• 參數(shù)資料
    型號: TLV320AIC34_1
    廠商: Texas Instruments, Inc.
    元件分類: Codec
    英文描述: FOUR-CHANNEL, LOW-POWER AUDIO CODEC FOR PORTABLE AUDIO/TELEPHONY
    中文描述: 四通道,低功耗音頻編解碼器的便攜式音頻/電話
    文件頁數(shù): 39/94頁
    文件大?。?/td> 905K
    代理商: TLV320AIC34_1
    www.ti.com
    B0155-01
    LB1
    RB2
    Atten
    LB2
    L
    +
    +
    +
    +
    +
    +
    +
    R
    To Left Channel
    To Right Channel
    DIGITAL INTERPOLATION FILTER
    The digital interpolation filter upsamples the output of the digital audio processing block by the required
    oversampling ratio before data is provided to the digital delta-sigma modulator and analog reconstruction filter
    stages. The filter provides a linear phase output with a group delay of 21/f
    S
    . In addition, programmable digital
    interpolation filtering is included to provide enhanced image filtering and reduce signal images caused by the
    upsampling process that are below 20 kHz. For example, upsampling an 8-kHz signal produces signal images at
    multiples of 8-kHz (i.e., 8 kHz, 16 kHz, 24 kHz, etc.). The images at 8 kHz and 16 kHz are below 20 kHz and still
    audible to the listener; therefore, they must be filtered heavily to maintain a good quality output. The interpolation
    filter is designed to maintain at least 65-dB rejection of images that land below 7.455 f
    S
    . To use the
    programmable interpolation capability, f
    S(ref)
    should be programmed to a higher rate (restricted to be in the range
    of 39 kHz to 53 kHz when the PLL is in use), and the actual f
    S
    is set using the NDAC divider. For example, if
    f
    S
    = 8 kHz is required, then f
    S(ref)
    can be set to 48 kHz, and the DAC f
    S
    set to f
    S(ref)
    /6. This ensures that all images
    of the 8-kHz data are sufficiently attenuated well beyond a 20-kHz audible frequency range.
    TLV320AIC34
    SLAS538A–OCTOBER 2007–REVISED NOVEMBER 2007
    Table 5. Default Digital Effects Processing Filter Coefficients,
    When in Independent Channel Processing Configuration
    Coefficients
    N1 = N4
    –27,034
    N0 = N3
    27,619
    D1 = D4
    32,131
    D2 = D5
    –31,506
    N2 = N5
    26,461
    The digital processing also includes capability to implement 3-D processing algorithms by providing means to
    process the mono mix of the stereo input, and then combine this with the individual channel signals for stereo
    output playback. The architecture of this processing mode, and the programmable filters available for use in the
    system, are shown in
    Figure 29
    . Note that the programmable attenuation block provides a method of adjusting
    the level of 3-D effect introduced into the final stereo output. This, combined with the fully programmable biquad
    filters in the system, enables the user to optimize fully the audio effects for a particular system and provide
    extensive differentiation from other systems using the same device.
    Figure 29. Architecture of the Digital Audio Processing When 3-D Effects are Enabled
    It is recommended that the digital effects filters should be disabled while the filter coefficients are being modified.
    While new coefficients are being written to the device over the control port, it is possible that a filter using
    partially updated coefficients may actually implement an unstable system and lead to oscillation or objectionable
    audio output. By disabling the filters, changing the coefficients, and then re-enabling the filters, these types of
    effects can be entirely avoided.
    Copyright 2007, Texas Instruments Incorporated
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    TLV320AIC34
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