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761 lines
22 KiB
C
761 lines
22 KiB
C
/* GStreamer FAAC (Free AAC Encoder) plugin
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* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
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* Copyright (C) 2009 Mark Nauwelaerts <mnauw@users.sourceforge.net>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-faac
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* @see_also: faad
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*
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* faac encodes raw audio to AAC (MPEG-4 part 3) streams.
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch audiotestsrc wave=sine num-buffers=100 ! audioconvert ! faac ! matroskamux ! filesink location=sine.mkv
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* ]| Encode a sine beep as aac and write to matroska container.
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* </refsect2>
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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 <stdlib.h>
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#include <string.h>
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#include <gst/audio/multichannel.h>
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#include <gst/pbutils/codec-utils.h>
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#include "gstfaac.h"
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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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#define SINK_CAPS \
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"audio/x-raw-int, " \
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"endianness = (int) BYTE_ORDER, " \
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"signed = (boolean) true, " \
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"width = (int) 16, " \
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"depth = (int) 16, " \
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"rate = (int) {" SAMPLE_RATES "}, " \
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"channels = (int) [ 1, 6 ] "
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/* these don't seem to work? */
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#if 0
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"audio/x-raw-int, "
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"endianness = (int) BYTE_ORDER, "
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"signed = (boolean) true, "
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"width = (int) 32, "
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"depth = (int) { 24, 32 }, "
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"rate = (int) [ 8000, 96000], "
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"channels = (int) [ 1, 6]; "
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"audio/x-raw-float, "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) 32, "
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"rate = (int) [ 8000, 96000], " "channels = (int) [ 1, 6]"
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#endif
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#define SRC_CAPS \
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"audio/mpeg, " \
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"mpegversion = (int) 4, " \
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"channels = (int) [ 1, 6 ], " \
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"rate = (int) {" SAMPLE_RATES "}, " \
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"stream-format = (string) { adts, raw }, " \
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"base-profile = (string) { main, lc, ssr, ltp }; " \
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"audio/mpeg, " \
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"mpegversion = (int) 2, " \
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"channels = (int) [ 1, 6 ], " \
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"rate = (int) {" SAMPLE_RATES "}, " \
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"stream-format = (string) { adts, raw }, " \
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"profile = (string) { main, lc }"
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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 (SRC_CAPS));
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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 (SINK_CAPS));
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enum
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{
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PROP_0,
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PROP_QUALITY,
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PROP_BITRATE,
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PROP_RATE_CONTROL,
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PROP_PROFILE,
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PROP_TNS,
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PROP_MIDSIDE,
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PROP_SHORTCTL
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};
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enum
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{
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VBR = 1,
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ABR
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};
