Audio
System DesignECE 542 Spring 2005
Dr. T. Tran
Rice University
Introduction: The Wide World of Audio
Analog and Digital Audio
Algorithms for Audio Processing
OMAP Audio Algorithms
DSP Architectures for the Future of Audio
OMAP Audio Design
Class D Audio Amplifier
Agenda
2
Agenda: Introduction
3
Introduction: The Wide World of Audio
Analog and Digital Audio
Algorithms for Audio Processing
OMAP Audio Algorithms
DSP Architectures for the Future of Audio
OMAP Audio Design
Class D Audio Amplifier
Introduction: Wide World of AudioThe Wide World of Audio
The audio life cycleAudio takes a journey from lips to ears
Audio in the 21st CenturyMore and more technologies are converging onto consumer entertainment electronics
Different phases, different facesNot just a ‘one size fits all’ approach
What we have to doBecome a reliable partner with our customers
How TI can meet the customers’ needsDelivering outstanding audio technologies and audio system solutions
4
The Audio Life Cycle
CaptureCapture
CreateCreate
ManipulateManipulate
ProduceProduceChannelDistribution
ChannelDistribution
StoreStore
Personal DistributionPersonal
Distribution
ConsumeConsume
Audio in the 21st CenturyIn the “Good Old Days”...
A new music delivery format once every ~20 yearsPortability was only for those with very strong arms
Some signs of change in the last five years:Significant new physical formats - MD, SACD, DVDNew non physical formats – Streaming audio over networksNew broadcast sources - DAB, Digital TV, HDTVMore and more people are making their own music
Now every month brings something newThe musical WWW reaches critical massWith some audio standards even set by a software company...Soon, every tune in the world can reach every ear in the world (provided they have IP addresses...)GOOD AUDIO EXPERIENCE MAKES BETTER VIDEO EXPERIENCE, NOT THE OTHER WAY AROUND!
Phonograph
Boom Box
MP3 Player
Agenda: Analog & Digital Audio
Introduction: The Wide World of Audio
Analog and Digital Audio
Algorithms for Audio Processing
OMAP Audio Algorithms
DSP Architectures for the Future of Audio
OMAP Audio Design
Class D Audio Amplifier
Analog and Digital Audio
ADC DSP DAC
Typical Audio System
AMP
CD,DVD
Analog Digital Digital Analog High-PowerAnalog
AnalogAudio
DigitalAudio
AnalogAudio
AnalogAudio
AnalogAudio
AnalogAudio
The whole world is analog and digital only plays a bit part
Analog Audio: Inputs
Microphone inputElectret (moving capacitive plates) or dynamic (moving coil) condenser microphone: Electret requires DC bias to functionVery low level input (~100mV): pre-amp of about 20dB requiredMost critical input as far as noise sensitivity: requires special care!Mic performance is very important for good speech recognition, video phone, karaoke, and full-duplex speakerphone
Microphone Front-End IssuesYou need a high performance solution with low component count, small board area and component bulk, simple integration into your digital control environment
PGA2500 programmable gain 10dB to 65dB in 1dB step.Extremely low noise and low distortion, 1nV/√Hz and 0.00002% THD
Simple interface to A-to-D converters: careful matching of supply voltages and signal levels can create an elegant input stage design
Line level inputsLine level amplitude varies greatly depending on the audio source, for example portable CD to DVD player.Typically, Line In = 200mVrms to 2Vrms, where 1Vrms = 2.828Vp-p
Issues related to line level:• Portable devices can’t drive audio system to the
maximum power. • High-end DVD players can overdrive the audio
system inputs and cause severe distortion.Audio Engineering Society (AES) has published many standards to define the audio levels for different types of equipment but only a few manufactures follow this recommendation.
Analog Audio: Inputs
Example of audio line level input problem
Analog Audio: Inputs
ADC
3.3V
ground
L-AudioR-Audio2Vrms/
5.656Vp-p
Input level is clipped at 3.3V, overdriving the CODEC
One solution to the clipping problem
Analog Audio: Inputs
ADC
3.3V
ground
L-AudioR-Audio
Volt.Divider
Add a voltage divider network to limit the AC input signals to theCODEC supply voltage. For example,
R1R2
)21(2656.53.3RR
RVV+
= R2 = 1.4 R1 Let R1 = 20K -> R2 = 28K,Keep input impedance greater than 20k
Analog to Digital Conversion Issues
You need enough dynamic range and performance to cope with almost any situation at almost any sample rate, but without burning lots of power, taking lots of board space or spending lots of money
A clear case of where the best product is in fact the best compromise between competing requirements; in real-world products, you don’t want the extra cost, board area and heat that a few extra dB’s brings - and you’ll lose a lot of versatility and performance if you try to save any more money
This function is the ‘gateway to the soul’ of your equipment, and you want the reassurance of the long experience of the Burr-Brown division of TI
ADCSR
Deci.By N
AnalogFilter
AudioInput
Nfs
DigitalLPF
AIC23 CODEC
fsData rate
Nfs Data Rate
Oversample by N pushes the quantization noise way above the audio spectrum. Sampling at Nyquist rate (2 times the highest frequency component) requires higher order anti-aliasing filter.Digital LPF removes the noise generated by the ADC. Sample Rate (SR) Decimation reduces the data rate for audio processing
Oversampling ADC
Noi
se D
ensi
ty
Frequency
Noise shaping characteristic (sigma-delta A/D)
Nfs/2fs/2
Nfs Sample Rate
Digital LPF Passband
Oversampling ADC
Watch for these potential problemsAnalog input filter is required to limit the audio bandwidth. Without this, out-of-band noise folds back into the audio spectrum and degrades the CODEC performance.Isolate the analog power supply by using a high performance linear voltage regulator.The reference voltage Vref needs to be decoupled properly. This voltage is used for sampling threshold comparison so it needs to be very clean.Routing all the input signals carefully and away from high speed and high current switching signals.
