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771d954c04
Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
1677 lines
53 KiB
C
1677 lines
53 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-aacparse
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* @title: aacparse
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* @short_description: AAC parser
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* @see_also: #GstAmrParse
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*
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* This is an AAC parser which handles both ADIF and ADTS stream formats.
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*
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* As ADIF format is not framed, it is not seekable and stream duration cannot
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* be determined either. However, ADTS format AAC clips can be seeked, and parser
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* can also estimate playback position and clip duration.
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 filesrc location=abc.aac ! aacparse ! faad ! audioresample ! audioconvert ! alsasink
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* ]|
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*
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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 <gst/base/gstbitreader.h>
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#include <gst/pbutils/pbutils.h>
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#include "gstaudioparserselements.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, loas };"));
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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, mpegversion = (int) { 2, 4 };"));
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GST_DEBUG_CATEGORY_STATIC (aacparse_debug);
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#define GST_CAT_DEFAULT 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 LOAS_MAX_SIZE 3 /* Should be enough */
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#define RAW_MAX_SIZE 1 /* Correct framing is required */
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#define ADTS_HEADERS_LENGTH 7UL /* Total byte-length of fixed and variable
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headers prepended during raw to ADTS
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conversion */
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#define AAC_FRAME_DURATION(parse) (GST_SECOND/parse->frames_per_sec)
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static const gint loas_sample_rate_table[16] = {
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96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
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16000, 12000, 11025, 8000, 7350, 0, 0, 0
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};
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static const gint loas_channels_table[16] = {
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0, 1, 2, 3, 4, 5, 6, 8,
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0, 0, 0, 7, 8, 0, 8, 0
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};
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static gboolean gst_aac_parse_start (GstBaseParse * parse);
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static gboolean gst_aac_parse_stop (GstBaseParse * parse);
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static gboolean gst_aac_parse_sink_setcaps (GstBaseParse * parse,
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GstCaps * caps);
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static GstCaps *gst_aac_parse_sink_getcaps (GstBaseParse * parse,
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GstCaps * filter);
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static GstFlowReturn gst_aac_parse_handle_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame, gint * skipsize);
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static GstFlowReturn gst_aac_parse_pre_push_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame);
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static gboolean gst_aac_parse_src_event (GstBaseParse * parse,
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GstEvent * event);
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static gboolean gst_aac_parse_read_audio_specific_config (GstAacParse *
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aacparse, GstBitReader * br, gint * object_type, gint * sample_rate,
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gint * channels, gint * frame_samples);
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#define gst_aac_parse_parent_class parent_class
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G_DEFINE_TYPE (GstAacParse, gst_aac_parse, GST_TYPE_BASE_PARSE);
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GST_ELEMENT_REGISTER_DEFINE (aacparse, "aacparse",
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GST_RANK_PRIMARY + 1, GST_TYPE_AAC_PARSE);
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/**
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* gst_aac_parse_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_aac_parse_class_init (GstAacParseClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (aacparse_debug, "aacparse", 0,
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"AAC audio stream parser");
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gst_element_class_add_static_pad_template (element_class, &sink_template);
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gst_element_class_add_static_pad_template (element_class, &src_template);
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gst_element_class_set_static_metadata (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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parse_class->start = GST_DEBUG_FUNCPTR (gst_aac_parse_start);
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parse_class->stop = GST_DEBUG_FUNCPTR (gst_aac_parse_stop);
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parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_aac_parse_sink_setcaps);
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parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_aac_parse_sink_getcaps);
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parse_class->handle_frame = GST_DEBUG_FUNCPTR (gst_aac_parse_handle_frame);
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parse_class->pre_push_frame =
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GST_DEBUG_FUNCPTR (gst_aac_parse_pre_push_frame);
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parse_class->src_event = GST_DEBUG_FUNCPTR (gst_aac_parse_src_event);
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}
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/**
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* gst_aac_parse_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_aac_parse_init (GstAacParse * aacparse)
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{
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GST_DEBUG ("initialized");
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GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (aacparse));
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (aacparse));
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aacparse->last_parsed_sample_rate = 0;
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aacparse->last_parsed_channels = 0;
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}
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/**
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* gst_aac_parse_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_aac_parse_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, *peercaps;
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gboolean res = FALSE;
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const gchar *stream_format;
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guint8 codec_data[2];
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guint16 codec_data_data;
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gint sample_rate_idx;
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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_empty_simple ("audio/mpeg");
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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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aacparse->output_header_type = aacparse->header_type;
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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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case DSPAAC_HEADER_LOAS:
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stream_format = "loas";
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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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/* Generate codec data to be able to set profile/level on the caps */
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sample_rate_idx =
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gst_codec_utils_aac_get_index_from_sample_rate (aacparse->sample_rate);
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if (sample_rate_idx < 0)
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goto not_a_known_rate;
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codec_data_data =
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(aacparse->object_type << 11) |
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(sample_rate_idx << 7) | (aacparse->channels << 3);
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GST_WRITE_UINT16_BE (codec_data, codec_data_data);
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gst_codec_utils_aac_caps_set_level_and_profile (src_caps, codec_data, 2);
