An influential voice becomes quieter

If you grew up in the English Midlands, you or your parents might have listened to Ed Doolan, a radio presenter who came from Australia to join the big Birmingham independent station BRMB in 1974. Some years later, when independent radio was changing its tactics, he was recruited by, and became popular on, the BBC station in the same city.

His programmes contained plenty of campaigning, endless local relevance, and listener involvement in countless forms. Ed Doolan’s voice, opinions and style have become very familiar to me over the last forty years and clearly influenced my own much smaller and less successful career in front of radio microphones.

I no longer live in the Midlands, but always tuned the car radio to BBC WM when in the area to hear his conversations.

Lately, Ed Doolan has gradually reduced his radio workload. He went on air on BBC WM the other day to explain to the station’s host Caroline Martin why he had retired from live broadcasting.

To hear this fellow, only 23 years older than me, and familiar over four decades, say “I’ve got dementia” simply halted all I was doing this afternoon. His full interview is here:

http://www.bbc.co.uk/programmes/p02hpwf3

Real-time visual pitch display

Here’s a hypnotic (or nausea-inducing) way of watching and listening to BBC radio programmes. You’ll need a modern version of FFplay, the multi-media player that’s part of the FFmpeg suite, and the open-source “get_iplayer” program. The filter that does the work is called “showcqt”.

For this example, I’m using BBC Radio 3. You will, no doubt, see how the command line can be modified to accept any audio source.

Just type this. This is from a Cygwin command line, rather similar to Unix. Windows won’t be much different.

get_iplayer --stream --type=liveradio "BBC Radio 3" | ffplay -f lavfi "amovie='pipe\:0',asplit[a][b];[a]showcqt=fullhd=0:timeclamp=0.3:fps=30[out0]; [b]anull[out1]"

Or, as another example, here’s one of my favourite on-line streams, “The Departure Lounge”:

ffplay.exe" -f lavfi "amovie='http\://listen64.radionomy.com/TheDepartureLounge',asplit[a][b];[a]showcqt=fullhd=0:timeclamp=0.3:fps=30[out0]; [b]anull[out1]"

…and, after waiting a few seconds for buffering, you’ll get this:

Audio spectrum of a fragment of a song for soprano and piano, with turntable rumble visible in the lower frequencies
Audio spectrum of a fragment of a song for soprano and piano, with turntable rumble visible in the lower frequencies

The backslash in the “pipe\:0” is because colons must be escaped with a backslash in FFmpeg/FFplay filters.

Just out of interest, I have a Python project that outputs a handy video and audio scope that needs a little refinement, but you can download it here: https://github.com/Warblefly/FFmpeg-Scope/

The scope’s on-screen display includes a waveform monitor showing superimposed YUV levels with 16-235 markers to check BT601/709 broadcast limits, an EBU R128 loudness chart, a stereo audio sum/difference display, a colour vectorscope, a full-range video check monitor and timecode.

This is the kind of output it gives:

Screenshot of FFmpeg scope
Screenshot of FFmpeg scope

Recorded music podcasts from the UK? No.

Is it truly impossible to send out a British-made podcast where recorded music is played?

It would seem so. Phonographic Performance Limited (PPL), who licence nearly all record labels’ recorded music for public performance, do not offer an Internet-only licence to include recorded music in on-line podcasts. Broadcasts are fine, where you can’t skip forward in a show: but not podcasts that can be manipulated on demand.

If something on-line is merely replicating an already broadcast radio programme, it’s fine. But Internet-only radio from the UK, using music on most record labels, is still not allowed.

Isn’t that a curious anomaly? The Performing Right Society, and the Mechanical Copyright Protection Society, who licence the music, are fine about it. But the record labels, represented by PPL are not.

One of the greatest powers of radio is to introduce music that is new to an audience, by allowing an expert curator to showcase records they have chosen. John Peel, late of BBC Radio 1 is an example that comes immediately to mind; likewise Lucie Skeaping or Andrew MacGregor, both of BBC Radio 3. But with an increasing number of young people turning exclusively to on-line sources, why can’t the Internet be allowed to broaden the range of curators (presenters, if you like) to include those without current BBC or independent radio contracts?

A discussion about this is going on right now, on the “Radio Today” website. Perhaps PPL will join in? Or maybe I’ll just phone them for a chat and report back?

