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• Using Sigmatel STAC 92XX on Windows 7 RTM x64, cannot get 5.1 surround sound to work

  - by Roy Rico

  I have a Dell XPS 420. I've installed Windows 7 RTM (x64) I have this audio device: SIGMATEL STAC 92XX C-Major HD Audio I have tried using the windows 7 basic driver, and also the Vista 64-bit driver from Dell's website (details): Date: 10/29/2007 Version: 6.10.0.5511, A04 File Size: 7 MB With both drivers, I get normal stereo sound from the driver, but my 5.1 surround sound doesn't work. Has anyone experienced this? Is there a fix?

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• Burn 24/96 flac files to play on standalone player

  - by takeshin

  I have vinyl record rip in 24/96 flac format. Each track is almost 200 MB big, so the album won't fit on CD. How to burn these files on a DVD to play with the same quality on standalone DVD player? My player supports SACD, DVD Audio and DVD video as well. My OS is Ubuntu Lucid (preferred), but I have also WinXp with Nero installed. BTW, is there any difference between DVD+ and DVD- for audio?

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• Replace sound in another YouTube video

  - by Tom

  I have received permission from someone to translate the audio in their movies. The problem I am facing is that the video quality is quite poor and the author does not have the original videos any more. How can I replace the audio in the YouTube videos without further degrading the quality of the videos? Thanks, Tom

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• Any plugins for Skype that support "Soundboard" usage?

  - by Axxmasterr

  I would like to find a program or plugin for Skype that allows you to pipe sound samples in to the outgoing audio stream when you are on a call. Ideally it would have some sort of soundboard functionality so that I could have a group of audio samples at the touch of a button. I'd also prefer something that supports mp3 but wav support will also do.

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• Debian sound on hdmi instead of jack

  - by Hans de Jong

  I installed debian (gnome) and i can't get my sound working. When i use inxi -A i get the following result: Audio: Card-1: Advanced Micro Devices [AMD] nee ATI Cayman/Antilles HDMI Audio [Radeon HD 6900 Series] driver: snd_hda_intel Card-2: Advanced Micro Devices [AMD] nee ATI SBx00 Azalia (Intel HDA) driver: snd_hda_intel Sound: Advanced Linux Sound Architecture ver: 1.0.24 My feeling tells me my sound output is on the HDMI instead of my jackplug on my motherboard. How can i change this to my motherboard sound output?

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• Which connector do I need for a "line level" subwoofer?

  - by Ben Brocka

  I've got a separate pair of speakers and I'm looking at adding a subwoofer (this, specifically). I noticed on the detail page it's inputs are listed as such: Inputs: Speaker level, line level If I'm not mistaken "line level" are the standard 3.5 audio jacks on your motherboard/sound card, right? My motherboard has the standard 6 ports for sound, if I get a subwoofer like this can I simply plug the input into the orange 3.5 jack? My audio software supports up to 7.1 so software-wise, 2.1 wouldn't be a problem.

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• Using Sigmatel STAC 92XX on Windows 7 RTM x64, cannot get 5.1 surround sound to work

  - by Roy Rico

  I have a Dell XPS 420. I've installed Windows 7 RTM (x64) I have this audio device: SIGMATEL STAC 92XX C-Major HD Audio I have tried using the windows 7 basic driver, and also the Vista 64-bit driver from Dell's website (details): Date: 10/29/2007 Version: 6.10.0.5511, A04 File Size: 7 MB With both drivers, I get normal stereo sound from the driver, but my 5.1 surround sound doesn't work. Has anyone experienced this? Is there a fix?

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• Changing default playback device on Windows 8

  - by emartel

  Previously, on Vista and Windows 7, changing the Default Playback device would occur instantly. For example, audio is coming out of my speakers, I right click the Volume Control, click Playback Devices then I select another device and click Set Default. Audio would be transferred immediately. Unfortunately, now, with Windows 8, I need to kill whatever process what outputting sound, and restart it for the change to take effect. Is there something that can be done about it so that changes are taken into account immediately?

