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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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599401a7f9
This patch adds the get_frame_overhead() vfunc so that baseparse can accurately calculate the min/avg/max bitrates for aacparse. Note: The bitrate was being incorrectly calculated for ADTS streams (it's not in the header as the code suggests).
739 lines
22 KiB
C
739 lines
22 KiB
C
/* GStreamer AAC parser plugin
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* Copyright (C) 2008 Nokia Corporation. All rights reserved.
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*
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* Contact: Stefan Kost <stefan.kost@nokia.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., 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-aacparse
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* @short_description: AAC parser
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* @see_also: #GstAmrParse
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*
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* <refsect2>
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* <para>
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* This is an AAC parser. It can handle both ADIF and ADTS stream formats.
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* The parser inherits from #GstBaseParse and therefore in only needs to
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* implement AAC-specific functionality.
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* </para>
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* <para>
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* As ADIF format is not framed, it is not seekable. From the same reason
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* stream duration cannot be calculated either. Instead, AAC clips that are
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* in ADTS format can be seeked, and parser also is able to calculate their
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* playback position and clip duration.
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* </para>
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* gst-launch filesrc location=abc.aac ! aacparse ! faad ! audioresample ! audioconvert ! alsasink
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* </programlisting>
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* </para>
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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 <string.h>
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#include "gstaacparse.h"
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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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"framed = (boolean) true, " "mpegversion = (int) { 2, 4 }, "
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"stream-format = (string) { raw, adts, adif };"));
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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/mpeg, "
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"framed = (boolean) false, " "mpegversion = (int) { 2, 4 };"));
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GST_DEBUG_CATEGORY_STATIC (gst_aacparse_debug);
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#define GST_CAT_DEFAULT gst_aacparse_debug
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#define ADIF_MAX_SIZE 40 /* Should be enough */
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#define ADTS_MAX_SIZE 10 /* Should be enough */
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#define AAC_FRAME_DURATION(parse) (GST_SECOND/parse->frames_per_sec)
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static void gst_aacparse_finalize (GObject * object);
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gboolean gst_aacparse_start (GstBaseParse * parse);
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gboolean gst_aacparse_stop (GstBaseParse * parse);
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static gboolean gst_aacparse_sink_setcaps (GstBaseParse * parse,
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GstCaps * caps);
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gboolean gst_aacparse_check_valid_frame (GstBaseParse * parse,
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GstBuffer * buffer, guint * size, gint * skipsize);
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GstFlowReturn gst_aacparse_parse_frame (GstBaseParse * parse,
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GstBuffer * buffer);
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gboolean gst_aacparse_convert (GstBaseParse * parse,
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GstFormat src_format,
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gint64 src_value, GstFormat dest_format, gint64 * dest_value);
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gboolean gst_aacparse_is_seekable (GstBaseParse * parse);
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gint gst_aacparse_get_frame_overhead (GstBaseParse * parse, GstBuffer * buffer);
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gboolean gst_aacparse_event (GstBaseParse * parse, GstEvent * event);
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#define _do_init(bla) \
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GST_DEBUG_CATEGORY_INIT (gst_aacparse_debug, "aacparse", 0, \
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"AAC audio stream parser");
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GST_BOILERPLATE_FULL (GstAacParse, gst_aacparse, GstBaseParse,
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GST_TYPE_BASE_PARSE, _do_init);
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static inline gint
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gst_aacparse_get_sample_rate_from_index (guint sr_idx)
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{
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static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000, 44100,
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32000, 24000, 22050, 16000, 12000, 11025, 8000
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};
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if (sr_idx < G_N_ELEMENTS (aac_sample_rates))
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return aac_sample_rates[sr_idx];
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GST_WARNING ("Invalid sample rate index %u", sr_idx);
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return 0;
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}
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/**
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* gst_aacparse_base_init:
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* @klass: #GstElementClass.
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*
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*/
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static void
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gst_aacparse_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 (&sink_template));
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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_set_details_simple (element_class,
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"AAC audio stream parser", "Codec/Parser/Audio",
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"Advanced Audio Coding parser", "Stefan Kost <stefan.kost@nokia.com>");
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}
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/**
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* gst_aacparse_class_init:
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* @klass: #GstAacParseClass.
