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27th April 2010, 20:19 | #1 | Link | |
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qaac - a new command line AAC/ALAC encoder using QuickTime API
for who is interested...
http://sites.google.com/site/qaacpage/home see also https://forum.doom9.org/showthread.p...01#post1925401 Quote:
Code:
qaac 0.17 Usage: qaac [options] infiles.... "-" as infile means stdin. In ADTS output mode, "-" as outfile means stdout. Main options: -d <dirname> Output directory, default is cwd -a, --abr <bitrate> AAC ABR mode / bitrate -V, --tvbr <n> AAC True VBR mode / quality [0-127] -v, --cvbr <bitrate> AAC Constrained VBR mode / bitrate -c, --cbr <bitrate> AAC CBR mode / bitrate --he HE AAC mode (Can't use TVBR) -A, --alac ALAC encoding mode -q, --quality <n> AAC encoding Quality [0-2] -r, --rate <option> Sample rate option (AAC only) Specify one of the followings: keep: Try to preserve the original rate auto: Let QuickTime choose the optimal one <number>: Literal rate in Hz -s, --silent Don't be verbose -n, --nice Give lower process priority --downmix <mono|stereo> Downmix to mono/stereo --no-optimize Don't optimize MP4 container file after encoding --adts ADTS(raw AAC)output, instead of m4a(AAC only) --ignorelength Assume WAV input and ignore the data chunk length -R, --raw Raw PCM input -S, --stat Save bitrate statistics into file Code:
Options for single input mode only: -o <filename> Output filename Options for Raw PCM input only: --raw-channels <n> Number of channels, default 2 --raw-rate <n> Sample rate, default 44100 --raw-format <str> Sample format, default S16L Sample format spec: 1st char: S(igned) | U(nsigned) | F(loat) 2nd part: Bitwidth Last part: L(ittle Endian) | B(ig Endian) Cases are ignored. u8b is OK. Code:
Tagging options(single input only): --title <string> --artist <string> --band <string> --album <string> --grouping <string> --composer <string> --comment <string> --genre <string> --date <string> --track <number[/total]> --disk <number[/total]> --compilation N.B. I am not the developer of this tool! _
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28th April 2010, 17:30 | #3 | Link |
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apple aac encoding engine...
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10th May 2010, 18:40 | #5 | Link | |
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new version...
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11th May 2010, 20:09 | #6 | Link | |
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new version...
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12th May 2010, 19:19 | #7 | Link | |
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fix bug...
Quote:
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21st May 2010, 19:42 | #8 | Link | |
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bug fix...
Quote:
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22nd May 2010, 19:15 | #9 | Link | |
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new bug fix...
Quote:
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24th May 2010, 17:35 | #12 | Link |
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and the aac (mpeg4 adts) output to stdout...
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29th May 2010, 13:59 | #14 | Link |
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What... Nobody has made a GUI yet!
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30th May 2010, 00:04 | #15 | Link |
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Have a question, hoping someone can help.
I am trying to use this for my subsonic server so I can stream via aac instead of mp3 to my mobile device. [5/29/10 6:51:18 PM EDT] DEBUG TranscodeInputStream Starting transcoder: [c:\subsonic\transcode\ffmpeg] [-i] [G:\Music\iTunes\Arctic Monkeys\Arctic Monkeys EP\03 Fake Tales Of San Francisco.mp3] [-f] [wav] [-] [5/29/10 6:51:18 PM EDT] DEBUG TranscodeInputStream Starting transcoder: [c:\subsonic\transcode\qaac] [-a] [128] [--adts] [-] [-] [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) initializing QTML...done [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) QuickTime 7.6.6 [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) Method: Average Bit Rate [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) Bitrate: 128 [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) Quality: Best [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) 0/0 samples processed [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) 0/0 samples processed [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) Overall bitrate: 2.06719kbps [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) [5/29/10 6:51:19 PM EDT] DEBUG InputStreamReaderThread (c:\subsonic\transcode\qaac) File contains data in an unknown format. So as shown above, I decoding the mp3 with this command ffmpeg -i %s -f wav - where %s is the file name, and decoding to stdin. And then trying to encode to stout with qaac -a %b --adts - - where %b is the bitrate. %s and %b get filled in automatically by subsonic as shown above. Even if I try: C:\subsonic\transcode>ffmpeg -i "G:\Music\iTunes\Arctic Monkeys\Arctic Monkeys EP\03 Fake Tales Of San Francisco.mp3" -f wav - | qaac -a 128 -o out.mp4 - FFmpeg veinritsiailiozinng QSTMVL.N..