VS1053b Datasheet
VS1053b -
Ogg Vorbis/MP3/AAC/WMA/FLAC/
MIDI AUDIO CODEC CIRCUIT
Features
•Decodes
Ogg Vorbis;
MP3 = MPEG 1 & 2 audio layer III (CBR
+VBR +ABR);
MP1/MP2 = layers I & II optional;
MPEG4 / 2 AAC-LC(+PNS),
HE-AAC v2 (Level 3) (SBR + PS);
WMA 4.0/4.1/7/8/9 all profiles (5-384 kbps);
General MIDI 1 / SP-MIDI format 0 files;
FLAC with software plugin;
WAV (PCM + IMA ADPCM)
•Encodes Ogg Vorbis w/ software plugin
•Encodes stereo IMA ADPCM / PCM
•Streaming support for MP3 and WAV
•EarSpeaker Spatial Processing
•Bass and treble controls
•Operates with a single 12..13 MHz clock
•Can also be used with a 24..26 MHz clock
•Internal PLL clock multiplier
•Low-power operation
•High-quality on-chip stereo DAC with no
phase error between channels
•Zero-cross detection for smooth volume
change
•Stereo earphone driver capable of driv-
ing a 30 Ωload
•Quiet power-on and power-off
•I2S output interface for external DAC
•Separate voltages for analog, digital, I/O
•On-chip RAM for user code and data
•Serial control and data interfaces
•Can be used as a slave co-processor
•SPI flash boot for special applications
•UART for debugging purposes
•New functions may be added with soft-
ware and up to 8 GPIO pins
•Lead-free RoHS-compliant package
Description
VS1053b is an Ogg Vorbis/MP3/AAC/WMA/
FLAC/WAVMIDI audio decoder as well as an
PCM/IMA ADPCM/Ogg Vorbis encoder on a
single chip. It contains a high-performance,
proprietary low-power DSP processor core
VS_DSP4, data memory, 16 KiB instruction
RAM and 0.5+ KiB data RAM for user appli-
cations running simultaneously with any built-
in decoder, serial control and input data in-
terfaces, up to 8 general purpose I/O pins,
an UART, as well as a high-quality variable-
sample-rate stereo ADC (mic, line, line + mic
or 2×line) and stereo DAC, followed by an
earphone amplifier and a common voltage buffer.
VS1053b receives its input bitstream through
a serial input bus, which it listens to as a
system slave. The input stream is decoded
and passed through a digital volume control
to an 18-bit oversampling, multi-bit, sigma-
delta DAC. The decoding is controlled via a
serial control bus. In addition to the basic de-
coding, it is possible to add application spe-
cific features, like DSP effects, to the user
RAM memory.
Optional factory-programmable unique chip
ID provides basis for digital rights manage-
ment or unit identification features.
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VS1053b Datasheet CONTENTS
Contents
VS1053 1
Table of Contents 2
List of Figures 5
1 Licenses 6
2 Disclaimer 6
3 Definitions 6
4 Characteristics & Specifications 7
4.1 Absolute Maximum Ratings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
4.2 Recommended Operating Conditions . . . . . . . . . . . . . . . . . . . . . . . . 7
4.3 AnalogCharacteristics ................................ 8
4.4 PowerConsumption ................................. 9
4.5 DigitalCharacteristics................................. 9
4.6 Switching Characteristics - Boot Initialization . . . . . . . . . . . . . . . . . . . . 10
5 Packages and Pin Descriptions 11
5.1 Packages ....................................... 11
5.1.1 LQFP-48.................................. 11
6 Connection Diagram, LQFP-48 14
7 SPI Buses 16
7.1 SPIBusPinDescriptions............................... 16
7.1.1 VS10xx Native Modes (New Mode, recommended) . . . . . . . . . . 16
7.1.2 VS1001 Compatibility Mode (deprecated, do not use in new designs) 16
7.2 DataRequestPinDREQ............................... 17
7.3 Serial Protocol for Serial Data Interface (SPI / SDI) . . . . . . . . . . . . . . . . 18
7.3.1 SDI in VS10xx Native Modes (New Mode, recommended) . . . . . . 18
7.3.2 SDI Timing Diagram in VS10xx Native Modes (New Mode) . . . . . . 19
7.3.3 SDI in VS1001 Compatibility Mode (deprecated, do not use in new
designs) .................................. 20
7.3.4 Passive SDI Mode (deprecated, do not use in new designs) . . . . . 20
7.4 Serial Protocol for Serial Command Interface (SPI / SCI) . . . . . . . . . . . . . 21
7.4.1 SCIRead ................................. 21
7.4.2 SCIWrite ................................. 22
7.4.3 SCIMultipleWrite............................. 22
7.4.4 SCI Timing Diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
7.5 SPI Examples with SM_SDINEW and SM_SDISHARED set . . . . . . . . . . . 24
7.5.1 TwoSCIWrites .............................. 24
7.5.2 TwoSDIBytes............................... 24
7.5.3 SCI Operation in Middle of Two SDI Bytes . . . . . . . . . . . . . . . 25
8 Supported Audio Decoder Formats 26
8.1 Supported MP3 (MPEG layer III) Formats . . . . . . . . . . . . . . . . . . . . . 26
8.2 Supported MP2 (MPEG layer II) Formats . . . . . . . . . . . . . . . . . . . . . . 27
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VS1053b Datasheet CONTENTS
8.3 Supported MP1 (MPEG layer I) Formats . . . . . . . . . . . . . . . . . . . . . . 27
8.4 Supported Ogg Vorbis Formats . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
8.5 Supported AAC (ISO/IEC 13818-7 and ISO/IEC 14496-3) Formats . . . . . . . 28
8.6 SupportedWMAFormats .............................. 30
8.7 SupportedFLACFormats .............................. 31
8.8 Supported RIFF WAV Formats . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
8.9 SupportedMIDIFormats............................... 32
9 Functional Description 34
9.1 MainFeatures..................................... 34
9.2 DataFlowofVS1053b ................................ 35
9.3 EarSpeaker Spatial Processing . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
9.4 Serial Data Interface (SDI) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
9.5 Serial Control Interface (SCI) . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
9.6 SCIRegisters ..................................... 38
9.6.1 SCI_MODE(RW)............................. 39
9.6.2 SCI_STATUS(RW) ............................ 41
9.6.3 SCI_BASS(RW) ............................. 42
9.6.4 SCI_CLOCKF(RW)............................ 43
9.6.5 SCI_DECODE_TIME (RW) . . . . . . . . . . . . . . . . . . . . . . . 44
9.6.6 SCI_AUDATA(RW) ............................ 44
9.6.7 SCI_WRAM(RW)............................. 44
9.6.8 SCI_WRAMADDR (W) . . . . . . . . . . . . . . . . . . . . . . . . . . 45
9.6.9 SCI_HDAT0 and SCI_HDAT1 (R) . . . . . . . . . . . . . . . . . . . . 45
9.6.10 SCI_AIADDR(RW)............................ 48
9.6.11 SCI_VOL(RW) .............................. 48
9.6.12 SCI_AICTRL[x] (RW) . . . . . . . . . . . . . . . . . . . . . . . . . . . 48
10 Operation 49
10.1 Clocking ........................................ 49
10.2 HardwareReset.................................... 49
10.3 SoftwareReset .................................... 49
10.4 LowPowerMode ................................... 50
10.5 PlayandDecode ................................... 50
10.5.1 Playing a Whole File . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
10.5.2 Cancelling Playback . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
10.5.3 FastPlay.................................. 51
10.5.4 Fast Forward and Rewind without Audio . . . . . . . . . . . . . . . . 52
10.5.5 Maintaining Correct Decode Time . . . . . . . . . . . . . . . . . . . . 52
10.6 FeedingPCMData .................................. 53
10.7 OggVorbisRecording ................................ 53
10.8 PCM/ADPCMRecording .............................. 54
10.8.1 Activating PCM / ADPCM Recording Mode . . . . . . . . . . . . . . . 54
10.8.2 Reading PCM / IMA ADPCM Data . . . . . . . . . . . . . . . . . . . . 55
10.8.3 Adding a PCM RIFF Header . . . . . . . . . . . . . . . . . . . . . . . 56
10.8.4 Adding an IMA ADPCM RIFF Header . . . . . . . . . . . . . . . . . . 57
10.8.5 Playing ADPCM Data . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
10.8.6 Sample Rate Considerations . . . . . . . . . . . . . . . . . . . . . . . 58
10.8.7 Record Monitoring Volume . . . . . . . . . . . . . . . . . . . . . . . . 59
10.9 SPIBoot........................................ 60
10.10Real-TimeMIDI .................................... 60
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VS1053b Datasheet CONTENTS
10.11ExtraParameters ................................... 61
10.11.1 Common Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
10.11.2WMA.................................... 63
10.11.3AAC .................................... 64
10.11.4Midi..................................... 65
10.11.5OggVorbis................................. 66
10.12SDITests ....................................... 67
10.12.1OldSineTest ............................... 67
10.12.2 New Sine and Sweep Tests . . . . . . . . . . . . . . . . . . . . . . . 68
10.12.3PinTest .................................. 68
10.12.4SCITest .................................. 68
10.12.5MemoryTest................................ 69
11 VS1053b Registers 70
11.1 Who Needs to Read This Chapter . . . . . . . . . . . . . . . . . . . . . . . . . . 70
11.2 TheProcessorCore ................................. 70
11.3 VS1053b Hardware DAC Audio Paths . . . . . . . . . . . . . . . . . . . . . . . . 71
11.4 VS1053b Hardware ADC Audio Paths . . . . . . . . . . . . . . . . . . . . . . . . 72
11.5 VS1053bMemoryMap................................ 73
11.6 SCIHardwareRegisters ............................... 73
11.7 Serial Data Interface (SDI) Registers . . . . . . . . . . . . . . . . . . . . . . . . 73
11.8 DACRegisters..................................... 74
11.9 PLLController..................................... 74
11.10GPIO.......................................... 76
11.11InterruptControl.................................... 77
11.12 UART (Universal Asynchronous Receiver/Transmitter) . . . . . . . . . . . . . . 78
11.12.1UARTRegisters.............................. 78
11.12.2 Status UART_STATUS . . . . . . . . . . . . . . . . . . . . . . . . . . 78
11.12.3DataUART_DATA............................. 79
11.12.4 Data High UART_DATAH . . . . . . . . . . . . . . . . . . . . . . . . . 79
11.12.5 Divider UART_DIV . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
11.12.6 UART Interrupts and Operation . . . . . . . . . . . . . . . . . . . . . 80
11.13Timers ......................................... 81
11.13.1TimerRegisters.............................. 81
11.13.2 Configuration TIMER_CONFIG . . . . . . . . . . . . . . . . . . . . . 81
11.13.3 Configuration TIMER_ENABLE . . . . . . . . . . . . . . . . . . . . . 82
11.13.4 Timer X Startvalue TIMER_Tx[L/H] . . . . . . . . . . . . . . . . . . . 82
11.13.5 Timer X Counter TIMER_TxCNT[L/H] . . . . . . . . . . . . . . . . . . 82
11.13.6TimerInterrupts.............................. 82
11.14I2SDACInterface................................... 83
11.15 Analog-to-Digital Converter (ADC) . . . . . . . . . . . . . . . . . . . . . . . . . . 84
11.16 Resampler SampleRate Converter (SRC) . . . . . . . . . . . . . . . . . . . . . 85
11.17 Sidestream Sigma-Delta Modulator (SDM) . . . . . . . . . . . . . . . . . . . . . 86
12 Version Changes 87
12.1 Changes Between VS1033c and VS1053a/b Firmware, 2007-03-08 . . . . . . . 87
13 Latest Document Version Changes 89
14 Contact Information 90
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VS1053b Datasheet LIST OF FIGURES
List of Figures
1 Pin configuration, LQFP-48. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
2 VS1053b in LQFP-48 packaging. . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
3 Typical connection diagram using LQFP-48. . . . . . . . . . . . . . . . . . . . . . 14
4 SDI in VS10xx Native Mode, single-byte transfer . . . . . . . . . . . . . . . . . . 18
5 SDI in VS10xx Native Mode, multi-byte transfer, X≥1............... 18
6 SDItimingdiagram................................... 19
7 SDI in VS1001 Mode - one byte transfer. Do not use in new designs! . . . . . . . 20
8 SDI in VS1001 Mode - two byte transfer. Do not use in new designs! . . . . . . . 20
9 SCIwordread ..................................... 21
10 SCIwordwrite ..................................... 22
11 SCImultiplewordwrite ................................ 22
12 SPItimingdiagram................................... 23
13 TwoSCIoperations................................... 24
14 TwoSDIbytes ..................................... 24
15 Two SDI bytes separated by an SCI operation . . . . . . . . . . . . . . . . . . . . 25
16 DataflowofVS1053b.................................. 35
17 EarSpeaker externalized sound sources vs. normal inside-the-head sound . . . 36
18 VS1053b ADC and DAC data paths with some data registers . . . . . . . . . . . 71
19 VS1053b ADC and DAC data paths with some data registers . . . . . . . . . . . 72
20 RS232 serial interface protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78
21 I2Sinterface,192kHz.................................. 83
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VS1053b Datasheet 3 DEFINITIONS
1 Licenses
VS1053b contains WMA decoding technology from Microsoft.
This product is protected by certain intellectual property rights of Microsoft and cannot
be used or further distributed without a license from Microsoft.
VS1053b contains AAC technology (ISO/IEC 13818-7 and ISO/IEC 14496-3) which cannot be
used without a proper license from Via Licensing Corporation or individual patent holders.
VS1053b contains spectral band replication (SBR) and parametric stereo (PS) technologies
developed by Coding Technologies. Licensing of SBR is handled within MPEG4 through Via
Licensing Corporation. Licensing of PS is handled with Coding Technologies.
See http://www.codingtechnologies.com/licensing/aacplus.htm for more information.
To the best of our knowledge, if the end product does not play a specific format that otherwise
would require a customer license: MPEG 1.0/2.0 layers I and II, WMA, or AAC, the respective
license should not be required. Decoding of MPEG layers I and II are disabled by default,
and WMA and AAC format exclusion can be easily performed based on the contents of the
SCI_HDAT1 register. Also PS and SBR decoding can be separately disabled.
2 Disclaimer
All properties and figures are subject to change.
3 Definitions
BByte, 8 bits.
bBit.
Ki “Kibi” = 210 = 1024 (IEC 60027-2).
Mi “Mebi” = 220 = 1048576 (IEC 60027-2).
VS_DSP VLSI Solution’s DSP core.
WWord. In VS_DSP, instruction words are 32-bit and data words are 16-bit wide.
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VS1053b Datasheet
4 CHARACTERISTICS & SPECIFICATIONS
4 Characteristics & Specifications
4.1 Absolute Maximum Ratings
Parameter Symbol Min Max Unit
Analog Positive Supply AVDD -0.3 3.6 V
Digital Positive Supply CVDD -0.3 1.85 V
I/O Positive Supply IOVDD -0.3 3.6 V
Current at Any Non-Power Pin1±50 mA
Voltage at Any Digital Input -0.3 IOVDD+0.32V
Operating Temperature -40 +85 ◦C
Storage Temperature -65 +150 ◦C
1Higher current can cause latch-up.
2Must not exceed 3.6 V
4.2 Recommended Operating Conditions
Parameter Symbol Min Typ Max Unit
Ambient Operating Temperature -40 +85 ◦C
Analog and Digital Ground 1AGND DGND 0.0 V
Positive Analog, VREF=1.23V 2AVDD12 2.6 2.8 3.6 V
Positive Analog, VREF=1.65V 2AVDD16 3.3 3.3 3.6 V
Positive Digital CVDD 1.7 1.8 1.85 V
I/O Voltage IOVDD 1.8 2.8 3.6 V
Input Clock Frequency 3XTALI 12 12.288 13 MHz
Internal Clock Frequency CLKI 12 36.864 55.3 MHz
Internal Clock Multiplier 4CLKM 1.0×3.0×4.5×
Master Clock Duty Cycle 40 50 60 %
1Must be connected together as close the device as possible for latch-up immunity.
2Reference voltage can be internally selected between 1.23V and 1.65V, see section 9.6.2.
3The maximum sample rate that can be played with correct speed is XTALI/256 (or XTALI/512
if SM_CLK_RANGE is set). Thus, XTALI must be at least 12.288 MHz (24.576 MHz) to be able
to play 48 kHz at correct speed.
4Reset value is 1.0×. Recommended SC_MULT=3.5×, SC_ADD=1.0×(SCI_CLOCKF=0x8800).
Do not exceed maximum specification for CLKI.
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VS1053b Datasheet
4 CHARACTERISTICS & SPECIFICATIONS
4.3 Analog Characteristics
Unless otherwise noted: AVDD=3.3V, CVDD=1.8V, IOVDD=2.8V, REF=1.65V, TA=-30..+85◦C,
XTALI=12..13MHz, Internal Clock Multiplier 3.5×. DAC tested with 1307.894 Hz full-scale output
sinewave, measurement bandwidth 20..20000 Hz, analog output load: LEFT to GBUF 30 Ω,
RIGHT to GBUF 30 Ω. Microphone test amplitude 48 mVpp (differential), fs=1 kHz, Line input
test amplitude 2.52 Vpp, fs=1 kHz.
DAC Characteristics
Parameter Symbol Min Typ Max Unit
DAC Resolution 18 bits
Total Harmonic Distortion, -3 dB of full-scale THD 0.04 %
Third Harmonic Distortion, -3 dB of full-scale 0.01 %
Dynamic Range (DAC unmuted, A-weighted) IDR 100 dB
S/N Ratio (full scale signal) SNR 94 dB
Interchannel Isolation (Cross Talk), 600Ω+ GBUF 80 dB
Interchannel Isolation (Cross Talk), 30Ω+ GBUF 53 dB
Interchannel Gain Mismatch -0.5 0.5 dB
Frequency Response -0.1 0.1 dB
Full Scale Output Voltage LEVEL16 27501mVpp
Full Scale Output Voltage, VREF = 1.2 V LEVEL12 20501mVpp
Deviation from Linear Phase 5 ◦
Analog Output Load Resistance AOLR 16 302Ω
Analog Output Load Capacitance 100 pF
DC level (CBUF, LEFT, RIGHT) VREF16 1.65 V
DC level (CBUF, LEFT, RIGHT), VREF = 1.2 V VREF12 1.23 V
1double can be achieved with +-to-+ wiring for mono difference sound.
2AOLR may be much lower, but below Typical distortion performance may be compromised.
ADC Characteristics
Parameter Symbol Min Typ Max Unit
Microphone input amplifier gain MGAIN 26 dB
Microphone input amplitude (differential) MLEV16 64 1801mVpp AC
Microphone input amplitude (diff.), VREF = 1.2 V MLEV12 48 1401mVpp AC
Microphone Total Harmonic Distortion MTHD 0.03 0.07 %
Microphone S/N Ratio MSNR 60 72 dB
Microphone input impedances, per pin MIMP 45 kΩ
Line input amplitude LLEV16 2500 28001mVpp AC
Line input amplitude, VREF = 1.2 V LLEV12 1900 21001mVpp AC
Line input Total Harmonic Distortion LTHD 0.005 0.014 %
Line input S/N Ratio LSNR 85 90 dB
Line input impedance LIMP 80 kΩ
1Harmonic Distortion increases above typical amplitude.
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VS1053b Datasheet
4 CHARACTERISTICS & SPECIFICATIONS
4.4 Power Consumption
Internal clock multiplier 3.0×. TA=+25◦C. IOVDD =2.8 V, AVDD = 2.6 V, CVDD = 1.8 ˙
V.