A Government Falls

For some reason this morning, while watching the featureless sky outside this window and waiting for Prime Minister’s Questions to start, I’m reminded of a turning point in British political history.

Thanks to the UK Parliamentary Recording Unit, you can hear the exact moment in 1979 when James Callaghan’s Labour government was challenged, by Margaret Thatcher’s Conservative opposition, to a vote of no confidence. As everyone knows, the vote was carried and thus an election was forced leading to a succession of Conservative governments.

The speeches surrounding this motion, by the two party leaders, can be heard in longer form by clicking this link.

Converting video for DVD with FFmpeg

Here’s another quick command line for FFmpeg. It converts interlaced video and audio into deinterlaced DVD-ready files. Your output will be a VOB file, ready to be split into a file of the correct size by any DVD authoring program (e.g. DVDStyler) without any further recoding.

The command line you see below was written for a recent film show, where interlaced material had been supplied on DVD, where the projector would not resize interlaced video correctly, and where the only replay device was a standard DVD player.

This command line is careful to apply the appropriate flags to the bitstream to signal that the video uses broadcast levels, and encodes colour according to ITU Rec.601, the standard for European (PAL) SD television.

Two filterchains are in use. The video filterchain first de-interlaces the incoming video, then applies noise-reduction because the files given to me were already noisy and, therefore, would waste bandwidth after encoding. The audio filterchain delays the sound by just over a frame: I found this to be necessary, possibly because of delays introduced by the video coder and the video filter.

ffmpeg -i VIDEO_INPUT -target pal-dvd -vf "w3fdif, hqdn3d" -af "adelay=50|50" -color_range 1 -colorspace 5 -color_primaries 5 -color_trc 5 VIDEO_OUTPUT.VOB

MXF Op-Atom files for Avid

THIS POST HAS BEEN PARTLY SUPERSEDED BY THE MUCH FASTER METHOD SHOWN HERE: http://johnwarburton.net/blog/?p=50731 BUT SOME METADATA IS OMITTED.

This post shows how to convert almost any kind of video and audio into native Avid Op-Atom MXF files, suitable for placement directly in Avid’s MXF media files directory. The method is fast, and uses only open source software. Crucially, conversion takes place on any machine, not just an Avid-equipped computer.

A side note regarding AMA: it’s sometimes (?) a little flaky when linking to files that aren’t from a small subset of QuickTime, or that have their own manufacturer-tested plugins.

In this example, I am importing footage into a 25fps HD project. The Avid codec is its own DNxHD, running at 145MBit/s.

Use FFmpeg to convert your incoming footage into uncompressed audio files, and into Avid’s native video format. Note that the video is not encapsulated beyond the raw DNxHD format: but this format contains almost enough information about the file to enable import to take place. Frame rate, for example, seems to be missing.

So, convert the incoming video into DNxHD and uncompressed audio with FFmpeg like this:

ffmpeg -i "bach.flv" -vcodec dnxhd -b:v 145M -an -sws_flags lanczos -vf "scale=1920:1080, smartblur=1.0:-1.0" bach-video.dnxhd -vn -ar 48000 -acodec pcm_s16le bach-audio.wav

I have scaled the video to the correct size using what I consider to be the best scaling algorithm (Lanczos), and have added a little crispness to avoid too much softening. Obviously, you will not want to do this to footage that is already the correct dimensions and does not need restoration.

Now, we must prepare these files for Avid, in the same way that Avid itself imports files. They must be encapsulated as Avid-flavour MXFs (Op-Atom). Here, the BBC and EBU-supported raw2bmx utility, from bmxlib, comes into play. Again, this is open source software, and this is a very simple command line. Much more metadata can be included, and you’ll need to think about this if you’re going to reconform the project at any stage.

On this command line, I instruct raw2bmx to wrap both the video file and the stereo audio file into MXF. The project name is given, as is a tape name. The output file location together with the file prefix is given.

You will also need to specify the frame rate, using the ‘-f’ option, if your footage is not 25fps. The rates acceptable are: 23976, 24, 25, 2997, 30, 50, 5994 and 60. The incoming DNxHD is specified by “–vc3_1080p_1237”, naming the codec, picture size and flavour. All such flavours are listed in the help for raw2bmx.

raw2bmx -t avid -f 25 --project BACH --clip "BACH001" -o "I:\Avid MediaFiles\MXF\1\BACH001" --vc3_1080p_1237 bach-video.dnxhd --wave bach-audio.wav

In your Avid Mediafiles directory, a number of MXF files will appear: Avid’s Media Tool will pick these up as clips with combined video and audio (if that’s what you’re converting), and you can drag the clips to whichever bin you wish. Note that the raw2bmx tool is terse in its progress reporting. It prints nothing until the end of the wrapping process.

