[Libav-user] wav VS raw

[ReSearchIT Eng]

Hi Eugen,

Thanks for the quick reply.
For case 1(wav), I tried now again removing the parameters as suggested.
Removing them does not make any change (see below).
Note: I came to the long set of parameters, in order to have the two
commands as close as possible, to leave no "doubts".

*Case 1: (WAV2AWB)*
*Command*: ./ffmpeg.exe -loglevel debug -y -vn -i 684.wav -ac 1 -ab 23850
-ar 16000 -f amr -acodec libvo_amrwbenc 640bytes_fromwav_less_details.awb)
*Outputs*:
Guessed Channel Layout for  Input Stream #0.0 : mono
Input #0, wav, from '684.wav':
  Duration: 00:00:00.02, bitrate: 256 kb/s
    Stream #0:0, 1, 1/16000: Audio: pcm_s16le *([1][0][0][0] / 0x0001)*,
16000 Hz, mono, s16, 256 kb/s
Output #0, amr, to '640bytes_fromwav_less_details.awb':
  Metadata:
    encoder         : Lavf55.34.101
    Stream #0:0, 0, 1/90000: Audio: amr_wb (libvo_amrwbenc), 16000 Hz,
mono, s16, 23 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le -> libvo_amrwbenc)


*Case 2: (RAW2AWB:)*
*Command*: ./ffmpeg.exe -loglevel debug -y -vn -f s16le -ac 1 -ar 16000
-acodec pcm_s16le -sample_fmt s16 -i 684.wav -ac 1 -ab 23850 -ar 16000 -f
amr -acodec libvo_amrwbenc -sample_fmt s16 640bytes_fromwav.awb
*Outputs:*
Input #0, s16le, from '640.raw':
  Duration: 00:00:00.02, bitrate: 256 kb/s
    Stream #0:0, 1, 1/16000: Audio: pcm_s16le, 16000 Hz, mono, s16, 256 kb/s
Output #0, amr, to '640bytes_fromrRaw_outs16.awb':
  Metadata:
    encoder         : Lavf55.34.101
    Stream #0:0, 0, 1/90000: Audio: amr_wb (libvo_amrwbenc), 16000 Hz,
mono, s16, 23 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le -> libvo_amrwbenc)

Still, but when I compare the binary output is different.

The only difference I noticed in the details on the screen is:
Audio: pcm_s16le *([1][0][0][0] / 0x0001) *
vs
Audio: pcm_s16le

Is this difference something important or it simply states that it uses one
stream/channel?

Thanks,
Sebastian


On Tue, Apr 8, 2014 at 3:47 PM, Carl Eugen Hoyos <cehoyos at ag.or.at> wrote:

> ReSearchIT Eng <researchiteng at ...> writes:
>
> > ./ffmpeg.exe -loglevel debug -y -vn -f s16le -ac 1
> > -ar 16000 -acodec pcm_s16le -sample_fmt s16 -i 684.wav
>
> Your questions are difficult to understand...
>
> Lets start with the simple things:
> Please remove everything starting with "-f" until "s16":
> If "ffmpeg -i 684.wav" does not work as expected, please
> report back / try to set some values that you seem to believe
> are necessary (but shouldn't be).
> Once that works, try to understand what "-f" does and why you
> should only use it for raw pcm files (in this case).
>
> (I wonder what -sample_fmt means: I test often and I believe
> I never used it...)
>
> Carl Eugen
>
> _______________________________________________
> Libav-user mailing list
> Libav-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/libav-user
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://ffmpeg.org/pipermail/libav-user/attachments/20140408/6961c719/attachment.html>