XRESET active
Parameter Min Typ Max Unit
Power Supply Consumption IOVDD 0.3 3.0 µA
Power Supply Consumption AVDD 0.6 5.0 µA
Power Supply Consumption CVDD 18 35.0 µA
Full-scale sine in sine test mode
Parameter Min Typ Max Unit
Power Supply Consumption AVDD, no load 5 mA
Power Supply Consumption AVDD, output load 30 Ω+ GBUF 30 37 60 mA
Power Supply Consumption CVDD 8 10 15 mA
128 kbit/s Ogg Vorbis audio plaback, full volume
Parameter Min Typ Max Unit
Power Supply Consumption AVDD, no load 5 mA
Power Supply Consumption AVDD, output load 30 Ω11 mA
Power Supply Consumption AVDD, output load 30 Ω+ GBUF 11 mA
Power Supply Consumption CVDD 11 mA
4.5 Digital Characteristics
Parameter Min Max Unit
High-Level Input Voltage (xRESET, XTALI, XTALO) 0.7×IOVDD IOVDD+0.31V
High-Level Input Voltage (other input pins) 0.7×CVDD IOVDD+0.31V
Low-Level Input Voltage -0.2 0.3×CVDD V
High-Level Output Voltage at XTALO = -0.1 mA 0.7×IOVDD V
Low-Level Output Voltage at XTALO = 0.1 mA 0.3×IOVDD V
High-Level Output Voltage at IO= -1.0 mA 0.7×IOVDD V
Low-Level Output Voltage at IO= 1.0 mA 0.3×IOVDD V
Input Leakage Current -1.0 1.0 µA
SPI Input Clock Frequency 2CLKI
7MHz
Rise time of all output pins, load = 50 pF 50 ns
1Must not exceed 3.6V
2Value for SCI reads. SCI and SDI writes allow CLKI
4.
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VS1053b Datasheet
4 CHARACTERISTICS & SPECIFICATIONS
4.6 Switching Characteristics - Boot Initialization
Parameter Symbol Min Max Unit
XRESET active time 2 XTALI
XRESET inactive to software ready 22000 500001XTALI
Power on reset, rise time to CVDD 10 V/s
1DREQ rises when initialization is complete. You should not send any data or commands
before that.
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VS1053b Datasheet
5 PACKAGES AND PIN DESCRIPTIONS
5 Packages and Pin Descriptions
5.1 Packages
LPQFP-48 is a lead (Pb) free and also RoHS compliant package. RoHS is a short name of
Directive 2002/95/EC on the restriction of the use of certain hazardous substances in electrical
and electronic equipment.
5.1.1 LQFP-48
1
48
Figure 1: Pin configuration, LQFP-48.
LQFP-48 package dimensions are at http://www.vlsi.fi/ .
Figure 2: VS1053b in LQFP-48 packaging.
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VS1053b Datasheet
5 PACKAGES AND PIN DESCRIPTIONS
Pad Name LQFP
Pin
Pin
Type
Function
MICP / LINE1 1 AI Positive differential mic input, self-biasing / Line-in 1
MICN 2 AI Negative differential mic input, self-biasing
XRESET 3 DI Active low asynchronous reset, schmitt-trigger input
DGND0 4 DGND Core & I/O ground
CVDD0 5 CPWR Core power supply
IOVDD0 6 IOPWR I/O power supply
CVDD1 7 CPWR Core power supply
DREQ 8 DO Data request, input bus
GPIO2 / DCLK19 DIO General purpose IO 2 / serial input data bus clock
GPIO3 / SDATA110 DIO General purpose IO 3 / serial data input
GPIO6 / I2S_SCLK311 DIO General purpose IO 6 / I2S_SCLK
GPIO7 /
I2S_SDATA3
12 DIO General purpose IO 7 / I2S_SDATA
XDCS / BSYNC113 DI Data chip select / byte sync
IOVDD1 14 IOPWR I/O power supply
VCO 15 DO For testing only (Clock VCO output)
DGND1 16 DGND Core & I/O ground
XTALO 17 AO Crystal output
XTALI 18 AI Crystal input
IOVDD2 19 IOPWR I/O power supply
DGND2 20 DGND Core & I/O ground
DGND3 21 DGND Core & I/O ground
DGND4 22 DGND Core & I/O ground
XCS 23 DI Chip select input (active low)
CVDD2 24 CPWR Core power supply
GPIO5 / I2S_MCLK325 DIO General purpose IO 5 / I2S_MCLK
RX 26 DI UART receive, connect to IOVDD if not used
TX 27 DO UART transmit
SCLK 28 DI Clock for serial bus
SI 29 DI Serial input
SO 30 DO3 Serial output
CVDD3 31 CPWR Core power supply
XTEST 32 DI Reserved for test, connect to IOVDD
GPIO0 33 DIO Gen. purp. IO 0 (SPIBOOT), use 100 kΩpull-down
resistor2
GPIO1 34 DIO General purpose IO 1
GND 35 DGND I/O Ground
GPIO4 /
I2S_LROUT3
36 DIO General purpose IO 4 / I2S_LROUT
AGND0 37 APWR Analog ground, low-noise reference
AVDD0 38 APWR Analog power supply
RIGHT 39 AO Right channel output
AGND1 40 APWR Analog ground
AGND2 41 APWR Analog ground
GBUF 42 AO Common buffer for headphones, do NOT connect to
ground!
AVDD1 43 APWR Analog power supply
RCAP 44 AIO Filtering capacitance for reference
AVDD2 45 APWR Analog power supply
LEFT 46 AO Left channel output
AGND3 47 APWR Analog ground
LINE2 48 AI Line-in 2 (right channel)
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VS1053b Datasheet
5 PACKAGES AND PIN DESCRIPTIONS
1First pin function is active in New Mode, latter in Compatibility Mode.
2If GPIO0 is high, SPI Boot is tried. See Chapter 10.9 for details.
3If GPIO0 is low and GPIO1 is high, Real-Time MIDI mode is entered. See Chapter 10.10 for
details.
4If I2S_CF_ENA is ’0’ the pins are used for GPIO. See Chapter 11.14 for details.
Pin types:
Type Description
DI Digital input, CMOS Input Pad
DO Digital output, CMOS Input Pad
DIO Digital input/output
DO3 Digital output, CMOS Tri-stated Output
Pad
AI Analog input
Type Description
AO Analog output
AIO Analog input/output
APWR Analog power supply pin
DGND Core or I/O ground pin
CPWR Core power supply pin
IOPWR I/O power supply pin
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VS1053b Datasheet
6 CONNECTION DIAGRAM, LQFP-48
6 Connection Diagram, LQFP-48
Figure 3: Typical connection diagram using LQFP-48.
Figure 3 shows a typical connection diagram for VS1053.
Figure Note 1: Connect either Microphone In or Line In, but not both at the same time.
Note: This connection assumes SM_SDINEW is active (see Chapter 9.6.1). If also SM_SDISHARE
is used, xDCS should be tied low or high (see Chapter 7.1.1).
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6 CONNECTION DIAGRAM, LQFP-48
The common buffer GBUF can be used for common voltage (1.23 V) for earphones. This will
eliminate the need for large isolation capacitors on line outputs, and thus the audio output pins
from VS1053b may be connected directly to the earphone connector.
GBUF must NOT be connected to ground under any circumstances. If GBUF is not used,
LEFT and RIGHT must be provided with coupling capacitors. To keep GBUF stable, you should
always have the resistor and capacitor even when GBUF is not used. See application notes for
details.
Unused GPIO pins should have a pull-down resistor. Unused line and microphone inputs should
not be connected.
If UART is not used, RX should be connected to IOVDD and TX be unconnected.
Do not connect any external load to XTALO.
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VS1053b Datasheet 7 SPI BUSES
7 SPI Buses
The SPI Bus - which was originally used in some Motorola devices - has been used for both
VS1053b’s Serial Data Interface SDI (Chapters 7.3 and 9.4) and Serial Control Interface SCI
(Chapters 7.4 and 9.5).
7.1 SPI Bus Pin Descriptions
7.1.1 VS10xx Native Modes (New Mode, recommended)
These modes are active on VS1053b when SM_SDINEW is set to 1 (default at startup). DCLK
and SDATA are not used for data transfer and they can be used as general-purpose I/O pins
(GPIO2 and GPIO3). BSYNC function changes to data interface chip select (XDCS).
SDI Pin SCI Pin Description
XDCS XCS Active low chip select input. A high level forces the serial interface into
standby mode, ending the current operation. A high level also forces serial
output (SO) to high impedance state. If SM_SDISHARE is 1, pin
XDCS is not used, but the signal is generated internally by inverting
XCS.
SCK Serial clock input. The serial clock is also used internally as the master
clock for the register interface.
SCK can be gated or continuous. In either case, the first rising clock edge
after XCS has gone low marks the first bit to be written.
SI Serial input. If a chip select is active, SI is sampled on the rising CLK edge.
- SO Serial output. In reads, data is shifted out on the falling SCK edge.
In writes SO is at a high impedance state.
7.1.2 VS1001 Compatibility Mode (deprecated, do not use in new designs)
This mode is active when SM_SDINEW is set to 0. In this mode, DCLK, SDATA and BSYNC
are active.
SDI Pin SCI Pin Description
- XCS Active low chip select input. A high level forces the serial interface into
standby mode, ending the current operation. A high level also forces serial
output (SO) to high impedance state.
BSYNC - SDI data is synchronized with a rising edge of BSYNC.
DCLK SCK Serial clock input. The serial clock is also used internally as the master
clock for the register interface.
SCK can be gated or continuous. In either case, the first rising clock edge
after XCS has gone low marks the first bit to be written.
SDATA SI Serial input. SI is sampled on the rising SCK edge, if XCS is low.
- SO Serial output. In reads, data is shifted out on the falling SCK edge.
In writes SO is at a high impedance state.
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VS1053b Datasheet 7 SPI BUSES
7.2 Data Request Pin DREQ
The DREQ pin/signal is used to signal if VS1053b’s 2048-byte FIFO is capable of receiving
data. If DREQ is high, VS1053b can take at least 32 bytes of SDI data or one SCI command.
DREQ is turned low when the stream buffer is too full and for the duration of an SCI command.
Because of the 32-byte safety area, the sender may send up to 32 bytes of SDI data at a
time without checking the status of DREQ, making controlling VS1053b easier for low-speed
microcontrollers.
Note: DREQ may turn low or high at any time, even during a byte transmission. Thus, DREQ
should only be used to decide whether to send more bytes. A transmission that has already
started doesn’t need to be aborted.
Note: In VS1053b DREQ also goes down while an SCI operation is in progress.
There are cases when you still want to send SCI commands when DREQ is low. Because
DREQ is shared between SDI and SCI, you can not determine if an SCI command has been
executed if SDI is not ready to receive data. In this case you need a long enough delay after
every SCI command to make certain none of them are missed. The SCI Registers table in
Chapter 9.6 gives the worst-case handling time for each SCI register write.
Note: The status of DREQ can also be read through SCI with the following code. For details on
SCI registers, see Chapter 7.4.
// This example reads status of DREQ pin through the SPI/SCI register
// interface.
#define SCI_WRAMADDR 7
#define SCI_WRAM 6
while (!endOfFile) {
int dreq;
WriteSciReg(SCI_WRAMADDR, 0xC012); // Send address of DREQ register
dreq = ReadSciReg(SCI_WRAM) & 1; // Read value of DREQ (in bit 0)
if (dreq) {
// DREQ high: send 1-32 bytes audio data
} else {
// DREQ low: wait 5 milliseconds (so that VS10xx doesn't get
// continuous SCI operations)
}
} /* while (!endOfFile) */
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VS1053b Datasheet 7 SPI BUSES
7.3 Serial Protocol for Serial Data Interface (SPI / SDI)
The serial data interface operates in slave mode so DCLK signal must be generated by an
external circuit.
Data (SDATA signal) can be clocked in at either the rising or falling edge of DCLK (Chapter 9.6).
VS1053b assumes its data input to be byte-sychronized. SDI bytes may be transmitted either
MSb or LSb first, depending of register SCI_MODE bit SM_SDIORD (Chapter 9.6.1).
The firmware is able to accept the maximum bitrate the SDI supports.
7.3.1 SDI in VS10xx Native Modes (New Mode, recommended)
DCLK
D7 D6 D5 D4 D3 D1D2 D0SDATA
XDCS
Figure 4: SDI in VS10xx Native Mode, single-byte transfer
In VS10xx native modes (SM_NEWMODE is 1), byte synchronization is achieved by XDCS, as
shown in Figure 4. The state of XDCS may not change while a data byte transfer is in progress.
XDCS does not need to be deactivated and reactivated for every byte transfer, as shown in
Figure 5. However, to maintain data synchronization even if there are occasional clock glitches,
it is recommended to deactivate and reactivate XDCS every now and then, for example after
each 32 bytes of data.
Note that when sending data through SDI you have to check the Data Request Pin DREQ at
least after every 32 bytes (Chapter 7.2).
DCLK
SDATA D7 D6 D5 D4 D3 D1D2 D0 D7 D6 D5 ... D3 D1D2 D0
Byte XByte 1 Byte 2
...
XDCS
Figure 5: SDI in VS10xx Native Mode, multi-byte transfer, X≥1
If SM_SDISHARE is 1, the XDCS signal is internally generated by inverting the XCS input.
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VS1053b Datasheet 7 SPI BUSES
7.3.2 SDI Timing Diagram in VS10xx Native Modes (New Mode)
SCK
SI
tXCSS tXCSHtWL tWH
tH
tSU
tXCS
xDCS
D7 D6 D5 D4 D3 D2 D1 D0
Figure 6: SDI timing diagram
Figure 6 presents SDI bus timing.
Symbol Min Max Unit
tXCSS 5 ns
tSU 0 ns
tH 2 CLKI cycles
tWL 2 CLKI cycles
tWH 2 CLKI cycles
tXCSH 1 CLKI cycles
tXCS 0 CLKI cycles
Note: xDCS is not required to go high between bytes, so tXCS is 0.
Note: Although the timing is derived from the internal clock CLKI, the system always starts up in
1.0×mode, thus CLKI=XTALI. After you have configured a higher clock through SCI_CLOCKF
and waited for DREQ to rise, you can use a higher SPI speed as well.
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7.3.3 SDI in VS1001 Compatibility Mode (deprecated, do not use in new designs)
DCLK
D7 D6 D5 D4 D3 D1D2 D0SDATA
BSYNC
Figure 7: SDI in VS1001 Mode - one byte transfer. Do not use in new designs!
When VS1053b is running in VS1001 compatibility mode, a BSYNC signal must be generated
to ensure correct bit-alignment of the input bitstream, as shown in Figures 7 and 8.
The first DCLK sampling edge (rising or falling, depending on selected polarity), during which
the BSYNC is high, marks the first bit of a byte (LSB, if LSB-first order is used, MSB, if MSB-first
order is used). If BSYNC is ’1’ when the last bit is received, the receiver stays active and next
8 bits are also received.
DCLK
D7 D6 D5 D4 D3 D1D2 D0 D7 D6 D5 D4 D3 D2 D1 D0SDATA
BSYNC
Figure 8: SDI in VS1001 Mode - two byte transfer. Do not use in new designs!
7.3.4 Passive SDI Mode (deprecated, do not use in new designs)
If SM_NEWMODE is 0 and SM_SDISHARE is 1, the operation is otherwise like the VS1001
compatibility mode, but bits are only received while the BSYNC signal is ’1’. Rising edge of
BSYNC is still used for synchronization.
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VS1053b Datasheet 7 SPI BUSES
7.4 Serial Protocol for Serial Command Interface (SPI / SCI)
The serial bus protocol for the Serial Command Interface SCI (Chapter 9.5) consists of an
instruction byte, address byte and one 16-bit data word. Each read or write operation can read
or write a single register. Data bits are read at the rising edge, so the user should update data
at the falling edge. Bytes are always send MSb first. XCS should be low for the full duration of
the operation, but you can have pauses between bits if needed.
The operation is specified by an 8-bit instruction opcode. The supported instructions are read
and write. See table below.
Instruction
Name Opcode Operation
READ 0b0000 0011 Read data
WRITE 0b0000 0010 Write data
Note: VS1053b sets DREQ low after each SCI operation. The duration depends on the opera-
tion. It is not allowed to finish a new SCI/SDI operation before DREQ is high again.
7.4.1 SCI Read
0 1 2 3 4 5 6 7 8 9 10 11 12 13 30 3114 15 16 17
0 0 0 0 0 0 1 1 0 0 0 0
3 2 1 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
15 14 1 0
X
instruction (read) address data out
XCS
SCK
SI
SO
don’t care don’t care
DREQ
execution
Figure 9: SCI word read
VS1053b registers are read from using the following sequence, as shown in Figure 9. First,
XCS line is pulled low to select the device. Then the READ opcode (0x3) is transmitted via
the SI line followed by an 8-bit word address. After the address has been read in, any further
data on SI is ignored by the chip. The 16-bit data corresponding to the received address will be
shifted out onto the SO line.
XCS should be driven high after data has been shifted out.
DREQ is driven low for a short while when in a read operation by the chip. This is a very short
time and doesn’t require special user attention.
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7.4.2 SCI Write
0 1 2 3 4 5 6 7 8 9 10 11 12 13 30 3114 15 16 17
0 0 0 0 0 0 1 0 0 0 0
3 2 1 0 1 0
X
address
XCS
SCK
SI
15 14
data out
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0SO 0 0 0 0 X
0
instruction (write)
DREQ
execution
Figure 10: SCI word write
VS1053b registers are written from using the following sequence, as shown in Figure 10. First,
XCS line is pulled low to select the device. Then the WRITE opcode (0x2) is transmitted via the
SI line followed by an 8-bit word address.
After the word has been shifted in and the last clock has been sent, XCS should be pulled high
to end the WRITE sequence.
After the last bit has been sent, DREQ is driven low for the duration of the register update,
marked “execution” in the figure. The time varies depending on the register and its contents
(see table in Chapter 9.6 for details). If the maximum time is longer than what it takes from the
microcontroller to feed the next SCI command or SDI byte, status of DREQ must be checked
before finishing the next SCI/SDI operation.
7.4.3 SCI Multiple Write
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17
0 0 0 0 0 0 1 0 0 0 0
3 2 1 0
address
XCS
SCK
SI
15 14
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0SO 0 0
0
instruction (write)
DREQ
1 0
X
0 0 X
execution
10 15 14
data out 1 data out 2
0 0 0 0
execution
X
31
30 32 3329
d.out n
m−2m−1
Figure 11: SCI multiple word write
VS1053b allows for the user to send multiple words to the same SCI register, which allows
fast SCI uploads, shown in Figure 11. The main difference to a single write is that instead of
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VS1053b Datasheet 7 SPI BUSES
bringing XCS up after sending the last bit of a data word, the next data word is sent immediately.
After the last data word, XCS is driven high as with a single word write.
After the last bit of a word has been sent, DREQ is driven low for the duration of the register
update, marked “execution” in the figure. The time varies depending on the register and its
contents (see table in Chapter 9.6 for details). If the maximum time is longer than what it takes
from the microcontroller to feed the next SCI command or SDI byte, status of DREQ must be
checked before finishing the next SCI/SDI operation.
7.4.4 SCI Timing Diagram
Figure 12: SPI timing diagram
The SCI timing diagram is presented in Figure 12.
Symbol Min Max Unit
tXCSS 5 ns
tSU 0 ns
tH 2 CLKI cycles
tZ 0 ns
tWL 2 CLKI cycles
tWH 2 CLKI cycles
tV 2 (+ 25 ns1) CLKI cycles
tXCSH 1 CLKI cycles
tXCS 2 CLKI cycles
tDIS 10 ns
125 ns is when pin loaded with 100 pF capacitance. The time is shorter with lower capacitance.
Note: Although the timing is derived from the internal clock CLKI, the system always starts up in
1.0×mode, thus CLKI=XTALI. After you have configured a higher clock through SCI_CLOCKF
and waited for DREQ to rise, you can use a higher SPI speed as well.
Note: Because tWL + tWH + tH is 6×CLKI + 25 ns, the maximum speed for SCI reads is CLKI/7.
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7.5 SPI Examples with SM_SDINEW and SM_SDISHARED set
7.5.1 Two SCI Writes
01 2 3 30 31
1 0 1 0
0 0 0 0 0 0
X X
XCS
SCK
SI
2
32 33 61 62 63
SCI Write 1 SCI Write 2
DREQ
DREQ up before finishing next SCI write
Figure 13: Two SCI operations
Figure 13 shows two consecutive SCI operations. Note that xCS must be raised to inactive
state between the writes. Also DREQ must be respected as shown in the figure.