D-to-A Conversion IssuesThe required state-of-the-art in replay keeps rising; better dynamic range, less distortion, lower solution cost, greater filtering flexibility for both DSD and PCM operation over a wide Fs range
You try to make your clocks as good as possible but they are never perfect; you need good tolerance to clock jitter
Not made of money: you need wide dynamic range and THD in real applications using affordable external components, not made out of Unobtanium
Good control of out-of-band noise is critical in broadcast applications, and in general any application where the D-to-A output feeds any kind of converter or modulator
Format flexibility (PCM, DSD) and filter transient response choice to deliver low delay and good sound quality
Oversampling DAC
DAC AnalogLPF
DigitalAudioInput
Nfs
Inter.Filter
AIC23 CODEC
AnalogAudio
Nfs Data Rate
Interpolation filter increases the sample rate by N for the oversample DAC to process the data.Analog lowpass filter is required to remove the sampling noise. This filter is included in the TLV320AIC23 device.
Oversampling DAC
Watch for these potential problems Isolate the analog power supply by using a high performance linear voltage regulator.Isolate the analog filter supply from the DAC supply as recommended by the manufacture.The reference voltage Vref needs to be decoupled properly. This voltage is used for sampling threshold comparison so it needs to be very clean.
TLV320AIC23 Stereo CODEC
1.0X
1.0X
1.0X
1.5X
Ω50k
Ω50k
VADC
VDAC
VMID
2:1MUX
+12 to -34dB,1.5dB Steps
Σ−∆ADC
2:1MUX
+12 to -34dB,1.5dB Steps
Σ−∆ADC
Ω50k
Ω10k
VMID
Mute,0db, 20dB
Σ−∆DAC
+6 to -73dB,1dB Steps
HeadphoneDriver
Σ−∆DAC
+6 to -73dB,1dB Steps
HeadphoneDriver
DigitalFilters
DigitalAudio
Interface
ControlInterface
OSC
Σ
Σ
Divider(1X, 1/2X)
AVDD
VMID
AGND
MICBIAS
RLINEIN
MICIN
LLINEIN
HPGND
HPVDD
RHPOUT
ROUT
LOUT
LHPOUT
XTI/MCLK
XTO
CLKOUT
/CS
SDIN
SCLK
MODE
DVDD
DGND
LRCIN
DIN
LRCOUT
DOUT
BCLK
VADC
VDAC
BVDD
TLV320AIC23 Stereo CODEC
DSP
0.1uF 10uF
3.3V
0.1uF 10uF
10uF
3.3V
5.6K
5.6K 220pF
0.1uF
5.6K
5.6K 220pF
0.1uF
5.6K
47K
1.0uF
3.3V
5.6K
47K
1.0uF
5.6K
47K
220uF
5.6K
47K
220uF
100
0.1uF
3.3V
1.5 - 3.3V
0.1uF10uF
10uF 0.1uF
220pF47K
680
Rmic
1uF
AVDD (14)
AGND (15)
HPVDD (8)
HPGND (11)
LHPOUT (9)
LOUT (12)
ROUT (13)
RHPOUT (10)
CLKOUT (2)
VMID (16)
XTI/MCLK (25)
XTO(26)
BVDD (1)
DGND (28)
LLINEIN (20)
RLINEIN (19)
MICBIAS (17)
MICIN (18)
DVDD (27)
15pF 15pF
LRCIN (5)
DIN (4)
DOUT (6)
LRCOUT (7)
BCLK (3)
CSB (21)
SDIN (23)
SCLK (24)
MODE (22)
SCLK (24)
3.3V
10K
BCLKX1 (33)
BDX1 (36)
BDX0 (24)
BFSX1(35)
BDR0 (30)
BFSR0 (29)BCLKR0 (27)
BFSX0 (23)
DSP PIN NUMBERREFERS TO COMMON80-PIN CONNECTOR
AIC23
FSA 3/21/01
Analog inputfilters
S/PDIF Digital Interfaces
S/PDIF, Sony/Philips Digital Interface, has been adapted as an I/O interface for digital audio and other consumer electronics equipment.
S/PDIF connector comes in two types, optical and/or RCA. Optical provides higher performance due to noise isolation but it costs more because of the additional optical transceiver.
Low jitter is a must for this interface, embedded clock and data on a single conductor.
