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945 lines
27 KiB
C
945 lines
27 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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*
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* The #GstFaac:profile property determines the AAC profile, where the default
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* LC (Low Complexity) profile is most widely used, supported and suitable for
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* general use. The other profiles are very rarely used and often not supported.
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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 "gstfaac.h"
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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) [ 8000, 96000 ], " \
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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, 2 }, " \
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"channels = (int) [ 1, 6 ], " \
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"rate = (int) [ 8000, 96000 ], " \
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"stream-format = (string) { adts, raw } "
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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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ARG_0,
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ARG_BITRATE,
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ARG_PROFILE,
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ARG_TNS,
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ARG_MIDSIDE,
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ARG_SHORTCTL
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};
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static void gst_faac_base_init (GstFaacClass * klass);
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static void gst_faac_class_init (GstFaacClass * klass);
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static void gst_faac_init (GstFaac * faac);
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static void gst_faac_finalize (GObject * object);
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static void gst_faac_reset (GstFaac * faac);
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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_sink_event (GstPad * pad, GstEvent * event);
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static gboolean gst_faac_configure_source_pad (GstFaac * faac);
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static gboolean gst_faac_sink_setcaps (GstPad * pad, GstCaps * caps);
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static GstCaps *gst_faac_sink_getcaps (GstPad * pad);
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static GstFlowReturn gst_faac_push_buffers (GstFaac * faac, gboolean force);
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static GstFlowReturn gst_faac_chain (GstPad * pad, GstBuffer * data);
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static GstStateChangeReturn gst_faac_change_state (GstElement * element,
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GstStateChange transition);
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static GstElementClass *parent_class = NULL;
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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_MPEGVERSION 4
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#define FAAC_DEFAULT_OUTPUTFORMAT 0 /* RAW */
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#define FAAC_DEFAULT_BITRATE 128 * 1000
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#define FAAC_DEFAULT_PROFILE LOW
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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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GType
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gst_faac_get_type (void)
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{
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static GType gst_faac_type = 0;
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if (!gst_faac_type) {
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static const GTypeInfo gst_faac_info = {
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sizeof (GstFaacClass),
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(GBaseInitFunc) gst_faac_base_init,
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NULL,
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(GClassInitFunc) gst_faac_class_init,
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NULL,
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NULL,
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sizeof (GstFaac),
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0,
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(GInstanceInitFunc) gst_faac_init,
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};
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const GInterfaceInfo preset_interface_info = {
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NULL, /* interface_init */
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NULL, /* interface_finalize */
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NULL /* interface_data */
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};
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gst_faac_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstFaac", &gst_faac_info, 0);
