Design and implementation of Up-Mixing algorithms for Home.pptx
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On PANDA OMAP4430 targetboard
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Project by:
M.SAI KIRAN 10UQ1A0456V.SAI MANOGNYA 10UQ1A0495V.REKHA 10UQ1A0499P.MOUNIKA 10UQ1A0466
Btech 4thyear.
[Electronics &communication engineering].
Vignan institutes of technology and aeronautical
engineering,deshmukhi,A.P.
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Introduction.
Architecture of PANDA OMAP-4430 & CORTEX A-9.
Basics of embedded audio processing.
Post processing & Block processing.
Surround sounds & its applications.
Stereo to 5.1 channel conversion process byup-mixing technique.
Developing C code using FILE i/o operations andcompiling in a LINUX environment using GCCcompiler.
Integrate the code developed in C on Panda boardand test it with File I/O.
Conclusion.
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With the growing proliferation of multi channel audioin consumer electronics, there are situations wherethe number of channels of either the audio content orthe reproducing loud speakers is limited and it isnecessary to upmix or downmix the channels toimprove the listening experience.
In the upmixing process, the surround channels canbe created from the two-channel stereo inputs byusing two approaches:
1) uses the decorrelated part of the stereo inputs asthe surround channels.
2) the other approach produces the surroundchannels by simulating the reverberant sound field inthe background.
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The PandaBoardis a low-power, low-cost single-board computer developmentplatform based on Texas instrumentsOMAP4430 system on a chip(SoC).
The board has been available to the public atthe subsidized price of US 174since 27October 2010.
It is a community supported developmentplatform.
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Operating systems: The device runs the Linuxkernel, with either traditional distributions orthe Android or Mozilla Firefox OS userenvironment.
Optimised versions of Android and Ubuntu areavailable from the Linaro Foundation.
Linaro has selected the Panda Board to be one ofthe hardware platforms they support withmonthly build images.
Graphics: The Panda Board has an integratedSGX540 graphics processor and provides 1080pHDMI output.
This GPU supports OpenGL ES 2.0, OpenGL ES1.1, OpenVG 1.1 and EGL 1.3.
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Dual-core ARM Cortex-A9 MPCore withSymmetric Multiprocessing (SMP) at 1 GHz each.Allows for 150% performance increase overprevious ARM Cortex-A8 cores.
Full HD (1080p) multi-standard video
encode/decode. Imagination Technologies POWERVR SGX540
graphics core supporting all major API's includingOpenGL ES v2.0, OpenGL ES v1.1, OpenVG v1.1and EGL v1.3 and delivering 2x sustained
performance compared to the previous SGX530core. Low power audio.
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The Cortex-A9 and Cortex-A9 MPCore are twonew ARM processors designed to address therequirements for both single and multipleprocessor designs.
The common microarchitecture incorporates
features that provide enhanced architecturalfunctionality, performance and power efficiencyacross not only the processor core, but the entireSoC.
The single core processor offers higher
performance and increased power efficiency forexisting ARM11 class devices enabling enhancedfunctionality and lower power consumption forextended battery life in mobile designs.
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introduction: audio functionality plays a critical rolein embedded media processing.while audio takesless processing power in general than videoprocessing.to create analog signal representingsound wave we use transducers to convert
mechanical pressure energy to electrical energy,foran ideal transducer voltage level is directlyproportional to sound waves pressure.
Analog signals are digitized using A/D converter andif we need analog signal at output D/A converters are
used. Both A/D and D/A converters are available in one
package called AUDIO CODEC.all conversions mustobey SHANNON-NYQUISTsampling theorem.
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This theorem dictates that analog signal must besampled at rate equal to exceeding twice itshighest frequency component.(fs=2*fm)
Audio ADCS and DACSMost common digital representation of analogsignal is PCM(pulse code modulation) having an
ideal A/D coverter,but it introduces somequantisation noise.traditional PCM processinguses sigma-delta modulators to reduce number ofquantising steps reducing noise.DACS of same
modulators are used to get required analogoutput.
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Using an embedded DSP post processing enhances the
quality of listening when switched from stereo tohome theatre systems.
