Re: Radio File Format Questions
On Sun Dec 1 23:01:15 GMT 2013, Budgie wrote: Hi Vangelis, Top posting just to say thanks. You are most welcome... You have given me much to consider: More than I dared hope. Since this is a public mailing list, I tend to think there's some chance others may (or may not) benefit from what I write... If I get stuck, will start a new thread. If you stumble upon FFmpeg specific issues, please do consider that FFmpeg may be a helper utility used by get_iplayer, but this is the get_iplayer mailing list and it could be disputed by other list members (or its maintainer) that FFmpeg usage queries are outside the scope of this list - FWIW, there's ample documentation on the net on FFmpeg and a very active mailing list (that I frequented in the past) here: http://ffmpeg.org/pipermail/ffmpeg-user/ On 01/12/13 05:22, Vangelis forthnet wrote: On Sat Nov 30 23:14:55 GMT 2013, Budgie wrote: (gigantic snip) get_iplayer at lists.infradead.org http://lists.infradead.org/mailman/listinfo/get_iplayer (Please read this as a "do not repeat in future" friendly advice, not a reprimand:) You shouldn't have quoted the entirety of my long reply in your "just to say thanks" message; this was in direct disrespect of the list netiquette, found here: http://david.woodhou.se/email.html I have kept a copy of my mail in my "Sent Items" folder, the other list members have a copy of it in their "Inboxes" and those who are reading the list archives are just one click away from it if currently reading your latest reply! This type of behaviour will (if it hasn't already) get you on a "blacklist" by the maintainer of the list, Mr Woodhouse... Most Kind Regards, Vangelis ___ get_iplayer mailing list get_iplayer@lists.infradead.org http://lists.infradead.org/mailman/listinfo/get_iplayer
Re: Radio File Format Questions
Hi Vangelis,
Top posting just to say thanks. You have given me much to consider:
More than I dared hope. Will pursue over the next few days. If I get
stuck will start a new thread.
I have now proved to my satisfaction that my problem is due to issues at
the BBC end.
My reference to flac was an aberration on my part.
Budgie.
On 01/12/13 05:22, Vangelis forthnet wrote:
On Sat Nov 30 23:14:55 GMT 2013, Budgie wrote:
Hi Vangelis,
I have been working on other stuff and only now return to sort out my
problem files. Would you have time to help some more please?
Hello there Budgie :-) ; I was surprised, to say the least, that a 4.5
month old
thread was bumped... I'll try to do my best, but do keep in mind that I
have NO
knowledge of Open Source OSs, just the WindowsVista x86 I am running in
this aging laptop; I only hope you can extrapolate what I say to your
setup, else
the majority of the list members are savvy enough and can help you
further...
I do have ffprobe but do not see any mention of HE-AACv2 or
AAC-LC when I run it on a problem file. What I do see is :-
[Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo,
fltp, 48 kb/s (default).
You may want to further investigate the capabilities of ffprobe by
printing its help
content::
ffprobe -h > "FFprobe Help.txt" (this for Win).
What you'd need in your case is the more verbose command "-show_streams".
Sacrificing the brevity of my reply, I've opted to illustrate this with
an example:
Running the following command (from a UK IP),
get_iplayer --type=radio --pid=b03j5fxd --modes=flashaacstd --force -w
--file-prefix="aacla[b03fbbbt]" --tag-podcast-radio
will get you what you've labeled as "higher quality radio option"; this
in fact is an MP4 container with the Apple
implemented .m4a extention (for Ipod compatible audio-only content),
which "contains" a AAC-LC audio stream @128kbpsABR.
ffprobe -show_streams "aaclc[b03fbbbt].m4a"
will produce a list of info; I'd have expected that ffprobe under
"profile=" would say
"aac_low', but instead it says "unknown".