-r21231-Sherpya, Copyright (c) 2000-2010 F abrice Bellard, et al. built on Jan 16 2010 05:42:31 with gcc 4.2.5 20080919 (prerelease) [Sherpya] libavutil 50. 7. 0 / 50. 7. 0 libavcodec 52.47. 0 / 52.47. 0 libavformat 52.47. 0 / 52.47. 0 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.15. 0 / 1.15. 0 libswscale 0. 8. 0 / 0. 8. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mp3 @ 015ebec0]max_analyze_duration reached [mp3 @ 015ebec0]Estimating duration from bitrate, this may be inaccurate Input #0, mp3, from 'G:\Music\iTunes\Arctic Monkeys\Arctic Monkeys EP\03 Fake Ta les Of San Francisco.mp3': Metadata: TPE1 : Arctic Monkeys TALB : Arctic Monkeys EP TPE2 : Arctic Monkeys TIT2 : 03 Fake Tales Of San Francisco TPUB : Domino/Ada TYER : 2005 TDRC : 2005 Duration: 00:02:58.17, start: 0.000000, bitrate: 128 kb/s Stream #0.0: Audio: mp3, 44100 Hz, 2 channels, s16, 128 kb/s Output #0, wav, to 'pipe:': Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding [mp3 @ 02b88260]overread, skip -6 enddists: -4 -4 done QuickTime 7.6.6 <stdin> Method: Average Bit Rate Bitrate: 128 Quality: Best 0/0 samples processed Overall bitrate: 2.06719kbps av_interleaved_write_frame(): Error while opening file |
30th May 2010, 16:58 | #16 | Link |
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this is working for me:
Code:
ffmpeg -i input.mp3 -acodec pcm_f32le -f wav - | sox -t wav --ignore-length - -t wav - | qaac -s -a 128 -o output.mp4 - _
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30th May 2010, 17:59 | #17 | Link |
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Interesting, because this fails for me:
c:\subsonic\transcode\ffmpeg -i "G:\Music\iTunes\Ace Of Base\The Bridge\01 Beautiful Life.mp3" -acodec pcm_f32le -f wav - | c:\subsonic\transcode\sox -t wav --ignore-length - -t wav - | qaac -s -a 128 --adts E:\tempACC.aac - Duration: 00:03:39.27, start: 0.000000, bitrate: 217 kb/s Stream #0.0: Audio: mp3, 44100 Hz, 2 channels, s16, 32 kb/s c:\subsonic\transcode\sox WARN wav: wave header missing FmtExt chunk c:\subsonic\transcode\sox WARN wav: Length in output .wav header will be wrong s ince can't seek to fix it Output #0, wav, to 'pipe:': Stream #0.0: Audio: pcm_f32le, 44100 Hz, 2 channels, flt, 2822 kb/s Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding [mp3 @ 02c48260]Header missing Error while decoding stream #0.0 E:\tempACC.aac: The system cannot find the file specified. c:\subsonic\transcode\sox FAIL sox: `-' error writing output file: Broken pipe av_interleaved_write_frame(): Error while opening file but this works: C:\subsonic\transcode>c:\subsonic\transcode\ffmpeg -i "G:\Music\iTunes\Ace Of Ba se\The Bridge\01 Beautiful Life.mp3" -acodec pcm_f32le -f wav - | c:\subsonic\tr anscode\sox -t wav --ignore-length - -t wav - | qaac -s -a 128 -o E:\tempAAC.m4a - Duration: 00:03:39.27, start: 0.000000, bitrate: 217 kb/s Stream #0.0: Audio: mp3, 44100 Hz, 2 channels, s16, 32 kb/s Output #0, wav, to 'pipe:': c:\subsonic\transcode\sox WARN Stream #0.0wav: : Audio: pcm_f32le, 44100 Hz, 2 channels, flt, 2822 kb/swave header missing FmtExt chunk Sc:\subsonic\transcode\sox WARN twav: rLength in output .wav header will be wron g since can't seek to fix ite am mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding [mp3 @ 01478260]Header missing Error while decoding stream #0.0 Error while decoding stream #0.0ate=2822.4kbits/s size= 75474kB time=219.06 bitrate=2822.4kbits/s video:0kB audio:75474kB global headers:0kB muxing overhead 0.000088% So maybe its the --adts mode thats not working for me? |
30th May 2010, 18:18 | #18 | Link |
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you forgot "-o"...line should be:
Code:
c:\subsonic\transcode\ffmpeg -i "G:\Music\iTunes\Ace Of Base\The Bridge\01 Beautiful Life.mp3" -acodec pcm_f32le -f wav - | c:\subsonic\transcode\sox -t wav --ignore-length - -t wav - | qaac -s -a 128 --adts -o E:\tempACC.aac -
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30th May 2010, 20:04 | #19 | Link |
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You are right, this seems to work:
c:\subsonic\transcode\ffmpeg -i "G:\Music\iTunes\Ace Of Base\ The Bridge\01 Beautiful Life.mp3" -acodec pcm_f32le -f wav - | c:\subsonic\trans code\sox -t wav --ignore-length - -t wav - | c:\subsonic\transcode\qaac -s -a 12 8 --adts -o - - It works on the command line, but no subsonic, but atleast I know qaac is working. Thanks a lot |
4th June 2010, 18:39 | #20 | Link | |
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new version...
Quote:
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