New TI SPACT technology completely eliminates performance degradation through plastic optical connections (long cables, low supply voltage...) even at high sample rate
Effect of Interface Jitter
0.002
0.020
0.004
0.006
0.008
0.010
0.012
0.014
0.016
0.018
20 20k50 100 200 500 1k 2k 5k 10k
Audio Signal Frequency [Hz]
5 ns 2 ns
1 ns
0.5 ns250 ps
THD+N [%]
0 ps
‘Analog’ DIR 1.5 ns
Effect of Interface Jitter
0.002
0.020
0.004
0.006
0.008
0.010
0.012
0.014
0.016
0.018
20 20k50 100 200 500 1k 2k 5k 10k
Audio Signal Frequency [Hz]
5 ns 2 ns 1.5 ns
1 ns
0.5 ns250 ps
THD+N [%]
0 ps
‘Analog’ DIRSPACT DIR
Algorithms for Audio Processing
Introduction: The Wide World of Audio
Analog and Digital Audio
Algorithms for Audio Processing
OMAP Audio Algorithms
DSP Architectures for the Future of Audio
OMAP Audio Design
Class D Audio Amplifier
Lossless vs. Lossy Compression
Coding that is bit exact, i.e., the original signal is recovered bit for bit. Examples of lossless coding include numerical coders:
Arithmetic coding, Ziv-Lempel (LZW) coding / WinZipHuffman coding / FaxRun Length CodingMeridian Lossless Packing (MLP) / DVD Audio
Coders that create an approximate reproduction of the input signal, i.e., “sounds like”the original speech or audio or “looks like” the original image. Examples of lossy coding includes source coders:
Sub-band coding / MP3Transform coding / JPEG2000Vector quantization / MPEG-4
Lossless Coding Lossy Coding
Lossless Coding
Assign the most frequent events to the shortest code words.Non-duplicating prefix, for example 110 and 11011 can’t be code words
Huffman Coding
Huffman coding example:Bus Status Probability Code wordOn time 0.5 0Late 0.35 10Early 0.125 110Wrecked 0.025 111
0
111
00
Run Length CodingSpecial start and stop codes for repeated data.Only send the value and how many times the value is repeated; for example 5555 5555 is coded as 85.
Frequency domainPrevalent when signal is approximately periodic
Expressed using non-uniform frequency scale derived from physiology of cochlea
Subband coding (AC-3, DTS, MP3, etc.)
Time domainPrevalent when signal contains large transient componentsDue to mechanical processes and early neural processingMasking components occurring just before and after a transient
Human Audio PerceptionProperties of auditory perception can be used to exploit irrelevancy in audio that hides noise where you can not perceive itStrategy: Identify signal components for which the ear is less sensitive to distortionEffects are typically signal dependentEffects are broadly categorized into:
critical bands
96 (dB)
22 kHz0
Hearing threshold
Number of quantization bits
Exploiting Perceptual Effects
Typically encoders/decoders perform these steps: 1. Analyzes input in frequency domain (critical subbands).2. The ear masks (does not hear) neighborhood low amplitude signals of a
certain frequency in the presence of a nearby higher-amplitude signal. Thus the psychoacoustic model helps ignore these low amplitude signals well (frequency irrelevancy).
3. The psychoacoustic model also helps gauge which subbands the ear does not hear as well. Scales the number of bits in relation to hearing threshold and masking signals.
SideInformationand optionalAncillary Data
Digital Audio Input
EncodedAudioBitstream32
Sub-bandFilterbank
Bits
trea
mFo
rmat
ting
Run Length
Encoding
Bit Allocation,
Quantization,and Coding
Block Diagram of an MP3 Encoder(by layer)
ExternalControl
Layer 3
Layer 2enhancements
Layer 1
Modified Discrete Cosine
Transform
1024 Points
FFT
PsychoacousticModel
+
HuffmanCoding
Widely Used Audio StandardsWidely Used ISO: MPEG Audio Coders:
MPEG-1 Audio Layers 1, 2 and 3 (MP3 is the popular name for MPEG-1 Audio Layer 3)MPEG-2 Audio Layers 1, 2 and 3: 32-256 kb/s/ch European DVD and HDTVMPEG-2 Advanced Audio Coder (AAC): 8-160 kb/s/ch Japanese HDTVMPEG-4 Speech and Audio Coders - MPEG-2 AAC (with extensions)
Other Audio Coders:Sony ATRAC: Primarily 2-ch coders used in Mini-Disk players at 144 kb/s/chMicrosoft WMA: Designed for 1- and 2-ch streaming, 5 kb/s/ch to 128 kb/s/ch, 8 to 48 kHz sampling Real Audio: Designed for 1- and 2-ch streaming, 5 to 96 kb/s/ch, 8 to 48 kHz samplingDolby Digital AC-3: Theater/Home multichannel standard. 5.1 channels, US DVD (5.1 ch/320 kb/s), US HDTVDTS (Digital Theater System): Home multichannel competitor. 5.1-ch system w/48 kHz sampling.