Recent builds of FFmpeg can be downloaded here, and the bmxlib project is on Sourceforge here.

A Lot To Learn

Today, Thursday 21st August, the GCSE exam results come out. In my schooldays, we went through the same results procedures for our O-levels and CSEs, although coursework generally wasn’t assessed. This was the first time we’d ever experienced result nerves, as the staff rifled through sealed envelopes until the correct name was found.

It was considered normal at my large, good, comprehensive school to take somewhere between four and ten exams. Today, teenagers regularly sit many more than this, and marvellous alternative qualifications are available for young people whose examination skills don’t match their real-world virtuosity.

We had most of the benefits that modern times bring: safe food and water, the National Health Service, easy transport with much cheaper petrol, luxuries spread around more classes than in our parents’ time, and lots of entertainment on record and cassette tape.

But we didn’t have the Internet with the immense, often anonymous, social pressures it brings to young minds.

A sixteen-year-old today can debate directly over Twitter with, for example, Richard Dawkins, Buzz Aldrin or Lily Allen; but he or she is also subject to anonymous and permanent criticism or attack on any aspect of their life, real or imagined, from any corner of the globe. Likewise, almost every media outlet was heavily edited: we had newspapers, radio and tv, but zines and self-published information were much more scarce than they are today. Blogs or instant social networks, outside radio hams and CBers, were just a dream. Now, teenagers must think editorially from their earliest exposure to the Internet, or be misled.

For sixteen-year-olds today, it seems to me that there’s much more to learn, and to refute, than there was for us in 1980, thirty-four years ago.

Timecode overlay with FFmpeg

This post describes how to use FFmpeg, a free and open-source program, to burn filename and timecode automatically into any number of video files, and then save them in a form suitable for network viewing.

In the olden days, video rushes would be burned to DVD through a VTR or, latterly, timecode plugin on Avid so as to give everyone timecoded copies.

Today, the free and open-source FFmpeg program can complete this task in the background on almost any modern computer. In this office, it’s making burnt-in timecode visible on three machines: a Mac PowerBook G4 made in 2004 (PowerPC processor) now running Debian Jessie GNU/Linux, a Seagate GoFlex Home caddy with an ARM5-compatible processor running GNU/Linux, and a Windows PC.

Here is a command line for a Windows machine that downconverts all files of a certain extension in a directory, and burns timecode onto them, along with the filename. At the moment, the timecode just starts at zero for each clip: it is no trouble to write an extra routine to read any embedded timecode and use that instead. The command also slaps a simple autolevel on the soundtracks (because this is for off-line logging) and also adds a slow-acting video AGC to make shots palatable if they need severe grading. The font I use looks clear on screen: it is a free font, downloadable from a number of sources. You could use arial.ttf instead, because it is already on all Windows machines.

In case you’re not familiar with command lines, the backslashes are escape characters, that rob the following character of any special meaning. For example, a colon : has a special meaning to FFmpeg, but preceding it with a backslash, \:, causes the colon to be treated as an ordinary printable character.

This line is for 25fps material, reducing the footage the size 512×288.

ffmpeg -i <VIDEOFILE> -n -acodec libfdk_aac -b:a 40k ^
-profile:a aac_he_v2 -vcodec libx264 -crf 22 ^
-vf "yadif, colormatrix=bt709:bt601, pp=al, scale=512:288, smartblur=1.0:-1.0, drawtext=fontfile='c\:\\windows\\fonts\\LiberationSans-Bold.ttf':text='<VIDEOFILE>\ \ \ \ \ \ ':x=120:y=h-lh-1:fontsize=16:fontcolor=white:shadowcolor=black:shadowx=1:shadowy=1:timecode='00\:00\:00\:00':timecode_rate=25" ^
-x264opts colorprim=bt470bg:fullrange=off:transfer=bt470bg:colormatrix=bt470bg ^
-af "compand=0.0|0.0:0.8|0.8:-90/-40|0/0:6:0:-30:0" ^
<VIDEOFILE>.mp4