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• PulseAudio on Cygwin: Failed to create secure directory: Unknown error 13

  - by Nithin

  I am unable to run PulseAudio on Cygwin. Operating System: Windows 8 Pro 64 bit Cygwin Setup.exe Version: 2.831 (64 bit) PulseAudio Version: 2.1-1 When I run: pulseaudio -vv this is the output: D: [(null)] core-util.c: setpriority() worked. I: [(null)] core-util.c: Successfully gained nice level -11. I: [(null)] main.c: This is PulseAudio 2.1 D: [(null)] main.c: Compilation host: x86_64-unknown-cygwin D: [(null)] main.c: Compilation CFLAGS: -ggdb -O2 -pipe -fdebug-prefix-map=/usr/src/ports/pulseaudio/pulseaudio-2.1-1/build=/usr/src/debug/pulseaudio-2.1-1 -fdebug-prefix-map=/usr/src/ports/pulseaudio/pulseaudio-2.1-1/src/pulseaudio-2.1=/usr/src/debug/pulseaudio-2.1-1 -Wall -W -Wextra -Wno-long-long -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option D: [(null)] main.c: Running on host: CYGWIN_NT-6.2 x86_64 1.7.25(0.270/5/3) 2013-08-31 20:37 D: [(null)] main.c: Found 4 CPUs. I: [(null)] main.c: Page size is 65536 bytes D: [(null)] main.c: Compiled with Valgrind support: no D: [(null)] main.c: Running in valgrind mode: no D: [(null)] main.c: Running in VM: no D: [(null)] main.c: Optimized build: yes D: [(null)] main.c: FASTPATH defined, only fast path asserts disabled. I: [(null)] main.c: Machine ID is 5d8bd07cb924c67197184e42527f2603. E: [(null)] core-util.c: Failed to create secure directory: Unknown error 13 When I instead run pulseaudio -vv --start the output is this: E: [autospawn] core-util.c: Failed to create secure directory: Unknown error 13 W: [autospawn] lock-autospawn.c: Cannot access autospawn lock. E: [(null)] main.c: Failed to acquire autospawn lock When I ran strace pulseaudio -vv, the red-colored lines in the output were: 28 1637050 [main] pulseaudio 5104 fhandler_pty_slave::write: (669): pty output_mutex(0xBC) released 26 1637076 [main] pulseaudio 5104 write: 7 = write(2, 0x3FE171079, 7) 42 1637118 [main] pulseaudio 5104 fhandler_pty_slave::write: pty0, write(0x60003BB40, 51) 27 1637145 [main] pulseaudio 5104 fhandler_pty_slave::write: (654): pty output_mutex (0xBC): waiting -1 ms 23 1637168 [main] pulseaudio 5104 fhandler_pty_slave::write: (654): pty output_mutex: acquired Failed to create secure directory: Unknown error 13 21 1637189 [main] pulseaudio 5104 fhandler_pty_slave::write: (669): pty output_mutex(0xBC) released 29 1637218 [main] pulseaudio 5104 write: 51 = write(2, 0x60003BB40, 51) 46 1637264 [main] pulseaudio 5104 fhandler_pty_slave::write: pty0, write(0x3FE17106F, 4) 24 1637288 [main] pulseaudio 5104 fhandler_pty_slave::write: (654): pty output_mutex (0xBC): waiting -1 ms 24 1637312 [main] pulseaudio 5104 fhandler_pty_slave::write: (654): pty output_mutex: acquired Please can someone help me?

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• Alsa hardware volume with PulseAudio

  - by Jan Hudec

  Before installing pulseaudio, I was able to control volume for the front (meaning on the front panel, the "headphone" jack) and rear (meaning on the back panel, the "line out" jack) separately. When I installed pulseaudio, it became possible to control volume for each playing process separately, but the individual controls for outputs disappeared. While the default device in alsa now routes via pulseaudio, the sysdefault device provides access to the hardware. But kmix does not seem to let me show them now. Is there any way to beat kmix into showing the sysdefault device too? Or something else X-based that would not fight with kmix too much? The system is Debian Jessie (testing) amd64, updated, KDE version 4:4.13.3-1.

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• Skype 2.1beta for Linux and sound quality

  - by vava

  I've been using Skype 2.1beta for Linux with my bluetooth headset and quality of the sound is just awful. But not always though, if I call echo service, quality is acceptable, but when I call real people there's echo, sound is crippling, there's pauses, voice is unrecognizable, all sorts of quality problems in one call. If I use newest Skype under WIndows with the same headset to call to the same people, quality is more than normal. So, is there some settings I can tweak, like tell Skype which codec to use or maybe there's noise cancellation plugin for PulseAudio I can use or any other system setting I can try to play with?

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• Limit PulseAudio volume to prevent overdrive

  - by el.pescado

  Is it possible to limit maximum volume in PulseAudio? Currently, PulseAudio sets PCM channel too loud which results in distorted sound. I use aumix to turn volume down, but whenever any other sound is played (IM notification etc), PA plays with knobs, turning master volume down and PCM up. aumix ++++++++++++++++++++O+++++<Vol ++++++++++++++++++++O+++++ Pcm ++++++++++++O+++++++++++++ becomes: aumix ++++++++++++++++O+++++++++<Vol ++++++++++++++++++++++++O+ Pcm ++++++++++++O+++++++++++++ I use OpenSUSE 11.2, pulseaudio 0.9.21, ALSA 1.0.21 and ALC889A sound codec.