However, you should pay attention to the following set of info:
codec_time_base=1/44100
sample_rate=44100
channels=2
time_base=1/44100
bit_rate=127999
This is indicative of the flashaacstd radio mode.
get_iplayer --type=radio --pid=b03j5fxd --modes=flashaaclow --force -w
--file-prefix="aache2[b03fbbbt]" --tag-podcast-radio
will fetch the "lower quality radio option", the one that IS NOT
SUPPORTED by your hardware player.
ffprobe -show_streams "aache2[b03fbbbt].m4a"
this time produces the next set of info:
codec_time_base=1/22050
sample_rate=44100
channels=1
time_base=1/44100
bit_rate=47999
which is indicative of the flashaaclow radiomode.
(codec_time_base=1/22050 signifies the presence of SBR, while
channels=1 may mean either a monaural stream, or, as is the case here,
the presence of Parametric Stereo [=PS]; AAC+SBR+PS= HE-AACv2 !
Please read further at
http://en.wikipedia.org/wiki/AAC%2B )
Of course, with MediaInfo installed (on Windows), all is much simpler.
The programme injects a right-click context menu entry, which, when
selected, shows this info:
"aaclc[b03fbbbt].m4a"
Audio
ID : 1
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : LC
Codec ID : 40
Duration : 30mn 0s
Bit rate mode: Variable
Bit rate : 128 Kbps
Maximum bit rate : 192 Kbps
Channel(s) : 2 channels
Channel positions: Front: L R
Sampling rate: 44.1 KHz
"aache2[b03fbbbt].m4a"
Audio
ID : 1
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : HE-AACv2 / HE-AAC / LC
Codec ID : 40
Duration : 30mn 0s
Bit rate mode: Constant
Bit rate : 48.0 Kbps
Channel(s) : 2 channels / 1 channel / 1 channel
Channel positions: Front: L R / Front: C / Front: C
Sampling rate: 44.1 KHz / 44.1 KHz / 22.05 KHz
I would urge you to install MediaInfo for your distro, if you
haven't done already, because it is way more practical in examining
media files than the CLI ffprobe - please ask for further help, if needed,
about installing it on your OS, as I am clueless in this field... :-{
I interpret this to mean that I have downloaded the flashaaclow
version which from your advice I interpret to be the HE-AACv2 encoded
version. Is that correct?Judging only by the declared bitrate (= 48
kb/s), then YES - but if you had the patience to
read through my elaboration above, then you'd have figured this out
already :-)
Since I have no idea why I get these files from time to time
I can safely say tha
Re: Radio File Format Questions
On Sat Nov 30 23:14:55 GMT 2013, Budgie wrote:
Hi Vangelis,
I have been working on other stuff and only now return to sort out my
problem files. Would you have time to help some more please?
Hello there Budgie :-) ; I was surprised, to say the least, that a 4.5 month
old
thread was bumped... I'll try to do my best, but do keep in mind that I have
NO
knowledge of Open Source OSs, just the WindowsVista x86 I am running in
this aging laptop; I only hope you can extrapolate what I say to your setup,
else
the majority of the list members are savvy enough and can help you
further...
I do have ffprobe but do not see any mention of HE-AACv2 or
AAC-LC when I run it on a problem file. What I do see is :-
[Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo,
fltp, 48 kb/s (default).
You may want to further investigate the capabilities of ffprobe by printing
its help
content::
ffprobe -h > "FFprobe Help.txt" (this for Win).
What you'd need in your case is the more verbose command "-show_streams".
Sacrificing the brevity of my reply, I've opted to illustrate this with an
example:
Running the following command (from a UK IP),
get_iplayer --type=radio --pid=b03j5fxd --modes=flashaacstd --force -w --file-prefix="aacla[b03fbbbt]"
--tag-podcast-radio
will get you what you've labeled as "higher quality radio option"; this in
fact is an MP4 container with the Apple
implemented .m4a extention (for Ipod compatible audio-only content), which
"contains" a AAC-LC audio stream @128kbpsABR.
ffprobe -show_streams "aaclc[b03fbbbt].m4a"
will produce a list of info; I'd have expected that ffprobe under "profile="
would say
"aac_low', but instead it says "unknown".
