ADSP21065例程代码 音频处理及其他 ADSP2XXX源代码 在VDSP++下应用文件列表: adsp21065l audio demo .....................\21klab.txt .....................\21klab.zip .....................\ad1819a .....................\.......\ad1819a_65l_ezlab_audio_drivers.zip .....................\.......\an_ad1819a_21065l.pdf .....................\.......\readme.doc .....................\.......\ws_ftp.log .....................\chorus.zip .....................\delays.zip .....................\detune.zip .....................\dynamic_range_controls .....................\......................\Compander .....................\......................\.........\AD1819a_initialization.asm .....................\......................\.........\AD1819a_initialization.BAK .....................\......................\.........\Clear_SPT1_regs.asm .....................\......................\.........\Codec_Processing_ISR.asm .....................\......................\.........\Compander.asm .....................\......................\.........\Compander.dpj .....................\......................\.........\COMPANDER.DXE .....................\......................\.........\compander_memory.map .....................\......................\.........\def21065l.h .....................\......................\.........\EZLAB_21065L_debugger.ldf .....................\......................\.........\Init_065L_EZLAB.asm .....................\......................\.........\ISR_table.asm .....................\......................\.........\new65Ldefs.h .....................\......................\.........\SDRAM_Init.asm .....................\......................\Compressor .....................\......................\..........\AD1819a_initialization.asm .....................\......................\..........\AD1819a_initialization.BAK .....................\......................\..........\Clear_SPT1_regs.asm .....................\......................\..........\Codec_Processing_ISR.asm .....................\......................\..........\Compressor.asm .....................\......................\..........\Compressor.dpj .....................\......................\..........\COMPRESSOR.DXE .....................\......................\..........\compressor_memory.map .....................\......................\..........\def21065l.h .....................\......................\..........\EZLAB_21065L_debugger.ldf .....................\......................\..........\Init_065L_EZLAB.asm .....................\......................\..........\ISR_table.asm .....................\......................\..........\new65Ldefs.h .....................\......................\..........\SDRAM_Init.asm .....................\......................\Expander .....................\......................\........\AD1819a_initialization.asm .....................\......................\........\AD1819a_initialization.BAK .....................\......................\........\Clear_SPT1_regs.asm .....................\......................\........\Codec_Processing_ISR.asm .....................\......................\........\def21065l.h .....................\......................\........\Expander.asm .....................\......................\........\Expander.dpj .....................\......................\........\EXPANDER.DXE .....................\......................\........\expander.map .....................\......................\........\expander_memory.map .....................\......................\........\EZLAB_21065L_debugger.ldf .....................\......................\........\Init_065L_EZLAB.asm .....................\......................\........\ISR_table.asm .....................\......................\........\new65Ldefs.h .....................\......................\........\SDRAM_Init.asm .....................\......................\Noise Gate - Downward Expander .....................\......................\..............................\AD1819a_initialization.asm .....................\......................\..............................\AD1819a_initialization.BAK .....................\......................\..............................\Clear_SPT1_regs.asm .....................\......................\..............................\Codec_Processing_ISR.asm .....................\......................\..............................\compressor.map .....................\......................\..............................\def21065l.h .....................\......................\..............................\EZLAB_21065L_debugger.ldf .....................\......................\..............................\Init_065L_EZLAB.asm .....................\......................\..............................\ISR_table.asm .....................\......................\..............................\new65Ldefs.h .....................\......................\..............................\Noise Gate.dpj .....................\......................\..............................\NOISE GATE.DXE .....................\......................\..............................\noisegate_memory.map .....................\......................\..............................\Noise_Gate.asm .....................\......................\..............................\SDRAM_Init.asm .....................\dynamic_range_controls.zip .....................\filters.zip .....................\flanger.zip .....................\guitar_distortion.zip .....................