Need for post-processing: The digital to analog converters, audio cables,
speakers, and other parts of the signal chain may have
poor frequency response and all of these needequalization. There can be 'clicks', 'pops' and otherartifacts present in sound due to poor mastering or
just because of low bit-rate encoding. All of theseissues, that deteriorate the listening experience, can
be addressed using audio post processing techniques. Audio post processing can be used to improve the
user experience of the audio player in multiple ways
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Equalizers:which modify the frequencyenvelope
Loudness Control:which deal with theloudness variation of the audio signal
Surround Sound:which deals with creation ofa realistic and directional audio scene
System Audio Effects:which deal with
enhancing the user experience of the audioplayer.
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With the migration to digital sound and theavailability of low-cost DSPs, audio equalizershave gone from a "good to have" to a "musthave" feature.
Equalizers can be used to compensate fordistortion introduced by sound reproductionsystem.
They may also be used to modify audiocontent to suit the listener's preference.
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A graphic equalizer is a minimumrequirement in most audio players nowadays.
Achieving a flat frequency spectrum at acommon gain setting for all filters provides amajor challenge for the designer.
Often the design uses octave spacing to coverthe entire audio frequency spectrum.
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http://m.eet.com/media/1073309/Figure%203.jpg -
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its based on filling a buffer of specific lengthbefore passing data to processing function andits more efficient than sample processing.
It reduces the overhead of calling a processing
function for each sample. many embedded processors have multiple
ALUS that can parellize computation of block ofdata.
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Sound:its a longitudinal displacement wavethat propagates through air or othermedium.its defined using amplitude &frequency.
The surround sound movie soundtrack allows
the audience to hear sounds coming from allaround them, and plays a large part inrealizing what movie makers call "suspendeddisbelief".
"Suspended disbelief" is when the audience iscompletely captivated by the movieexperience and is no longer aware of theirreal-world surroundings.
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There is one center speaker which carries most ofthe dialog (since the actors usually speak whilemaking their onscreen appearance), and part of the
soundtrack. There are left and right front speakers that carry
most of the soundtrack (music and sound effects).
There is a pair of surround sound speakers that is
placed to the side (and slightly above) of theaudience to provide the surround sound andambient effects.
Finally, a subwoofer can be used to reproduce the
low and very low frequency effects (LFE) that comewith certain movies (e.g., the foot-stomping basseffects in "Jurassic Park" and "Godzilla")
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Dolby Digital
formerly known as Dolby AC-3, where AC-3 isshort for audio coding 3.
DTS Digital Surround
Dolby Surround Pro-Logic
Dolby Digital EX, THX Surround EX & DTS ExtendedSurround (DTS-ES)
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R R
RS
C
LEF
LS
L L
UP-MIXING
STEREO TO 5.1CHANNEL
SURROUNDSOUNDSYSTEM
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surround sound can be extracted from stereosound system using upmixing algorithms whichhave application is home theatre system for true
sound experience.
extraction of surround audio is done by creatingCenter(C) Channel by band pass filtering theinput.
Surround channel high-frequency absorption issimulated by high pass filtering and by thedelaying input.
Low Frequency Enhancement (LFE) channel isderived from the center channel using low passfilters.
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There are different filters like Finite ImpulseResponse (FIR), Infinite Impulse Response (IIR).
Problem of FIR filters is that requires morenumber of filter coefficients when compared toIIR which requires less number of coefficients forgetting same frequency response specification.
IIR Filters require calculation of filter coefficientsfrom Matlab. In this project we use second orderFIR filter.
External sound card is needed for outputs of
remaining channles as left and right channels aredefault.
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A c code is developed using file i/ooperations and different FIR filters for bandpass,low pass and delay filter are designed inc programming language for 2ndorder filter.
General FIR filter equation is:
() ()=0
Where x(k) is length of filter and h(k) is filter
coefficient.
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After writing c code it is to be compiledusing GCC compiler in a linux operatingsystem.
The output wave file is to be checked usingaplay command to listen for true surroundsound.
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The c code which is developed is to bedumped into the panda target board and anexternal sound card is connected to the portwhich is present in panda board.
All the speakers are given connection and theoutput wave file is to be listened from targetboard to experience true surround soundhaving centre,left and right surround,sub-woofer and left and right channel sounds.
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Hence stereo to 5.1 channel conversion isdone which finds application in home theatresystems or car sound systems to experiencetrue surround sound using up-mixing
algorithm which is tested on pandaOMAP4430 target board.
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Thank you