On 30/10/2019 12:06, Budge wrote: > On 29/10/2019 16:50, RS wrote: >> >> >> On 27/10/2019 21:08, Budge wrote: >>> On 27/10/2019 20:56, Budge wrote: >> >>>> >>>> Further to this thread as it has developed I find I have two example >>>> files both downloaded with GiP. Using ffprobe, one is shown as:- >>>> >>>> Duration: 02:33:00.99, start: 0.000000, bitrate: 321 kb/s >>>> Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, >>>> fltp, 320 kb/s (default) >>>> Metadata: >>>> handler_name : SoundHandler >>>> Stream #0:1: Video: mjpeg (Baseline), yuvj420p(pc, >>>> bt470bg/unknown/unknown), 150x84 [SAR 72:72 DAR 25:14], 90k tbr, 90k >>>> tbn, 90k tbc (attached pic) >>>> >>>> and the second:- >>>> >>>> Duration: 02:37:00.06, start: 0.000000, bitrate: 321 kb/s >>>> Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, >>>> stereo, fltp, 320 kb/s (default) >>>> Metadata: >>>> handler_name : SoundHandler >>>> Stream #0:1: Video: mjpeg (Baseline), yuvj420p(pc, >>>> bt470bg/unknown/unknown), 86x48 [SAR 72:72 DAR 43:24], 90k tbr, 90k tbn, >>>> 90k tbc (attached pic). >>>> >>>> Among the differences I note first is AAC, I assume HE? at 48000Hz and >>>> the second is AAC (LC) at 44100Hz. >>>> >>>> The first is later and dated 2017-01-19 and the second 2014-05-15. >>>> >>>> It is the earlier download that works and I still believe the problem >>>> has been from my incorrect setting up of GiP. Both files play on all my >>>> players except the Linn DS devices. >>>> >>>> I am about to pass the problem over to Linn but before I do please could >>>> somebody suggest why the two files have different codecs. >>>> >>> >>> Correction. Not HE above, just AAC. >>> Budge >>> >> The principal difference between the two files is that the one that >> works has a sampling rate of 48kHz and the one that doesn't has a >> sampling rate of 44.1kHz. One possibility is that Linn does not support >> a 44.1kHz sampling rate, although it would be very surprising if it didn't. >> >> Since "Linn recommends FLAC" you could try converting the 44.1kHz >> sampling rate file to FLAC at the same sampling rate. You could also >> try resampling the 44.1kHz sampling rate file to 48kHz. >> >> What puzzles me is how you got a file with a sampling rate of 44.1kHz >> and a bit rate of 320kbit/s. I am not saying that is not a valid >> combination; it is. What I am saying is that as far as I am aware it is >> not a combination available from the BBC. >> >> Many radio programmes are available as podcasts. You can download a >> podcast from the BBC website. You will be offered a choice of bit >> rates, 64kbit/s and 128kbit/s. The file will be in MP3 format with a >> sampling rate of 44.1kHz. >> >> If you download a radio programme with get_iplayer without using the >> --raw option you will get a M4A/AAC file with a sampling rate of 48kHz. >> You will have a choice of bit rates, in some cases up to 320kbit/s. >> >> In the last few months programmes with podcasts have also had podcast >> versions which can be downloaded with get_iplayer. More recently still >> for some programmes the only download available with get_iplayer has >> been the podcast version. In either case the file format has been >> M4A/AAC and the sampling rate has been 48kHz. >> >> One example of a Radio 3 programme with a podcast is The Listening Service. >> get_iplayer --pid=m0009jzd --info >> shows that it has both original and podcast versions and, for both, bit >> rates up to 320kbit/s are available. If you download it you will see >> the sampling rate is 48kHz. >> >> If you go to the programme's website >> https://www.bbc.co.uk/programmes/b078n25h/episodes/downloads >> the download buttons will offer a choice of 64kbit/s and 128kbit/s bit >> rates. If you download one of them you will get a MP3 file with a >> sampling rate of 44.1kHz. >> >> There is no option which combines a 44.1kHz sampling rate with a bit >> rate of 320kbitj/s. >> >> I have just read your post again, and it seems I have got the one that >> works and the one that doesn't the wrong way round. It seems even more >> unlikely that your Linn device would not support a 48kHz sampling rate, >> although 44.1kHz is the CD standard. >> >> Best wishes >> Richard >> >> >> _______________________________________________ >> get_iplayer mailing list >> get_iplayer@lists.infradead.org >> http://lists.infradead.org/mailman/listinfo/get_iplayer > Hi Richard, > Many thanks for the information and for your explanations. > > Unfortunately all these files are historical downloads which I am trying > to clean up and I believe (but cannot easily check,) these problem files > have resulted from downloads following an operating system change or new > installation, following which I have made errors which have munged the > downloads until discovered and corrected! > > Almost all of the problems can be linked to playing, or not as the case > may be, on Linn DS devices. > > Last time I tried running through ffmpeg things became worse even when > just "copying" through > > I am trying again to raise this with Linn. Meanwhile I will look again > at my samples as I could easily get them muddled but afaik my normal > classical (third programme) downloads are:- > > Stream #0:0(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, > fltp, 320 kb/s (default) > > which is what you advise and these play fine. > > Let us see how we get on with Linn. > Thanks again, > Alastair. > > _______________________________________________ > get_iplayer mailing list > get_iplayer@lists.infradead.org > http://lists.infradead.org/mailman/listinfo/get_iplayer >
At the risk of being slightly OT because normal downloads of classical music are all fine with Linn DS and all the others I would like to raise again my problem because I have had essentially the same reply from Linn that I had in 2017. Here is what they say:- "The engineer advised that he received a file from you and advised that if you convert the track to FLAC or ALAC that it plays. The problems are caused by the file being split into an enormous number of tiny chunks (over 400,000 audio blocks for a 9,200 second track); any encoder which reduced this would allow the file to play. I passed the first track you supplied to me to them and they advised that have advised that the track appears to be corrupt, and transcoding the track should allow it to then be played on the DSM." Leaving aside the suggestion that the file had been corrupted, which in one case is credible even though it plays fine on eg VLC or RPi with mplayer and upmpdcli, can anybody explain what Linn are on about? I shall try and find my earlier thread and try again to get to the bottom of this. Budge _______________________________________________ get_iplayer mailing list get_iplayer@lists.infradead.org http://lists.infradead.org/mailman/listinfo/get_iplayer