However, you should pay attention to the following set of info:
codec_time_base=1/44100
sample_rate=44100
channels=2
time_base=1/44100
bit_rate=127999
This is indicative of the flashaacstd radio mode.
get_iplayer --type=radio --pid=b03j5fxd --modes=flashaaclow --force -w --file-prefix="aache2[b03fbbbt]"
--tag-podcast-radio
will fetch the "lower quality radio option", the one that IS NOT SUPPORTED
by your hardware player.
ffprobe -show_streams "aache2[b03fbbbt].m4a"
this time produces the next set of info:
codec_time_base=1/22050
sample_rate=44100
channels=1
time_base=1/44100
bit_rate=47999
which is indicative of the flashaaclow radiomode.
(codec_time_base=1/22050 signifies the presence of SBR, while
channels=1 may mean either a monaural stream, or, as is the case here,
the presence of Parametric Stereo [=PS]; AAC+SBR+PS= HE-AACv2 !
Please read further at
http://en.wikipedia.org/wiki/AAC%2B )
Of course, with MediaInfo installed (on Windows), all is much simpler.
The programme injects a right-click context menu entry, which, when
selected, shows this info:
"aaclc[b03fbbbt].m4a"
Audio
ID : 1
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : LC
Codec ID : 40
Duration : 30mn 0s
Bit rate mode: Variable
Bit rate : 128 Kbps
Maximum bit rate : 192 Kbps
Channel(s) : 2 channels
Channel positions: Front: L R
Sampling rate: 44.1 KHz
"aache2[b03fbbbt].m4a"
Audio
ID : 1
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : HE-AACv2 / HE-AAC / LC
Codec ID : 40
Duration : 30mn 0s
Bit rate mode: Constant
Bit rate : 48.0 Kbps
Channel(s) : 2 channels / 1 channel / 1 channel
Channel positions: Front: L R / Front: C / Front: C
Sampling rate: 44.1 KHz / 44.1 KHz / 22.05 KHz
I would urge you to install MediaInfo for your distro, if you
haven't done already, because it is way more practical in examining
media files than the CLI ffprobe - please ask for further help, if needed,
about installing it on your OS, as I am clueless in this field... :-{
I interpret this to mean that I have downloaded the flashaaclow
version which from your advice I interpret to be the HE-AACv2 encoded
version. Is that correct?Judging only by the declared bitrate (= 48 kb/s),
then YES - but if you had the patience to
read through my elaboration above, then you'd have figured this out already
:-)
Since I have no idea why I get these files from time to time
I can safely say that most probably it's something on the Beeb's side - in
the latest
month I came across 4 or 5 instances where a radio show was unavailable in
the
flashaacstd mode for no apparent reason; the radio shows preceding &
succeeding it
might've been, not the specific one in between I was after. And the
situation remained
so for the whole duration of the 7day avai
Re: Radio File Format Questions
Budgie wrote: >Since I have no idea why I get these files from time to time I assume I >do not have get_iplayer set up correctly so that is cannot download in >this format. First request then is what should I put in my options file >to ensure I only get the higher quality radio options? I think you need to set --modes= to a list of the modes you're willing to download. Suppose that the default is something like: flashaachigh,flashaacstd,flashaaclow,wma but that at the time that your fetch is attempted only the lower quality files are actually working properly from the content provider... then GiP has probably tried the high quality ones, they've failed and the lower quality one has been fetched. If you code for example --modes=flashaachigh,flashaacstd then I think only those quality levels would be attempted. -- Jeremy Nicoll - my opinions are my own. ___ get_iplayer mailing list get_iplayer@lists.infradead.org http://lists.infradead.org/mailman/listinfo/get_iplayer
Re: Radio File Format Questions