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static void gst_faac_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_faac_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static gboolean gst_faac_configure_source_pad (GstFaac * faac);
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static GstCaps *gst_faac_getcaps (GstAudioEncoder * enc);
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static gboolean gst_faac_start (GstAudioEncoder * enc);
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static gboolean gst_faac_stop (GstAudioEncoder * enc);
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static gboolean gst_faac_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_faac_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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GST_DEBUG_CATEGORY_STATIC (faac_debug);
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#define GST_CAT_DEFAULT faac_debug
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#define FAAC_DEFAULT_QUALITY 100
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#define FAAC_DEFAULT_BITRATE 128 * 1000
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#define FAAC_DEFAULT_RATE_CONTROL VBR
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#define FAAC_DEFAULT_TNS FALSE
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#define FAAC_DEFAULT_MIDSIDE TRUE
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#define FAAC_DEFAULT_SHORTCTL SHORTCTL_NORMAL
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GST_BOILERPLATE (GstFaac, gst_faac, GstAudioEncoder, GST_TYPE_AUDIO_ENCODER);
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static void
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gst_faac_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_set_details_simple (element_class, "AAC audio encoder",
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"Codec/Encoder/Audio",
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"Free MPEG-2/4 AAC encoder",
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"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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GST_DEBUG_CATEGORY_INIT (faac_debug, "faac", 0, "AAC encoding");
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}
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#define GST_TYPE_FAAC_RATE_CONTROL (gst_faac_brtype_get_type ())
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static GType
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gst_faac_brtype_get_type (void)
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{
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static GType gst_faac_brtype_type = 0;
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if (!gst_faac_brtype_type) {
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static GEnumValue gst_faac_brtype[] = {
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{VBR, "VBR", "VBR encoding"},
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{ABR, "ABR", "ABR encoding"},
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{0, NULL, NULL},
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};
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gst_faac_brtype_type = g_enum_register_static ("GstFaacBrtype",
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gst_faac_brtype);
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}
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return gst_faac_brtype_type;
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}
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#define GST_TYPE_FAAC_SHORTCTL (gst_faac_shortctl_get_type ())
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static GType
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gst_faac_shortctl_get_type (void)
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{
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static GType gst_faac_shortctl_type = 0;
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if (!gst_faac_shortctl_type) {
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static GEnumValue gst_faac_shortctl[] = {
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{SHORTCTL_NORMAL, "SHORTCTL_NORMAL", "Normal block type"},
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{SHORTCTL_NOSHORT, "SHORTCTL_NOSHORT", "No short blocks"},
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{SHORTCTL_NOLONG, "SHORTCTL_NOLONG", "No long blocks"},
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{0, NULL, NULL},
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};
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gst_faac_shortctl_type = g_enum_register_static ("GstFaacShortCtl",
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gst_faac_shortctl);
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}
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return gst_faac_shortctl_type;
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}
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static void
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gst_faac_class_init (GstFaacClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->set_property = gst_faac_set_property;
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gobject_class->get_property = gst_faac_get_property;
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base_class->start = GST_DEBUG_FUNCPTR (gst_faac_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_faac_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_faac_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_faac_handle_frame);
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base_class->getcaps = GST_DEBUG_FUNCPTR (gst_faac_getcaps);
