bump version

Since channel map was not initialized, first channel was copied to every
other channel.

.gitignore

@@ -0,0 +1,22 @@
+*.o
+*~
+Makefile
+Makefile.in
+aclocal.m4
+autom4te.cache
+compile
+config.guess
+config.h
+config.h.in
+config.log
+config.status
+config.sub
+configure
+depcomp
+install-sh
+missing
+missings/.deps/
+src/.deps/
+src/.dirstamp
+stamp-h1
+fdkaac

ChangeLog

@@ -1,7 +1,121 @@
-2013-10-25  nu774 <honeycomb77@gmail.com>
+2014-01-18  nu774 <honeycomb77@gmail.com>
 
   * update ChangeLog [HEAD]
 
+  * bump version [v0.5.2]
+
+  * fix reading of caf file without chan chunk
+
+2013-11-17  nu774 <honeycomb77@gmail.com>
+
+  * Merge pull request #9 from rbrito/fix-typo [origin/master]
+
+2013-11-18  Rogério Brito <rbrito@ime.usp.br>
+
+  * man: Regen manpage with hyphens escaped.
+
+  * README: Remove trailing whitespaces that end up in the manpages.
+
+  * README: Fix typo in bandwidth to match CLI options.
+
+2013-11-08  nu774 <honeycomb77@gmail.com>
+
+  * add genman.sh, update fdkaac.1
+
+  * update ChangeLog
+
+  * bump version [v0.5.1]
+
+  * fix to use libFDKAAC signaling mode 1
+
+2013-11-05  nu774 <honeycomb77@gmail.com>
+
+  * fix README
+
+2013-11-04  nu774 <honeycomb77@gmail.com>
+
+  * rename README.md -> README
+
+  * Update README -> README.md, generate groff manpage from README.md
+
+  * update ChangeLog
+
+  * update git2changelog to accept non-ascii output
+
+  * add manpage
+
+2013-11-03  nu774 <honeycomb77@gmail.com>
+
+  * fix gcc warnings
+
+  * Merge pull request #7 from rbrito/misc-fixes
+
+2013-11-03  Rogério Brito <rbrito@ime.usp.br>
+
+  * gitignore: Add list of files to ignore.
+
+2013-11-03  nu774 <honeycomb77@gmail.com>
+
+  * update ChangeLog
+
+  * bump version [v0.5.0]
+
+  * add --sbr-ratio to support AACENC_SBR_RATIO appeared on libFDK 3.4.12
+
+  * support 7.1 channel mode added on FDK 3.4.12
+
+2013-10-30  nu774 <honeycomb77@gmail.com>
+
+  * update ChangeLog
+
+  * bump version [v0.4.2]
+
+  * use tell() to obtain data chunk offset
+
+  * rename aacenc_result_t -> aacenc_frame_t, simplify write_sample()
+
+  * prepend 1 sample zero padding in case of SBR and enc_delay is odd
+
+  * cleanup interface of aac_encode_frame()
+
+  * add some copyright notice
+
+2013-10-29  nu774 <honeycomb77@gmail.com>
+
+  * smart padding for better gapless playback
+
+  * fix unused variable warning
+
+  * fix warning: cast size_t as sprintf() arg to int
+
+  * fix vcxproj
+
+  * fix pcm_seek() to inline
+
+2013-10-27  nu774 <honeycomb77@gmail.com>
+
+  * bump version [v0.4.1]
+
+  * add --include-sbr-delay
+
+  * fix help message: show -I as shorthand for --ignorelength
+
+  * remove --sbr-signaling
+
+2013-10-26  nu774 <honeycomb77@gmail.com>
+
+  * re-fix #ifdef cond for lrint() [old]
+
+  * tag mapping: add recorded date and tempo, remove performer->artist
+
+2013-10-25  nu774 <honeycomb77@gmail.com>
+
+  * fix MSVC12 build issue
+
+  * fix build issue on platform where fileno is a naive macro
+
+  * update ChangeLog
+
   * bump version [v0.4.0]
 
   * update README

MSVC/fdkaac.vcxproj

@@ -98,6 +98,8 @@ copy ..\fdk-aac\libSYS\include\machine_type.h include\fdk-aac\ </Command>
     <ClCompile Include="..\src\aacenc.c" />
     <ClCompile Include="..\src\caf_reader.c" />
     <ClCompile Include="..\src\compat_win32.c" />
+    <ClCompile Include="..\src\extrapolater.c" />
+    <ClCompile Include="..\src\lpc.c" />
     <ClCompile Include="..\src\lpcm.c" />
     <ClCompile Include="..\src\m4af.c" />
     <ClCompile Include="..\src\main.c" />
@@ -111,7 +113,10 @@ copy ..\fdk-aac\libSYS\include\machine_type.h include\fdk-aac\ </Command>
   <ItemGroup>
     <ClInclude Include="..\missings\getopt.h" />
     <ClInclude Include="..\src\aacenc.h" />
+    <ClInclude Include="..\src\caf_reader.h" />
+    <ClInclude Include="..\src\catypes.h" />
     <ClInclude Include="..\src\compat.h" />
+    <ClInclude Include="..\src\lpc.h" />
     <ClInclude Include="..\src\lpcm.h" />
     <ClInclude Include="..\src\m4af.h" />
     <ClInclude Include="..\src\m4af_endian.h" />

MSVC/fdkaac.vcxproj.filters

@@ -24,6 +24,12 @@
     <ClCompile Include="..\src\compat_win32.c">
       <Filter>Source Files</Filter>
     </ClCompile>
+    <ClCompile Include="..\src\extrapolater.c">
+      <Filter>Source Files</Filter>
+    </ClCompile>
+    <ClCompile Include="..\src\lpc.c">
+      <Filter>Source Files</Filter>
+    </ClCompile>
     <ClCompile Include="..\src\lpcm.c">
       <Filter>Source Files</Filter>
     </ClCompile>
@@ -62,6 +68,9 @@
     <ClInclude Include="..\src\compat.h">
       <Filter>Header Files</Filter>
     </ClInclude>
+    <ClInclude Include="..\src\lpc.h">
+      <Filter>Header Files</Filter>
+    </ClInclude>
     <ClInclude Include="..\src\lpcm.h">
       <Filter>Header Files</Filter>
     </ClInclude>

Makefile.am

@@ -6,6 +6,8 @@ bin_PROGRAMS = fdkaac
 fdkaac_SOURCES = \
     src/aacenc.c               \
     src/caf_reader.c           \
+    src/extrapolater.c         \
+    src/lpc.c                  \
     src/lpcm.c                 \
     src/m4af.c                 \
     src/main.c                 \
@@ -16,6 +18,8 @@ fdkaac_SOURCES = \
     src/progress.c             \
     src/wav_reader.c
 
+dist_man_MANS = man/fdkaac.1
+
 fdkaac_LDADD = \
     @LIBICONV@ -lfdk-aac -lm
 

README

@@ -1,122 +1,332 @@
-==========================================================================
-fdkaac - command line frontend encoder for libfdk-aac
-==========================================================================
+% FDKAAC(1)
+% nu774 <honeycomb77@gmail.com>
+% November, 2013
 
-Prerequisites
--------------
-You need libfdk-aac.
-On Posix environment, you will also need GNU gettext (for iconv.m4) and
-GNU autoconf/automake.
+NAME
+====
 
-How to build on Posix environment
----------------------------------
-First, you need to build libfdk-aac and install on your system.
-Once you have done it, the following will do the task.
-(MinGW build can be done the same way, and doesn't require gettext/iconv)
+fdkaac - command line frontend for libfdk-aac encoder
 
-$ autoreconf -i
-$ ./configure && make && make install
+SYNOPSIS
+========
 
-How to build on MSVC
---------------------
-First you have to extract libfdk-aac source here, so that directory tree will
-look like the following:
-+- fdk-aac ---+-documentation
-|             +-libAACdec
-|             +-libAACenc
-|             :
-+- m4
-+- missings
-+- MSVC
-+- src
+**fdkaac** [OPTIONS] [FILE]
 
-MSVC solution for Visual Studio 2010 is under MSVC directory.
+DESCRIPTION
+===========
 
-Available input format
-----------------------
-WAV, RF64, CAF, RAW, upto 32bit int / 64bit float format is supported.
-Metadata in CAF info chunk can be read and copied to the resulting m4a.
-This is especially useful and works well when you pipe from ffmpeg via CAF.
-For example, you can copy tag from original "foo.flac" to "foo.m4a"
-through the following pipeline:
+**fdkaac** reads linear PCM audio in either WAV, raw PCM, or CAF format,
+and encodes it into either M4A / AAC file.
 
-$ ffmpeg -i foo.flac -f caf - | fdkaac -m3 - -o foo.m4a
+If the input file is "-", data is read from stdin. Likewise, if the
+output file is "-", data is written to stdout if one of streamable AAC
+transport formats are selected by **-f**.
 
-Since FDK AAC encoder is implemented based on fixed point integer,
-encoder itself handles 16bit input only.
-Therefore, when feeding non-integer input, be careful so that input doesn't
-exceed 0dBFS to avoid hard clips.
-You might also want to apply dither/noise shape beforehand when your input
-has higher resolution.
+When CAF input and M4A output is used, tags in CAF file are copied into
+the resulting M4A.
 
-Note that fdkaac doesn't automatically resample for you
-when input samplerate is not supported by AAC spec.
+OPTIONS
+=======
 
-Tagging Options
----------------
-Generic tagging options like --tag, --tag-from-file, --long-tag allows you
-to set arbitrary tags.
-Available tags and their fcc (four char code) for --tag and --tag-from-file
-can be found at http://code.google.com/p/mp4v2/wiki/iTunesMetadata
+-h, --help
+:   Show command help
 
-For tags such as Artist where first char of fcc is copyright sign,
-you can skip first char and just say like --tag="ART:Foo Bar" or
---tag-from-file=lyr:/path/to/your/lyrics.txt
+-o \<FILE\>
+:   Output filename.
 