\phaser.zip .....................\pitch_shifter.zip .....................\readme.doc .....................\reverbs.zip .....................\rotating_speaker .....................\................\AD1819a_initialization.asm ... ...Brief Description of all available audio demosChorus Effects: Chorus Effect with FeedbackChorusing effect simulating 2 voices/musical instruments (SPORT1 Rx ISR count update method - delay calculation determined by a counter incremented in the serial port's audio processing routine). Stereo Chorus with Feedback Stereo chorusing effect simulating 3 voices/musical instruments (SPORT1 Rx ISR count update method - delay calculation determined by a counter incremented in the serial port's audio processing routine).DeTune Effects: Two Voice - ISR Update Method Pitch Shift Effect Simulating 2 Voices Slightly Out Of Pitch (ISR Update Method - delay calculation determined by a SPORT0 rx ISR counter). Two Voice -Timer Update Method Pitch Shift Effect Simulating 2 Voices Slightly Out Of Pitch (Timer0 Update Method - delay calculation determined by the on-chip programmable timer).Digital Delay Effects: ADT Slapback Slapback ('Doubling') Effect (also called Mono Automatic Double Tracking). The effect is created by adding a delayed signal together with the original, where a single input source is mixed with the delay, and the result is sent to both output channels. Automatic Double Tracking Stereo Automatic Double Tracking (Stereo Doubling) - "Concert Announcer Simulation." This effect is set up to playback the original "dry" signal in one stereo channel and the delayed signal in the other channel. This creates the impression of a stereo effect using a single mono source. Digital Delay - Single Reflection A Single digital delay consists of delaying an input signal using a delay-line, andadding the delayed signal back to the original. An FIR comb filter results. Multiple DelayMultiple digital delays can be created by delaying an input signal using multiple delay-lines, and adding the delayed signals back to the original. Multi-Tap Delay Multiple delayed values of an input signal can be combined easily to produce multiple reflections of the input. This can be done by having multiple taps pointing to different previous inputs stored into the delay line, or by having separate memory buffers at different sizes where input samples are stored. Reverb DelayA reverberating delay consists of adding delayed replicas of an input together with the direct input signal. In this implementation, the direct signal is combined with a gain controlled IIR comb filter. Slapback Delay Slapback ('Doubling') Effect (also called Mono Automatic Double Tracking). The effect is created by adding a delayed signal together with the original, where a single input source is mixed with the delay, and the result is sent to both output channels. Stereo Delay Stereo delay effects can be implemented by coupling the left and right channels. The left and right channels are coupled by introducing cross-feedback coefficients, so that the delayed output of one channel is fed into the input of the other channel.Digital Reverberation Effects Griesinger's Plate ReverbDigital Plate-Class Reverberation Audio Effect which works well for single instruments such as a guitar, keyboard, violin, etc. Griesinger's Plate Reverb w delay modulationTo improve reverberation responses for drums or other impulse signals, linear interpolation is used to slowly modulate the nominal tap point for two of the tank diffusers. Works well on vocals and drums, but not recommended for guitars/keyboards because of perceived vibrato from delay modulation. Moorer's Reverb This effect uses a low pass comb filter for each reverb stage to enlarge the density of the response. The technique involves 6 parallel comb filters with a low pass output,summing their outputs and then sending the summed result to an allpass filter before producing the final result. ReverbBuilding Blocks ToolKit All Pass C Canonical IIR comb filters tend to magnify input signal frequencies near comb filter peak frequencies. Allpass filters can be used to prevent this 'coloration' of the input since it has a relatively flat magnitude response for all frequencies. All Pass C Transpose Similar to the Canonical All pass filter except the filter structure is slightly different. IIR Comb Filter Comb filters are used for noise reduction of periodic signals, signal enhancement, averaging signals, and are inherent in digital audio effects such as delays, chorus and flange effects. Comb filters work by adding a signal with a delayed and scaledversion of the same signal. LowPass IIR Comb Filter The use of a lowpass comb filter in reverb algorithms causes each echo to spread out, or smear. This results in a much more diffused, mellowed response, and also more closely replicates the fact that higher frequencies diminish faster than lower frequencies in reverberant spaces. Stereo Allpass Filter Same as the previously described all pass filter, except this one works in stereo. Stereo IIR Comb Filter Same as the previously described IIR comb filter, except this one works in stereo. Schroeder's Reverb This effect uses four IIR comb filters in parallel, with their output results added together, followed by 2 cascaded allpass filters stages. Also, Schroeder's Reverb keeps the high reflections ringing longer since plain IIR comb filters are used.Dynamic Range Control Effects CompanderCombination of a compressor and expander. The peaks signal levels