On 14/07/13 13:37, Vangelis forthnet wrote: On Sat Jul 13 15:52:02 BST 2013, Budgie wrote: As usual, a couple of questions. Is the file format HE-AAC v2 the normal output for a low bit rate download or is it another, to me, anomaly? Hello. Yes, 'flashaaclow' radiomode yields an audio file packaged in an MP4 container (whose format profile is "Apple audio with iTunes info", hence the .m4a extention), which in it contains a raw ADTS (audio data transport stream) .aac file encoded in HE-AACv2 as you correctly state; HE-AAC is AAC+SBR, v2 indicates the use of PS (parametric stereo). The encode uses a VBR with a mean value of 48kbps. NB that if you come from a non-UK IP, this is the only audio quality available to you for National Stations. If in the UK, the default high quality mode (= flashaac/flashaacstd) is again an .m4a file, but the audio stream contained therein is encoded in AAC LC (no SBR, no PS) @ 128kbps (320kbps for Radio 3) ABR, that's why it is more compatible with software/ hardware players. Depending on the player used, the PS part may be skipped (audio plays in mono), or both PS+SBR skipped, in which case audio plays in mono and in very low quality, since only half the sampling rate is used. In my Windows setup I haven't come across a software player that does not play at least the AAC part of a HE-AACv2 encode. But hardware players (like your network player here: http://www.linn.co.uk/all-products/network-music-players/sneaky-ds ) behave differently; the features list of yours only mentions a "generic AAC" decoding support, so it may be expected that it does not support HE-AAC (try a World Service download) or HE-AACv2, as you have found out. On your laptop, any ffmpeg based software player (FFplay, + the ones you mentioned) can play fully HE-AACv2 audio streams. What programme can I use to find out the detailed information of what is in each .m4a file? As a generic multimedia file "investigator", you can use the CLI FFprobe, http://ffmpeg.org/ffprobe.html which, together with FFplay, is part of the FFmpeg package - if it isn't available for your OS, maybe its fork "avprobe" is: http://libav.org/avprobe.html As a personal choice though, I'd recommend MediaInfo - it comes both as a GUI & CLI and is available for a plethora of OSes, including yours (openSUSE 12.2) here: http://mediaarea.net/el/MediaInfo/Download/openSUSE what would you recommend I run to change the format of the sound file and to what format? dinkypumpkin in your answer to you has kindly suggested a recode from HE-AACv2 -> AAC-LC through FFmpeg (or avconv). If your FFmpeg is built with support for one of the non-free AAC encoders (libfaac or the far better libfdk_aac), then I guess it'd be fine, but the native encoder (-c:a aac -strict -2) lacks in performance, especially in music parts - for speech is fine. If I can humbly share my opinion, I have found that a transcode from HE-AACv2 @48kbps -> LAME MP3 @ 96kbps (for spoken content) / 112 (or even 128) kbps (for music content) is more than adequate and I would propose that, since your SneakyDS does play MP3 files. Regards. Vangelis Hi Vangelis, I have been working on other stuff and only now return to sort out my problem files. Would you have time to help some more please? First question concerns diagnosis of the files which do not play on Linn device. I do have ffprobe but do not see any mention of HE-AACv2 or AAC-LC when I run it on a problem file. What I do see is :- [CODE] Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 48 kb/s (default). [CODE] I interpret this to mean that I have the downloaded the flashaaclow version which from your advice I interpret to be the HE-AACv2 encoded version. Is that correct? Since I have no idea why I get these files from time to time I assume I do not have get_iplayer set up correctly so that is cannot download in this format. First request then is what should I put in my options file to ensure I only get the higher quality radio options? Second question is please could you help with ffmpeg command line to convert these files to files that will play as suggested previously by dinkypumpkin. I regret my knowledge is not up to doing it without more help with command options. Finally I note I could transcode on downloading and save these files as mp3 files but is it also possible to transcode to flac? Grateful for any further help when you have time. Regards, Budgie ___ get_iplayer mailing list get_iplayer@lists.infradead.org http://lists.infradead.org/mailman/listinfo/get_iplayer
Re: Radio File Format Questions