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g_type_add_interface_static (gst_faac_type, GST_TYPE_PRESET,
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&preset_interface_info);
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}
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return gst_faac_type;
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}
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static void
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gst_faac_base_init (GstFaacClass * 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_PROFILE (gst_faac_profile_get_type ())
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static GType
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gst_faac_profile_get_type (void)
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{
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static GType gst_faac_profile_type = 0;
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if (!gst_faac_profile_type) {
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static GEnumValue gst_faac_profile[] = {
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{MAIN, "MAIN", "Main profile"},
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{LOW, "LC", "Low complexity profile"},
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{SSR, "SSR", "Scalable sampling rate profile"},
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{LTP, "LTP", "Long term prediction profile"},
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{0, NULL, NULL},
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};
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gst_faac_profile_type = g_enum_register_static ("GstFaacProfile",
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gst_faac_profile);
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}
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return gst_faac_profile_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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GstElementClass *gstelement_class = GST_ELEMENT_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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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_faac_finalize);
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/* properties */
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g_object_class_install_property (gobject_class, ARG_BITRATE,
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g_param_spec_int ("bitrate", "Bitrate (bps)", "Bitrate in bits/sec",
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8 * 1000, 320 * 1000, FAAC_DEFAULT_BITRATE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, ARG_PROFILE,
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g_param_spec_enum ("profile", "Profile", "MPEG/AAC encoding profile",
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GST_TYPE_FAAC_PROFILE, FAAC_DEFAULT_PROFILE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, ARG_TNS,
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g_param_spec_boolean ("tns", "TNS", "Use temporal noise shaping",
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FAAC_DEFAULT_TNS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, ARG_MIDSIDE,
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g_param_spec_boolean ("midside", "Midside", "Allow mid/side encoding",
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FAAC_DEFAULT_MIDSIDE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, ARG_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_READWRITE | G_PARAM_STATIC_STRINGS));
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/* virtual functions */
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gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_faac_change_state);
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}
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static void
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gst_faac_init (GstFaac * faac)
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{
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faac->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
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gst_pad_set_chain_function (faac->sinkpad,
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GST_DEBUG_FUNCPTR (gst_faac_chain));
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gst_pad_set_setcaps_function (faac->sinkpad,
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GST_DEBUG_FUNCPTR (gst_faac_sink_setcaps));
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gst_pad_set_getcaps_function (faac->sinkpad,
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GST_DEBUG_FUNCPTR (gst_faac_sink_getcaps));
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gst_pad_set_event_function (faac->sinkpad,
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GST_DEBUG_FUNCPTR (gst_faac_sink_event));
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gst_element_add_pad (GST_ELEMENT (faac), faac->sinkpad);
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faac->srcpad = gst_pad_new_from_static_template (&src_template, "src");
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gst_pad_use_fixed_caps (faac->srcpad);
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gst_element_add_pad (GST_ELEMENT (faac), faac->srcpad);