-Currently, --tag-from-file just stores file contents into m4a without any
-character encoding / line terminater conversion.
-Therefore, only use UTF-8 (without BOM) when setting text tags by this option.
+-p, --profile \<n\>
+:   Target profile (MPEG4 audio object type, AOT)
 
-On the other hand, --tag / --long-tag (and other command line arguments) are
-converted from locale character encoding to UTF-8 on Posix environment.
-On Windows, command line arguments are always treated as Unicode.
+    2
+    :   MPEG-4 AAC LC (default)
 
-Tagging using JSON
-------------------
-With --tag-from-json, fdkaac can read JSON file and set tags from it.
-By default, tags are assumed to be in the root object(dictionary) like this:
+    5
+    :   MPEG-4 HE-AAC (SBR)
 
-{
-  "title": "No Expectations",
-  "artist": "The Rolling Stones",
-  "album": "Beggars Banquet",
-  "track": 2
-}
+    29
+    :   MPEG-4 HE-AAC v2 (SBR+PS)
 
-In this case, you can simply specify the filename like:
---tag-from-json=/path/to/json
+    23
+    :   MPEG-4 AAC LD
 
-If the object containing tags is placed somewhere else, you can optionally
-specify the path of the object with dotted notation.
+    39
+    :   MPEG-4 AAC ELD
 
-{
-  "format" : {
-    "filename" : "Middle Curse.flac",
-    "nb_streams" : 1,
-    "format_name" : "flac",
-    "format_long_name" : "raw FLAC",
-    "start_time" : "N/A",
-    "duration" : "216.146667",
-    "size" : "11851007.000000",
-    "bit_rate" : "438628.000000",
-    "tags" : {
-      "ALBUM" : "Scary World Theory",
-      "ARTIST" : "Lali Puna",
-      "DATE" : "2001",
-      "DISCID" : "9208CC0A",
-      "TITLE" : "Middle Curse",
-      "TRACKTOTAL" : "10",
-      "track" : "2"
-    }
-  }
-}
+    129
+    :   MPEG-2 AAC LC
 
-In this example, tags are placed under the object "format.tags".
-("format" is a child of the root, and "tags" is a child of the "format").
-In this case, you can say:
---tag-from-json=/path/to/json?format.tags
+    132
+    :   MPEG-2 HE-AAC (SBR)
 
-For your information, ffprobe of ffmpeg project (or avprobe of libav) can
-output media information/metadata in json format like this.
+    156
+    :   MPEG-2 HE-AAC v2 (SBR+PS)
+
+-b, --bitrate \<n\>
+:   Target bitrate (for CBR)
+
+-m, --bitrate-mode \<n\>
+:   Bitrate configuration mode. Available VBR quality value depends on
+    other parameters such as profile, sample rate, or number of
+    channels.
+
+    0
+    :   CBR (default)
+
+    1-5
+    :   VBR (higher value -\> higher bitrate)
+
+-w, --bandwidth \<n\>
+:   Frequency bandwidth (lowpass cut-off frequency) in Hz. Available on
+    AAC LC only.
+
+-a, --afterburner \<n\>
+:   Configure afterburner mode. When enabled, quality is increased at
+    the expense of additional computational workload.
+
+    0
+    :   Off
+
+    1
+    :   On (default)
+
+-L, --lowdelay-sbr \<n\>
+:   Configure SBR activity on AAC ELD.
+
+    -1
+    :   Use ELD SBR auto configuration
+
+    0
+    :   Disable SBR on ELD (default)
+
+    1
+    :   Enable SBR on ELD
+
+-s, --sbr-ratio \<n\>
+:   Controls activation of downsampled SBR.
+
+    0
+    :   Use lib default (default)
+
+    1
+    :   Use downsampled SBR (default for ELD+SBR)
+
+    2
+    :   Use dual-rate SBR (default for HE-AAC)
+
+    Dual-rate SBR is what is normally used for HE-AAC, where AAC is
+    encoded at half the sample rate of SBR, hence "dual rate". On the
+    other hand, downsampled SBR uses same sample rate for both of AAC
+    and SBR (single rate), therefore downsampled SBR typically consumes
+    more bitrate.
+
+    Downsampled SBR is newly introduced feature in FDK encoder library
+    version 3.4.12. When libfdk-aac in the system doesn't support this,
+    dual-rate SBR will be used. When available, dual-rate SBR is the
+    default for HE-AAC and downsampled SBR is the default for ELD+SBR.
+
+    Note that downsampled HE-AAC is not so common as dual-rate one. When
+    downsampled HE-AAC is selected, **fdkaac** is forced to choose
+    explicit hierarchical SBR signaling, which (at least) iTunes doesn't
+    accept.
+
+-f, --transport-format \<n\>
+:   Transport format. Tagging and gapless playback is only available on
+    M4A. Streaming to stdout is only available on others.
+
+    0
+    :   M4A (default)
+
+    1
+    :   ADIF
+
+    2
+    :   ADTS
+
+    6
+    :   LATM MCP=1
+
+    7
+    :   LATM MCP=0
+
+    10
+    :   LOAS/LATM (LATM within LOAS)
+
+-C, --adts-crc-check
+:   Add CRC protection on ADTS header.
+
+-h, --header-period \<n\>
+:   StreamMuxConfig/PCE repetition period in the transport layer.
+
+-G, --gapless-mode \<n\>
+:   Method to declare amount of encoder delay (and padding) in M4A
+    container. These values are mandatory for proper gapless playback on
+    player side.
+
+    0
+    :   iTunSMPB (default)
+
+    1
+    :   ISO standard (edts and sgpd)
+
+    2
+    :   Both
+
+--include-sbr-delay
+:   When specified, count SBR decoder delay in encoder delay.
+
+    This is not iTunes compatible and will lead to gapless playback
+    issue on LC only decoder, but this is the default behavior of FDK
+    library.
+
+    Whether counting SBR decoder delay in encoder delay or not result in
+    incompatibility in gapless playback. You should pick which one will
+    work for your favorite player.
+
+    However, it's better not to choose SBR at all if you want gapless
+    playback. LC doesn't have such issues.
+
+-I, --ignorelength
+:   Ignore length field of data chunk in input WAV file.
+
+-S, --silent
+:   Don't print progress messages.
+
+--moov-before-mdat
+:   Place moov box before mdat box in M4A container. This option might
+    be important for some hardware players, that are known to refuse
+    moov box placed after mdat box.
+
+-R, --raw
+:   Regard input as raw PCM.
+
+--raw-channels \<n\>
+:   Specify number of channels of raw input (default: 2)
+
+--raw-rate \<n\>
+:   Specify sample rate of raw input (default: 44100)
+
+--raw-format \<spec\>
+:   Specify sample format of raw input (default: "S16L"). **Spec** is as
+    the following (case insensitive):
+
+    1st char -- type of sample
+    :   **S** (igned) | **U** (nsigned) | **F** (loat)
+
+    2nd part (in digits)
+    :   bits per channel
+
+    Last char -- endianness (can be omitted)
+    :   **L** (ittle, default) | **B** (ig)
+
+--title \<string\>
+:   Set title tag.
+
+--artist \<string\>
+:   Set artist tag.
+
+--album \<string\>
+:   Set album tag.
+
+--genre \<string\>
+:   Set genre tag.
+
+--date \<string\>
+:   Set date tag.
+
+--composer \<string\>
+:   Set composer tag.
+
+--grouping \<string\>
+:   Set grouping tag.
+
+--comment \<string\>
+:   Set comment tag.
+
+--album-artist \<string\>
+:   Set album artist tag.
+
+--track \<number[/total]\>
+:   Set track tag, with or without number of total tracks.
+
+--disk \<number[/total]\>
+:   Set disk tag, with or without number of total discs.
+
+--tempo \<n\>
+:   Set tempo (BPM) tag.
+
+--tag \<fcc\>:\<value\>
+:   Set iTunes predefined tag with explicit fourcc key and value. See
+    [https://code.google.com/p/mp4v2/wiki/iTunesMetadata](https://code.google.com/p/mp4v2/wiki/iTunesMetadata)
+    for known predefined keys. You can omit first char of **fcc** when
+    it is the copyright sign.
+
+--tag-from-file \<fcc\>:\<filename\>
+:   Same as --tag, but set content of file as tag value.
+
+--long-tag \<name\>:\<value\>
+:   Set arbitrary tag as iTunes custom metadata. Stored in
+    com.apple.iTunes field.
+
+--tag-from-json \<filename[?dot\_notation]\>
+:   Read tags from JSON. By default, tags are assumed to be direct
+    children of the root object in JSON. Optionally you can specify
+    arbitrary dot notation to locate the object containing tags.
+
+
+EXAMPLES
+========
+
+Encode WAV file into a M4A file. MPEG4 AAC LC, VBR quality 3:
+
+    fdkaac -m3 foo.wav
+
+Encode WAV file into a M4A file. MPEG4 HE-AAC, bitrate 64kbps:
+
+    fdkaac -p5 -b64 foo.wav
+
+Piping from **ffmpeg** (you need version supporting CAF output):
+
+    ffmpeg -i foo.flac -f caf - | fdkaac -b128 - -o foo.m4a
+
+Import tags via json:
+
+    ffprobe -v 0 -of json -show_format foo.flac >foo.json
+
+    flac -dc foo.flac | \
+    fdkaac - -ox.m4a -m2 --import-tag-from-json=foo.json?format.tags
+
+NOTES
+=====
+
+Upto 32bit integer or 64bit floating point format is supported as input.
+However, FDK library is implemented based on fixed point math and only
+supports 16bit integer PCM. Therefore, be wary of clipping. You might
+want to dither/noise shape beforehand when your input has higher
+resolution.
+
+Following channel layouts are supported by the encoder.
+
+1ch
+:   C
+
+2ch
+:   L R
+
+3ch
+:   C L R
+
+4ch
+:   C L R Cs
+
+5ch
+:   C L R Ls Rs
+
+5.1ch
+:   C L R Ls Rs LFE
+
+7.1ch (front)
+:   C Lc Rc L R Ls Rs LFE
+
+7.1ch (rear)
+:   C L R Ls Rs Rls Rrs LFE
 
-Note that not all tags can be read/written this way.