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• Strange sound behavior

  - by caarlos0

  First of all, sorry for the tittle. English isn't my native language, and I can't find a word to describe the strange behavior I'm getting here. In the most simple explanation, is like sound keeps going down and up again... Think in a kid with that old radios that have a circled volume button. Think like these kid keeps "turning" the volume button. That is the behavior I'm getting here. At first, I believed that it was a pulseaudio issue, but, it isn't. I followed the wiki part I think that should be my problem, but it didn't work. After that, as I'm using XFCE, I didn't really need pulseaudio, so I removed it and stays with a clean alsa, hopping that will fix my problem. Sweet mistake. It really looks like a kid looking for trouble. I believed it worked, and, suddenly, here is the same issue again. BTW: I have a full-upgraded testing system (yeah, I upgraded to testing hopping for new pulseaudio version which fix the issue), no pulseaudio at all, just xfce, started with startxfce. What can I do to fix this? It's extremely annoying... sometimes I just want to throw my laptop in the wall because of this. Any extra info you need, please, tell me. Thanks in advance -- EDIT: My alsamixer is like this: And here is a video with the sound behavior.

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• choose a tool to create/maintain custom eclipse distrib

  - by raticulin

  I would like to settle on a tool to create/maintain my custom eclipse distrib (starting with next 3.6). By studying previous questions main contenders seem: Pulse Yoxos doing it yourself in eclipse Has anyone experiences in several of them and can comment on advantages etc?? My wishes are: by 'distrib' I mean: plugins, settings & preferences... be able to use the same eclipse setup in several workstations MAYBE sharing with other members of the team works across 3.5 and next 3.6: I don't know if it's possible. And anyway I would not object to customize the distrib once per new eclipse major release

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• The method split(String) is undefined for the type String

  - by pi

  I am using Pulse - the Plugin Manager for Eclipse and installed. I have the Eclipse 3.5 for mobile development(Pulsar) profile with a couple other profiles. I realized that the split() method called on a string from code such as below: String data = "one, two, three, four"; data.split(","); generates the error: "The method split(String) is undefined for the type String". I am aware that the split() method did not exist before Java's JRE 1.4 and perhaps could be the cause of the problem. The problem is I don't think I have jre/sdk versions installed. Perhaps there's one in-built with the Pulsar profile and needs editing - but I couldn't tell what settings (and where) needs tweaking. I have checked WindowsPreferencesJavaInstalled JREs and it's set to = jre1.4. Please help thanks.

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• Synchronizing audio and video using MP4Box / ffmpeg to concatenate files

  - by jdl2003

  I have two H.264 encoded MPEG-4 files that I need to concatenate. I have been using MP4Box for this task by first ensuring the files are encoded identically (even went so far as to compare output from h264_parse on their video tracks) and then concatenating with this command: MP4Box -cat file1.mp4 -cat file2.mp4 output_file.mp4 This works and the output file is playable, but on playback in Quicktime or VLC the second video's audio starts too soon, making the entire second part of the concatenated file out of sync. I have tried reencoding the output through ffmpeg with -vcodec copy and -acodec copy but the sync issue persists. I have also tried converting to MPEG-2 format first, concatenating with a simple cat file1.mpg file2.mpg > output.mpg and reencoding the result with ffmpeg. This was even worse. I know that I can pass commands to MP4Box to adjust the start time of the audio track, but I am trying to do this programmatically for any input video (in the same encoding of course). I am looking for possible solutions that would not require human intervention / manual adjustments. Or, at least, an understanding of what is happening to make the second part of the concatenated video go out of sync.

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• Sending microphone input over Remote Desktop 7.0

  - by Taylor Price

  I am using Remote Desktop 7 (the new version that came out with Windows 7) to control a Windows XP Pro machine. I have selected "Record from this computer" in the Remote Audio settings. When I connect to the machine, go to the control panel, open the sound panel, and go to the audio tab, I find that the default sound playback device is "Microsoft RDP Audio Driver". However, there is no default sound recording device. As expected, my IP phone thinks there is no recording device. If I am sitting in front of the computer with a mic plugged in, it works just fine. Has anybody else been able to get this work appropriately? Is there anything that I have to setup on the XP machine to get this working? Thanks in advance. Edit: As John T pointed out below, you have to be connecting to a Windows 7 Enterprise or Ultimate machine for this to work. I've also found out that Multi-monitor support has the same requirement.