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/* properties */
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g_object_class_install_property (gobject_class, PROP_QUALITY,
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g_param_spec_int ("quality", "Quality (%)",
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"Variable bitrate (VBR) quantizer quality in %", 1, 1000,
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FAAC_DEFAULT_QUALITY,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_BITRATE,
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g_param_spec_int ("bitrate", "Bitrate (bps)",
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"Average Bitrate (ABR) in bits/sec", 8 * 1000, 320 * 1000,
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FAAC_DEFAULT_BITRATE,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_RATE_CONTROL,
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g_param_spec_enum ("rate-control", "Rate Control (ABR/VBR)",
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"Encoding bitrate type (VBR/ABR)", GST_TYPE_FAAC_RATE_CONTROL,
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FAAC_DEFAULT_RATE_CONTROL,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_TNS,
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g_param_spec_boolean ("tns", "TNS", "Use temporal noise shaping",
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FAAC_DEFAULT_TNS,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_MIDSIDE,
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g_param_spec_boolean ("midside", "Midside", "Allow mid/side encoding",
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FAAC_DEFAULT_MIDSIDE,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_SHORTCTL,
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g_param_spec_enum ("shortctl", "Block type",
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"Block type encorcing",
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GST_TYPE_FAAC_SHORTCTL, FAAC_DEFAULT_SHORTCTL,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_faac_init (GstFaac * faac, GstFaacClass * klass)
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{
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}
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static void
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gst_faac_close_encoder (GstFaac * faac)
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{
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if (faac->handle)
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faacEncClose (faac->handle);
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faac->handle = NULL;
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}
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static gboolean
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gst_faac_start (GstAudioEncoder * enc)
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{
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GstFaac *faac = GST_FAAC (enc);
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GST_DEBUG_OBJECT (faac, "start");
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return TRUE;
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}
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static gboolean
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gst_faac_stop (GstAudioEncoder * enc)
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{
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GstFaac *faac = GST_FAAC (enc);
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GST_DEBUG_OBJECT (faac, "stop");
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gst_faac_close_encoder (faac);
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return TRUE;
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}
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static const GstAudioChannelPosition aac_channel_positions[][8] = {
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{GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
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{GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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{
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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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},
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{
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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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{
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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_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
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{
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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_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE}
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};
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static GstCaps *
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gst_faac_getcaps (GstAudioEncoder * enc)
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{
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static volatile gsize sinkcaps = 0;
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if (g_once_init_enter (&sinkcaps)) {
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GstCaps *tmp = gst_caps_new_empty ();
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GstStructure *s, *t;
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gint i, c;