MLP (Meridian Lossless packing): Lossless coder for DVD-Audio. Multichannel system up to 24bit/192kHz
Other Standardized Audio Processes:Dolby PrologicDTS NEO:6THX
Audio Effects Classification
Amplitude based effectsVolume, Panning, Tremolo, Noise gating, Compression/Expansion, …
Time delayModulated delay: Chorus, Flanger, Phaser, Vibrato, …Fixed delay: Echo/Delay, Reverb, …
Waveform shaping effectsVacuum tube simulator, Non-linear effects
Frequency filtersNoise Reduction, Pitch-shifter, Vocoding, Equalizer, …
Compression/Expansion
Reverb
Acoustic Effects Library ExampleEqualizer Adjust the level of signal frequencies components*
Boost
Cut
CutoffFrequency
Gain
Frequency
Lowpass Shelving Filter Frequency Response (Magnitude)
1
CutoffFrequency
Gain
2
Highpass Shelving Filter Frequency Response (Magnitude)
Boost
Cut
Frequency
Σb0
b1
b2
-a1
-a2
z-1
z-1
z-1
X(z) H(z)X(z)
z-1
Gain
Peaking Filter Frequency Response (Magnitude)
CenterFrequency
Cut
Frequency
Boost
*From Parametric Equalization on the TMS320C6000 DSP (SPRA867) app note and code for C6711 DSK
OMAP Audio Algorithms
Introduction: The Wide World of Audio
Analog and Digital Audio
Algorithms for Audio Processing
OMAP Audio Algorithms
DSP Architectures for the Future of Audio
OMAP Audio Design
Class D Audio Amplifier
System Software (Play, Stop, FF, Rew, Repeat)EQ (5-band)Volume (Stereo)SRC(Stereo)AC3 Decode MP3 decodeMP3 encode AAC decodeAAC encodeWAV PlaybackWAV RecordADPCM PlaybackADPCM RecordATRAC-3 decodeACELP.NET decode
TwinVQDolby Digital 2 channel Class - CDolby Digital 5.1 channel Class -CReal G2 decodeWMA decode WMA w/ DRM decodeWMA encodeLucent ePACLiquid Audio SP3 (ver. 1.0) MPEG-1 Layer I EncoderMPEG-1 Layer I DecoderMPEG-1 Layer II EncoderMPEG-1 Layer II DecoderAC-3 2 ch Encoder (DDCE) Class AIntertrust723.1
Audio Algorithms for C54xx/C55xx DSPs
Audio Algorithms
AAC DecoderFeatures
MPEG-4 AAC Decoder with simple and Low complexity Profile.Decoder outputs pcm file format from 16-bit AAC Encoded samplesall sample rates (8 kHz to 48 kHz)all bit rates (8 kbit/s to 288 kbit/s)mono, stereo, dual-channel, joint stereo The given decoder supports both; ADIF (Audio Data Interchange Format) and ADTS (Audio Data Transport Stream) input formats.automatic bitstream synchronization and error recoveryaudible fast-forward/rewind modeTI eXpressDSP compliantC5500 large memory model object code library
PerformanceLow memory library
25.5 MHz (peak) and 23.5 MHz (average) for stereo 44.1 kHz at 128 kbit/s13.5 Kwords of SARAM (const and program) and 11 Kwords of DARAM including real-time I/O buffers and stack
Normal memory library (with more memory)20.5 MHz (peak) and 18.5 MHz (average) for stereo 44.1 kHz at 128 kbit/s21 kwords of SARAM ( const and program ) and 11 Kwords of DARAM including real-time I/O buffers and stack
AAC EncoderFeatures
The given Encoder implements AAC Low complexity (LC) profile of the standard up to 2 channels (stereo).The given encoder supports both; ADIF (Audio Data Interchange Format) and ADTS (Audio Data Transport Stream) input formats.It supports sampling frequency from 8 kHz to 96 kHz as specified by the standard.Encoder outputs AAC file format from 16- bit raw PCM samples.It supports the bit-rate up to maximum depending upon sampling frequency as specified by the standard (see 8.2.2 of ISO/IEC 13818-7). It supports all necessary tools and features, so that the given algorithm is standard compliant.