7.5.2 Two SDI Bytes
1 2 3
XCS
SCK
SI
7 6 5 4 3 1 0 7 6 5 2 1 0
X
SDI Byte 1 SDI Byte 2
0 6 7 8 9 13 14 15
DREQ
Figure 14: Two SDI bytes
SDI data is synchronized with a raising edge of xCS as shown in Figure 14. However, every
byte doesn’t need separate synchronization.
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VS1053b Datasheet 7 SPI BUSES
7.5.3 SCI Operation in Middle of Two SDI Bytes
01
XCS
SCK
SI
7
7 6 5 1
0 0
0 7 6 5 1 0
SDI Byte SCI Operation SDI Byte
8 9 39 40 41 46 47
X
DREQ high before end of next transfer
DREQ
Figure 15: Two SDI bytes separated by an SCI operation
Figure 15 shows how an SCI operation is embedded in between SDI operations. xCS edges
are used to synchronize both SDI and SCI. Remember to respect DREQ as shown in the figure.
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8 SUPPORTED AUDIO DECODER FORMATS
8 Supported Audio Decoder Formats
Conventions
Mark Description
+ Format is supported
? Format is supported but not thoroughly tested
- Format exists but is not supported
Format doesn’t exist
8.1 Supported MP3 (MPEG layer III) Formats
MPEG 1.01:
Samplerate / Hz Bitrate / kbit/s
32 40 48 56 64 80 96 112 128 160 192 224 256 320
48000 + + + + + + + + + + + + + +
44100 + + + + + + + + + + + + + +
32000 + + + + + + + + + + + + + +
MPEG 2.01:
Samplerate / Hz Bitrate / kbit/s
8 16 24 32 40 48 56 64 80 96 112 128 144 160
24000 + + + + + + + + + + + + + +
22050 + + + + + + + + + + + + + +
16000 + + + + + + + + + + + + + +
MPEG 2.51:
Samplerate / Hz Bitrate / kbit/s
8 16 24 32 40 48 56 64 80 96 112 128 144 160
12000 + + + + + + + + + + + + + +
11025 + + + + + + + + + + + + + +
8000 + + + + + + + + + + + + + +
1Also all variable bitrate (VBR) formats are supported.
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8.2 Supported MP2 (MPEG layer II) Formats
Note: Layer I / II decoding must be specifically enabled from register SCI_MODE.
MPEG 1.0:
Samplerate / Hz Bitrate / kbit/s
32 48 56 64 80 96 112 128 160 192 224 256 320 384
48000 + + + + + + + + + + + + + +
44100 + + + + + + + + + + + + + +
32000 + + + + + + + + + + + + + +
MPEG 2.0:
Samplerate / Hz Bitrate / kbit/s
8 16 24 32 40 48 56 64 80 96 112 128 144 160
24000 + + + + + + + + + + + + + +
22050 + + + + + + + + + + + + + +
16000 + + + + + + + + + + + + + +
8.3 Supported MP1 (MPEG layer I) Formats
Note: Layer I / II decoding must be specifically enabled from register SCI_MODE.
MPEG 1.0:
Samplerate / Hz Bitrate / kbit/s
32 64 96 128 160 192 224 256 288 320 352 384 416 448
48000 + + + + + + + + + + + + + +
44100 + + + + + + + + + + + + + +
32000 + + + + + + + + + + + + + +
MPEG 2.0:
Samplerate / Hz Bitrate / kbit/s
32 48 56 64 80 96 112 128 144 160 176 192 224 256
24000 ? ? ? ? ? ? ? ? ? ? ? ? ? ?
22050 ? ? ? ? ? ? ? ? ? ? ? ? ? ?
16000 ? ? ? ? ? ? ? ? ? ? ? ? ? ?
8.4 Supported Ogg Vorbis Formats
Parameter Min Max Unit
Channels 2
Window size 64 4096 samples
Samplerate 100 48000 Hz
Bitrate 500 kbit/sec
Only floor 1 is supported. No known current encoder uses floor 0. All one- and two-channel
Ogg Vorbis files should be playable with this decoder.
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8 SUPPORTED AUDIO DECODER FORMATS
8.5 Supported AAC (ISO/IEC 13818-7 and ISO/IEC 14496-3) Formats
VS1053b decodes MPEG2-AAC-LC-2.0.0.0 and MPEG4-AAC-LC-2.0.0.0 streams, i.e. the low
complexity profile with maximum of two channels can be decoded. If a stream contains more
than one element and/or element type, you can select which one to decode from the 16 single-
channel, 16 channel-pair, and 16 low-frequency elements. The default is to select the first one
that appears in the stream.
Dynamic range control (DRC) is supported and can be controlled by the user to limit or enhance
the dynamic range of the material that contains DRC information.
Both Sine window and Kaiser-Bessel-derived window are supported. For MPEG4 pseudo-
random noise substitution (PNS) is supported. Short frames (120 and 960 samples) are not
supported.
Spectral Band Replication (SBR) level 3, and Parametric Stereo (PS) level 3 are supported (HE-
AAC v2). Level 3 means that maximum of 2 channels, samplerates up to and including 48 kHz
without and with SBR (with or without PS) are supported. Also, both mixing modes (Raand Rb),
IPD/OPD synthesis and 34 frequency bands resolution are implemented. The downsampled
synthesis mode (core coder rates >24 kHz and <=48 kHz with SBR) is implemented.
SBR and PS decoding can also be disabled. Also different operating modes can be selected.
See
config1
and
sbrAndPsStatus
in section 10.11 : "Extra parameters".
If enabled, the internal clock (CLKI) is automatically increased if AAC decoding needs a higher
clock. PS and SBR operation is automatically switched off if the internal clock is too slow for
correct decoding. Generally HE-AAC v2 files need 4.5×clock to decode both SBR and PS
content. This is why 3.5×+ 1.0×clock is the recommended default.
For AAC the streaming ADTS format is recommended. This format allows easy rewind and fast
forward because resynchronization is easily possible.
In addition to ADTS (.aac), MPEG2 ADIF (.aac) and MPEG4 AUDIO (.mp4 / .m4a) files are
played, but these formats are less suitable for rewind and fast forward operations. You can still
implement these features by using the safe jump points table, or using slightly less robust but
much easier automatic resync mechanism (see Section 10.5.4).
Because 3GPP (.3gp) and 3GPPv2 (.3g2) files are just MPEG4 files, those that contain only
HE-AAC or HE-AACv2 content are played.
Note: To be able to play the .3gp, .3g2, .mp4 and .m4a files, the mdat atom must be the
last atom in the MP4 file. Because VS1053b receives all data as a stream, all metadata must
be available before the music data is received. Several MP4 file formatters do not satisfy this
requirement and some kind of conversion is required. This is also why the streamable ADTS
format is recommended.
Programs exist that optimize the .mp4 and .m4a into so-called streamable format that has the
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8 SUPPORTED AUDIO DECODER FORMATS
mdat atom last in the file, and thus suitable for web servers’ audio streaming. You can use this
kind of tool to process files for VS1053b too. For example
mp4creator -optimize file.mp4
.
AAC12:
Samplerate / Hz Maximum Bitrate kbit/s - for 2 channels
≤96 132 144 192 264 288 384 529 576
48000 + + + + + + + + +
44100 + + + + + + + +
32000 + + + + + + +
24000 + + + + + +
22050 + + + + +
16000 + + + +
12000 + + +
11025 + +
8000 +
164000 Hz, 88200 Hz, and 96000 Hz AAC files are played at the highest possible samplerate
(48000 Hz with 12.288 MHz XTALI).
2Also all variable bitrate (VBR) formats are supported. Note that the table gives the maximum
bitrate allowed for two channels for a specific samplerate as defined by the AAC specification.
The decoder does not actually have a fixed lower or upper limit.
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8 SUPPORTED AUDIO DECODER FORMATS
8.6 Supported WMA Formats
Windows Media Audio codec versions 2, 7, 8, and 9 are supported. All WMA profiles (L1,
L2, and L3) are supported. Previously streams were separated into Classes 1, 2a, 2b, and
3. The decoder has passed Microsoft’s conformance testing program. Windows Media Audio
Professional is a different codec and is not supported.
WMA 4.0 / 4.1:
Samplerate Bitrate / kbit/s
/ Hz 5 6 8 10 12 16 20 22 32 40 48 64 80 96 128 160 192
8000 + + + +
11025 + +
16000 + + + +
22050 + + + +
32000 + + + + + +
44100 + + + + + + +
48000 + +
WMA 7:
Samplerate Bitrate / kbit/s
/ Hz 5 6 8 10 12 16 20 22 32 40 48 64 80 96 128 160 192
8000 + + + +
11025 + +
16000 + + + +
22050 + + + +
32000 + + + +
44100 + + + + + + + +
48000 + +
WMA 8:
Samplerate Bitrate / kbit/s
/ Hz 5 6 8 10 12 16 20 22 32 40 48 64 80 96 128 160 192
8000 + + + +
11025 + +
16000 + + + +
22050 + + + +
32000 + + + +
44100 + + + + + + + +
48000 + + +
WMA 9:
Samplerate Bitrate / kbit/s
/ Hz 5 6 8 10 12 16 20 22 32 40 48 64 80 96 128 160 192 256 320
8000 + + + +
11025 + +
16000 + + + +
22050 + + + +
32000 + + + +
44100 + + + + + + + + + + +
48000 + + + + +
In addition to these expected WMA decoding profiles, all other bitrate and samplerate com-
binations are supported, including variable bitrate WMA streams. Note that WMA does not
consume the bitstream as evenly as MP3, so you need a higher peak transfer capability for
clean playback at the same bitrate.
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8 SUPPORTED AUDIO DECODER FORMATS
8.7 Supported FLAC Formats
Up to 48 kHz and 24-bit FLAC files are supported with the VS1053b Patches w/ FLAC Decoder
plugin that is available at http://www.vlsi.fi/en/support/software/vs10xxpatches.html . Read the
accompanying documentation of the plugin for details.
8.8 Supported RIFF WAV Formats
The most common RIFF WAV subformats are supported, with 1 or 2 audio channels.
Format Name Supported Comments
0x01 PCM + 16 and 8 bits, any samplerate ≤48kHz
0x02 ADPCM -
0x03 IEEE_FLOAT -
0x06 ALAW -
0x07 MULAW -
0x10 OKI_ADPCM -
0x11 IMA_ADPCM + Any samplerate ≤48kHz
0x15 DIGISTD -
0x16 DIGIFIX -
0x30 DOLBY_AC2 -
0x31 GSM610 -
0x3b ROCKWELL_ADPCM -
0x3c ROCKWELL_DIGITALK -
0x40 G721_ADPCM -
0x41 G728_CELP -
0x50 MPEG -
0x55 MPEGLAYER3 + For supported MP3 modes, see Chapter 8.1
0x64 G726_ADPCM -
0x65 G722_ADPCM -
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8 SUPPORTED AUDIO DECODER FORMATS
8.9 Supported MIDI Formats
General MIDI and SP-MIDI format 0 files are played. Format 1 and 2 files must be converted to
format 0 by the user. The maximum polyphony is 64, the maximum sustained polyphony is 40.
Actual polyphony depends on the internal clock rate (which is user-selectable), the instruments
used, whether the reverb effect is enabled, and the possible global postprocessing effects en-
abled, such as bass enhancer, treble control or EarSpeaker spatial processing. The polyphony
restriction algorithm makes use of the SP-MIDI MIP table, if present, and uses smooth note
removal.
43 MHz (3.5×input clock) achieves 19-31 simultaneous sustained notes. The instantaneous
amount of notes can be larger. This is a fair compromise between power consumption and
quality, but higher clocks can be used to increase polyphony.
Reverb effect can be controlled by the user. In addition to reverb automatic and reverb off
modes, 14 different decay times can be selected. These roughly correspond to different room
sizes. Also, each midi song decides how much effect each instrument gets. Because the reverb
effect uses about 4 MHz of processing power the automatic control enables reverb only when
the internal clock is at least 3.0×.
In VS1053b both EarSpeaker and MIDI reverb can be on simultaneously. This is ideal for
listening MIDI songs with headphones.
New instruments have been implemented in addition to the 36 that are available in VS1003.
VS1053b now has unique instruments in the whole GM1 instrument set and one bank of GM2
percussions.
Supported MIDI messages:
•meta: 0x51 : set tempo
•other meta: MidiMeta() called
•device control: 0x01 : master volume
•channel message: 0x80 note off, 0x90 note on, 0xc0 program, 0xe0 pitch wheel
•channel message 0xb0: parameter
–0x00: bank select (0 is default, 0x78 and 0x7f is drums, 0x79 melodic)
–0x06: RPN MSB: 0 = bend range, 2 = coarse tune
–0x07: channel volume
–0x0a: pan control
–0x0b: expression (changes volume)
–0x0c: effect control 1 (sets global reverb decay)
–0x26: RPN LSB: 0 = bend range
–0x40: hold1
–0x42: sustenuto
–0x5b effects level (channel reverb level)
–0x62,0x63,0x64,0x65: NRPN and RPN selects
–0x78: all sound off
–0x79: reset all controllers
–0x7b, 0x7c, 0x7d: all notes off
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8 SUPPORTED AUDIO DECODER FORMATS
VS1053b Melodic Instruments (GM1)
1 Acoustic Grand Piano 33 Acoustic Bass 65 Soprano Sax 97 Rain (FX 1)
2 Bright Acoustic Piano 34 Electric Bass (finger) 66 Alto Sax 98 Sound Track (FX 2)
3 Electric Grand Piano 35 Electric Bass (pick) 67 Tenor Sax 99 Crystal (FX 3)
4 Honky-tonk Piano 36 Fretless Bass 68 Baritone Sax 100 Atmosphere (FX 4)
5 Electric Piano 1 37 Slap Bass 1 69 Oboe 101 Brightness (FX 5)
6 Electric Piano 2 38 Slap Bass 2 70 English Horn 102 Goblins (FX 6)
7 Harpsichord 39 Synth Bass 1 71 Bassoon 103 Echoes (FX 7)
8 Clavi 40 Synth Bass 2 72 Clarinet 104 Sci-fi (FX 8)
9 Celesta 41 Violin 73 Piccolo 105 Sitar
10 Glockenspiel 42 Viola 74 Flute 106 Banjo
11 Music Box 43 Cello 75 Recorder 107 Shamisen
12 Vibraphone 44 Contrabass 76 Pan Flute 108 Koto
13 Marimba 45 Tremolo Strings 77 Blown Bottle 109 Kalimba
14 Xylophone 46 Pizzicato Strings 78 Shakuhachi 110 Bag Pipe
15 Tubular Bells 47 Orchestral Harp 79 Whistle 111 Fiddle
16 Dulcimer 48 Timpani 80 Ocarina 112 Shanai
17 Drawbar Organ 49 String Ensembles 1 81 Square Lead (Lead 1) 113 Tinkle Bell
18 Percussive Organ 50 String Ensembles 2 82 Saw Lead (Lead) 114 Agogo
19 Rock Organ 51 Synth Strings 1 83 Calliope Lead (Lead 3) 115 Pitched Percussion
20 Church Organ 52 Synth Strings 2 84 Chiff Lead (Lead 4) 116 Woodblock
21 Reed Organ 53 Choir Aahs 85 Charang Lead (Lead 5) 117 Taiko Drum
22 Accordion 54 Voice Oohs 86 Voice Lead (Lead 6) 118 Melodic Tom
23 Harmonica 55 Synth Voice 87 Fifths Lead (Lead 7) 119 Synth Drum
24 Tango Accordion 56 Orchestra Hit 88 Bass + Lead (Lead 8) 120 Reverse Cymbal
25 Acoustic Guitar (nylon) 57 Trumpet 89 New Age (Pad 1) 121 Guitar Fret Noise
26 Acoustic Guitar (steel) 58 Trombone 90 Warm Pad (Pad 2) 122 Breath Noise
27 Electric Guitar (jazz) 59 Tuba 91 Polysynth (Pad 3) 123 Seashore
28 Electric Guitar (clean) 60 Muted Trumpet 92 Choir (Pad 4) 124 Bird Tweet
29 Electric Guitar (muted) 61 French Horn 93 Bowed (Pad 5) 125 Telephone Ring
30 Overdriven Guitar 62 Brass Section 94 Metallic (Pad 6) 126 Helicopter
31 Distortion Guitar 63 Synth Brass 1 95 Halo (Pad 7) 127 Applause
32 Guitar Harmonics 64 Synth Brass 2 96 Sweep (Pad 8) 128 Gunshot
VS1053b Percussion Instruments (GM1+GM2)
27 High Q 43 High Floor Tom 59 Ride Cymbal 2 75 Claves
28 Slap 44 Pedal Hi-hat [EXC 1] 60 High Bongo 76 Hi Wood Block
29 Scratch Push [EXC 7] 45 Low Tom 61 Low Bongo 77 Low Wood Block
30 Scratch Pull [EXC 7] 46 Open Hi-hat [EXC 1] 62 Mute Hi Conga 78 Mute Cuica [EXC 4]
31 Sticks 47 Low-Mid Tom 63 Open Hi Conga 79 Open Cuica [EXC 4]
32 Square Click 48 High Mid Tom 64 Low Conga 80 Mute Triangle [EXC 5]
33 Metronome Click 49 Crash Cymbal 1 65 High Timbale 81 Open Triangle [EXC 5]
34 Metronome Bell 50 High Tom 66 Low Timbale 82 Shaker
35 Acoustic Bass Drum 51 Ride Cymbal 1 67 High Agogo 83 Jingle bell
36 Bass Drum 1 52 Chinese Cymbal 68 Low Agogo 84 Bell tree
37 Side Stick 53 Ride Bell 69 Cabasa 85 Castanets
38 Acoustic Snare 54 Tambourine 70 Maracas 86 Mute Surdo [EXC 6]
39 Hand Clap 55 Splash Cymbal 71 Short Whistle [EXC 2] 87 Open Surdo [EXC 6]
40 Electric Snare 56 Cowbell 72 Long Whistle [EXC 2]
41 Low Floor Tom 57 Crash Cymbal 2 73 Short Guiro [EXC 3]
42 Closed Hi-hat [EXC 1] 58 Vibra-slap 74 Long Guiro [EXC 3]
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9 FUNCTIONAL DESCRIPTION
9 Functional Description
9.1 Main Features
VS1053b is based on a proprietary digital signal processor, VS_DSP. It contains all the code
and data memory needed for Ogg Vorbis, MP3, AAC, WMA and WAV PCM + ADPCM audio
decoding and a MIDI synthesizer, together with serial interfaces, a multirate stereo audio DAC
and analog output amplifiers and filters. Also PCM/ADPCM audio encoding is supported us-
ing a microphone amplifier and/or line-level inputs and a stereo A/D converter. With software
plugins the chip can also decode lossless FLAC as well as record the high-quality Ogg Vorbis
format. A UART is provided for debugging purposes.
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9.2 Data Flow of VS1053b
Volume
control
Audio
FIFO S.rate.conv.
and DAC R
Bitstream
FIFO
SDI
L
SCI_VOL
SM_ADPCM=0
2048 stereo
samples
Bass
enhancer
SB_AMPLITUDE=0
SB_AMPLITUDE!=0
AIADDR = 0
AIADDR != 0
User
Application
ST_AMPLITUDE=0
ST_AMPLITUDE!=0
Ear
Speaker
MP3 MP2 MP1
WAV ADPCM
WMA AAC
MIDI Vorbis
Treble
control
Figure 16: Data flow of VS1053b.
First, depending on the audio data, and provided ADPCM encoding mode is not set, Ogg
Vorbis, PCM WAV or IMA ADPCM WAV is received and decoded from the SDI bus.
After decoding, if SCI_AIADDR is non-zero, application code is executed from the address
pointed to by that register. For more details, see Application Notes for VS10XX.
Then data may be sent to the Bass Enhancer and Treble Control depending on the SCI_BASS
register.