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static const int rates[] = {
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8000, 11025, 12000, 16000, 22050, 24000,
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32000, 44100, 48000, 64000, 88200, 96000
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};
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GValue rates_arr = { 0, };
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GValue tmp_v = { 0, };
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g_value_init (&rates_arr, GST_TYPE_LIST);
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g_value_init (&tmp_v, G_TYPE_INT);
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for (i = 0; i < G_N_ELEMENTS (rates); i++) {
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g_value_set_int (&tmp_v, rates[i]);
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gst_value_list_append_value (&rates_arr, &tmp_v);
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}
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g_value_unset (&tmp_v);
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s = gst_structure_new ("audio/x-raw-int",
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"signed", G_TYPE_BOOLEAN, TRUE,
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"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
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gst_structure_set_value (s, "rate", &rates_arr);
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for (i = 1; i <= 6; i++) {
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GValue chanpos = { 0 };
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GValue pos = { 0 };
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t = gst_structure_copy (s);
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gst_structure_set (t, "channels", G_TYPE_INT, i, NULL);
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g_value_init (&chanpos, GST_TYPE_ARRAY);
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g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
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for (c = 0; c < i; c++) {
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g_value_set_enum (&pos, aac_channel_positions[i - 1][c]);
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gst_value_array_append_value (&chanpos, &pos);
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}
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g_value_unset (&pos);
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gst_structure_set_value (t, "channel-positions", &chanpos);
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g_value_unset (&chanpos);
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gst_caps_append_structure (tmp, t);
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}
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gst_structure_free (s);
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g_value_unset (&rates_arr);
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GST_DEBUG_OBJECT (enc, "Generated sinkcaps: %" GST_PTR_FORMAT, tmp);
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g_once_init_leave (&sinkcaps, (gsize) tmp);
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}
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return gst_audio_encoder_proxy_getcaps (enc, (GstCaps *) sinkcaps);
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}
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static gboolean
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gst_faac_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
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{
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GstFaac *faac = GST_FAAC (enc);
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faacEncHandle *handle;
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gint channels, samplerate, width;
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gulong samples, bytes, fmt = 0, bps = 0;
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gboolean result = FALSE;
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/* base class takes care */
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channels = GST_AUDIO_INFO_CHANNELS (info);
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samplerate = GST_AUDIO_INFO_RATE (info);
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width = GST_AUDIO_INFO_WIDTH (info);
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if (GST_AUDIO_INFO_IS_INTEGER (info)) {
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switch (width) {
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case 16:
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fmt = FAAC_INPUT_16BIT;
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bps = 2;
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break;
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case 24:
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case 32:
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fmt = FAAC_INPUT_32BIT;
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bps = 4;
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break;
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default:
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g_return_val_if_reached (FALSE);
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}
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} else {
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fmt = FAAC_INPUT_FLOAT;
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bps = 4;
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}