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faac->adapter = gst_adapter_new ();
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/* default properties */
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faac->bitrate = FAAC_DEFAULT_BITRATE;
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faac->profile = FAAC_DEFAULT_PROFILE;
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faac->shortctl = FAAC_DEFAULT_SHORTCTL;
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faac->outputformat = FAAC_DEFAULT_OUTPUTFORMAT;
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faac->tns = FAAC_DEFAULT_TNS;
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faac->midside = FAAC_DEFAULT_MIDSIDE;
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gst_faac_reset (faac);
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}
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static void
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gst_faac_reset (GstFaac * faac)
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{
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faac->handle = NULL;
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faac->samplerate = -1;
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faac->channels = -1;
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faac->offset = 0;
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gst_adapter_clear (faac->adapter);
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}
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static void
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gst_faac_finalize (GObject * object)
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{
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GstFaac *faac = (GstFaac *) object;
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g_object_unref (faac->adapter);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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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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gst_adapter_clear (faac->adapter);
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faac->offset = 0;
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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_sink_getcaps (GstPad * pad)
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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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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,
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"depth", G_TYPE_INT, 16, "rate", GST_TYPE_INT_RANGE, 8000, 96000, NULL);
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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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GST_DEBUG_OBJECT (pad, "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_caps_ref ((GstCaps *) sinkcaps);
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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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caps = gst_pad_get_allowed_caps (faac->srcpad);
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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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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 (faac, "use ADTS format for output");
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faac->outputformat = 1;
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} else if (strcmp (str, "raw") == 0) {
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GST_DEBUG_OBJECT (faac, "use RAW format for output");
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faac->outputformat = 0;
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} else {
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GST_DEBUG_OBJECT (faac, "unknown stream-format: %s", str);
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faac->outputformat = 0;
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}
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}
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}
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if (caps)
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gst_caps_unref (caps);
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}
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static gboolean
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gst_faac_sink_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
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GstStructure *structure = gst_caps_get_structure (caps, 0);
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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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if (!gst_caps_is_fixed (caps))
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goto refuse_caps;
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if (!gst_structure_get_int (structure, "channels", &channels) ||
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!gst_structure_get_int (structure, "rate", &samplerate)) {
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goto refuse_caps;
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}
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if (gst_structure_has_name (structure, "audio/x-raw-int")) {
|
|
gst_structure_get_int (structure, "width", &width);
|
|
switch (width) {
|
|
case 16:
