19 12
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请教录音问题在

请教录音问题在

因用了近十年的电脑坏了!近期去配了一台,由于考虑不周,配置较低。声卡不是独立的,是主板带的集成声卡。现在录音时总感觉到,录音后葫芦丝的声音滞后比伴奏慢。请朋友们支支招,看要如何处理。在此先谢谢大家了!

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原帖由 南山一虫 于 2009-12-23 13:26 发表 因用了近十年的电脑坏了!近期去配了一台,由于考虑不周,配置较低。声卡不是独立的,是主板带的集成声卡。现在录音时总感觉到,录音后葫芦丝的声音滞后比伴奏慢。请朋友们支支招,看要如何处理。在此先谢谢大家了!
我的也是这毛病,换个AA试试。
为了健康    为了快乐    为了友谊    为了和谐    吹起迷人的葫芦丝     让我们永远永远在一起

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    The Multitrack tab in the Settings dialog box provides options that let you optimize performance during recording, playback, and mixdown.

    Playback Buffer Size

    Determines the buffer size (in seconds) used when sending data to your sound card when playing back a multitrack session. Different sound card drivers may require different memory buffer size settings. Adobe Audition's default settings should work fine for most sound cards. If you hear choppiness (skips or dropouts) in multitrack playback, adjust the buffer size. (Choppiness in multitrack playback can also be attributed to the background mixing process not being far enough ahead). A larger buffer size requires more computer memory. The default setting is 1.

    Playback Buffers

    Specifies the number of buffers Adobe Audition uses for playback in the multitrack environment. If you experience break-up in your audio, try reducing the number of buffers. Increasing this number might also be helpful for some configurations. The default setting is 10.

    Recording Buffer Size

    Reserves memory for recording in a multitrack session by entering a buffer size (in seconds). Different sound card drivers may require different memory buffer size settings. Adobe Audition's default settings should work fine for most sound cards. If you experience dropouts while recording in multitrack (especially when playback seems fine), try increasing this setting. (First, be sure the background mixing process is sufficiently complete when you go to record as this may cause the same symptom.) A larger buffer size requires more computer memory. The default setting is 2 seconds.

    Recording Buffers

    Specifies the number of buffers used for recording in the multitrack environment. If you experience break-ups in your audio, try reducing the number of buffers. Increasing this number may also help for some configurations. The default setting is 10.

    Background Mixing Priority

    Specifies the priority level of the background mixing process in a multitrack session. Lower values indicate a higher level of priority above other system events. You can use fractional numbers (such as 0.8). The default setting is 2.

    Open Order

    Determines the order in which Adobe Audition opens a sound card's playback (in) and record (out) ports for use in the multitrack environment. This order is relevant only for older sound cards that don't support full-duplex capability.

    Start Order

    Determines the order in which Adobe Audition starts a sound card's playback (in) and record (out) ports for use in the multitrack environment. This order is relevant only for older sound cards that don't support full-duplex capability.

    Correct For Drift In Recordings

    Synchronizes the master audio playback device (generally, the first Out device listed in the session--the one on Track 1) and the record device of the waveform being recorded. If the true sample rates on the cards differ enough that the recording would have drifted out of sync with the original if both were played back at exactly the same sample rate, then the recording is corrected by resampling to make it the proper length. This option only works with new record tracks, not with recording on top of existing waveforms, or punch-ins.

    Note: On sound cards that support sample accurate devices (that is, synchronized device starting, and all devices keyed off of the same clock) you don't need to select this option. This option allows for some measure of near sample-accurate synchronization across different sound cards, or when using with a single sound card that doesn't use the same clock for playback and recording (which is common in consumer and other low-end sound cards).

    Correct For Start Sync In Recordings

    Compares the exact true time that the record device started with the time the master playback device started. If different, the recorded block's position is adjusted so the recording starts in perfect sync with the playback. This option only works with new record tracks, not with recording on top of existing waveforms, or punch-ins.

    If this option is enabled, and you do a loopback test (by connecting the audio Out to the audio In and recording some ticks) and each recording is still a fixed amount out of sync, then you can adjust for this by entering this amount (in milliseconds) in the Latency field of Options > Device Properties for the recording device being used. To compute milliseconds, look at the difference in samples, multiply by 1000, and then divide by the sample rate. For instance, if the recording consistently appears 27 samples ahead of the playback, the latency would be 27 x 1000 / 44,100, or about 0.61 milliseconds. (The reason for the milliseconds format and not samples is because at various sample rates this latency will be different in terms of samples, but will be the same in terms of milliseconds.)