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*
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*/
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static void
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gst_aacparse_class_init (GstAacParseClass * klass)
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{
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
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object_class->finalize = gst_aacparse_finalize;
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parse_class->start = GST_DEBUG_FUNCPTR (gst_aacparse_start);
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parse_class->stop = GST_DEBUG_FUNCPTR (gst_aacparse_stop);
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parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_aacparse_sink_setcaps);
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parse_class->is_seekable = GST_DEBUG_FUNCPTR (gst_aacparse_is_seekable);
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parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_aacparse_parse_frame);
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parse_class->check_valid_frame =
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GST_DEBUG_FUNCPTR (gst_aacparse_check_valid_frame);
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parse_class->get_frame_overhead =
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GST_DEBUG_FUNCPTR (gst_aacparse_get_frame_overhead);
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}
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/**
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* gst_aacparse_init:
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* @aacparse: #GstAacParse.
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* @klass: #GstAacParseClass.
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*
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*/
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static void
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gst_aacparse_init (GstAacParse * aacparse, GstAacParseClass * klass)
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{
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/* init rest */
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gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 1024);
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GST_DEBUG ("initialized");
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}
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/**
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* gst_aacparse_finalize:
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* @object:
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*
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*/
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static void
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gst_aacparse_finalize (GObject * object)
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{
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GstAacParse *aacparse;
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aacparse = GST_AACPARSE (object);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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/**
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* gst_aacparse_set_src_caps:
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* @aacparse: #GstAacParse.
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* @sink_caps: (proposed) caps of sink pad
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*
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* Set source pad caps according to current knowledge about the
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* audio stream.
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*
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* Returns: TRUE if caps were successfully set.
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*/
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static gboolean
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gst_aacparse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
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{
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GstStructure *s;
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GstCaps *src_caps = NULL;
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gboolean res = FALSE;
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const gchar *stream_format;
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GST_DEBUG_OBJECT (aacparse, "sink caps: %" GST_PTR_FORMAT, sink_caps);
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if (sink_caps)
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src_caps = gst_caps_copy (sink_caps);
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else
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src_caps = gst_caps_new_simple ("audio/mpeg", NULL);
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gst_caps_set_simple (src_caps, "framed", G_TYPE_BOOLEAN, TRUE,
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"mpegversion", G_TYPE_INT, aacparse->mpegversion, NULL);
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switch (aacparse->header_type) {
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case DSPAAC_HEADER_NONE:
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stream_format = "raw";
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break;
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case DSPAAC_HEADER_ADTS:
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stream_format = "adts";
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break;
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case DSPAAC_HEADER_ADIF:
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stream_format = "adif";
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break;
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default:
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stream_format = NULL;
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}
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s = gst_caps_get_structure (src_caps, 0);
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if (!gst_structure_has_field (s, "rate") && aacparse->sample_rate > 0)
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gst_structure_set (s, "rate", G_TYPE_INT, aacparse->sample_rate, NULL);
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if (!gst_structure_has_field (s, "channels") && aacparse->channels > 0)
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gst_structure_set (s, "channels", G_TYPE_INT, aacparse->channels, NULL);
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if (!gst_structure_has_field (s, "stream-format") && stream_format)
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gst_structure_set (s, "stream-format", G_TYPE_STRING, stream_format, NULL);
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GST_DEBUG_OBJECT (aacparse, "setting src caps: %" GST_PTR_FORMAT, src_caps);
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gst_pad_use_fixed_caps (GST_BASE_PARSE (aacparse)->srcpad);
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res = gst_pad_set_caps (GST_BASE_PARSE (aacparse)->srcpad, src_caps);
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gst_caps_unref (src_caps);
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return res;
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}
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/**
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* gst_aacparse_sink_setcaps:
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* @sinkpad: GstPad
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* @caps: GstCaps
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*
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* Implementation of "set_sink_caps" vmethod in #GstBaseParse class.
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*
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* Returns: TRUE on success.
