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GST_ERROR_OBJECT (self, "Unable to set sample rate %d: %d\n",
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GST_AUDIO_INFO_RATE (info), err);
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return FALSE;
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}
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if (GST_AUDIO_INFO_CHANNELS (info) == 1) {
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channel_mode = MODE_1;
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self->need_reorder = FALSE;
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self->aac_positions = NULL;
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} else {
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guint64 in_channel_mask, out_channel_mask;
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gint i;
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for (i = 0; i < G_N_ELEMENTS (channel_layouts); i++) {
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if (channel_layouts[i].channels != GST_AUDIO_INFO_CHANNELS (info))
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continue;
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gst_audio_channel_positions_to_mask (&GST_AUDIO_INFO_POSITION (info, 0),
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GST_AUDIO_INFO_CHANNELS (info), FALSE, &in_channel_mask);
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gst_audio_channel_positions_to_mask (channel_layouts[i].positions,
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channel_layouts[i].channels, FALSE, &out_channel_mask);
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if (in_channel_mask == out_channel_mask) {
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channel_mode = channel_layouts[i].mode;
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self->need_reorder =
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memcmp (channel_layouts[i].positions,
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&GST_AUDIO_INFO_POSITION (info, 0),
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GST_AUDIO_INFO_CHANNELS (info) *
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sizeof (GstAudioChannelPosition)) != 0;
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self->aac_positions = channel_layouts[i].positions;
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break;
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}
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}
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if (i == G_N_ELEMENTS (channel_layouts)) {
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GST_ERROR_OBJECT (self, "Couldn't find a valid channel layout");
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return FALSE;
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}
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}
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if ((err = aacEncoder_SetParam (self->enc, AACENC_CHANNELMODE,
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channel_mode)) != AACENC_OK) {
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GST_ERROR_OBJECT (self, "Unable to set channel mode %d: %d", channel_mode,
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err);
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return FALSE;
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}
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/* MPEG channel order */
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if ((err = aacEncoder_SetParam (self->enc, AACENC_CHANNELORDER,
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0)) != AACENC_OK) {
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GST_ERROR_OBJECT (self, "Unable to set channel order %d: %d", channel_mode,
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err);
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return FALSE;
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}
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bitrate = self->bitrate;
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/* See
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* http://wiki.hydrogenaud.io/index.php?title=Fraunhofer_FDK_AAC#Recommended_Sampling_Rate_and_Bitrate_Combinations
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*/
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if (bitrate == 0) {
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if (GST_AUDIO_INFO_CHANNELS (info) == 1) {
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if (GST_AUDIO_INFO_RATE (info) < 16000) {
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bitrate = 8000;
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} else if (GST_AUDIO_INFO_RATE (info) == 16000) {
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bitrate = 16000;
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} else if (GST_AUDIO_INFO_RATE (info) < 32000) {
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bitrate = 24000;
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} else if (GST_AUDIO_INFO_RATE (info) == 32000) {
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bitrate = 32000;
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} else if (GST_AUDIO_INFO_RATE (info) <= 44100) {
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bitrate = 56000;
