The "STA013" and "STA015" integrated circuits by STMicroelectronics are flexible MP3 decoders with good performance. The STA015 is almost fully backwards compatible with the STA013, so this article will focus on the STA013 and mention differences with the STA015 where applicable.
The goal of this article is to collect information that is gained from working with the chip, and that is not readily available in the datasheet or the application notes for the STA013 and STA015 ICs. As such, this article augments the datasheet and the application notes, but it does not replace them.
A brief technical introduction
The STA013 uses an I2C interface for "control" and an SPI-compatible interface for data. The digital PCM data that it provides on its outputs can be in one of various standard formats (among which I2S). Typically, you can connect the STA013 directly to a Digital-Analogue Converter (DAC), because the STA013 also generates the (oversampling) clock for the DAC.
To feed the compressed data into the decoder, the STA013 supports a serial data interface that is compatible with the SPI bus ("Serial Peripheral Interface", developed by Motorola) and that allows a bit stream of up to 20 MHz. Control commands for the decoder go over the I2C bus, at a maximum frequency of 400 kHz. The decoder is, in fact, a DSP (digital signal processor) with a fairly large number of internal registers, many of which are undocumented.
To get the STA013 started, you will first need to upload a "configuration file" with over 2000 assignments of values to registers. STMicroelectronics provides this configuration file as an ASCII file with a register number and a value per line. After loading the file, you need to configure a few more registers depending on the crystal (clock frequency) and the desired input and output interfaces. Some references in the STMicroelectronics documentation call this file the patch file, which suggests that this configuration file fixes issues that have been found after the STA013 went into production. However, without this patch, the STA013 will not run. The STA015 has the patches built-in and does not need to be configured.
The "data feed" to the STA013 goes via a serial protocol that is compatible with SPI. In this data protocol, the microcontroller must first select the STA013 chip by toggling the "
BIT_EN" line high; then it clocks in the eight bits of data in each byte, starting with the most significant bit. The
BIT_EN can stay high during the transfer of all bits/bytes; there is no need to pulse it between two bytes. Normally, SPI is also defined for 16-bit data transfers, but the STA013 requires 8-bit data transfers.
As the SPI protocol has no acknowledge for accepted bytes (unlike I2C), the slave device must either accept anything that the master device pushes into it, or have some other means to signal the master when to stop sending. The STA013 handles synchronization through the
DATA_REQ line (a separate pin on the IC). The master device should check the status of this line after writing each full byte. The datasheet for the STA015, a successor to the STA013, documents that it drops the
DATA_REQ low when its internal buffer has room for at most four additional bytes.
Audio data buffers
The STA013 processes the MP3 audio data per frame. A frame is an independent block containing a fixed number of compressed samples that follow a 4-byte header (see also the next section of this paper). The format of the MP3 files were designed this way to enable streaming of music files (e.g. over the Internet). You can pick up any MPEG audio stream mid-way, starting at the first frame header that you detect. There is never a need to seek from the start of the file to the desired position. The frame header allows the STA013 to skip over any non-audio data, such as an ID3v2 tag. A sequence of 11 bits that are set (1) indicates the start of a frame.
Due to the "bit reservoir" that most layer 3 streams use, frames are, in practice, not independent of each other. When an MP3 decoder must pick up a stream halfway, it will mostly produce silence or an arbitrary noise for the first few frames.
In a lesser sense, the Modified Discrete Cosine Transform (MDCT) that MP3 encoding uses also makes frames interdependent. MDCT is an overlapping encoding where the samples in the first half of a frame combine with those that are actually part of the predecessing frame. When starting in the middle of an audio clip, the first frame is only partially decoded.
Our experiments show that the STA013 keeps its
DATA_REQ high until it has loaded a complete frame. The size of a frame, in bytes, depends on a number of parameters, notably the bitrate and the sampling frequency; it grows to over 1 kiB of 320 kb/s 44.1 kHz tracks. The STA013 needs to have that amount of memory internally. It might have a larger buffer size, so that it can accept the next frame while playing an active frame.
We attached the
DATA_REQline to an interrupt and pushed a high-precision time stamp in a queue at every interrupt. Our statements about the size of the internal buffer of the STA013 are based on observations of the interrupt timestamps when sending tracks with different sampling frequencies and bitrates to the STA013. With tracks at 44.1 kHz, we observed pulses of the
DATA_REQline at intervals of 26.1 ms, regardless of the bitrate of the track. When sending a track sampled at 48 kHz, the
DATA_REQpulsed at 24 ms. These measured values match the durations of the MP3 frames at 44.1 kHz and 48 kHz respectively.
An e-mail exchange with a STMicroelectronics engineer confirmed our findings about the buffer size. He described the design as a dual-buffer: a back-buffer that holds a complete frame (which the decoder works on) and a front-buffer for prefetching the bytes for the next frame.
Additionally, the STA013 has an internal MP3 parser that skips any non-MP3 data, such as ID3 tags. You can therefore feed an MP3 file into the decoder without needing to parse it yourself. This parser runs parallel with the reception of input data: when a large ID3 tag is present in a file, it is immediately rejected and the
DATA_REQ stays high until the STA013 has received a full valid frame. If you would feed a non-MP3 file into the STA013, the
DATA_REQ will never go low at all, regardless of the size of the file that you push into the chip.
The STA013 can give you the header of the frame that it has just started decoding. Or rather, it gives you three of the four bytes of the header (since the first byte of an MP3 frame header is all 1s, it never contains relevant information). This data the I2C bus. The bitrate for the active block is in this header information. In variable bitrate files, this field may change from one frame to the next. Another register in the STA013 gives you the average bitrate. This register will allow you to estimate the duration of the music sequence --it remains, however, and estimate.
