Audio Quality comparison of PCM, DAB, DAB+, FM and AM

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DAB Radio The purpose of this article is to give a reasonable audio quality comparison of uncompressed PCM audio, DAB, DAB+, FM and AM. The broadcast formats were simulated in the manner described in the following methodology – whilst not quite as good as using full transmission modulation and demodulation chains, it is hoped that these simulations are “good enough” for comparison purposes.

Photo by bods

Audio Comparison Methodology

First of all a sample of music was ripped from CD using dB PowerAmp software and saved as 16bit PCM Wav file – ie uncompressed audio. Then copies of this file were processed as following to simulate the various transmission formats:

  • DAB – file converted to MP2 with three formats using dB PowerAmp software. Formats selected were 80kbps Stereo, 80kbps Mono and 128kbps Stereo. These files were then converted back to PCM wav files for compatability reasons
  • DAB+ – file converted to AAC HE v2 (which we call AAC+). Formats selected were 56kbps Stereo, 32kbps Stereo, 16kbps Stereo and 12kbps Stereo. These files were then converted back to PCM wav files for compatability reasons
  • FM – File was band-limited to 15kHz. Another version also had audio processing applied using the “Broadcast” preset in the multi-band audio processor function of Adobe Audition 3.0
  • AM – File was processed using AM simulator preset on Reaper software

Associated Broadcast Consultants opinion is that 80kbps Stereo is inadequate quality, but becomes bearable at 80kbps Mono, albeit with total loss of stereo image. 128kbps MP2 seems to give a good approximation of FM quality both audibly and on the spectral charts. The AAC+ codec (used in DAB+) is remarkably good at lower bit rates – still acceptable at 32kbps, but becoming a bit “YouTubey” at 16kbps and horrible at 12kbps.  However we feel it still has “something missing” at 56kbps which is the highest bitrate possible for that codec (above that it becomes normal AAC).  Unsurprisingly AM sounds the worst, but this is probably exagerrated by the lack of proper AM audio processing which would make a professional AM broadcast sound much better.

Audio Quality Comparison files

The audio samples have been combined into one 6 minute comparison file. Ideally you should listen to the WAV file (123MB), but if you are in a rush and/or have poor broadband, the 320kbps MP3 file still clearly shows the difference in audio quality:-

WAV File (large – 123MB)

MP3 File (smaller – 14MB)

(Just click to play or Right click and file save or save target as to download the files)

Spectral Charts

To supplement listening tests, we also did screen captures of the audio spectrum of each file. This clearly shows the band limiting of the FM file. It also shows a surprisingly heavy band-limiting in the MP2 80kbps Stereo file (almost to AM quality). In contrast the 80kpbs mono MP2 file has much better audio spectrum, and this is clearly audible on the test file. The 12kbps AAC+ picture is interesting.  It appears to show aliasing – that is audio components between 0-4kHz are repeated from 4-8kHz and 8-12kHz.  This could be why it sounds so bad! It could be avoided by bandlimiting the audio to 4Khz before encoding.  We are not sure if this is a problem with the codec used, or a limitation of the AAC HE v2 codec specification.

Original Audio PCM WAV MP2 80kbps Stereo MP2 80kbps Mono FM Unprocessed FM Processed DAB+ (AAC+ or AAC HE V2) – 56kbps DAB+ (AAC+ or AAC HE V2) – 32kbps DAB+ (AAC+ or AAC HE V2) – 16kbps DAB+ (AAC+ or AAC HE V2) – 12kbps AM Europe (4.5kHz)

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8 thoughts on “Audio Quality comparison of PCM, DAB, DAB+, FM and AM

  1. radiohead Post author

    We agree technically – but practically we believe WAV is more universally supported than FLAC – and that aspect is important on any website

  2. Christian Schubert

    Unfortunately I cannot download the files due to low bandwith connection. But I want to add some points.

    The thin vertical lines in your spectra seem to be the result of clipping of the audio. If you use a original wave file going to 0 dBFS (digital fullscale) most of the time (like almost every today’s pop and rock music production does) and do any kind of psychoacoustic coding you will get artefacts. These artefacts can increase or decrease every single sample when played back. In average this will not change the audio level, but every single sample that is close to 0 dBFS can clip due to these artefacts. The lower the bitrate the higher the “overshoot”. Typical values for MP2 can be some dB when coded to 128 oder 192 kbps. Above 256 kbps the overshoots will become less important.

