S/PDIF (Sony/Philips Digital Interconnect Format)
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HDTV Audio In the December ‘07 issue, we examined the various ways to hook up pieces of your home entertainment system to your HDTV. We specifically focused on the different video interfaces. We’ll continue now with the choices for passing audio from one device to another. by Jeff Mazur nce again, the most common the ham operator who lives next and also known as TOSLINK — uses 1 connection by far is the standard door!). To solve this issue — as well as mm fiber terminated in a 5 mm Oanalog stereo pair using RCA complete the total conversion to binary connector. While earlier cables were jacks and cables. With good quality 1s and 0s — there are three basic ways restricted to less than 15 feet, you can cable and connectors, this method can to pass audio signals digitally between now buy high quality TOSLINK cables provide excellent results. The most devices: coax, optical, and HDMI. up to 100 feet in length. TOSLINK can common issue with analog audio con- carry data signals of up to 125 Mbits/s, nections is its susceptibility to picking S/PDIF (Sony/Philips which allows for three audio channels. up hum and/or other extraneous Digital Interconnect However, it is usually used to carry a signals, especially from components Format) single pair of stereo audio signals. within your system (or perhaps from As an electrical signal, S/PDIF is Named after the two companies represented by a roughly 1V digital Figures 2-4 are courtesy of Wikipedia, the that developed this interface, S/PDIF pulse train using Biphase Mark Code free encyclopedia (licensed to the public is a means to carry audio between (BMC) to carry the audio data. While under the GNU Free Documentation License). devices in a digital format. The signals no specific sampling rate or bit depth can be carried over is specified in the standard, audio is standard 75 ohm coaxial usually carried as either 48 kHz (DAT) cable using RCA jacks or 44.1 kHz (CD) data with either 20 (or BNC connectors in or 24 bit samples. We’ll describe the professional equipment) actual data format in a moment. or via optical fiber (glass or plastic, usually HDMI terminated with F05 connectors). See Figure 1. We’ve already discussed the The optical connec- HDMI interface that can carry digital tion — created by Toshiba video between devices. HDMI also includes support for up to eight FIGURE 1. Digital audio channels of uncompressed digital connections (top, coax and bottom, optical). audio at a 192 kHz sample rate with a 60 February 2008 24 bits/sample, as well as compressed values. Under ideal conditions, it also (PCM). This approach simply takes the streams such as Dolby Digital, or DTS. represents the maximum signal to noise output from an Analog-to-Digital HDMI also supports one-bit audio, ratio (SNR), which is related to the num- Converter (ADC) and places the bits such as that used on Super Audio CDs ber of bits by the following formula: into a continuous bitstream. at rates up to 11.3 MHz. With version Figure 2 shows a sine wave (in 1.3, HDMI now also supports lossless SNR = 20 log 2N = approx (6 x N) dB red) that is sampled and quantized compressed streams such as Dolby using simple PCM. At each sample TrueHD and DTS-HD Master Audio. where N = number of bits. point, the digital representation of the signal’s analog value is sampled and Digital Audio Basic For example, a 20-bit converter then held until the next sample point. theoretically could obtain an SNR of This produces an approximation of the Digital audio connections can be 120 dB (if there are no other sources original signal, which is easily encoded used to connect various components of noise). In practice, the maximum as digital data. For example, if the sine of your home entertainment system signal level is usually reduced by 20 wave in Figure 2 is quantized into 16 such as from a cable or satellite STB dB of headroom to prevent clipping. values (i.e., four bits), we would (Set Top Box) to the TV. Since audio is This still leaves an SNR of approxi- generate the following data samples: transmitted digitally in the ATSC DTV mately 100 dB. In comparison, normal 1001, 1011, 1100, 1101, 1110, 1110, signal, this will often be the best audio tape typically only achieves an 1111, 1111, 1111, 1110, etc. choice. Other components (e.g., a CD SNR of about 60 dB. We could transmit these PCM player) also handle audio natively in a As you can see, digitizing an ana- samples as four-bit parallel data with a digital form. However, devices that log signal is all about compromise. You separate clock