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• QuickTime Player sounds much better than iTunes

  - by Gene Goykhman

  I am playing a 320 kpbs encoded music MP3 in iTunes and the sound is substantially worse than the exact same file played back in QuickTime Player (Max OS X 10.8.5). I have maxed out system volume and iTunes playback volume. I have disabled all the audio processing features in iTunes (equalization, sound enhancer, etc.) The audio coming from iTunes still sounds resampled and/or processed, whereas QuickTime Player appears to be playing it "as is". Even when I Get Info on the MP3 file in Finder and play it back directly from the Get Info window it sounds good. It's just iTunes that seems to be mangling the song. I can notice a difference on virtually all my music, so it's not just one particular MP3. I suspect the issue is that iTunes is doing some kind of audio processing but I can't find a way to turn it off. This is the newest iTunes (11.1), but the problem has probably been going on for a while... I just switched to decent earbuds and started noticing the difference. What's the best way to force iTunes to play back the file as-is, or as close as possible to how QuickTime Player/Finder would play it?

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• Speakers silent, headphones work in Ubuntu 9.04

  - by CarlF

  I'm running Ubuntu 9.04. Worked fine for months, then I rebooted yesterday after weeks of continuous operation. Now audio won't play through the speakers. The USB headset works fine, but the Conexant audio (CX20549) does not. Weirdly, it thinks it's playing. pavumeter shows appropriate levels, volume looks OK in alsamixer, but no sound. I did find this page: http://www.eugeneteplitsky.com/fixing-silent-pulseaudio-in-ubuntu-9-04/ Unfortunately the advice there doesn't help me. For one thing, the syntax for the alsa-base.conf file is apparently not actually documented anywhere. For another, my chipset isn't listed in the kernel.org docs he links to! EDIT: would upgrading to 9.10 help? Is there a major change in the audio subsystem between 9.04 and 9.10? Any suggestions? EDIT 2: This is stranger than I thought. Sound works normally in Xine, but is silent in Audacity, VLC and mplayer. What the?

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• avconv gets killed if mkv has subtitles

  - by Lukas Knuth

  What I'm trying to do is to take a movie (in an Matroska container), convert all audio tracks to AC3 and don't touch anything else. I'm using this line: avconv -i infile.mkv -map 0 -vcodec copy -scodec copy -acodec ac3 -ab 256k outfile.mkv This works fine, except when there are subtitles embedded. Then, after some time processing with no progress, avconv just "dies" (output shortened, these seem to be the interesting parts): [matroska,webm @ 0xf867a0] max_analyze_duration reached [matroska,webm @ 0xf867a0] Estimating duration from bitrate, this may be inaccurate ... Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' ... Stream #0.0(eng): Video: H264 / 0x34363248, yuv420p, 1280x536 [PAR 1:1 DAR 160:67], q=2-31, 1k tbn, 1k tbc (default) Stream #0.1(ger): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s (default) Stream #0.2(eng): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s Stream #0.3(ger): Subtitle: dvdsub (default) (forced) Metadata: title : forced Stream #0.4(ger): Subtitle: dvdsub Metadata: title : complete Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (dca -> ac3) Stream #0:2 -> #0:2 (dca -> ac3) Stream #0:3 -> #0:3 (copy) Stream #0:4 -> #0:4 (copy) Input stream #0:2 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 Input stream #0:1 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 frame= 2606 fps=1303 q=-1.0 size= 3kB time=107.36 bitrate= 0.2kbits/s ... frame=96141 fps=813 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s frame=96251 fps=810 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s ... frame=97015 fps=397 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s Getötet ["Killed", in English] I have no idea why this happens, as there is no error-output. I'd like to just copy the subtitles over, not touch them at all. If that won't work, they can be completely dropped.

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• Is ffmpeg incorrectly interpreting .aif files?

  - by marue

  Being on an Ubuntu 10.04 server i installed the ffmpeg packages with apt. ffmpeg is working afterwards, and doing as it should. Almost. For testing purposes i uploaded a few audiofiles. One of them, an aif file, is not being correctly interpreted. While on my workhorse (Mac SnowLeopard) ffmpeg tells the format as Stream #0.0: Audio: pcm_s24be, 44100 Hz, 2 channels, s32, 2116 kb/s my Ubuntu server says it is: Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s which is the wrong bitdepth. Ubuntu then fails to convert the file with the error message [pcm_s24be @ 0xcd4b580]invalid PCM packet Error while decoding stream #0.0 which certainly is not true. The file is perfectly valid. Are there any know issues for ffmpeg interpreting the aif format? How can i find out which version of the aif-codec ffmpeg is using? Any ideas how to approach this issue? ffprobe output: FFprobe version SVN-r20090707, Copyright (c) 2007-2009 Stefano Sabatini libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 built on Jan 20 2010 00:13:01, gcc: 4.4.3 20100116 (prerelease) Input #0, aiff, from 'testfile.aif': Duration: 00:00:04.00, start: 0.000000, bitrate: 2117 kb/s Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s update 2: Forcing the conversion with -sample_fmt s32 doesn't change anything. Strange thing is: Even without using -sample_fmt s32 i just realized that the conversion is working and creates valid audiofiles. There just is the error message from above.