|
|
fmt = FAAC_INPUT_16BIT;
|
|
bps = 2;
|
|
break;
|
|
case 24:
|
|
case 32:
|
|
fmt = FAAC_INPUT_32BIT;
|
|
bps = 4;
|
|
break;
|
|
default:
|
|
g_return_val_if_reached (FALSE);
|
|
}
|
|
} else if (gst_structure_has_name (structure, "audio/x-raw-float")) {
|
|
fmt = FAAC_INPUT_FLOAT;
|
|
bps = 4;
|
|
}
|
|
|
|
if (!fmt)
|
|
goto refuse_caps;
|
|
|
|
/* If the encoder is initialized, do not
|
|
reinitialize it again if not necessary */
|
|
if (faac->handle) {
|
|
if (samplerate == faac->samplerate && channels == faac->channels &&
|
|
fmt == faac->format)
|
|
return TRUE;
|
|
|
|
/* clear out pending frames */
|
|
gst_faac_push_buffers (faac, TRUE);
|
|
|
|
gst_faac_close_encoder (faac);
|
|
}
|
|
|
|
if (!(handle = faacEncOpen (samplerate, channels, &samples, &bytes)))
|
|
goto setup_failed;
|
|
|
|
/* ok, record and set up */
|
|
faac->format = fmt;
|
|
faac->bps = bps;
|
|
faac->handle = handle;
|
|
faac->bytes = bytes;
|
|
faac->samples = samples;
|
|
faac->channels = channels;
|
|
faac->samplerate = samplerate;
|
|
|
|
gst_faac_negotiate (faac);
|
|
|
|
/* finish up */
|
|
result = gst_faac_configure_source_pad (faac);
|
|
|
|
done:
|
|
gst_object_unref (faac);
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
setup_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (faac, LIBRARY, SETTINGS, (NULL), (NULL));
|
|
goto done;
|
|
}
|
|
refuse_caps:
|
|
{
|
|
GST_WARNING_OBJECT (faac, "refused caps %" GST_PTR_FORMAT, caps);
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_faac_configure_source_pad (GstFaac * faac)
|
|
{
|
|
GstCaps *allowed_caps;
|
|
GstCaps *srccaps;
|
|
gboolean ret = FALSE;
|
|
gint n, ver, mpegversion = 2;
|
|
faacEncConfiguration *conf;
|
|
guint maxbitrate;
|
|
|
|
mpegversion = FAAC_DEFAULT_MPEGVERSION;
|
|
|
|
allowed_caps = gst_pad_get_allowed_caps (faac->srcpad);
|
|
GST_DEBUG_OBJECT (faac, "allowed caps: %" GST_PTR_FORMAT, allowed_caps);
|
|
|
|
if (allowed_caps) {
|
|
if (gst_caps_is_empty (allowed_caps))
|
|
goto empty_caps;
|
|
|
|
if (!gst_caps_is_any (allowed_caps)) {
|
|
for (n = 0; n < gst_caps_get_size (allowed_caps); n++) {
|
|
GstStructure *s = gst_caps_get_structure (allowed_caps, n);
|
|
|
|
if (gst_structure_get_int (s, "mpegversion", &ver) &&
|
|
(ver == 4 || ver == 2)) {
|
|
mpegversion = ver;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
gst_caps_unref (allowed_caps);
|
|
}
|
|
|
|
/* we negotiated caps update current configuration */
|
|
conf = faacEncGetCurrentConfiguration (faac->handle);
|
|
conf->mpegVersion = (mpegversion == 4) ? MPEG4 : MPEG2;
|
|
conf->aacObjectType = faac->profile;
|
|
conf->allowMidside = faac->midside;
|
|
conf->useLfe = 0;
|
|
conf->useTns = faac->tns;
|
|
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, mpegversion,
|
|
"channels", G_TYPE_INT, faac->channels,
|
|
"rate", G_TYPE_INT, faac->samplerate,
|
|
"stream-format", G_TYPE_STRING, (faac->outputformat ? "adts" : "raw"),
|
|
NULL);
|
|
|
|
if (!faac->outputformat) {
|
|
GstBuffer *codec_data;
|
|
guint8 *config = NULL;
|
|
gulong config_len = 0;
|
|
|
|
/* get the config string */
|
|
GST_DEBUG_OBJECT (faac, "retrieving decoder info");
|
|
faacEncGetDecoderSpecificInfo (faac->handle, &config, &config_len);
|
|
|
|
/* copy it into a buffer */
|
|
codec_data = gst_buffer_new_and_alloc (config_len);
|
|
memcpy (GST_BUFFER_DATA (codec_data), config, config_len);
|
|
free (config);
|
|
|
|
/* add to caps */
|
|
gst_caps_set_simple (srccaps,
|
|
"codec_data", GST_TYPE_BUFFER, codec_data, NULL);
|
|
|
|
gst_buffer_unref (codec_data);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (faac, "src pad caps: %" GST_PTR_FORMAT, srccaps);
|
|
|
|
ret = gst_pad_set_caps (faac->srcpad, srccaps);
|
|
gst_caps_unref (srccaps);
|
|
|
|
return ret;
|
|
|
|
/* ERROR */
|
|
empty_caps:
|
|
{
|
|
gst_caps_unref (allowed_caps);
|
|
return FALSE;
|
|
}
|
|
set_failed:
|
|
{
|
|
GST_WARNING_OBJECT (faac, "Faac doesn't support the current configuration");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_faac_push_buffers (GstFaac * faac, gboolean force)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
gint av, frame_size, size, ret_size;
|
|
GstBuffer *outbuf;
|
|
guint64 timestamp, distance;
|
|
const guint8 *data;
|
|
|
|
/* samples already considers channel count */
|
|
frame_size = faac->samples * faac->bps;
|
|
|
|
while (G_LIKELY (ret == GST_FLOW_OK)) {
|
|
|
|
av = gst_adapter_available (faac->adapter);
|
|
|
|
GST_LOG_OBJECT (faac, "pushing: force: %d, frame_size: %d, av: %d, "
|
|
"offset: %d", force, frame_size, av, faac->offset);
|
|
|
|
/* idea:
|
|
* - start of adapter corresponds with what has already been encoded
|
|
* (i.e. really returned by faac)
|
|
* - start + offset is what needs to be fed to faac next
|
|
* That way we can timestamp the output based
|
|
* on adapter provided timestamp (and duration is a fixed frame duration) */
|
|
|
|
/* not enough data for one frame and no flush forcing */
|
|
if (!force && (av < frame_size + faac->offset))
|
|
break;
|
|
|
|
if (G_LIKELY (av - faac->offset >= frame_size)) {
|
|
GST_LOG_OBJECT (faac, "encoding a frame");
|
|
data = gst_adapter_peek (faac->adapter, faac->offset + frame_size);
|
|
data += faac->offset;
|
|
size = frame_size;
|
|
} else if (av - faac->offset > 0) {
|
|
GST_LOG_OBJECT (faac, "encoding leftover");
|
|
data = gst_adapter_peek (faac->adapter, av);
|
|
data += faac->offset;
|
|
size = av - faac->offset;
|
|
} else {
|
|
GST_LOG_OBJECT (faac, "emptying encoder");
