set the channel description: Developer preview of Rust SDK

Hey all, just sharing it here: https://livekit-users.slack.com/archives/C01KZ61HZRR/p1673461147751939 Would love to hear if these APIs (particularly around events) make sense to y'all. We wanted to build something that feels natively Rust.

Im curious to know .. with this Rust client, could we use this server side also to join rooms, somewhat like Go SDK does or its really aimed for client SDK?

Do you have an ETA on release of publishing tracks?

Hey Jonathan, we are working on it! Perhaps sometimes this month.

I recently became aware of LiveKit, the project looks great. We are working on a Rust-based project with WebRTC-based functionality, and are evaluating LiveKit for our needs. After successfully testing out a self-hosted server with the example web frontend (example.livekit.io), I tried out the client-sdk-rust project's simple_room example. I understand the Rust client is very early in development. I have a few questions questions: 1. I am unable to get video to display in the Rust egui UI - even when another client (via example.livekit.io) is connected. Furthermore, I'm seeing crashes (on two different Windows 10 machines) - after a web client joins, I get hard crashes with:

error: process didn't exit successfully: c:\FF\client-sdk-rust\examples\target\release\simple_room.exe (exit code: 0xc0000005, STATUS_ACCESS_VIOLATION)

Is this a known issue? 2. More generally - I note the readme states one of the goals as:

Build a standalone, cross-platform LiveKit client SDK for Rustaceans.

Our project is cross platform (desktop, web, iOS, Android) with common client-side code built in Rust (UI in other languages, varies by platform). When the stated goal is "cross-platform" - what is the scope, in terms of platform support? I see Windows, Mac, iOS, and Android in the webrtc-sys build.rs I assume Linux is planned, but WebAssembly is out of scope? iOS and Android? I ask because it would be great to have a single Rust codebase that can target all platforms (desktop, iOS, Android), other than the UI layer on top of course. Is that the intention here? Of course, we can instead integrate with platform-specific LiveKit clients (Android, iOS, React Native, etc), which is workable alternative. 3. Are you able to share any sort of roadmap or ETAs for the Rust client? i.e., do you have an estimate for when it will be largely feature complete? Thanks.

Does/will the rust sdk support token creation? 🤔

Hey folks, we plan to adopt LiveKit for our media stream infra. Currently, the biggest blocker is the C++ SDK as we’d require it for our publisher. @dry-elephant-14928 mentioned that you are considering building it on top of Rust sdk. I want to go ahead and explore a bit as we also have our timeline and want to make sure this is something we can make it on time. Since I am new to Rust, want to check are you going to use cpp_abi or cxx? Any instructions on where I can start with?

👋 Hey all! I've been working with livekit using the CLI for some time, and am now working to port some of the CLI hacks to Rust code. My current method was using gstreamer to put my stream into TCP socket, and then publish to a room using

livekit-cli join-room .. --publish h264://..

-- this isn't ideal since: 1. It requires a separate process for each participant 2. (more importantly) It uses the server credentials, rather than a join token I'd like to take my GStreamer pipeline directly to Livekit using the Rust SDK, but I'm having trouble understanding how I can create a publish video track that didn't originate from livekit webrtc

Are there instructions for prerequisites for using livekit crate? I already noticed that it was failing until I installed protobuf with

brew install protobuf

Now, I’m getting a linking error:

error: failed to run custom build command for `webrtc-sys v0.1.1 (<https://github.com/livekit/client-sdk-rust#main#a5caf129>)`

Caused by:
  process didn't exit successfully: `/Users/cole/autoplay/craft/target/debug/build/webrtc-sys-ae8729e557fd0c78/build-script-build` (signal: 6, SIGABRT: process abort signal)
  --- stderr
  dyld[7331]: Library not loaded: @rpath/libbz2.1.dylib
    Referenced from: <0AFDF875-749F-3995-B4DC-93C95538B617> /Users/cole/autoplay/craft/target/debug/build/webrtc-sys-ae8729e557fd0c78/build-script-build
    Reason: tried: '/Users/cole/autoplay/craft/target/debug/deps/libbz2.1.dylib' (no such file), '/Users/cole/autoplay/craft/target/debug/libbz2.1.dylib' (no such file), '/Users/cole/.rustup/toolchains/stable-aarch64-apple-darwin/lib/rustlib/aarch64-apple-darwin/lib/libbz2.1.dylib' (no such file), '/Users/cole/.rustup/toolchains/stable-aarch64-apple-darwin/lib/libbz2.1.dylib' (no such file), '/Users/cole/lib/libbz2.1.dylib' (no such file), '/usr/local/lib/libbz2.1.dylib' (no such file), '/usr/lib/libbz2.1.dylib' (no such file, not in dyld cache), '/Users/cole/autoplay/craft/target/debug/build/zstd-sys-68a5717b1c36b3f2/out/libbz2.1.dylib' (no such file)

Now that video is working for my gstreamer -> livekit demo, I'll start adding audio today, and then source discovery.

