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This project implements a multi-function multimedia system that allows the user to sing with the music video and generate some fancy sound effects. In recent decades, multimedia becomes quite popular in our daily life. In fact, multimedia system has existed for a long time. People enjoy the time viewing photos, listening to music, and watching movies. With the development of technology, we are able to create, modify and edit multimedia data. Also, thanks to the emergence of the digital era, we have more tools that help us to achieve these goals.
For example, we could store the analog data in digital format to make it more robust and lasting. Besides, we are able to synthesize the data and generate some amazing effects.
Our final project is to design an intelligent multimedia system. This system allows users to use the microphone as the analog input to speak and sing. For example, we could adjust the volume of the voice and music.
Besides, we could generate the echo of our voice as if we are in a theatre. Furthermore, since we could transform the analog signal into digital format, we are able to manipulate the audio data.
In this project, we used several popular mathematical models in the field of Audio Signal Processing that help us to simulate specific sound effects.
The kernel of the system is controlled dsc808 the ATMega32 microcontroller, same as the one we used in the regular lab sections. The voice signal input is generated from the microphone, and the music signal is generated from any music application such as a MP3 player or PC.
We passed both of the audio signals through a pre-amplifier daasheet boost its signal strength. The digital data can then be propagated to the microcontroller.
Besides, we use the RS serial port datasgeet communicate between the microcontroller and the PC. This design allows us to transfer the music data to the PC from this multimedia system in real time. In the microcontroller, we could capture both of the voice signal and music signal and mix them to generate special effects. The output signal can eatasheet be propagated to a digital-to-analog chip DAC to transform the digital data to analog signal.
Finally, the signal will be passed to the speaker and generate a wonderful melody. User could apply the featured effects to the voice signal via this GUI.
The final project is a good experience for us to integrate all the knowledge that we gained in this course and some other related courses. We chose this topic because we are all interested in multimedia signal processing. Also, we thought that this topic would be a good chance for us to combine our knowledge in various fields. Also, we could understand more about the analog circuit when datashedt integrated all the hardware components.
The overall design block of this system is shows as follows. In this projectwe used two different channels for our audio signal inputs: Initially both of the signals are in analog format, so we needed two analog-to-digital converters to transform the analog signal into digital format before feeding into the microcontroller. Since there is only one analog-to-digital converter embedded in ATMega32, we used the internal analog-to-digital converter for the voice channel and an external analog-to-digital converter chip ADC for the music channel.
The signal could be manipulated in the microcontroller. The signal will be propagated to the digital-to-analog converter chip DAC to be transferred into analog format. Finally, the processed signal is passed to a speaker to generate the sound. It provides a user-friendly graphical interface to allow the dax808 to control the system by a mouse click.
How to use a DAC with the arduino
The mouse click is then turned into a command to the microcontroller via the COM Port. There were so some minor changes we made in the system architecture while we were working on this project.
Some of the changes were made since we encountered some limitation in the hardware. For example, in a common karaoke system, we need at least two channels for one voice input and one music input.
However, there is only one internal analog-to-digital converter in ATMega If we use it for both voice and music signals alternatively, the sample rate will be reduced to less than half of the original one. This would lower the sound quality. In order to solve this problem, we decided to use an external analog-to-digital converter chip ADC for the music channel.
In this way, the microcontroller can sample both the voice and music signals in 8KHz in maximum concurrently which the produces a higher sound quality. Besides, initially we decided to output the processed signal to the speaker in the same approach that we learned in Lab 2.
However, we found that the sound quality was pretty bad since this approach used PWM mode of the microcontroller to generate an approximate signal and hence the accuracy of signal output was lowered.
To cope with this problem, we used a digital-to-analog chip DAC to transform the digital output to an analog output that makes the sound more dulcet. For the choice of user interface, we initially decided to use the buttons on the STK board as the control panel.
However, we found that using a PC can provide a much flexible and user-friendly control interface. All the changes we made help datashheet the performance of datashedt multimedia system on the whole. Since all of our team members are ECE MEng students, Cornell University owns all copyrights and potential, patents related to this project. Since the ideas and implementation of our project hardware are solely our own, there should not be any trademark, patent, or copyright infringement issues.
Also we used the timers to control the sampling frequency of voice and music signals as well as mixed signal output. In order to facilitate the communication between a PC and the microcontroller, we make use of the high speed serial port.
The console program configures the COM Port to run at a baud rate of The program gets the command from the graphical user dahasheet and transmits it dwtasheet the microcontroller, and at the same time receives the real time digital audio data from the microcontroller.
The Java application on top of the console program is to provide graphical user interface for the multimedia system and the useful operations such as displaying the voice waveform, recording the voice in a wave file and sending an email attached with the wave file. The following diagram demonstrates the general data flow between the microcontroller, the xac808 program and the Java application.
The dataflow between the microcontroller, the console program and the Java application. The following is the user interface of our program. The left panel is to display the waveform datashset the audio signal. The right panel is effect datwsheet and volume control panel. All the buttons on the bottom are to control the operations of the multimedia system. PC Graphical User Interface. In our multimedia system, we implemented eatasheet Graphical User Interface written in Java running on the PC to control the whole embedded system.
The UART Receive Complete Interrupt is enabled to allow dtaasheet program jump to the interrupt service routine to receive the command packets. The received data is then passed to processCommandPacket to be saved into the command packet buffer. A state machine is implemented in the packet reception.
The code is as follows. There are 7 different commands the PC can send to the microcontroller. They are listed as below. The packet structure for command packet has 4 bytes. The first 2 bytes are for 16 bits command. The sac808 2 bytes are for 16 bits payload length of the command packet. The command packet is stored to the dsc808 commandPacket and processed by the checkCommand function later. The checkCommand function handles the command packet sent from the PC software.
The code is shown as follows. The sampled audio signal is sent to the PC in the timer interrupt. Dattasheet the multimedia system, we have designed some special effects for voice processing. Therefore we implemented a sample buffer of size bytes to store the signal samples and reuse the buffer continuously. The implementation is as follows.
Our program used the internal Analog-to-Digital Converter of ATMega32 microcontroller to obtain the voice signal from a microphone via an Op-Amp circuit to amplify the signal strength. The conversion rate is up to 15, samples at maximum per second. So the internal ADC suits for our system requirement. The reference voltage is 5V.
The Op-Amp circuit has a 2. Therefore, for no signal, the digital representation is theoretically But in our tests, we found that the datasheey data of the dc808 signal was in fact This is because the DC-offset is not exactly 2. In the function processSignalthe following code is to get the voice signal from ADC, store the signal to a sample buffer and then start another conversion.
Pin 4 of PortD is used to control the WR pin of ADC in order to command it to start another conversion by setting it to be 0 and then 1 immediately.
Clarification on Digital to analog conversion
The mixed dqc808 of voice and music signals is eventually output to PortC for digital-to-analog conversion using DAC chip and the analog output of DAC is connected to a speaker to output the signal.
The sampling rate of both voice and music is controlled by Timer0. When there is a Timer0 compare interrupt, the function processSignal is invoked to get the voice and music samples and do the selected effects.
Timer1 is assigned with prescaler 8. The clock signal is equal to. The multimedia system allows the users to select preferred sampling frequency for compare and contrast purpose. Lower sampling frequency can also be chosen to lower sample size and hence reduce the storage of the recorded data in the PC.