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s = gst_caps_get_structure (src_caps, 0);
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if (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 (aacparse->channels > 0)
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gst_structure_set (s, "channels", G_TYPE_INT, aacparse->channels, NULL);
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if (stream_format)
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gst_structure_set (s, "stream-format", G_TYPE_STRING, stream_format, NULL);
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peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (aacparse), NULL);
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if (peercaps && !gst_caps_can_intersect (src_caps, peercaps)) {
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GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
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"Caps can not intersect");
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if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
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GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
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"Input is ADTS, trying raw");
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gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "raw",
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NULL);
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if (gst_caps_can_intersect (src_caps, peercaps)) {
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GstBuffer *codec_data_buffer;
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GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
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"Caps can intersect, we will drop the ADTS layer");
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aacparse->output_header_type = DSPAAC_HEADER_NONE;
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/* The codec_data data is according to AudioSpecificConfig,
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ISO/IEC 14496-3, 1.6.2.1 */
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codec_data_buffer = gst_buffer_new_and_alloc (2);
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gst_buffer_fill (codec_data_buffer, 0, codec_data, 2);
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gst_caps_set_simple (src_caps, "codec_data", GST_TYPE_BUFFER,
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codec_data_buffer, NULL);
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gst_buffer_unref (codec_data_buffer);
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}
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} else if (aacparse->header_type == DSPAAC_HEADER_NONE) {
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GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
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"Input is raw, trying ADTS");
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gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "adts",
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NULL);
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if (gst_caps_can_intersect (src_caps, peercaps)) {
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GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
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"Caps can intersect, we will prepend ADTS headers");
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aacparse->output_header_type = DSPAAC_HEADER_ADTS;
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}
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}
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}
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if (peercaps)
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gst_caps_unref (peercaps);
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aacparse->last_parsed_channels = 0;
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aacparse->last_parsed_sample_rate = 0;
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GST_DEBUG_OBJECT (aacparse, "setting src caps: %" GST_PTR_FORMAT, src_caps);
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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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not_a_known_rate:
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GST_ERROR_OBJECT (aacparse, "Not a known sample rate: %d",
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aacparse->sample_rate);
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gst_caps_unref (src_caps);
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return FALSE;
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}
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/**
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* gst_aac_parse_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_aac_parse_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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const GValue *value;
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aacparse = GST_AAC_PARSE (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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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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if (buf && gst_buffer_get_size (buf) >= 2) {
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GstMapInfo map;
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GstBitReader br;
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if (!gst_buffer_map (buf, &map, GST_MAP_READ))
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return FALSE;
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gst_bit_reader_init (&br, map.data, map.size);
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gst_aac_parse_read_audio_specific_config (aacparse, &br,
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&aacparse->object_type, &aacparse->sample_rate, &aacparse->channels,
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&aacparse->frame_samples);
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aacparse->header_type = DSPAAC_HEADER_NONE;
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aacparse->mpegversion = 4;
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gst_buffer_unmap (buf, &map);
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GST_DEBUG ("codec_data: object_type=%d, sample_rate=%d, channels=%d, "
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"samples=%d", aacparse->object_type, aacparse->sample_rate,
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aacparse->channels, aacparse->frame_samples);
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/* arrange for metadata and get out of the way */
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gst_aac_parse_set_src_caps (aacparse, caps);
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if (aacparse->header_type == aacparse->output_header_type)
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gst_base_parse_set_passthrough (parse, TRUE);
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/* input is already correctly framed */
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gst_base_parse_set_min_frame_size (parse, RAW_MAX_SIZE);
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} else {
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return FALSE;
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}
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/* caps info overrides */
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gst_structure_get_int (structure, "rate", &aacparse->sample_rate);
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gst_structure_get_int (structure, "channels", &aacparse->channels);
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} else {
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const gchar *stream_format =
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gst_structure_get_string (structure, "stream-format");
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if (g_strcmp0 (stream_format, "raw") == 0) {
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GST_ERROR_OBJECT (parse, "Need codec_data for raw AAC");
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return FALSE;
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} else {
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aacparse->sample_rate = 0;
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aacparse->channels = 0;
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aacparse->header_type = DSPAAC_HEADER_NOT_PARSED;
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gst_base_parse_set_passthrough (parse, FALSE);
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}
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}
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return TRUE;
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}
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/**
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* gst_aac_parse_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_aac_parse_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_aac_parse_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_aac_parse_check_adts_frame (GstAacParse * aacparse,
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const guint8 * data, const guint avail, gboolean drain,
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guint * framesize, guint * needed_data)
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{
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guint crc_size;
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*needed_data = 0;
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/* Absolute minimum to perform the ADTS syncword,
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layer and sampling frequency tests */
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if (G_UNLIKELY (avail < 3)) {