genman.sh

@@ -0,0 +1,2 @@
+#!/bin/sh
+pandoc -s -f markdown -t man README >fdkaac.1 && mv -f fdkaac.1 man/fdkaac.1

git2changelog.py

@@ -39,6 +39,6 @@ with Popen(GITLOG_CMD, shell=False, stdout=PIPE).stdout as pipe:
     commits = parse_gitlog(pipe)
     commits_by_date_author = groupby(commits, key=lambda x: (x.date, x.author))
     for (date, author), commits in commits_by_date_author:
-        output('{0}  {1}\n\n'.format(date, author))
+        output(u'{0}  {1}\n\n'.format(date, author).encode('utf-8'))
         for c in commits:
-            output('  * {0}{1}\n\n'.format(c.subject, c.ref))
+            output(u'  * {0}{1}\n\n'.format(c.subject, c.ref).encode('utf-8'))

man/fdkaac.1

@@ -0,0 +1,493 @@
+.TH FDKAAC 1 "November, 2013" 
+.SH NAME
+.PP
+fdkaac \- command line frontend for libfdk\-aac encoder
+.SH SYNOPSIS
+.PP
+\f[B]fdkaac\f[] [OPTIONS][FILE]
+.SH DESCRIPTION
+.PP
+\f[B]fdkaac\f[] reads linear PCM audio in either WAV, raw PCM, or CAF
+format, and encodes it into either M4A / AAC file.
+.PP
+If the input file is "\-", data is read from stdin.
+Likewise, if the output file is "\-", data is written to stdout if one
+of streamable AAC transport formats are selected by \f[B]\-f\f[].
+.PP
+When CAF input and M4A output is used, tags in CAF file are copied into
+the resulting M4A.
+.SH OPTIONS
+.TP
+.B \-h, \-\-help
+Show command help
+.RS
+.RE
+.TP
+.B \-o <FILE>
+Output filename.
+.RS
+.RE
+.TP
+.B \-p, \-\-profile <n>
+Target profile (MPEG4 audio object type, AOT)
+.RS
+.TP
+.B 2
+MPEG\-4 AAC LC (default)
+.RS
+.RE
+.TP
+.B 5
+MPEG\-4 HE\-AAC (SBR)
+.RS
+.RE
+.TP
+.B 29
+MPEG\-4 HE\-AAC v2 (SBR+PS)
+.RS
+.RE
+.TP
+.B 23
+MPEG\-4 AAC LD
+.RS
+.RE
+.TP
+.B 39
+MPEG\-4 AAC ELD
+.RS
+.RE
+.TP
+.B 129
+MPEG\-2 AAC LC
+.RS
+.RE
+.TP
+.B 132
+MPEG\-2 HE\-AAC (SBR)
+.RS
+.RE
+.TP
+.B 156
+MPEG\-2 HE\-AAC v2 (SBR+PS)
+.RS
+.RE
+.RE
+.TP
+.B \-b, \-\-bitrate <n>
+Target bitrate (for CBR)
+.RS
+.RE
+.TP
+.B \-m, \-\-bitrate\-mode <n>
+Bitrate configuration mode.
+Available VBR quality value depends on other parameters such as profile,
+sample rate, or number of channels.
+.RS
+.TP
+.B 0
+CBR (default)
+.RS
+.RE
+.TP
+.B 1\-5
+VBR (higher value \-> higher bitrate)
+.RS
+.RE
+.RE
+.TP
+.B \-w, \-\-bandwidth <n>
+Frequency bandwidth (lowpass cut\-off frequency) in Hz.
+Available on AAC LC only.
+.RS
+.RE
+.TP
+.B \-a, \-\-afterburner <n>
+Configure afterburner mode.
+When enabled, quality is increased at the expense of additional
+computational workload.
+.RS
+.TP
+.B 0
+Off
+.RS
+.RE
+.TP
+.B 1
+On (default)
+.RS
+.RE
+.RE
+.TP
+.B \-L, \-\-lowdelay\-sbr <n>
+Configure SBR activity on AAC ELD.
+.RS
+.TP
+.B \-1
+Use ELD SBR auto configuration
+.RS
+.RE
+.TP
+.B 0
+Disable SBR on ELD (default)
+.RS
+.RE
+.TP
+.B 1
+Enable SBR on ELD
+.RS
+.RE
+.RE
+.TP
+.B \-s, \-\-sbr\-ratio <n>
+Controls activation of downsampled SBR.
+.RS
+.TP
+.B 0
+Use lib default (default)
+.RS
+.RE
+.TP
+.B 1
+Use downsampled SBR (default for ELD+SBR)
+.RS
+.RE
+.TP
+.B 2
+Use dual\-rate SBR (default for HE\-AAC)
+.RS
+.RE
+.PP
+Dual\-rate SBR is what is normally used for HE\-AAC, where AAC is
+encoded at half the sample rate of SBR, hence "dual rate".
+On the other hand, downsampled SBR uses same sample rate for both of AAC
+and SBR (single rate), therefore downsampled SBR typically consumes more
+bitrate.
+.PP
+Downsampled SBR is newly introduced feature in FDK encoder library
+version 3.4.12.
+When libfdk\-aac in the system doesn\[aq]t support this, dual\-rate SBR
+will be used.
+When available, dual\-rate SBR is the default for HE\-AAC and
+downsampled SBR is the default for ELD+SBR.
+.PP
+Note that downsampled HE\-AAC is not so common as dual\-rate one.
+When downsampled HE\-AAC is selected, \f[B]fdkaac\f[] is forced to
+choose explicit hierarchical SBR signaling, which (at least) iTunes
+doesn\[aq]t accept.
+.RE
+.TP
+.B \-f, \-\-transport\-format <n>
+Transport format.
+Tagging and gapless playback is only available on M4A.
+Streaming to stdout is only available on others.
+.RS
+.TP
+.B 0
+M4A (default)
+.RS
+.RE
+.TP
+.B 1
+ADIF
+.RS
+.RE
+.TP
+.B 2
+ADTS
+.RS
+.RE
+.TP
+.B 6
+LATM MCP=1
+.RS
+.RE
+.TP
+.B 7
+LATM MCP=0
+.RS
+.RE
+.TP
+.B 10
+LOAS/LATM (LATM within LOAS)
+.RS
+.RE
+.RE
+.TP
+.B \-C, \-\-adts\-crc\-check
+Add CRC protection on ADTS header.
+.RS
+.RE
+.TP
+.B \-h, \-\-header\-period <n>
+StreamMuxConfig/PCE repetition period in the transport layer.
+.RS
+.RE
+.TP
+.B \-G, \-\-gapless\-mode <n>
+Method to declare amount of encoder delay (and padding) in M4A
+container.
+These values are mandatory for proper gapless playback on player side.
+.RS
+.TP
+.B 0
+iTunSMPB (default)
+.RS
+.RE
+.TP
+.B 1
+ISO standard (edts and sgpd)
+.RS
+.RE
+.TP
+.B 2
+Both
+.RS
+.RE
+.RE
+.TP
+.B \-\-include\-sbr\-delay
+When specified, count SBR decoder delay in encoder delay.
+.RS
+.PP
+This is not iTunes compatible and will lead to gapless playback issue on
+LC only decoder, but this is the default behavior of FDK library.
+.PP
+Whether counting SBR decoder delay in encoder delay or not result in
+incompatibility in gapless playback.
+You should pick which one will work for your favorite player.
+.PP
+However, it\[aq]s better not to choose SBR at all if you want gapless
+playback.
+LC doesn\[aq]t have such issues.
+.RE
+.TP
+.B \-I, \-\-ignorelength
+Ignore length field of data chunk in input WAV file.
+.RS
+.RE
+.TP
+.B \-S, \-\-silent
+Don\[aq]t print progress messages.
+.RS
+.RE
+.TP
+.B \-\-moov\-before\-mdat
+Place moov box before mdat box in M4A container.
+This option might be important for some hardware players, that are known
+to refuse moov box placed after mdat box.
+.RS
+.RE
+.TP
+.B \-R, \-\-raw
+Regard input as raw PCM.
+.RS
+.RE
+.TP
+.B \-\-raw\-channels <n>
+Specify number of channels of raw input (default: 2)
+.RS
+.RE
+.TP
+.B \-\-raw\-rate <n>
+Specify sample rate of raw input (default: 44100)
+.RS
+.RE
+.TP
+.B \-\-raw\-format <spec>
+Specify sample format of raw input (default: "S16L").
+\f[B]Spec\f[] is as the following (case insensitive):
+.RS
+.TP
+.B 1st char \-\- type of sample
+\f[B]S\f[] (igned) | \f[B]U\f[] (nsigned) | \f[B]F\f[] (loat)
+.RS
+.RE
+.TP
+.B 2nd part (in digits)
+bits per channel
+.RS
+.RE
+.TP
+.B Last char \-\- endianness (can be omitted)
+\f[B]L\f[] (ittle, default) | \f[B]B\f[] (ig)
+.RS
+.RE
+.RE
+.TP
+.B \-\-title <string>
+Set title tag.
+.RS
+.RE
+.TP
+.B \-\-artist <string>
+Set artist tag.
+.RS
+.RE
+.TP
+.B \-\-album <string>
+Set album tag.
+.RS
+.RE
+.TP
+.B \-\-genre <string>
+Set genre tag.
+.RS
+.RE
+.TP
+.B \-\-date <string>
+Set date tag.
+.RS
+.RE
+.TP
+.B \-\-composer <string>
+Set composer tag.
+.RS
+.RE
+.TP
+.B \-\-grouping <string>
+Set grouping tag.
+.RS
+.RE
+.TP
+.B \-\-comment <string>
+Set comment tag.
+.RS
+.RE
+.TP
+.B \-\-album\-artist <string>
+Set album artist tag.
+.RS
+.RE
+.TP
+.B \-\-track <number[/total]>
+Set track tag, with or without number of total tracks.
+.RS
+.RE
+.TP
+.B \-\-disk <number[/total]>
+Set disk tag, with or without number of total discs.
+.RS
+.RE
+.TP
+.B \-\-tempo <n>
+Set tempo (BPM) tag.
+.RS
+.RE
+.TP
+.B \-\-tag <fcc>:<value>
+Set iTunes predefined tag with explicit fourcc key and value.
+See <https://code.google.com/p/mp4v2/wiki/iTunesMetadata> for known
+predefined keys.
+You can omit first char of \f[B]fcc\f[] when it is the copyright sign.
+.RS
+.RE
+.TP
+.B \-\-tag\-from\-file <fcc>:<filename>
+Same as \-\-tag, but set content of file as tag value.
+.RS
+.RE
+.TP
+.B \-\-long\-tag <name>:<value>
+Set arbitrary tag as iTunes custom metadata.
+Stored in com.apple.iTunes field.
+.RS
+.RE
+.TP
+.B \-\-tag\-from\-json <filename[?dot_notation]>
+Read tags from JSON.
+By default, tags are assumed to be direct children of the root object in
+JSON.
+Optionally you can specify arbitrary dot notation to locate the object
+containing tags.
+.RS
+.RE
+.SH EXAMPLES
+.PP
+Encode WAV file into a M4A file.
+MPEG4 AAC LC, VBR quality 3:
+.IP
+.nf
+\f[C]
+fdkaac\ \-m3\ foo.wav
+\f[]
+.fi
+.PP
+Encode WAV file into a M4A file.
+MPEG4 HE\-AAC, bitrate 64kbps:
+.IP
+.nf
+\f[C]
+fdkaac\ \-p5\ \-b64\ foo.wav
+\f[]
+.fi
+.PP
+Piping from \f[B]ffmpeg\f[] (you need version supporting CAF output):
+.IP
+.nf
+\f[C]
+ffmpeg\ \-i\ foo.flac\ \-f\ caf\ \-\ |\ fdkaac\ \-b128\ \-\ \-o\ foo.m4a
+\f[]
+.fi
+.PP
+Import tags via json:
+.IP
+.nf
+\f[C]
+ffprobe\ \-v\ 0\ \-of\ json\ \-show_format\ foo.flac\ >foo.json
+
+flac\ \-dc\ foo.flac\ |\ \\
+fdkaac\ \-\ \-ox.m4a\ \-m2\ \-\-import\-tag\-from\-json=foo.json?format.tags
+\f[]
+.fi
+.SH NOTES
+.PP
+Upto 32bit integer or 64bit floating point format is supported as input.
+However, FDK library is implemented based on fixed point math and only
+supports 16bit integer PCM.
+Therefore, be wary of clipping.
+You might want to dither/noise shape beforehand when your input has
+higher resolution.
+.PP
+Following channel layouts are supported by the encoder.
+.TP
+.B 1ch
+C
+.RS
+.RE
+.TP
+.B 2ch
+L R
+.RS
+.RE
+.TP
+.B 3ch
+C L R
+.RS
+.RE
+.TP
+.B 4ch
+C L R Cs
+.RS
+.RE
+.TP
+.B 5ch
+C L R Ls Rs
+.RS
+.RE
+.TP
+.B 5.1ch
+C L R Ls Rs LFE
+.RS
+.RE
+.TP
+.B 7.1ch (front)
+C Lc Rc L R Ls Rs LFE
+.RS
+.RE
+.TP
+.B 7.1ch (rear)
+C L R Ls Rs Rls Rrs LFE
+.RS
+.RE
+.SH AUTHORS
+nu774 <honeycomb77@gmail.com>.

