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选择了ASIO,怎么在SAW中实时监听还是有延时?

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2355
#1 08-4-29 00:00

选择了ASIO,怎么在SAW中实时监听还是有延时?

打开SAW录音 实时监听还是有延迟,选择的是ASIO,Buffer已调到64还是可以听出有延迟,请问怎样解决?

4632
#2 08-4-29 00:15
会不会是你声卡的本身还存在延迟呢?曾经有一个朋友用APOGEE的声卡调节到BUFFER 64都存在延迟~~

2355
#3 08-4-29 00:37
不存在啊,我一直用nuendo都是正常的,而且刚刚还测试了audition都正常,声卡是FF800,平常用Nuendo录音,Buffer调到512都感觉不到这种延迟,是不是软件还需怎样设置呢?

13860
#4 08-4-29 01:34
SAW中也要调Buffer,你调到512、256甚至128,就不延时了。

2355
#5 08-4-29 01:53
谢谢!好像SAW的buffer是跟着声卡一起动的,开始没有刷新 现在用128sample似乎不明显了


再请教另外两个问题:
1、 out preload buffers这个选项是表示什么意思呢?

2、录音的时候,在Input Source里上选择了话筒输入的通道,回放却要在Input Source里选择Multitrk,不然的话一直在监听,有没有其他的窍门可以不用这么复杂吗?

4632
#6 08-4-29 09:09
输入输出的BUFFER,我好像听小马说过这个问题

13669
#7 08-4-29 09:51
In PreLoad Buffers
This setting is available when the Audio Device Setup window is toggled to the In Device selections.
This setting controls how many blank buffers are preloaded to the audio device before recording
begins. This gives the system some breathing room while other events are stealing CPU time, to keep
the recording from glitching. If this setting is too low for your system, audio stuttering or glitching
during recording may occur. This setting combines with the In Buffer Size setting to determine the
final latency.
Use the In PreLoad Buffers listbox to choose a setting between 1 and 10. Higher values decrease the
chance of glitched recordings.

這設置是在錄音時控制多少個空白緩衝作為預備,  讓其他事件在使用CPU同時, 得以有喘息時間, 避免錄音時因急趕而產生
雜音, 它配合 BUFFER SIZE來決定最終的延遲時間, 參數由1-10, 數字越高可減場產生雜音的機會

In Buffer Size
This setting is available when the Audio Device Setup window is toggled to the In Device selections.
Use the In Buffer Size listbox to choose from eight different settings ranging from 64 to 4096 samples.
This setting controls the response time of the recording meters. Lower values create a more real-time
display. If using values equal to or below 1024, a suggested setting for the In PreLoad Buffer value
would be between 6 and 10. The higher buffer values do not affect the visible response of the meter.
F If the buffer size is set too low, audio static and/or glitching might occur, especially during heavy
CPU loads such as SRP Recording. A good starting combination might 8 buffers at 1024 size.
When you are finished with all adjustments, click OK, then save your default settings by clicking on
the File menu and selecting Preferences File | Save Default.

設置分別有八-九個選項, (32) 64-4096 採樣, 它控制錄音時的電平反應, 參數越低越實時顯示, 如用 1024採樣, 建議PRELOAD BUFFER在 6-10之間, 這個PRELOAD BUFFER數字不影響電平反應時間, 但正如上面所說, PRELOAD BUFFER數字越少, 可能會引起錯音時有雜音

Out PreLoad Buffers
This setting controls how many buffers of data are preloaded to the audio device before playback
begins. This affects the response time of live controls such as mutes, solos, faders, etc. If this setting
is too low for your system, audio stuttering or glitching during playback may occur, especially when
minimizing or sizing windows, etc. on a complex session. This setting combines with the Out Buffer
Size setting to determine the final latency.
Use the Out PreLoad Buffers listbox to choose a setting between 1 and 10. Lower values will give you
faster response times.

這設置是在播放時控制多少個空白緩衝作為預備,  它影響即時操作反應, 如 SOLO、MUTE、推子 等...如設置參數太低, 可能會引至播放不流蜴或雜音, 尤其在進行比較複雜的工程縮小SAW的窗口時, 它配合 OIUT BUFFER SIZE來決定最終延遲時間。 PRELOAD BUFFER可遺參數為 1-10, 數字越小, 反應越快

Out Buffer Size
Use the Out Buffer Size listbox to choose from eight different settings ranging from 64 to 4096
samples.
The combination of the Out Buffer Size and the Out PreLoad Buffer value control the latency of the
real-time engine. Lower latency values give live, instantaneous response and the feel of a real
physical console when adjusting controls and switches. Try different combinations to obtain the
lowest comfortable working range while still not overloading your CPU. Good starting combinations
are 4 to 6 buffers at 1024 size. If your system can handle it, 3 or 4 buffers at 512 size feels just like
physical hardware.
F Some combinations that result in the same low latency value will cause the system to glitch while
other combinations will perform comfortably. For instance, 2 buffers at 1024 size might cause
static or glitching on certain audio devices, but the same latency value can be obtained by using
4 buffers at 512 size, and the audio device may play back perfectly.

設置參數由 (32) 64-4096採樣, OUT BUFFER 和 OUT BUFFER PRELOAD配合得以控制 實時引擎的延遲值, 低延遲讓操作時有如真實使用硬件的感覺, 嘗試不同參數以取得不同系統中最理想的設置, 建議以 4或 6 BUFFERS X 1024 作為開始, 當 3 或 4 X 512 , 操作已跟硬件無異。 2 X 1024 和 4 X 512 基本是一樣的值, 但在不同系統下有不一樣的表現, 故此最好是動手試試。

[ 本帖最后由 himhui 于 08-4-29 11:11 编辑 ]

2355
#8 08-4-29 13:04
谢谢Himhui,非常详细!
这个问题还没有解答,继续求助:
录音的时候,在Input Source里上选择了话筒输入的通道,回放却要在Input Source里选择Multitrk,不然的话一直在监听,有没有其他的窍门可以不用这么复杂吗

8027
#9 08-4-29 13:34
录音时,不是必须要在Input Source窗口中设置成外部端口,直接设置成MT,然后在录音面板中输入选择mono/stereo device即可。

13669
#10 08-4-29 13:37
看看

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2355
#11 08-4-30 11:31
呵呵 谢谢两位 回家试试
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