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1452 lines
44 KiB
C
1452 lines
44 KiB
C
/* GStreamer MPEG audio parser
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* Copyright (C) 2006-2007 Jan Schmidt <thaytan@mad.scientist.com>
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* Copyright (C) 2010 Mark Nauwelaerts <mnauw users sf net>
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* Copyright (C) 2010 Nokia Corporation. All rights reserved.
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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-mpegaudioparse
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* @short_description: MPEG audio parser
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* @see_also: #GstAmrParse, #GstAACParse
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*
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* Parses and frames mpeg1 audio streams. Provides seeking.
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch-1.0 filesrc location=test.mp3 ! mpegaudioparse ! mad ! autoaudiosink
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* ]|
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* </refsect2>
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*/
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/* FIXME: we should make the base class (GstBaseParse) aware of the
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* XING seek table somehow, so it can use it properly for things like
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* accurate seeks. Currently it can only do a lookup via the convert function,
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* but then doesn't know what the result represents exactly. One could either
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* add a vfunc for index lookup, or just make mpegaudioparse populate the
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* base class's index via the API provided.
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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 "gstmpegaudioparse.h"
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#include <gst/base/gstbytereader.h>
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#include <gst/pbutils/pbutils.h>
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GST_DEBUG_CATEGORY_STATIC (mpeg_audio_parse_debug);
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#define GST_CAT_DEFAULT mpeg_audio_parse_debug
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#define MPEG_AUDIO_CHANNEL_MODE_UNKNOWN -1
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#define MPEG_AUDIO_CHANNEL_MODE_STEREO 0
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#define MPEG_AUDIO_CHANNEL_MODE_JOINT_STEREO 1
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#define MPEG_AUDIO_CHANNEL_MODE_DUAL_CHANNEL 2
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#define MPEG_AUDIO_CHANNEL_MODE_MONO 3
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#define CRC_UNKNOWN -1
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#define CRC_PROTECTED 0
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#define CRC_NOT_PROTECTED 1
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#define XING_FRAMES_FLAG 0x0001
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#define XING_BYTES_FLAG 0x0002
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#define XING_TOC_FLAG 0x0004
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#define XING_VBR_SCALE_FLAG 0x0008
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#define MIN_FRAME_SIZE 6
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/mpeg, "
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"mpegversion = (int) 1, "
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"layer = (int) [ 1, 3 ], "
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"mpegaudioversion = (int) [ 1, 3], "
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"rate = (int) [ 8000, 48000 ], "
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"channels = (int) [ 1, 2 ], " "parsed=(boolean) true")
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);
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/mpeg, mpegversion = (int) 1")
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);
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static void gst_mpeg_audio_parse_finalize (GObject * object);
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static gboolean gst_mpeg_audio_parse_start (GstBaseParse * parse);
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static gboolean gst_mpeg_audio_parse_stop (GstBaseParse * parse);
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static GstFlowReturn gst_mpeg_audio_parse_handle_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame, gint * skipsize);
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static GstFlowReturn gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame);
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static gboolean gst_mpeg_audio_parse_convert (GstBaseParse * parse,
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GstFormat src_format, gint64 src_value,
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GstFormat dest_format, gint64 * dest_value);
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static GstCaps *gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse,
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GstCaps * filter);
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static void gst_mpeg_audio_parse_handle_first_frame (GstMpegAudioParse *
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mp3parse, GstBuffer * buf);
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#define gst_mpeg_audio_parse_parent_class parent_class
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G_DEFINE_TYPE (GstMpegAudioParse, gst_mpeg_audio_parse, GST_TYPE_BASE_PARSE);
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#define GST_TYPE_MPEG_AUDIO_CHANNEL_MODE \
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(gst_mpeg_audio_channel_mode_get_type())
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static const GEnumValue mpeg_audio_channel_mode[] = {
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{MPEG_AUDIO_CHANNEL_MODE_UNKNOWN, "Unknown", "unknown"},
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{MPEG_AUDIO_CHANNEL_MODE_MONO, "Mono", "mono"},
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{MPEG_AUDIO_CHANNEL_MODE_DUAL_CHANNEL, "Dual Channel", "dual-channel"},
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{MPEG_AUDIO_CHANNEL_MODE_JOINT_STEREO, "Joint Stereo", "joint-stereo"},
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{MPEG_AUDIO_CHANNEL_MODE_STEREO, "Stereo", "stereo"},
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{0, NULL, NULL},
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};
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static GType
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gst_mpeg_audio_channel_mode_get_type (void)
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{
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static GType mpeg_audio_channel_mode_type = 0;
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if (!mpeg_audio_channel_mode_type) {
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mpeg_audio_channel_mode_type =
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g_enum_register_static ("GstMpegAudioChannelMode",
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mpeg_audio_channel_mode);
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}
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return mpeg_audio_channel_mode_type;
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}
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static const gchar *
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gst_mpeg_audio_channel_mode_get_nick (gint mode)
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{
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guint i;
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for (i = 0; i < G_N_ELEMENTS (mpeg_audio_channel_mode); i++) {
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if (mpeg_audio_channel_mode[i].value == mode)
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return mpeg_audio_channel_mode[i].value_nick;
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}
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return NULL;
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}
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static void
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gst_mpeg_audio_parse_class_init (GstMpegAudioParseClass * klass)
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{
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GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (mpeg_audio_parse_debug, "mpegaudioparse", 0,
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"MPEG1 audio stream parser");
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object_class->finalize = gst_mpeg_audio_parse_finalize;
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parse_class->start = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_start);
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parse_class->stop = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_stop);
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parse_class->handle_frame =
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GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_handle_frame);
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parse_class->pre_push_frame =
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GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_pre_push_frame);
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parse_class->convert = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_convert);
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parse_class->get_sink_caps =
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GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_get_sink_caps);
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/* register tags */
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#define GST_TAG_CRC "has-crc"
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#define GST_TAG_MODE "channel-mode"
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gst_tag_register (GST_TAG_CRC, GST_TAG_FLAG_META, G_TYPE_BOOLEAN,
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"has crc", "Using CRC", NULL);
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gst_tag_register (GST_TAG_MODE, GST_TAG_FLAG_ENCODED, G_TYPE_STRING,
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"channel mode", "MPEG audio channel mode", NULL);
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g_type_class_ref (GST_TYPE_MPEG_AUDIO_CHANNEL_MODE);
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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, "MPEG1 Audio Parser",
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"Codec/Parser/Audio",
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"Parses and frames mpeg1 audio streams (levels 1-3), provides seek",
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"Jan Schmidt <thaytan@mad.scientist.com>,"
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"Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
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}
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static void
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gst_mpeg_audio_parse_reset (GstMpegAudioParse * mp3parse)
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{
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mp3parse->channels = -1;
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mp3parse->rate = -1;
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mp3parse->sent_codec_tag = FALSE;
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mp3parse->last_posted_crc = CRC_UNKNOWN;
