Sound Devices Firmware

[Nguyet Mahrenholz]

Iam very impressed. I figured that SD never seeming to get to this meant that the 6xx machines were not capable of it somehow. There hasn't been a new model in that series in some years now, and the new run of MixPres have come out since then, so it is all the more impressive that SD would devote the resources to making this happen. I understand that many recordists using SD gear may not understand what a big deal this is, UNTIL they end up on a shoot with a paranoid director and a backfield of ambitious, meddlesome clients and agency folks. A manually operated client-headphone mute system is an embarrassing accident waiting to happen. On a fast moving shoot it is inevitable that you will blow it at some point, probably every day, even with blinking lights etc etc. For working soundies this automute thing is as big a deal as auto-mix! Thanks!

I'm not sure about older, but the 664 is an analog mixer with a digital recorder integrated, whereas the 633 and 688 are both digital mixers. Unfortunately features can't be added to an analog device through a firmware update.

SD has added to the 664 features over the years via firmware. But not being a digital mixer means it can't have purely digital features like automix. On the other hand it does do a few things the later 6xx machines can't (mix bus or "cascade" input, for instance).

Sound Devices has released a 8-Series v10.0 firmware update. This Free user-driven update includes 32-bit float recording, streamlined functionality, and more flexible audio routing options. The firmware update is a direct result of customer feedback.

The firmware introduces the ability to record ISO tracks in 32-bit floating point, eliminating clipping in the file, no matter the gain setting. It is good to see Sound Devices integrate 32-bit float recording capabilities in their flagship mixer/recorder series.

Users can also choose whether playback audio goes to all outputs or just headphones, while leaving live audio to pass through to other outputs. This is beneficial for production crews, who can continue hearing live audio without interruption while the sound mixer can check playback of takes.

On Wednesday, May 29th, 2024, at 11:00am CDT (4:00pm GMT), Paul Isaacs will be joined by Sound Devices Technical Support Expert Laura Sillanp for a live-streaming event where they will demonstrate some of the new features in the v10.01 firmware. Attendees will be able to submit questions to be answered during the session or following the live stream. The event is open to the public, but registration in advance is required at -series-v10-01-live-stream/. To learn more about 8-Series mixer-recorders and to download v10.01 firmware, visit www.sounddevices.com/8-series.

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I have taken a deep dive into linux lately. I recently reinstalled my fedora to fedora 36 server with the custom install. I have setup i3WM and some other custom stuff and it works great for the most part. But the internal sound card does not seem to work.

I assume I am just missing some drivers or something for my intel soundcard. I had a simaler install a copple a weeks ago but when I installed that I hooked off sound and video during the install, if I remember correctly (been some monts since I did) This time there were no such option. Sound is however working with external usb devices e.g. headset and thunderbolt dock.

Finally found a solution to this problem. For anyone else who might have this problem as well. Here is the link to the post who saved me.

-36-no-sound-coming-out-of-hdmi-intel-i5-8250u-setting-intel-iommu-on-igfx-off-in-grub-doesnt-help/75160/8?u=vegard

Sound Devices has released a firmware update, v10.0, for its 8-Series mixer recorders (833, 888, and Scorpio models), adding features based on user feedback. This free update introduces 32-bit float recording for ISO tracks, preventing clipping regardless of gain settings, and a new Daylight Mode for improved LCD visibility in bright conditions. Users can now rename record folders directly on the device, choose playback audio routing, and benefit from an enhanced SD-Remote app with a comprehensive Routing Matrix tab. This is great for production crews, to continue hearing live audio without interruption while the sound mixer can check playback of takes.

The biggest news is that the MixPre-6 and -10T now have Ambisonic support. Ambisonic compatibility has been a popular request from those working in 360 field recording such as VR development and game developers.

