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LPCNet for FreeDV

Experimental version of LPCNet that has been used to develop FreeDV 2020 - a HF radio Digital Voice mode for over the air experimentation with Neural Net speech coding. Possibly the first use of Neural Net speech coding in real world operation.

Quickstart

$ cd ~
$ git clone https://github.com/drowe67/LPCNet.git
$ cd LPCNet && mkdir build_linux && cd build_linux
$ cmake ..
$ make

Unquantised LPCNet:

$ cd ~/LPCNet/build_linux/src
$ sox ../../wav/wia.wav -t raw -r 16000 - | ./dump_data --c2pitch --test - - | ./test_lpcnet - - | aplay -f S16_LE -r 16000

LPCNet at 1733 bits/s using direct-split quantiser:

sox ../../wav/wia.wav -t raw -r 16000 - | ./lpcnet_enc -s | ./lpcnet_dec -s | aplay -f S16_LE -r 16000

Manually Selecting SIMD Technology

Cmake will select the fastest SIMD available (AVX/SSSE/None), however you can manually select e.g.:

make -DDISABLE_CPU_OPTIMIZATION=ON -DSSE=ON ..

CTests

$ cd ~/LPCNet/build_linux
$ ctest

Note, due to precision/library issues several tests (1-3) will only pass on some machines.

Building Debian packages

To build Debian packages, simply run the "cpack" command after running "make". This will generate the following packages:

  • lpcnet: Contains the .so and .a files for linking/executing applications dependent on LPCNet.
  • lpcnet-dev: Contains the header files for development using LPCNet.
  • lpcnet-tools: Contains tools for use with LPCNet.

Once generated, they can be installed with "dpkg -i".

Reading Further

  1. Original LPCNet Repo with more instructions and background
  2. LPCNet: DSP-Boosted Neural Speech Synthesis
  3. Sample model files

Credits

Thanks Jean-Marc Valin for making LPCNet available, and Richard for the CMake build system.

Cross Compiling for Windows

This code has been cross compiled to Windows using Fedora Linux 30, see the freedv-gui README.md, and build_windows.sh script.

Speech Material for Training

Suitable training material can be obtained from the McGill University Telecommunications & Signal Processing Laboratory. Download the ISO and extract the 16k-LP7 directory, the src/concat.sh script can be used to generate a headerless file of training samples.

cd 16k-LP7
sh /path/to/LPCNet/src/concat.sh

Quantiser Experiments

The quantiser files used for these experiments (pred_v2.tgz and split.tgz) are here

Exploring Features

Install GNU Octave (if thats your thing).

Extract a feature file, fire up Octave, and mesh plot the 18 cepstrals for the first 100 frame (1 second):

$ ./dump_data --test speech_orig_16k.s16 speech_orig_16k_features.f32
$ cd src
$ octave --no-gui
octave:3> f=load_f32("../speech_orig_16k_features.f32",55);
nrows: 1080
octave:4> mesh(f(1:100,1:18))

Uniform Quantisation

Listen to the effects of 4dB step uniform quantisation on cepstrals:

$ cat ~/Downloads/wia.wav | ./dump_data --test - - | ./quant_feat -u 4 | ./test_lpcnet - - | play -q -r 16000 -s -2 -t raw -

This lets us listen to the effect of quantisation error. Once we think it sounds OK, we can compute the variance (average squared quantiser error). A 4dB step size means the error PDF is uniform in the range of -2 to +2 dB. A uniform PDF has variance of (b-a)^2/12, so (2--2)^2/12 = 1.33 dB^2. We can then try to design a quantiser (e.g. multi-stage VQ) to achieve that variance.

Training a Predictive VQ

Clone and build codec2:

$ git clone https://github.com/drowe67/codec2.git
$ cd codec2 && mkdir build_linux && cd build_linux && cmake ../ && sudo make install

In train_pred2.sh, adjust PATH for the location of codec2-dev on your machine.

Generate 5E6 vectors using the -train option on dump_data to apply a bunch of different filters, then run the predictive VQ training script

$ cd LPCNet
$ ./dump_data --train all_speech.s16 all_speech_features_5e6.f32 /dev/null
$ ./train_pred2.sh

Mbest VQ search

Keeps M best candidates after each stage:

cat ~/Downloads/speech_orig_16k.s16 | ./dump_data --test - - | ./quant_feat --mbest 5 -q pred2_stage1.f32,pred2_stage2.f32,pred2_stage3.f32 > /dev/null

In this example, the VQ error variance was reduced from 2.68 to 2.28 dB^2 (I think equivalent to 3 bits), and the number of outliers >2dB reduced from 15% to 10%.