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*needed_data = 3;
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return FALSE;
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}
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/* Syncword and layer tests */
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if ((data[0] == 0xff) && ((data[1] & 0xf6) == 0xf0)) {
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/* Sampling frequency test */
|
|
if (G_UNLIKELY ((data[2] & 0x3C) >> 2 == 15))
|
|
return FALSE;
|
|
|
|
/* This looks like an ADTS frame header but
|
|
we need at least 6 bytes to proceed */
|
|
if (G_UNLIKELY (avail < 6)) {
|
|
*needed_data = 6;
|
|
return FALSE;
|
|
}
|
|
|
|
*framesize = gst_aac_parse_adts_get_frame_len (data);
|
|
|
|
/* If frame has CRC, it needs 2 bytes
|
|
for it at the end of the header */
|
|
crc_size = (data[1] & 0x01) ? 0 : 2;
|
|
|
|
/* CRC size test */
|
|
if (*framesize < 7 + crc_size) {
|
|
*needed_data = 7 + crc_size;
|
|
return FALSE;
|
|
}
|
|
|
|
/* In EOS mode this is enough. No need to examine the data further.
|
|
We also relax the check when we have sync, on the assumption that
|
|
if we're not looking at random data, we have a much higher chance
|
|
to get the correct sync, and this avoids losing two frames when
|
|
a single bit corruption happens. */
|
|
if (drain || !GST_BASE_PARSE_LOST_SYNC (aacparse)) {
|
|
return TRUE;
|
|
}
|
|
|
|
if (*framesize + ADTS_MAX_SIZE > avail) {
|
|
/* We have found a possible frame header candidate, but can't be
|
|
sure since we don't have enough data to check the next frame */
|
|
GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
|
|
*framesize + ADTS_MAX_SIZE, avail);
|
|
*needed_data = *framesize + ADTS_MAX_SIZE;
|
|
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
|
|
*framesize + ADTS_MAX_SIZE);
|
|
return FALSE;
|
|
}
|
|
|
|
if ((data[*framesize] == 0xff) && ((data[*framesize + 1] & 0xf6) == 0xf0)) {
|
|
guint nextlen = gst_aac_parse_adts_get_frame_len (data + (*framesize));
|
|
|
|
GST_LOG ("ADTS frame found, len: %d bytes", *framesize);
|
|
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
|
|
nextlen + ADTS_MAX_SIZE);
|
|
return TRUE;
|
|
}
|
|
}
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_aac_parse_latm_get_value (GstAacParse * aacparse, GstBitReader * br,
|
|
guint32 * value)
|
|
{
|
|
guint8 bytes, i, byte;
|
|
|
|
*value = 0;
|
|
if (!gst_bit_reader_get_bits_uint8 (br, &bytes, 2))
|
|
return FALSE;
|
|
for (i = 0; i <= bytes; ++i) {
|
|
*value <<= 8;
|
|
if (!gst_bit_reader_get_bits_uint8 (br, &byte, 8))
|
|
return FALSE;
|
|
*value += byte;
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_aac_parse_get_audio_object_type (GstAacParse * aacparse, GstBitReader * br,
|
|
guint8 * audio_object_type)
|
|
{
|
|
if (!gst_bit_reader_get_bits_uint8 (br, audio_object_type, 5))
|
|
return FALSE;
|
|
if (*audio_object_type == 31) {
|
|
if (!gst_bit_reader_get_bits_uint8 (br, audio_object_type, 6))
|
|
return FALSE;
|
|
*audio_object_type += 32;
|
|
}
|
|
GST_LOG_OBJECT (aacparse, "audio object type %u", *audio_object_type);
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_aac_parse_get_audio_sample_rate (GstAacParse * aacparse, GstBitReader * br,
|
|
gint * sample_rate)
|
|
{
|
|
guint8 sampling_frequency_index;
|
|
if (!gst_bit_reader_get_bits_uint8 (br, &sampling_frequency_index, 4))
|
|
return FALSE;
|
|
GST_LOG_OBJECT (aacparse, "sampling_frequency_index: %u",
|
|
sampling_frequency_index);
|
|
if (sampling_frequency_index == 0xf) {
|
|
guint32 sampling_rate;
|
|
if (!gst_bit_reader_get_bits_uint32 (br, &sampling_rate, 24))
|
|
return FALSE;
|
|
*sample_rate = sampling_rate;
|
|
} else {
|
|
*sample_rate = loas_sample_rate_table[sampling_frequency_index];
|
|
if (!*sample_rate)
|
|
return FALSE;
|
|
}
|
|
aacparse->last_parsed_sample_rate = *sample_rate;
|
|
return TRUE;
|
|
}
|
|
|
|
/* See table 1.13 in ISO/IEC 14496-3 */
|
|
static gboolean
|
|
gst_aac_parse_read_audio_specific_config (GstAacParse * aacparse,
|
|
GstBitReader * br, gint * object_type, gint * sample_rate, gint * channels,
|
|
gint * frame_samples)
|
|
{
|
|
guint8 audio_object_type;
|
|
guint8 G_GNUC_UNUSED extension_audio_object_type;
|
|
guint8 channel_configuration, extension_channel_configuration;
|
|
gboolean G_GNUC_UNUSED sbr = FALSE, ps = FALSE;
|
|
|
|
if (!gst_aac_parse_get_audio_object_type (aacparse, br, &audio_object_type))
|
|
return FALSE;
|
|
if (object_type)
|
|
*object_type = audio_object_type;
|
|
|
|
if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
|
|
return FALSE;
|
|
|
|
if (!gst_bit_reader_get_bits_uint8 (br, &channel_configuration, 4))
|
|
return FALSE;
|
|
*channels = loas_channels_table[channel_configuration];
|
|
GST_LOG_OBJECT (aacparse, "channel_configuration: %d", channel_configuration);
|
|
if (!*channels)
|
|
return FALSE;
|
|
|
|
if (audio_object_type == 5 || audio_object_type == 29) {
|
|
extension_audio_object_type = 5;
|
|
sbr = TRUE;
|
|
if (audio_object_type == 29) {
|
|
ps = TRUE;
|
|
/* Parametric stereo. If we have a one-channel configuration, we can
|
|
* override it to stereo */
|
|
if (*channels == 1)
|
|
*channels = 2;
|
|
}
|
|
|
|
GST_LOG_OBJECT (aacparse,
|
|
"Audio object type 5 or 29, so rereading sampling rate (was %d)...",
|
|
*sample_rate);
|
|
if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
|
|
return FALSE;
|
|
|
|
if (!gst_aac_parse_get_audio_object_type (aacparse, br, &audio_object_type))
|
|
return FALSE;
|
|
|
|
if (audio_object_type == 22) {
|
|
/* extension channel configuration */
|
|
if (!gst_bit_reader_get_bits_uint8 (br, &extension_channel_configuration,
|
|
4))
|
|
return FALSE;
|
|
GST_LOG_OBJECT (aacparse, "extension channel_configuration: %d",
|
|
extension_channel_configuration);
|
|
*channels = loas_channels_table[extension_channel_configuration];
|
|
if (!*channels)
|
|
return FALSE;
|
|
}
|
|
} else {
|
|
extension_audio_object_type = 0;
|
|
}
|
|
|
|
GST_INFO_OBJECT (aacparse, "Parsed AudioSpecificConfig: %d Hz, %d channels",
|
|
*sample_rate, *channels);
|
|
|
|
if (frame_samples && audio_object_type == 23) {
|
|
guint8 frame_flag;
|
|
/* Read the Decoder Configuration (GASpecificConfig) if present */
|
|
/* We only care about the first bit to know what the number of samples
|
|
* in a frame is */
|
|
if (!gst_bit_reader_get_bits_uint8 (br, &frame_flag, 1))
|
|
return FALSE;
|
|
*frame_samples = frame_flag ? 960 : 1024;
|
|
}
|
|
|
|
/* There's LOTS of stuff next, but we ignore it for now as we have
|
|
what we want (sample rate and number of channels */
|
|
GST_DEBUG_OBJECT (aacparse,
|
|
"Need more code to parse humongous LOAS data, currently ignored");
|
|
aacparse->last_parsed_channels = *channels;
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_aac_parse_read_loas_config (GstAacParse * aacparse, const guint8 * data,
|
|
guint avail, gint * sample_rate, gint * channels, gint * version)
|
|
{
|
|
GstBitReader br;
|
|
guint8 u8, v, vA;
|
|
|
|
/* No version in the bitstream, but the spec has LOAS in the MPEG-4 section */
|
|
if (version)
|
|
*version = 4;
|
|
|
|
gst_bit_reader_init (&br, data, avail);
|
|
|
|
/* skip sync word (11 bits) and size (13 bits) */
|
|
if (!gst_bit_reader_skip (&br, 11 + 13))
|
|
return FALSE;
|
|
|
|
/* First bit is "use last config" */
|
|
if (!gst_bit_reader_get_bits_uint8 (&br, &u8, 1))
|
|
return FALSE;
|
|
if (u8) {
|
|
GST_LOG_OBJECT (aacparse, "Frame uses previous config");
|
|
if (!aacparse->last_parsed_sample_rate || !aacparse->last_parsed_channels) {
|
|
GST_DEBUG_OBJECT (aacparse,
|
|
"No previous config to use. We'll look for more data.");
|
|
return FALSE;
|
|
}
|
|
*sample_rate = aacparse->last_parsed_sample_rate;
|
|
*channels = aacparse->last_parsed_channels;
|
|
return TRUE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (aacparse, "Frame contains new config");
|
|
|
|
/* audioMuxVersion */
|
|
if (!gst_bit_reader_get_bits_uint8 (&br, &v, 1))
|
|
return FALSE;
|
|
if (v) {
|
|
/* audioMuxVersionA */
|
|
if (!gst_bit_reader_get_bits_uint8 (&br, &vA, 1))
|
|
return FALSE;
|
|
} else
|
|
vA = 0;
|
|
|
|
GST_LOG_OBJECT (aacparse, "v %d, vA %d", v, vA);
|
|
if (vA == 0) {
|
|
guint8 same_time, subframes, num_program, prog;
|
|
if (v == 1) {
|
|
guint32 value;
|
|
/* taraBufferFullness */
|
|
if (!gst_aac_parse_latm_get_value (aacparse, &br, &value))
|
|
return FALSE;
|
|
}
|
|
if (!gst_bit_reader_get_bits_uint8 (&br, &same_time, 1))
|
|
return FALSE;
|
|
if (!gst_bit_reader_get_bits_uint8 (&br, &subframes, 6))
|
|
return FALSE;
|
|
if (!gst_bit_reader_get_bits_uint8 (&br, &num_program, 4))
|
|
return FALSE;
|
|
GST_LOG_OBJECT (aacparse, "same_time %d, subframes %d, num_program %d",
|
|
same_time, subframes, num_program);
|
|
|
|
for (prog = 0; prog <= num_program; ++prog) {
|
|
guint8 num_layer, layer;
|
|
if (!gst_bit_reader_get_bits_uint8 (&br, &num_layer, 3))
|
|
return FALSE;
|
|
GST_LOG_OBJECT (aacparse, "Program %d: %d layers", prog, num_layer);
|
|
|
|
for (layer = 0; layer <= num_layer; ++layer) {
|
|
guint8 use_same_config;
|
|
if (prog == 0 && layer == 0) {
|
|
use_same_config = 0;
|
|
} else {
|
|
if (!gst_bit_reader_get_bits_uint8 (&br, &use_same_config, 1))
|
|
return FALSE;
|
|
}
|
|
if (!use_same_config) {
|
|
if (v == 0) {
|
|
if (!gst_aac_parse_read_audio_specific_config (aacparse, &br, NULL,
|
|
sample_rate, channels, NULL))
|
|
return FALSE;
|
|
} else {
|
|
guint32 asc_len;
|
|
if (!gst_aac_parse_latm_get_value (aacparse, &br, &asc_len))
|
|
return FALSE;
|
|
if (!gst_aac_parse_read_audio_specific_config (aacparse, &br, NULL,
|
|
sample_rate, channels, NULL))
|
|
return FALSE;
|
|
if (!gst_bit_reader_skip (&br, asc_len))
|
|
return FALSE;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
GST_LOG_OBJECT (aacparse, "More data ignored");
|
|
} else {
|
|
GST_WARNING_OBJECT (aacparse, "Spec says \"TBD\"...");
|
|
return FALSE;
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_aac_parse_loas_get_frame_len:
|
|
* @data: block of data containing a LOAS header.