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*/
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static gboolean
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gst_aacparse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
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{
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GstAacParse *aacparse;
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GstStructure *structure;
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gchar *caps_str;
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aacparse = GST_AACPARSE (parse);
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structure = gst_caps_get_structure (caps, 0);
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caps_str = gst_caps_to_string (caps);
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GST_DEBUG_OBJECT (aacparse, "setcaps: %s", caps_str);
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g_free (caps_str);
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/* This is needed at least in case of RTP
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* Parses the codec_data information to get ObjectType,
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* number of channels and samplerate */
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if (gst_structure_has_field (structure, "codec_data")) {
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const GValue *value = gst_structure_get_value (structure, "codec_data");
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if (value) {
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GstBuffer *buf = gst_value_get_buffer (value);
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const guint8 *buffer = GST_BUFFER_DATA (buf);
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guint sr_idx;
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sr_idx = ((buffer[0] & 0x07) << 1) | ((buffer[1] & 0x80) >> 7);
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aacparse->object_type = (buffer[0] & 0xf8) >> 3;
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aacparse->sample_rate = gst_aacparse_get_sample_rate_from_index (sr_idx);
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aacparse->channels = (buffer[1] & 0x78) >> 3;
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aacparse->header_type = DSPAAC_HEADER_NONE;
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aacparse->mpegversion = 4;
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GST_DEBUG ("codec_data: object_type=%d, sample_rate=%d, channels=%d",
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aacparse->object_type, aacparse->sample_rate, aacparse->channels);
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/* arrange for metadata and get out of the way */
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gst_aacparse_set_src_caps (aacparse, caps);
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gst_base_parse_set_passthrough (parse, TRUE);
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} else
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return FALSE;
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}
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return TRUE;
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}
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/**
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* gst_aacparse_adts_get_frame_len:
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* @data: block of data containing an ADTS header.
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*
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* This function calculates ADTS frame length from the given header.
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*
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* Returns: size of the ADTS frame.
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*/
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static inline guint
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gst_aacparse_adts_get_frame_len (const guint8 * data)
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{
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return ((data[3] & 0x03) << 11) | (data[4] << 3) | ((data[5] & 0xe0) >> 5);
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}
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/**
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* gst_aacparse_check_adts_frame:
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* @aacparse: #GstAacParse.
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* @data: Data to be checked.
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* @avail: Amount of data passed.
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* @framesize: If valid ADTS frame was found, this will be set to tell the
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* found frame size in bytes.
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* @needed_data: If frame was not found, this may be set to tell how much
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* more data is needed in the next round to detect the frame
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* reliably. This may happen when a frame header candidate
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* is found but it cannot be guaranteed to be the header without
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* peeking the following data.
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*
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* Check if the given data contains contains ADTS frame. The algorithm
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* will examine ADTS frame header and calculate the frame size. Also, another
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* consecutive ADTS frame header need to be present after the found frame.
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* Otherwise the data is not considered as a valid ADTS frame. However, this
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* "extra check" is omitted when EOS has been received. In this case it is
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* enough when data[0] contains a valid ADTS header.
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*
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* This function may set the #needed_data to indicate that a possible frame
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* candidate has been found, but more data (#needed_data bytes) is needed to
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* be absolutely sure. When this situation occurs, FALSE will be returned.
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*
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* When a valid frame is detected, this function will use
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* gst_base_parse_set_min_frame_size() function from #GstBaseParse class
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* to set the needed bytes for next frame.This way next data chunk is already
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* of correct size.
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*
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* Returns: TRUE if the given data contains a valid ADTS header.
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*/
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static gboolean
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gst_aacparse_check_adts_frame (GstAacParse * aacparse,
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const guint8 * data,
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const guint avail, guint * framesize, guint * needed_data)
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{
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if (G_UNLIKELY (avail < 2))
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return FALSE;
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if ((data[0] == 0xff) && ((data[1] & 0xf6) == 0xf0)) {
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*framesize = gst_aacparse_adts_get_frame_len (data);
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/* In EOS mode this is enough. No need to examine the data further */
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if (gst_base_parse_get_drain (GST_BASE_PARSE (aacparse))) {
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return TRUE;
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}
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if (*framesize + ADTS_MAX_SIZE > avail) {
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/* We have found a possible frame header candidate, but can't be
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sure since we don't have enough data to check the next frame */
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GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
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*framesize + ADTS_MAX_SIZE, avail);
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*needed_data = *framesize + ADTS_MAX_SIZE;
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gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
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*framesize + ADTS_MAX_SIZE);
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return FALSE;
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}
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if ((data[*framesize] == 0xff) && ((data[*framesize + 1] & 0xf6) == 0xf0)) {
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guint nextlen = gst_aacparse_adts_get_frame_len (data + (*framesize));
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GST_LOG ("ADTS frame found, len: %d bytes", *framesize);
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gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
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nextlen + ADTS_MAX_SIZE);
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return TRUE;
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}
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}
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return FALSE;
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}
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/**
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* gst_aacparse_detect_stream:
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* @aacparse: #GstAacParse.