Streaming of WIA broadcast material

Interesting mix of speakers and recording conditions, some not so great microphones. Faster speech than the training material.

Basic unquantised LPCNet model:

sox -r 16000 ~/Downloads/wianews-2019-01-20.s16 -t raw - trim 200 | ./dump_data --c2pitch --test - - | ./test_lpcnet - - | aplay -f S16_LE -r 16000

Fully quantised at (44+8)/0.03 = 1733 bits/s:

sox -r 16000 ~/Downloads/wianews-2019-01-20.s16 -t raw - trim 200 | ./dump_data --c2pitch --test - - | ./quant_feat -g 0.25 -o 6 -d 3 -w --mbest 5 -q pred_v2_stage1.f32,pred_v2_stage2.f32,pred_v2_stage3.f32,pred_v2_stage4.f32 | ./test_lpcnet - - | aplay -f S16_LE -r 16000

Fully quantised encoder/decoder programs

Same thing as above with quantisation code packaged up into library functions. Between quant_enc and quant_dec are 52 bit frames every 30ms:

cat ~/Downloads/speech_orig_16k.s16 | ./dump_data --c2pitch --test - - | ./quant_enc | ./quant_dec | ./test_lpcnet - - | aplay -f S16_LE -r 16000

Same thing with everything integrated into stand alone encoder and decoder programs:

cat ~/Downloads/speech_orig_16k.s16 | ./lpcnet_enc | ./lpcnet_dec | aplay -f S16_LE -r 16000

The bit stream interface is 1 bit/char, as I find that convenient for my digital voice over radio experiments. The decimation rate, number of VQ stages, and a few other parameters can be set as command line options, for example 20ms frame rate, 3 stage VQ (2050 bits/s):

cat ~/Downloads/speech_orig_16k.s16 | ./lpcnet_enc -d 2 -n 3 | ./lpcnet_dec -d 2 -n 3 | aplay -f S16_LE -r 16000

You'll need the same set of parameters for the encoder as decoder.

Useful additions would be:

  1. Run time loading of .h5 NN models.
  2. A --packed option to pack the quantised bits tightly, which would make the programs useful for storage applications.

Direct Split VQ

Four stage VQ of log magnitudes (Ly), 11 bits (2048 entries) per stage, First 3 stages 18 elements wide; final stage 12 elements wide. During training this acheived similar variance to 4 stage predictive quantiser (measured on 12 bands). Same bit rate, but direct quantisation means more robust to bit errors and especially packet loss.

sox ~/Desktop/deep/quant/wia.wav -t raw - | ./dump_data --c2pitch --test - - | ./quant_feat -d 3 -i -p 0 --mbest 5 -q split_stage1.f32,split_stage2.f32,split_stage3.f32,split_stage4.f32 | ./test_lpcnet - - | aplay -f S16_LE -r 16000

Compare this to four stage predictive VQ of Cepstrals (DCT of Ly), 11 bits (2048 entries) per stage, 18 element wide vectors. We quantise the predictor output.

sox ~/Desktop/deep/quant/wia.wav -t raw -  | ./dump_data --c2pitch --test - - | ./quant_feat -d 3 -w --mbest 5 -q pred_v2_stage1.f32,pred_v2_stage2.f32,pred_v2_stage3.f32,pred_v2_stage4.f32 | ./test_lpcnet - - | aplay -f S16_LE -r 16000

Both are decimated by a factor of 3 (so 30ms update of parameters, 30*44=1733 bits/s).

Effect of Bit Errors

Random 1 Bit Error Rate (BER):

Predictive: sox wav/wia.wav -t raw -r 16000 - | ./lpcnet_enc | ./lpcnet_dec -b 0.01 | aplay -f S16_LE -r 16000

Direct-split: sox wav/wia.wav -t raw -r 16000 - | ./lpcnet_enc -s | ./lpcnet_dec -s -b 0.01 | aplay -f S16_LE -r 16000

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Experimental Neural Net speech coding for FreeDV

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