fdk-aac-dabplus Package

This package contains an DAB+ encoder that uses the standalone library of the Fraunhofer FDK AAC code from Android, patched for 960-transform to do DAB+ broadcast encoding.

The main tool is the dabplus-enc encoder, which can read audio from a file (raw or wav), from an ALSA source or from JACK, and encode to a file, a pipe, or to a ZeroMQ output compatible with ODR-DabMux.

The ALSA input supports experimental sound card clock drift compensation, that can compensate for imprecise sound card clocks.

The JACK input does not automatically connect to anything. The encoder runs at the rate defined by the system clock, and therefore sound card clock drift compensation is also used.

dabplus-enc includes support for DAB MOT Slideshow and DLS, contributed by CSP.

To encode DLS and Slideshow data, the mot-encoder tool reads images from a folder and DLS text from a file, and generates the PAD data for the encoder.

For detailed usage, see the usage screen of the different tools.

More information is available on the Opendigitalradio wiki

How to build

Requirements:

boost-thread and boost-system
ImageMagick magickwand (for MOT slideshow)
The alsa libraries (libasound2)
Download and install libfec from https://github.com/Opendigitalradio/ka9q-fec
Download and install ZeroMQ from http://download.zeromq.org/zeromq-4.0.3.tar.gz
JACK audio connection kit (optional)

This package:

git clone https://github.com/mpbraendli/fdk-aac-dabplus.git
cd fdk-aac-dabplus
./bootstrap
./configure
make
sudo make install

If you want to use the JACK input, please use

./configure --enable-jack

See the possible scenarios below on how to use the tools
use mot-encoder to encode images into MOT Slideshow

How to use

We assume that you have a ODR-DabMux configured for an ZeroMQ input on port 9000.

ALSASRC="default"
DST="tcp://yourserver:9000"
BITRATE=64

AAC encoder configuration

By default, when not overridden by the --aaclc, --sbr or --ps options, the encoder is configured according to bitrate and number of channels.

If only one channel is used, SBR (Spectral-Band Replication, also called HE-AAC) is enabled up to 64kbps. AAC-LC is used for higher bitrates.

If two channels are used, PS (Parametric Stereo, also called HE-AAC v2) is enabled up to 48kbps. Between 56kbps and 80kbps, SBR is enabled. 88kbps and higher are using AAC-LC.

Scenario 1

Live Stream from ALSA sound card at 32kHz, with ZMQ output for ODR-DabMux:

dabplus-enc -d $ALSASRC -c 2 -r 32000 -b $BITRATE -o $DST -l

To enable sound card drift compensation, add the option -D:

dabplus-enc -d $ALSASRC -c 2 -r 32000 -b $BITRATE -o $DST -D -l

You might see U and O appearing on the terminal. They correspond to audio underruns and overruns that happen due to the different speeds at which the audio is captured from the soundcard, and encoded into HE-AACv2.

High occurrence of these will lead to audible artifacts.

Scenario 2

Play some local audio source from a file, with ZMQ output for ODR-DabMux. The problem with playing a file is that dabplus-enc cannot directly be used, because ODR-DabMux does not back-pressure the encoder, which will therefore encode much faster than realtime.

While this issue is sorted out, the following trick is a very flexible solution: use the alsa virtual loop soundcard snd-aloop in the following way:

modprobe snd-aloop

This creates a new audio card (usually 'hw:1' but have a look at /proc/asound/card to be sure) that can then be used for the alsa encoder.

./dabplus-enc -d hw:1 -c 2 -r 32000 -b 64 -o $DST -l

Then, you can use any media player that has an alsa output to play whatever source it supports:

cd your/preferred/music
mplayer -ao alsa:device=hw=1.1 -srate 32000 -shuffle *

Important: you must specify the correct sample rate on both "sides" of the virtual sound card.

Scenario 3

Live Stream encoding and preparing for DAB muxer, with ZMQ output, at 32kHz, using sox. This illustrates the fifo input over standard input of dabplus-enc.

sox -t alsa $ALSASRC -b 16 -t raw - rate 32k channels 2 | \
dabplus-enc -r 32000 -l \
-i - -b $BITRATE -f raw -o $DST -p 53

The -p 53 sets the padlen, compatible with the default mot-encoder setting. mot-encoder needs to be given the same value for this option.

Scenario 4

Live Stream encoding and preparing for DAB muxer, with FIFO to odr-dabmux, 48kHz, using arecord.

arecord -t raw -f S16_LE -c 2 -r 48000 -D plughw:CARD=Loopback,DEV=0,SUBDEV=0 | \
dabplus-enc -l -b $BITRATE -f raw -c 2 -r 48000 -i /dev/stdin -o - | \
mbuffer -q -m 10k -P 100 -s 360 > station1.fifo

Here we are using the ALSA plughw feature.

Scenario 5

Live Stream resampling (to 32KHz) and encoding from FIFO and preparing for DAB muxer, with FIFO to odr-dabmux using mplayer. If there are no data in FIFO, encoder generates silence.

mplayer -quiet -loop 0 -af resample=32000:nowaveheader,format=s16le,channels=2 -ao pcm:file=/tmp/aac.fifo:fast <FILE/URL> &
dabplus-enc -l -f raw --fifo-silence -i /tmp/aac.fifo -r 32000 -c 2 -b 72 -o /dev/stdout \
mbuffer -q -m 10k -P 100 -s 1080 > station1.fifo

Note: Do not use /dev/stdout for pcm oputput in mplayer. Mplayer log messages on stdout.

Scenario 6

Wave file encoding, for non-realtime processing

dabplus-enc -b $BITRATE -i wave_file.wav -o station1.dabp

Scenario 7

JACK input: Instead of -i (file input) or -d (ALSA input), use -j name, where name specifies the JACK name for the encoder:

dabplus-enc -j myenc -l -b $BITRATE -f raw -o $DST

The samplerate of the JACK server should be 32kHz or 48kHz.

Return values

dabplus-enc returns:

0 if it encoded the whole input file
1 if some options were not understood, or encoder initialisation failed
2 if the silence timeout was reached
3 if the AAC encoder failed
4 it the ZeroMQ send failed
5 if the ALSA input had a fault

Usage of MOT Slideshow and DLS

mot-encoder reads images from the specified folder, and generates the PAD data for the encoder. This is communicated through a fifo to the encoder. It also reads DLS from a file, and includes this information in the PAD.

It can read all file formats supported by ImageMagick, and by default resizes them to 320x240 pixels, and compresses them as JPEG. If the input file is already a JPEG file of the correct size, and smaller than 50kB, it is sent without further compression. If the input file is a PNG that satisfies the same criteria, it is transmitted as PNG without any recompression.

dabplus-enc can insert the PAD data from mot-encoder into the bitstream. The mp2 encoder toolame-dab can also read mot-encoder data.

This is an ongoing development. Make sure you use the same pad length option for mot-encoder and the audio encoder. Only some pad lengths are supported, please see mot-encoder's help. Only pad lengths 34, 42 and 58 seem to be working with some receivers, 23 and 26 appear to be broken.

Known Limitations

mot-encoder encodes slides in a 10 second interval, which is not linked to the rate at which the encoder reads the PAD data. It also doesn't prioritise DLS transmission over Slides.

Some receivers did not decode audio anymore between v0.3.0 and v0.5.0, because of a change implemented to get PAD to work. The change was subsequently reverted in v0.5.1 because it was deemed essential that audio decoding works on all receivers. Work to get both functional audio and PAD on all receivers is ongoing.

Version 0.4.0 of the encoder changed the ZeroMQ framing. It will only work with ODR-DabMux v0.7.0 and later.