On 15/07/2013 21:10, Vangelis forthnet wrote: ffmpeg -i foo.m4a -vn -c:a libmp3lame -ab 96k -ac 2 -ar 44.1k foo.mp3 (NB that the m4a file's metadata will be lost during the conversion) If you have a relatively recent version of ffmpeg, a lot of the metadata will be preserved. You'll lose the cover art, but the major ID3 frames will be populated. openSUSE 12.3 packages ffmpeg 1.0.6, which is recent enough to do the job. ___ get_iplayer mailing list get_iplayer@lists.infradead.org http://lists.infradead.org/mailman/listinfo/get_iplayer
Re: Radio File Format Questions
On Sun Jul 14 22:48:46 BST 2013, Budgie wrote: Very many thanks for your instructive and helpful reply. You are most welcome; apologies I did not get back to you sooner. I am currently mourning the sudden loss of an external USB HDD 1TB in capacity, which was full with close to 900GB worth of files (the majority of which are irreplaceable), accumulated in the course of nearly 3 years. It will cost me a small fortune to buy a replacement disk and recover about 85% of the data that can be salvaged from the failed disk :-( :-( This will help me investigate the several other files which are causing me problems. More often than not, the first "Listen Again" radiomode that is made available online by the beeb after a radio show has finished is the "flashaaclow" one (in GiP's terms), which they call the "Lower Bandwidth Version". I don't know if this is by design, but it's my personal observation. For example, after the Official Chart Show on Radio 1 finishes at 19:00 BST on a Sunday, the "flashaaclow" mode appears first 1,5-2 hours later, while it usually takes another 1-2 hours for the "flashaacstd" mode to be made available (this depends on the load on the encoders chain...). If you are doing your audio downloads in an automated way via a PVR list and you have not explicitly asked for the "flashaacstd" mode, then, when your PVR list is executed, it may sometimes download the low quality version of a programme, because at that time it was the only one available. This is possibly why you end up with some HE-AACv2 .m4a files inside your downloads folder... for the spoken word I would be happy to use mp3. This is how I receive BBC podcasts of spoken word FYI, these are mono files encoded @ 64kbps constant bitrate (CR); the source used for the encode is probably a high quality master, that's why they do sound quite good, but TBH I do not like the monaural sound, especially when listened to through headphones. If I wish to transcode the problem HE-AACv2 file to mp3 should I do this with ffmpeg or another program? Any audio conversion software capable of fully decoding HE-AACv2 and encoding to mp3 should do the task; I have no clue what are your choices in your platform (OpenSuSE 12.2). FFmpeg is fine for this - I would use something like this: ffmpeg -i foo.m4a -vn -c:a libmp3lame -ab 96k -ac 2 -ar 44.1k foo.mp3 (NB that the m4a file's metadata will be lost during the conversion) On Mon Jul 15 18:31:47 BST 2013, Budgie wrote: Yes I am using ffmpeg. First try gave me twice the file size as advised by Vangelis but I am looking at options to set lower bit rate for output file. You never mentioned that file size is an issue for you. HE-AACv2 is a very efficient encoder at low bitrates (22-64kbps), that is why it is now used very widely for internet radio streaming (to cut down on bandwidth costs). In comparison, LAME MP3 lacks considerably in this field. If you are prepared to compromise with some quality loss, you could experiment with bitrates lower than 96kbps, or try a variable bitrate (VBR) scheme, but I wouldn't try values lower than 64kbps (unless you are about to listen to the end result on a mobile phone with a cheap set of earphones...) Just my two "eurocents" ... Vangelis. ___ get_iplayer mailing list get_iplayer@lists.infradead.org http://lists.infradead.org/mailman/listinfo/get_iplayer
Re: Radio File Format Questions
Use ffmpeg but to make things easier, use Winff which is a gui interface with ffmpeg. I use it on both Windows and Linux and it works a treat. Clive On 14/07/13 22:48, Budgie wrote: If I wish to transcode the problem HE-AACv2 file to mp3 should I do this with ffmpeg or another program? Many thanks again, Budgie ___ get_iplayer mailing list get_iplayer@lists.infradead.org http://lists.infradead.org/mailman/listinfo/get_iplayer ___ get_iplayer mailing list get_iplayer@lists.infradead.org http://lists.infradead.org/mailman/listinfo/get_iplayer