src/aacenc.c

@@ -13,6 +13,17 @@
 #include <string.h>
 #include "aacenc.h"
 
+int aacenc_is_sbr_ratio_available()
+{
+#if AACENCODER_LIB_VL0 < 3 || (AACENCODER_LIB_VL0==3 && AACENCODER_LIB_VL1<4)
+    return 0;
+#else
+    LIB_INFO lib_info;
+    aacenc_get_lib_info(&lib_info);
+    return lib_info.version > 0x03040000;
+#endif
+}
+
 int aacenc_is_sbr_active(const aacenc_param_t *params)
 {
     switch (params->profile) {
@@ -26,66 +37,31 @@ int aacenc_is_sbr_active(const aacenc_param_t *params)
     return 0;
 }
 
-static const unsigned aacenc_sampling_freq_tab[] = {
-    96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 
-    16000, 12000, 11025, 8000, 7350, 0, 0, 0
-};
-
-static
-unsigned sampling_freq_index(unsigned rate)
+int aacenc_is_dual_rate_sbr(const aacenc_param_t *params)
 {
-    unsigned i;
-    for (i = 0; aacenc_sampling_freq_tab[i]; ++i)
-        if (aacenc_sampling_freq_tab[i] == rate)
-            return i;
-    return 0xf;
+    if (params->profile == AOT_PS || params->profile == AOT_MP2_PS)
+        return 1;
+    else if (params->profile == AOT_SBR || params->profile == AOT_MP2_SBR)
+        return params->sbr_ratio == 0 || params->sbr_ratio == 2;
+    else if (params->profile == AOT_ER_AAC_ELD && params->lowdelay_sbr)
+        return params->sbr_ratio == 2;
+    return 0;
 }
 
-/*
- * Append backward compatible SBR/PS signaling to implicit signaling ASC,
- * if SBR/PS is present.
- */
-int aacenc_mp4asc(const aacenc_param_t *params,
-                  const uint8_t *asc, uint32_t ascsize,
-                  uint8_t *outasc, uint32_t *outsize)
+void aacenc_get_lib_info(LIB_INFO *info)
 {
-    unsigned asc_sfreq = aacenc_sampling_freq_tab[(asc[0]&0x7)<<1 |asc[1]>>7];
-
-    switch (params->profile) {
-    case AOT_SBR:
-    case AOT_PS:
-        if (*outsize < ascsize + 3)
-            return -1;
-        memcpy(outasc, asc, ascsize);
-        /* syncExtensionType:11 (value:0x2b7) */
-        outasc[ascsize+0] = 0x2b << 1;
-        outasc[ascsize+1] = 0x7 << 5;
-        /* extensionAudioObjectType:5 (value:5)*/
-        outasc[ascsize+1] |= 5;
-        /* sbrPresentFlag:1 (value:1) */
-        outasc[ascsize+2] = 0x80;
-        /* extensionSamplingFrequencyIndex:4 */
-        outasc[ascsize+2] |= sampling_freq_index(asc_sfreq << 1) << 3;
-        if (params->profile == AOT_SBR) {
-            *outsize = ascsize + 3;
-            break;
+    LIB_INFO *lib_info = 0;
+    lib_info = calloc(FDK_MODULE_LAST, sizeof(LIB_INFO));
+    if (aacEncGetLibInfo(lib_info) == AACENC_OK) {
+        int i;
+        for (i = 0; i < FDK_MODULE_LAST; ++i) {
+            if (lib_info[i].module_id == FDK_AACENC) {
+                memcpy(info, &lib_info[i], sizeof(LIB_INFO));
+                break;
+            }
         }
-        if (*outsize < ascsize + 5)
-            return -1;
-        /* syncExtensionType:11 (value:0x548) */
-        outasc[ascsize+2] |= 0x5;
-        outasc[ascsize+3] = 0x48;
-        /* psPresentFlag:1 (value:1) */
-        outasc[ascsize+4] = 0x80;
-        *outsize = ascsize + 5;
-        break;
-    default:
-        if (*outsize < ascsize)
-            return -1;
-        memcpy(outasc, asc, ascsize);
-        *outsize = ascsize;
     }
-    return 0;
+    free(lib_info);
 }
 
 static
@@ -93,10 +69,10 @@ int aacenc_channel_mode(const pcm_sample_description_t *format)
 {
     uint32_t chanmask = format->channel_mask;
 
-    if (format->channels_per_frame > 6)
+    if (format->channels_per_frame > 8)
         return 0;
     if (!chanmask) {
-        static uint32_t defaults[] = { 0x4, 0x3, 0x7, 0, 0x37, 0x3f };
+        static uint32_t defaults[] = { 0x4, 0x3, 0x7, 0, 0x37, 0x3f, 0, 0x63f };
         chanmask = defaults[format->channels_per_frame - 1];
     }
     switch (chanmask) {
@@ -108,6 +84,10 @@ int aacenc_channel_mode(const pcm_sample_description_t *format)
     case 0x107: return MODE_1_2_1;
     case 0x607: return MODE_1_2_2;
     case 0x60f: return MODE_1_2_2_1;
+#if AACENCODER_LIB_VL0 > 3 || (AACENCODER_LIB_VL0==3 && AACENCODER_LIB_VL1>=4)
+    case 0xff:  return MODE_1_2_2_2_1;
+    case 0x63f: return MODE_7_1_REAR_SURROUND;
+#endif
     }
     return 0;
 }
@@ -118,8 +98,11 @@ int aacenc_init(HANDLE_AACENCODER *encoder, const aacenc_param_t *params,
 {
     int channel_mode;
     int aot;
+    LIB_INFO lib_info;
 
     *encoder = 0;
+    aacenc_get_lib_info(&lib_info);
+
     if ((channel_mode = aacenc_channel_mode(format)) == 0) {
         fprintf(stderr, "ERROR: unsupported channel layout\n");
         goto FAIL;
@@ -145,13 +128,21 @@ int aacenc_init(HANDLE_AACENCODER *encoder, const aacenc_param_t *params,
         fprintf(stderr, "ERROR: unsupported sample rate\n");
         goto FAIL;
     }
-    aacEncoder_SetParam(*encoder, AACENC_CHANNELMODE, channel_mode);
+    if (aacEncoder_SetParam(*encoder, AACENC_CHANNELMODE,
+                            channel_mode) != AACENC_OK) {
+        fprintf(stderr, "ERROR: unsupported channel mode\n");
+        goto FAIL;
+    }
     aacEncoder_SetParam(*encoder, AACENC_BANDWIDTH, params->bandwidth);
     aacEncoder_SetParam(*encoder, AACENC_CHANNELORDER, 1);
     aacEncoder_SetParam(*encoder, AACENC_AFTERBURNER, !!params->afterburner);
 