above the upper threshold are compressed, while lower level signals above the lower threshold are expanded. Compressor Keeps the level of the signal within a specific dynamic range, using a technique called 'gain reduction'. The objective of a compressor is to make sure that the signal level does not go beyond a desired level, thus restricting the dynamic range. Expander Opposite of a compressor, with opposite input/output ratios. This effect is often used to recover the full dynamic range of a compresses signal. Also useful for reducing noise in tape recording and wireless systems by increasing the level of the audio signal relative to low-level noise beyond a low threshold. Noise Gate - Downward Expander Reduces the amount of gain below a certain threshold to reduce or eliminate noise produced when no audio signal is present, while still allowing the signal to pass through. This is useful after processing multiple audio effects that can introduce noise above the noise floor of the AD1819a DACs.Flanger Flanger with Feedback Flanging consists of modulating a delayed replica of an input by a few milliseconds and adding this delayed signal together with the input, which will then cause phase shifting and spreading of the audio signal as a result of comb filtering. The delay is modulated using a low frequency sinusoid. The effect works best on drums, guitars, keyboards, and some vocals. Stereo Flanger Same as previously described Flanger, but works in stereo.Guitar Distortion SimulationThis routine will square the input and cause it to clip at positive and negative full scale to simulate the effect of saturating an op amp or overdriving audio amplifier speakers (The Hendrix Effect).Phaser Effect This effect is produced by passing an audio signal through a narrow notch filter and combining a proportion of the filter's output with the direct sound. The frequency of the notch is then varied using a low-frequency oscillator.Pitch Shifter 1 Single Pitch Shift An interesting and commonly used effect is changing the pitch of an instrument or voice. The pitch shifter requires using a sawtooth/ramp wavetable to achieve a 'linear' process of dropping and adding samples in playback from an input buffer. 2 Part Harmony w CrossFade This version crossfades between two buffers to prevent 'clicks pops' of input and output pointer wraparounds. A slight audible modulation occurs at higher pitch shift values.Rotating Speaker EmulationThis effect consists of a combination of the tremolo and vibrato effects, combined a low-pass filter with a variable cutoff frequency. The resulting sound heard by the ears is a cyclical increase/decrease in volume, a doppler effect from the speaker rotating, and a cyclical drop in high frequencies whenever the speaker is facing away from the listener.Tremolo ISR Count Method Tremolo consists of panning the output result between the left and right output stereo channels at a slow periodic rate. This is achieved multiplying the amplitude of the incoming signal with a low frequency sinusoid. The stereo example pans the output to the left speaker for +sine values and pans the output to the right speaker for -sine values.Vibrato Vibrato ISR_Count_Method This effect is achieved by evenly modulating the pitch of the signal. It is similar to a guitarist moving the 'whammy' bar, or a violinist creating vibrato with cyclical movement of the playing hand.WaveMusic DemoThis routine generates a song using tones generated from a sine function approximation routine.Digital Filters FIR Filter - HighPass LowPass Standard Finite Impulse Response Filter.Directions for loading, building, and using an audio effect on the ADSP-21065L.1.) Plug the power supply into the board and attach the serial cable to the board and your computer.2.) Open the VisualDSP Environment shortcut in the start menu.3.) Inside the VisualDSP Environment, open the ..\Audio Demos subdirectory.4.) Choose which audio effect you wish to use.(ie: Rotating Speaker Emulation)5.) Go to the subdirectory of the effect you want to use.6.) Open the project file (.dpj) for the effect you want to use.7.) Go to Project, Rebuild All, in order to rebuild the effect for use.8.) When the build has completed, go to Project, Debug, in order to start the debugger.9.) Hit the reset button when the dialog box requests a reset of the board, then hit OK.10.) You should now get a dialog box that says “Communication Success”, hit OK.11.) You should now get a dialog box says “Target Halted at Breakpoint”, hit OK.12.) Go to Debug, Run to start the program on the board.13.) Start the input (ie: cd player, microphone, etc.)14.) You should be hearing output at this point.TroubleshootingIf you don’t hear any sound coming out of the output:àYou must configure the system for either Line in or Mic in input.Configuring the system for either a Line in or Mic in input: Hardware Settings: Both jumpers, jp1 and jp2, need to be on either the Line in or the Mic in setting on the board. Software Settings: Open up the project file(.dpj) inside the Visual DSP Environment for whichever effect you want to use (ie: rotating speaker emulation). Open the file called AD1819a_initialization.asm. If you are using a microphone input set activate the line:#define MIC_INPUT 0and comment out the line:#define LINE_INPUT 1If you are using a Line in input, then activate the line:#define LINE_INPUT 1and comment out the line:#define MIC_INPUT 0Recompile the code through the Visual DSP Environment using the steps shown above.……