[FFmpeg-devel] [RFC 00/10] AV_SAMPLE_FMT_DSD
pross at xvid.org
Wed May 7 10:59:06 CEST 2014
On Wed, May 07, 2014 at 01:40:51PM +0530, anshul wrote:
> On 05/07/2014 01:08 PM, Peter Ross wrote:
> >On Tue, May 06, 2014 at 08:38:21PM +0530, Anshul wrote:
> >>Peter Ross <pross at xvid.org> wrote:
> >>Is dop also related to default behaviour of audio transcoding ?
> >Can you restate/rephrase the question?
> >DoP is a method of transporting DSD (one-bit audio) samples over PCM.
> >This is done expressly for the purpose of feeding an external DSD DAC.
> >./ffmpeg -i sample.dff -acodec dop_s24le out.wav
> >then play back out.wav file to dac
> >-- Peter
> >(A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)
> >ffmpeg-devel mailing list
> >ffmpeg-devel at ffmpeg.org
> If i have aac audio with pcm raw data inside, if i transcode from
> aac to mp3 then in mp3 there will be pcm or
> this dsd sample format.
> I dont know lot about dsd, i am worried about my audio's pcm.
> (default behaviour of FFmpeg)
> I use below command
> ffmpeg -i some.aac out.mp3
* the default behaviour will not change *
to use DoP encoding, you must deliberately request it using '-acodec dop_24le'
(A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)
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