-    if (aot == AOT_ER_AAC_ELD && params->lowdelay_sbr)
-        aacEncoder_SetParam(*encoder, AACENC_SBR_MODE, 1);
+    aacEncoder_SetParam(*encoder, AACENC_SBR_MODE, params->lowdelay_sbr);
+
+#if AACENCODER_LIB_VL0 > 3 || (AACENCODER_LIB_VL0==3 && AACENCODER_LIB_VL1>=4)
+    if (lib_info.version > 0x03040000)
+        aacEncoder_SetParam(*encoder, AACENC_SBR_RATIO, params->sbr_ratio);
+#endif
 
     if (aacEncoder_SetParam(*encoder, AACENC_TRANSMUX,
                             params->transport_format) != AACENC_OK) {
@@ -187,7 +178,7 @@ FAIL:
 int aac_encode_frame(HANDLE_AACENCODER encoder,
                      const pcm_sample_description_t *format,
                      const int16_t *input, unsigned iframes,
-                     uint8_t **output, uint32_t *olen, uint32_t *osize)
+                     aacenc_frame_t *output)
 {
     uint32_t ilen = iframes * format->channels_per_frame;
     AACENC_BufDesc ibdesc = { 0 }, obdesc = { 0 };
@@ -205,12 +196,14 @@ int aac_encode_frame(HANDLE_AACENCODER encoder,
     unsigned channel_mode, obytes;
 
     channel_mode = aacEncoder_GetParam(encoder, AACENC_CHANNELMODE);
-    obytes = 6144 / 8 * channel_mode + 7;
-    if (!*output || *osize < obytes) {
-        *osize = obytes;
-        *output = realloc(*output, obytes);
+    obytes = 6144 / 8 * channel_mode;
+    if (!output->data || output->capacity < obytes) {
+        uint8_t *p = realloc(output->data, obytes);
+        if (!p) return -1;
+        output->capacity = obytes;
+        output->data = p;
     }
-    obufs[0] = *output;
+    obufs[0] = output->data;
     obuf_sizes[0] = obytes;
 
     iargs.numInSamples = ilen ? ilen : -1; /* -1 for signaling EOF */
@@ -230,6 +223,6 @@ int aac_encode_frame(HANDLE_AACENCODER encoder,
         fprintf(stderr, "ERROR: aacEncEncode() failed\n");
         return -1;
     }
-    *olen = oargs.numOutBytes;
-    return oargs.numInSamples;
+    output->size = oargs.numOutBytes;
+    return oargs.numInSamples / format->channels_per_frame;
 }

src/aacenc.h

@@ -15,6 +15,7 @@
     unsigned bandwidth; \
     unsigned afterburner; \
     unsigned lowdelay_sbr; \
+    unsigned sbr_ratio; \
     unsigned sbr_signaling; \
     unsigned transport_format; \
     unsigned adts_crc_check; \
@@ -24,11 +25,18 @@ typedef struct aacenc_param_t {
     AACENC_PARAMS
 } aacenc_param_t;
 
+typedef struct aacenc_frame_t {
+    uint8_t *data;
+    uint32_t size, capacity;
+} aacenc_frame_t;
+
+int aacenc_is_sbr_ratio_available();
+
 int aacenc_is_sbr_active(const aacenc_param_t *params);
 
-int aacenc_mp4asc(const aacenc_param_t *params,
-                  const uint8_t *asc, uint32_t ascsize,
-                  uint8_t *outasc, uint32_t *outsize);
+int aacenc_is_dual_rate_sbr(const aacenc_param_t *params);
+
+void aacenc_get_lib_info(LIB_INFO *info);
 
 int aacenc_init(HANDLE_AACENCODER *encoder, const aacenc_param_t *params,
                 const pcm_sample_description_t *format,
@@ -37,6 +45,6 @@ int aacenc_init(HANDLE_AACENCODER *encoder, const aacenc_param_t *params,
 int aac_encode_frame(HANDLE_AACENCODER encoder,
                      const pcm_sample_description_t *format,
                      const int16_t *input, unsigned iframes,
-                     uint8_t **output, uint32_t *olen, uint32_t *osize);
+                     aacenc_frame_t *output);
 
 #endif

src/caf_reader.c

@@ -193,12 +193,10 @@ int caf_parse(caf_reader_t *reader, int64_t *data_length)
         } else if (fcc == M4AF_FOURCC('d','a','t','a')) {
             TRY_IO(pcm_skip(&reader->io, 4)); /* mEditCount */
             *data_length = (chunk_size == ~0ULL) ? chunk_size : chunk_size - 4;
-            reader->data_offset += 12;
+            reader->data_offset = pcm_tell(&reader->io);
             break;
         } else
             TRY_IO(pcm_skip(&reader->io, chunk_size));
-
-        reader->data_offset += (chunk_size + 8);
     }
     ENSURE(reader->sample_format.channels_per_frame);
     ENSURE(fcc == M4AF_FOURCC('d','a','t','a'));
@@ -227,6 +225,7 @@ pcm_reader_t *caf_open(pcm_io_context_t *io,
     memcpy(&reader->io, io, sizeof(pcm_io_context_t));
     reader->tag_callback = tag_callback;
     reader->tag_ctx = tag_ctx;
+    memcpy(reader->chanmap, "\000\001\002\003\004\005\006\007", 8);
 
     if (caf_parse(reader, &data_length) < 0) {
         free(reader);

src/extrapolater.c

@@ -0,0 +1,208 @@
+/* 
+ * Copyright (C) 2013 nu774
+ * For conditions of distribution and use, see copyright notice in COPYING
+ */
+#if HAVE_CONFIG_H
+#  include "config.h"
+#endif
+#if HAVE_STDINT_H
+#  include <stdint.h>
+#endif
+#include <stdlib.h>
+#include <string.h>
+#include <assert.h>
+#include "pcm_reader.h"
+#include "lpc.h"
+
+typedef int16_t sample_t;
+
+typedef struct buffer_t {
+    sample_t *data;
+    unsigned count;    /* count in frames */
+    unsigned capacity; /* size in bytes   */
+} buffer_t;
+
+typedef struct extrapolater_t {
+    pcm_reader_vtbl_t *vtbl;
+    pcm_reader_t *src;
+    pcm_sample_description_t format;
+    buffer_t buffer[2];
+    unsigned nbuffer;
+    int (*process)(struct extrapolater_t *, void *, unsigned);
+} extrapolater_t;
+
+#define LPC_ORDER 32
+
+static inline pcm_reader_t *get_source(pcm_reader_t *reader)
+{
+    return ((extrapolater_t *)reader)->src;
+}
+
+static const
+pcm_sample_description_t *get_format(pcm_reader_t *reader)
+{
+    return pcm_get_format(get_source(reader));
+}
+
+static int64_t get_length(pcm_reader_t *reader)
+{
+    return pcm_get_length(get_source(reader));
+}
+
+static int64_t get_position(pcm_reader_t *reader)
+{
+    return pcm_get_position(get_source(reader));
+}
+
+static int realloc_buffer(buffer_t *bp, size_t size)
+{
+    if (bp->capacity < size) {
+        void *p = realloc(bp->data, size);
+        if (!p) return -1;
+        bp->data = p;
+        bp->capacity = size;
+    }
+    return 0;
+}
+
+static void reverse_buffer(sample_t *data, unsigned nframes, unsigned nchannels)
+{
+    unsigned i = 0, j = nchannels * (nframes - 1), n;
+
+    for (; i < j; i += nchannels, j -= nchannels) {
+        for (n = 0; n < nchannels; ++n) {
+            sample_t tmp = data[i + n];
+            data[i + n] = data[j + n];
+            data[j + n] = tmp;
+        }
+    }
+}
+
+static int fetch(extrapolater_t *self, unsigned nframes)
+{
+    const pcm_sample_description_t *sfmt = pcm_get_format(self->src);
+    buffer_t *bp = &self->buffer[self->nbuffer];
+    int rc = 0;
+
+    if (realloc_buffer(bp, nframes * sfmt->bytes_per_frame) == 0) {
+        rc = pcm_read_frames(self->src, bp->data, nframes);
+        bp->count = rc > 0 ? rc : 0;
+    }
+    if (rc > 0)
+        self->nbuffer ^= 1;
+    return bp->count;
+}
+
+static int extrapolate(extrapolater_t *self, const buffer_t *bp,
+                        void *dst, unsigned nframes)
+{
+    const pcm_sample_description_t *sfmt = pcm_get_format(self->src);
+    unsigned i, n = sfmt->channels_per_frame;
+    float lpc[LPC_ORDER];
+
+    for (i = 0; i < n; ++i) {
+        vorbis_lpc_from_data(bp->data + i, lpc, bp->count, LPC_ORDER, n);
+        vorbis_lpc_predict(lpc, &bp->data[i + n * (bp->count - LPC_ORDER)],
+                           LPC_ORDER, (sample_t*)dst + i, nframes, n);
+    }
+    return nframes;
+}
+
+static int process1(extrapolater_t *self, void *buffer, unsigned nframes);
+static int process2(extrapolater_t *self, void *buffer, unsigned nframes);
+static int process3(extrapolater_t *self, void *buffer, unsigned nframes);
+
+static int process0(extrapolater_t *self, void *buffer, unsigned nframes)
+{
+    const pcm_sample_description_t *sfmt = pcm_get_format(self->src);
+    unsigned nchannels = sfmt->channels_per_frame;
+    buffer_t *bp = &self->buffer[self->nbuffer];
+
+    if (fetch(self, nframes) < 2 * LPC_ORDER)
+        memset(buffer, 0, nframes * sfmt->bytes_per_frame);
+    else {
+        reverse_buffer(bp->data, bp->count, nchannels);
+        extrapolate(self, bp, buffer, nframes);
+        reverse_buffer(buffer, nframes, nchannels);
+        reverse_buffer(bp->data, bp->count, nchannels);
+    }
+    self->process = bp->count ? process1 : process2;
+    return nframes;
+}
+
+static int process1(extrapolater_t *self, void *buffer, unsigned nframes)
+{
+    const pcm_sample_description_t *sfmt = pcm_get_format(self->src);
+    buffer_t *bp = &self->buffer[self->nbuffer ^ 1];
+
+    assert(bp->count <= nframes);
+    memcpy(buffer, bp->data, bp->count * sfmt->bytes_per_frame); 
+    if (!fetch(self, nframes))
+        self->process = process2;
+    return bp->count;
+}
+
+static int process2(extrapolater_t *self, void *buffer, unsigned nframes)
+{
+    const pcm_sample_description_t *sfmt = pcm_get_format(self->src);
+    buffer_t *bp = &self->buffer[self->nbuffer];
+    buffer_t *bbp = &self->buffer[self->nbuffer ^ 1];
+
+    if (bp->count < 2 * LPC_ORDER) {
+        size_t total = bp->count + bbp->count;
+        if (bbp->count &&
+            realloc_buffer(bbp, total * sfmt->bytes_per_frame) == 0)
+        {
+            memcpy(bbp->data + bbp->count * sfmt->channels_per_frame,
+                   bp->data, bp->count * sfmt->bytes_per_frame);
+            bbp->count = total;
+            bp->count = 0;
+            bp = bbp;
+            self->nbuffer ^= 1;
+        }
+    }
+    self->process = process3;
+
+    if (bp->count >= 2 * LPC_ORDER)
+        extrapolate(self, bp, buffer, nframes);
+    else
+        memset(buffer, 0, nframes * sfmt->bytes_per_frame);
+    return nframes;
+}
+
+static int process3(extrapolater_t *self, void *buffer, unsigned nframes)
+{
+    return 0;
+}
+
+static int read_frames(pcm_reader_t *reader, void *buffer, unsigned nframes)
+{
+    extrapolater_t *self = (extrapolater_t *)reader;
+    return self->process(self, buffer, nframes);
+}
+
+static void teardown(pcm_reader_t **reader)
+{
+    extrapolater_t *self = (extrapolater_t *)*reader;
+    pcm_teardown(&self->src);
+    free(self->buffer[0].data);
+    free(self->buffer[1].data);
+    free(self);
+    *reader = 0;
+}
+
+static pcm_reader_vtbl_t my_vtable = {
+    get_format, get_length, get_position, read_frames, teardown
+};
+
+pcm_reader_t *extrapolater_open(pcm_reader_t *reader)
+{
+    extrapolater_t *self = 0;
+
+    if ((self = calloc(1, sizeof(extrapolater_t))) == 0)
+        return 0;
+    self->src = reader;
+    self->vtbl = &my_vtable;
+    self->process = process0;
+    return (pcm_reader_t *)self;
+}