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/* clean up in case of re-configure */
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gst_faac_close_encoder (faac);
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if (!(handle = faacEncOpen (samplerate, channels, &samples, &bytes)))
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goto setup_failed;
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/* ok, record and set up */
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faac->format = fmt;
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faac->bps = bps;
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faac->handle = handle;
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faac->bytes = bytes;
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faac->samples = samples;
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faac->channels = channels;
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faac->samplerate = samplerate;
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/* finish up */
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result = gst_faac_configure_source_pad (faac);
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/* report needs to base class */
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gst_audio_encoder_set_frame_samples (enc, samples);
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gst_audio_encoder_set_frame_max (enc, 1);
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done:
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return result;
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/* ERRORS */
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setup_failed:
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{
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GST_ELEMENT_ERROR (faac, LIBRARY, SETTINGS, (NULL), (NULL));
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goto done;
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}
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}
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/* check downstream caps to configure format */
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static void
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gst_faac_negotiate (GstFaac * faac)
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{
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GstCaps *caps;
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/* default setup */
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faac->profile = LOW;
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faac->mpegversion = 4;
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faac->outputformat = 0;
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caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (faac));
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GST_DEBUG_OBJECT (faac, "allowed caps: %" GST_PTR_FORMAT, caps);
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if (caps && gst_caps_get_size (caps) > 0) {
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GstStructure *s = gst_caps_get_structure (caps, 0);
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const gchar *str = NULL;
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gint i = 4;
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|
|
if ((str = gst_structure_get_string (s, "stream-format"))) {
|
|
if (strcmp (str, "adts") == 0) {
|
|
GST_DEBUG_OBJECT (faac, "use ADTS format for output");
|
|
faac->outputformat = 1;
|
|
} else if (strcmp (str, "raw") == 0) {
|
|
GST_DEBUG_OBJECT (faac, "use RAW format for output");
|
|
faac->outputformat = 0;
|
|
} else {
|
|
GST_DEBUG_OBJECT (faac, "unknown stream-format: %s", str);
|
|
faac->outputformat = 0;
|
|
}
|
|
}
|
|
|
|
if ((str = gst_structure_get_string (s, "profile"))) {
|
|
if (strcmp (str, "main") == 0) {
|
|
faac->profile = MAIN;
|
|
} else if (strcmp (str, "lc") == 0) {
|
|
faac->profile = LOW;
|
|
} else if (strcmp (str, "ssr") == 0) {
|
|
faac->profile = SSR;
|
|
} else if (strcmp (str, "ltp") == 0) {
|
|
faac->profile = LTP;
|
|
} else {
|
|
faac->profile = LOW;
|
|
}
|
|
}
|
|
|
|
if (!gst_structure_get_int (s, "mpegversion", &i) || i == 4) {
|
|
faac->mpegversion = 4;
|
|
} else {
|
|
faac->mpegversion = 2;
|
|
}
|
|
}
|
|
|
|
if (caps)
|
|
gst_caps_unref (caps);
|
|
}
|
|
|
|
static gboolean
|
|
gst_faac_configure_source_pad (GstFaac * faac)
|
|
{
|
|
GstCaps *srccaps;
|
|
gboolean ret = FALSE;
|
|
faacEncConfiguration *conf;
|
|
guint maxbitrate;
|
|
|
|
/* negotiate stream format */
|
|
gst_faac_negotiate (faac);
|
|
|
|
/* we negotiated caps update current configuration */
|
|
conf = faacEncGetCurrentConfiguration (faac->handle);
|
|
conf->mpegVersion = (faac->mpegversion == 4) ? MPEG4 : MPEG2;
|
|
conf->aacObjectType = faac->profile;
|
|
conf->allowMidside = faac->midside;
|
|
conf->useLfe = 0;
|
|
conf->useTns = faac->tns;
|
|
|
|
if (faac->brtype == VBR) {
|
|
conf->quantqual = faac->quality;
|
|
} else if (faac->brtype == ABR) {
|
|
conf->bitRate = faac->bitrate / faac->channels;
|
|
}
|
|
|
|
conf->inputFormat = faac->format;
|
|
conf->outputFormat = faac->outputformat;
|
|
conf->shortctl = faac->shortctl;
|
|
|
|
/* check, warn and correct if the max bitrate for the given samplerate is
|
|
* exceeded. Maximum of 6144 bit for a channel */
|
|
maxbitrate =
|
|
(unsigned int) (6144.0 * (double) faac->samplerate / (double) 1024.0 +
|
|
.5);
|
|
if (conf->bitRate > maxbitrate) {
|
|
GST_ELEMENT_WARNING (faac, RESOURCE, SETTINGS, (NULL),
|
|
("bitrate %lu exceeds maximum allowed bitrate of %u for samplerate %d. "
|
|