Performance73 MHz (peak) and 64 MHz (average) for stereo 44.1 kHz at 128 kbit/s27 kwords data RAM including real-time I/O buffers14 kwords data ROM23 kwords program ROM
MP3 DecoderFeatures
layer-3 only16-bit PCM outputall sample rates (8 kHz to 48 kHz)all bit rates (8 kbit/s to 320 kbit/s)variable bit rate (VBR) modemono, stereo, dual-channel, joint stereoCRC error protectionautomatic bitstream synchronization and error recoveryaudible fast-forward/rewind modeTI eXpressDSP compliantC5500 large memory model object code library
Performance20.97 MHz (peak) and 15.38 MHz (average) for stereo 44.1 kHz at 128 kbit/s11 kwords data RAM including real-time I/O buffers and stack8 kwords data ROM10 kwords program ROMTested against MPEG-1 and -2 compliance suites
MP3pro Decoder 2/2Features
Spectral Band Replication Technology using ancillary data field ( designed by -Coding Technologies.)Supports .mp3 filesSupports a maximum of up to two channelsAll mode (stereo/joint stereo/dual channel/single channel)Ancillary data extractionVariable bit-rate decoding
Performance66 MHz (peak) and 59 MHz (average) for stereo 44.1 kHz at 96 kbit/s25.5 Kwords of Program memorymp3 part of mp3PRO decoder = 10Kwords , PRO part of mp3PRO decoder = 15.5Kwords35 Kwords of Data memorymp3 part of mp3PRO decoder = 17Kwords,PRO part of mp3PRO decoder = 18Kwords
MP3 Encoder
Performance93 MHz (peak) and 61 MHz (average) for stereo 44.1 kHz at 128 kbit/s24.2 kwords data RAM including real-time I/O buffers8.3 kwords data ROM30 kwords program ROM
ADPCM Decoder
DescriptionThe ADPCM decoder supports all the bit rates and sampling frequencies, which are specified in ITU G726 Standard. I.e. Sample Rates 8, 11.025, 16000, 22.050,32 and 44.1 KHz and 2,3,4 and 5 bit encoding/decoding of Mono streams.
Performance3.78 MIPS (peak) and 3.76 MIPS (average) for mono 8 kHz20.72 MIPS (peak) and 20.5 MIPS (average) for mono 44.1 kHz977 words data RAM1069 words program ROM
ADPCM Encoder
DescriptionThe ADPCM encoder supports all the bit rates and sampling frequencies, which are specified in ITU G726 Standard i.e. Sample Rates 8,11.025,16000,22.050,32 and 44.1 KHz and at 2,3,4 and 5 bit encoding/decoding. Mono encoding and decoding is supported.
Performance4.71 MIPS (peak) and 4.68 MIPS (average) for mono 8 kHz25.137 MIPS (peak) and 24.86 MIPS (average) for mono 44.1 kHz990 words data RAM1750 words program ROM
WMA Decoder
DescriptionThe WMA C55x all bit rate decoder supports all the bit rates, which are specified in WMA decoder Standard. It supports version 2, version 7 as well as version 8 encoded bit streams.
Performance44.4 MHz (peak) and 17.5 MHz (average) for stereo 44.1 kHz at 128 kbit/s27 kwords data RAM11.5 kwords data ROM16.5 kwords program ROM
5-band EqualizerFeatures
N bandsGains adjustable from –15db to +15 db in 1db steps Distortion free flat spectrum at 0 dB gainNo unpleasant audible artifacts during gain changes16-bit mono PCM input and output audio dataCenter frequencies and bandwidths are configurable off-lineImplemented by 2nd order IIR sections in cascadeTI eXpressDSP xDAIS compliant
Performance7.64 MIPS (peak) and 7.58 MIPS (average) for 5 bands4.24 MIPS (peak) and 4.24 MIPS (average) for 3 bands208 words data RAM including real-time I/O buffers266 words data ROM1427 words program ROM
Speech RecognitionSolutions from Scansoft: VoCon ASR 1/2
Key Features
VoCon is optimized for small vocabularies and offers the following advanced features:Continuous speech inputVoCon can recognize natural speech, so there is no need to leave gaps between words.Speaker Dependent RecognitionUser creates own vocabulary via user word training.Speaker Independent RecognitionVocabulary consists of optimized acoustic models corresponding to pre-defined grammar and word list.Whole-word and Phoneme-based RecognitionEnsures high recognition accuracy and the design of specific words and acronyms.Keyword ActivationFilters and recognizes keywords out of natural spoken sentences.Out-of-Vocabulary RejectionRejects words that are not part of the vocabulary.Speaker AdaptationAdapts speaker-specific language characteristics to speaker-independent systems.Dialogue CapabilitiesSupport the design of the user interface via grammars and integration of word probabilities.Context-switching capabilitiesEnsures shortest turnaround time when switching vocabulary.