Next, headphone processing is performed, if the EarSpeaker spatial processing is active.
After that the data to the Audio FIFO, which holds the data until it is read by the Audio interrupt
and fed to the samplerate converter and DACs. The size of the audio FIFO is 2048 stereo
(2×16-bit) samples, or 8 KiB.
The samplerate converter upsamples all different samplerates to XTALI/2, or 128 times the
highest usable samplerate with 18-bit precision. Volume control is performed in the upsampled
domain. New volume settings are loaded only when the upsampled signal crosses the zero
point (or after a timeout). This zero-crossing detection almost completely removes all audible
noise that occurs when volume is suddenly changed.
The samplerate conversion to a common samplerate removes the need for complex PLL-based
clocking schemes and allows almost unlimited sample rate accuracy with one fixed input clock
frequency. With a 12.288 MHz clock, the DA converter operates at 128 ×48 kHz, i.e. 6.144
MHz, and creates a stereo in-phase analog signal. The oversampled output is low-pass filtered
by an on-chip analog filter. This signal is then forwarded to the earphone amplifier.
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9.3 EarSpeaker Spatial Processing
While listening to headphones the sound has a tendency to be localized inside the head. The
sound field becomes flat and lacking the sensation of dimensions. This is an unnatural, awk-
ward and sometimes even disturbing situation. This phenomenon is often referred in literature
as ‘lateralization’, meaning ’in-the-head’ localization. Long-term listening to lateralized sound
may lead to listening fatigue.
All real-life sound sources are external, leaving traces to the acoustic wavefront that arrives to
the ear drums. From these traces, the auditory system of the brain is able to judge the distance
and angle of each sound source. In loudspeaker listening the sound is external and these
traces are available. In headphone listening these traces are missing or ambiguous.
EarSpeaker processes sound to make listening via headphones more like listening to the same
music from real loudspeakers or live music. Once EarSpeaker processing is activated, the
instruments are moved from inside to the outside of the head, making it easier to separate
the different instruments (see figure 17). The listening experience becomes more natural and
pleasant, and the stereo image is sharper as the instruments are widely on front of the listener
instead of being inside the head.
Figure 17: EarSpeaker externalized sound sources vs. normal inside-the-head sound
Note that EarSpeaker differs from any common spatial processing effects, such as echo, reverb,
or bass boost. EarSpeaker accurately simulates the human auditory model and real listening
environment acoustics. Thus is does not change the tonal character of the music by introducing
artificial effects.
EarSpeaker processing can be parameterized to a few different modes, each simulating a little
different type of acoustical situation, suiting different personal preferences and types of record-
ing. See section 9.6.1 for how to activate different modes.
•Off: Best option when listening through loudspeakers or if the audio to be played contains
binaural preprocessing.
•minimal: Suited for listening to normal musical scores with headphones, very subtle.
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•normal: Suited for listening to normal musical scores with headphones, moves sound
source further away than minimal.
•extreme: Suited for old or ’dry’ recordings, or if the audio to be played is artificial, for
example generated MIDI.
9.4 Serial Data Interface (SDI)
The serial data interface is meant for transferring compressed data for the different decoders of
VS1053b.
If the input of the decoder is invalid or it is not received fast enough, analog outputs are auto-
matically muted.
Also several different tests may be activated through SDI as described in Chapter 10.
9.5 Serial Control Interface (SCI)
The serial control interface is compatible with the SPI bus specification. Data transfers are
always 16 bits. VS1053b is controlled by writing and reading the registers of the interface.
The main controls of the serial control interface are:
•control of the operation mode, clock, and builtin effects
•access to status information and header data
•receiving encoded data in recording mode
•uploading and controlling user programs
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9.6 SCI Registers
VS1053b sets DREQ low when it detects an SCI operation (this delay is 16 to 40 CLKI cycles
depending on whether an interrupt service routine is active) and restores it when it has pro-
cessed the operation. The duration depends on the operation. If DREQ is low when an SCI
operation is performed, it also stays low after SCI operation processing.
If DREQ is high before a SCI operation, do not start a new SCI/SDI operation before DREQ is
high again. If DREQ is low before a SCI operation because the SDI can not accept more data,
make certain there is enough time to complete the operation before sending another.
SCI registers, prefix SCI_
Reg Type Reset Time1Abbrev[bits] Description
0x0 rw 0x4000680 CLKI4MODE Mode control
0x1 rw 0x000C380 CLKI STATUS Status of VS1053b
0x2 rw 0 80 CLKI BASS Built-in bass/treble control
0x3 rw 0 1200 XTALI5CLOCKF Clock freq + multiplier
0x4 rw 0 100 CLKI DECODE_TIME Decode time in seconds
0x5 rw 0 450 CLKI2AUDATA Misc. audio data
0x6 rw 0 100 CLKI WRAM RAM write/read
0x7 rw 0 100 CLKI WRAMADDR Base address for RAM
write/read
0x8 r 0 80 CLKI HDAT0 Stream header data 0
0x9 r 0 80 CLKI HDAT1 Stream header data 1
0xA rw 0 210 CLKI2AIADDR Start address of application
0xB rw 0 80 CLKI VOL Volume control
0xC rw 0 80 CLKI2AICTRL0 Application control register 0
0xD rw 0 80 CLKI2AICTRL1 Application control register 1
0xE rw 0 80 CLKI2AICTRL2 Application control register 2
0xF rw 0 80 CLKI2AICTRL3 Application control register 3
1This is the worst-case time that DREQ stays low after writing to this register. The user may
choose to skip the DREQ check for those register writes that take less than 100 clock cycles to
execute and use a fixed delay instead.
2In addition, the cycles spent in the user application routine must be counted.
3Firmware changes the value of this register immediately to 0x48 (analog enabled), and after
a short while to 0x40 (analog drivers enabled).
4When mode register write specifies a software reset the worst-case time is 22000 XTALI
cycles.
5If the clock multiplier is changed, writing to CLOCKF register may force internal clock to run
at 1.0×XTALI for a while. Thus it is not a good idea to send SCI or SDI bits while this register
update is in progress.
6Firmware changes the value of this register immediately to 0x4800.
Reads from all SCI registers complete in under 100 CLKI cycles, except a read from AIADDR
in 200 cycles. In addition the cycles spent in the user application routine must be counted to
the read time of AIADDR, AUDATA, and AICTRL0..3.
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9.6.1 SCI_MODE (RW)
SCI_MODE is used to control the operation of VS1053b and defaults to 0x4800 (SM_SDINEW
set).
Bit Name Function Value Description
0 SM_DIFF Differential 0 normal in-phase audio
1 left channel inverted
1 SM_LAYER12 Allow MPEG layers I & II 0 no
1 yes
2 SM_RESET Soft reset 0 no reset
1 reset
3 SM_CANCEL Cancel decoding current file 0 no
1 yes
4 SM_EARSPEAKER_LO EarSpeaker low setting 0 off
1 active
5 SM_TESTS Allow SDI tests 0 not allowed
1 allowed
6 SM_STREAM Stream mode 0 no
1 yes
7 SM_EARSPEAKER_HI EarSpeaker high setting 0 off
1 active
8 SM_DACT DCLK active edge 0 rising
1 falling
9 SM_SDIORD SDI bit order 0 MSb first
1 MSb last
10 SM_SDISHARE Share SPI chip select 0 no
1 yes
11 SM_SDINEW VS10xx native SPI modes 0 no
1 yes
12 SM_ADPCM PCM/ADPCM recording active 0 no
1 yes
13 - - 0 right
1 wrong
14 SM_LINE1 MIC / LINE1 selector 0 MICP
1 LINE1
15 SM_CLK_RANGE Input clock range 0 12..13 MHz
1 24..26 MHz
When SM_DIFF is set, the player inverts the left channel output. For a stereo input this creates
virtual surround, and for a mono input this creates a differential left/right signal.
SM_LAYER12 enables MPEG 1.0 and 2.0 layer I and II decoding in addition to layer III.
Software reset is initiated by setting SM_RESET to 1. This bit is cleared automatically.
If you want to stop decoding a in the middle, set SM_CANCEL, and continue sending data
honouring DREQ. When SM_CANCEL is detected by a codec, it will stop decoding and return
to the main loop. The stream buffer content is discarded and the SM_CANCEL bit cleared.
SCI_HDAT1 will also be cleared. See Chapter 10.5.2 for details.
Bits SM_EARSPEAKER_LO and SM_EARSPEAKER_HI control the EarSpeaker spatial pro-
cessing. If both are 0, the processing is not active. Other combinations activate the processing
and select 3 different effect levels: LO = 1, HI = 0 selects minimal, LO = 0, HI = 1 selects nor-
mal, and LO = 1, HI = 1 selects extreme. EarSpeaker takes approximately 12 MIPS at 44.1 kHz
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samplerate.
If SM_TESTS is set, SDI tests are allowed. For more details on SDI tests, look at Chapter 10.12.
SM_STREAM activates VS1053b’s stream mode. In this mode, data should be sent with as
even intervals as possible and preferable in blocks of less than 512 bytes, and VS1053b makes
every attempt to keep its input buffer half full by changing its playback speed up to 5%. For best
quality sound, the average speed error should be within 0.5%, the bitrate should not exceed
160 kbit/s and VBR should not be used. For details, see Application Notes for VS10XX. This
mode only works with MP3 and WAV files.
SM_DACT defines the active edge of data clock for SDI. When ’0’, data is read at the rising
edge, when ’1’, data is read at the falling edge.
When SM_SDIORD is clear, bytes on SDI are sent MSb first. By setting SM_SDIORD, the user
may reverse the bit order for SDI, i.e. bit 0 is received first and bit 7 last. Bytes are, however,
still sent in the default order. This register bit has no effect on the SCI bus.
Setting SM_SDISHARE makes SCI and SDI share the same chip select, as explained in Chap-
ter 7.1, if also SM_SDINEW is set.
Setting SM_SDINEW will activate VS10xx native serial modes as described in Chapters 7.1.1 and 7.3.1.
Note, that this bit is set as a default when VS1053b is started up.
By activating SM_ADPCM and SM_RESET at the same time, the user will activate IMA ADPCM
recording mode (see section 10.8).
SM_LINE_IN is used to select the left-channel input for ADPCM recording. If ’0’, differential
microphone input pins MICP and MICN are used; if ’1’, line-level MICP/LINEIN1 pin is used.
SM_CLK_RANGE activates a clock divider in the XTAL input. When SM_CLK_RANGE is set,
the clock is divided by 2 at the input. From the chip’s point of view e.g. 24 MHz becomes
12 MHz. SM_CLK_RANGE should be set as soon as possible after a chip reset.
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9.6.2 SCI_STATUS (RW)
SCI_STATUS contains information on the current status of VS1053b. It also controls some
low-level things that the user does not usually have to care about.
Name Bits Description
SS_DO_NOT_JUMP 15 Header in decode, do not fast forward/rewind
SS_SWING 14:12 Set swing to +0 dB, +0.5 dB, .., or +3.5 dB
SS_VCM_OVERLOAD 11 GBUF overload indicator ’1’ = overload
SS_VCM_DISABLE 10 GBUF overload detection ’1’ = disable
9:8 reserved
SS_VER 7:4 Version
SS_APDOWN2 3 Analog driver powerdown
SS_APDOWN1 2 Analog internal powerdown
SS_AD_CLOCK 1 AD clock select, ’0’ = 6 MHz, ’1’ = 3 MHz
SS_REFERENCE_SEL 0 Reference voltage selection, ’0’ = 1.23 V, ’1’ = 1.65 V
SS_DO_NOT_JUMP is set when a WAV, Ogg Vorbis, WMA, MP4, or AAC-ADIF header is
being decoded and jumping to another location in the file is not allowed. If you use soft reset or
cancel, clear this bit yourself or it can be accidentally left set.
If AVDD is at least 3.3 V, SS_REFERENCE_SEL can be set to select 1.65 V reference voltage
to increase the analog output swing.
SS_AD_CLOCK can be set to divide the AD modulator frequency by 2 if XTALI/2 is too much.
SS_VER is 0 for VS1001, 1 for VS1011, 2 for VS1002, 3 for VS1003, 4 for VS1053 and VS8053,
5 for VS1033, 7 for VS1103, and 6 for VS1063.
SS_APDOWN2 controls analog driver powerdown. SS_APDOWN1 controls internal analog
powerdown. These bit are meant to be used by the system firmware only.
If the user wants to powerdown VS1053b with a minimum power-off transient, set SCI_VOL to
0xffff, then wait for at least a few milliseconds before activating reset.
VS1053b contains GBUF protection circuit which disconnects the GBUF driver when too much
current is drawn, indicating a short-circuit to ground. SS_VCM_OVERLOAD is high while the
overload is detected. SS_VCM_DISABLE can be set to disable the protection feature.
SS_SWING allows you to go above the 0 dB volume setting. Value 0 is normal mode, 1 gives
+0.5 dB, and 2 gives +1.0 dB. Settings from 3 to 7 cause the DAC modulator to be overdriven
and should not be used. You can use SS_SWING with I2S to control the amount of headroom.
Note: Due to a firmware bug in the VS1053b volume calculation routine clears SS_AD_CLOCK
and SS_REFERENCE_SEL bits. Writes to SCI_STATUS or SCI_VOL, and sample rate changes
(if bass enhancer or treble control are active) causes the volume calculation routine to be called.
See the VS1053b Patches w/ FLAC Decoder plugin for a workaround:
http://www.vlsi.fi/en/support/software/vs10xxpatches.html
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9.6.3 SCI_BASS (RW)
Name Bits Description
ST_AMPLITUDE 15:12 Treble Control in 1.5 dB steps (-8..7, 0 = off)
ST_FREQLIMIT 11:8 Lower limit frequency in 1000 Hz steps (1..15)
SB_AMPLITUDE 7:4 Bass Enhancement in 1 dB steps (0..15, 0 = off)
SB_FREQLIMIT 3:0 Lower limit frequency in 10 Hz steps (2..15)
The Bass Enhancer VSBE is a bass boosting DSP algorithm, which tries to take the most out
of the users earphones without causing clipping.
VSBE is activated when SB_AMPLITUDE is non-zero. SB_AMPLITUDE should be set to the
user’s preferences, and SB_FREQLIMIT to roughly 1.5 times the lowest frequency the user’s
audio system can reproduce. For example setting SCI_BASS to 0x00f6 will have 15 dB en-
hancement below 60 Hz. When the bass enhancer is in use with VS1053b, the bass portion is
calculated in mono.
Note: Because VSBE tries to avoid clipping, it gives the best bass boost with dynamical music
material, or when the playback volume is not set to maximum. It also does not create bass: the
source material must have some bass to begin with.
Treble Control VSTC is activated when ST_AMPLITUDE is non-zero. For example setting
SCI_BASS to 0x7a00 will have 10.5 dB treble enhancement at and above 10 kHz.
Bass Enhancer uses about 2.1 MIPS and Treble Control 1.2 MIPS at 44100 Hz samplerate.
Both can be on simultaneously.
In VS1053b bass and treble initialization and volume change is delayed until the next batch of
samples are sent to the audio FIFO. Thus, unlike with earlier VS10XX chips, audio interrupts
can no longer be missed when SCI_BASS or SCI_VOL is written to.
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9.6.4 SCI_CLOCKF (RW)
The external clock multiplier SCI register SCI_CLOCKF, which has changed slightly since
VS1003 and VS1033, is presented in the table below.
SCI_CLOCKF bits
Name Bits Description
SC_MULT 15:13 Clock multiplier
SC_ADD 12:11 Allowed multiplier addition
SC_FREQ 10: 0 Clock frequency
SC_MULT activates the built-in clock multiplier. This will multiply XTALI to create a higher CLKI.
When the multiplier is changed by more than 0.5×, the chip runs at 1.0×clock for a few hundres
clock cycles. The values are as follows:
SC_MULT MASK CLKI
0 0x0000 XTALI
1 0x2000 XTALI×2.0
2 0x4000 XTALI×2.5
3 0x6000 XTALI×3.0
4 0x8000 XTALI×3.5
5 0xa000 XTALI×4.0
6 0xc000 XTALI×4.5
7 0xe000 XTALI×5.0
SC_ADD tells how much the decoder firmware is allowed to add to the multiplier specified by
SC_MULT if more cycles are temporarily needed to decode a WMA or AAC stream. The values
are:
SC_ADD MASK Multiplier addition
0 0x0000 No modification is allowed
1 0x0800 XTALI×1.0
2 0x1000 XTALI×1.5
3 0x1800 XTALI×2.0
If SC_FREQ is non-zero, it tells that the input clock XTALI is running at something else than
12.288 MHz. XTALI is set in 4 kHz steps. The formula for calculating the correct value for this
register is XT ALI−8000000
4000 (XTALI is in Hz).
Note: because maximum samplerate is XT ALI
256 , all samplerates are not available if XTALI <
12.288 MHz.
Note: Automatic clock change can only happen when decoding WMA and AAC files. Automatic
clock change is done one 0.5×at a time. This does not cause a drop to 1.0×clock and you can
use the same SCI and SDI clock throughout the file.
Example: If SCI_CLOCKF is 0x8BE8, SC_MULT = 4, SC_ADD = 1 and SC_FREQ = 0x3E8 = 1000.
This means that XTALI = 1000 ×4000 + 8000000 = 12 MHz. The clock multiplier is set to
3.5×XTALI = 42 MHz, and the maximum allowed multiplier that the firmware may automatically
choose to use is (3.5+1.0)×XTALI = 54 MHz.
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9.6.5 SCI_DECODE_TIME (RW)
When decoding correct data, current decoded time is shown in this register in full seconds.
The user may change the value of this register. In that case the new value should be written
twice to make absolutely certain that the change is not overwritten by the firmware. A write to
SCI_DECODE_TIME also resets the
byteRate
calculation.
SCI_DECODE_TIME is reset at every hardware and software reset. It is no longer cleared
when decoding of a file ends to allow the decode time to proceed automatically with looped
files and with seamless playback of multiple files.
With fast playback (see the
playSpeed
extra parameter) the decode time also counts faster.
Some codecs (WMA and Ogg Vorbis) can also indicate the absolute play position, see the
positionMsec
extra parameter in section 10.11.
9.6.6 SCI_AUDATA (RW)
When decoding correct data, the current samplerate and number of channels can be found
in bits 15:1 and 0 of SCI_AUDATA, respectively. Bits 15:1 contain the samplerate divided by
two, and bit 0 is 0 for mono data and 1 for stereo. Writing to SCI_AUDATA will change the
samplerate directly.
Example: 44100 Hz stereo data reads as 0xAC45 (44101).
Example: 11025 Hz mono data reads as 0x2B10 (11024).
Example: Writing 0xAC80 sets samplerate to 44160 Hz, stereo mode does not change.
To reduce digital power consumption when idle, you can write a low samplerate to SCI_AUDATA.
Note: Ogg Vorbis decoding overrides AUDATA change. If you want to fine-tune samplerate in
streaming applications with Ogg Vorbis, use SCI_CLOCKF to control the playback rate instead
of AUDATA.
9.6.7 SCI_WRAM (RW)
SCI_WRAM is used to upload application programs and data to instruction and data RAMs.
The start address must be initialized by writing to SCI_WRAMADDR prior to the first write/read
of SCI_WRAM. As 16 bits of data can be transferred with one SCI_WRAM write/read, and the
instruction word is 32 bits long, two consecutive writes/reads are needed for each instruction
word. The byte order is big-endian (i.e. most significant words first). After each full-word
write/read, the internal pointer is autoincremented.
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9.6.8 SCI_WRAMADDR (W)
SCI_WRAMADDR is used to set the program address for following SCI_WRAM writes/reads.
Use an address offset from the following table to access X, Y, I or peripheral memory.
WRAMADDR Dest. addr. Bits/ Description
Start. . . End Start. . . End Word
0x1800. . . 0x18XX 0x1800. . . 0x18XX 16 X data RAM
0x5800. . . 0x58XX 0x1800. . . 0x18XX 16 Y data RAM
0x8040. . . 0x84FF 0x0040. . . 0x04FF 32 Instruction RAM
0xC000. . . 0xFFFF 0xC000. . . 0xFFFF 16 I/O
Only user areas in X, Y, and instruction memory are listed above. Other areas can be accessed,
but should not be written to unless otherwise specified.