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} else {
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bitrate = 160000;
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}
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} else if (GST_AUDIO_INFO_CHANNELS (info) == 2) {
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if (GST_AUDIO_INFO_RATE (info) < 16000) {
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bitrate = 16000;
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} else if (GST_AUDIO_INFO_RATE (info) == 16000) {
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bitrate = 24000;
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} else if (GST_AUDIO_INFO_RATE (info) < 22050) {
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bitrate = 32000;
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} else if (GST_AUDIO_INFO_RATE (info) < 32000) {
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bitrate = 40000;
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} else if (GST_AUDIO_INFO_RATE (info) == 32000) {
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bitrate = 96000;
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} else if (GST_AUDIO_INFO_RATE (info) <= 44100) {
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bitrate = 112000;
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} else {
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bitrate = 320000;
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}
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} else {
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/* 5, 5.1 */
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if (GST_AUDIO_INFO_RATE (info) < 32000) {
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bitrate = 160000;
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} else if (GST_AUDIO_INFO_RATE (info) <= 44100) {
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bitrate = 240000;
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} else {
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bitrate = 320000;
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}
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}
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}
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if ((err = aacEncoder_SetParam (self->enc, AACENC_TRANSMUX,
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transmux)) != AACENC_OK) {
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GST_ERROR_OBJECT (self, "Unable to set transmux %d: %d", transmux, err);
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return FALSE;
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}
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if ((err = aacEncoder_SetParam (self->enc, AACENC_BITRATE,
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bitrate)) != AACENC_OK) {
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GST_ERROR_OBJECT (self, "Unable to set bitrate %d: %d", bitrate, err);
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return FALSE;
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}
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if ((err = aacEncEncode (self->enc, NULL, NULL, NULL, NULL)) != AACENC_OK) {
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GST_ERROR_OBJECT (self, "Unable to initialize encoder: %d", err);
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return FALSE;
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}
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if ((err = aacEncInfo (self->enc, &enc_info)) != AACENC_OK) {
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GST_ERROR_OBJECT (self, "Unable to get encoder info: %d", err);
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return FALSE;
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}
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gst_audio_encoder_set_frame_max (enc, 1);
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gst_audio_encoder_set_frame_samples_min (enc, enc_info.frameLength);
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gst_audio_encoder_set_frame_samples_max (enc, enc_info.frameLength);
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gst_audio_encoder_set_hard_min (enc, FALSE);
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self->outbuf_size = enc_info.maxOutBufBytes;
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self->samples_per_frame = enc_info.frameLength;
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src_caps = gst_caps_new_simple ("audio/mpeg",
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"mpegversion", G_TYPE_INT, mpegversion,
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"channels", G_TYPE_INT, GST_AUDIO_INFO_CHANNELS (info),
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"framed", G_TYPE_BOOLEAN, TRUE,
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"rate", G_TYPE_INT, GST_AUDIO_INFO_RATE (info), NULL);
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/* raw */
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if (transmux == 0) {
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GstBuffer *codec_data =
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gst_buffer_new_wrapped (g_memdup (enc_info.confBuf, enc_info.confSize),
