Transcript Progress Presentation

Final Year Project
Progress January 2007
By Daire O’Neill 4EE
Overview of progress to date
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Have installed Ubuntu and become familiar with the
basics: using shell commands,configuring for internet,
installing tarballs and rpm packages. Have also used
open source audio programs successfully, implementing
JACK and ALSA
Have done research and developed ideas to solve the
issue of synchronisation of audio files
Have investigated using Python for writing additional
scripts, for open source programs
Currently setting up basic FTP client/server network,
for transfer of audio files
Synchronisation of audio files
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An issue that was previously highlighted was the
accurate synchronisation of audio files
Due to variations in clock speeds on different
computers, an audio file recorded on a PC may
not play back with precisely the same speed on a
different PC
The difference is small enough not to be noticed
for normal playback, but is an issue when
synchronising multiple files for longer playback
Synchronisation of audio files
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One solution may be to provide a front-end
hardware device for capture of audio into the
digital domain
Provided that the interface is manufactured
within sufficiently strict limits, the timing
characteristics for 2 interfaces should be very
similar for a given audio signal
Hardware interface
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An interface such as this may comprise of a high
quality ADC chip, microphone and line level
inputs, volume control and level indicator
By connecting with a PC via a USB (or Firewire)
interface, it works with the software to record an
analogue signal and transfer the digital stream to
the hard disk of the PC for storage
To complete feasibility report on such an
interface…
Synchronisation of audio files
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Once accurate capture of the audio signal has
been achieved, synchronised playback of various
audio files is possible even with different PCs
and clock speeds.
This is because the playback of each audio file is
always driven from the same clock, so the
different files will playback in sync relative to
each other.
Synchronisation of audio files
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A file that was downloaded from the internet
(where it was recorded and uploaded by a
different user) may play back on your machine at
a slightly slower speed than which it was
originally recorded
But, it will be synchronised with the measure ( of
the playback software, provided that the original
digital capture was accurate.
To explain this in more detail…
Example: Synchronisation
Capture
Recording Software
Audio
Sample
Sample Data
Sample No.
200
Voltage
1.87654 V
Measure
3.255 beats
Capture
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As well as recording the value of the audio
signal (voltage) for each sample instant, the value
of the measure (number of beats) for that
sample is also stored:
Sample No.
200
Voltage
1.87654 V
Measure
3.255 beats
Playback
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In most audio playback devices, a collection of
audio samples is stored in a buffer, and are
reproduced (converted back to analogue) at the
same rate at which they were captured,
e.g 44.1 kHz for most digital audio
Variations in clock speed, (i.e. the consistency of
the 44.1 khz rate) produce timing problems
when playing back multiple samples for
synchronisation
Solution?
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A different approach is required for successful
synchronisation…
Rather than letting the clock “just run”, samples should
be played back and aligned according to their position
in the ‘measure’ at which they were captured
Once the capture is accurate (achieved by using
standard interfaces), then any sample captured in this
way should play back accurately on any PC running the
same software
Playback
Measure 3.249 3.250 3.251 3.252 3.253 3.254 3.255 3.256 3.257
Signal
Sample Data
Sample No.
200
Voltage
1.87654 V
Measure
3.255 beats
Synchronisation of audio files
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So, regardless of the variation in clock speed on
playback, accurate timing is provided due to each
sample being ‘tagged’ with the value of the measure
(beats) at each sample instant-this will not vary between
machines running the same software
A different clock speed will result in the overall
collection of samples being played faster or slower than
normal (different sampling rate), but all samples will be
synchronised relative to each other
Another example
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Another example of using a variable sampling
rate can be shown with the issue of ‘network
speakers’-speakers that have an Ethernet or
wireless connection to receive a digital stream,
which is then converted to analogue and output
as sound
A significant delay (greater than 30 ms) between
these 2 audio streams (left and right channels)
can produce an audible echo, or Haas effect
Synchronisation of audio files
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The solution is to use a dynamic sampling rate
for playback, that changes (speeds audio up or
down) to compensate for the difference in
timing between the audio files
Networking
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I am currently in the process of setting up a
basic FTP client/server system in Ubuntu.
Vsftpd is the software I plan to use to
implement this
File Transfer Protocol (FTP)
Questions or
recommendations?