On PANDA OMAP4430 targetboard

Project by: M.SAI KIRAN 10UQ1A0456 V.SAI MANOGNYA 10UQ1A0495 V.REKHA 10UQ1A0499 P.MOUNIKA 10UQ1A0466

Btech 4th year. [Electronics &communication engineering]. Vignan institutes of technology and aeronautical engineering,deshmukhi,A.P.

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 by up-mixing technique. Developing C code using FILE i/o operations and compiling in a LINUX environment using GCC compiler. Integrate the code developed in C on Panda board and test it with File I/O. Conclusion.

With the growing proliferation of multi channel audio in consumer electronics, there are situations where the number of channels of either the audio content or the reproducing loud speakers is limited and it is necessary to upmix or downmix the channels to improve the listening experience. In the upmixing process, the surround channels can be created from the two-channel stereo inputs by using two approaches: 1) uses the decorrelated part of the stereo inputs as the surround channels. 2) the other approach produces the surround channels by simulating the reverberant sound field in the background.

The PandaBoard is a low-power, lowcost single-board computer development platform based on Texas instruments OMAP4430 system on a chip(SoC). The board has been available to the public at the subsidized price of US$174 since 27 October 2010. It is a community supported development platform.

Operating systems: The device runs the Linux kernel, with either traditional distributions or the Android or Mozilla Firefox OS user environment. Optimised versions of Android and Ubuntu are available from the Linaro Foundation. Linaro has selected the Panda Board to be one of the hardware platforms they support with monthly build images. Graphics: The Panda Board has an integrated SGX540 graphics processor and provides 1080p HDMI output. This GPU supports OpenGL ES 2.0, OpenGL ES 1.1, OpenVG 1.1 and EGL 1.3.

Dual-core ARM Cortex-A9 MPCore with Symmetric Multiprocessing (SMP) at 1 GHz each. Allows for 150% performance increase over previous ARM Cortex-A8 cores. Full HD (1080p) multi-standard video encode/decode. Imagination Technologies POWERVR SGX540 graphics core supporting all major API's including OpenGL ES v2.0, OpenGL ES v1.1, OpenVG v1.1 and EGL v1.3 and delivering 2x sustained performance compared to the previous SGX530 core. Low power audio.

The Cortex-A9 and Cortex-A9 MPCore are two new ARM processors designed to address the requirements for both single and multiple processor designs. The common microarchitecture incorporates features that provide enhanced architectural functionality, performance and power efficiency across not only the processor core, but the entire SoC. The single core processor offers higher performance and increased power efficiency for existing ARM11 class devices enabling enhanced functionality and lower power consumption for extended battery life in mobile designs.

introduction: audio functionality plays a critical role in embedded media processing.while audio takes less processing power in general than video processing.to create analog signal representing sound wave we use transducers to convert mechanical pressure energy to electrical energy,for an ideal transducer voltage level is directly proportional to sound waves pressure. Analog signals are digitized using A/D converter and if 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 must obey SHANNON-NYQUIST sampling theorem.

This theorem dictates that analog signal must be sampled at rate equal to exceeding twice its highest frequency component.(fs=2*fm) Audio ADCS and DACS: Most common digital representation of analog signal is PCM(pulse code modulation) having an ideal A/D coverter,but it introduces some quantisation noise.traditional PCM processing uses sigma-delta modulators to reduce number of quantising steps reducing noise.DACS of same modulators are used to get required analog output.

Using an embedded DSP post processing enhances the quality of listening when switched from stereo to home 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 need equalization. There can be 'clicks', 'pops' and other artifacts present in sound due to poor mastering or just because of low bit-rate encoding. All of these issues, 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

Equalizers:which modify the frequency envelope Loudness Control:which deal with the loudness variation of the audio signal Surround Sound:which deals with creation of a realistic and directional audio scene System Audio Effects:which deal with enhancing the user experience of the audio player.

With the migration to digital sound and the availability of low-cost DSPs, audio equalizers have gone from a "good to have" to a "must have" feature. Equalizers can be used to compensate for distortion introduced by sound reproduction system. They may also be used to modify audio content to suit the listener's preference.

A graphic equalizer is a minimum requirement in most audio players nowadays. Achieving a flat frequency spectrum at a common gain setting for all filters provides a major challenge for the designer. Often the design uses octave spacing to cover the entire audio frequency spectrum.

its based on filling a buffer of specific length before passing data to processing function and its 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 of data.

Sound:its a longitudinal displacement wave that propagates through air or other medium.its defined using amplitude & frequency. The surround sound movie soundtrack allows the audience to hear sounds coming from all around them, and plays a large part in realizing what movie makers call "suspended disbelief". "Suspended disbelief" is when the audience is completely captivated by the movie experience and is no longer aware of their real-world surroundings.

There is one center speaker which carries most of the dialog (since the actors usually speak while making 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 the audience to provide the surround sound and ambient effects. Finally, a subwoofer can be used to reproduce the low and very low frequency effects (LFE) that come with certain movies (e.g., the foot-stomping bass effects in "Jurassic Park" and "Godzilla")

Dolby Digital formerly known as Dolby AC-3, where AC-3 is short for audio coding 3. DTS Digital Surround Dolby Surround Pro-Logic Dolby Digital EX, THX Surround EX & DTS Extended Surround (DTS-ES)

R RS
UP-MIXING STEREO TO 5.1 CHANNEL SURROUND SOUND SYSTEM

C LEF

LS L

surround sound can be extracted from stereo sound system using upmixing algorithms which have application is home theatre system for true sound experience. extraction of surround audio is done by creating Center(C) Channel by band pass filtering the input. Surround channel high-frequency absorption is simulated by high pass filtering and by the delaying input. Low Frequency Enhancement (LFE) channel is derived from the center channel using low pass filters.

There are different filters like Finite Impulse Response (FIR), Infinite Impulse Response (IIR). Problem of FIR filters is that requires more number of filter coefficients when compared to IIR which requires less number of coefficients for getting same frequency response specification. IIR Filters require calculation of filter coefficients from Matlab. In this project we use second order FIR filter. External sound card is needed for outputs of remaining channles as left and right channels are default.

A c code is developed using file i/o operations and different FIR filters for band pass,low pass and delay filter are designed in c programming language for 2nd order filter. General FIR filter equation is: () = =0 () Where x(k) is length of filter and h(k) is filter coefficient.

After writing c code it is to be compiled using GCC compiler in a linux operating system. The output wave file is to be checked using aplay command to listen for true surround sound.

The c code which is developed is to be dumped into the panda target board and an external sound card is connected to the port which is present in panda board. All the speakers are given connection and the output wave file is to be listened from target board to experience true surround sound having centre,left and right surround,subwoofer and left and right channel sounds.

Hence stereo to 5.1 channel conversion is done which finds application in home theatre systems or car sound systems to experience true surround sound using up-mixing algorithm which is tested on panda OMAP4430 target board.

Thank

you