What is the --mode= command for R3 high quality aac streams please

[bat guano]

----------------------------------------
> Subject: Re: Re: What is the --mode= command for R3 high quality aac streams please
> From: richard at richsim900.plus.com
> To: get_iplayer at lists.infradead.org
> Date: Thu, 16 Jun 2011 14:32:50 +0100
>
> bat guano wrote:
> >
> >
> > ----------------------------------------
> >> Date: Sat, 14 May 2011 09:42:50 +0100
> >> From: get_iplayer at cjnash.com
> >> To: batguano999 at hotmail.com
> >> CC: roadcone at gmx.com; get_iplayer at lists.infradead.org
> >> Subject: Re: What is the --mode= command for R3 high quality aac
> >> streams please
> >>
> >> bat guano wrote:
> >>>
> >>> ----------------------------------------
> >>>> Date: Fri, 13 May 2011 22:07:30 +0100
> >>>> From: clivebuc at gmail.com
> >>>> To: get_iplayer at lists.infradead.org
> >>>> Subject: Re: What is the --mode= command for R3 high quality aac
> >>>> streams please
> >>>>
> >>>>
> >>>>
> >>>> On 13/05/2011 21:44, bat guano wrote:
> >>>>>
> >>>>> ----------------------------------------
> >>>>>> Date: Fri, 13 May 2011 20:25:56 +0100
> >>>>>> From: clivebuc at gmail.com
> >>>>>> To: get_iplayer at lists.infradead.org
> >>>>>> Subject: What is the --mode= command for R3 high quality aac
> >>>>>> streams please
> >>>>>>
> >>>>>> Hello,
> >>>>>>
> >>>>>> It seems that R3 is broadcasting 320K aac streams for their
> >>>>>> evening live
> >>>>>> concerts. Can anyone guide me as to the correct --mode=?? switch
> >>>>>> to
> >>>>>> secure those streams please?
> >>>>>>
> >>>>>> Thank you.
> >>>
> >>>>>> Clive
> >>>>> Hi
> >>>>> It was mentioned in an email some months ago.
> >>>>> I can't find the email now, but this is the command to download
> >>>>> the 320Kbps aac stream in an flv container:-
> >>>>>
> >>>>> get_iplayer --get --type=liveradio
> "http://www.bbc.co.uk/mediaselector/4/gtis/?server=cp60703.live.edgefcs.net&identifier=Special_Event1_UK@s6485&kind=akamai&application=live"
> >>>>>
> >>>> Thanks for the replay batguano - I can see this is to access the
> >>>>> live
> >>>> stream (which I have never tried but now I know how). What I was
> >>>> after
> >>>> is the mode to d/l the file later. My default mode is
> >>>> --mode=flashaac
> >>>> for R4 and R4Ex but that gets me 128K - is there a 320K equivalent
> >>>> please?
> >>>>
> >>>> Thanks.
> >>>>
> >>>> Clive
> >>> Hi
> >>> I don't think that the Radio 3 'listen again' shows are available in
> >>> 320Kbps format... but maybe I'm wrong.
> >>> See if someone else corrects me.
> >>>
> >>> By the way, when using my previous command to download the live
> >>> 320Kbps stream, it gives me better results if I include:-
> >>> --rtmp-liveradio-opts --live
> >>> in the command.
> >>> Like this:-
> >>> get_iplayer --get --type=liveradio --rtmp-liveradio-opts --live
> "http://www.bbc.co.uk/mediaselector/4/gtis/?server=cp60703.live.edgefcs.net&identifier=Special_Event1_UK@s6485&kind=akamai&application=live"
> >>>
> >>>
> >> Can you say a little more about what kind of better results you get
> >> when
> >> you use this option?
> >>
> >> Simon
> >>>
> > *** >
> > Can you say a little more about what kind of better results you get
> > when
> > you use this option?
> > Simon
> > *** >
> >
> > Hi Simon
> > If I don't use '--rtmp-liveradio-opts --live', when I come to unpack
> > the flv with FFmpeg it "sometimes" gives errors.
> > Like this:-
> > ffmpeg -i foo.flv -acodec copy foo.aac
> > [adts @ 0x9f432e0] Application provided invalid, non monotonically
> > increasing dts to muxer in stream 0: 4973220 >= 4971150
> >
> > When this happens, I've tried using absf but it doesn't seem to cure
> > it.
> > Like this:-
> > ffmpeg -i foo.flv -acodec copy -absf aac_adtstoasc foo.aac
> >
> > Using '--rtmp-liveradio-opts --live' adds '--live' to the command for
> > RTMPDump.
> >
> > I've come across this before when downloading other (non BBC) streams
> > with RTMPDump.
> >
> > For live streams (and sometimes even for non-live streams) RTMPDump
> > tries to download too fast.
> > Then it trips up, then it tries to resume.
> > So it produces weird timestamps.
> > Using '--live' option with RTMPDump will prevent this.
> >
>
> For information:
>
> I found the '--rtmp-liveradio-opts --live' option doesn't always prevent
> invalid timestamps, although it helps prevent it.
>
> More details here:
>
> http://ffmpeg.org/pipermail/ffmpeg-user/2011-June/001323.html
>
> If you get the error 'invalid, non monotonically increasing dts to muxer
> in stream' I found a workaround. Details here:
>
> http://ffmpeg.org/pipermail/ffmpeg-user/2011-June/001342.html
>
>
>
>

**************************************************
Richard said:-
>> Tried to re-encode the flv with:
>
> ffmpeg -i input.flv -acodec libfaac -ab 320k output.m4a
>
> This produced a playable m4a but it encoded at 152 kbps instead of
> 320kbps. I wanted 320 kbps sound quality.

Found a workaround. Re-encoded the flv with:

ffmpeg -i input.flv -acodec aac -strict experimental -ab 325k output.m4a

This produced playable m4a encoded at 300 kbps. Odd that libfaac should
encode at 152 kbps (instead of 320 kpbs) yet acc (which ffmpeg reports
as an experimental encoder) encodes at 300 kpbs.
***********************************************************


Hi
I don't know anything about FFmpeg's internal "experimental" aac encoder.

But libfaac is based on faac encoder.
faac is a VBR encoder.

Using faac --long-help shows:-
-q <quality>    Set default variable bitrate (VBR) quantizer quality in percent.
        (default: 100, averages at approx. 120 kbps VBR for a normal
        stereo input file with 16 bit and 44.1 kHz sample rate; max.
        value 500, min. 10).

So -aq <quality> is used with FFmpeg libfaac to set the quantizer quality of the VBR.
 
When -ab <bitrate> is used with FFmpeg libfaac it sets the ABR.
(That's the average bitrate of the variable bitrate).

Using faac --long-help shows:-
 -b <bitrate>    Set average bitrate (ABR) to approximately <bitrate> kbps.
        (max. value 152 kbps/stereo with a 16 kHz cutoff, can be raised
        with a higher -c setting).

It seems the ABR is capped at 152Kbps unless that value "c" is changed.
I'm not sure how to change that "c" value with FFmpeg, or even how to calculate a suitable value for it.