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• Problems with 5.1 digital out on Ubuntu 12.04

  - by user895319

  I've recently bought a new PC, installed Ubuntu and am now unable to get 5.1 digital sound working. Simple analogue stereo works fine on both the front and rear connectors. On my old box I connected the coax connection from my soundcard to my surround sound amplifier, set Settings-Sound to "Digital Stereo Duplex" and it worked. My old soundcard doesn't fit in my new machine so I'm using the built-in sound hardware. I'm connecting the combination output socket on the back of the PC via the same cable to my surround amp as before. The MB is an MSI Global H61M-P31 with an RealTek ALC887 sound chip. When I go to Settings-Sound I only see "Headphone Built-in Audio" and "Analogue Output Built-in Audio" - no digitial options. The output from aplay -l is: default Playback/recording through the PulseAudio sound server sysdefault:CARD=PCH HDA Intel PCH, ALC887-VD Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers dmix:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Hardware device with all software conversions While googling for ALC887 I've seen some references to "ALC887 -VD Analog" and some to "ALC887 -VD Digital". Does anyone know if I need to force it to chance mode somehow? It's worth mentioning that when I set the output to 5.1 digital surround in Windows 7 on the same machine I still don't get any sound so it's not a unique Linux problem. Thanks for any help.

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• 5.1 Surround Channels are Jumbled

  - by stickynips

  I had this exact setup working previously, but after a reformat it went screwy on me. I have an Onkyo A/V Receiver hooked up to my PC, via optical S/PDIF. Attached to the receiver is a 5.1 speaker setup (tested and working fine with my Xbox via the receiver). It seems to me that the audio channels are getting mixed up somehow between the PC and the receiver. I have a 5.1 test file which plays a sounds through each speaker individually. The channels are mixed as such: "Left Front" plays through my Right Front speaker "Center" plays through my Left Front speaker "Right Front" plays through my Center speaker "Left Rear" plays through my Subwoofer "Right Rear" plays through my Left Rear speaker I've tried downloading the latest Realtek HD Audio Drivers and the Realtek HD Audio Manager, but neither makes any difference. If there's a way I can manually rearrange the channels I believe it would fix the problem, but as far as I know this is impossible. edit: Sorry, I've forgotten some basic info. I'm running Windows 7 x64. The sound card is Realtek ALC892 embedded in a GIGABYTE GA-890GPA-UD3H AM3 motherboard.

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• How to get the mic on the Creative X-Mod soundcard working correctly?

  - by Nyamiou The Galeanthrope

  Well, I have this problem for a while now. When my computer start the mic seem to work but it's like it's muted. I have to go to a terminal and type alsamixer -c 1 and then I set up PCM Capture Source on Line and set up it back to Mic to get the mic actually working. Is there is a way to do this automatically or to solve the problem. I use a special workaround on this card because of the bug #429642. My workaround is having this at the end of my /usr/share/pulseaudio/alsa-mixer/profile-sets/default.conf : [Mapping xmod-stereo-out] device-strings = surround51:%f description = Analog Stereo Creative Xmod channel-map = front-left,front-right paths-output = analog-output analog-output-headphones analog-output-mono analog-output-lfe-on-mono paths-input = analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line priority = 10 Maybe the bug come from here, maybe I have to change something. Thank you for any help.

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• Renoise and dssi and jack

  - by Bojan

  This may be a little complicated, but, is jack a necessity? I mean, i use renoise, and, since i dont have the need for low latencies, do i really need to use it? My basic setup ( or workflow ) is that i use csound to render stuff to wav, then import it as a sample in renoise. That goes with field recordings, my own samples, etc. So, i dont need ultra low latencies, and i dont need to patch "cords", but i want to use dssi plugins, and dssi-vst. What would be something of a minimum requirements of apps that should work. Can renoise load dssi-vst plugins by itself or do i need to use jack to patch thru or something third, i tried to read lot of articles but i got lost in the different setups...

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