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enc_info.confSize);
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gst_caps_set_simple (src_caps, "codec_data", GST_TYPE_BUFFER, codec_data,
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"stream-format", G_TYPE_STRING, "raw", NULL);
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gst_buffer_unref (codec_data);
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} else if (transmux == 1) {
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gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "adif",
|
|
NULL);
|
|
} else if (transmux == 2) {
|
|
gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "adts",
|
|
NULL);
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
gst_codec_utils_aac_caps_set_level_and_profile (src_caps, enc_info.confBuf,
|
|
enc_info.confSize);
|
|
|
|
ret = gst_audio_encoder_set_output_format (enc, src_caps);
|
|
gst_caps_unref (src_caps);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_fdkaacenc_handle_frame (GstAudioEncoder * enc, GstBuffer * inbuf)
|
|
{
|
|
GstFdkAacEnc *self = GST_FDKAACENC (enc);
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstAudioInfo *info;
|
|
GstMapInfo imap, omap;
|
|
GstBuffer *outbuf;
|
|
AACENC_BufDesc in_desc = { 0 };
|
|
AACENC_BufDesc out_desc = { 0 };
|
|
AACENC_InArgs in_args = { 0 };
|
|
AACENC_OutArgs out_args = { 0 };
|
|
gint in_id = IN_AUDIO_DATA, out_id = OUT_BITSTREAM_DATA;
|
|
gint in_sizes, out_sizes;
|
|
gint in_el_sizes, out_el_sizes;
|
|
AACENC_ERROR err;
|
|
|
|
info = gst_audio_encoder_get_audio_info (enc);
|
|
|
|
if (!inbuf) {
|
|
in_args.numInSamples = -1;
|
|
} else {
|
|
if (self->need_reorder) {
|
|
inbuf = gst_buffer_copy (inbuf);
|
|
gst_buffer_map (inbuf, &imap, GST_MAP_READWRITE);
|
|
gst_audio_reorder_channels (imap.data, imap.size,
|
|
GST_AUDIO_INFO_FORMAT (info), GST_AUDIO_INFO_CHANNELS (info),
|
|
&GST_AUDIO_INFO_POSITION (info, 0), self->aac_positions);
|
|
} else {
|
|
gst_buffer_map (inbuf, &imap, GST_MAP_READ);
|
|
}
|
|
|
|
in_args.numInSamples = imap.size / GST_AUDIO_INFO_BPS (info);
|
|
|
|
in_sizes = imap.size;
|
|
in_el_sizes = 2;
|
|
in_desc.bufferIdentifiers = &in_id;
|
|
in_desc.numBufs = 1;
|
|
in_desc.bufs = (void *) &imap.data;
|
|
in_desc.bufSizes = &in_sizes;
|
|
in_desc.bufElSizes = &in_el_sizes;
|
|
}
|
|
|
|
outbuf = gst_audio_encoder_allocate_output_buffer (enc, self->outbuf_size);
|
|
if (!outbuf) {
|
|
ret = GST_FLOW_ERROR;
|
|
goto out;
|
|
}
|
|
|
|
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
|
|
out_sizes = omap.size;
|
|
out_el_sizes = 1;
|
|
out_desc.bufferIdentifiers = &out_id;
|
|
out_desc.numBufs = 1;
|
|
out_desc.bufs = (void *) &omap.data;
|
|
out_desc.bufSizes = &out_sizes;
|
|
out_desc.bufElSizes = &out_el_sizes;
|
|
|
|
if ((err = aacEncEncode (self->enc, &in_desc, &out_desc, &in_args,
|
|
&out_args)) != AACENC_OK) {
|
|
if (!inbuf && err == AACENC_ENCODE_EOF)
|
|
goto out;
|
|
|
|
GST_ERROR_OBJECT (self, "Failed to encode data: %d", err);
|
|
ret = GST_FLOW_ERROR;
|
|
goto out;
|
|
}
|
|
|
|
if (inbuf) {
|
|
gst_buffer_unmap (inbuf, &imap);
|
|
if (self->need_reorder)
|
|
gst_buffer_unref (inbuf);
|
|
inbuf = NULL;
|
|
}
|
|
|
|
if (!out_args.numOutBytes)
|
|
goto out;
|
|
|
|
gst_buffer_unmap (outbuf, &omap);
|
|
gst_buffer_set_size (outbuf, out_args.numOutBytes);
|
|
|
|
ret = gst_audio_encoder_finish_frame (enc, outbuf, self->samples_per_frame);
|
|
outbuf = NULL;
|
|
|
|
out:
|
|
if (outbuf) {
|
|
gst_buffer_unmap (outbuf, &omap);
|
|
gst_buffer_unref (outbuf);
|
|
}
|
|
if (inbuf) {
|
|
gst_buffer_unmap (inbuf, &imap);
|
|
if (self->need_reorder)
|
|
gst_buffer_unref (inbuf);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_fdkaacenc_init (GstFdkAacEnc * self)
|
|
{
|
|
self->bitrate = DEFAULT_BITRATE;
|
|
self->enc = NULL;
|
|
|
|
gst_audio_encoder_set_drainable (GST_AUDIO_ENCODER (self), TRUE);
|
|
}
|
|
|
|
static void
|
|
gst_fdkaacenc_class_init (GstFdkAacEncClass * klass)
|
|
{
|
|
GObjectClass *object_class = G_OBJECT_CLASS (klass);
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
|
|
GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
|
|
|
|
object_class->set_property = GST_DEBUG_FUNCPTR (gst_fdkaacenc_set_property);
|
|
object_class->get_property = GST_DEBUG_FUNCPTR (gst_fdkaacenc_get_property);
|
|
|
|
base_class->start = GST_DEBUG_FUNCPTR (gst_fdkaacenc_start);
|
|
base_class->stop = GST_DEBUG_FUNCPTR (gst_fdkaacenc_stop);
|
|
base_class->set_format = GST_DEBUG_FUNCPTR (gst_fdkaacenc_set_format);
|
|
base_class->getcaps = GST_DEBUG_FUNCPTR (gst_fdkaacenc_get_caps);
|
|
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_fdkaacenc_handle_frame);
|
|
|
|
g_object_class_install_property (object_class, PROP_BITRATE,
|
|
g_param_spec_int ("bitrate",
|
|
"Bitrate",
|
|
"Target Audio Bitrate (0 = fixed value based on "
|
|
" sample rate and channel count)",
|
|
0, G_MAXINT, DEFAULT_BITRATE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
gst_element_class_add_static_pad_template (element_class, &sink_template);
|
|
gst_element_class_add_static_pad_template (element_class, &src_template);
|
|
|
|
gst_element_class_set_static_metadata (element_class, "FDK AAC audio encoder",
|
|
"Codec/Encoder/Audio", "FDK AAC audio encoder",
|
|
"Sebastian Dröge <sebastian@centricular.com>");
|
|
|
|
GST_DEBUG_CATEGORY_INIT (gst_fdkaacenc_debug, "fdkaacenc", 0,
|
|
"fdkaac encoder");
|
|
}
|