|
|
*
|
|
* This function calculates LOAS frame length from the given header.
|
|
*
|
|
* Returns: size of the LOAS frame.
|
|
*/
|
|
static inline guint
|
|
gst_aac_parse_loas_get_frame_len (const guint8 * data)
|
|
{
|
|
return (((data[1] & 0x1f) << 8) | data[2]) + 3;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_aac_parse_check_loas_frame:
|
|
* @aacparse: #GstAacParse.
|
|
* @data: Data to be checked.
|
|
* @avail: Amount of data passed.
|
|
* @framesize: If valid LOAS frame was found, this will be set to tell the
|
|
* found frame size in bytes.
|
|
* @needed_data: If frame was not found, this may be set to tell how much
|
|
* more data is needed in the next round to detect the frame
|
|
* reliably. This may happen when a frame header candidate
|
|
* is found but it cannot be guaranteed to be the header without
|
|
* peeking the following data.
|
|
*
|
|
* Check if the given data contains contains LOAS frame. The algorithm
|
|
* will examine LOAS frame header and calculate the frame size. Also, another
|
|
* consecutive LOAS frame header need to be present after the found frame.
|
|
* Otherwise the data is not considered as a valid LOAS frame. However, this
|
|
* "extra check" is omitted when EOS has been received. In this case it is
|
|
* enough when data[0] contains a valid LOAS header.
|
|
*
|
|
* This function may set the #needed_data to indicate that a possible frame
|
|
* candidate has been found, but more data (#needed_data bytes) is needed to
|
|
* be absolutely sure. When this situation occurs, FALSE will be returned.
|
|
*
|
|
* When a valid frame is detected, this function will use
|
|
* gst_base_parse_set_min_frame_size() function from #GstBaseParse class
|
|
* to set the needed bytes for next frame.This way next data chunk is already
|
|
* of correct size.
|
|
*
|
|
* LOAS can have three different formats, if I read the spec correctly. Only
|
|
* one of them is supported here, as the two samples I have use this one.
|
|
*
|
|
* Returns: TRUE if the given data contains a valid LOAS header.
|
|
*/
|
|
static gboolean
|
|
gst_aac_parse_check_loas_frame (GstAacParse * aacparse,
|
|
const guint8 * data, const guint avail, gboolean drain,
|
|
guint * framesize, guint * needed_data)
|
|
{
|
|
*needed_data = 0;
|
|
|
|
/* 3 byte header */
|
|
if (G_UNLIKELY (avail < 3)) {
|
|
*needed_data = 3;
|
|
return FALSE;
|
|
}
|
|
|
|
if ((data[0] == 0x56) && ((data[1] & 0xe0) == 0xe0)) {
|
|
*framesize = gst_aac_parse_loas_get_frame_len (data);
|
|
GST_DEBUG_OBJECT (aacparse, "Found possible %u byte LOAS frame",
|
|
*framesize);
|
|
|
|
/* In EOS mode this is enough. No need to examine the data further.
|
|
We also relax the check when we have sync, on the assumption that
|
|
if we're not looking at random data, we have a much higher chance
|
|
to get the correct sync, and this avoids losing two frames when
|
|
a single bit corruption happens. */
|
|
if (drain || !GST_BASE_PARSE_LOST_SYNC (aacparse)) {
|
|
return TRUE;
|
|
}
|
|
|
|
if (*framesize + LOAS_MAX_SIZE > avail) {
|
|
/* We have found a possible frame header candidate, but can't be
|
|
sure since we don't have enough data to check the next frame */
|
|
GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
|
|
*framesize + LOAS_MAX_SIZE, avail);
|
|
*needed_data = *framesize + LOAS_MAX_SIZE;
|
|
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
|
|
*framesize + LOAS_MAX_SIZE);
|
|
return FALSE;
|
|
}
|
|
|
|
if ((data[*framesize] == 0x56) && ((data[*framesize + 1] & 0xe0) == 0xe0)) {
|
|
guint nextlen = gst_aac_parse_loas_get_frame_len (data + (*framesize));
|
|
|
|
GST_LOG ("LOAS frame found, len: %d bytes", *framesize);
|
|
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
|
|
nextlen + LOAS_MAX_SIZE);
|
|
return TRUE;
|
|
} else {
|
|
GST_DEBUG_OBJECT (aacparse, "That was a false positive");
|
|
}
|
|
}
|
|
return FALSE;
|
|
}
|
|
|
|
/* caller ensure sufficient data */
|
|
static inline void
|
|
gst_aac_parse_parse_adts_header (GstAacParse * aacparse, const guint8 * data,
|
|
gint * rate, gint * channels, gint * object, gint * version)
|
|
{
|
|
|
|
if (rate) {
|
|
gint sr_idx = (data[2] & 0x3c) >> 2;
|
|
|
|
*rate = gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
|
|
}
|
|
if (channels) {
|
|
*channels = ((data[2] & 0x01) << 2) | ((data[3] & 0xc0) >> 6);
|
|
if (*channels == 7)
|
|
*channels = 8;
|
|
}
|
|
|
|
if (version)
|
|
*version = (data[1] & 0x08) ? 2 : 4;
|
|
if (object)
|
|
*object = ((data[2] & 0xc0) >> 6) + 1;
|
|
}
|
|
|
|
/**
|
|
* gst_aac_parse_detect_stream:
|
|
* @aacparse: #GstAacParse.
|
|
* @data: A block of data that needs to be examined for stream characteristics.
|
|
* @avail: Size of the given datablock.
|
|
* @framesize: If valid stream was found, this will be set to tell the
|
|
* first frame size in bytes.
|
|
* @skipsize: If valid stream was found, this will be set to tell the first
|
|
* audio frame position within the given data.