Re: Radio File Format Questions
On Sun Jul 14 13:44:55 BST 2013, Chris Marriott wrote: What do we get if we use the "aactomp3" flag on "get_iplayer"? That's the way that I've always done it personally, and the results are entirely satisfactory (to my ear) for spoken word programmes (which are all I ever download). Hello Chris! From GiP's "longhelp" (get_iplayer --longhelp) I quote: Recording Options: --aactomp3 Transcode AAC audio to MP3 with ffmpeg (CBR 128k unless --mp3vbr is specified) --mp3vbr Set LAME VBR mode to N (0 to 9) for AAC transcoding. 0 = target bitrate 245 Kbit/s, 9 = target bitrate 65 Kbit/s (requires --aactomp3) that is if you are downloading a "flashaac" radiomode (no matter if it is the low/std variant) and you have specified the --aactomp3 switch, you will end up with an .mp3 audio file transcoded @ 128kbps constant bitrate (CBR). If it is the flashaaclow mode you are recording (which I find is sufficient for spoken word content), then with --aactomp3 you do have the original quality preserved, but at a 2.6 times the original file size (=128/48). In such a case I would experiment with the --mp3vbr switch at values larger than 7, which will produce smaller mp3 files - beware though that some hardware (/ portable) mp3 players are "peaky" about VBR files; they may report wrong duration or not play the mp3 file at all... If it is the flashaacstd mode (default from within the UK) you are recording, --aactomp3 produces a same size (to the initial .aac source) audio file, with no noticeable loss of quality for spoken content. This is not the case for Radio 3 content, please also refer to my post earlier in the month: http://lists.infradead.org/pipermail/get_iplayer/2013-July/004425.html And I will repeat myself, but only transcode if you really have to! Now, as far as the OP (Budgie) is concerned, the way I understood it is that he has already on disk some "flashaaclow" audio files that have presumably expired (so he cannot re-download them using the --aactomp3 switch) but needs to listen to them on his Network Player that does not support the encoding format of the files (HE-AACv2 = AAC+SBR+PS). In order to do this, he must re-encode them to a format supported by his Player... I hope I made things clearer now :-) Vangelis. ___ get_iplayer mailing list get_iplayer@lists.infradead.org http://lists.infradead.org/mailman/listinfo/get_iplayer
Re: Radio File Format Questions
-Original Message- From: Vangelis forthnet Sent: Sunday, July 14, 2013 1:37 PM To: get_iplayer@lists.infradead.org Subject: Re: Radio File Format Questions If I can humbly share my opinion, I have found that a transcode from HE-AACv2 @48kbps -> LAME MP3 @ 96kbps (for spoken content) / 112 (or even 128) kbps (for music content) is more than adequate and I would propose that, since your SneakyDS does play MP3 files. What do we get if we use the "aactomp3" flag on "get_iplayer"? That's the way that I've always done it personally, and the results are entirely satisfactory (to my ear) for spoken word programmes (which are all I ever download). Chris ___ get_iplayer mailing list get_iplayer@lists.infradead.org http://lists.infradead.org/mailman/listinfo/get_iplayer
Re: Radio File Format Questions
On Sat Jul 13 15:52:02 BST 2013, Budgie wrote: As usual, a couple of questions. Is the file format HE-AAC v2 the normal output for a low bit rate download or is it another, to me, anomaly? Hello. Yes, 'flashaaclow' radiomode yields an audio file packaged in an MP4 container (whose format profile is "Apple audio with iTunes info", hence the .m4a extention), which in it contains a raw ADTS (audio data transport stream) .aac file encoded in HE-AACv2 as you correctly state; HE-AAC is AAC+SBR, v2 indicates the use of PS (parametric stereo). The encode uses a VBR with a mean value of 48kbps. NB that if you come from a non-UK IP, this is the only audio quality available to you for National Stations. If in the UK, the default high quality mode (= flashaac/flashaacstd) is again an .m4a file, but the audio stream contained