9.6.9 SCI_HDAT0 and SCI_HDAT1 (R)
For WAV files, SCI_HDAT1 contains 0x7665 (“ve”). SCI_HDAT0 contains the data rate mea-
sured in bytes per second for all supported RIFF WAVE formats: mono and stereo 8-bit or
16-bit PCM, mono and stereo IMA ADPCM. To get the bitrate of the file, multiply the value by 8.
For AAC ADTS streams, SCI_HDAT1 contains 0x4154 (“AT”). For AAC ADIF files, SCI_HDAT1
contains 0x4144 (“AD”). For AAC .mp4 / .m4a files, SCI_HDAT1 contains 0x4D34 (“M4”).
SCI_HDAT0 contains the average data rate in bytes per second. To get the bitrate of the file,
multiply the value by 8.
For WMA files, SCI_HDAT1 contains 0x574D (“WM”) and SCI_HDAT0 contains the data rate
measured in bytes per second. To get the bitrate of the file, multiply the value by 8.
For MIDI files, SCI_HDAT1 contains 0x4D54 (“MT”) and SCI_HDAT0 contains the average data
rate in bytes per second. To get the bitrate of the file, multiply the value by 8.
For Ogg Vorbis files, SCI_HDAT1 contains 0x4F67 “Og”. SCI_HDAT0 contains the average
data rate in bytes per second. To get the bitrate of the file, multiply the value by 8.
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For MP3 files, SCI_HDAT1 is between 0xFFE0 and 0xFFFF. SCI_HDAT1 / 0 contain the follow-
ing:
Bit Function Value Explanation
HDAT1[15:5] syncword 2047 stream valid
HDAT1[4:3] ID 3 ISO 11172-3 MPG 1.0
2 ISO 13818-3 MPG 2.0 (1/2-rate)
1 MPG 2.5 (1/4-rate)
0 MPG 2.5 (1/4-rate)
HDAT1[2:1] layer 3 I
2 II
1 III
0 reserved
HDAT1[0] protect bit 1 No CRC
0 CRC protected
HDAT0[15:12] bitrate see bitrate table
HDAT0[11:10] samplerate 3 reserved
2 32/16/ 8 kHz
1 48/24/12 kHz
0 44/22/11 kHz
HDAT0[9] pad bit 1 additional slot
0 normal frame
HDAT0[8] private bit not defined
HDAT0[7:6] mode 3 mono
2 dual channel
1 joint stereo
0 stereo
HDAT0[5:4] extension see ISO 11172-3
HDAT0[3] copyright 1 copyrighted
0 free
HDAT0[2] original 1 original
0 copy
HDAT0[1:0] emphasis 3 CCITT J.17
2 reserved
1 50/15 microsec
0 none
When read, SCI_HDAT0 and SCI_HDAT1 contain header information that is extracted from
MP3 stream currently being decoded. After reset both registers are cleared, indicating no data
has been found yet.
The “samplerate” field in SCI_HDAT0 is interpreted according to the following table:
“samplerate” ID=3 ID=2 ID=0,1
3 - - -
2 32000 16000 8000
1 48000 24000 12000
0 44100 22050 11025
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9 FUNCTIONAL DESCRIPTION
The “bitrate” field in HDAT0 is read according to the following table. Notice that for variable
bitrate stream the value changes constantly.
Layer I Layer II Layer III
“bitrate” ID=3 ID=0,1,2 ID=3 ID=0,1,2 ID=3 ID=0,1,2
kbit/s kbit/s kbit/s
15 forbidden forbidden forbidden forbidden forbidden forbidden
14 448 256 384 160 320 160
13 416 224 320 144 256 144
12 384 192 256 128 224 128
11 352 176 224 112 192 112
10 320 160 192 96 160 96
9 288 144 160 80 128 80
8 256 128 128 64 112 64
7 224 112 112 56 96 56
6 192 96 96 48 80 48
5 160 80 80 40 64 40
4 128 64 64 32 56 32
3 96 56 56 24 48 24
2 64 48 48 16 40 16
1 32 32 32 8 32 8
0------
The average data rate in bytes per second can be read from memory, see the
byteRate
extra
parameter. This variable contains the byte rate for all codecs. To get the bitrate of the file,
multiply the value by 8.
The bitrate calculation is not automatically reset between songs, but it can also be reset without
a software or hardware reset by writing to SCI_DECODE_TIME.
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9.6.10 SCI_AIADDR (RW)
SCI_AIADDR indicates the start address of the application code written earlier with SCI_WRAMADDR
and SCI_WRAM registers. If no application code is used, this register should not be initialized,
or it should be initialized to zero. For more details, see Application Notes for VS10XX.
Note: Reading AIADDR is not recommended. It can cause samplerate to be set to a very low
value.
9.6.11 SCI_VOL (RW)
SCI_VOL is a volume control for the player hardware. The most significant byte of the volume
register controls the left channel volume, the low part controls the right channel volume. The
channel volume sets the attenuation from the maximum volume level in 0.5 dB steps. Thus,
maximum volume is 0x0000 and total silence is 0xFEFE.
Note, that after hardware reset the volume is set to full volume. Resetting the software does
not reset the volume setting.
Setting SCI_VOL to 0xFFFF will activate analog powerdown mode.
Example: for a volume of -2.0 dB for the left channel and -3.5 dB for the right channel: (2.0/0.5)
= 4, 3.5/0.5 = 7 →SCI_VOL = 0x0407.
Example: SCI_VOL = 0x2424 →both left and right volumes are 0x24 * -0.5 = -18.0 dB.
In VS1053b bass and treble initialization and volume change is delayed until the next batch of
samples are sent to the audio FIFO. Thus, audio interrupts can no longer be missed during a
write to SCI_BASS or SCI_VOL.
This delays the volume setting slightly, but because the volume control is now done in the DAC
hardware instead of performing it to the samples going into the audio FIFO, the overall volume
change response is better than before. Also, the actual volume control has zero-cross detec-
tion, which almost completely removes all audible noise that occurs when volume is suddenly
changed.
9.6.12 SCI_AICTRL[x] (RW)
SCI_AICTRL[x] registers ( x=[0 .. 3] ) can be used to access the user’s application program.
The AICTRL registers are also used with PCM/ADPCM encoding mode.
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10 Operation
10.1 Clocking
VS1053b operates on a single, nominally 12.288 MHz fundamental frequency master clock.
This clock can be generated by external circuitry (connected to pin XTALI) or by the internal
clock crystal interface (pins XTALI and XTALO). This clock is used by the analog parts and
determines the highest available samplerate. With 12.288 MHz clock all samplerates up to
48000 Hz are available.
VS1053b can also use 24..26 MHz clocks when SM_CLK_RANGE in the SCI_MODE register is
set to 1. The system clock is then divided by 2 at the clock input and the chip gets a 12..13 MHz
input clock.
10.2 Hardware Reset
When the XRESET -signal is driven low, VS1053b is reset and all the control registers and
internal states are set to the initial values. XRESET-signal is asynchronous to any external
clock. The reset mode doubles as a full-powerdown mode, where both digital and analog parts
of VS1053b are in minimum power consumption stage, and where clocks are stopped. Also
XTALO is grounded.
When XRESET is asseted, all output pins go to their default states. All input pins will go to
high-impedance state (to input state), except SO, which is still controlled by the XCS.
After a hardware reset (or at power-up) DREQ will stay down for around 22000 clock cycles,
which means an approximate 1.8 ms delay if VS1053b is run at 12.288 MHz. After this the
user should set such basic software registers as SCI_MODE, SCI_BASS, SCI_CLOCKF, and
SCI_VOL before starting decoding. See section 9.6 for details.
If the input clock is 24..26 MHz, SM_CLK_RANGE should be set as soon as possible after a
chip reset without waiting for DREQ.
Internal clock can be multiplied with a PLL. Supported multipliers through the SCI_CLOCKF
register are 1.0×. . . 5.0×the input clock. Reset value for Internal Clock Multiplier is 1.0×. If
typical values are wanted, the Internal Clock Multiplier needs to be set to 3.5×after reset. Wait
until DREQ rises, then write value 0x9800 to SCI_CLOCKF (register 3). See section 9.6.4 for
details.
10.3 Software Reset
In some cases the decoder software has to be reset. This is done by activating bit SM_RESET
in register SCI_MODE (Chapter 9.6.1). Then wait for at least 2 µs, then look at DREQ. DREQ
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will stay down for about 22000 clock cycles, which means an approximate 1.8 ms delay if
VS1053b is run at 12.288 MHz. After DREQ is up, you may continue playback as usual.
As opposed to all earlier VS10XX chips, it is not recommended to do a software reset between
songs. This way the user may be sure that even files with low samplerates or bitrates are played
right to their end.
10.4 Low Power Mode
If you need to keep the system running while not decoding data, but need to lower the power
consumption, you can use the following tricks.
•Select the 1.0×clock by writing 0x0000 to SCI_CLOCKF. This disables the PLL and saves
some power.
•Write a low non-zero value, such as 0x0010 to SCI_AUDATA. This will reduce the sam-
plerate and the number of audio interrupts required. Between audio interrupts the VSDSP
core will just wait for an interrupt, thus saving power.
•Turn off all audio post-processing (tone controls and EarSpeaker).
•If possible for the application, write 0xffff to SCI_VOL to disable the analog drivers.
To return from low-power mode, revert register values in reverse order.
Note: The low power mode consumes significantly more electricity than hardware reset.
10.5 Play and Decode
This is the normal operation mode of VS1053b. SDI data is decoded. Decoded samples are
converted to analog domain by the internal DAC. If no decodable data is found, SCI_HDAT0
and SCI_HDAT1 are set to 0.
When there is no input for decoding, VS1053b goes into idle mode (lower power consumption
than during decoding) and actively monitors the serial data input for valid data.
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10.5.1 Playing a Whole File
This is the default playback mode.
1. Send an audio file to VS1053b.
2. Read extra parameter value endFillByte (Chapter 10.11).
3. Send at least 2052 bytes of endFillByte[7:0].
4. Set SCI_MODE bit SM_CANCEL.
5. Send at least 32 bytes of endFillByte[7:0].
6. Read SCI_MODE. If SM_CANCEL is still set, go to 5. If SM_CANCEL hasn’t cleared
after sending 2048 bytes, do a software reset (this should be extremely rare).
7. The song has now been successfully sent. HDAT0 and HDAT1 should now both contain
0 to indicate that no format is being decoded. Return to 1.
10.5.2 Cancelling Playback
Cancelling playback of a song is a normal operation when the user wants to jump to another
song while doing playback.
1. Send a portion of an audio file to VS1053b.
2. Set SCI_MODE bit SM_CANCEL.
3. Continue sending audio file, but check SM_CANCEL after every 32 bytes of data. If it
is still set, goto 3. If SM_CANCEL doesn’t clear after 2048 bytes or one second, do a
software reset (this should be extremely rare).
4. When SM_CANCEL has cleared, read extra parameter value endFillByte (Chapter 10.11).
5. Send 2052 bytes of endFillByte[7:0].
6. HDAT0 and HDAT1 should now both contain 0 to indicate that no format is being decoded.
You can now send the next audio file.
10.5.3 Fast Play
VS1053b allows fast audio playback. If your microcontroller can feed data fast enough to the
VS1053b, this is the preferred way to fast forward audio.
1. Start sending an audio file to VS1053b.
2. To set fast play, set extra parameter value playSpeed (Chapter 10.11).
3. Continue sending audio file.
4. To exit fast play mode, write 1 to playSpeed.
To estimate whether or not your microcontroller can feed enough data to VS1053b in fast play
mode, see contents of extra parameter value byteRate (Chapter 10.11). Note that byteRate
contains the data speed of the file played back at nominal speed even when fast play is active.
Note: Play speed is not reset when song is changed.
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10.5.4 Fast Forward and Rewind without Audio
To do fast forward and rewind you need the capability to do random access to the audio file.
Unfortunately fast forward and rewind isn’t available at all times, like when file headers are
being read.
1. Send a portion of an audio file to VS1053b.
2. When random access is required, read SCI_STATUS bit SS_DO_NOT_JUMP. If that bit
is set, random access cannot be performed, so go back to 1.
3. Read extra parameter value endFillByte (Chapter 10.11).
4. Send at least 2048 bytes of endFillByte[7:0].
5. Jump forwards or backwards in the file.
6. Continue sending the file.
Note: It is recommended that playback volume is decreased by e.g. 10 dB when fast forward-
ing/rewinding.
Note: Register DECODE_TIME does not take jumps into account.
Note: Midi is not suitable for random-access. You can implement fast forward using the
playSpeed
extra parameter to select 1-128×play speed. SCI_DECODE_TIME also speeds
up. If necessary, rewind can be implemented by restarting decoding of a MIDI file and fast
playing to the appropriate place. SCI_DECODE_TIME can be used to decide when the right
place has been reached.
10.5.5 Maintaining Correct Decode Time
When fast forward and rewind operations are performed, there is no way to maintain correct
decode time for most files. However, WMA and Ogg Vorbis files offer exact time information in
the file. To use accurate time information whenever possible, use the following algorithm:
1. Start sending an audio file to VS1053b.
2. Read extra parameter value pair positionMsec (Chapter 10.11).
3. If positionMsec is -1, show you estimation of decoding time using DECODE_TIME (and
your estimate of file position if you have performed fast forward / rewind operations).
4. If positionMsec is not -1, use this time to show the exact position in the file.
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10.6 Feeding PCM Data
VS1053b can be used as a PCM decoder by sending a WAV file header. If the length sent in the
WAV header is 0xFFFFFFFF, VS1053b will stay in PCM mode indefinitely (or until SM_CANCEL
has been set). 8-bit (unsigned) linear and 16-bit (signed, 2’s complement) linear audio is sup-
ported in mono or stereo. A WAV header looks like this:
File Offset Field Name Size Bytes Description
0 ChunkID 4
"RIFF"
4 ChunkSize 4 0xff 0xff 0xff 0xff
8 Format 4
"WAVE"
12 SubChunk1ID 4
"fmt "
16 SubChunk1Size 4 0x10 0x0 0x0 0x0 16
20 AudioFormat 2 0x1 0x0 Linear PCM
22 NumOfChannels 2 C0 C1 1 for mono, 2 for stereo
24 SampleRate 4 S0 S1 S2 S3 0x1f40 for 8 kHz
28 ByteRate 4 R0 R1 R2 R3 0x3e80 for 8 kHz 16-bit mono
32 BlockAlign 2 A0 A1 0x02 0x00 for mono, 0x04 0x00 for stereo 16-bit
34 BitsPerSample 2 B0 B1 0x10 0x00 for 16-bit data
52 SubChunk2ID 4
"data"
56 SubChunk2Size 4 0xff 0xff 0xff 0xff Data size
The rules to calculate the four variables are as follows:
•S=sample rate in Hz, e.g. 44100 for 44.1 kHz.
•For 8-bit data B= 8, and for 16-bit data B= 16.
•For mono data C= 1, for stereo data C= 2.
•A=C×B
8.
•R=S×A.
Example: A 44100 Hz 16-bit stereo PCM header would read as follows:
0000 52 49 46 46 ff ff ff ff 57 41 56 45 66 6d 74 20 |RIFF....WAVEfmt |
0100 10 00 00 00 01 00 02 00 44 ac 00 00 10 b1 02 00 |........D.......|
0200 04 00 10 00 64 61 74 61 ff ff ff ff |....data....|
10.7 Ogg Vorbis Recording
Ogg Vorbis is an open file format that allows for very high sound quality with low to medium
bitrates.
Ogg Vorbis recording is activated by loading the Ogg Vorbis Encoder Application to the 16
KiB program RAM memory of the VS1053b. After activation, encoder results can be read
from registers SCI_HDAT0 and SCI_HDAT1, much like when using PCM/ADPCM recording
(Chapter 10.8).
Three profiles are provided: one for high-quality stereo recording at a bitrate of approx. 140
kbit/s, and two for speech-quality mono recording at a bitrates between 15 and 30 kbit/s.
To use the Ogg Vorbis Encoder application, please load the application from VLSI Solution’s
Web page http://www.vlsi.fi/en/support/software/vs10xxapplications.html and read the accom-
panying documentation.
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10.8 PCM / ADPCM Recording
This chapter explains how to record a RIFF/WAV file in PCM or IMA ADPCM format. IME
ADPCM is a widely supported ADPCM format and many PC audio playback programs can play
it. IMA ADPCM recording gives a compression ratio of almost 4:1 compared to linear, 16-bit
audio. This makes it possible to record for example ono 8 kHz audio at 32.44 kbit/s.
VS1053 has a stereo ADC, thus also two-channel (separate AGC, if AGC enabled) and stereo
(common AGC, if AGC enabled) modes are available. Mono recording mode selects either the
left or right channel. Left channel is either MIC or LINE1 depending on the SCI_MODE register.
If absolute best quality stereo PCM recording at 48 kHz is required, download and use the
VS1053 WAV PCM Recorder Application plugin, available for download at
http://www.vlsi.fi/en/support/software/vs10xxapplications.html . If you use it, follow the instruc-
tions of the plugin documentation instead of this datasheet.
10.8.1 Activating PCM / ADPCM Recording Mode
Register Bits Description
SCI_MODE 2, 12, 14 Start ADPCM mode, select MIC/LINE1
SCI_AICTRL0 15..0 Sample rate 8000..48000 Hz (read at recording startup)
SCI_AICTRL1 15..0 Recording gain (1024 = 1×) or 0 for automatic gain control
SCI_AICTRL2 15..0 Maximum autogain amplification (1024 = 1×, 65535 = 64×)
SCI_AICTRL3 1..0 0 = joint stereo (common AGC),
1 = dual channel (separate AGC),
2 = left channel,
3 = right channel
2 0 = IMA ADPCM mode, 1 = LINEAR PCM mode
15..3 reserved, set to 0
PCM / IMA ADPCM recording mode is activated by setting bit SM_ADPCM in SCI_MODE
and loading and starting a patch code. Line input 1 is used instead of differential mic in-
put if SM_LINE1 is set. Before activating ADPCM recording, user must write the right val-
ues to SCI_AICTRL0 and SCI_AICTRL3. These values are only read at recording startup.
SCI_AICTRL1 and SCI_AICTRL2 can be altered anytime, but it is preferable to write good init
values before activation.
SCI_AICTRL1 controls linear recording gain. 1024 is equal to digital gain 1, 512 is equal to
digital gain 0.5 and so on. If the user wants to use automatic gain control (AGC), SCI_AICTRL1
should be set to 0. Typical speech applications usually are better off using AGC, as this takes
care of relatively uniform speech loudness in recordings.
SCI_AICTRL2 controls the maximum AGC gain. This can be used to limit the amplification of
noise when there is no signal. If SCI_AICTRL2 is zero, the maximum gain is initialized to 65535
(64×), i.e. whole range is used.
For example:
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WriteVS10xxRegister(SCI_AICTRL0, 16000U);
WriteVS10xxRegister(SCI_AICTRL1, 0);
WriteVS10xxRegister(SCI_AICTRL2, 4096U);
WriteVS10xxRegister(SCI_AICTRL3, 0);
WriteVS10xxRegister(SCI_MODE, ReadVS10xxRegister(SCI_MODE) |
SM_ADPCM | SM_LINE1);
#ifdef I_HAVE_THE_VS1053B_PATCHES_PACKAGE
/* Strongly recommended to use the VS1053b Patches package. Get it at
http://www.vlsi.fi/en/support/software/vs10xxpatches.html */
LoadVS1053PatchesPackage(); /* Loads patches and starts recording */
#else
/* If not using the VS1053b Patches package, you need to do these steps */
WriteVS10xxPatch();
#endif
selects 16 kHz, stereo mode with automatic gain control and maximum amplification of 4×.
