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Stuttering text-to-speech and how I Solved it

Table of Contents

Introduction

Currently the best solution for text-to-speech on the Raspberry Pi is the eSpeak software speech synthesiser.

The speech-dispatcher program can use other software speech synthesisers, but historically we have only used the speakup screen-reader so far and the best way to connect speakup to a text-to-speech engine is with the espeakup program, not using speech-dispatcher.

This page does not go into the intense debate that rages about the pros and cons of eSpeak. Some claim it sounds horrible. But because of it’s small size and huge language support it remains the best option.

History

All was well with eSpeak text-to-speech on the Raspberry Pi until approximately April 2013.
At about that time, a change was made to the ALSA driver to introduce real DMA.

Unfortunately, the above change broke eSpeak tts. After that time, it would stutter very, very badly within a few seconds of beginning to speak.

If that was not bad enough, a kernel oops regularly occurs when using the ALSA driver.

I established that this was what was happening when the console froze by connecting the Pi to another Linux machine via the console which runs on the UART available on the GPIO bus. Using this it is possible to see what happens when the kernel oops occurs and the debug information is sent out of the UART.

This is caused by the Video Core Hardware Interface Queue (VCHIQ) process passing the kernel a null pointer.

The VCHIQ process is responsible for queueing audio and video into the Graphics Processing Unit (GPU).

In efforts to get around this I tried a lot of things:

  • Using YASR instead of speakup in the console
  • Changing the niceness and the priority of the espeakup program
  • Using speech-dispatcher instead of espeakup to connect speakup to TTS
  • Many other tweaks

Nothing worked.

Solution

I think it was late in 2013 I learned about the OpenMAX library and specifically the Integration Layer library.

OpenMAX, hereinafter referred to as OMX, is a software system that seeks to standardise the interface to graphics and audio hardware on different platforms.

I believe it was created to make graphics and sound programming easier for the multitude of mobile devices and smart phones now in use.

In a standard Raspbian Raspberry Pi, there is a directory:

/opt/vc

This directory contains libraries and other code that relates to the video core, hence VC.

Note at this point that the term video core should not be mistaken for something that only refers to video, for the GPU also renders sound.

In the above directory I found examples of code that renders sound on the GPU.

So I set about learning how to interface to the OMX Integration Layer Client.

eSpeak Specifics

One of the great things about eSpeak is it’s ability to be used in a mode that will return pulse-code modulated (PCM) audio data to the calling program.

In this mode, a fragment of text is passed to eSpeak to be rendered into synthetic speech, and the converted PCM is returned in a callback function.

The PCM data can then be used however it is required, for example it can be written to a .wav file, some other file, or processed in some way and then passed to some mechanism to be played over the output device.

So, what I had to do was:

  1. Write code that would receive PCM data and queue it into the GPU via the OMX Integration Layer Client.
  2. Adapt espeakup to call eSpeak in it’s callback mode and link my OMXILC library with it.

So, I wrote the library. It took a long time and an even longer time to debug.

It taught me a lot about concurrency and the so called producer/consumer problem, in which different threads of execution either produce data, or consume it.

Concurrency can be defined as the need for different threads of execution, or even different processes, to have access to a common resource without “treading on each other’s toes”.

Then I forked the code of espeakup and created piespeakup.

piespeakup contains a callback function which then queues the text-to-speech audio returned from eSpeak into the GPU.

The OMX library I wrote contains a circular buffer which receives this data, which is constantly filled and drained in the classic producer/consumer sequence. piespeakup being the producer of PCM audio, and the OMX library passing the TTS audio to VCHIQ as the consumer.

Result, no involvement from the broken ALSA driver.

One of the hairiest problems I had to solve was one of latency. The time taken for the espeak program to render text into PCM data and return it to the calling program, and how this impacts on, in particular, the quality of small chunks of speech, at the beginning and end of each utterance. For a longertime I had it working but the speech was very severely clipped at the end of each utterance.

It is not commonly understood that espeak renders text passed to it in often very small portions rather than as, for example, a whole sentence in one go.

Hacker Public Radio Podcast

I did a podcast for Hacker Public Radio to demonstrate the fixed TTS audio. Click here to go to the HPR page and this podcast.

Installing Both Components

First you need to install espeak. If you are using Raspbian, that distro splits the espeak program from the library and you need to install it like this:

$ sudo apt-get install libespeak-dev

If you are using Arch:

$ sudo pacman -S espeak

Follow these instructions to install the two components which will give stutter-free console speech with SpeakUp.

In the instructions below I have given the version numbers of both components as 1.0.0, of course it may change, check the Web site before you start.

The OMX Library

This has been tested on both Raspbian and Arch Linux.

Follow these instructions, in each line the dollar sign represents the prompt. And note that Downloads starts with a capital ‘D’:

$ wget http://www.raspberryvi.org/Downloads/ilctts-1.0.0.tar.gz
$ tar zxf ilctts-1.0.0.tar.gz
$ cd ilctts-1.0.0
$ ./configure
$ make
$ sudo make install
$ sudo ldconfig

Now, to load the SpeakUp kernel modules either reboot before installing the next component or:

$ sudo modprobe speakup_soft

Installing piespeakup

Follow these steps. Again the dollar sign is the prompt:

$ wget http://www.raspberryvi.org/Downloads/piespeakup-1.0.0.tar.gz
	$ tar zxf piespeakup-1.0.0.tar.gz
		$ cd piespeakup-1.0.0
			$ ./configure
				$ make
					$ sudo make install

Note that in both instances of wget above the word Downloads has a capital ‘D’.

					At this point before we enable and start the 
					`piespeakup` service we have to make sure the 
					`speakup` kernel modules are loaded or the service 
					won't start.
					
					If you either rebooted after installing the OMX 
					library or manually instaled the kernel modules they 
					will be there.
					
					Check like this:
					
						$ lsmod | grep speakup
						
						You should see two modules, `speakup_soft` 
						and it's dependancy `speakup`.
						
						Now enable `piespeakup`:
						
							$ sudo systemctl enable piespeakup
							
							And start it:
							
$ sudo systemctl start piespeakup

Configuration

There is a file at:

/etc/systemd/system/piespeakup.conf

In this file is where we can change from the analogue audio jack (3.5mm socket) to HDMI.

The file contains the line:

ExecStart=/usr/local/piespeakup --device=local

Change it to:

ExecStart=/usr/local/piespeakup --device=hdmi

To switch to HDMI, after which you will need to restart the piespeakup service.

DO NOT remove the line above this which reads:

ExecStart=

Because the original ExecStart directive needs to be blanked before it is reset. I will add more configuration options at a later date.

When both the OMX library and piespeakup are installed, and when the speakup kernel modules are loaded correctly, the Pi should come up speaking when it is rebooted.

It is interesting to note that this console audio does NOT use the ALSA driver so it does not suffer from the classic problem for accessibility on the Linux desktop, which is that when the user logs into the desktop and speech-dispatcher is configured to use pulseaudio console audio is silenced because of some configuration of pulseaudio, a solution for which I have never seen.

Note also that there is currently a bug in the speech-dispatcher espeak module; sd_espeak which causes it to crash regularly, which makes it impossible to reliably use speech-dispatcher configured for ALSA.

This does not matter to use currently since the sd_espeak module will also stutter very badly. I need to write an OMX version os the sd_espeak module or an OMX audio driver for speech-dispatcher.