The voltage regulator used is the MCP1700-330, which can provide up to 250 mA of current. There are 4 capacitors associated with the regulator. C1 and C2 are the input capacitors; they stabilize the 5V input. C3 and C4 are the output capacitors, they stabilize the 3.3V output
There is a jumper that allows you to skip the regulator and use the 'built in' 3.3V supply from the Arduino. However, it is not suggested as that supply is not guaranteed to provide the current necessary.
At the bottom are the power supplies. There are 2 mechanical ground connections and a logic ground. There is also the logic power connection, connected to the 3.3v regulator.
CS is the select line, used to tell the MMC that we want to send it data. This line is pulled low (to ground) when we want to send data to the card. That means we need to make sure when we dont have anything connected, the pin is pulled high to ~3.3V. We use R6 as the pullup and zener diode D1 to keep the voltage at 3.3V. R1 allows the diode to bias properly when the Arduino pulls the pin high.
SD/MMC and FAT16 manuals if you're interested.
Image from wikipediaThe audio is encoded in PCM format. This means "pulse Code Modulation". Lets say it's a 16bit, 22khz wave. The audio waveform is sliced up 22,000 times a second and a corresponding value (up to 16 bits - from 0 to 65,635) is read from the waveform, then that value is stored in the file. Each sample is a unique value. The file is not compressed. This means the files are very large but the quality is very very good.
The SD card can provide 512 bytes at a time. This is buffered inside the Arduino's RAM so that we have smooth playback. (Techinally, its a double-buffer which means we read 256 bytes and play 256 bytes, then swap.) The audio interrupt picks one sample at a time and sends the data to the DAC (digital/analog converter).
The microcontroller/Arduino uses the DAC_CS (chip select), DAC_CLK (data clock), DAC_DI (data), and DAC_LATCH (convert the digital to analog) pins to send the sample data over. The DAC also has a Vref input, this is the reference voltage that it uses to define the maximum analog value it can generate. There is a very low low-pass filter connected to it (C6 and R8) so that any digital noise (there is -a lot-) will not make it into the audio signal.
There is another low-pass filter connected to the output of the DAC (R7 and C8). This is for filtering out the 'square wave' component you see in the recreated-audio wave. Even though the noise is only 1/4096'ths of the signal (about 1.2mV) it's still noise and these two components filter out anything above 11KHz. The reason the filter cut-off frequency is 11KHz and not 22KHz is that if you sample at 22KHz you will only be able to reproduce frequencies at half that rate, 11KHz. This is the Nyquist theory. It is sneaky but true. If you try to sample 16KHz waveform at 22KHz it will actually sound much -lower-, it will play at 6KHz (it is 'mirrored' around 11KHz).
The analog signal then goes into a high-output, rail-to-rail opamp. This op-amp can provide up to 100mA per channel. The two channels are hooked up in parallel for up to 200mA output (at 5V). This means it can provide 1/8 W into an 8ohm speaker (or 1/4 W into 4ohm speaker). This isn't enough for a boom-box but its good for headphones and small speakers. The output is filtered through a bypass capacitor C9 which will keep any DC voltage from going to the speaker, which could damage it.
The headphone jack is stereo, which both mono channels connected in parallel. This gives the most power output. There are internal switches in the jack so that when the headphones are removed, the audio flows to the 'speaker connection' next to the jack.