It is strongly recommended to always run the VS1053b Patches package, available at
http://www.vlsi.fi/en/support/software/vs10xxpatches.html , when encoding. If you are not run-
ning the VS1053b Patches package, you need to implement WriteVS10xxPatch() for the exam-
ple above. The function must perform the following SCI writes:
Register Reg. No Values
SCI_WRAMADDR 0x7 0x8050
SCI_WRAM 0x6
0x0000,0x1790,0xf400,0x5400,0x0000,0x0a10,0xf400,0x5600,
0xb080,0x0024,0x0007,0x9257,0x3f00,0x0024,0x0030,0x0297,
0x3f00,0x0024,0x0000,0x004d,0x0014,0x958f,0x0000,0x1b4e,
0x280f,0xe100,0x0006,0x2016,0x2a00,0x17ce,0x3e12,0xb817,
0x3e14,0xf812,0x3e01,0xb811,0x0007,0x9717,0x0020,0xffd2,
0x0030,0x11d1,0x3111,0x8024,0x3704,0xc024,0x3b81,0x8024,
0x3101,0x8024,0x3b81,0x8024,0x3f04,0xc024,0x2808,0x4800,
0x36f1,0x9811,0x2814,0x9c91,0x0000,0x004d,0x2814,0x9940,
0x003f,0x0013,
SCI_AIADDR 0xa 0x0050
This small and incomplete patch is also available from VLSI Solution’s web page
http://www.vlsi.fi/en/support/software/vs10xxpatches.html by the name of VS1053b IMA AD-
PCM Encoder Fix.
10.8.2 Reading PCM / IMA ADPCM Data
After PCM / IMA ADPCM recording has been activated, registers SCI_HDAT0 and SCI_HDAT1
have new functions.
The PCM / IMA ADPCM sample buffer is 1024 16-bit words. The fill status of the buffer can
be read from SCI_HDAT1. If SCI_HDAT1 is greater than 0, you can read as many 16-bit words
from SCI_HDAT0. If the data is not read fast enough, the buffer overflows and returns to empty
state.
Note: if SCI_HDAT1 ≥768, it may be better to wait for the buffer to overflow and clear before
reading samples. That way you may avoid buffer aliasing.
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In IMA ADPCM mode each mono IMA ADPCM block is 128 words, i.e. 256 bytes, and stereo
IMA ADPCM block is 256 words, i.e. 512 bytes. If you wish to interrupt reading data and
possibly continue later, please stop at the boundary. This way complete compression blocks
are skipped and the encoded stream stays valid.
10.8.3 Adding a PCM RIFF Header
To make your PCM file a RIFF / WAV file, you have to add a header to the data. The following
shows a header for a mono file. Note that 2- and 4-byte values are little-endian (lowest byte
first).
File Offset Field Name Size Bytes Description
0 ChunkID 4
"RIFF"
4 ChunkSize 4 F0 F1 F2 F3 File size - 8
8 Format 4
"WAVE"
12 SubChunk1ID 4
"fmt "
16 SubChunk1Size 4 0x10 0x0 0x0 0x0 20
20 AudioFormat 2 0x01 0x0 0x1 for PCM
22 NumOfChannels 2 C0 C1 1 for mono, 2 for stereo
24 SampleRate 4 R0 R1 R2 R3 0x1f40 for 8 kHz
28 ByteRate 4 B0 B1 B2 B3 0x3e80 for 8 kHz mono
32 BlockAlign 2 0x02 0x00 2 for mono, 4 for stereo
34 BitsPerSample 2 0x10 0x00 16 bits / sample
36 SubChunk3ID 4
"data"
40 SubChunk3Size 4 D0 D1 D2 D3 Data size (File Size-36)
44 Samples... Audio samples
The values in the table are calculated as follows:
R=Fs(see Chapter 10.8.1 to see how to calculate Fs)
B= 2 ×Fs×C
If you know beforehand how much you are going to record, you may fill in the complete header
before any actual data. However, if you don’t know how much you are going to record, you have
to fill in the header size datas Fand Dafter finishing recording.
The PCM data is read from SCI_HDAT0 and written into file as follows. The low 8 bits of
SCI_HDAT0 should be written as the first byte to a file, then the high 8 bits (little-endian order).
Note that this is different from how ADPCM data is written to a file (see Chapter 10.8.4).
Below is an example of a valid header for a 44.1 kHz mono PCM file that has a final length of
1798768 (0x1B7270) bytes:
0000 52 49 46 46 68 72 1b 00 57 41 56 45 66 6d 74 20 |RIFFhr..WAVEfmt |
0010 10 00 00 00 01 00 01 00 80 bb 00 00 00 77 01 00 |.............w..|
0020 02 00 10 00 64 61 74 61 44 72 1b 00 |....dataDr......|
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10.8.4 Adding an IMA ADPCM RIFF Header
To make your IMA ADPCM file a RIFF / WAV file, you have to add a header to the data. The
following shows a header for a mono file. Note that 2- and 4-byte values are little-endian (lowest
byte first).
File Offset Field Name Size Bytes Description
0 ChunkID 4
"RIFF"
4 ChunkSize 4 F0 F1 F2 F3 File size - 8
8 Format 4
"WAVE"
12 SubChunk1ID 4
"fmt "
16 SubChunk1Size 4 0x14 0x0 0x0 0x0 20
20 AudioFormat 2 0x11 0x0 0x11 for IMA ADPCM
22 NumOfChannels 2 C0 C1 1 for mono, 2 for stereo
24 SampleRate 4 R0 R1 R2 R3 0x1f40 for 8 kHz
28 ByteRate 4 B0 B1 B2 B3 0xfd7 for 8 kHz mono
32 BlockAlign 2 0x00 0x01 256 for mono, 512 for stereo
34 BitsPerSample 2 0x04 0x00 4-bit ADPCM
36 ByteExtraData 2 0x02 0x00 2
38 ExtraData 2 0xf9 0x01 Samples per block (505)
40 SubChunk2ID 4
"fact"
44 SubChunk2Size 4 0x4 0x0 0x0 0x0 4
48 NumOfSamples 4 S0 S1 S2 S3
52 SubChunk3ID 4
"data"
56 SubChunk3Size 4 D0 D1 D2 D3 Data size (File Size-60)
60 Block1 256 First ADPCM block, 512 bytes for stereo
316 . . . More ADPCM data blocks
If we have naudio blocks, the values in the table are as follows:
F=n×C×256 + 52
R=Fs(see Chapter 10.8.1 to see how to calculate Fs)
B=Fs×C×256
505
S=n×505.D=n×C×256
If you know beforehand how much you are going to record, you may fill in the complete header
before any actual data. However, if you don’t know how much you are going to record, you have
to fill in the header size datas F,Sand Dafter finishing recording.
The 128 words (256 words for stereo) of an ADPCM block are read from SCI_HDAT0 and
written into file as follows. The high 8 bits of SCI_HDAT0 should be written as the first byte to a
file, then the low 8 bits (big-endian order). Note that this is contrary to the native byte order of
some 16-bit microcontrollers, and you may have to take extra care to do this right. Note also,
that this is different from how PCM data is written to a file (see Chapter 10.8.3).
To see if you have written the mono file in the right way check bytes 2 and 3 (the first byte
counts as byte 0) of each 256-byte block. Byte 2 should be in the range 0..88 and byte 3 should
be zero. For stereo you check bytes 2, 3, 6, and 7 of each 512-byte block. Bytes 2 and 6 should
be in the range 0..88. Bytes 3 and 7 should be zero.
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Below is an example of a valid header for a 44.1 kHz stereo IMA ADPCM file that has a final
length of 10038844 (0x992E3C) bytes:
0000 52 49 46 46 34 2e 99 00 57 41 56 45 66 6d 74 20 |RIFF4...WAVEfmt |
0010 14 00 00 00 11 00 02 00 44 ac 00 00 a7 ae 00 00 |........D.......|
0020 00 02 04 00 02 00 f9 01 66 61 63 74 04 00 00 00 |........fact....|
0030 14 15 97 00 64 61 74 61 00 2e 99 00 |....data....|
10.8.5 Playing ADPCM Data
In order to play back your PCM / IMA ADPCM recordings, you have to have a file with a header
as described in Chapter 10.8.3 or Chapter 10.8.4. If this is the case, all you need to do is to
provide the ADPCM file through SDI as you would with any audio file.
10.8.6 Sample Rate Considerations
VS10xx chips that support IMA ADPCM playback are capable of playing back ADPCM files with
any sample rate. However, some other programs may expect IMA ADPCM files to have some
exact sample rates, like 8000 or 11025 Hz. Also, some programs or systems do not support
sample rates below 8000 Hz.
If you want better quality with the expense of increased data rate, you can use higher sample
rates, for example 16 kHz.
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10.8.7 Record Monitoring Volume
In VS1053b writing to the SCI_VOL register during IMA ADPCM encoding does not change the
volume. You need to set a suitable volume before activating the IMA ADPCM mode, or you can
use the VS1053 hardware volume control register DAC_VOL directly.
For example:
WriteVS10xxRegister(SCI_WRAMADDR, 0xc045); /*DAC_VOL*/
WriteVS10xxRegister(SCI_WRAM, 0x0101); /*-6.0 dB*/
The hardware volume control DAC_VOL (address 0xc045) allows 0.5 dB steps for both left (high
8 bits) and right channel (low 8 bits). The low 4 bits of both 8-bit values set the attenuation in
6 dB steps (range 0..15), the high 4 bits in 0.5 dB steps (range 0..11). For examples, see table
below.
dB DAC_VOL
-0.0 0x0000
-0.5 0xb1b1
-1.0 0xa1a1
-1.5 0x9191
-2.0 0x8181
-2.5 0x7171
-3.0 0x6161
-3.5 0x5151
-4.0 0x4141
-4.5 0x3131
-5.0 0x2121
-5.5 0x1111
-6.0 0x0101
dB DAC_VOL
-6.5 0xb2b2
: :
-12.0 0x0202
-12.5 0xb3b3
: :
-18.0 0x0303
-18.5 0xb4b4
: :
-24.0 0x0404
-24.5 0xb5b5
: :
-30.0 0x0505
-30.5 0xb6b6
dB DAC_VOL
: :
-36.0 0x0606
-36.5 0xb7b7
: :
-42.0 0x0707
-42.5 0xb8b8
: :
-48.0 0x0808
-48.5 0xb9b9
: :
-54.0 0x0909
-54.5 0xbaba
: :
dB DAC_VOL
-60.0 0x0a0a
-60.5 0xbbbb
: :
-66.0 0x0b0b
-66.5 0xbcbc
: :
-72.0 0x0c0c
-72.5 0xbdbd
: :
-78.0 0x0d0d
-78.5 0xbebe
: :
-84.0 0x0e0e
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10.9 SPI Boot
If GPIO0 is set with a pull-up resistor to 1 at boot time, VS1053b tries to boot from external SPI
memory.
SPI boot redefines the following pins:
Normal Mode SPI Boot Mode
GPIO0 xCS
GPIO1 CLK
DREQ MOSI
GPIO2 MISO
The memory has to be an SPI Bus Serial EEPROM with 16-bit or 24-bit addresses. The serial
speed used by VS1053b is 245 kHz with the nominal 12.288 MHz clock. The first three bytes
in the memory have to be 0x50, 0x26, 0x48.
10.10 Real-Time MIDI
If GPIO0 is low and GPIO1 is high during boot, real-time MIDI mode is activated. In this mode
the PLL is configured to 4.0×, the UART is configured to the MIDI data rate 31250 bps, and
real-time MIDI data is then read from UART and SDI. Both input methods should not be used
simultaneously. If you use SDI, first send 0x00 and then send the MIDI data byte.
EarSpeaker setting can be configured with GPIO2 and GPIO3. The state of GPIO2 and GPIO3
are only read at startup.
Real-Time MIDI can also be started with a small patch code using SCI.
Note: The real-time MIDI parser in VS1053b does not know how to skip SysEx messages. An
improved version can be loaded into IRAM if needed.
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10.11 Extra Parameters
The following structure is in X memory at address 0x1e02 (note the different location than in
VS1033) and can be used to change some extra parameters or get useful information.
#define PARAMETRIC_VERSION 0x0003
struct parametric {
/* configs are not cleared between files */
u_int16 version; /*1e02 - structure version */
u_int16 config1; /*1e03 ---- ---- ppss RRRR PS mode, SBR mode, Reverb */
u_int16 playSpeed; /*1e04 0,1 = normal speed, 2 = twice, 3 = three times etc. */
u_int16 byteRate; /*1e05 average byterate */
u_int16 endFillByte; /*1e06 byte value to send after file sent */
u_int16 reserved[16]; /*1e07..15 file byte offsets */
u_int32 jumpPoints[8]; /*1e16..25 file byte offsets */
u_int16 latestJump; /*1e26 index to lastly updated jumpPoint */
u_int32 positionMsec /*1e27-28 play position, if known (WMA, Ogg Vorbis) */
s_int16 resync; /*1e29 > 0 for automatic m4a, ADIF, WMA resyncs */
union {
struct {
u_int32 curPacketSize;
u_int32 packetSize;
} wma;
struct {
u_int16 sceFoundMask; /*1e2a SCE's found since last clear */
u_int16 cpeFoundMask; /*1e2b CPE's found since last clear */
u_int16 lfeFoundMask; /*1e2c LFE's found since last clear */
u_int16 playSelect; /*1e2d 0 = first any, initialized at aac init */
s_int16 dynCompress; /*1e2e -8192=1.0, initialized at aac init */
s_int16 dynBoost; /*1e2f 8192=1.0, initialized at aac init */
u_int16 sbrAndPsStatus; /*0x1e30 1=SBR, 2=upsample, 4=PS, 8=PS active */
} aac;
struct {
u_int32 bytesLeft;
} midi;
struct {
s_int16 gain; /* 0x1e2a proposed gain offset in 0.5dB steps, default = -12 */
} vorbis;
} i;
};
Notice that reading two-word variables through the SCI_WRAMADDR and SCI_WRAM inter-
face is not protected in any way. The variable can be updated between the read of the low and
high parts. The problem arises when both the low and high parts change values. To determine
if the value is correct, you should read the value twice and compare the results.
The following example shows what happens when
bytesLeft
is decreased from 0x10000 to
0xffff and the update happens between low and high part reads or after high part read.
Read Invalid
Address Value
0x1e2a 0x0000 change after this
0x1e2b 0x0000
0x1e2a 0xffff
0x1e2b 0x0000
Read Valid
Address Value
0x1e2a 0x0000
0x1e2b 0x0001 change after this
0x1e2a 0xffff
0x1e2b 0x0000
No Update
Address Value
0x1e2a 0x0000
0x1e2b 0x0001
0x1e2a 0x0000
0x1e2b 0x0001
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You can see that in the invalid read the low part wraps from 0x0000 to 0xffff while the high part
stays the same. In this case the second read gives a valid answer, otherwise always use the
value of the first read. The second read is needed when it is possible that the low part wraps
around, changing the high part, i.e. when the low part is small.
bytesLeft
is only decreased
by one at a time, so a reread is needed only if the low part is 0.
10.11.1 Common Parameters
These parameters are common for all codecs. Other fields are only valid when the correspond-
ing codec is active. The currently active codec can be determined from SCI_HDAT1.
Parameter Address Usage
version 0x1e02 Structure version – 0x0003
config1 0x1e03 Miscellaneous configuration
playSpeed 0x1e04 0,1 = normal speed, 2 = twice, 3 = three times etc.
byteRate 0x1e05 average byterate
endFillByte 0x1e06 byte to send after file
jumpPoints[8] 0x1e16-25 Packet offsets for WMA and AAC
latestJump 0x1e26 Index to latest jumpPoint
positionMsec 0x1e27-28 File position in milliseconds, if available
resync 0x1e29 Automatic resync selector
The fuse-programmed ID is read at startup and copied into the
chipID
field. If not available,
the value will be all zeros. The
version
field can be used to determine the layout of the rest
of the structure. The version number is changed when the structure is changed. For VS1053b
the structure version is 3.
config1
controls MIDI Reverb and AAC’s SBR and PS settings.
playSpeed
makes it possible to fast forward songs. Decoding of the bitstream is performed,
but only each
playSpeed
frames are played. For example by writing 4 to
playSpeed
will play
the song four times as fast as normal, if you are able to feed the data with that speed. Write 0
or 1 to return to normal speed. SCI_DECODE_TIME will also count faster. All current codecs
support the
playSpeed
configuration.
byteRate
contains the average bitrate in bytes per second for every code. The value is updated
once per second and it can be used to calculate an estimate of the remaining playtime. This
value is also available in SCI_HDAT0 for all codecs except MP3, MP2, and MP1.
endFillByte
indicates what byte value to send after file is sent before SM_CANCEL.
jumpPoints
contain 32-bit file offsets. Each valid (non-zero) entry indicates a start of a packet
for WMA or start of a raw data block for AAC (ADIF, .mp4 / .m4a).
latestJump
contains the
index of the entry that was updated last. If you only read entry pointed to by
latestJump
you
do not need to read the entry twice to ensure validity. Jump point information can be used to
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implement perfect fast forward and rewind for WMA and AAC (ADIF, .mp4 / .m4a).
positionMsec
is a field that gives the current play position in a file in milliseconds, regardless
of rewind and fast forward operations. The value is only available in codecs that can determine
the play position from the stream itself. Currently WMA and Ogg Vorbis provide this information.
If the position is unknown, this field contains -1.
resync
field is used to force a resynchronization to the stream for WMA and AAC (ADIF, .mp4
/ .m4a) instead of ending the decode at first error. This field can be used to implement almost
perfect fast forward and rewind for WMA and AAC (ADIF, .mp4 / .m4a). The user should set this
field before performing data seeks if they are not in packet or data block boundaries. The field
value tells how many tries are allowed before giving up. The value 32767 gives infinite tries.
The
resync
field is set to 32767 after a reset to make resynchronization the default action, but
it can be cleared after reset to restore the old action. When
resync
is set, every file decode
should always end as described in Chapter 10.5.1.
Seek fields no longer exist. When resync is required, WMA and AAC codecs now enter broad-
cast/stream mode where file size information is ignored. Also, the file size and sample size in-
formation of WAV files are ignored when
resync
is non-zero. The user must use SM_CANCEL
or software reset to end decoding.
Note: WAV, WMA, ADIF, and .mp4 / .m4a files begin with a metadata or header section, which
must be fully processed before any fast forward or rewind operation. SS_DO_NOT_JUMP
(in SCI_STATUS) is clear when the header information has been processed and jumps are
allowed.
10.11.2 WMA
Parameter Address Usage
curPacketSize 0x1e2a/2b The size of the packet being processed
packetSize 0x1e2c/2d The packet size in ASF header
The ASF header packet size is available in
packetSize
. With this information and a packet
start offset from
jumpPoints
you can parse the packet headers and skip packets in ASF files.
WMA decoder can also increase the internal clock automatically when it detects that a file can
not be decoded correctly with the current clock. The maximum allowed clock is configured with
the SCI_CLOCKF register.
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10.11.3 AAC
Parameter Address Usage
config1 0x1e03(7:4) SBR and PS select
sceFoundMask 0x1e2a Single channel elements found
cpeFoundMask 0x1e2b Channel pair elements found
lfeFoundMask 0x1e2c Low frequency elements found
playSelect 0x1e2d Play element selection
dynCompress 0x1e2e Compress coefficient for DRC, -8192=1.0
dynBoost 0x1e2f Boost coefficient for DRC, 8192=1.0
sbrAndPsStatus 0x1e30 SBR and PS available flags
playSelect
determines which element to decode if a stream has multiple elements. The value
is set to 0 each time AAC decoding starts, which causes the first element that appears in the
stream to be selected for decoding. Other values are: 0x01 - select first single channel element
(SCE), 0x02 - select first channel pair element (CPE), 0x03 - select first low frequency element
(LFE), S∗16 + 5 - select SCE number S, P∗16 + 6 - select CPE number P, L∗16 + 7 -
select LFE number L. When automatic selection has been performed,
playSelect
reflects the
selected element.
sceFoundMask
,
cpeFoundMask
, and
lfeFoundMask
indicate which elements have been found in
an AAC stream since the variables have last been cleared. The values can be used to present
an element selection menu with only the available elements.
dynCompress
and
dynBoost
change the behavior of the dynamic range control (DRC) that is
present in some AAC streams. These are also initialized when AAC decoding starts.
sbrAndPsStatus
indicates spectral band replication (SBR) and parametric stereo (PS) status.