Speech Recognition
Solutions from Scansoft: VoCon ASR 2/2
Performance (implementation on C5402)about 80 MIPS100 kwords program ROM112 kwords data RAM
Scansoft ASR software VoCon, ASR3200
Speech RecognitionSolutions from Advanced Recognition Technologies: smART Speak XG
Key FeaturesName dialing - speaker independentIncorporates core modules (smARTspeak NG):Name dialing - speaker dependentContinuous digit dialing - speaker independent/ dependentMenu navigation and device control – speaker independent / dependentUnique adaptation to user’s voice for enhanced performance with accents, while maintaining an excellent out of the box experience
Performanceruns on ARM or DSPabout 8 MIPS46.7 kwords program ROM167 kwords constant data (continuous digit dialing, 20 fixed commands, 100names)30.5 kwords data RAM
Introduction: The Wide World of Audio
Analog and Digital Audio
Algorithms for Audio Processing
OMAP Audio Algorithms
DSP Architectures for the Future of Audio
OMAP Audio Design
Class D Audio Amplifier
DSP Architectures
DSP & Analog AV Solutions
ROOM #1 – 7.1 ANALOG AMPLIFICATION
PCM1738/1791/1792
OrPCM1608/DSD1608
Or PCM1606
A/V Receiver System Diagram
Tuner
2-6 channelsof A to D
Conversion
5.1 Analog In
OpticalDigital
Receiver
1-3 Optical Digital InDVD/DVP
Set op BoxCD
S/PDIF In (high Quality)DVD/DVPSet Top Box
Mul
tiple
xer
16-20 Bit 48 Khz
R L
OPA134
FILTER8 Ch DAC VOLUME
PGA2310 AM
PLIFIER
OPA134 PGA2310
1
8
ROOM #2 – 7.1 DIGITAL AMPLIFICATIONPOWER STAGE DIGITAL PWM
TAS5110
TAS5110
TAS5110
TAS5110
OR
REMOTE SPEAKER
PCM1804
-Or-
PCM1802
-Or-
PCM1800/2
S/P DIF Transceiver
DIX1700
PCM1738/1791/1792
PCM1608/DSD1608
PCM1606
HIGH
MID
LOW
TAS5015
TAS5012
TAS5010
Or
Or
OMAP5910
Encoder/DecoderDolby Digital (AC-3)
WMAMultichannel AAC
MP3Effects
Bass Management Treble Control
Dolby HeadphoneVirtual 3-D
DownmixingHall Effects
S/PDIF Out eg. Mini Disc
mjc11
Slide 50
mjc11 I added 'one big arrow from DA610/C6714 to Room #1 and Room#2marilyn connell, 4/30/2003
A/V Receiver System Diagram
150W x 5.1 Channels(0.09% THD @ 8 ohms BTL)
1 x $2.32 3 x $1.93 24 x $0.20
$12.91 ($2.15 per channel)
TAS50466-Channel
PWM Processor100 dB
TAS51922-Channel
Gate Driver
TAS51922-Channel
Gate Driver
TAS51922-Channel
Gate Driver
TAS31066-Channel
Digital AudioPost-Processor
AudioDecode
OMAP5910Processor
L
R
LS
RS
C
LFE
mjc11
Slide 51
mjc11 I added 'one big arrow from DA610/C6714 to Room #1 and Room#2marilyn connell, 4/30/2003
Typical requirement is 32-48 bit dynamic range with a 48-76 bit precision for accumulator
AudioProcessing
Speaker EqualizerRoom Equalizer
Typical requirement is 24-30 bit dynamic range with 48-64 bit precision for accumulator
Typical requirement is 16-24 bit dynamic range with 32-48 bit precision for accumulator
AudioSource
Audio HiFiPlayback
Typical A/D16~24-bit @ 11.25k~48kHz
Typical DAC24-bit @96kHz
Interface format to typical DAC’s are fixed point.
No Floating-point DAC!!
A/V Receiver Applications: The Whole Gamut
AudioEncode/Decode
Dolby Digital (AC-3)DTSMultichannel AACEtc…
AudioProcessing
Dolby Pro LogicDTS NEO:6GraphicEqualizer
MP3 Jukebox System Diagram
Mass StorageFlash
Hard DriveIDE CD
Stereo A/D, D/A+ Amplifier
TLV320AIC23
3V VoltageRegulator
Power Supply
TPS61010
1.5V VoltageRegulatorTPS62000
OMAP5910
STEREO 16Ω, S.E.
40mW
Connectivity
WiFi 802.11
USBConnector
LCD
EEPROM24WC256
Crystal12 MHz
Crystal32.768 KHz
Batteries
MonitorButtons
IIC
PLL
RTCGPIOParallel
Bus
USB
LCD
MMC/SD
McBSP
GPIO
MMCSD
Audio Application MP3 Jukebox: Reduced Gamut
HiFi AudioHome TheaterSuper Audio CDDVD AudioHDD serverAV receiverDVD Receiver
Outdoor, MobileHome
Automotive AudioCar Audio and Entertainment
Portable AudioPortable CD, MP3PDA AudioWireless Phone AudioPortable Digital Radio
General AudioMini Stereo systemBoombox
Hi Fidelity
Casual
16-bit words, 44.1 kHz is the recreation goal of the MP3 decoderTradeoff bit rate vs. storage space Typically in a noisier environment than the A/V Receiver (i.e. what SNR do you need when you are jogging?)