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* @data: A block of data that needs to be examined for stream characteristics.
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* @avail: Size of the given datablock.
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* @framesize: If valid stream was found, this will be set to tell the
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* first frame size in bytes.
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* @skipsize: If valid stream was found, this will be set to tell the first
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* audio frame position within the given data.
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*
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* Examines the given piece of data and try to detect the format of it. It
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* checks for "ADIF" header (in the beginning of the clip) and ADTS frame
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* header. If the stream is detected, TRUE will be returned and #framesize
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* is set to indicate the found frame size. Additionally, #skipsize might
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* be set to indicate the number of bytes that need to be skipped, a.k.a. the
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* position of the frame inside given data chunk.
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*
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* Returns: TRUE on success.
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*/
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static gboolean
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gst_aacparse_detect_stream (GstAacParse * aacparse,
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const guint8 * data, const guint avail, guint * framesize, gint * skipsize)
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{
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gboolean found = FALSE;
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guint need_data = 0;
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guint i = 0;
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GST_DEBUG_OBJECT (aacparse, "Parsing header data");
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/* FIXME: No need to check for ADIF if we are not in the beginning of the
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stream */
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/* Can we even parse the header? */
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if (avail < ADTS_MAX_SIZE)
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return FALSE;
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for (i = 0; i < avail - 4; i++) {
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if (((data[i] == 0xff) && ((data[i + 1] & 0xf6) == 0xf0)) ||
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strncmp ((char *) data + i, "ADIF", 4) == 0) {
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found = TRUE;
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if (i) {
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/* Trick: tell the parent class that we didn't find the frame yet,
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but make it skip 'i' amount of bytes. Next time we arrive
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here we have full frame in the beginning of the data. */
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*skipsize = i;
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return FALSE;
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}
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break;
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}
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}
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if (!found) {
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if (i)
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*skipsize = i;
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return FALSE;
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}
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if (gst_aacparse_check_adts_frame (aacparse, data, avail,
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framesize, &need_data)) {
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gint sr_idx;
|
|
GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize);
|
|
|
|
aacparse->header_type = DSPAAC_HEADER_ADTS;
|
|
sr_idx = (data[2] & 0x3c) >> 2;
|
|
|
|
aacparse->sample_rate = gst_aacparse_get_sample_rate_from_index (sr_idx);
|
|