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mp3parse->last_posted_channel_mode = MPEG_AUDIO_CHANNEL_MODE_UNKNOWN;
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mp3parse->freerate = 0;
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mp3parse->hdr_bitrate = 0;
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mp3parse->xing_flags = 0;
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mp3parse->xing_bitrate = 0;
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mp3parse->xing_frames = 0;
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mp3parse->xing_total_time = 0;
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mp3parse->xing_bytes = 0;
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mp3parse->xing_vbr_scale = 0;
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memset (mp3parse->xing_seek_table, 0, 100);
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memset (mp3parse->xing_seek_table_inverse, 0, 256);
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mp3parse->vbri_bitrate = 0;
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mp3parse->vbri_frames = 0;
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mp3parse->vbri_total_time = 0;
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mp3parse->vbri_bytes = 0;
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mp3parse->vbri_seek_points = 0;
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g_free (mp3parse->vbri_seek_table);
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mp3parse->vbri_seek_table = NULL;
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mp3parse->encoder_delay = 0;
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mp3parse->encoder_padding = 0;
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}
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static void
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gst_mpeg_audio_parse_init (GstMpegAudioParse * mp3parse)
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{
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gst_mpeg_audio_parse_reset (mp3parse);
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GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (mp3parse));
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (mp3parse));
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}
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static void
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gst_mpeg_audio_parse_finalize (GObject * object)
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{
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_mpeg_audio_parse_start (GstBaseParse * parse)
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{
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GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
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gst_base_parse_set_min_frame_size (GST_BASE_PARSE (mp3parse), MIN_FRAME_SIZE);
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GST_DEBUG_OBJECT (parse, "starting");
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gst_mpeg_audio_parse_reset (mp3parse);
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return TRUE;
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}
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static gboolean
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gst_mpeg_audio_parse_stop (GstBaseParse * parse)
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{
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GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
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GST_DEBUG_OBJECT (parse, "stopping");
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gst_mpeg_audio_parse_reset (mp3parse);
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return TRUE;
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}
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static const guint mp3types_bitrates[2][3][16] = {
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{
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{0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,},
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{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,},
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{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}
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},
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{
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{0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,},
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{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,},
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{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}
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},
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};
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static const guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
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{22050, 24000, 16000},
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{11025, 12000, 8000}
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};
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static inline guint
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mp3_type_frame_length_from_header (GstMpegAudioParse * mp3parse, guint32 header,
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guint * put_version, guint * put_layer, guint * put_channels,
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guint * put_bitrate, guint * put_samplerate, guint * put_mode,
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guint * put_crc)
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{
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guint length;
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gulong mode, samplerate, bitrate, layer, channels, padding, crc;
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gulong version;
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gint lsf, mpg25;
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if (header & (1 << 20)) {
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lsf = (header & (1 << 19)) ? 0 : 1;
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mpg25 = 0;
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} else {
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lsf = 1;
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mpg25 = 1;
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}
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version = 1 + lsf + mpg25;
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layer = 4 - ((header >> 17) & 0x3);
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crc = (header >> 16) & 0x1;
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bitrate = (header >> 12) & 0xF;
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bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
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if (!bitrate) {
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GST_LOG_OBJECT (mp3parse, "using freeform bitrate");
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bitrate = mp3parse->freerate;
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}
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samplerate = (header >> 10) & 0x3;
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samplerate = mp3types_freqs[lsf + mpg25][samplerate];
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/* force 0 length if 0 bitrate */
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padding = (bitrate > 0) ? (header >> 9) & 0x1 : 0;
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mode = (header >> 6) & 0x3;
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channels = (mode == 3) ? 1 : 2;
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switch (layer) {
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case 1:
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length = 4 * ((bitrate * 12) / samplerate + padding);
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break;
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case 2:
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length = (bitrate * 144) / samplerate + padding;
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break;
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default:
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case 3:
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length = (bitrate * 144) / (samplerate << lsf) + padding;
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break;
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}
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GST_DEBUG_OBJECT (mp3parse, "Calculated mp3 frame length of %u bytes",
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length);
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GST_DEBUG_OBJECT (mp3parse, "samplerate = %lu, bitrate = %lu, version = %lu, "
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"layer = %lu, channels = %lu, mode = %s", samplerate, bitrate, version,
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layer, channels, gst_mpeg_audio_channel_mode_get_nick (mode));
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if (put_version)
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*put_version = version;
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if (put_layer)
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*put_layer = layer;
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if (put_channels)
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*put_channels = channels;
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if (put_bitrate)
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*put_bitrate = bitrate;
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if (put_samplerate)
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*put_samplerate = samplerate;
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if (put_mode)
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*put_mode = mode;
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if (put_crc)
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*put_crc = crc;
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return length;
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}
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/* Minimum number of consecutive, valid-looking frames to consider
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* for resyncing */
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#define MIN_RESYNC_FRAMES 3
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/* Perform extended validation to check that subsequent headers match
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* the first header given here in important characteristics, to avoid
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* false sync. We look for a minimum of MIN_RESYNC_FRAMES consecutive
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* frames to match their major characteristics.
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*
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* If at_eos is set to TRUE, we just check that we don't find any invalid
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* frames in whatever data is available, rather than requiring a full
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* MIN_RESYNC_FRAMES of data.
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*
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* Returns TRUE if we've seen enough data to validate or reject the frame.
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* If TRUE is returned, then *valid contains TRUE if it validated, or false
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* if we decided it was false sync.
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* If FALSE is returned, then *valid contains minimum needed data.