|
|
*
|
|
* Examines the given piece of data and try to detect the format of it. It
|
|
* checks for "ADIF" header (in the beginning of the clip) and ADTS frame
|
|
* header. If the stream is detected, TRUE will be returned and #framesize
|
|
* is set to indicate the found frame size. Additionally, #skipsize might
|
|
* be set to indicate the number of bytes that need to be skipped, a.k.a. the
|
|
* position of the frame inside given data chunk.
|
|
*
|
|
* Returns: TRUE on success.
|
|
*/
|
|
static gboolean
|
|
gst_aac_parse_detect_stream (GstAacParse * aacparse,
|
|
const guint8 * data, const guint avail, gboolean drain,
|
|
guint * framesize, gint * skipsize)
|
|
{
|
|
gboolean found = FALSE;
|
|
guint need_data_adts = 0, need_data_loas;
|
|
guint i = 0;
|
|
|
|
GST_DEBUG_OBJECT (aacparse, "Parsing header data");
|
|
|
|
/* FIXME: No need to check for ADIF if we are not in the beginning of the
|
|
stream */
|
|
|
|
/* Can we even parse the header? */
|
|
if (avail < MAX (ADTS_MAX_SIZE, LOAS_MAX_SIZE)) {
|
|
GST_DEBUG_OBJECT (aacparse, "Not enough data to check");
|
|
return FALSE;
|
|
}
|
|
|
|
for (i = 0; i < avail - 4; i++) {
|
|
if (((data[i] == 0xff) && ((data[i + 1] & 0xf6) == 0xf0)) ||
|
|
((data[i] == 0x56) && ((data[i + 1] & 0xe0) == 0xe0)) ||
|
|
strncmp ((char *) data + i, "ADIF", 4) == 0) {
|
|
GST_DEBUG_OBJECT (aacparse, "Found signature at offset %u", i);
|
|
found = TRUE;
|
|
|
|
if (i) {
|
|
/* Trick: tell the parent class that we didn't find the frame yet,
|
|
but make it skip 'i' amount of bytes. Next time we arrive
|
|
here we have full frame in the beginning of the data. */
|
|
*skipsize = i;
|
|
return FALSE;
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
if (!found) {
|
|
if (i)
|
|
*skipsize = i;
|
|
return FALSE;
|
|
}
|
|
|
|
if (gst_aac_parse_check_adts_frame (aacparse, data, avail, drain,
|
|
framesize, &need_data_adts)) {
|
|
gint rate, channels;
|
|
|
|
GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize);
|
|
|
|
gst_aac_parse_parse_adts_header (aacparse, data, &rate, &channels,
|
|
&aacparse->object_type, &aacparse->mpegversion);
|
|
|
|
if (!channels || !framesize) {
|
|
GST_DEBUG_OBJECT (aacparse, "impossible ADTS configuration");
|
|
return FALSE;
|
|
}
|
|
|
|
aacparse->header_type = DSPAAC_HEADER_ADTS;
|
|
gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
|
|
aacparse->frame_samples, 2, 2);
|
|
|
|
GST_DEBUG ("ADTS: samplerate %d, channels %d, objtype %d, version %d",
|
|
rate, channels, aacparse->object_type, aacparse->mpegversion);
|
|
|
|
gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
if (gst_aac_parse_check_loas_frame (aacparse, data, avail, drain,
|
|
framesize, &need_data_loas)) {
|
|
gint rate = 0, channels = 0;
|
|
|
|
GST_INFO ("LOAS, framesize: %d", *framesize);
|
|
|
|
aacparse->header_type = DSPAAC_HEADER_LOAS;
|
|
|
|
if (!gst_aac_parse_read_loas_config (aacparse, data, avail, &rate,
|
|
&channels, &aacparse->mpegversion)) {
|
|
/* This is pretty normal when skipping data at the start of
|
|
* random stream (MPEG-TS capture for example) */
|
|
GST_LOG_OBJECT (aacparse, "Error reading LOAS config");
|
|
return FALSE;
|
|
}
|
|
|
|
if (rate && channels) {
|
|
gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
|
|
aacparse->frame_samples, 2, 2);
|
|
|
|
/* Don't store the sample rate and channels yet -
|
|
* this is just format detection. */
|
|
GST_DEBUG ("LOAS: samplerate %d, channels %d, objtype %d, version %d",
|
|
rate, channels, aacparse->object_type, aacparse->mpegversion);
|
|
}
|
|
|
|
gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
if (need_data_adts || need_data_loas) {
|
|
/* 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;
|
|
GstCaps *sinkcaps;
|
|
|
|
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_codec_utils_aac_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);
|
|
|
|
/* arrange for metadata and get out of the way */
|
|
sinkcaps = gst_pad_get_current_caps (GST_BASE_PARSE_SINK_PAD (aacparse));
|
|
gst_aac_parse_set_src_caps (aacparse, sinkcaps);
|
|
if (sinkcaps)
|
|
gst_caps_unref (sinkcaps);
|
|
|
|
/* not syncable, not easily seekable (unless we push data from start */
|
|
gst_base_parse_set_syncable (GST_BASE_PARSE_CAST (aacparse), FALSE);
|
|
gst_base_parse_set_passthrough (GST_BASE_PARSE_CAST (aacparse), TRUE);
|
|
gst_base_parse_set_average_bitrate (GST_BASE_PARSE_CAST (aacparse), 0);
|
|
|
|
*framesize = avail;
|
|
return TRUE;
|
|
}
|
|
|
|
/* This should never happen */
|
|
return FALSE;
|
|
}
|
|
|
|
/**
|
|
* gst_aac_parse_get_audio_profile_object_type
|
|
* @aacparse: #GstAacParse.
|
|
*
|
|
* Gets the MPEG-2 profile or the MPEG-4 object type value corresponding to the
|
|
* mpegversion and profile of @aacparse's src pad caps, according to the
|
|
* values defined by table 1.A.11 in ISO/IEC 14496-3.
|
|
*
|
|
* Returns: the profile or object type value corresponding to @aacparse's src
|
|
* pad caps, if such a value exists; otherwise G_MAXUINT8.
|
|
*/
|
|
static guint8
|
|
gst_aac_parse_get_audio_profile_object_type (GstAacParse * aacparse)
|
|
{
|
|
GstCaps *srccaps;
|
|
GstStructure *srcstruct;
|
|
const gchar *profile;
|
|
guint8 ret;
|
|
|
|
srccaps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse));
|
|
if (G_UNLIKELY (srccaps == NULL)) {
|
|
return G_MAXUINT8;
|
|
}
|
|
|
|
srcstruct = gst_caps_get_structure (srccaps, 0);
|
|
profile = gst_structure_get_string (srcstruct, "profile");
|
|
if (G_UNLIKELY (profile == NULL)) {
|
|
gst_caps_unref (srccaps);
|
|
return G_MAXUINT8;
|
|
}
|
|
|
|
if (g_strcmp0 (profile, "main") == 0) {
|
|
ret = (guint8) 0U;
|
|
} else if (g_strcmp0 (profile, "lc") == 0) {
|
|
ret = (guint8) 1U;
|
|
} else if (g_strcmp0 (profile, "ssr") == 0) {
|
|
ret = (guint8) 2U;
|
|
} else if (g_strcmp0 (profile, "ltp") == 0) {
|
|
if (G_LIKELY (aacparse->mpegversion == 4))
|
|
ret = (guint8) 3U;
|
|
else
|
|
ret = G_MAXUINT8; /* LTP Object Type allowed only for MPEG-4 */
|
|
} else {
|
|
ret = G_MAXUINT8;
|
|
}
|
|
|
|
gst_caps_unref (srccaps);
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_aac_parse_get_audio_channel_configuration
|
|
* @num_channels: number of audio channels.