That will unlock the ability to record, playback, and monitor both A-Format and B-Format (FuMa and AmbiX) Ambisonics up to 192 kHz. It can also do the same to Ambisonics decoded to binaural and stereo, although binaural is limited to lower sampling rates. In fact, Ambisonics, stereo, and binaural can be recorded at once!

Of course, a recorder introducing any noise is a serious problem. Beyond that, this was a significant issue for sound designers in particular. Any ultrasonic problems created in the upper spectrum would be revealed when pitch-shifting or resampling the file. Thankfully, firmware version 3.00 fixes the ultrasonic noise bug.

Besides the sound device drivers, ALSA also bundles a user space driven library for application developers. They can then use those ALSA drivers for high level API development. This enables direct (kernel) interaction with sound devices through ALSA libraries.

udev will automatically detect your hardware and select needed drivers at boot time, therefore, your sound should already be working. However, your sound may be initially muted. If it is, see #Unmuting the channels.

Install the alsa-utils package. This contains (among other utilities) the alsamixer(1) and amixer(1) utilities. amixer is a shell command to change audio settings, while alsamixer provides a more intuitive ncurses based interface for audio device configuration.

ALSA has some ability to intercept OSS calls and re-route them through ALSA instead. This emulation layer is useful e.g. for legacy applications which try to open /dev/dsp and write sound data to them directly. Without OSS or the emulation library, /dev/dsp will be missing, and the application will not produce any sound.

Evidently, both methods are mutually exclusive. You can decide for one of the two approaches depending on your requirements. To edit these units, see systemd#Editing provided units. You can check their status using systemctl.

sof-firmware is required for some laptop models (mainly since 2019) because they implement their drivers with firmware provided by the Sound Open Firmware project. Checking the journal will provide messages about the missing firmware (see BBS#275577).

To get full 5.1 or 7.1 surround sound, you will likely need to unmute other channels such as Front, Surround, Center, LFE (subwoofer) and Side. (Those are channel names with Intel HD Audio; they may vary with different hardware)

Everything depends on user preferences when it comes to different styles of configuration; however, one should avoid mixing different styles. Further information on basic configuration can be found in [3].

ALSA uses different data types for parameter values, which must be set in the users respective configuration file. Some keys accept multiple data types, while most do not. A list of configuration options and their respective type requirements for PCM plugins can be found in [4].

There are different operation modes for parsing nodes, the default mode is merge and create. If operation mode is either merge/create or merge, type checking is done. Only same type assignments can be merged, so strings cannot be merged with integers. Trying to define a simple assignment in default operation mode to a compound (and vice versa) will also not work.

Assuming that "defaults" node is set in /usr/share/alsa/alsa.conf, where "defaults.pcm.card" and its "ctl" counterpart have assignment values "0" (type integer), user wants to set default pcm and control device to (third) sound card "2" or "SB" for an Azalia sound card.

Using double quotes here automatically sets values data type to string, so in the above example, setting defaults.pcm.!card "2" would result in retaining last default device, in this case card 0. Using double quotes for strings is not mandatory as long as no special characters are used, which ideally should never be the case. This may be irrelevant in other assignments.

Use $ cat /proc/asound/modules to get the loaded sound modules and their order. This list is usually all that is needed for the loading order. Use $ lsmod grep snd to get a devices & modules list. This configuration assumes you have one mia sound card using snd_mia and one (e.g. onboard) card using snd_hda_intel.

You can also provide an index of -2 to instruct ALSA to never use a card as the primary one. Distributions such as Linux Mint and Ubuntu use the following settings to avoid USB and other "abnormal" drivers from getting index 0:

Probably, it is enough to set ALSA_CARD to the name of the device. First, get the names with aplay -l, then set ALSA_CARD to the name which comes after the colon and before the bracket; e.g. if you have

In this case as well, replace Audigy2 with the name of your device. You can get the names with aplay -l or you can also use PCMs like surround51. But if you need to use the microphone, it is a good idea to select full-duplex PCM as default.

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