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*/
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static gboolean
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gst_mp3parse_validate_extended (GstMpegAudioParse * mp3parse, GstBuffer * buf,
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guint32 header, int bpf, gboolean at_eos, gint * valid)
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{
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guint32 next_header;
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GstMapInfo map;
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gboolean res = TRUE;
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int frames_found = 1;
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int offset = bpf;
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gst_buffer_map (buf, &map, GST_MAP_READ);
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while (frames_found < MIN_RESYNC_FRAMES) {
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/* Check if we have enough data for all these frames, plus the next
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frame header. */
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if (map.size < offset + 4) {
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if (at_eos) {
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/* Running out of data at EOS is fine; just accept it */
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*valid = TRUE;
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goto cleanup;
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} else {
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*valid = offset + 4;
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res = FALSE;
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goto cleanup;
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}
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}
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next_header = GST_READ_UINT32_BE (map.data + offset);
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GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X, bpf=%d",
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offset, (unsigned int) header, (unsigned int) next_header, bpf);
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/* mask the bits which are allowed to differ between frames */
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#define HDRMASK ~((0xF << 12) /* bitrate */ | \
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(0x1 << 9) /* padding */ | \
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(0xf << 4) /* mode|mode extension */ | \
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(0xf)) /* copyright|emphasis */
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|
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if ((next_header & HDRMASK) != (header & HDRMASK)) {
|
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/* If any of the unmasked bits don't match, then it's not valid */
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GST_DEBUG_OBJECT (mp3parse, "next header doesn't match "
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"(header=%08X (%08X), header2=%08X (%08X), bpf=%d)",
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(guint) header, (guint) header & HDRMASK, (guint) next_header,
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(guint) next_header & HDRMASK, bpf);
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*valid = FALSE;
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goto cleanup;
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} else if (((next_header >> 12) & 0xf) == 0xf) {
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/* The essential parts were the same, but the bitrate held an
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invalid value - also reject */
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GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)");
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*valid = FALSE;
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goto cleanup;
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}
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|
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bpf = mp3_type_frame_length_from_header (mp3parse, next_header,
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NULL, NULL, NULL, NULL, NULL, NULL, NULL);
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|
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/* if no bitrate, and no freeform rate known, then fail */
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if (G_UNLIKELY (!bpf)) {
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GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate 0)");
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*valid = FALSE;
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goto cleanup;
|
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}
|
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|
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offset += bpf;
|
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frames_found++;
|
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}
|
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|
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*valid = TRUE;
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|
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cleanup:
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gst_buffer_unmap (buf, &map);
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return res;
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}
|
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|
|
static gboolean
|
|
gst_mpeg_audio_parse_head_check (GstMpegAudioParse * mp3parse,
|
|
unsigned long head)
|
|
{
|
|
GST_DEBUG_OBJECT (mp3parse, "checking mp3 header 0x%08lx", head);
|
|
/* if it's not a valid sync */
|
|
if ((head & 0xffe00000) != 0xffe00000) {
|
|
GST_WARNING_OBJECT (mp3parse, "invalid sync");
|
|
return FALSE;
|
|
}
|
|
/* if it's an invalid MPEG version */
|
|
if (((head >> 19) & 3) == 0x1) {
|
|
GST_WARNING_OBJECT (mp3parse, "invalid MPEG version: 0x%lx",
|
|
(head >> 19) & 3);
|
|
return FALSE;
|
|
}
|
|
/* if it's an invalid layer */
|
|
if (!((head >> 17) & 3)) {
|
|
GST_WARNING_OBJECT (mp3parse, "invalid layer: 0x%lx", (head >> 17) & 3);
|
|
return FALSE;
|
|
}
|
|
/* if it's an invalid bitrate */
|
|
if (((head >> 12) & 0xf) == 0xf) {
|
|
GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx", (head >> 12) & 0xf);
|
|
return FALSE;
|
|
}
|
|
/* if it's an invalid samplerate */
|
|
if (((head >> 10) & 0x3) == 0x3) {
|
|
GST_WARNING_OBJECT (mp3parse, "invalid samplerate: 0x%lx",
|
|
(head >> 10) & 0x3);
|
|
return FALSE;
|
|
}
|
|
|
|
if ((head & 0x3) == 0x2) {
|
|
/* Ignore this as there are some files with emphasis 0x2 that can
|
|
* be played fine. See BGO #537235 */
|
|
GST_WARNING_OBJECT (mp3parse, "invalid emphasis: 0x%lx", head & 0x3);
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* Determines possible freeform frame rate/size by looking for next
|
|
* header with valid bitrate (0 or otherwise valid) (and sufficiently
|
|
* matching current header).
|
|
*
|
|
* Returns TRUE if we've found such one, and *rate then contains rate