|
|
*
|
|
* Gets the Channel Configuration value, as defined by table 1.19 in ISO/IEC
|
|
* 14496-3, for a given number of audio channels.
|
|
*
|
|
* Returns: the Channel Configuration value corresponding to @num_channels, if
|
|
* such a value exists; otherwise G_MAXUINT8.
|
|
*/
|
|
static guint8
|
|
gst_aac_parse_get_audio_channel_configuration (gint num_channels)
|
|
{
|
|
if (num_channels >= 1 && num_channels <= 6) /* Mono up to & including 5.1 */
|
|
return (guint8) num_channels;
|
|
else if (num_channels == 8) /* 7.1 */
|
|
return (guint8) 7U;
|
|
else
|
|
return G_MAXUINT8;
|
|
|
|
/* FIXME: Add support for configurations 11, 12 and 14 from
|
|
* ISO/IEC 14496-3:2009/PDAM 4 based on the actual channel layout
|
|
*/
|
|
}
|
|
|
|
/**
|
|
* gst_aac_parse_get_audio_sampling_frequency_index:
|
|
* @sample_rate: audio sampling rate.
|
|
*
|
|
* Gets the Sampling Frequency Index value, as defined by table 1.18 in ISO/IEC
|
|
* 14496-3, for a given sampling rate.
|
|
*
|
|
* Returns: the Sampling Frequency Index value corresponding to @sample_rate,
|
|
* if such a value exists; otherwise G_MAXUINT8.
|
|
*/
|
|
static guint8
|
|
gst_aac_parse_get_audio_sampling_frequency_index (gint sample_rate)
|
|
{
|
|
switch (sample_rate) {
|
|
case 96000:
|
|
return 0x0U;
|
|
case 88200:
|
|
return 0x1U;
|
|
case 64000:
|
|
return 0x2U;
|
|
case 48000:
|
|
return 0x3U;
|
|
case 44100:
|
|
return 0x4U;
|
|
case 32000:
|
|
return 0x5U;
|
|
case 24000:
|
|
return 0x6U;
|
|
case 22050:
|
|
return 0x7U;
|
|
case 16000:
|
|
return 0x8U;
|
|
case 12000:
|
|
return 0x9U;
|
|
case 11025:
|
|
return 0xAU;
|
|
case 8000:
|
|
return 0xBU;
|
|
case 7350:
|
|
return 0xCU;
|
|
default:
|
|
return G_MAXUINT8;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_aac_parse_prepend_adts_headers:
|
|
* @aacparse: #GstAacParse.
|
|
* @frame: raw AAC frame to which ADTS headers shall be prepended.
|
|
*
|
|
* Prepends ADTS headers to a raw AAC audio frame.
|
|
*
|
|
* Returns: TRUE if ADTS headers were successfully prepended; FALSE otherwise.
|
|
*/
|
|
static gboolean
|
|
gst_aac_parse_prepend_adts_headers (GstAacParse * aacparse,
|
|
GstBaseParseFrame * frame)
|
|
{
|
|
GstMemory *mem;
|
|
guint8 *adts_headers;
|
|
gsize buf_size;
|
|
gsize frame_size;
|
|
guint8 id, profile, channel_configuration, sampling_frequency_index;
|
|
|
|
id = (aacparse->mpegversion == 4) ? 0x0U : 0x1U;
|
|
profile = gst_aac_parse_get_audio_profile_object_type (aacparse);
|
|
if (profile == G_MAXUINT8) {
|
|
GST_ERROR_OBJECT (aacparse, "Unsupported audio profile or object type");
|
|
return FALSE;
|
|
}
|
|
channel_configuration =
|
|
gst_aac_parse_get_audio_channel_configuration (aacparse->channels);
|
|
if (channel_configuration == G_MAXUINT8) {
|
|
GST_ERROR_OBJECT (aacparse, "Unsupported number of channels");
|
|
return FALSE;
|
|
}
|
|
sampling_frequency_index =
|
|
gst_aac_parse_get_audio_sampling_frequency_index (aacparse->sample_rate);
|
|
if (sampling_frequency_index == G_MAXUINT8) {
|
|
GST_ERROR_OBJECT (aacparse, "Unsupported sampling frequency");
|
|
return FALSE;
|
|
}
|
|
|
|
frame->out_buffer = gst_buffer_copy (frame->buffer);
|
|
buf_size = gst_buffer_get_size (frame->out_buffer);
|
|
frame_size = buf_size + ADTS_HEADERS_LENGTH;
|
|
|
|
if (G_UNLIKELY (frame_size >= 0x4000)) {
|
|
GST_ERROR_OBJECT (aacparse, "Frame size is too big for ADTS");
|
|
return FALSE;
|
|
}
|
|
|
|
adts_headers = (guint8 *) g_malloc0 (ADTS_HEADERS_LENGTH);
|
|
|
|
/* Note: no error correction bits are added to the resulting ADTS frames */
|
|
adts_headers[0] = 0xFFU;
|
|
adts_headers[1] = 0xF0U | (id << 3) | 0x1U;
|
|
adts_headers[2] = (profile << 6) | (sampling_frequency_index << 2) | 0x2U |
|
|
((channel_configuration & 0x4U) >> 2);
|
|
adts_headers[3] = ((channel_configuration & 0x3U) << 6) | 0x30U |
|
|
(guint8) (frame_size >> 11);
|
|
adts_headers[4] = (guint8) ((frame_size >> 3) & 0x00FF);
|
|
adts_headers[5] = (guint8) (((frame_size & 0x0007) << 5) + 0x1FU);
|
|
adts_headers[6] = 0xFCU;
|
|
|
|
mem = gst_memory_new_wrapped (0, adts_headers, ADTS_HEADERS_LENGTH, 0,
|
|
ADTS_HEADERS_LENGTH, adts_headers, g_free);
|
|
gst_buffer_prepend_memory (frame->out_buffer, mem);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_aac_parse_check_valid_frame:
|
|
* @parse: #GstBaseParse.
|
|
* @frame: #GstBaseParseFrame.
|
|
* @skipsize: How much data parent class should skip in order to find the
|
|
* frame header.
|
|
*
|
|
* Implementation of "handle_frame" vmethod in #GstBaseParse class.
|
|
*
|
|
* Also determines frame overhead.
|
|
* ADTS streams have a 7 byte header in each frame. MP4 and ADIF streams don't have
|
|
* a per-frame header. LOAS has 3 bytes.