src/lpc.c

@@ -0,0 +1,169 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+  function: LPC low level routines
+  last mod: $Id: lpc.c 16227 2009-07-08 06:58:46Z xiphmont $
+
+ ********************************************************************/
+
+/* Some of these routines (autocorrelator, LPC coefficient estimator)
+   are derived from code written by Jutta Degener and Carsten Bormann;
+   thus we include their copyright below.  The entirety of this file
+   is freely redistributable on the condition that both of these
+   copyright notices are preserved without modification.  */
+
+/* Preserved Copyright: *********************************************/
+
+/* Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
+Technische Universita"t Berlin
+
+Any use of this software is permitted provided that this notice is not
+removed and that neither the authors nor the Technische Universita"t
+Berlin are deemed to have made any representations as to the
+suitability of this software for any purpose nor are held responsible
+for any defects of this software. THERE IS ABSOLUTELY NO WARRANTY FOR
+THIS SOFTWARE.
+
+As a matter of courtesy, the authors request to be informed about uses
+this software has found, about bugs in this software, and about any
+improvements that may be of general interest.
+
+Berlin, 28.11.1994
+Jutta Degener
+Carsten Bormann
+
+*********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#if HAVE_STDINT_H
+# include <stdint.h>
+#endif
+
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+#include "lpc.h"
+#include "lpcm.h"
+
+/* Autocorrelation LPC coeff generation algorithm invented by
+   N. Levinson in 1947, modified by J. Durbin in 1959. */
+
+/* Input : n elements of time doamin data
+   Output: m lpc coefficients, excitation energy */
+
+float vorbis_lpc_from_data(short *data,float *lpci,int n,int m,int stride){
+  double *aut=malloc(sizeof(*aut)*(m+1));
+  double *lpc=malloc(sizeof(*lpc)*(m));
+  double error;
+  double epsilon;
+  int i,j;
+
+  /* autocorrelation, p+1 lag coefficients */
+  j=m+1;
+  while(j--){
+    double d=0; /* double needed for accumulator depth */
+    for(i=j;i<n;i++)d+=(double)data[i*stride]*data[(i-j)*stride]/1073741824.0;
+    aut[j]=d;
+  }
+
+  /* Generate lpc coefficients from autocorr values */
+
+  /* set our noise floor to about -100dB */
+  error=aut[0] * (1. + 1e-10);
+  epsilon=1e-9*aut[0]+1e-10;
+
+  for(i=0;i<m;i++){
+    double r= -aut[i+1];
+
+    if(error<epsilon){
+      memset(lpc+i,0,(m-i)*sizeof(*lpc));
+      goto done;
+    }
+
+    /* Sum up this iteration's reflection coefficient; note that in
+       Vorbis we don't save it.  If anyone wants to recycle this code
+       and needs reflection coefficients, save the results of 'r' from
+       each iteration. */
+
+    for(j=0;j<i;j++)r-=lpc[j]*aut[i-j];
+    r/=error;
+
+    /* Update LPC coefficients and total error */
+
+    lpc[i]=r;
+    for(j=0;j<i/2;j++){
+      double tmp=lpc[j];
+
+      lpc[j]+=r*lpc[i-1-j];
+      lpc[i-1-j]+=r*tmp;
+    }
+    if(i&1)lpc[j]+=lpc[j]*r;
+
+    error*=1.-r*r;
+
+  }
+
+ done:
+
+  /* slightly damp the filter */
+  {
+    double g = .99;
+    double damp = g;
+    for(j=0;j<m;j++){
+      lpc[j]*=damp;
+      damp*=g;
+    }
+  }
+
+  for(j=0;j<m;j++)lpci[j]=(float)lpc[j];
+
+  /* we need the error value to know how big an impulse to hit the
+     filter with later */
+
+  free(aut);
+  free(lpc);
+  return error;
+}
+
+void vorbis_lpc_predict(float *coeff,short *prime,int m,
+                        short *data,long n,int stride){
+
+  /* in: coeff[0...m-1] LPC coefficients
+         prime[0...m-1] initial values (allocated size of n+m-1)
+    out: data[0...n-1] data samples */
+
+  long i,j,o,p;
+  float y;
+  float *work=malloc(sizeof(*work)*(m+n));
+
+  if(!prime)
+    for(i=0;i<m;i++)
+      work[i]=0.f;
+  else
+    for(i=0;i<m;i++)
+      work[i]=prime[i*stride]/32768.0f;
+
+  for(i=0;i<n;i++){
+    y=0;
+    o=i;
+    p=m;
+    for(j=0;j<m;j++)
+      y-=work[o++]*coeff[--p];
+
+    work[o]=y;
+    data[i*stride]=lrint(pcm_clip(y*32768.0,-32768.0,32767.0));
+  }
+  free(work);
+}

src/lpc.h

@@ -0,0 +1,27 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+  function: LPC low level routines
+  last mod: $Id: lpc.h 16037 2009-05-26 21:10:58Z xiphmont $
+
+ ********************************************************************/
+
+#ifndef _V_LPC_H_
+#define _V_LPC_H_
+
+/* simple linear scale LPC code */
+extern float vorbis_lpc_from_data(short *data,float *lpc,int n,int m,int stride);
+
+extern void vorbis_lpc_predict(float *coeff,short *prime,int m,
+                               short *data,long n,int stride);
+
+#endif

src/lpcm.c

@@ -13,35 +13,6 @@
 #include "lpcm.h"
 #include "m4af_endian.h"
 
-#if defined(_MSC_VER) && _MSC_VER < 1800
-#  ifdef _M_IX86
-inline int lrint(double x)
-{
-    int n;
-    _asm {
-        fld x
-        fistp n
-    }
-    return n;
-}
-#  else
-#    include <emmintrin.h>
-inline int lrint(double x)
-{
-    return _mm_cvtsd_si32(_mm_load_sd(&x));
-}
-#  endif
-#endif
-
-static
-inline double pcm_clip(double n, double min_value, double max_value)
-{
-    if (n < min_value)
-        return min_value;
-    else if (n > max_value)
-        return max_value;
-    return n;
-}
 static
 inline float pcm_i2f(int32_t n)
 {

src/lpcm.h

@@ -31,6 +31,36 @@ typedef struct pcm_sample_description_t {
 #define PCM_BYTES_PER_CHANNEL(desc) \
     ((desc)->bytes_per_frame / (desc)->channels_per_frame)
 