Bit Usage
0 SBR present
1 upsampling active
2 PS present
3 PS active
Bits 7 to 4 in
config1
can be used to control the SBR and PS decoding. Bits 5 and 4 select
SBR mode and bits 7 and 6 select PS mode. These configuration bits are useful if your AAC
license does not cover SBR and/or PS.
config1(5:4) Usage
’00’ normal mode, upsample <24 kHz AAC files
’01’ do not automatically upsample <24 kHz AAC files, but
enable upsampling if SBR is encountered
’10’ never upsample
’11’ disable SBR (also disables PS)
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config1(7:6) Usage
’00’ normal mode, process PS if it is available
’01’ process PS if it is available, but in downsampled mode
’10’ reserved
’11’ disable PS processing
AAC decoder can also increase the internal clock automatically when it detects that a file can
not be decoded correctly with the current clock. The maximum allowed clock is configured with
the SCI_CLOCKF register.
If even the highest allowed clock is too slow to decode an AAC file with SBR and PS compo-
nents, the advanced decoding features are automatically dropped one by one until the file can
be played. First the parametric stereo processing is dropped (the playback becomes mono).
If that is not enough, the spectral band replication is turned into downsampled mode (reduced
bandwidth). As the last resort the spectral band replication is fully disabled. Dropped features
are restored at each song change.
10.11.4 Midi
Parameter Address Usage
config1 0x1e03 Miscellaneous configuration
bits [3:0] Reverb: 0 = auto (ON if clock >= 3.0×)
1 = off, 2 - 15 = room size
bytesLeft 0x1e2a/2b The number of bytes left in this track
The lowest 4 bits of
config1
controls the reverb effect.
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10.11.5 Ogg Vorbis
Parameter Address Usage
gain 0x1e2a Preferred Replay Gain offset
Ogg Vorbis decoding supports Replay Gain technology. The Replay Gain technology is used
to automatically give all songs a matching volume so that the user does not need to adjust the
volume setting between songs.
If the Ogg Vorbis decoder finds a REPLAYGAIN_ALBUM_GAIN tag in the song header, the tag
is parsed and the decoded gain setting is written to the
gain
parameter.
If REPLAYGAIN_ALBUM_GAIN is not available, REPLAYGAIN_TRACK_GAIN is used.
If even REPLAYGAIN_TRACK_GAIN is not available, a default of -6 dB (
gain
value -12) is set.
For more information about Replay Gain, see http://en.wikipedia.org/wiki/Replay_Gain and
http://www.replaygain.org/ .
The player software can use the gain value to adjust the volume level. Negative values mean
that the volume should be decreased, positive values mean that the volume should be in-
creased.
For example
gain
= -11 means that volume should be decreased by 5.5 dB (−11/2 = −5.5),
and left and right attenuation should be increased by 11. When
gain
= 2 volume should be
increased by 1 dB (2/2=1.0), and left and right attenuation should be decreased by 2. Because
volume setting can not go above +0 dB, the value should be saturated.
Gain Volume SCI_VOL (Volume-Gain)
-11 (-5.5 dB) 0 (+0.0 dB) 0x0b0b (-5.5 dB)
-11 (-5.5 dB) 3 (-1.5 dB) 0x0e0e (-7.0 dB)
+2 (+1.0 dB) 0 (+0.0 dB) 0x0000 (+0.0 dB)
+2 (+1.0 dB) 1 (-0.5 dB) 0x0000 (+0.0 dB)
+2 (+1.0 dB) 4 (-2.0 dB) 0x0202 (-1.0 dB)
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10.12 SDI Tests
There are several test modes in VS1053b, which allow the user to perform memory tests, SCI
bus tests, and several different sine wave tests.
All tests, except for the New Sine and Sweep Tests, are started in a similar way: do a hardware
reset to VS1053b, then set register SM_MODE bit SM_TESTS, and then send a test command
sequence to the SDI bus. Each test is started by sending a 4-byte special command sequence,
followed by 4 zeros. The sequences are described below.
10.12.1 Old Sine Test
Sine test is initialized with the 8-byte sequence 0x53 0xEF 0x6E n0 0 0 0, where ndefines the
sine test to use. nis defined as follows:
nbits
Name Bits Description
FsIdx 7:5 Samplerate index
S4:0 Sine skip speed
FsIdx F sFsIdx F s
0 44100 Hz 4 24000 Hz
1 48000 Hz 5 16000 Hz
2 32000 Hz 6 11025 Hz
3 22050 Hz 7 12000 Hz
The frequency of the sine to be output can now be calculated from F=Fs×S
128 .
Example: Sine test is activated with value 126, which is 0b01111110. Breaking nto its compo-
nents, FsIdx = 0b011 = 3 and thus Fs= 22050Hz.S= 0b11110 = 30, and thus the final sine
frequency F= 22050Hz ×30
128 ≈5168Hz.
To exit the sine test, send the sequence 0x45 0x78 0x69 0x74 0 0 0 0.
Note: Sine test signals go through the digital volume control, so it is possible to test channels
separately.
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10.12.2 New Sine and Sweep Tests
A more frequency-accurate sine test can be started and controlled from SCI. SCI_AICTRL0 and
SCI_AICTRL1 set the sine frequencies for left and right channel, respectively. These registers,
volume (SCI_VOL), and samplerate (SCI_AUDATA) can be set before or during the test. Write
0x4020 to SCI_AIADDR to start the test.
SCI_AICTRLn can be calculated from the desired frequency and DAC samplerate by:
SCI_AICT RLn =Fsin ×65536/Fs
The maximum value for SCI_AICTRLn is 0x8000U. For the best S/N ratio for the generated
sine, three LSb’s of the SCI_AICTRLn should be zero. The resulting frequencies Fsin can be
calculated from the DAC samplerate Fsand SCI_AICTRL0 / SCI_AICTRL1 using the following
equation.
Fsin =SCI_AICT RLn ×Fs/65536
Sine sweep test can be started by writing 0x4022 to SCI_AIADDR.
Both these tests use the normal audio path, thus also SCI_BASS, differential output mode, and
EarSpeaker settings have an effect.
10.12.3 Pin Test
Pin test is activated with the 8-byte sequence 0x50 0xED 0x6E 0x54 0 0 0 0. This test is meant
for chip production testing only.
10.12.4 SCI Test
Sci test is initialized with the 8-byte sequence 0x53 0x70 0xEE n0 0 0 0, where nis the
register number to test. The content of the given register is read and copied to SCI_HDAT0. If
the register to be tested is HDAT0, the result is copied to SCI_HDAT1.
Example: if nis 0, contents of SCI register 0 (SCI_MODE) is copied to SCI_HDAT0.
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10.12.5 Memory Test
Memory test mode is initialized with the 8-byte sequence 0x4D 0xEA 0x6D 0x54 0 0 0 0. After
this sequence, wait for 1100000 clock cycles. The result can be read from the SCI register
SCI_HDAT0, and ’one’ bits are interpreted as follows:
Bit(s) Mask Meaning
15 0x8000 Test finished
14:10 Unused
9 0x0200 Mux test succeeded
8 0x0100 Good MAC RAM
7 0x0080 Good I RAM
6 0x0040 Good Y RAM
5 0x0020 Good X RAM
4 0x0010 Good I ROM 1
3 0x0008 Good I ROM 2
2 0x0004 Good Y ROM
1 0x0002 Good X ROM 1
0 0x0001 Good X ROM 2
0x83ff All ok
Memory tests overwrite the current contents of the RAM memories.
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11 VS1053B REGISTERS
11 VS1053b Registers
11.1 Who Needs to Read This Chapter
User software is required when a user wishes to add some own functionality like DSP effects
to VS1053b.
However, most users of VS1053b don’t need to worry about writing their own code, or about
this chapter, including those who only download software plug-ins from VLSI Solution’s Web
site.
Note: Also see VS1063 Hardware Guide for more information, because the hardware is com-
patible with VS1053.
11.2 The Processor Core
VS_DSP is a 16/32-bit DSP processor core that also had extensive all-purpose processor fea-
tures. VLSI Solution’s free VSKIT Software Package contains all the tools and documentation
needed to write, simulate and debug Assembly Language or Extended ANSI C programs for the
VS_DSP processor core. VLSI Solution also offers a full Integrated Development Environment
VSIDE for full debug capabilities.
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11.3 VS1053b Hardware DAC Audio Paths
Registers Registers
SRC_LEFT
SRC_RIGHT
SDM_LEFT
SDM_RIGHT
SDM
Sigma−delta
Left
Right
CBUF
I2S
SRC
Resampler Sidestream
Analog
drivermodulator
DAC
SRC
Register
Registers
DAC_FCTLL
DAC_FCTLH
SRC_CONTROL SDM_CONTROL
RegisterRegister
DAC_VOL
Registers
DAC_LEFT
DAC_RIGHT
Figure 18: VS1053b ADC and DAC data paths with some data registers
Figure 18 presents the VS1053b Hardware DAC audio paths.
The main audio path starts from the DAC register (Chapter 11.8) to the high-fidelity, fully digital
DAC SRC (Digital-to-Analog Converter SampleRate Converter), which low-pass filters and in-
terpolates the data to the high samplerate of XTALI/2 (nominally 6.144 MHz). This 18-bit data is
then fed to the volume control. It then passes through the sigma-delta modulator to the analog
driver and analog Left and Right signals.
The user may resample and record the data with the Resampler SampleRate Converter (Chap-
ter 11.16). Because there is no automatic low-pass filtering, it is the user’s responsibility to avoid
aliasing distortion.
The user may add a PCM sidestream with the Sidestream Sigma-Delta Modulator input (Chap-
ter 11.17). As is the case with the Resampler SampleRate Converter, hardware doesn’t offer
low-pass filtering, so sufficient aliasing image rejection is the responsibility of the user.
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11.4 VS1053b Hardware ADC Audio Paths
Registers
Multiplexer
ADC decimator
ADC
amplifier
Microphone
MICP
LINE1
MICN
LINE2 ADC_DATA_LEFT
ADC_DATA_RIGHT
Figure 19: VS1053b ADC and DAC data paths with some data registers
Figure 18 presents the VS1053b Hardware ADC audio paths.
Analog audio may be fed up to two channels: one as a differential signal to MICN/MICP or as
a one-sided signal to Line1, and the other as a one-sided signal to Line2.
If microphone input for the left channel has been selected, audio is fed through a microphone
amplifier and that signal is selected by a multiplexer.
Audio is then downsampled to one of four allowed samplerates: XTALI/64, XTALI/128, XTALI/256
or XTALI/512. With the nominal 12.288 MHz crystal, these correspond to 192, 96, 48 or 24 kHz
samplerates, respectively (Chapter 11.17).
If the “3 MHz” option bit SS_AD_CLOCK in register SCI_STATUS has been set to 1, then
samplerates are divided by two, so the nominal samplerates become 96, 48, 24 and 12 kHz.
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11.5 VS1053b Memory Map
X-memory
Address Description
0x0000..0x17ff System RAM
0x1800..0x187f User RAM
0x1880..0x197f Stack
0x1980..0x3fff System RAM
0x4000..0xbfff ROM 32k
0xc000..0xc0ff Peripherals
0xc100..0xffff ROM 15.75k
Y-memory
Address Description
0x0000..0x17ff System RAM
0x1800..0x187f User RAM
0x1880..0x197f Stack
0x1980..0x3fff System RAM
0x4000..0xdfff ROM 40k
0xe000..0xffff System RAM
I-memory
Address Description
0x0000..0x004f System RAM
0x0050..0x0fff User RAM
0x1000..0x1fff -
0x2000..0xffff ROM 56k
and banked
0xc000..0xffff ROM4 16k
11.6 SCI Hardware Registers
SCI registers described in Chapter 9.6 can be found here between 0xC000..0xC00F. In addition
to these registers, there is one in address 0xC010, called SCI_CHANGE.
SCI registers, prefix SCI_
Reg Type Reset Abbrev[bits] Description
0xC010 r 0 CHANGE[5:0] Last SCI access address
SCI_CHANGE bits
Name Bits Description
SCI_CH_WRITE 4 1 if last access was a write cycle
SCI_CH_ADDR 3:0 SCI address of last access
SCI_CHANGE contains the last SCI register that has been accessed through the SCI bus, as
well as whether the access was a read or write operation.
11.7 Serial Data Interface (SDI) Registers
Whenever two bytes have been written to the SDI bus, an interrupt is generated and the data
can be read as a 16-bit big-endian value from the SDI registers. The user can control the DREQ
pin as if it was a general-purpose output through its own register bit.
SDI registers, prefix SER_
Reg Type Reset Abbrev[bits] Description
0xC011 r 0 DATA Last received 2 bytes, big-endian
0xC012 w 0 DREQ[0] DREQ pin control
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11.8 DAC Registers
DAC registers, prefix DAC_
Reg Type Reset Abbrev[bits] Description
0xC013 rw 0 FCTLL DAC frequency control, 16 LSbs
0xC014 rw 0 FCTLH DAC frequency control 4MSbs, PLL control
0xC015 rw 0 LEFT DAC left channel PCM value
0xC016 rw 0 RIGHT DAC right channel PCM value
0xC045 rw 0 VOL DAC hardware volume
The internal 20-bit register DAC_FCTL is calculated from DAC_FCTLH and DAC_FCTLL reg-
isters as follows: DAC_FCTL = (DAC_FCTLH & 15) ×65536 + DAC_FCTLL. Highest supported
value for DAC_FCTL is 0x80000.
If we define C= DAC_FCTL and X= XTALI in Hz, then the resulting samplerate fsof the asso-
ciated DAC SampleRate Converter is fs=C×X×2−27.
Example:
If C= 0x80000 and X= 12.288 MHz then fs= 524288 ×(12.288 ×106)×2−27 = 48000 (Hz).
Note: FCTLH bits 13:4 are used for the PLL Controller. See Chapter 11.9 for details.
DAC_VOL bits
Name Bits Description
LEFT_FINE 15:12 Left channel gain +0.0 dB. . .+5.5 dB (0 to 11)
LEFT_COARSE 11:8 Left channel attenuation in -6 dB steps
RIGHT_FINE 7:4 Right channel volume +0.0 dB. . .+5.5 dB (0 to
11)
RIGHT_COARSE 3:0 Right channel attenuation in -6 dB steps
Normally DAC_VOL is handled by the firmware. DAC_VOL depends on SCI_VOL and the bass
and treble settings in SCI_BASS (and optionally SS_SWING bits in SCI_STATUS).
11.9 PLL Controller
The Phase-Locked Loop (PLL) controller is used to generate clock frequencies that are higher
than the incoming (crystal-based) clock frequency. The PLL output is used by the CPU core
and some peripherals.
Configurable features include:
•VCO Enable/Disable
•Select VCO or input clock to be output clock
•Route VCO frequency to output pin
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•Select PLL clock multiplier
At the core of the PLL controller is the VCO, a high frequency oscillator, whose oscillation
frequency is adjusted to be an integer multiple of some input frequency. As the name “Phase-
Locked Loop” suggests, this is done by comparing the phase of the input frequency against the
phase of a signal which is derived from the VCO output through frequency division.
If the system is stable, e.g. the comparison phase difference remains virtually zero, the PLL is
said to be “in lock”. This means that the output frequency of the VCO is stable and reliable.
The PLL is preceded by a division-by-two unit. Thus, with a nominal XTALI = 12.288 MHz, the
internal clock frequency CLKI can be adjusted with an accuracy of XTALI/2 = 6.144 MHz.
PLL control lies in DAC_FCTL bits 13:4. To see what bits 3:0 do, see Chapter 11.8.
FREQCTLH PLL bits, prefix FCH_
Name Bits Description
PLL_LOCKED 13 0=lock failed since last test (read-only)
PLL_SET_LOCK 12 1:Sets FCH_PLL_LOCKED to 1 to start lock test
PLL_VCO_OUT_ENA 11 Route VCO to GPIO pin (VS1000:second cs pin)
PLL_FORCE_PLL 9 1:System clock is VCO / 0:System clock is inclk
PLL_DIV_INCLK 8 divide inclk by 2 (for 1.5, 2.5 or 3.5 x clk)
PLL_RATE 7:4 PLL rate control
The PLL locked status can be checked by generating a high-active pulse (writing first “1” , then
“0”) to FCH_PLL_SET_LOCK and reading FCH_PLL_LOCKED. FCH_PLL_LOCKED is set to
“1” along with the high level of FCH_PLL_SET_LOCK and to “0” whenever the PLL falls out of
lock. So if the “1” remains in FCH_PLL_LOCKED, PLL is in sync.
The PLL controller’s operation is optimized for frequencies around 12. . . 13 MHz. If you use an
24. . . 26 MHz input clock, set the extra clock divider bit SM_CLK_RANGE in register SCI_MODE
to 1 before activating the PLL.
It’s recommended to change the PLL rate in small steps and wait for the PLL to stabilize after
each change. For diagnostic purposes, the PLL clock output (VCO) can be routed to an I/O pin
so it can be scanned with an oscilloscope.
FCH_PLL_RATE (bits 7:4) control PLL multiplication rate. PLL multiplier is (FCH_PLL_RATE
+ 1). When FCH_PLL_RATE is 0, the VCO is powered down and output clock is forced to be
input clock (same as if FCH_PLL_FORCE_PLL = 0).
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11.10 GPIO
GPIO registers, prefix GPIO_
Reg Type Reset Abbrev[bits] Description
0xC017 rw 0 DDR[9:0] Direction
0xC018 r 0 IDATA[11:0] Values read from the pins
0xC019 rw 0 ODATA[8:0] Values set to the pins
GPIO_DIR is used to set the direction of the GPIO pins. 1 means output. GPIO_ODATA
remembers its values even if a GPIO_DIR bit is set to input.
GPIO_IDATA is used to read the pin states. In VS1053 also the SDI and SCI input pins
can be read through GPIO_IDATA: SCLK = GPIO_IDATA[8], XCS = GPIO_IDATA[9], SI =
GPIO_IDATA[10], and XDCS = GPIO_IDATA[11].
In addition to data direction control for GPIO pins 0 to 7, there are two additional control bits in
GPIO_DIR. GPIO_DIR[8] switches SO into software control, so that the value in GPIO_ODATA[8]
is shown on SO whenever xCS is low. When GPIO_DIR[9] is set to ’1’ SCI and SDI are dis-
abled. When using SCLK, xCS, SI and xDCS as general-purpose input, GPIO_DIR[9] prevents
transitions in them from getting random data from SDI and/or SCI.
GPIO registers don’t generate interrupts.
Note that in VS1053b the VSDSP registers can be read and written through the SCI_WRAMADDR
and SCI_WRAM registers. You can thus use the GPIO pins quite conveniently.
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11.11 Interrupt Control
Interrupt registers, prefix INT_
Reg Type Reset Abbrev[bits] Description
0xC01A rw 0 ENABLE[9:0] Interrupt enable
0xC01B w 0 GLOB_DIS[-] Write to add to interrupt counter
0xC01C w 0 GLOB_ENA[-] Write to subtract from interrupt counter
0xC01D rw 0 COUNTER[4:0] Interrupt counter
INT_ENABLE controls the interrupts. The control bits are as follows:
INT_ENABLE bits
Name Bits Description
INT_EN_SDM 9 Enable Sigma Delta Modulator interrupt
INT_EN_SRC 8 Enable SampleRate Converter interrupt
INT_EN_TIM1 7 Enable Timer 1 interrupt
INT_EN_TIM0 6 Enable Timer 0 interrupt
INT_EN_RX 5 Enable UART RX interrupt
INT_EN_TX 4 Enable UART TX interrupt
INT_EN_ADC 3 Enable AD modulator interrupt
INT_EN_SDI 2 Enable Data interrupt
INT_EN_SCI 1 Enable SCI interrupt
INT_EN_DAC 0 Enable DAC interrupt
Note: It may take up to 6 clock cycles before changing INT_ENABLE has any effect.
