Help with afconvert in terminal

B4tm4n

Registered
I'm trying to use afconvert to convert an ac3 file in to an aac file.
And I'm getting no where here is how far I've got.

Code:
MacBook:~ B4tm4n$ afconvert -v -f mp4f -d aac -b 128000 /Users/B4tm4n/Desktop/\[mkvextract\]/audio2.ac3 
Input file: audio2.ac3, 0 frames
Error: Couldn't set client format (-50)

What am I doing wrong, I just can't figure it out.

Here is the usage info from terminal.

Code:
Usage:
afconvert [option...] input_file [output_file]

Options: (may appear before or after arguments)
    { -f | --file } file_format:
        'adts' = AAC ADTS (.aac, .adts)
                   data_formats: 'aac ' 
        'ac-3' = AC3 (.ac3)
                   data_formats: 'ac-3' 
        'AIFC' = AIFC (.aif, .aiff, .aifc)
                   data_formats: BEI8 BEI16 BEI24 BEI32 BEF32 
                                 BEF64 'ulaw' 'alaw' 'MAC3' 'MAC6' 'ima4' 
                                 'QDMC' 'QDM2' 'Qclp' 'agsm' 
        'AIFF' = AIFF (.aif, .aiff)
                   data_formats: BEI8 BEI16 BEI24 BEI32 
        'caff' = Apple CAF File (.caf)
                   data_formats: '.mp1' '.mp2' '.mp3' 'FlAd' 'MAC3' 
                                 'MAC6' 'PesA' 'QDM2' 'QDMC' 'Qclp' 'Qclq' 
                                 'TS\x00\x02' 'TS\x00\x06' 'TS\x00\x07' 'TS\x00\x11' 'TS\x00E' 'TS\x00U' 
                                 'WMA1' 'WMA2' 'WMA3' 'XiVs' 'aac ' 'ac-3' 
                                 'agsm' 'alac' 'alaw' 'drms' 'dvca' 'dvi ' 
                                 'ima4' 'lpc ' BEI8 BEI16 BEI24 BEI32 
                                 BEF32 BEF64 LEI16 LEI24 LEI32 LEF32 
                                 LEF64 'ms\x00\x02' 'ms\x00\x11' 'ms\x001' 'ms\x00U' 'ms\x01`' 
                                 'ms\x01a' 'ms \x00' 'samr' 'ulaw' 'vdva' 
        'MPG3' = MPEG Layer 3 (.mp3, .mpeg)
                   data_formats: '.mp3' 
        'mp4f' = MPEG4 Audio (.mp4)
                   data_formats: 'aac ' 
        'm4af' = MPEG4 Audio (.m4a)
                   data_formats: 'aac ' 'alac' 
        'NeXT' = NeXT/Sun (.snd, .au)
                   data_formats: BEI8 BEI16 BEI24 BEI32 BEF32 
                                 BEF64 'ulaw' 
        'Sd2f' = Sound Designer II (.sd2)
                   data_formats: BEI8 BEI16 BEI24 BEI32 
        'WAVE' = WAVE (.wav)
                   data_formats: LEUI8 LEI16 LEI24 LEI32 LEF32 
                                 LEF64 'ulaw' 'alaw' 
    { -d | --data } data_format[@sample_rate_hz][/format_flags][#frames_per_packet] :
        [-][BE|LE]{F|[U]I}{8|16|24|32|64}          (PCM)
            e.g.   BEI16   F32@44100
        or a data format appropriate to file format, as above
        format_flags: hex digits, e.g. '80'
        bitdepth on non-PCM formats can be specified, e.g.: alac-24
        Frames per packet can be specified for some encoders, e.g.: samr#12
    { -c | --channels } number_of_channels
        add/remove channels without regard to order
    { -l | --channellayout } layout_tag
        layout_tag: name of a constant from CoreAudioTypes.h
          (prefix "kAudioChannelLayoutTag_" may be omitted)
        if specified once, applies to output file; if twice, the first
          applies to the input file, the second to the output file
    { -b | --bitrate } bit_rate_bps
         e.g. 128000
    { -q | --quality } codec_quality
        codec_quality: 0-127
    { -r | --src-quality } src_quality
        src_quality (sample rate converter quality): 0-127
    { -v | --verbose }
        print progress verbosely
    { -s | --strategy } strategy
        bitrate strategy for encoded file
        0 for CBR, 1 for ABR, 2 for VBR
    { -t | --tag }
        If encoding to CAF, store the source file's format and name in a user chunk.
        If decoding from CAF, use the destination format and filename found in a user chunk.
    --prime-method method
        decode priming method (see AudioConverter.h)

If anyone can help me find the right order of commands I would be greatly thankfull.

Cheers.
 
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