|
|
* (or *rate contains 0 if decided no freeframe size could be determined).
|
|
* If not enough data, returns FALSE.
|
|
*/
|
|
static gboolean
|
|
gst_mp3parse_find_freerate (GstMpegAudioParse * mp3parse, GstMapInfo * map,
|
|
guint32 header, gboolean at_eos, gint * _rate)
|
|
{
|
|
guint32 next_header;
|
|
const guint8 *data;
|
|
guint available;
|
|
int offset = 4;
|
|
gulong samplerate, rate, layer, padding;
|
|
gboolean valid;
|
|
gint lsf, mpg25;
|
|
|
|
available = map->size;
|
|
data = map->data;
|
|
|
|
*_rate = 0;
|
|
|
|
/* pick apart header again partially */
|
|
if (header & (1 << 20)) {
|
|
lsf = (header & (1 << 19)) ? 0 : 1;
|
|
mpg25 = 0;
|
|
} else {
|
|
lsf = 1;
|
|
mpg25 = 1;
|
|
}
|
|
layer = 4 - ((header >> 17) & 0x3);
|
|
samplerate = (header >> 10) & 0x3;
|
|
samplerate = mp3types_freqs[lsf + mpg25][samplerate];
|
|
padding = (header >> 9) & 0x1;
|
|
|
|
for (; offset < available; ++offset) {
|
|
/* Check if we have enough data for all these frames, plus the next
|
|
frame header. */
|
|
if (available < offset + 4) {
|
|
if (at_eos) {
|
|
/* Running out of data; failed to determine size */
|
|
return TRUE;
|
|
} else {
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
valid = FALSE;
|
|
next_header = GST_READ_UINT32_BE (data + offset);
|
|
if ((next_header & 0xFFE00000) != 0xFFE00000)
|
|
goto next;
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X",
|
|
offset, (unsigned int) header, (unsigned int) next_header);
|
|
|
|
if ((next_header & HDRMASK) != (header & HDRMASK)) {
|
|
/* If any of the unmasked bits don't match, then it's not valid */
|
|
GST_DEBUG_OBJECT (mp3parse, "next header doesn't match "
|
|
"(header=%08X (%08X), header2=%08X (%08X))",
|
|
(guint) header, (guint) header & HDRMASK, (guint) next_header,
|
|
(guint) next_header & HDRMASK);
|
|
goto next;
|
|
} else if (((next_header >> 12) & 0xf) == 0xf) {
|
|
/* The essential parts were the same, but the bitrate held an
|
|
invalid value - also reject */
|
|
GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)");
|
|
goto next;
|
|
}
|
|
|
|
valid = TRUE;
|
|
|
|
next:
|
|
/* almost accept as free frame */
|
|
if (layer == 1) {
|
|
rate = samplerate * (offset - 4 * padding + 4) / 48000;
|
|
} else {
|
|
rate = samplerate * (offset - padding + 1) / (144 >> lsf) / 1000;
|
|
}
|
|
|
|
if (valid) {
|
|
GST_LOG_OBJECT (mp3parse, "calculated rate %lu", rate * 1000);
|
|
if (rate < 8 || (layer == 3 && rate > 640)) {
|
|
GST_DEBUG_OBJECT (mp3parse, "rate invalid");
|
|
if (rate < 8) {
|
|
/* maybe some hope */
|
|
continue;
|
|
} else {
|
|
GST_DEBUG_OBJECT (mp3parse, "aborting");
|
|
/* give up */
|
|
break;
|
|
}
|
|
}
|
|
*_rate = rate * 1000;
|
|
break;
|
|
} else {
|
|
/* avoid indefinite searching */
|
|
if (rate > 1000) {
|
|
GST_DEBUG_OBJECT (mp3parse, "exceeded sanity rate; aborting");
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_mpeg_audio_parse_handle_frame (GstBaseParse * parse,
|
|
GstBaseParseFrame * frame, gint * skipsize)
|
|
{
|
|
GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
|
|
GstBuffer *buf = frame->buffer;
|
|
GstByteReader reader;
|
|
gint off, bpf = 0;
|
|
gboolean lost_sync, draining, valid, caps_change;
|
|
guint32 header;
|
|
guint bitrate, layer, rate, channels, version, mode, crc;
|
|
GstMapInfo map;
|
|
gboolean res = FALSE;
|
|
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
if (G_UNLIKELY (map.size < 6)) {
|
|
*skipsize = 1;
|
|
goto cleanup;
|
|
}
|
|
|
|
gst_byte_reader_init (&reader, map.data, map.size);
|
|
|
|
off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffe00000, 0xffe00000,
|
|
0, map.size);
|
|
|
|
GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off);
|
|
|
|
/* didn't find anything that looks like a sync word, skip */
|
|
if (off < 0) {
|
|
*skipsize = map.size - 3;
|
|
goto cleanup;
|
|
}
|
|
|
|
/* possible frame header, but not at offset 0? skip bytes before sync */
|
|
if (off > 0) {
|
|
*skipsize = off;
|
|
goto cleanup;
|
|
}
|
|
|
|
/* make sure the values in the frame header look sane */
|
|
header = GST_READ_UINT32_BE (map.data);
|
|
if (!gst_mpeg_audio_parse_head_check (mp3parse, header)) {
|
|
*skipsize = 1;
|
|
goto cleanup;
|
|
}
|
|
|
|
GST_LOG_OBJECT (parse, "got frame");
|
|
|
|
lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
|
|
draining = GST_BASE_PARSE_DRAINING (parse);
|
|
|
|
if (G_UNLIKELY (lost_sync))
|
|
mp3parse->freerate = 0;
|
|
|
|
bpf = mp3_type_frame_length_from_header (mp3parse, header,
|
|
&version, &layer, &channels, &bitrate, &rate, &mode, &crc);
|
|
|
|
if (channels != mp3parse->channels || rate != mp3parse->rate ||
|
|
layer != mp3parse->layer || version != mp3parse->version)
|
|
caps_change = TRUE;
|
|
else
|
|
caps_change = FALSE;
|
|
|
|
/* maybe free format */
|
|
if (bpf == 0) {
|
|
GST_LOG_OBJECT (mp3parse, "possibly free format");
|
|
if (lost_sync || mp3parse->freerate == 0) {
|
|
GST_DEBUG_OBJECT (mp3parse, "finding free format rate");
|
|
if (!gst_mp3parse_find_freerate (mp3parse, &map, header, draining,
|
|
&valid)) {
|
|
/* not enough data */
|
|
gst_base_parse_set_min_frame_size (parse, valid);
|
|
*skipsize = 0;
|
|
goto cleanup;
|
|
} else {
|
|
GST_DEBUG_OBJECT (parse, "determined freeform size %d", valid);
|
|
mp3parse->freerate = valid;
|
|
}
|
|
}
|
|
/* try again */
|
|
bpf = mp3_type_frame_length_from_header (mp3parse, header,
|
|
&version, &layer, &channels, &bitrate, &rate, &mode, &crc);
|
|
if (!bpf) {
|
|
/* did not come up with valid freeform length, reject after all */
|
|
*skipsize = 1;
|
|
goto cleanup;
|
|
}
|
|
}
|
|
|
|
if (!draining && (lost_sync || caps_change)) {
|
|
if (!gst_mp3parse_validate_extended (mp3parse, buf, header, bpf, draining,
|
|
&valid)) {
|
|
/* not enough data */
|
|
gst_base_parse_set_min_frame_size (parse, valid);
|
|
*skipsize = 0;
|
|
goto cleanup;
|
|
} else {
|
|
if (!valid) {
|
|
*skipsize = off + 2;
|
|
goto cleanup;
|
|
}
|
|
}
|
|
} else if (draining && lost_sync && caps_change && mp3parse->rate > 0) {
|
|
/* avoid caps jitter that we can't be sure of */
|
|
*skipsize = off + 2;
|
|
goto cleanup;
|
|
}
|
|
|
|
/* restore default minimum */
|
|
gst_base_parse_set_min_frame_size (parse, MIN_FRAME_SIZE);
|
|
|
|
res = TRUE;
|
|
|
|
/* metadata handling */
|
|
if (G_UNLIKELY (caps_change)) {
|
|
GstCaps *caps = gst_caps_new_simple ("audio/mpeg",
|
|
"mpegversion", G_TYPE_INT, 1,
|
|
"mpegaudioversion", G_TYPE_INT, version,
|
|
"layer", G_TYPE_INT, layer,
|
|
"rate", G_TYPE_INT, rate,
|
|
"channels", G_TYPE_INT, channels, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
|
|
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
|
|
gst_caps_unref (caps);
|
|
|
|
mp3parse->rate = rate;
|
|
mp3parse->channels = channels;
|
|
mp3parse->layer = layer;
|
|
mp3parse->version = version;
|
|
|
|
/* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */
|
|
if (mp3parse->layer == 1)
|
|
mp3parse->spf = 384;
|
|
else if (mp3parse->layer == 2)
|
|
mp3parse->spf = 1152;
|
|
else if (mp3parse->version == 1) {
|
|
mp3parse->spf = 1152;
|
|
} else {
|
|
/* MPEG-2 or "2.5" */
|
|
mp3parse->spf = 576;
|
|
}
|
|
|
|
/* lead_in:
|
|
* We start pushing 9 frames earlier (29 frames for MPEG2) than
|
|
* segment start to be able to decode the first frame we want.
|
|
* 9 (29) frames are the theoretical maximum of frames that contain
|
|
* data for the current frame (bit reservoir).