signal to indicate when handle audio as an analog signal — need to sample at a high enough rate each sample was taken. This is cumber- including the equipment used to so as not to miss changes in the signal some, however, and requires the use of record or create TV audio at its source that occur between the samples. And multi-conductor cables. Most data — must first convert the analog signal we need enough bits to represent each transmission today is done in a serial to digital. This process is known as sample so that the difference between fashion. This requires that each bit of digitizing and is a good place to start the actual analog value and its closest the PCM sample be clocked out onto a when discussing digital audio. digital representation (a.k.a., quantiza- single serial data line. At the receiving To digitize an analog signal, we tion error) is not very much. Of course, end of this data stream, a shift register basically perform two separate increasing either of these values means will convert the serial data back into functions. First, the signal is sampled that there will be more digital data that parallel data words. To keep the receiv- at regular intervals to determine its needs to be carried and processed. er in sync with the transmitter, some value at each discrete point in time. On the positive side, once a signal form of clock recovery is necessary. This is usually the function of a has been digitized it can be transmit- One of the easiest ways to do this sample-and-hold circuit. Next, each ted much more efficiently and without is to make sure that the serial data sample is quantized, or converted many of the side effects of noise and changes polarities at least once during from an analog voltage to a particular distortion present in the communica- each bit-time. This is the basis for sever- digital representation of that value. tion channel used. More importantly, al different coding schemes, including The sampling rate determines it can be compressed digitally so that Biphase Mark Code (BMC) — the sig- what frequencies can be carried redundant and/or unessential data naling method used by both TOSLINK digitally; information theory tells us can be discarded. This is one of the and the professional digital audio for- that only frequencies below one-half main reasons that our TV signals are mat established by, and referred to as, of the sampling frequency (also undergoing the transition to digital. AES/EBU (Audio Engineering Society referred to as the Nyquist frequency) can be represented accurately. Signals PCM above this limit will cause extraneous frequencies (i.e., distortion) to appear There are many due to an effect known as aliasing. ways to represent In other words, we need at least each sample as a dig- two samples per cycle of the highest ital signal. The most frequency we wish to digitize. common technique The quantization of each sample is known as Pulse- determines how many bits will be used Code Modulation to represent each sample. The more bits, the higher the precision will be of FIGURE 2. Analog- each sample. This translates into the to-digital conversion of a signal using dynamic range of a signal, or the differ- Pulse Code ence between its lowest and highest Modulation (PCM). February 2008 61 FIGURE 3. Serialization of rate, compression, emphasis modes. digital data using Biphase Mark Coding (BMC). • Byte 1: Indicates if the audio nized to a common 27 stream is stereo, mono, or some other MHz timebase. Even so, combination. a frame of NTSC video has a duration of: • Byte 2: Audio word length. 1 / 29.97 = 33.366… ms • Byte 3: Used only for multichannel and the European Broadcasting Union). applications. With BMC, the data stream At 48 kHz, an audio frame has a changes value at the beginning of duration of: • Byte 4: Suitability of the signal as a each data bit. A logic 1 is represented sampling rate reference. by having the stream change value 1 / 48,000 = 20.833… µs again during the middle of its bit time; • Byte 5: Reserved. it does not change for a logic 0 (see This makes a complete audio block Figure 3). BMC coding provides easy 192 x 20.833 = 3,999.4 µs. The number • Bytes 6–9 and 10–13: Two slots of synchronization since there is at least of audio samples per video frame, four bytes each for transmitting ASCII one change in polarity for every bit. however, is not an integer number: characters. Also, the polarity of the actual signal is not important since information is 33366 / 20.833 = 1601.6 audio • Bytes 14–17: Four-byte/32-bit sam- conveyed by the number of transitions samples/video frame ple address, incrementing every frame.