 

Here’s a step-by-step explanation of this command.

ffmpeg
FFmpeg command name
-i <VIDEOFILE>
Use your VIDEOFILE as input
-n
Overwrite existing files without question. I use this because the command is run from a FOR…DO loop in a batch file, and may be making revisions of many earlier files.
-acodec libfdk_aac -b:a 40k -profile:a aac_he_v2
Choose Fraunhofer’s AAC codec for audio, instruct the coder to use the HEAAC V2 flavour of the codec, and use 40kbit/s as the bitrate. AAC is the successor codec to MP3, used very widely in applications such as iTunes; HE means the “High Efficiency” version of the codec, which uses spectral band replication to code the upper frequencies of its input; and Version 2 adds a more efficient form of representing the difference between two stereo channels. The Fraunhofer codec is the best codec in the market, and its source code has been released primarily for use in Android development. However, its licence is not compatible with FFmpeg’s licence, and so it must be compiled into FFmpeg by hand, as I do.
-vcodec libx264 -crf 22
Use x264 as the video codec. This open source project is widely regarded as the most accurate H.264 codec. The coder is instructed to use a constant quality, represented by the “constant rate factor” or crf parameter. 22 is a fair trade-off between bandwidth and quality for our purposes, and is suitable for distribution over a LAN or good ADSL line.
-vf "yadif, colormatrix=bt709:bt601, pp=al, scale=512:288, smartblur=1.0:-1.0, drawtext=fontfile='c\:\\windows\\fonts\\LiberationSans-Bold.ttf':text='<VIDEOFILE>\ \ \ \ \ \ ':x=120:y=h-lh-1:fontsize=16:fontcolor=white:shadowcolor=black:shadowx=1:shadowy=1:timecode='00\:00\:00\:00':timecode_rate=25"

This is a video filter chain. It does several jobs. In order, they are:

  1. De-interlace the video. My incoming source is interlaced, but the eventual film, and web, destinations demand progressive-scan video
  2. Change the colour matrix from the HD standard to the SD (and below) standard. Many external sources will confirm how YUV representations of real-world colour pictures are calculated, and how the international standards differ in this representation between high-definition and standard-definition pictures.
  3. Add the auto-level filter from the post-production-processing library (libpostproc). This is a simple, slow-acting automatic gain control for the video. It goes before the scaling process in case of overshoots caused by the change in size or any sharpening.
  4. Scaling then takes place: a raster size of 512 x 288 gives sufficient detail for logging, but does not eat up too much bandwidth after coding
  5. The smartblur filter is a semi-intelligent edge-detection algorithm that, in this case, is asked to work on neighbouring pixels, and sharpen them. The negative number literally means “the opposite of blur”
  6. The drawtext filter writes on the video. This command (with appropriate escape characters):
    1. chooses a font by pointing to its filename;
    2. colours it white;
    3. positions the line of text appropriately (“lh” means “line height”);
    4. adds a black shadow for clarity;
    5. includes the filename;
    6. tells the timecode counter where to start (timecode display is implicit when a start frame is given);
    7. instructs the counter to count 25 frames per second.
  7. The filter chain ends here. Its output is now fed to the x264 coder.

-x264opts colorprim=bt470bg:fullrange=off:transfer=bt470bg:colormatrix=bt470bg

These are private options for the x264 coder.

  • The three options specifying “bt470bg” instruct the coder exactly how to interpret, and tell the decoder how to interpret, the conversion from YUV to RGB for display. In this case, I have chosen “ITU Recommendation BT.470 systems G and R”, the standard colour encoding for PAL and SECAM standard definition video. I specify this exactly because some displays assume colour matrices wrongly if they are not explicitly given instructions. Some others get it wrong anyway, but we must try.
  • The fullrange instruction tells the x264 coder that the incoming video is at studio levels (16 <= Y <= 235), and the coder includes this information in its instructions to the decoder and display. Again, it ought to be assumed that YUV-encoded video is already at studio levels, but sometimes decoders get this wrong.
-af "compand=0.0|0.0:0.8|0.8:-90/-40|0/0:6:0:-30:0"
An audio filter is described here.
compand
is a general audio level alteration process.
0.0|0.0
These two zeroes describe the channel attack times for the level-detection algorithm (0.0 seconds, instant)
0.8|0.8
These are the decay time constants for the stereo channels (0.8s)
-90/-40|0/0
A simple curve for gain amplification is described: raise sounds at -90dB up to -40dB, then alter levels proportionally until 0dB (full scale) remains at 0dB
6
This defines the softness of the knee down at -40dB: the curve is 6dB wide
0
This zero instructs the filter to apply 0dB gain make-up
-30
-30dB is the initial volume that the filter should assume, to avoid very loud audio at the start of each file
0
This final zero instructs the filter to act without delaying the side-chain, thus disabling its look-ahead function. This has been included for simplicity’s sake: I didn’t have much time to tweak this figure.
<FILENAME>
The output filename: same as the input filename but with the extension .mp4