"Setting bitrate to %u", conf->bitRate, maxbitrate,
|
|
faac->samplerate, maxbitrate));
|
|
conf->bitRate = maxbitrate;
|
|
}
|
|
|
|
/* default 0 to start with, libfaac chooses based on bitrate */
|
|
conf->bandWidth = 0;
|
|
|
|
if (!faacEncSetConfiguration (faac->handle, conf))
|
|
goto set_failed;
|
|
|
|
/* let's see what really happened,
|
|
* note that this may not really match desired rate */
|
|
GST_DEBUG_OBJECT (faac, "average bitrate: %lu kbps",
|
|
(conf->bitRate + 500) / 1000 * faac->channels);
|
|
GST_DEBUG_OBJECT (faac, "quantization quality: %ld", conf->quantqual);
|
|
GST_DEBUG_OBJECT (faac, "bandwidth: %d Hz", conf->bandWidth);
|
|
|
|
/* now create a caps for it all */
|
|
srccaps = gst_caps_new_simple ("audio/mpeg",
|
|
"mpegversion", G_TYPE_INT, faac->mpegversion,
|
|
"channels", G_TYPE_INT, faac->channels,
|
|
"rate", G_TYPE_INT, faac->samplerate,
|
|
"stream-format", G_TYPE_STRING, (faac->outputformat ? "adts" : "raw"),
|
|
NULL);
|
|
|
|
/* DecoderSpecificInfo is only available for mpegversion=4 */
|
|
if (faac->mpegversion == 4) {
|
|
guint8 *config = NULL;
|
|
gulong config_len = 0;
|
|
|
|
/* get the config string */
|
|
GST_DEBUG_OBJECT (faac, "retrieving decoder info");
|
|
faacEncGetDecoderSpecificInfo (faac->handle, &config, &config_len);
|
|
|
|
if (!gst_codec_utils_aac_caps_set_level_and_profile (srccaps, config,
|
|
config_len)) {
|
|
free (config);
|
|
gst_caps_unref (srccaps);
|
|
goto invalid_codec_data;
|
|
}
|
|
|
|
if (!faac->outputformat) {
|
|
GstBuffer *codec_data;
|
|
|
|
/* copy it into a buffer */
|
|
codec_data = gst_buffer_new_and_alloc (config_len);
|
|
memcpy (GST_BUFFER_DATA (codec_data), config, config_len);
|
|
|
|
/* add to caps */
|
|
gst_caps_set_simple (srccaps,
|
|
"codec_data", GST_TYPE_BUFFER, codec_data, NULL);
|
|
|
|
gst_buffer_unref (codec_data);
|
|
}
|
|
|
|
free (config);
|
|
} else {
|
|
const gchar *profile;
|
|
|
|
/* Add least add the profile to the caps */
|
|
switch (faac->profile) {
|
|
case MAIN:
|
|
profile = "main";
|
|
break;
|
|
case LTP:
|
|
profile = "ltp";
|
|
break;
|
|
case SSR:
|
|
profile = "ssr";
|
|
break;
|
|
case LOW:
|
|
default:
|
|
profile = "lc";
|
|
break;
|
|
}
|
|
gst_caps_set_simple (srccaps, "profile", G_TYPE_STRING, profile, NULL);
|
|
/* FIXME: How to get the profile for mpegversion==2? */
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (faac, "src pad caps: %" GST_PTR_FORMAT, srccaps);
|
|
|
|
ret = gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (faac), srccaps);
|
|
gst_caps_unref (srccaps);
|
|
|
|
return ret;
|
|
|
|
/* ERROR */
|
|
set_failed:
|
|
{
|
|
GST_WARNING_OBJECT (faac, "Faac doesn't support the current configuration");
|
|
return FALSE;
|
|
}
|
|
invalid_codec_data:
|
|
{
|
|
GST_ERROR_OBJECT (faac, "Invalid codec data");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_faac_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
|
|
{
|
|
GstFaac *faac = GST_FAAC (enc);
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstBuffer *out_buf;
|
|
gint size, ret_size;
|
|
const guint8 *data;
|
|
|
|
out_buf = gst_buffer_new_and_alloc (faac->bytes);
|
|
|
|
if (G_LIKELY (in_buf)) {
|
|
data = GST_BUFFER_DATA (in_buf);
|
|
size = GST_BUFFER_SIZE (in_buf);
|
|
} else {
|
|
data = NULL;
|
|
size = 0;
|
|
}
|
|
|
|
if (G_UNLIKELY ((ret_size = faacEncEncode (faac->handle, (gint32 *) data,
|
|
size / faac->bps, GST_BUFFER_DATA (out_buf),
|
|
GST_BUFFER_SIZE (out_buf))) < 0))
|
|
goto encode_failed;
|
|
|
|
GST_LOG_OBJECT (faac, "encoder return: %d", ret_size);
|
|
if (ret_size > 0) {
|
|
GST_BUFFER_SIZE (out_buf) = ret_size;
|
|
ret = gst_audio_encoder_finish_frame (enc, out_buf, faac->samples);
|
|
}
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
encode_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (faac, LIBRARY, ENCODE, (NULL), (NULL));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_faac_set_property (GObject * object,
|
|
guint prop_id, const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstFaac *faac = GST_FAAC (object);
|
|
|
|
GST_OBJECT_LOCK (faac);
|
|
|
|
switch (prop_id) {
|
|
case PROP_QUALITY:
|
|
faac->quality = g_value_get_int (value);
|
|
break;
|
|
case PROP_BITRATE:
|
|
faac->bitrate = g_value_get_int (value);
|
|
break;
|
|
case PROP_RATE_CONTROL:
|
|
faac->brtype = g_value_get_enum (value);
|
|
break;
|
|
case PROP_TNS:
|
|
faac->tns = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_MIDSIDE:
|
|
faac->midside = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_SHORTCTL:
|
|
faac->shortctl = g_value_get_enum (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
|
|
GST_OBJECT_UNLOCK (faac);
|
|
}
|
|
|
|
static void
|
|
gst_faac_get_property (GObject * object,
|
|
guint prop_id, GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstFaac *faac = GST_FAAC (object);
|
|
|
|
GST_OBJECT_LOCK (faac);
|
|
|
|
switch (prop_id) {
|
|
case PROP_QUALITY:
|
|
g_value_set_int (value, faac->quality);
|
|
break;
|
|
case PROP_BITRATE:
|
|
g_value_set_int (value, faac->bitrate);
|
|
break;
|
|
case PROP_RATE_CONTROL:
|
|
g_value_set_enum (value, faac->brtype);
|
|
break;
|
|
case PROP_TNS:
|
|
g_value_set_boolean (value, faac->tns);
|
|
break;
|
|
case PROP_MIDSIDE:
|
|
g_value_set_boolean (value, faac->midside);
|
|
break;
|
|
case PROP_SHORTCTL:
|
|
g_value_set_enum (value, faac->shortctl);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
|
|
GST_OBJECT_UNLOCK (faac);
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "faac", GST_RANK_SECONDARY,
|
|
GST_TYPE_FAAC);
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
"faac",
|
|
"Free AAC Encoder (FAAC)",
|
|
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
|