    So if you do coding experiments please do not use wave files going to 0 dBFS. Attenuate the file prior to coding to -6 dBFS in peak level, then you will be safe.

    According to years of listening experience MP2 (“the classic” DAB coding and also used on DVB-C, DVB-S and DVB-T) can deliver following results with a good codec:

    96 kbps MP2 mono – very good for portable use, good for HiFi use
    128 kbps MP2 joint stereo – audible artefacts in portable use, bad for HiFi use
    160 kbps MP2 joint stereo – audible artefacts in portable use, but sometimes acceptable, bad for HiFi use
    192 kbps MP2 joint stereo or linear stereo – no audible artefacts in portable use, good for HiFi use
    256 kbps MP2 linear stereo – excellent in portable use, very good for HiFi use
    320 or 384 kbps MP2 linear stereo – excellent in portable use, very good – excellent for HiFi use

    However, quality at a given bitrate depends on codec settings and codec quality. There was a codec called RE660 manufactured by Barco / Scientific Atlanta in the 90s that was widely used for satellite transmission. The RE660 had a bad implementation of joint stereo mode resulting in a bad audio quality at 192 kbps (!). In linear stereo this codec was absolutely ok.

    Another important pint is sound processing in radio stations. These terribly distorted and “discolored” sounds will result in a lower quality of the MP2 or AAC sound.

    Germany’s public broadcasters use 320 kbps on satellite, you can hear the high quality in the cultural programmes that are transmitted without heavy sound processing. 384 kbps is used inside the broadcast houses to store audio files ready for transmission.

    The new DAB+ codec AAC is more sophisticated. The quality depends on some settings.

    A typical setting for stereo ist 96 kbps HE-AAC (with spectral band replication SBR). It delivers bright treble – but this is artificial above approx. 11 kHz. It sounds very clear for the 1st moment but when you listen longer you will hear that this treble is annoying. It feels like it does not belong to the original audio (and this is the truth). So old recordings from the 60s will have a extremely brilliant treble (which of course is not the truth) and speech or the voice of a (female) singer seems not to belong to the original recording.

    You can use LC-AAC instead at 96 kbps, the LC codec is coding the complete frequency range and not cutting at 11 kHz and adding artificial treble instead. The result ist slightly better – but only on receivers that are capable of good-quality handling of the LC codec. Some receivers produce at 96 kbps LC-AAC a terrible sound that sounds like 32 kbps or so. Due to this some broadcasters in Germany decided to go back to the HE codec with artificial treble.

    LC-AAC is delivering high quality on all receivers at bitrates of 128 kbps or higher. This is sometimes used for cultural programmes.

    The typical “low cost” setting 72 or 88 kbps HE-AAC sounds terrible and any good FM reception will sound better. This is particularly the case when the “net” bitrate is significantly lower than the “gross” bitrate because of additional transmitted pictures etc.

    My private preference:

    MP2 above 256 or 320 kbps > MP2 at good coded 192 kbps = good FM reception > MP2 at 160 kbps joint stereo = LC-AAC at 144 or 128 kbps > LC-AAC at 128 kbps > HE-AAC at 96 kbps . All with less bitrate is not acceptable.

  3. radiohead Post author

    Some great points Christian – thanks for taking the time to share them with us. To avoid clipping I generally normalise to 92% – maybe I need to reduce it still further!
    Let’s hope that “Small Scale DAB” in the UK leads to sufficient DAB capacity to implement the kind of bit rates that you recommend! Our ears will benefit!

    1. Gagarin Miljkovich

      Why do you think “Small Scale DAB” isn’t used by BBC and the B.I.G commercial broadcasters?

      Answer: To get a rugged DAB multiplex you need a really heavy RF-signal. And when it is a heavy RF-signal, with 500-20.000 watts, it isn’t anymore “Small Scale DAB”. BBC and the B.I.G commercial broadcasters know that.

      With lower RF power you don’t get any rugged signal. It’s a joke…

  4. radiohead Post author

    Thanks Cees. But I don’t think Opus is implemented in the DAB+ standard is it? Maybe one for DAB++!

    1. Gagarin Miljkovich

      Opus is a very good audio codec.

      When I suggested around two years ago that Opus should be added to the international digital radio standards like DRM and DRM+, the guys behind todays AAC critisized it. They said that the patents around Opus are disputed.

      Surely they would say that. The royalty on DAB+/DRM+-codec is generating a heavy income.

      “France Telecom claims patent on Opus, gets rebuffed, with technical analysis of claims


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