|
|
*
|
|
* 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
|
|
*
|
|
* Returns: a #GstFlowReturn.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_aac_parse_handle_frame (GstBaseParse * parse,
|
|
GstBaseParseFrame * frame, gint * skipsize)
|
|
{
|
|
GstMapInfo map;
|
|
GstAacParse *aacparse;
|
|
gboolean ret = FALSE;
|
|
gboolean lost_sync;
|
|
GstBuffer *buffer;
|
|
guint framesize;
|
|
gint rate = 0, channels = 0;
|
|
|
|
aacparse = GST_AAC_PARSE (parse);
|
|
buffer = frame->buffer;
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
|
|
*skipsize = -1;
|
|
lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
|
|
|
|
if (aacparse->header_type == DSPAAC_HEADER_ADIF ||
|
|
aacparse->header_type == DSPAAC_HEADER_NONE) {
|
|
/* There is nothing to parse */
|
|
framesize = map.size;
|
|
ret = TRUE;
|
|
|
|
} else if (aacparse->header_type == DSPAAC_HEADER_NOT_PARSED || lost_sync) {
|
|
|
|
ret = gst_aac_parse_detect_stream (aacparse, map.data, map.size,
|
|
GST_BASE_PARSE_DRAINING (parse), &framesize, skipsize);
|
|
|
|
} else if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
|
|
guint needed_data = 1024;
|
|
|
|
ret = gst_aac_parse_check_adts_frame (aacparse, map.data, map.size,
|
|
GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
|
|
|
|
if (!ret && needed_data) {
|
|
GST_DEBUG ("buffer didn't contain valid frame");
|
|
*skipsize = 0;
|
|
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
|
|
needed_data);
|
|
}
|
|
|
|
} else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
|
|
guint needed_data = 1024;
|
|
|
|
ret = gst_aac_parse_check_loas_frame (aacparse, map.data,
|
|
map.size, GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
|
|
|
|
if (!ret && needed_data) {
|
|
GST_DEBUG ("buffer didn't contain valid frame");
|
|
*skipsize = 0;
|
|
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),
|
|
ADTS_MAX_SIZE);
|
|
}
|
|
|
|
if (G_UNLIKELY (!ret))
|
|
goto exit;
|
|
|
|
if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
|
|
/* see above */
|
|
frame->overhead = 7;
|
|
|
|
gst_aac_parse_parse_adts_header (aacparse, map.data,
|
|
&rate, &channels, NULL, NULL);
|
|
|
|
GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels);
|
|
|
|
if (G_UNLIKELY (rate != aacparse->sample_rate
|
|
|| channels != aacparse->channels)) {
|
|
aacparse->sample_rate = rate;
|
|
aacparse->channels = channels;
|
|
|
|
if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
|
|
/* If linking fails, we need to return appropriate error */
|
|
ret = GST_FLOW_NOT_LINKED;
|
|
}
|
|
|
|
gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
|
|
aacparse->sample_rate, aacparse->frame_samples, 2, 2);
|
|
}
|
|
} else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
|
|
gboolean setcaps = FALSE;
|
|
|
|
/* see above */
|
|
frame->overhead = 3;
|
|
|
|
if (!gst_aac_parse_read_loas_config (aacparse, map.data, map.size, &rate,
|
|
&channels, NULL) || !rate || !channels) {
|
|
/* This is pretty normal when skipping data at the start of
|
|
* random stream (MPEG-TS capture for example) */
|
|
GST_DEBUG_OBJECT (aacparse, "Error reading LOAS config. Skipping.");
|
|
/* Since we don't fully parse the LOAS config, we don't know for sure
|
|
* how much to skip. Just skip 1 to end up to the next marker and
|
|
* resume parsing from there */
|
|
*skipsize = 1;
|
|
goto exit;
|
|
}
|
|
|
|
if (G_UNLIKELY (rate != aacparse->sample_rate
|
|
|| channels != aacparse->channels)) {
|
|
aacparse->sample_rate = rate;
|
|
aacparse->channels = channels;
|
|
setcaps = TRUE;
|
|
GST_INFO_OBJECT (aacparse, "New LOAS config: %d Hz, %d channels", rate,
|
|
channels);
|
|
}
|
|
|
|
/* We want to set caps both at start, and when rate/channels change.
|
|
Since only some LOAS frames have that info, we may receive frames
|
|
before knowing about rate/channels. */
|
|
if (setcaps
|
|
|| !gst_pad_has_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse))) {
|
|
if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
|
|
/* If linking fails, we need to return appropriate error */
|
|
ret = GST_FLOW_NOT_LINKED;
|
|
}
|
|
|
|
gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
|
|
aacparse->sample_rate, aacparse->frame_samples, 2, 2);
|
|
}
|
|
}
|
|
|
|
if (aacparse->header_type == DSPAAC_HEADER_NONE
|
|
&& aacparse->output_header_type == DSPAAC_HEADER_ADTS) {
|
|
if (!gst_aac_parse_prepend_adts_headers (aacparse, frame)) {
|
|
GST_ERROR_OBJECT (aacparse, "Failed to prepend ADTS headers to frame");
|
|
ret = GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
exit:
|
|
gst_buffer_unmap (buffer, &map);
|
|
|
|
if (ret) {
|
|
/* found, skip if needed */
|
|
if (*skipsize > 0)
|
|
return GST_FLOW_OK;
|
|
*skipsize = 0;
|
|
} else {
|
|
if (*skipsize < 0)
|
|
*skipsize = 1;
|
|
}
|
|
|
|
if (ret && framesize <= map.size) {
|
|
return gst_base_parse_finish_frame (parse, frame, framesize);
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_aac_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
|
|
{
|
|
GstAacParse *aacparse = GST_AAC_PARSE (parse);
|
|
|
|
if (!aacparse->sent_codec_tag) {
|
|
GstTagList *taglist;
|
|
GstCaps *caps;
|
|
|
|
/* codec tag */
|
|
caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
|
|
if (caps == NULL) {
|
|
if (GST_PAD_IS_FLUSHING (GST_BASE_PARSE_SRC_PAD (parse))) {
|
|
GST_INFO_OBJECT (parse, "Src pad is flushing");
|
|
return GST_FLOW_FLUSHING;
|
|
} else {
|
|
GST_INFO_OBJECT (parse, "Src pad is not negotiated!");
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
}
|
|
}
|
|
|
|
taglist = gst_tag_list_new_empty ();
|
|
gst_pb_utils_add_codec_description_to_tag_list (taglist,
|
|
GST_TAG_AUDIO_CODEC, caps);
|
|
gst_caps_unref (caps);
|
|
|
|
gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE);
|
|
gst_tag_list_unref (taglist);
|
|
|
|
/* also signals the end of first-frame processing */
|
|
aacparse->sent_codec_tag = TRUE;
|
|
}
|
|
|
|
/* As a special case, we can remove the ADTS framing and output raw AAC. */
|
|
if (aacparse->header_type == DSPAAC_HEADER_ADTS
|
|
&& aacparse->output_header_type == DSPAAC_HEADER_NONE) {
|
|
guint header_size;
|
|
GstMapInfo map;
|
|
frame->out_buffer = gst_buffer_make_writable (frame->buffer);
|
|
frame->buffer = NULL;
|
|
gst_buffer_map (frame->out_buffer, &map, GST_MAP_READ);
|
|
header_size = (map.data[1] & 1) ? 7 : 9; /* optional CRC */
|
|
gst_buffer_unmap (frame->out_buffer, &map);
|
|
gst_buffer_resize (frame->out_buffer, header_size,
|
|
gst_buffer_get_size (frame->out_buffer) - header_size);
|
|
}
|
|
|
|
frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP;
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_aac_parse_start:
|
|
* @parse: #GstBaseParse.