|
|
*
|
|
* lead_out:
|
|
* Some mp3 streams have an offset in the timestamps, for which we have to
|
|
* push the frame *after* the end position in order for the decoder to be
|
|
* able to decode everything up until the segment.stop position. */
|
|
gst_base_parse_set_frame_rate (parse, mp3parse->rate, mp3parse->spf,
|
|
(version == 1) ? 10 : 30, 2);
|
|
}
|
|
|
|
mp3parse->hdr_bitrate = bitrate;
|
|
|
|
/* For first frame; check for seek tables and output a codec tag */
|
|
gst_mpeg_audio_parse_handle_first_frame (mp3parse, buf);
|
|
|
|
/* store some frame info for later processing */
|
|
mp3parse->last_crc = crc;
|
|
mp3parse->last_mode = mode;
|
|
|
|
cleanup:
|
|
gst_buffer_unmap (buf, &map);
|
|
|
|
if (res && bpf <= map.size) {
|
|
return gst_base_parse_finish_frame (parse, frame, bpf);
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static void
|
|
gst_mpeg_audio_parse_handle_first_frame (GstMpegAudioParse * mp3parse,
|
|
GstBuffer * buf)
|
|
{
|
|
const guint32 xing_id = 0x58696e67; /* 'Xing' in hex */
|
|
const guint32 info_id = 0x496e666f; /* 'Info' in hex - found in LAME CBR files */
|
|
const guint32 vbri_id = 0x56425249; /* 'VBRI' in hex */
|
|
const guint32 lame_id = 0x4c414d45; /* 'LAME' in hex */
|
|
gint offset_xing, offset_vbri;
|
|
guint64 avail;
|
|
gint64 upstream_total_bytes = 0;
|
|
guint32 read_id_xing = 0, read_id_vbri = 0;
|
|
GstMapInfo map;
|
|
guint8 *data;
|
|
guint bitrate;
|
|
|
|
if (mp3parse->sent_codec_tag)
|
|
return;
|
|
|
|
/* Check first frame for Xing info */
|
|
if (mp3parse->version == 1) { /* MPEG-1 file */
|
|
if (mp3parse->channels == 1)
|
|
offset_xing = 0x11;
|
|
else
|
|
offset_xing = 0x20;
|
|
} else { /* MPEG-2 header */
|
|
if (mp3parse->channels == 1)
|
|
offset_xing = 0x09;
|
|
else
|
|
offset_xing = 0x11;
|
|
}
|
|
|
|
/* The VBRI tag is always at offset 0x20 */
|
|
offset_vbri = 0x20;
|
|
|
|
/* Skip the 4 bytes of the MP3 header too */
|
|
offset_xing += 4;
|
|
offset_vbri += 4;
|
|
|
|
/* Check if we have enough data to read the Xing header */
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
data = map.data;
|
|
avail = map.size;
|
|
|
|
if (avail >= offset_xing + 4) {
|
|
read_id_xing = GST_READ_UINT32_BE (data + offset_xing);
|
|
}
|
|
if (avail >= offset_vbri + 4) {
|
|
read_id_vbri = GST_READ_UINT32_BE (data + offset_vbri);
|
|
}
|
|
|
|
/* obtain real upstream total bytes */
|
|
if (!gst_pad_peer_query_duration (GST_BASE_PARSE_SINK_PAD (mp3parse),
|
|
GST_FORMAT_BYTES, &upstream_total_bytes))
|
|
upstream_total_bytes = 0;
|
|
|
|
if (read_id_xing == xing_id || read_id_xing == info_id) {
|
|
guint32 xing_flags;
|
|
guint bytes_needed = offset_xing + 8;
|
|
gint64 total_bytes;
|
|
GstClockTime total_time;
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id);
|
|
|
|
/* Move data after Xing header */
|
|
data += offset_xing + 4;
|
|
|
|
/* Read 4 base bytes of flags, big-endian */
|
|
xing_flags = GST_READ_UINT32_BE (data);
|
|
data += 4;
|
|
if (xing_flags & XING_FRAMES_FLAG)
|
|
bytes_needed += 4;
|
|
if (xing_flags & XING_BYTES_FLAG)
|
|
bytes_needed += 4;
|
|
if (xing_flags & XING_TOC_FLAG)
|
|
bytes_needed += 100;
|
|
if (xing_flags & XING_VBR_SCALE_FLAG)
|
|
bytes_needed += 4;
|
|
if (avail < bytes_needed) {
|
|
GST_DEBUG_OBJECT (mp3parse,
|
|
"Not enough data to read Xing header (need %d)", bytes_needed);
|
|
goto cleanup;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "Reading Xing header");
|
|
mp3parse->xing_flags = xing_flags;
|
|
|
|
if (xing_flags & XING_FRAMES_FLAG) {
|
|
mp3parse->xing_frames = GST_READ_UINT32_BE (data);
|
|
if (mp3parse->xing_frames == 0) {
|
|
GST_WARNING_OBJECT (mp3parse,
|
|
"Invalid number of frames in Xing header");
|
|
mp3parse->xing_flags &= ~XING_FRAMES_FLAG;
|
|
} else {
|
|
mp3parse->xing_total_time = gst_util_uint64_scale (GST_SECOND,
|
|
(guint64) (mp3parse->xing_frames) * (mp3parse->spf),
|
|
mp3parse->rate);
|
|
}
|
|
|
|
data += 4;
|
|
} else {
|
|
mp3parse->xing_frames = 0;
|
|
mp3parse->xing_total_time = 0;
|
|
}
|
|
|
|
if (xing_flags & XING_BYTES_FLAG) {
|
|
mp3parse->xing_bytes = GST_READ_UINT32_BE (data);
|
|
if (mp3parse->xing_bytes == 0) {
|
|
GST_WARNING_OBJECT (mp3parse, "Invalid number of bytes in Xing header");
|
|
mp3parse->xing_flags &= ~XING_BYTES_FLAG;
|
|
}
|
|
data += 4;
|
|
} else {
|
|
mp3parse->xing_bytes = 0;
|
|
}
|
|
|
|
/* If we know the upstream size and duration, compute the
|
|
* total bitrate, rounded up to the nearest kbit/sec */
|
|
if ((total_time = mp3parse->xing_total_time) &&
|
|
(total_bytes = mp3parse->xing_bytes)) {
|
|
mp3parse->xing_bitrate = gst_util_uint64_scale (total_bytes,
|
|
8 * GST_SECOND, total_time);
|
|
mp3parse->xing_bitrate += 500;
|
|
mp3parse->xing_bitrate -= mp3parse->xing_bitrate % 1000;
|
|
}
|
|
|
|
if (xing_flags & XING_TOC_FLAG) {
|
|
int i, percent = 0;
|
|
guchar *table = mp3parse->xing_seek_table;
|
|
guchar old = 0, new;
|
|
guint first;
|
|
|
|
first = data[0];
|
|
GST_DEBUG_OBJECT (mp3parse,
|
|
"Subtracting initial offset of %d bytes from Xing TOC", first);
|
|
|
|
/* xing seek table: percent time -> 1/256 bytepos */
|
|
for (i = 0; i < 100; i++) {
|
|
new = data[i] - first;
