# Help with afconvert in terminal



## B4tm4n (Nov 28, 2007)

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.


```
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.


```
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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