aacparse->mpegversion = (data[1] & 0x08) ? 2 : 4;
|
|
aacparse->object_type = (data[2] & 0xc0) >> 6;
|
|
aacparse->channels = ((data[2] & 0x01) << 2) | ((data[3] & 0xc0) >> 6);
|
|
|
|
gst_base_parse_set_frame_props (GST_BASE_PARSE (aacparse),
|
|
aacparse->sample_rate, 1024, 50);
|
|
|
|
GST_DEBUG ("ADTS: samplerate %d, channels %d, objtype %d",
|
|
aacparse->sample_rate, aacparse->channels, aacparse->object_type);
|
|
|
|
return TRUE;
|
|
} else if (need_data) {
|
|
/* This tells the parent class not to skip any data */
|
|
*skipsize = 0;
|
|
return FALSE;
|
|
}
|
|
|
|
if (avail < ADIF_MAX_SIZE)
|
|
return FALSE;
|
|
|
|
if (memcmp (data + i, "ADIF", 4) == 0) {
|
|
const guint8 *adif;
|
|
int skip_size = 0;
|
|
int bitstream_type;
|
|
int sr_idx;
|
|
|
|
aacparse->header_type = DSPAAC_HEADER_ADIF;
|
|
aacparse->mpegversion = 4;
|
|
|
|
/* Skip the "ADIF" bytes */
|
|
adif = data + i + 4;
|
|
|
|
/* copyright string */
|
|
if (adif[0] & 0x80)
|
|
skip_size += 9; /* skip 9 bytes */
|
|
|
|
bitstream_type = adif[0 + skip_size] & 0x10;
|
|
aacparse->bitrate =
|
|
((unsigned int) (adif[0 + skip_size] & 0x0f) << 19) |
|
|
((unsigned int) adif[1 + skip_size] << 11) |
|
|
((unsigned int) adif[2 + skip_size] << 3) |
|
|
((unsigned int) adif[3 + skip_size] & 0xe0);
|
|
|
|
/* CBR */
|
|
if (bitstream_type == 0) {
|
|
#if 0
|
|
/* Buffer fullness parsing. Currently not needed... */
|
|
guint num_elems = 0;
|
|
guint fullness = 0;
|
|
|
|
num_elems = (adif[3 + skip_size] & 0x1e);
|
|
GST_INFO ("ADIF num_config_elems: %d", num_elems);
|
|
|
|
fullness = ((unsigned int) (adif[3 + skip_size] & 0x01) << 19) |
|
|
((unsigned int) adif[4 + skip_size] << 11) |
|
|
((unsigned int) adif[5 + skip_size] << 3) |
|
|
((unsigned int) (adif[6 + skip_size] & 0xe0) >> 5);
|
|
|
|
GST_INFO ("ADIF buffer fullness: %d", fullness);
|
|
#endif
|
|
aacparse->object_type = ((adif[6 + skip_size] & 0x01) << 1) |
|
|
((adif[7 + skip_size] & 0x80) >> 7);
|
|
sr_idx = (adif[7 + skip_size] & 0x78) >> 3;
|
|
}
|
|
/* VBR */
|
|
else {
|
|
aacparse->object_type = (adif[4 + skip_size] & 0x18) >> 3;
|
|
sr_idx = ((adif[4 + skip_size] & 0x07) << 1) |
|
|
((adif[5 + skip_size] & 0x80) >> 7);
|
|
}
|
|
|
|
/* FIXME: This gives totally wrong results. Duration calculation cannot
|
|
be based on this */
|
|
aacparse->sample_rate = gst_aacparse_get_sample_rate_from_index (sr_idx);
|
|
|
|
/* baseparse is not given any fps,
|
|
* so it will give up on timestamps, seeking, etc */
|
|
|
|
/* FIXME: Can we assume this? */
|
|
aacparse->channels = 2;
|
|
|
|
GST_INFO ("ADIF: br=%d, samplerate=%d, objtype=%d",
|
|
aacparse->bitrate, aacparse->sample_rate, aacparse->object_type);
|
|
|
|
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 512);
|
|
|
|
*framesize = avail;
|
|
return TRUE;
|
|
}
|
|
|
|
/* This should never happen */
|
|
return FALSE;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_aacparse_check_valid_frame:
|
|
* @parse: #GstBaseParse.
|
|
* @buffer: #GstBuffer.
|
|
* @framesize: If the buffer contains a valid frame, its size will be put here
|
|
* @skipsize: How much data parent class should skip in order to find the
|
|
* frame header.
|
|
*
|
|
* Implementation of "check_valid_frame" vmethod in #GstBaseParse class.
|
|
*
|
|
* Returns: TRUE if buffer contains a valid frame.
|
|
*/
|
|
gboolean
|
|
gst_aacparse_check_valid_frame (GstBaseParse * parse,
|
|
GstBuffer * buffer, guint * framesize, gint * skipsize)
|
|
{
|
|
const guint8 *data;
|
|
GstAacParse *aacparse;
|
|
gboolean ret = FALSE;
|
|
gboolean sync;
|
|
|
|
aacparse = GST_AACPARSE (parse);
|
|
data = GST_BUFFER_DATA (buffer);
|
|
|
|
sync = gst_base_parse_get_sync (parse);
|
|
|
|
if (aacparse->header_type == DSPAAC_HEADER_ADIF ||
|
|
aacparse->header_type == DSPAAC_HEADER_NONE) {
|
|
/* There is nothing to parse */
|
|
*framesize = GST_BUFFER_SIZE (buffer);
|
|
ret = TRUE;
|
|
|
|
} else if (aacparse->header_type == DSPAAC_HEADER_NOT_PARSED || sync == FALSE) {
|
|
|
|
ret = gst_aacparse_detect_stream (aacparse, data, GST_BUFFER_SIZE (buffer),
|
|
framesize, skipsize);
|
|
|
|
} else if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
|
|
guint needed_data = 1024;
|
|
|
|
ret = gst_aacparse_check_adts_frame (aacparse, data,
|
|
GST_BUFFER_SIZE (buffer), framesize, &needed_data);
|
|
|
|
if (!ret) {
|
|
GST_DEBUG ("buffer didn't contain valid frame");
|
|
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
|
|
needed_data);
|
|
}
|
|
|
|
} else {
|
|
GST_DEBUG ("buffer didn't contain valid frame");
|
|
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 1024);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_aacparse_parse_frame:
|
|
* @parse: #GstBaseParse.