|
|
if (old > new) {
|
|
GST_WARNING_OBJECT (mp3parse, "Skipping broken Xing TOC");
|
|
mp3parse->xing_flags &= ~XING_TOC_FLAG;
|
|
goto skip_toc;
|
|
}
|
|
mp3parse->xing_seek_table[i] = old = new;
|
|
}
|
|
|
|
/* build inverse table: 1/256 bytepos -> 1/100 percent time */
|
|
for (i = 0; i < 256; i++) {
|
|
while (percent < 99 && table[percent + 1] <= i)
|
|
percent++;
|
|
|
|
if (table[percent] == i) {
|
|
mp3parse->xing_seek_table_inverse[i] = percent * 100;
|
|
} else if (percent < 99 && table[percent]) {
|
|
gdouble fa, fb, fx;
|
|
gint a = percent, b = percent + 1;
|
|
|
|
fa = table[a];
|
|
fb = table[b];
|
|
fx = (b - a) / (fb - fa) * (i - fa) + a;
|
|
mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100);
|
|
} else if (percent == 99) {
|
|
gdouble fa, fb, fx;
|
|
gint a = percent, b = 100;
|
|
|
|
fa = table[a];
|
|
fb = 256.0;
|
|
fx = (b - a) / (fb - fa) * (i - fa) + a;
|
|
mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100);
|
|
}
|
|
}
|
|
skip_toc:
|
|
data += 100;
|
|
} else {
|
|
memset (mp3parse->xing_seek_table, 0, 100);
|
|
memset (mp3parse->xing_seek_table_inverse, 0, 256);
|
|
}
|
|
|
|
if (xing_flags & XING_VBR_SCALE_FLAG) {
|
|
mp3parse->xing_vbr_scale = GST_READ_UINT32_BE (data);
|
|
data += 4;
|
|
} else
|
|
mp3parse->xing_vbr_scale = 0;
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "Xing header reported %u frames, time %"
|
|
GST_TIME_FORMAT ", %u bytes, vbr scale %u", mp3parse->xing_frames,
|
|
GST_TIME_ARGS (mp3parse->xing_total_time), mp3parse->xing_bytes,
|
|
mp3parse->xing_vbr_scale);
|
|
|
|
/* check for truncated file */
|
|
if (upstream_total_bytes && mp3parse->xing_bytes &&
|
|
mp3parse->xing_bytes * 0.8 > upstream_total_bytes) {
|
|
GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; "
|
|
"invalidating Xing header duration and size");
|
|
mp3parse->xing_flags &= ~XING_BYTES_FLAG;
|
|
mp3parse->xing_flags &= ~XING_FRAMES_FLAG;
|
|
}
|
|
|
|
/* Optional LAME tag? */
|
|
if (avail - bytes_needed >= 36 && GST_READ_UINT32_BE (data) == lame_id) {
|
|
gchar lame_version[10] = { 0, };
|
|
guint tag_rev;
|
|
guint32 encoder_delay, encoder_padding;
|
|
|
|
memcpy (lame_version, data, 9);
|
|
data += 9;
|
|
tag_rev = data[0] >> 4;
|
|
GST_DEBUG_OBJECT (mp3parse, "Found LAME tag revision %d created by '%s'",
|
|
tag_rev, lame_version);
|
|
|
|
/* Skip all the information we're not interested in */
|
|
data += 12;
|
|
/* Encoder delay and end padding */
|
|
encoder_delay = GST_READ_UINT24_BE (data);
|
|
encoder_delay >>= 12;
|
|
encoder_padding = GST_READ_UINT24_BE (data);
|
|
encoder_padding &= 0x000fff;
|
|
|
|
mp3parse->encoder_delay = encoder_delay;
|
|
mp3parse->encoder_padding = encoder_padding;
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "Encoder delay %u, encoder padding %u",
|
|
encoder_delay, encoder_padding);
|
|
}
|
|
} else if (read_id_vbri == vbri_id) {
|
|
gint64 total_bytes, total_frames;
|
|
GstClockTime total_time;
|
|
guint16 nseek_points;
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "Found VBRI header marker 0x%x", vbri_id);
|
|
|
|
if (avail < offset_vbri + 26) {
|
|
GST_DEBUG_OBJECT (mp3parse,
|
|
"Not enough data to read VBRI header (need %d)", offset_vbri + 26);
|
|
goto cleanup;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "Reading VBRI header");
|
|
|
|
/* Move data after VBRI header */
|
|
data += offset_vbri + 4;
|
|
|
|
if (GST_READ_UINT16_BE (data) != 0x0001) {
|
|
GST_WARNING_OBJECT (mp3parse,
|
|
"Unsupported VBRI version 0x%x", GST_READ_UINT16_BE (data));
|
|
goto cleanup;
|
|
}
|
|
data += 2;
|
|
|
|
/* Skip encoder delay */
|
|
data += 2;
|
|
|
|
/* Skip quality */
|
|
data += 2;
|
|
|
|
total_bytes = GST_READ_UINT32_BE (data);
|
|
if (total_bytes != 0)
|
|
mp3parse->vbri_bytes = total_bytes;
|
|
data += 4;
|
|
|
|
total_frames = GST_READ_UINT32_BE (data);
|
|
if (total_frames != 0) {
|
|
mp3parse->vbri_frames = total_frames;
|
|
mp3parse->vbri_total_time = gst_util_uint64_scale (GST_SECOND,
|
|
(guint64) (mp3parse->vbri_frames) * (mp3parse->spf), mp3parse->rate);
|
|
}
|
|
data += 4;
|
|
|
|
/* If we know the upstream size and duration, compute the
|
|
* total bitrate, rounded up to the nearest kbit/sec */
|
|
if ((total_time = mp3parse->vbri_total_time) &&
|
|
(total_bytes = mp3parse->vbri_bytes)) {
|
|
mp3parse->vbri_bitrate = gst_util_uint64_scale (total_bytes,
|
|
8 * GST_SECOND, total_time);
|
|
mp3parse->vbri_bitrate += 500;
|
|
mp3parse->vbri_bitrate -= mp3parse->vbri_bitrate % 1000;
|
|
}
|
|
|
|
nseek_points = GST_READ_UINT16_BE (data);
|
|
data += 2;
|
|
|
|
if (nseek_points > 0) {
|
|
guint scale, seek_bytes, seek_frames;
|
|
gint i;
|
|
|
|
mp3parse->vbri_seek_points = nseek_points;
|
|
|
|
scale = GST_READ_UINT16_BE (data);
|
|
data += 2;
|
|
|
|
seek_bytes = GST_READ_UINT16_BE (data);
|
|
data += 2;
|
|
|
|
seek_frames = GST_READ_UINT16_BE (data);
|
|
|
|
if (scale == 0 || seek_bytes == 0 || seek_bytes > 4 || seek_frames == 0) {
|
|
GST_WARNING_OBJECT (mp3parse, "Unsupported VBRI seek table");
|
|
goto out_vbri;
|
|
}
|
|
|
|
if (avail < offset_vbri + 26 + nseek_points * seek_bytes) {
|
|
GST_WARNING_OBJECT (mp3parse,
|
|