|
|
*
|
|
* Implementation of "start" vmethod in #GstBaseParse class.
|
|
*
|
|
* Returns: TRUE if startup succeeded.
|
|
*/
|
|
static gboolean
|
|
gst_aac_parse_start (GstBaseParse * parse)
|
|
{
|
|
GstAacParse *aacparse;
|
|
|
|
aacparse = GST_AAC_PARSE (parse);
|
|
GST_DEBUG ("start");
|
|
aacparse->frame_samples = 1024;
|
|
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), ADTS_MAX_SIZE);
|
|
aacparse->sent_codec_tag = FALSE;
|
|
aacparse->last_parsed_channels = 0;
|
|
aacparse->last_parsed_sample_rate = 0;
|
|
aacparse->object_type = 0;
|
|
aacparse->bitrate = 0;
|
|
aacparse->header_type = DSPAAC_HEADER_NOT_PARSED;
|
|
aacparse->output_header_type = DSPAAC_HEADER_NOT_PARSED;
|
|
aacparse->channels = 0;
|
|
aacparse->sample_rate = 0;
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_aac_parse_stop:
|
|
* @parse: #GstBaseParse.
|
|
*
|
|
* Implementation of "stop" vmethod in #GstBaseParse class.
|
|
*
|
|
* Returns: TRUE is stopping succeeded.
|
|
*/
|
|
static gboolean
|
|
gst_aac_parse_stop (GstBaseParse * parse)
|
|
{
|
|
GST_DEBUG ("stop");
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
remove_fields (GstCaps * caps)
|
|
{
|
|
guint i, n;
|
|
|
|
n = gst_caps_get_size (caps);
|
|
for (i = 0; i < n; i++) {
|
|
GstStructure *s = gst_caps_get_structure (caps, i);
|
|
|
|
gst_structure_remove_field (s, "framed");
|
|
}
|
|
}
|
|
|
|
static void
|
|
add_conversion_fields (GstCaps * caps)
|
|
{
|
|
guint i, n;
|
|
|
|
n = gst_caps_get_size (caps);
|
|
for (i = 0; i < n; i++) {
|
|
GstStructure *s = gst_caps_get_structure (caps, i);
|
|
|
|
if (gst_structure_has_field (s, "stream-format")) {
|
|
const GValue *v = gst_structure_get_value (s, "stream-format");
|
|
|
|
if (G_VALUE_HOLDS_STRING (v)) {
|
|
const gchar *str = g_value_get_string (v);
|
|
|
|
if (strcmp (str, "adts") == 0 || strcmp (str, "raw") == 0) {
|
|
GValue va = G_VALUE_INIT;
|
|
GValue vs = G_VALUE_INIT;
|
|
|
|
g_value_init (&va, GST_TYPE_LIST);
|
|
g_value_init (&vs, G_TYPE_STRING);
|
|
g_value_set_string (&vs, "adts");
|
|
gst_value_list_append_value (&va, &vs);
|
|
g_value_set_string (&vs, "raw");
|
|
gst_value_list_append_value (&va, &vs);
|
|
gst_structure_set_value (s, "stream-format", &va);
|
|
g_value_unset (&va);
|
|
g_value_unset (&vs);
|
|
}
|
|
} else if (GST_VALUE_HOLDS_LIST (v)) {
|
|
gboolean contains_raw = FALSE;
|
|
gboolean contains_adts = FALSE;
|
|
guint m = gst_value_list_get_size (v), j;
|
|
|
|
for (j = 0; j < m; j++) {
|
|
const GValue *ve = gst_value_list_get_value (v, j);
|
|
const gchar *str;
|
|
|
|
if (G_VALUE_HOLDS_STRING (ve) && (str = g_value_get_string (ve))) {
|
|
if (strcmp (str, "adts") == 0)
|
|
contains_adts = TRUE;
|
|
else if (strcmp (str, "raw") == 0)
|
|
contains_raw = TRUE;
|
|
}
|
|
}
|
|
|
|
if (contains_adts || contains_raw) {
|
|
GValue va = G_VALUE_INIT;
|
|
GValue vs = G_VALUE_INIT;
|
|
|
|
g_value_init (&va, GST_TYPE_LIST);
|
|
g_value_init (&vs, G_TYPE_STRING);
|
|
g_value_copy (v, &va);
|
|
|
|
if (!contains_raw) {
|
|
g_value_set_string (&vs, "raw");
|
|
gst_value_list_append_value (&va, &vs);
|
|
}
|
|
if (!contains_adts) {
|
|
g_value_set_string (&vs, "adts");
|
|
gst_value_list_append_value (&va, &vs);
|
|
}
|
|
|
|
gst_structure_set_value (s, "stream-format", &va);
|
|
|
|
g_value_unset (&vs);
|
|
g_value_unset (&va);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_aac_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
|
|
{
|
|
GstCaps *peercaps, *templ;
|
|
GstCaps *res;
|
|
|
|
templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
|
|
|
|
if (filter) {
|
|
GstCaps *fcopy = gst_caps_copy (filter);
|
|
/* Remove the fields we convert */
|
|
remove_fields (fcopy);
|
|
add_conversion_fields (fcopy);
|
|
peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), fcopy);
|
|
gst_caps_unref (fcopy);
|
|
} else
|
|
peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), NULL);
|
|
|
|
if (peercaps) {
|
|
peercaps = gst_caps_make_writable (peercaps);
|
|
/* Remove the fields we convert */
|
|
remove_fields (peercaps);
|
|
add_conversion_fields (peercaps);
|
|
|
|
res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (peercaps);
|
|
gst_caps_unref (templ);
|
|
} else {
|
|
res = templ;
|
|
}
|
|
|
|
if (filter) {
|
|
GstCaps *intersection;
|
|
|
|
intersection =
|
|
gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (res);
|
|
res = intersection;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_aac_parse_src_event (GstBaseParse * parse, GstEvent * event)
|
|
{
|
|
GstAacParse *aacparse = GST_AAC_PARSE (parse);
|
|
|
|
if (GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
|
|
aacparse->last_parsed_channels = 0;
|
|
aacparse->last_parsed_sample_rate = 0;
|
|
}
|
|
|
|
return GST_BASE_PARSE_CLASS (parent_class)->src_event (parse, event);
|
|
}
|