DSP Architectures for the Future of Audio
CDQuality
And… Often other concerns like cost, power consumption, portability etc. outweigh performance for a « leaner » system
2002
Dolby™• Dolby Digital™ (AC3)• ProLogic™• ProLogic™ II• Dolby Headphone™
Fraunhoffer™• MPEG AAC (LC)
DTS™• Consumer 5.1™• ES 6.1™• Neo 6.1™• DTS 96/24™
Func
tiona
lity
Code Composer Studio
DSP/BIOS
xDAIS Algorithm Standard
Third Parties
Fully Programmable
Fixed Function devices
Low pin count
GUI tools
DA610/C6713
C6712C5509
C5401C5402
TAS3103
TAS3002/3004
Floating Point C67xx Core
Fixed Point C54xx & C55xx Core
Fixed Function Digital Audio Processors
Digital Audio Devices
TAS3001
Speaker EqualizationEffects
MP3 (encode/decode)AAC (encode/decode)Windows Media Audio™ (decode)ATRAC3™ (decode)Security features
C6711
C5410
5910C5510C5416
mjc12
Slide 55
mjc12 Rita I added C5502 but it needs positioningmarilyn connell, 4/30/2003
Introduction: The Wide World of Audio
Analog and Digital Audio
Algorithms for Audio Processing
OMAP Audio Algorithms
DSP Architectures for the Future of Audio
OMAP Audio Design
Class D Audio Amplifier
OMAP Audio Design
R610K
U1TLV320AIC23PW
LOUT 12
ROUT13
LLINEIN19
RLINEIN20
MICIN18
MICBIAS17
DIN4
LRCIN5
BCLK3
XTL_MCLK25XTO
26
CS#21
MODE22
BVDD 1
BVDD 27
DG
ND
28
LHPOUT9
RHPOUT 10
HP
GN
D11
AG
ND
15
VMID16
HPVDD8
AVDD14
CLKOUT 2
LRCOUT7
DOUT6
SDIN23
SCLK24
R710K
C4000.1UF S0T-23
Q1DK/MMBT3904ICT-ND
VCC_I/O
R4220K
MCBSP1_BCLK
R32.2K
R2100
+ C2100UF_(D)
+ C510.0UF
+ C410.0UF
RLINE_IN
LLINE_IN
MCBSP1_SYNC
R130
R90
VCC_I/O MCBSP1_DOUT
I2C_DATA
MCBSP1_DIN
I2C_CLK
2
3
1
3
1
2
MK1DK/P9925-ND
12
OMAP Audio Design
Microphone Preamp-Need to bypass the biasvoltage-Use op amp instead of transistor-> much betternoise rejection-Need to terminate McBSPbus
L210uH
GND AUDIO
GND AUDIO
C?
100nFGND AUDIO
C135B
470nF
+C13910UF16V
C142100nF
GND AUDIO
C143100nF
C144100nF
+C14110UF16V
+C14010UF16V
GND AUDIO
R187F
10K
GND AUDIO
VCC_AUDIO
C135B 100pF
R187C 100K
L110uH
VCC_AUDIO
+
C127B
1UF16V
R159B 10K
+
-
U119A
OPA348
1
34
52
RT120K
RT220K
+CT110UF16V
ampbias
R187E
5.6K
R187D 5.6K
+
C141B
10UF16V
GND AUDIO
J7
MIC I/F
123
GND AUDIO
ampbias
U43
TLV320AIC23
SCLK 24
XTO26 XTI/MCLK25
CLKOUT2
DGND 28
LRCOUT7
DOUT6
DIN4
LRCIN5
BCLK3
BVDD1
LHPOUT9
HPGND 11
HPVDD8
RHPOUT10LOUT12 ROUT13
AVDD 14
AGND15
MICBIAS17
VMID 16
MICIN18
RLINEIN19
LLINEIN 20
DVDD 27
MODE22 CS/21
SDIN23
OMAP Audio Design
Low noise micPre-amp circuit
OMAP Audio DesignU1TLV320AIC23PW
LOUT12
ROUT13
LLINEIN19
RLINEIN20
MICIN18
MICBIAS17
DIN4
LRCIN5
BCLK3
XTL_MCLK25
XTO26
CS#21
MODE22
BVDD1
BVDD 27
DG
ND
28
LHPOUT9
RHPOUT 10
HP
GN
D1
1A
GN
D1
5
VMID16
HPVDD 8
AVDD14
CLKOUT2
LRCOUT7
DOUT6
SDIN23
SCLK24
L6
100NH D/K#PCD1201CT-ND
SG-710(CA)
Q200DK/MMBT3904ICT-ND
Q201DK/MMBT3904ICT-ND
U35DK/SG8002CAPCB-ND(16.9344MHZ)
OUT3
GND2
EN1
VCC4
R1210K
VCC_I/O
C9050.1UF
U36DK/SG8002CAPCB-ND(12.000MHZ)
OUT3
GND2
EN1
VCC4
TP20TP
1
U22SN74AHC1G04DCKR
24
53
1
C9060.1UF
CLK_SEL
MCLK, Oversampling Clock = OS x Frame Sync-> OS = 256, 250, 272 or 384 oversampleFrame Sync = 44.1KHz, 48KHz or 32KHzBCLK has to be sync’ed with MCLKMaster CODEC drives BCLK & Frame Sync
Audio Clock Design
OMAP Audio DesignU1TLV320AIC23PW
LOUT12
ROUT13
LLINEIN19
RLINEIN20
MICIN18
MICBIAS17
DIN4
LRCIN5
BCLK3
XTL_MCLK25
XTO26
CS#21
MODE22
BVDD1
BVDD27
DGND
28
LHPOUT9
RHPOUT10
HPGN
D11
AGND
15
VMID16
HPVDD8
AVDD14
CLKOUT2
LRCOUT7
DOUT6
SDIN23
SCLK24
L6
100NH D/K#PCD1201CT-ND
SG-710(CA)
PW
Q200DK/MMBT3904ICT-ND
Q201DK/MMBT3904ICT-ND
U35DK/SG8002CAPCB-ND(16.9344MHZ)
OUT3
GND2
EN1
VCC4
C9000.1UF
+C90110.0UF
MCBSP1_CLKS
R19833
R1210K
MCBSP1_BCLK VCC_I/O
LLINE_IN
RLINE_IN
VCC_3V3MCBSP1_SYNC
VCC_I/O
C9050.1UF
U36DK/SG8002CAPCB-ND(12.000MHZ)