Handling the audio frame-by-frame makes sense for handling variable bitrate files, but it also opens possibilities for or making sound loops. To make a loop, you basically clock in the first block of an audio file again after having completed the last block. The STA013 has no knowledge of files, only frames and there is no need to "stop" the MP3 playback in the STA013 at the end of one file and to restart it at the next. Making seamless loops, without plops or gaps, is not quite so easy, though --see the separate article "Gapless looping MP3 tracks".
Stopping a track (clearing buffers)
To abort a playing track, you can simply send a command over the I2C bus to set the "play" register to zero. However, this does not reset the internal buffers of the STA013. If you resume the track at the same position, there is no problem. However, if you wished to stop the track and play a different track at some later time, you will notice that the STA013 first pushes out some samples of the stopped track. Evidently, these samples were still in a buffer.
The only way that we have found to clear this buffer is to reset the STA013; sending a "soft reset" command is sufficient. The drawback is that, after the reset, you need to send the configuration file again. Sending those 2007 register settings over I2C takes some time. In our code, with I2C running at 400 kHz, the complete reset procedure took roughly 220 ms. This only applies to the STA013, as the STA015 does not have a patch file.
The description of the bass and treble controls is very brief in the datasheets of the STA013 and STA015. There are five settings in total: bass filtre frequency, bass gain, treble filtre frequency, treble gain and a general tone control attentuation. To start with the latter: when you use a positive gain for bass or treble, the volume must be attenuated by at least the same amount, because the gain cannot raise above unity gain. For example: if the bass gain is 6 dB and the treble gain is 12 dB, the tone control attenuation should be 12 dB (the highest of the two). If you set the tone control attenuation to 6 dB, the treble filter will not reach the 12 dB level, as it clips to unity gain.
The bass and treble controls each create a double-knee curve. The picture shows a curve with the treble frequency set at 1.5 kHz and the gain at 18 dB; the bass controls are not used (zero-gain).
For filters, the cut-off frequency is usually defined as the frequency where the gain is +3dB or -3dB from the adjoining horizontal section of the curve. A double-knee curve has two cut-off frequencies. From inspection and experiments, we have found that, at a gain of +18 dB (or -18 dB), the cut-off frequencies of the filter are at approximately:
fc1 = 0.25 · f
fc2 = 1.75 · f
where f is the frequency passed to the STA013 for its bass or treble settings. As is apparent from these figures, the frequency that you pass to the STA013 lies exactly in the middle of the two cut-off frequencies. An alternative (one that I would prefer) would be to pass in the frequency where the gain is half that of the total gain. This frequency is at 0.66 · f.
The roll-off rate (the slope of the curve) is approximately:
S = 4.6 dB / octave
When the gain is lower then ±18 dB, the roll-off rate decreases proportionally. I prefer to consider the cut-off frequencies as being independent of the gain, even though this is not quite accurate: since the cut-off frequency is defined as the point where the gain is ±3 dB from the horizontal section, it depends on the roll-off rate. The picture at the right has the same settings as the previous picture, except that the treble gain is at +9 dB.
Frame size and frame duration
There is a wide-spread misconception that a frame of an MP3 file is defined at 26 ms. Even if this may be valid for most MP3 files, it is inaccurate. Typically, an MP3 file stands for MPEG 1, layer 3, at 44100 Hz. For these files, each frame contains 1152 samples, and their duration is indeed approximately (but not exactly) 26 ms. The sampling frequency configurable, however, and MP3 audio files sampled at other rates than the ubiquitous 44.1 kHz certainly exist. Depending on the MPEG version and the layer, the number of samples per frame may also differ from 1152. The article "MPEG Audio Frame Header" (see references) has all the details.
To calculate the size in bytes of a frame, the criterion are the bitrate and the number of frames per second. The bitrate gives the number of bits per second, and this number includes the headers of the frames. Divide this number by 8 to get the number of bytes per second; then divide it again by the number of frames per second to get the size of a frame in bytes. The number of frames per second is, of course, the inverse of the frame duration. There are three caveats in the calculation of the frame length that you must take into account:
- The number of bytes per frame may come out of the calculation as a fractional value. In that case, the MP3 file has frames with a byte size that is the computed value rounded down and frames with the computed value rounded up. The header of a frame indicates the occasional extra byte with the "padding bit".
- The frame sizes are rounded up to a "slot size". The size of a slot is just one byte for layers 2 and 3, but it is four bytes for layer 1.
- It has been reported that, in practice, the frame size is sometimes a little "off spec". Some encoders appear to "pad" frames with a few extra bytes on specific circumstances. An MP3 decoder must be robust against this and it can check for the "sync marker" (eleven consecutive 1 bits starting at a byte boundary) to find the real start of a next frame.
For the common case of MPEG 1 layer 3, the duration of a frame is:
Note that the bitrate value is the number of bits of compressed data that must be pushed into a decoder (and that the bitrate value includes the size of the frame headers). The MP3 compression algorithm is a combination of domain transform and quantization, plus a statistical lossless compression scheme called "Huffman compression". In the domain transform and quantization phase, you may calculate deterministically what your compression ratio is, but you have no idea to how many bytes this will be reduced in the Huffman coding step. If you quantize too coarsely, quality suffers; if you quantize too finely, the frame size will exceed the set bitrate. However, you will only know the size of the frame until after the Huffman coding phase has finished. As you can imagine, an encoder uses an iterative procedure to find the optimal settings for each frame.
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