    Note: On sound cards that support sample accurate devices (that is, synchronized device starting, and all devices keyed off of the same clock) you don't need to select this option. This option allows for some measure of near sample-accurate synchronization across different sound cards or a situation where a single sound card uses different clocks for playback and recording. (This situation is common for consumer and other low-end sound cards.)

    Delete Old Takes After Merging

    Automatically deletes any unused takes created during a punch-in when you select a take. If you don't select this option, unused takes remain available to the Session (in the Insert menu) and occupy hard drive space.

    Crossfade Time

    Determines the amount of time (in milliseconds) over which crossfading occurs when a take created using punch-in is merged back into the surrounding waveform.

    Mixdowns

    Determines the bit-resolution that is used when performing a mixdown. Regardless of the session format (16-bit or 32-bit), you can generate mixdowns at either 16-bit or 32-bit quality with this option. The default is 16-bit. Click Dithering Option to specify how to dither the 16-bit mixdown.

    Track Record

    Specifies how waveforms are created when recording directly into the Multitrack View: as mono or stereo, and as 16-bit or 32-bit.

    Pre-Mixing

    Determines the bit size used for the background mixing process. Best quality is achieved by leaving this at the default 32-bit setting. However, if you're using multiple sound cards, it may be advantageous and faster to choose 16-bit for pre-mixing, as less data will be transferred across the hard drives. For single output device situations, or faster hard drives, 32-bit is better as it provides optimization at mixdown.

    Panning Mode

    Specifies the method used for panning waveforms in a multitrack session.

    • L/R Cut (log): Pans left by reducing the volume of the right channel, and pans right by reducing the left channel volume. The channel being panned to doesn't increase in volume as panning gets closer to 100%.
    • Equal-power Sine: Pans left and right channels with equal power, so a hard pan left will contain the same loudness as both channels together. This results in an increase of 3 dB RMS on the channel being panned to when at 100%.

    Note: Because panning can actually make one channel louder than the original waveform, audible clipping can occur in 16-bit sessions. To avoid this, work in the 32-bit realm if you're using the Equal-power Sine panning method.

    Auto Zero-Cross Edits

    Automatically adjusts the beginning and end points of all Cut, Copy, and Paste-type edits to the nearest place where the waveform crosses the center line (zero amplitude point).

    If the amplitudes aren't lined up on both sides of the selection, the endpoints are at different amplitudes. This often results in an audible pop or click at that point.

    Smooth Auto-Scrolling During Playback

    Enables smooth scrolling when playing back audio in Multitrack View. By default Adobe Audition uses a paging method of scrolling in Multitrack View instead of the smooth scrolling technique used in Edit View. This saves on system resources.

    Save Locked Track Files After Closing Sessions

    Saves the temporary files associated with locked tracks. When you reopen the session, Adobe Audition uses the temporary file instead of mixing down the locked tracks.

附件

Snap3.jpg (47.89 KB)

2009-12-23 17:44

Snap3.jpg

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回复 2# 竹林古筝 的帖子

我现在用的就是AA。

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回复 3# reghlsbbs 的帖子

谢谢reghlsbbs 朋友!可我还是一头雾水搞不明白。不知在电脑的哪里还是AA录音软件的什么地方能找到朋友所提示的设置对话框。再就是英文提示我是一抹黑。

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这个是Adobe Audition(即AA)1.5版本的截图,她的绿色汉化版在本论坛有下载。 要想找到上面的界面很简单,启动AA后在菜单栏依次选:选项-设置-多轨,然后调整录音缓冲区的两个值(即Recording Buffer Size 和Recording Buffers)边录边调整直到满意为止。 另外,以我的经验目前再烂的机子跑AA1.5也不成问题,哪怕是512M物理内存

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引用:
原帖由 reghlsbbs 于 2009-12-24 09:06 发表 这个是Adobe Audition(即AA)1.5版本的截图,她的绿色汉化版在本论坛有下载。要想找到上面的界面很简单,启动AA后在菜单栏依次选:选项-设置-多轨,然后调整录音缓冲区的两个值(即Recording Buffer Size 和Recor ...

谢谢reghlsbbs朋友的再次指教!我现在用的是AA3.0版本。可是在菜单栏里怎么也找不到朋友所指的界面。

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最后一次奉劝楼主机子不好别用AA3.0,那家伙耗资源比1.5大多啦

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记住:适合于自己的才是最好!

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AA3.0耗资源不说,就你的声卡而言为解决延时问题还必须装ASIO4ALL软件,有你烦的啦

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