OUT3
GND2
EN1
VCC4
U2TLV320AIC23PW
LOUT 12
ROUT13
LLINEIN19
RLINEIN20
MICIN18
MICBIAS17
DIN4
LRCIN5
BCLK3
XTL_MCLK25XTO
26
CS#21
MODE22
BVDD1
BVDD27
DGND
28
LHPOUT 9
RHPOUT10
HPGN
D11
AGND
15
VMID16
HPVDD8
AVDD14
CLKOUT2
LRCOUT7
DOUT6
SDIN23
SCLK24
PW
C9000.1UF
+C90110.0UF
MCBSP1_CLKS
R19833
VCC_I/O
LLINE_IN
RLINE_IN
VCC_3V3
TP20TP
1
MCBSP2_DOUT
I2C_DATA
MCBSP2_DIN
I2C_CLK
TP20TP
1R130
R90
DO NOT POPULATE
U22SN74AHC1G04DCKR
24
53
1
C9060.1UF
VCC_I/O MCBSP1_DOUT
I2C_DATA
MCBSP1_DIN
CLK_SEL
I2C_CLK
MULTIPLE CODECs DESIGN-Master device drives all FS and BCLKinputs of slave devices.-Only McBSP1 has SRG, generating different BCLK from one CLKS input.-Only one master is allowed for Sync CODECs
Introduction: The Wide World of Audio
Analog and Digital Audio
Algorithms for Audio Processing
OMAP Audio Algorithms
DSP Architectures for the Future of Audio
OMAP Audio Design
Class D Audio Amplifier
Class D Audio Amplifier
Power Amplification Issues
All equipment designers are striving for more compact, lighter, cheaper products with less heat generation and uncritical power require-ments, while preserving great sound quality
Switching amplifiers, both digital-input and analog-input, are the way to go for efficiency improvement and bulk reduction (smaller, or even no, heatsinks). Customers nervous about switching designs; TI has world-class experts to advise.
Small form factor is sexier - and cheaper to ship! Ultra low profile PCB design enables sub-50mm total product height
TI is pioneering Power Supply Compensation for digital-input amplifiers
Ease-of-use features of the latest analog-input switching amplifiers make them the solution of choice for sleek products like LCD television
The “Linear Danger Zone”
Pdiss (W)555045
353025201510
50
0.1 1 10Po each channel (W)
Power Dissipation (Sine Wave)
40Vs = 14.4 VRL = 4 x 4 Ohm
CLASS AB
Danger Zone!
TDAATDAA
Legacy Audio Amplifiers (Linear Class AB)
Efficiency: 40-60%• Bulky and heavy• Hot and noisy• Shorter battery life• Power hogs
Emerging Audio Amplifiers (TDAA Class D)
Efficiency: ~ 90%• Light and streamlined• Cool and quiet• Longer battery life • Energy efficient• Consumer friendly• Cosmetically pleasing
Class D Amplifier
Class D Amplifier
Higher Efficiency>90% Theoretical, / >85% Achievable
• Lower Current RequirementLonger Battery LifeLess Power Input for same Power Output
• Less Heat DissipationSmaller PackagesLess Heat Sinking
Smaller Power Supply Requirement• Less Weight
In General – “Greener”
Class D Amplifier
Superior Distortion CharacteristicsNo Slew-rate induced TIM (Transient Inter-Modulation Distortion)Very Low Feedback Required
Excellent Transient ResponseExcellent Open-loop Characteristics
Low Open-loop Output ImpedanceLess Back-EMF Interference
Psycho-acoustically Speaking“Tighter Bass Response”“Crisper/Cleaner High End”
TI Class D Amplifier
PCM to PWM
Processor
CD/DVD/DAB
/SACD/MP3
SwitchingOutput stage
24 bit32-96kHz
PCM DigitalPWMControl
V+
TAS50XXfamily
TAS51XXfamily
Digital Upsampler,Linearity Corrector
and Modulator
DigitalSource
TI Class D Amplifier
“Thunderbird”Power Amplifier
AC Power Strip TAS5100Digital Audio PWM Power Output Stage
TAS5100Digital Audio PWM Power Output Stage
TAS5010Digital Audio
PWM Processor
TAS3002 Digital Audio
Processor
Volume Control
Parallel Port Interface
DIR1703 S/PDIF
Receiver
I2S Interface
I2S Bus
I2S BusI2S Input
I2C BusAnalog Input
Analog Output
PC Software
32kHz - 192kHz
32kHz - 96kHz
Thunderbird Power Amplifier
Toslink CableCD
Player
24 bit44.1 kHz
PCM8 ohm Speaker
8 ohm Speaker
AC Power Strip 75 Watt Power Supply
TI Class D Amplifier EVM
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