Writing any value to INT_GLOB_DIS adds one to the interrupt counter INT_COUNTER and
effectively disables all interrupts. It may take up to 6 clock cycles before writing to this register
has any effect.
Writing any value to INT_GLOB_ENA subtracts one from the interrupt counter INT_COUNTER,
unless it already was 0, in which case nothing happens. If, after the operation INT_COUNTER
becomes zero, interrupts selected with INT_ENABLE are restored. An interrupt routine should
always write to this register as the last thing it does, because interrupts automatically add one
to the interrupt counter, but subtracting it back to its initial value is the responsibility of the user.
It may take up to 6 clock cycles before writing this register has any effect.
By reading INT_COUNTER the user may check if the interrupt counter is correct or not. If the
register is not 0, interrupts are disabled.
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11.12 UART (Universal Asynchronous Receiver/Transmitter)
The RS232 UART implements a serial interface using RS232 standard 8N1 (8 data bits, no
parity, 1 stop bit).
Start
bit D0 D1 D2 D3 D4 D5 D6 D7 Stop
bit
Figure 20: RS232 serial interface protocol
When the line is idling, it stays in logic high state. When a byte is transmitted, the transmission
begins with a start bit (logic zero) and continues with data bits (LSB first) and ends up with a
stop bit (logic high). 10 bits are sent for each 8-bit byte frame.
11.12.1 UART Registers
UART registers
Reg Type Reset Abbrev[bits] Description
0xC028 r 0 UART_STATUS[4:0] Status
0xC029 r/w 0 UART_DATA[7:0] Data
0xC02A r/w 0 UART_DATAH[15:8] Data High
0xC02B r/w 0 UART_DIV Divider
11.12.2 Status UART_STATUS
A read from the status register returns the transmitter and receiver states.
UART_STATUS bits
Name Bits Description
UART_ST_FRAMEERR 4 Framing error (stop bit was 0)
UART_ST_RXORUN 3 Receiver overrun
UART_ST_RXFULL 2 Receiver data register full
UART_ST_TXFULL 1 Transmitter data register full
UART_ST_TXRUNNING 0 Transmitter running
UART_ST_FRAMEERR is set if the stop bit of the received byte was 0.
UART_ST_RXORUN is set if a received byte overwrites unread data when it is transferred from
the receiver shift register to the data register, otherwise it is cleared.
UART_ST_RXFULL is set if there is unread data in the data register.
UART_ST_TXFULL is set if a write to the data register is not allowed (data register full).
UART_ST_TXRUNNING is set if the transmitter shift register is in operation.
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11.12.3 Data UART_DATA
A read from UART_DATA returns the received byte in bits 7:0, bits 15:8 are returned as ’0’. If
there is no more data to be read, the receiver data register full indicator will be cleared.
A receive interrupt will be generated when a byte is moved from the receiver shift register to
the receiver data register.
A write to UART_DATA sets a byte for transmission. The data is taken from bits 7:0, other
bits in the written value are ignored. If the transmitter is idle, the byte is immediately moved
to the transmitter shift register, a transmit interrupt request is generated, and transmission is
started. If the transmitter is busy, the UART_ST_TXFULL will be set and the byte remains in the
transmitter data register until the previous byte has been sent and transmission can proceed.
11.12.4 Data High UART_DATAH
The same as UART_DATA, except that bits 15:8 are used.
11.12.5 Divider UART_DIV
UART_DIV Bits
Name Bits Description
UART_DIV_D1 15:8 Divider 1 (0..255)
UART_DIV_D2 7:0 Divider 2 (6..255)
The divider is set to 0x0000 in reset. The ROM boot code must initialize it correctly depending
on the master clock frequency to get the correct bit speed. The second divider (D2) must be
from 6 to 255.
The communication speed f=fm
(D1+1)×(D2), where fmis the master clock frequency, and fis
the TX/RX speed in bps.
Divider values for common communication speeds at 26 MHz master clock:
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Example UART Speeds, fm= 49.152 MHz
Comm. Speed [bps] UART_DIV_D1 UART_DIV_D2
4800 255 40
9600 255 20
14400 233 15
19200 255 10
28800 243 7
38400 159 8
57600 121 7
115200 60 7
11.12.6 UART Interrupts and Operation
Transmitter operates as follows: After an 8-bit word is written to the transmit data register it will
be transmitted instantly if the transmitter is not busy transmitting the previous byte. When the
transmission begins a TX_INTR interrupt will be sent. Status bit [1] informs the transmitter data
register empty (or full state) and bit [0] informs the transmitter (shift register) empty state. A
new word must not be written to transmitter data register if it is not empty (bit [1] = ’0’). The
transmitter data register will be empty as soon as it is shifted to transmitter and the transmission
is begun. It is safe to write a new word to transmitter data register every time a transmit interrupt
is generated.
Receiver operates as follows: It samples the RX signal line and if it detects a high to low
transition, a start bit is found. After this it samples each 8 bit at the middle of the bit time (using
a constant timer), and fills the receiver (shift register) LSB first. Finally the data in the receiver
is moved to the reveive data register, the stop bit state is checked (logic high = ok, logic low =
framing error) for status bit[4], the RX_INTR interrupt is sent, status bit[2] (receive data register
full) is set, and status bit[2] old state is copied to bit[3] (receive data overrun). After that the
receiver returns to idle state to wait for a new start bit. Status bit[2] is zeroed when the receiver
data register is read.
RS232 communication speed is set using two clock dividers. The base clock is the processor
master clock. Bits 15-8 in these registers are for first divider and bits 7-0 for second divider. RX
sample frequency is the clock frequency that is input for the second divider.
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11.13 Timers
There are two 32-bit timers that can be initialized and enabled independently of each other. If
enabled, a timer initializes to its start value, written by a processor, and starts decrementing
every clock cycle. When the value goes past zero, an interrupt is sent, and the timer initializes
to the value in its start value register, and continues downcounting. A timer stays in that loop
as long as it is enabled.
A timer has a 32-bit timer register for down counting and a 32-bit TIMER1_LH register for
holding the timer start value written by the processor. Timers have also a 2-bit TIMER_ENA
register. Each timer is enabled (1) or disabled (0) by a corresponding bit of the enable register.
11.13.1 Timer Registers
Timer registers, prefix TIMER_
Reg Type Reset Abbrev Description
0xC030 r/w 0 CONFIG[7:0] Timer configuration
0xC031 r/w 0 ENABLE[1:0] Timer enable
0xC034 r/w 0 T0L Timer0 startvalue - LSBs
0xC035 r/w 0 T0H Timer0 startvalue - MSBs
0xC036 r/w 0 T0CNTL Timer0 counter - LSBs
0xC037 r/w 0 T0CNTH Timer0 counter - MSBs
0xC038 r/w 0 T1L Timer1 startvalue - LSBs
0xC039 r/w 0 T1H Timer1 startvalue - MSBs
0xC03A r/w 0 T1CNTL Timer1 counter - LSBs
0xC03B r/w 0 T1CNTH Timer1 counter - MSBs
11.13.2 Configuration TIMER_CONFIG
TIMER_CONFIG Bits
Name Bits Description
TIMER_CF_CLKDIV 7:0 Master clock divider
TIMER_CF_CLKDIV is the master clock divider for all timer clocks. The generated internal
clock frequency fi=fm
c+1 , where fmis the master clock frequency and cis TIMER_CF_CLKDIV.
Example: With a 12 MHz master clock, TIMER_CF_DIV=3 divides the master clock by 4, and
the output/sampling clock would thus be fi=12MHz
3+1 = 3MHz.
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11.13.3 Configuration TIMER_ENABLE
TIMER_ENABLE Bits
Name Bits Description
TIMER_EN_T1 1 Enable timer 1
TIMER_EN_T0 0 Enable timer 0
11.13.4 Timer X Startvalue TIMER_Tx[L/H]
The 32-bit start value TIMER_Tx[L/H] sets the initial counter value when the timer is reset. The
timer interrupt frequency ft=fi
c+1 where fiis the master clock obtained with the clock divider
(see Chapter 11.13.2 and cis TIMER_Tx[L/H].
Example: With a 12 MHz master clock and with TIMER_CF_CLKDIV=3, the master clock fi=
3MHz. If TIMER_TH=0, TIMER_TL=99, then the timer interrupt frequency ft=3M Hz
99+1 =
30kHz.
11.13.5 Timer X Counter TIMER_TxCNT[L/H]
TIMER_TxCNT[L/H] contains the current counter values. By reading this register pair, the user
may get knowledge of how long it will take before the next timer interrupt. Also, by writing to
this register, a one-shot different length timer interrupt delay may be realized.
11.13.6 Timer Interrupts
Each timer has its own interrupt, which is asserted when the timer counter underflows.
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11.14 I2S DAC Interface
The 16-bit I2S Interface makes it possible to attach an external DAC to the system.
Note: The samplerate of the audio file and the I2S rate are independent. All audio will be
automatically converted to 6.144 MHz for VS1053 DAC and to the configured I2S rate using a
high-quality sample-rate converter.
Note: In VS1053b the I2S pins share different GPIO pins than in VS1033 to be able to use SPI
boot and I2S in the same application.
I2S registers, prefix I2S_
Reg Type Reset Abbrev Description
0xC040 r/w 0 CONFIG[3:0] I2S configuration
I2S_CONFIG Bits
Name Bits Description
I2S_CF_MCLK_ENA 3 Enables the MCLK output (12.288 MHz)
I2S_CF_ENA 2 Enables I2S, otherwise pins are GPIO
I2S_CF_SRATE 1:0 I2S rate, "10" = 192, "01" = 96, "00" = 48 kHz
I2S_CF_ENA enables the I2S interface. After reset I2S is disabled and the pins are used for
GPIO inputs.
I2S_CF_MCLK_ENA enables the MCLK output. The frequency is either directly the input clock
(nominal 12.288 MHz), or half the input clock when mode register bit SM_CLK_RANGE is set
to 1 (24-26 MHz input clock).
I2S_CF_SRATE controls the output samplerate. When set to 48 kHz, SCLK is MCLK divided
by 8, when 96 kHz SCLK is MCLK divided by 4, and when 192 kHz SCLK is MCLK divided by
2. I2S_CF_SRATE can only be changed when I2S_CF_ENA is 0.
MCLK
SDATA
SCLK
LROUT
MSB LSB MSB
Left Channel Word Right Channel Word
Figure 21: I2S interface, 192 kHz.
To enable I2S first write 0xc017 to SCI_WRAMADDR and 0x00f0 to SCI_WRAM, then write
0xc040 to SCI_WRAMADDR and 0x000c to SCI_WRAM.
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11.15 Analog-to-Digital Converter (ADC)
ADC modulator registers control Analog-to-Digital conversions of VS1053b.
ADC Decimator registers, prefix ADC_
Reg Type Reset Abbrev[bits] Description
0xC042 rw 0 CONTROL[4:0] ADC control
0xC043 r 0 DATA_LEFT ADC left channel data
0xC044 r 0 DATA_RIGHT ADC right channel data
ADC_CONTROL controls the ADC and its associated decimator unit.
ADC_CONTROL Bits
Name Bits Description
ADC_MODU2_PD 4 Right channel powerdown
ADC_MODU1_PD 3 Left channel powerdown
ADC_DECIM_FACTOR 2:1 ADC Decimator factor:
- 3 = downsample to XTALI/512 (nominal 24 kHz)
- 2 = downsample to XTALI/256 (nominal 48 kHz)
- 1 = downsample to XTALI/128 (nominal 96 kHz)
- 0 = downsample to XTALI/64 (nominal 192 kHz)
ADC_ENABLE 0 Set to activate ADC converter and decimator
Note: Setting bit SS_AD_CLOCK in register SCI_STATUS will halve the operation speed of the
A/D unit, and thus halve the resulting samplerate.
Each time a new (stereo) sample has been generated, an ADC interrupt is generated.
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11.16 Resampler SampleRate Converter (SRC)
The resampler SRC makes it possible to catch audio from the DAC path.
Note: hardware makes no attempts at low-pass filtering data. If the SRC samplerate is lower
than the DAC samplerate, aliasing may and will occur.
Resampler SRC registers, prefix SRC_
Reg Type Reset Abbrev[bits] Description
0xC046 rw 0 CONTROL[12:0] SRC control
0xC047 r 0 DATA_LEFT SRC left channel data
0xC048 r 0 DATA_RIGHT SRC right channel data
SRC_CONTROL Bits
Name Bits Description
SRC_ENABLE 12 Set to enable SRC
SRC_DIV 11:0 Set samplerate to XTALI/2/(SRC_DIV+1)
Each time a new (stereo) sample has been generated, an SRC interrupt is generated.
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11.17 Sidestream Sigma-Delta Modulator (SDM)
The Sidestream Sigma-Delta Modulator makes it possible to insert a digital side stream on top
of existing audio.
Note: The SDM provides a direct, low-delay side channel to the Sigma-Delta DACs of VS10xx.
It makes no attempts at low-pass filtering data. Thus there will be practically no image rejection.
If using low samplerates, this may cause audible aliasing distortion.
Sidestream SDM registers, prefix SDM_
Reg Type Reset Abbrev[bits] Description
0xC049 rw 0 CONTROL[12:0] SDM control
0xC04A rw 0 DATA_LEFT SDM left channel data
0xC04B rw 0 DATA_RIGHT SDM right channel data
SDM_CONTROL Bits
Name Bits Description
SDM_ENABLE 12 Set to enable SDM
SDM_DIV 11:0 Set samplerate to XTALI/2/(SDM_DIV+1)
Each time a new (stereo) sample is needed, an SDM interrupt is generated.
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12 Version Changes
This chapter describes the lastest and most important changes done to VS1053b
12.1 Changes Between VS1033c and VS1053a/b Firmware, 2007-03-08
Completely new or major changes:
•I2S pins are now in GPIO4-GPIO7 and do not overlap with SPI boot pins.
•No software reset required between files when used correctly.
•Ogg Vorbis decoding added. Non-fatal ogg or vorbis decode errors cause automatic
resync. This allows easy rewind and fast forward. Decoding ends if the "last frame" flag
is reached or SM_CANCEL is set.
•HE-AAC v2 Level 3 decoding added. It is possible to disable PS and SBR processing and
control the upsampling modes through
parametric_x.control1
.
•Like the WMA decoder, the AAC decoder uses the clock adder (see SCI_CLOCKF) if it
needs more clock to decode the file. HE-AAC features are dropped one by one, if the file
can not be decoded correctly even with the highest allowed clock. Parametric stereo is
the first feature to be dropped, then downsampled mode is used, and as the final resort
Spectral Band Replication is disabled. Features are automatically restored for the next
file.
•Completely new volume control with zero-cross detection prevents pops when volume is
changed.
•Audio FIFO underrun detection (with slow fade to zero) instead of looping the audio buffer
content.
•Average bitrate calculation (
byteRate
) for all codecs.
•All codecs support fast play mode with selectable speeds for the best-quality fast forward
operation. Fast play also advances DECODE_TIME faster.
•WMA and Ogg Vorbis provide an absolute decode position in milliseconds.
•When SM_CANCEL is detected, the firmware also discards the stream buffer contents.
•Bit SCIST_DO_NOT_JUMP in SCI_STATUS is ’1’ when jumps in the file should not be
done: during header processing and with Midi files.
•IMA ADPCM encode now supports stereo encoding and selectable samplerate.
Other changes or additions:
•Delayed volume and bass/treble control calculation reduces the time the corresponding
SCI operations take. This delayed handling and the new volume control hardware pre-
vents audio samples from being missed during volume change.
•SCI_DECODE_TIME only cleared at hardware and software reset to allow files to be
played back-to-back or looped.
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•Read and write to YRAM at 0xe000..0xffff added to SCI_WRAMADDR/SCI_WRAM.
•The
resync
parameter (parametric_x.resync) is set to 32767 after reset to allow inifinite
resynchronization attempts (or until SM_CANCEL is set). Old operation can be restored
by writing 0 to
resync
after reset.
•WMA,AAC: more robust resync.
•WMA,AAC: If resync is performed, broadcast mode is automatically activated. The broad-
cast mode disables file size checking, and decoding continues until SM_CANCEL is set
or reset is performed.
•Treble control fixed (volume change could cause bad artefacts).
•MPEG Layer I mono fixed.
•MPEG Layer II half-rate decoding fixed (frame size was calculated wrong).
•MPEG Layer II accuracy problem fixed, invalid grouped values set to 0.
•WAV parser now skips unknown RIFF chunks.
•IMA ADPCM: Maximum blocksize is now 4096 bytes (4088 samples stereo, 8184 mono).
Thus, now also plays 44100Hz stereo.
•Rt-midi: starts if in reset GPIO0=’0’, GPIO1=’1’, GPIO2&3 give earSpeaker setup.
•NewSinTest() and NewSinSweep() added (AIADDR = 0x4020/0x4022) AICTRL0 and AIC-
TRL1 set sin frequency for left/right.
•Clears memory before SPI boot and not in InitHardware().
Known quirks, bugs, or features in VS1053b:
•Setting volume clears SS_REFERENCE_SEL and SS_AD_CLOCK bits. See Chap-
ter 9.6.2.
•Software reset clears GPIO_DDR, also affects I2S pins.
•Ogg Vorbis occasionally overflows in windowing causing a small glitch to audio. Patch
available (VS1053b Patches w/ FLAC Decoder plugin at
http://www.vlsi.fi/en/support/software/vs10xxpatches.html).
•IMA ADPCM encoding requires short patch to start. Patch available in Chapter 10.8.1.
•There are also fixes for some other issues, we recommend you use the latest version of
the
VS1053b Patches w/ FLAC Decoder package from
http://www.vlsi.fi/en/support/software/vs10xxpatches.html.
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13 Latest Document Version Changes
This chapter describes the latest and most important changes to this document.
Version 1.31, 2017-11-17
•Split table in Chapter 4.3, Analog Characteristics, into two.
•Removed MP2 and MP3 license descriptions from Chapter 1, Licenses, and other rele-
vant places, as all their patents have expired.
•Improved imafix in Chapter 10.8.1.
Version 1.30, 2016-12-21
•Updated numbers in Chapter 4, Characteristics & Specifications, according to re-qualification
results.
•Added mention of 8N1 format to Chapter 11.12, UART (Universal Asynchronous Re-
ceiver/Transmitter).
Version 1.22, 2014-12-19
•Added mention of RIFF 8-bit and 16-bit data signedness to Chapter 10.6, Feeding PCM
Data.
•Updated telephone number in Chapter 14, Contact Information.
Version 1.21, 2014-08-13
•Clarified how Ogg Vorbis set Replay Gain parameters in Chapter 10.11.5, Ogg Vorbis.
•Explained that Bass Enhancer handles the bass part of the audio signal in mono in Chap-
ter 9.6.3, SCI_BASS.
•Added GPIO_DDR[9:8] and GPIO_ODATA[8] explanation to Chapter 11.10, GPIO.
•Moved value from tV(min) to tV(max) in Chapter 7.4.4, SPI Timing Diagram.
•Chapter 10.8, PCM / ADPCM Recording, has been updated to better explain how to
record using the VS1053b Patches package.
•Clarified byte endianness when reading PCM or ADPCM audio data in Chapter 10.8.3,
Adding a PCM RIFF Header, and Chapter 10.8.4, Adding an IMA ADPCM RIFF Header.
Version: 1.31, 2017-11-17 89
VS1053b Datasheet
14 CONTACT INFORMATION
14 Contact Information
VLSI Solution Oy
Entrance G, 2nd floor
Hermiankatu 8
FI-33720 Tampere
FINLAND
URL: http://www.vlsi.fi/
Phone: +358-50-462-3200
Commercial e-mail: sales@vlsi.fi
For technical support or suggestions regarding this document, please participate at
http://www.vsdsp-forum.com/
For confidential technical discussions, contact
support@vlsi.fi
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