"Not enough data to read VBRI seek table (need %d)",
|
|
offset_vbri + 26 + nseek_points * seek_bytes);
|
|
goto out_vbri;
|
|
}
|
|
|
|
if (seek_frames * nseek_points < total_frames - seek_frames ||
|
|
seek_frames * nseek_points > total_frames + seek_frames) {
|
|
GST_WARNING_OBJECT (mp3parse,
|
|
"VBRI seek table doesn't cover the complete file");
|
|
goto out_vbri;
|
|
}
|
|
|
|
data = map.data;
|
|
data += offset_vbri + 26;
|
|
|
|
/* VBRI seek table: frame/seek_frames -> byte */
|
|
mp3parse->vbri_seek_table = g_new (guint32, nseek_points);
|
|
if (seek_bytes == 4)
|
|
for (i = 0; i < nseek_points; i++) {
|
|
mp3parse->vbri_seek_table[i] = GST_READ_UINT32_BE (data) * scale;
|
|
data += 4;
|
|
} else if (seek_bytes == 3)
|
|
for (i = 0; i < nseek_points; i++) {
|
|
mp3parse->vbri_seek_table[i] = GST_READ_UINT24_BE (data) * scale;
|
|
data += 3;
|
|
} else if (seek_bytes == 2)
|
|
for (i = 0; i < nseek_points; i++) {
|
|
mp3parse->vbri_seek_table[i] = GST_READ_UINT16_BE (data) * scale;
|
|
data += 2;
|
|
} else /* seek_bytes == 1 */
|
|
for (i = 0; i < nseek_points; i++) {
|
|
mp3parse->vbri_seek_table[i] = GST_READ_UINT8 (data) * scale;
|
|
data += 1;
|
|
}
|
|
}
|
|
out_vbri:
|
|
|
|
GST_DEBUG_OBJECT (mp3parse, "VBRI header reported %u frames, time %"
|
|
GST_TIME_FORMAT ", bytes %u", mp3parse->vbri_frames,
|
|
GST_TIME_ARGS (mp3parse->vbri_total_time), mp3parse->vbri_bytes);
|
|
|
|
/* check for truncated file */
|
|
if (upstream_total_bytes && mp3parse->vbri_bytes &&
|
|
mp3parse->vbri_bytes * 0.8 > upstream_total_bytes) {
|
|
GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; "
|
|
"invalidating VBRI header duration and size");
|
|
mp3parse->vbri_valid = FALSE;
|
|
} else {
|
|
mp3parse->vbri_valid = TRUE;
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (mp3parse,
|
|
"Xing, LAME or VBRI header not found in first frame");
|
|
}
|
|
|
|
/* set duration if tables provided a valid one */
|
|
if (mp3parse->xing_flags & XING_FRAMES_FLAG) {
|
|
gst_base_parse_set_duration (GST_BASE_PARSE (mp3parse), GST_FORMAT_TIME,
|
|
mp3parse->xing_total_time, 0);
|
|
}
|
|
if (mp3parse->vbri_total_time != 0 && mp3parse->vbri_valid) {
|
|
gst_base_parse_set_duration (GST_BASE_PARSE (mp3parse), GST_FORMAT_TIME,
|
|
mp3parse->vbri_total_time, 0);
|
|
}
|
|
|
|
/* tell baseclass how nicely we can seek, and a bitrate if one found */
|
|
/* FIXME: fill index with seek table */
|
|
#if 0
|
|
seekable = GST_BASE_PARSE_SEEK_DEFAULT;
|
|
if ((mp3parse->xing_flags & XING_TOC_FLAG) && mp3parse->xing_bytes &&
|
|
mp3parse->xing_total_time)
|
|
seekable = GST_BASE_PARSE_SEEK_TABLE;
|
|
|
|
if (mp3parse->vbri_seek_table && mp3parse->vbri_bytes &&
|
|
mp3parse->vbri_total_time)
|
|
seekable = GST_BASE_PARSE_SEEK_TABLE;
|
|
#endif
|
|
|
|
if (mp3parse->xing_bitrate)
|
|
bitrate = mp3parse->xing_bitrate;
|
|
else if (mp3parse->vbri_bitrate)
|
|
bitrate = mp3parse->vbri_bitrate;
|
|
else
|
|
bitrate = 0;
|
|
|
|
gst_base_parse_set_average_bitrate (GST_BASE_PARSE (mp3parse), bitrate);
|
|
|
|
cleanup:
|
|
gst_buffer_unmap (buf, &map);
|
|
}
|
|
|
|
static gboolean
|
|
gst_mpeg_audio_parse_time_to_bytepos (GstMpegAudioParse * mp3parse,
|
|
GstClockTime ts, gint64 * bytepos)
|
|
{
|
|
gint64 total_bytes;
|
|
GstClockTime total_time;
|
|
|
|
/* If XING seek table exists use this for time->byte conversion */
|
|
if ((mp3parse->xing_flags & XING_TOC_FLAG) &&
|
|
(total_bytes = mp3parse->xing_bytes) &&
|
|
(total_time = mp3parse->xing_total_time)) {
|
|
gdouble fa, fb, fx;
|
|
gdouble percent =
|
|
CLAMP ((100.0 * gst_util_guint64_to_gdouble (ts)) /
|
|
gst_util_guint64_to_gdouble (total_time), 0.0, 100.0);
|
|
gint index = CLAMP (percent, 0, 99);
|
|
|
|
fa = mp3parse->xing_seek_table[index];
|
|
if (index < 99)
|
|
fb = mp3parse->xing_seek_table[index + 1];
|
|
else
|
|
fb = 256.0;
|
|
|
|
fx = fa + (fb - fa) * (percent - index);
|
|
|
|
*bytepos = (1.0 / 256.0) * fx * total_bytes;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
if (mp3parse->vbri_seek_table && (total_bytes = mp3parse->vbri_bytes) &&
|
|
(total_time = mp3parse->vbri_total_time)) {
|
|
gint i, j;
|
|
gdouble a, b, fa, fb;
|
|
|
|
i = gst_util_uint64_scale (ts, mp3parse->vbri_seek_points - 1, total_time);
|
|
i = CLAMP (i, 0, mp3parse->vbri_seek_points - 1);
|
|
|
|
a = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time,
|
|
mp3parse->vbri_seek_points));
|
|
fa = 0.0;
|
|
for (j = i; j >= 0; j--)
|
|
fa += mp3parse->vbri_seek_table[j];
|
|
|
|
if (i + 1 < mp3parse->vbri_seek_points) {
|
|
b = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time,
|
|
mp3parse->vbri_seek_points));
|
|
fb = fa + mp3parse->vbri_seek_table[i + 1];
|
|
} else {
|
|
b = gst_guint64_to_gdouble (total_time);
|
|
fb = total_bytes;
|
|
}
|
|
|
|
*bytepos = fa + ((fb - fa) / (b - a)) * (gst_guint64_to_gdouble (ts) - a);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_mpeg_audio_parse_bytepos_to_time (GstMpegAudioParse * mp3parse,
|
|
gint64 bytepos, GstClockTime * ts)
|
|
{
|
|
gint64 total_bytes;
|
|
GstClockTime total_time;
|
|
|
|
/* If XING seek table exists use this for byte->time conversion */
|
|
if ((mp3parse->xing_flags & XING_TOC_FLAG) &&
|
|
