Backup Configuration
Hi all, I'm looking for help to backup my laptop installation. Not the downloaded media files, rather the preferences & PVR list information. Can any one advise how this is done, what is it I need to backup? I'm running Windows Home Premium 64bit. Cheers! M. -- http://www.fastmail.fm - Access all of your messages and folders wherever you are ___ 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