|
|
* @buffer: #GstBuffer.
|
|
*
|
|
* Implementation of "parse_frame" vmethod in #GstBaseParse class.
|
|
*
|
|
* Returns: GST_FLOW_OK if frame was successfully parsed and can be pushed
|
|
* forward. Otherwise appropriate error is returned.
|
|
*/
|
|
GstFlowReturn
|
|
gst_aacparse_parse_frame (GstBaseParse * parse, GstBuffer * buffer)
|
|
{
|
|
GstAacParse *aacparse;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
|
|
aacparse = GST_AACPARSE (parse);
|
|
|
|
if (!aacparse->src_caps_set) {
|
|
if (!gst_aacparse_set_src_caps (aacparse,
|
|
GST_PAD_CAPS (GST_BASE_PARSE (aacparse)->sinkpad))) {
|
|
/* If linking fails, we need to return appropriate error */
|
|
ret = GST_FLOW_NOT_LINKED;
|
|
}
|
|
aacparse->src_caps_set = TRUE;
|
|
}
|
|
|
|
gst_buffer_set_caps (buffer, GST_PAD_CAPS (parse->srcpad));
|
|
return ret;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_aacparse_start:
|
|
* @parse: #GstBaseParse.
|
|
*
|
|
* Implementation of "start" vmethod in #GstBaseParse class.
|
|
*
|
|
* Returns: TRUE if startup succeeded.
|
|
*/
|
|
gboolean
|
|
gst_aacparse_start (GstBaseParse * parse)
|
|
{
|
|
GstAacParse *aacparse;
|
|
|
|
aacparse = GST_AACPARSE (parse);
|
|
GST_DEBUG ("start");
|
|
aacparse->src_caps_set = FALSE;
|
|
gst_base_parse_set_passthrough (parse, FALSE);
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_aacparse_stop:
|
|
* @parse: #GstBaseParse.
|
|
*
|
|
* Implementation of "stop" vmethod in #GstBaseParse class.
|
|
*
|
|
* Returns: TRUE is stopping succeeded.
|
|
*/
|
|
gboolean
|
|
gst_aacparse_stop (GstBaseParse * parse)
|
|
{
|
|
GstAacParse *aacparse;
|
|
|
|
aacparse = GST_AACPARSE (parse);
|
|
GST_DEBUG ("stop");
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_aacparse_is_seekable:
|
|
* @parse: #GstBaseParse.
|
|
*
|
|
* Implementation of "is_seekable" vmethod in #GstBaseParse class.
|
|
*
|
|
* Returns: TRUE if the current stream is seekable.
|
|
*/
|
|
gboolean
|
|
gst_aacparse_is_seekable (GstBaseParse * parse)
|
|
{
|
|
GstAacParse *aacparse;
|
|
|
|
aacparse = GST_AACPARSE (parse);
|
|
GST_DEBUG_OBJECT (aacparse, "IS_SEEKABLE: %d",
|
|
aacparse->header_type != DSPAAC_HEADER_ADIF);
|
|
|
|
/* Not seekable if ADIF header is found */
|
|
return (aacparse->header_type != DSPAAC_HEADER_ADIF);
|
|
}
|
|
|
|
/**
|
|
* gst_aacparse_get_frame_overhead:
|
|
* @parse: #GstBaseParse.
|
|
* @buffer: #GstBuffer.
|
|
*
|
|
* Implementation of "get_frame_overhead" vmethod in #GstBaseParse class. ADTS
|
|
* streams have a 7 byte header in each frame. MP4 and ADIF streams don't have
|
|
* a per-frame header.
|
|
*
|
|
* We're making a couple of simplifying assumptions:
|
|
*
|
|
* 1. We count Program Configuration Elements rather than searching for them
|
|
* in the streams to discount them - the overhead is negligible.
|
|
*
|
|
* 2. We ignore CRC. This has a worst-case impact of (num_raw_blocks + 1)*16
|
|
* bits, which should still not be significant enough to warrant the
|
|
* additional parsing through the headers
|
|
*/
|
|
gint
|
|
gst_aacparse_get_frame_overhead (GstBaseParse * parse, GstBuffer * buffer)
|
|
{
|
|
GstAacParse *aacparse = GST_AACPARSE (parse);
|
|
|
|
if (aacparse->header_type == DSPAAC_HEADER_ADTS)
|
|
return 7;
|
|
else
|
|
return 0;
|
|
}
|