|
|
data = NULL;
|
|
size = 0;
|
|
}
|
|
|
|
outbuf = gst_buffer_new_and_alloc (faac->bytes);
|
|
|
|
if (G_UNLIKELY ((ret_size = faacEncEncode (faac->handle, (gint32 *) data,
|
|
size / faac->bps, GST_BUFFER_DATA (outbuf),
|
|
faac->bytes)) < 0)) {
|
|
gst_buffer_unref (outbuf);
|
|
goto encode_failed;
|
|
}
|
|
|
|
GST_LOG_OBJECT (faac, "encoder return: %d", ret_size);
|
|
|
|
/* consumed, advanced view */
|
|
faac->offset += size;
|
|
g_assert (faac->offset <= av);
|
|
|
|
if (G_UNLIKELY (!ret_size)) {
|
|
gst_buffer_unref (outbuf);
|
|
if (size)
|
|
continue;
|
|
else
|
|
break;
|
|
}
|
|
|
|
/* deal with encoder lead-out */
|
|
if (G_UNLIKELY (av == 0 && faac->offset == 0)) {
|
|
GST_DEBUG_OBJECT (faac, "encoder returned additional data");
|
|
/* continuous with previous output, ok to have 0 duration */
|
|
timestamp = faac->next_ts;
|
|
} else {
|
|
/* after some caching, finally some data */
|
|
/* adapter gives time */
|
|
timestamp = gst_adapter_prev_timestamp (faac->adapter, &distance);
|
|
}
|
|
|
|
if (G_LIKELY ((av = gst_adapter_available (faac->adapter)) >= frame_size)) {
|
|
/* must have then come from a complete frame */
|
|
gst_adapter_flush (faac->adapter, frame_size);
|
|
faac->offset -= frame_size;
|
|
size = frame_size;
|
|
} else {
|
|
/* otherwise leftover */
|
|
gst_adapter_clear (faac->adapter);
|
|
faac->offset = 0;
|
|
size = av;
|
|
}
|
|
|
|
GST_BUFFER_SIZE (outbuf) = ret_size;
|
|
if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (timestamp)))
|
|
GST_BUFFER_TIMESTAMP (outbuf) = timestamp +
|
|
GST_FRAMES_TO_CLOCK_TIME (distance / faac->channels / faac->bps,
|
|
faac->samplerate);
|
|
GST_BUFFER_DURATION (outbuf) =
|
|
GST_FRAMES_TO_CLOCK_TIME (size / faac->channels / faac->bps,
|
|
faac->samplerate);
|
|
faac->next_ts =
|
|
GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
|
|
|
|
/* perhaps check/set DISCONT based on timestamps ? */
|
|
|
|
GST_LOG_OBJECT (faac, "Pushing out buffer time: %" GST_TIME_FORMAT
|
|
" duration: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
|
|
|
|
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (faac->srcpad));
|
|
ret = gst_pad_push (faac->srcpad, outbuf);
|
|
}
|
|
|
|
/* in case encoder returns less than expected, clear our view as well */
|
|
if (G_UNLIKELY (force)) {
|
|
#ifndef GST_DISABLE_GST_DEBUG
|
|
if ((av = gst_adapter_available (faac->adapter)))
|
|
GST_WARNING_OBJECT (faac, "encoder left %d bytes; discarding", av);
|
|
#endif
|
|
gst_adapter_clear (faac->adapter);
|
|
faac->offset = 0;
|
|
}
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
encode_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (faac, LIBRARY, ENCODE, (NULL), (NULL));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_faac_sink_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstFaac *faac;
|
|
gboolean ret;
|
|
|
|
faac = GST_FAAC (gst_pad_get_parent (pad));
|
|
|
|
GST_LOG_OBJECT (faac, "received %s", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
{
|
|
if (faac->handle) {
|
|
/* flush first */
|
|
GST_DEBUG_OBJECT (faac, "Pushing out remaining buffers because of EOS");
|
|
gst_faac_push_buffers (faac, TRUE);
|
|
}
|
|
|
|
ret = gst_pad_event_default (pad, event);
|
|
break;
|
|
}
|
|
default:
|
|
ret = gst_pad_event_default (pad, event);
|
|
break;
|
|
|
|
}
|
|
gst_object_unref (faac);
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_faac_chain (GstPad * pad, GstBuffer * inbuf)
|
|
{
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
GstFaac *faac;
|
|
|
|
faac = GST_FAAC (gst_pad_get_parent (pad));
|
|
|
|
if (!faac->handle)
|
|
goto no_handle;
|
|
|
|
GST_LOG_OBJECT (faac, "Got buffer time: %" GST_TIME_FORMAT " duration: %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)));
|
|
|
|
gst_adapter_push (faac->adapter, inbuf);
|
|
|
|
result = gst_faac_push_buffers (faac, FALSE);
|
|
|
|
done:
|
|
gst_object_unref (faac);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
no_handle:
|
|
{
|
|
GST_ELEMENT_ERROR (faac, CORE, NEGOTIATION, (NULL),
|
|
("format wasn't negotiated before chain function"));
|
|
gst_buffer_unref (inbuf);
|
|
result = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
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 ARG_BITRATE:
|
|
faac->bitrate = g_value_get_int (value);
|
|
break;
|
|
case ARG_PROFILE:
|
|
faac->profile = g_value_get_enum (value);
|
|
break;
|
|
case ARG_TNS:
|
|
faac->tns = g_value_get_boolean (value);
|
|
break;
|
|
case ARG_MIDSIDE:
|
|
faac->midside = g_value_get_boolean (value);
|
|
break;
|
|
case ARG_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 ARG_BITRATE:
|
|
g_value_set_int (value, faac->bitrate);
|
|
break;
|
|
case ARG_PROFILE:
|
|
g_value_set_enum (value, faac->profile);
|
|
break;
|
|
case ARG_TNS:
|
|
g_value_set_boolean (value, faac->tns);
|
|
break;
|
|
case ARG_MIDSIDE:
|
|
g_value_set_boolean (value, faac->midside);
|
|
break;
|
|
case ARG_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 GstStateChangeReturn
|
|
gst_faac_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
GstFaac *faac = GST_FAAC (element);
|
|
|
|
/* upwards state changes */
|
|
switch (transition) {
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
/* downwards state changes */
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
{
|
|
gst_faac_close_encoder (faac);
|
|
gst_faac_reset (faac);
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
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)
|