+#if defined(_MSC_VER) && _MSC_VER < 1800
+#  ifdef _M_IX86
+static inline int lrint(double x)
+{
+    int n;
+    _asm {
+        fld x
+        fistp n
+    }
+    return n;
+}
+#  else
+#    include <emmintrin.h>
+static inline int lrint(double x)
+{
+    return _mm_cvtsd_si32(_mm_load_sd(&x));
+}
+#  endif
+#endif
+
+static
+inline double pcm_clip(double n, double min_value, double max_value)
+{
+    if (n < min_value)
+        return min_value;
+    else if (n > max_value)
+        return max_value;
+    return n;
+}
+
 int pcm_convert_to_native_sint16(const pcm_sample_description_t *format,
                                  const void *input, uint32_t nframes,
                                  int16_t *result);

src/m4af.c

@@ -864,7 +864,6 @@ void m4af_write_sbgp_box(m4af_ctx_t *ctx, uint32_t track_idx)
 static
 void m4af_write_sgpd_box(m4af_ctx_t *ctx, uint32_t track_idx)
 {
-    m4af_track_t *track = &ctx->track[track_idx];
     m4af_write(ctx,
                "\0\0\0\026"  /* size: 22           */
                "sgpd"        /* type               */
@@ -1050,7 +1049,6 @@ void m4af_write_elst_box(m4af_ctx_t *ctx, uint32_t track_idx)
 static
 void m4af_write_edts_box(m4af_ctx_t *ctx, uint32_t track_idx)
 {
-    m4af_track_t *track = &ctx->track[track_idx];
     int64_t pos = m4af_tell(ctx);
     m4af_write(ctx, "\0\0\0\0edts", 8);
     m4af_write_elst_box(ctx, track_idx);

src/main.c

@@ -54,7 +54,8 @@ static void handle_signals(void)
 {
     int i, sigs[] = { SIGINT, SIGHUP, SIGTERM };
     for (i = 0; i < sizeof(sigs)/sizeof(sigs[0]); ++i) {
-        struct sigaction sa = { 0 };
+        struct sigaction sa;
+        memset(&sa, 0, sizeof sa);
         sa.sa_handler = signal_handler;
         sa.sa_flags |= SA_RESTART;
         sigaction(sigs[i], &sa, 0);
@@ -132,7 +133,14 @@ PROGNAME " %s\n"
 " -a, --afterburner <n>         Afterburner\n"
 "                                 0: Off\n"
 "                                 1: On(default)\n"
-" -L, --lowdelay-sbr            Enable ELD-SBR (AAC ELD only)\n"
+" -L, --lowdelay-sbr <-1|0|1>   Configure SBR activity on AAC ELD\n"
+"                                -1: Use ELD SBR auto configurator\n"
+"                                 0: Disable SBR on ELD (default)\n"
+"                                 1: Enable SBR on ELD\n"
+" -s, --sbr-ratio <0|1|2>       Controls activation of downsampled SBR\n"
+"                                 0: Use lib default (default)\n"
+"                                 1: downsampled SBR (default for ELD+SBR)\n"
+"                                 2: dual-rate SBR (default for HE-AAC)\n"
 " -f, --transport-format <n>    Transport format\n"
 "                                 0: RAW (default, muxed into M4A)\n"
 "                                 1: ADIF\n"
@@ -228,7 +236,7 @@ static
 int parse_options(int argc, char **argv, aacenc_param_ex_t *params)
 {
     int ch;
-    unsigned n;
+    int n;
 
 #define OPT_INCLUDE_SBR_DELAY    M4AF_FOURCC('s','d','l','y')
 #define OPT_MOOV_BEFORE_MDAT     M4AF_FOURCC('m','o','o','v')
@@ -247,7 +255,8 @@ int parse_options(int argc, char **argv, aacenc_param_ex_t *params)
         { "bitrate-mode",     required_argument, 0, 'm' },
         { "bandwidth",        required_argument, 0, 'w' },
         { "afterburner",      required_argument, 0, 'a' },
-        { "lowdelay-sbr",     no_argument,       0, 'L' },
+        { "lowdelay-sbr",     required_argument, 0, 'L' },
+        { "sbr-ratio",        required_argument, 0, 's' },
         { "transport-format", required_argument, 0, 'f' },
         { "adts-crc-check",   no_argument,       0, 'C' },
         { "header-period",    required_argument, 0, 'P' },
@@ -325,7 +334,18 @@ int parse_options(int argc, char **argv, aacenc_param_ex_t *params)
             params->afterburner = n;
             break;
         case 'L':
-            params->lowdelay_sbr = 1;
+            if (sscanf(optarg, "%d", &n) != 1 || n < -1 || n > 1) {
+                fprintf(stderr, "invalid arg for lowdelay-sbr\n");
+                return -1;
+            }
+            params->lowdelay_sbr = n;
+            break;
+        case 's':
+            if (sscanf(optarg, "%u", &n) != 1 || n > 2) {
+                fprintf(stderr, "invalid arg for sbr-ratio\n");
+                return -1;
+            }
+            params->sbr_ratio = n;
             break;
         case 'f':
             if (sscanf(optarg, "%u", &n) != 1) {
@@ -472,16 +492,15 @@ int parse_options(int argc, char **argv, aacenc_param_ex_t *params)
 };
 
 static
-int write_sample(FILE *ofp, m4af_ctx_t *m4af,
-                 const void *data, uint32_t size, uint32_t duration)
+int write_sample(FILE *ofp, m4af_ctx_t *m4af, aacenc_frame_t *frame)
 {
     if (!m4af) {
-        fwrite(data, 1, size, ofp);
+        fwrite(frame->data, 1, frame->size, ofp);
         if (ferror(ofp)) {
             fprintf(stderr, "ERROR: fwrite(): %s\n", strerror(errno));
             return -1;
         }
-    } else if (m4af_write_sample(m4af, 0, data, size, duration) < 0) {
+    } else if (m4af_write_sample(m4af, 0, frame->data, frame->size, 0) < 0) {
         fprintf(stderr, "ERROR: failed to write m4a sample\n");
         return -1;
     }
@@ -489,51 +508,76 @@ int write_sample(FILE *ofp, m4af_ctx_t *m4af,
 }
 
 static
-int encode(pcm_reader_t *reader, HANDLE_AACENCODER encoder,
-           uint32_t frame_length, FILE *ofp, m4af_ctx_t *m4af,
-           int show_progress)
+int encode(aacenc_param_ex_t *params, pcm_reader_t *reader,
+           HANDLE_AACENCODER encoder, uint32_t frame_length, 
+           m4af_ctx_t *m4af)
 {
-    int16_t *ibuf = 0;
-    uint8_t *obuf = 0;
-    uint32_t olen;
-    uint32_t osize = 0;
+    int16_t *ibuf = 0, *ip;
+    aacenc_frame_t obuf[2] = {{ 0 }}, *obp;
+    unsigned flip = 0;
     int nread = 1;
-    int consumed;
     int rc = -1;
-    int frames_written = 0;
+    int remaining, consumed;
+    int frames_written = 0, encoded = 0;
     aacenc_progress_t progress = { 0 };
     const pcm_sample_description_t *fmt = pcm_get_format(reader);
 
     ibuf = malloc(frame_length * fmt->bytes_per_frame);
     aacenc_progress_init(&progress, pcm_get_length(reader), fmt->sample_rate);
-    do {
+
+    for (;;) {
+        /*
+         * Since we delay the write, we cannot just exit loop when interrupted.
+         * Instead, we regard it as EOF.
+         */
         if (g_interrupted)
             nread = 0;
-        else if (nread) {
+        if (nread > 0) {
             if ((nread = pcm_read_frames(reader, ibuf, frame_length)) < 0) {
                 fprintf(stderr, "ERROR: read failed\n");
                 goto END;
             }
-            if (show_progress)
+            if (!params->silent)
                 aacenc_progress_update(&progress, pcm_get_position(reader),
                                        fmt->sample_rate * 2);
         }
-        if ((consumed = aac_encode_frame(encoder, fmt, ibuf, nread,
-                                         &obuf, &olen, &osize)) < 0)
-            goto END;
-        if (olen > 0) {
-            if (write_sample(ofp, m4af, obuf, olen, frame_length) < 0)
+        ip = ibuf;
+        remaining = nread;
+        do {
+            obp = &obuf[flip];
+            consumed = aac_encode_frame(encoder, fmt, ip, remaining, obp);
+            if (consumed < 0) goto END;
+            if (consumed == 0 && obp->size == 0) goto DONE;
+            if (obp->size == 0) break;
+
+            remaining -= consumed;
+            ip += consumed * fmt->channels_per_frame;
+            flip ^= 1;
+            /*
+             * As we pad 1 frame at beginning and ending by our extrapolator,
+             * we want to drop them.
+             * We delay output by 1 frame by double buffering, and discard
+             * second frame and final frame from the encoder.
+             * Since sbr_header is included in the first frame (in case of
+             * SBR), we cannot discard first frame. So we pick second instead.
+             */
+            ++encoded;
+            if (encoded == 1 || encoded == 3)
+                continue;
+            obp = &obuf[flip];
+            if (write_sample(params->output_fp, m4af, obp) < 0)
                 goto END;
             ++frames_written;
-        }
-    } while (nread > 0 || olen > 0);
-
-    if (show_progress)
+        } while (remaining > 0);
+    }
+DONE:
+    if (!params->silent)
         aacenc_progress_finish(&progress, pcm_get_position(reader));
     rc = frames_written;
 END:
     if (ibuf) free(ibuf);
-    if (obuf) free(obuf);
+    if (obuf[0].data) free(obuf[0].data);
+    if (obuf[1].data) free(obuf[1].data);
     return rc;
 }
 
@@ -543,19 +587,11 @@ void put_tool_tag(m4af_ctx_t *m4af, const aacenc_param_ex_t *params,
 {
     char tool_info[256];
     char *p = tool_info;
-    LIB_INFO *lib_info = 0;
+    LIB_INFO lib_info;
 
     p += sprintf(p, PROGNAME " %s, ", fdkaac_version);
-
-    lib_info = calloc(FDK_MODULE_LAST, sizeof(LIB_INFO));
-    if (aacEncGetLibInfo(lib_info) == AACENC_OK) {
-        int i;
-        for (i = 0; i < FDK_MODULE_LAST; ++i)
-            if (lib_info[i].module_id == FDK_AACENC)
-                break;
-        p += sprintf(p, "libfdk-aac %s, ", lib_info[i].versionStr);
-    }
-    free(lib_info);
+    aacenc_get_lib_info(&lib_info);
+    p += sprintf(p, "libfdk-aac %s, ", lib_info.versionStr);
     if (params->bitrate_mode)
         sprintf(p, "VBR mode %d", params->bitrate_mode);
     else
@@ -607,7 +643,7 @@ char *generate_output_filename(const char *filename, const char *ext)
         const char *ext_org = strrchr(base, '.');
         if (ext_org) ilen = ext_org - base;
         p = malloc(ilen + ext_len + 1);
-        sprintf(p, "%.*s%s", ilen, base, ext);
+        sprintf(p, "%.*s%s", (int)ilen, base, ext);
     }
     return p;
 }
@@ -651,7 +687,7 @@ int parse_raw_spec(const char *spec, pcm_sample_description_t *desc)
 static pcm_io_vtbl_t pcm_io_vtbl = {
     read_callback, seek_callback, tell_callback
 };
-static pcm_io_vtbl_t pcm_io_vtbl_noseek = { read_callback, 0, 0 };
+static pcm_io_vtbl_t pcm_io_vtbl_noseek = { read_callback, 0, tell_callback };
 
 static
 pcm_reader_t *open_input(aacenc_param_ex_t *params)
@@ -709,10 +745,13 @@ pcm_reader_t *open_input(aacenc_param_ex_t *params)
             }
             break;
         default:
+            fprintf(stderr, "ERROR: unsupported input file\n");
             goto END;
         }
     }
-    return pcm_open_sint16_converter(reader);
+    if ((reader = pcm_open_sint16_converter(reader)) != 0)
+        reader = extrapolater_open(reader);
+    return reader;
 END:
     return 0;
 }
@@ -731,9 +770,9 @@ int main(int argc, char **argv)
     AACENC_InfoStruct aacinfo = { 0 };
     m4af_ctx_t *m4af = 0;
     const pcm_sample_description_t *sample_format;
-    int downsampled_timescale = 0;
     int frame_count = 0;
     int sbr_mode = 0;
+    unsigned scale_shift = 0;
 
     setlocale(LC_CTYPE, "");
     setbuf(stderr, 0);
@@ -746,16 +785,18 @@ int main(int argc, char **argv)
 
     sample_format = pcm_get_format(reader);
 