therein is encoded in AAC LC (no SBR, no PS) @ 128kbps (320kbps for Radio 3) ABR, that's why it is more compatible with software/ hardware players. Depending on the player used, the PS part may be skipped (audio plays in mono), or both PS+SBR skipped, in which case audio plays in mono and in very low quality, since only half the sampling rate is used. In my Windows setup I haven't come across a software player that does not play at least the AAC part of a HE-AACv2 encode. But hardware players (like your network player here: http://www.linn.co.uk/all-products/network-music-players/sneaky-ds ) behave differently; the features list of yours only mentions a "generic AAC" decoding support, so it may be expected that it does not support HE-AAC (try a World Service download) or HE-AACv2, as you have found out. On your laptop, any ffmpeg based software player (FFplay, + the ones you mentioned) can play fully HE-AACv2 audio streams. What programme can I use to find out the detailed information of what is in each .m4a file? As a generic multimedia file "investigator", you can use the CLI FFprobe, http://ffmpeg.org/ffprobe.html which, together with FFplay, is part of the FFmpeg package - if it isn't available for your OS, maybe its fork "avprobe" is: http://libav.org/avprobe.html As a personal choice though, I'd recommend MediaInfo - it comes both as a GUI & CLI and is available for a plethora of OSes, including yours (openSUSE 12.2) here: http://mediaarea.net/el/MediaInfo/Download/openSUSE what would you recommend I run to change the format of the sound file and to what format? dinkypumpkin in your answer to you has kindly suggested a recode from HE-AACv2 -> AAC-LC through FFmpeg (or avconv). If your FFmpeg is built with support for one of the non-free AAC encoders (libfaac or the far better libfdk_aac), then I guess it'd be fine, but the native encoder (-c:a aac -strict -2) lacks in performance, especially in music parts - for speech is fine. If I can humbly share my opinion, I have found that a transcode from HE-AACv2 @48kbps -> LAME MP3 @ 96kbps (for spoken content) / 112 (or even 128) kbps (for music content) is more than adequate and I would propose that, since your SneakyDS does play MP3 files. Regards. Vangelis ___ get_iplayer mailing list get_iplayer@lists.infradead.org http://lists.infradead.org/mailman/listinfo/get_iplayer
Re: Radio File Format Questions
On 13/07/2013 15:52, Budgie wrote: Pending Linn sorting out their DS player what would you recommend I run to change the format of the sound file and to what format? You should be able to do an AAC->AAC transcode with ffmpeg. The default settings should give you an output file with AAC-LC profile. The AAC-HE profile is presumably what the Linn player can't handle. http://ffmpeg.org/trac/ffmpeg/wiki/AACEncodingGuide ___ get_iplayer mailing list get_iplayer@lists.infradead.org http://lists.infradead.org/mailman/listinfo/get_iplayer
Radio File Format Questions
Further to my attempts to sort out my radio downloads, the size anomaly, which has been explained and non playing of some files, I found that the low bitrate file I mentioned earlier would not play on my Linn SneakyDS but the larger file did play. Both files played well on other renderers. I sought help on the Linn forum and this is the consequential post from bubbleguuum, addressed to Linn Development Forum:- [quote]Here's a HE-AAC v2 file that doesn't play on a DS. It is from a BBC podcast. It is an 48Kbps AAC SBR+PS file. foobar2000 and VLC play it. Playing it to a DS fails silently. [quote] As usual, a couple of questions. Is the file format HE-AAC v2 the normal output for a low bit rate download or is it another, to me, anomaly? What programme can I use to find out the detailed information of what is in each .m4a file? I have used AtomicParsley to discover tagging metadata and then qtfaststart to sort out order of atoms where for whatever reason, these have been screwed up. Pending Linn sorting out their DS player what would you recommend I run to change the format of the sound file and to what format? Regards, Budgie ___ get_iplayer mailing list get_iplayer@lists.infradead.org http://lists.infradead.org/mailman/listinfo/get_iplayer