The files this command line produces vary in bit-rate between 2Mbit/s and 500kbit/s, suitable for low quality LAN or Wi-fi use. A further shell batch command (this time run on my little ARM5 Seagate GoFlex Home) further downconverts them to around 150kbit/s, suitable for ADSL streaming over my slow line.

This may seem a complex command, but it does a lot of time-saving work in a single pass.

Making HTML5 AAC audio work

Web authors may be leaping to use the simple media embedding that HTML5 offers them: but there is a problem generated by the strictness of Microsoft’s Internet Explorer 9. It is particularly fussy, far more so than Google Chrome, about the mime-type of the media file.

As an example, a web page that includes this code:

<audio src="http://johnwarburton.net/aria.mp4" controls="true">

won’t play on Internet Explorer 9 when the file is served by an off-the-shelf Apache server running on Red Hat or Centos. The expected audio controls are replaced by a red cross. However, IE9 has the tools to analyse the problem (and, for that matter, so does Google Chrome, though the audio played first time). By pressing F12 and selecting the Console tab, one may query the Document Object Model for the error held by the audio object:

document.getElementsByTagName("audio")[0].error.code

One may need to adjust the index following ‘audio’ if the tag in question is not the first <audio> tag on the page. The error codes returned mean this:

  • MEDIA_ERR_ABORTED : 1
    The fetching process for the media resource was aborted by the user.
  • MEDIA_ERR_DECODE : 3
    An error has occurred in the decoding of the media resource, after the resource was established to be usable.
  • MEDIA_ERR_NETWORK : 2
    A network error has caused the user agent to stop fetching the media resource, after the resource was established to be usable
  • MEDIA_ERR_SRC_NOT_SUPPORTED : 4
    The media resource specified by the ‘src’ parameter was not usable.
The stock Apache server on Red Hat or Centos generates error 4 on an mp4 AAC audio file, and further investigation, again entirely within Internet Explorer 9, reveals the problem to be the server not sending the mime-type header which IE9 expects. Other browsers look at the file itself, but not IE9. Again, within the F12 debugging window, one may select the ‘Network’ tab, click the ‘Start Capture’ button, and reload the page to see the mime-types actually sent by the server. If the audio file doesn’t have type ‘audio/mp4’, the server’s mime-types file needs to be reconfigured to send it.
On the server, one needs to add this line to mime-types:
audio/mp4                       mp4

then restart the Apache server

service httpd restart

This cured the problem here.

Acknowledgement for this information is gladly given to the following blog from Microsoft: http://blogs.msdn.com/b/thebeebs/archive/2011/07/20/html5-video-not-working-in-ie9-some-tips-to-debug.aspx

 

Fit to print?

Love it or hate it, the Daily Mail has a huge readership both in print and on its website. So aware of its worldwide audience is the Mail that its website, when viewed within the United States of America, produces a whole new tranche of sensationalist, audience-winning and, yes, cleverly-written stories aimed specifically at that country’s population. One need only take a short hop across the border to Canada to type the same address in one’s browser, then to be greeted once more by the familiar UK front page.

Paul Dacre, the editor-in-chief of the Daily Mail, spoke passionately on Wednesday morning about the progress of press reporting standards in the UK, and also awakened the spectre of a straitjacketed press should Parliament vote for compulsory press regulation:

News, let me remind you, is often something that someone – the rich, the powerful, the privileged – doesn’t want printed…. Indeed, I would argue that Britain’s commercially viable free press – because it is in hock to nobody – is the only really free media in this country. Over regulate that press and you put democracy itself in peril.

Paul Dacre’s speech is on-line here, at The Guardian’s website.