-    /*
-     * We use explicit/hierarchical signaling for LOAS.
-     * Other than that, we request implicit signaling to FDK library, then 
-     * append explicit/backward-compatible signaling to ASC in case of MP4FF.
-     *
-     * Explicit/backward-compatible signaling of SBR is the most recommended
-     * way in MPEG4 part3 spec, and seems the only way supported by iTunes.
-     * Since FDK library does not support it, we have to do it on our side.
-     */
-    params.sbr_signaling = (params.transport_format == TT_MP4_LOAS) ? 2 : 0;
+    sbr_mode = aacenc_is_sbr_active((aacenc_param_t*)&params);
+    if (sbr_mode && !aacenc_is_sbr_ratio_available()) {
+        fprintf(stderr, "WARNING: Only dual-rate SBR is available "
+                        "for this version\n");
+        params.sbr_ratio = 2;
+    }
+    scale_shift = aacenc_is_dual_rate_sbr((aacenc_param_t*)&params);
+    params.sbr_signaling =
+        (params.transport_format == TT_MP4_LOAS) ? 2 :
+        (params.transport_format == TT_MP4_RAW)  ? 1 : 0;
+    if (sbr_mode && !scale_shift)
+        params.sbr_signaling = 2;
 
     if (aacenc_init(&encoder, (aacenc_param_t*)&params, sample_format,
                     &aacinfo) < 0)
@@ -773,29 +814,33 @@ int main(int argc, char **argv)
         goto END;
     }
     handle_signals();
-    sbr_mode = aacenc_is_sbr_active((aacenc_param_t*)&params);
+
     if (!params.transport_format) {
         uint32_t scale;
-        uint8_t mp4asc[32];
-        uint32_t ascsize = sizeof(mp4asc);
         unsigned framelen = aacinfo.frameLength;
-        if (sbr_mode)
-            downsampled_timescale = 1;
-        scale = sample_format->sample_rate >> downsampled_timescale;
+        scale = sample_format->sample_rate >> scale_shift;
         if ((m4af = m4af_create(M4AF_CODEC_MP4A, scale, &m4af_io,
                                 params.output_fp)) < 0)
             goto END;
-        aacenc_mp4asc((aacenc_param_t*)&params, aacinfo.confBuf,
-                      aacinfo.confSize, mp4asc, &ascsize);
-        m4af_set_decoder_specific_info(m4af, 0, mp4asc, ascsize);
+        m4af_set_decoder_specific_info(m4af, 0,
+                                       aacinfo.confBuf, aacinfo.confSize);
         m4af_set_fixed_frame_duration(m4af, 0,
-                                      framelen >> downsampled_timescale);
+                                      framelen >> scale_shift);
         m4af_set_vbr_mode(m4af, 0, params.bitrate_mode);
         m4af_set_priming_mode(m4af, params.gapless_mode + 1);
         m4af_begin_write(m4af);
     }
-    frame_count = encode(reader, encoder, aacinfo.frameLength,
-                         params.output_fp, m4af, !params.silent);
+    if (scale_shift && (aacinfo.encoderDelay & 1)) {
+        /*
+         * Since odd delay cannot be exactly expressed in downsampled scale,
+         * we push one zero frame to the encoder here, to make delay even
+         */
+        int16_t zero[8] = { 0 };
+        aacenc_frame_t frame = { 0 };
+        aac_encode_frame(encoder, sample_format, zero, 1, &frame);
+        free(frame.data);
+    }
+    frame_count = encode(&params, reader, encoder, aacinfo.frameLength, m4af);
     if (frame_count < 0)
         goto END;
     if (m4af) {
@@ -805,12 +850,11 @@ int main(int argc, char **argv)
 
         if (sbr_mode && params.profile != AOT_ER_AAC_ELD &&
             !params.include_sbr_delay)
-            delay -= 481 << 1;
-        if (sbr_mode && (delay & 1))
+            delay -= 481 << scale_shift;
+        if (scale_shift && (delay & 1))
             ++delay;
         padding = frame_count * aacinfo.frameLength - frames_read - delay;
-        m4af_set_priming(m4af, 0, delay >> downsampled_timescale,
-                         padding >> downsampled_timescale);
+        m4af_set_priming(m4af, 0, delay >> scale_shift, padding >> scale_shift);
         if (finalize_m4a(m4af, &params, encoder) < 0)
             goto END;
     }

src/metadata.c

@@ -1,3 +1,7 @@
+/* 
+ * Copyright (C) 2013 nu774
+ * For conditions of distribution and use, see copyright notice in COPYING
+ */
 #if HAVE_CONFIG_H
 #  include "config.h"
 #endif

src/metadata.h

@@ -1,3 +1,7 @@
+/* 
+ * Copyright (C) 2013 nu774
+ * For conditions of distribution and use, see copyright notice in COPYING
+ */
 #ifndef METADATA_H
 #define METADATA_H
 

src/pcm_reader.h

@@ -1,3 +1,7 @@
+/* 
+ * Copyright (C) 2013 nu774
+ * For conditions of distribution and use, see copyright notice in COPYING
+ */
 #ifndef PCM_READER_H
 #define PCM_READER_H
 
@@ -85,7 +89,7 @@ uint32_t bitcount(uint32_t bits)
 int pcm_read(pcm_io_context_t *io, void *buffer, uint32_t size);
 int pcm_skip(pcm_io_context_t *io, int64_t count);
 
-static int pcm_seek(pcm_io_context_t *io, int64_t off, int whence)
+static inline int pcm_seek(pcm_io_context_t *io, int64_t off, int whence)
 {
     return io->vtbl->seek ? io->vtbl->seek(io->cookie, off, whence) : -1;
 }
@@ -109,4 +113,6 @@ int apple_chan_chunk(pcm_io_context_t *io, uint32_t chunk_size,
 
 pcm_reader_t *pcm_open_sint16_converter(pcm_reader_t *reader);
 
+pcm_reader_t *extrapolater_open(pcm_reader_t *reader);
+
 #endif

src/pcm_readhelper.c

@@ -204,7 +204,7 @@ int apple_chan_chunk(pcm_io_context_t *io, uint32_t chunk_size,
 
     switch (mChannelLayoutTag) {
     case kAudioChannelLayoutTag_UseChannelBitmap:
-        ENSURE(bitcount(mask) == nchannels);
+        ENSURE(bitcount(mChannelBitmap) == nchannels);
         TRY_IO(pcm_skip(io, chunk_size - 12));
         fmt->channel_mask = mChannelBitmap;
         for (i = 0; i < nchannels; ++i)

src/pcm_sint16_converter.c

@@ -1,3 +1,7 @@
+/* 
+ * Copyright (C) 2013 nu774
+ * For conditions of distribution and use, see copyright notice in COPYING
+ */
 #if HAVE_CONFIG_H
 #  include "config.h"
 #endif

src/wav_reader.c

@@ -93,7 +93,6 @@ int riff_ds64(wav_reader_t *reader, int64_t *length)
     TRY_IO(pcm_scanl(&reader->io, "QQQL",
                      &riff_size, length, &sample_count, &table_size) != 4);
     TRY_IO(pcm_skip(&reader->io, (chunk_size - 27) & ~1));
-    reader->data_offset += (chunk_size + 9) & ~1;
 FAIL:
     return -1;
 }
@@ -163,7 +162,6 @@ int wav_parse(wav_reader_t *reader, int64_t *data_length)
            container == RIFF_FOURCC('R','F','6','4'));
     TRY_IO(pcm_read32le(&reader->io, &fcc));
     ENSURE(fcc == RIFF_FOURCC('W','A','V','E'));
-    reader->data_offset = 12;
 
     if (container == RIFF_FOURCC('R','F','6','4'))
         riff_ds64(reader, data_length);
@@ -174,12 +172,11 @@ int wav_parse(wav_reader_t *reader, int64_t *data_length)
         } else if (fcc == RIFF_FOURCC('d','a','t','a')) {
             if (container == RIFF_FOURCC('R','I','F','F'))
                 *data_length = chunk_size;
-            reader->data_offset += 8;
+            reader->data_offset = pcm_tell(&reader->io);
             break;
         } else {
             TRY_IO(pcm_skip(&reader->io, (chunk_size + 1) & ~1));
         }
-        reader->data_offset += (chunk_size + 9) & ~1;
     }
     if (fcc == RIFF_FOURCC('d','a','t','a'))
         return 0;

version.h

@@ -1,4 +1,4 @@
 #ifndef VERSION_H
 #define VERSION_H
-const char *fdkaac_version = "0.4.1";
+const char *fdkaac_version = "0.5.3";
 #endif