(total_bytes = mp3parse->xing_bytes) &&
|
|
(total_time = mp3parse->xing_total_time)) {
|
|
gdouble fa, fb, fx;
|
|
gdouble pos;
|
|
gint index;
|
|
|
|
pos = CLAMP ((bytepos * 256.0) / total_bytes, 0.0, 256.0);
|
|
index = CLAMP (pos, 0, 255);
|
|
fa = mp3parse->xing_seek_table_inverse[index];
|
|
if (index < 255)
|
|
fb = mp3parse->xing_seek_table_inverse[index + 1];
|
|
else
|
|
fb = 10000.0;
|
|
|
|
fx = fa + (fb - fa) * (pos - index);
|
|
|
|
*ts = (1.0 / 10000.0) * fx * gst_util_guint64_to_gdouble (total_time);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
if (mp3parse->vbri_seek_table &&
|
|
(total_bytes = mp3parse->vbri_bytes) &&
|
|
(total_time = mp3parse->vbri_total_time)) {
|
|
gint i = 0;
|
|
guint64 sum = 0;
|
|
gdouble a, b, fa, fb;
|
|
|
|
do {
|
|
sum += mp3parse->vbri_seek_table[i];
|
|
i++;
|
|
} while (i + 1 < mp3parse->vbri_seek_points
|
|
&& sum + mp3parse->vbri_seek_table[i] < bytepos);
|
|
i--;
|
|
|
|
a = gst_guint64_to_gdouble (sum);
|
|
fa = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time,
|
|
mp3parse->vbri_seek_points));
|
|
|
|
if (i + 1 < mp3parse->vbri_seek_points) {
|
|
b = a + mp3parse->vbri_seek_table[i + 1];
|
|
fb = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time,
|
|
mp3parse->vbri_seek_points));
|
|
} else {
|
|
b = total_bytes;
|
|
fb = gst_guint64_to_gdouble (total_time);
|
|
}
|
|
|
|
*ts = gst_gdouble_to_guint64 (fa + ((fb - fa) / (b - a)) * (bytepos - a));
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_mpeg_audio_parse_convert (GstBaseParse * parse, GstFormat src_format,
|
|
gint64 src_value, GstFormat dest_format, gint64 * dest_value)
|
|
{
|
|
GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
|
|
gboolean res = FALSE;
|
|
|
|
if (src_format == GST_FORMAT_TIME && dest_format == GST_FORMAT_BYTES)
|
|
res =
|
|
gst_mpeg_audio_parse_time_to_bytepos (mp3parse, src_value, dest_value);
|
|
else if (src_format == GST_FORMAT_BYTES && dest_format == GST_FORMAT_TIME)
|
|
res = gst_mpeg_audio_parse_bytepos_to_time (mp3parse, src_value,
|
|
(GstClockTime *) dest_value);
|
|
|
|
/* if no tables, fall back to default estimated rate based conversion */
|
|
if (!res)
|
|
return gst_base_parse_convert_default (parse, src_format, src_value,
|
|
dest_format, dest_value);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse,
|
|
GstBaseParseFrame * frame)
|
|
{
|
|
GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
|
|
GstTagList *taglist = NULL;
|
|
|
|
/* we will create a taglist (if any of the parameters has changed)
|
|
* to add the tags that changed */
|
|
if (mp3parse->last_posted_crc != mp3parse->last_crc) {
|
|
gboolean using_crc;
|
|
|
|
if (!taglist)
|
|
taglist = gst_tag_list_new_empty ();
|
|
|
|
mp3parse->last_posted_crc = mp3parse->last_crc;
|
|
if (mp3parse->last_posted_crc == CRC_PROTECTED) {
|
|
using_crc = TRUE;
|
|
} else {
|
|
using_crc = FALSE;
|
|
}
|
|
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_CRC,
|
|
using_crc, NULL);
|
|
}
|
|
|
|
if (mp3parse->last_posted_channel_mode != mp3parse->last_mode) {
|
|
if (!taglist)
|
|
taglist = gst_tag_list_new_empty ();
|
|
|
|
mp3parse->last_posted_channel_mode = mp3parse->last_mode;
|
|
|
|
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_MODE,
|
|
gst_mpeg_audio_channel_mode_get_nick (mp3parse->last_mode), NULL);
|
|
}
|
|
|
|
/* tag sending done late enough in hook to ensure pending events
|
|
* have already been sent */
|
|
if (taglist != NULL || !mp3parse->sent_codec_tag) {
|
|
GstCaps *caps;
|
|
|
|
if (taglist == NULL)
|
|
taglist = gst_tag_list_new_empty ();
|
|
|
|
/* codec tag */
|
|
caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
|
|
if (G_UNLIKELY (caps == NULL)) {
|
|
gst_tag_list_unref (taglist);
|
|
|
|
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;
|
|
}
|
|
}
|
|
gst_pb_utils_add_codec_description_to_tag_list (taglist,
|
|
GST_TAG_AUDIO_CODEC, caps);
|
|
gst_caps_unref (caps);
|
|
|
|
if (mp3parse->hdr_bitrate > 0 && mp3parse->xing_bitrate == 0 &&
|
|
mp3parse->vbri_bitrate == 0) {
|
|
/* We don't have a VBR bitrate, so post the available bitrate as
|
|
* nominal and let baseparse calculate the real bitrate */
|
|
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
|
|
GST_TAG_NOMINAL_BITRATE, mp3parse->hdr_bitrate, NULL);
|
|
}
|
|
|
|
/* also signals the end of first-frame processing */
|
|
mp3parse->sent_codec_tag = TRUE;
|
|
}
|
|
|
|
/* if the taglist exists, we need to update it so it gets sent out */
|
|
if (taglist) {
|
|
gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE);
|
|
gst_tag_list_unref (taglist);
|
|
}
|
|
|
|
/* usual clipping applies */
|
|
frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP;
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
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, "parsed");
|
|
}
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_mpeg_audio_parse_get_sink_caps (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);
|
|
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) {
|
|
/* Remove the parsed field */
|
|
peercaps = gst_caps_make_writable (peercaps);
|
|
remove_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;
|
|
}
|