Ok... so... contrary to my assumptions. Audio is harder to figure out than video 😉 I've just pushed a commit include the broken audio, but the basics are this: 1. I have a GStreamer audio appsink sample, which should be in

S16be

-- as Livekit expects. 2. I also generate a GStreamer

AudioInfo

which will report the n_samples (

1024

), channels (

2

) & rate (

48000

) 3. I can decode the buffer using the

AudioInfo

, which formats the buffer as a

&[u8]

4. I then convert the buffer slice into a

Vec<i16>

using

byteorder

(which handles the boundary edge conditions and such) Unfortunately, the audio track never seems to join Livekit... I swapped in the sine track as a test, and it joins up fine, so I suspect there's something invalid about my audio frame https://github.com/spiegela/gstreamer-livekit

👋 Hi There, I am trying to build

client-sdk-rust

on my ubuntu-22.10 and I am running into issues building with the following error.

error: failed to run custom build command for `livekit-protocol v0.1.0 (/home/ggovindan/client-sdk-rust/livekit-protocol)`

Caused by:
  process didn't exit successfully: `/home/ggovindan/client-sdk-rust/target/debug/build/livekit-protocol-745ac081750311d3/build-script-build` (exit status: 1)
  --- stderr
  Error: Custom { kind: Other, error: "protoc failed: Could not make proto path relative: protocol/livekit_egress.proto: No such file or directory\n" }

Appreciate any pointers to what I might be missing.

Hi, I am testing out the example program from the client-rust-sdk with a url and a token from an active livekit. I am getting segmentation fault. I could not find anything from the gdb. I was able to play the same url with

<http://example.livekit.io|example.livekit.io>

. Am I missing something? I am running on ubuntu 22.10. The segmentation fault seems to happen in

Room::connect(&url, &token).await;

I tried running it in gdb and got this

Thread 1 "spotwebrtc" received signal SIGSEGV, Segmentation fault.
___pthread_rwlock_wrlock (rwlock=0x0) at ./nptl/pthread_rwlock_wrlock.c:26

I just started getting this error from the Room client when creating a client using `RoomClient::with_api_key`:

invalid claims: token grants room_join but doesn't have an identity

The breakage starts with commit `1c12e74` and looks related to this change, I think?

I also hit an issue building on Ubuntu today, which seems to require an additional

protoc

argument:

#0 117.3 error: failed to run custom build command for `livekit-protocol v0.1.0 (<https://github.com/livekit/client-sdk-rust.git?rev=a5caf12#a5caf129>)`
#0 117.3 
#0 117.3 Caused by:
#0 117.3   process didn't exit successfully: `/usr/src/rtsp-publisher/target/release/build/livekit-protocol-11624bcec5533409/build-script-build` (exit status: 1)
#0 117.3   --- stderr
#0 117.3   Error: Custom { kind: Other, error: "protoc failed: livekit_room.proto: This file contains proto3 optional fields, but --experimental_allow_proto3_optional was not set.\n" }

Here's a PR to add the indicated build arg: https://github.com/livekit/client-sdk-rust/pull/65

I’ve been doing some testing to containerize the livekit rust client, and while I get the builds to complete,

ldd

shows everything linked, the seg-faults on startup. Any ideas?

So... regarding the issue linking

libpthread

through

webrtc-sys

on Ubuntu 22.04... I've tried nearly everything I can come up with. I thought it might not be discovering the path, so I copied

/lib/aarch64-linux-gnu/libpthread.so.0

to

/lib

,

/usr/lib

,

/usr/local/lib

as both

libpthread.so.0

and

libpthread.so

... I've tried a multitude of options in the

webrtc-sys

build script, but none of them have linked libpthread. I found this post, so I wonder if the issue couldn't be the GCC version?

Hello! First of all, I'd like to say that this rust client sdk is a marvel - really beautiful implementation and a great example of how to do a bunch of interesting things in rust! I'll be exploring this SDK for a while! I have a question though: I need to have access to the raw, uncompressed video frames coming from the RTP stream (remote tracks), and AFAIU the

libwebrtc

is higher level than that, so to achieve what I need I'd need to make changes to

libwebrtc

itself. Is my reasoning correct?

Q: I'm trying to compile (in a mac) the

libwebrtc

bundled in

webrtc-sys

- I'm setting the

LK_CUSTOM_WEBRTC

flag, but it seems to require

libwebrtc

to already have been built before. By running

build_macos.sh

it downloads depot_tools and attempts to compile, but I'm getting a

fatal error: 'cxxabi.h' file not found

error. Which steps am I missing?

A description of the higher level problem I'm trying to solve, so maybe you can help me to find a better alternative: • I want to do further processing (ML inference) with the video frames I receive from the media streams. A good way to do that is to have the video frames decoded by the GPU (nvdec or equivalent), so they are already in GPU memory, ready to be used for inference. Otherwise the decoded frames are copied to RAM, and then need to be copied again to the GPU.

Getting an error when attempting to build

livekit-protocol

on both mac and linux:

~/src/client-sdk-rust/livekit-protocol$ cargo build
warning: profiles for the non root package will be ignored, specify profiles at the workspace root:
package:   /home/gbuzogany/src/client-sdk-rust/livekit-ffi/Cargo.toml
workspace: /home/gbuzogany/src/client-sdk-rust/Cargo.toml
   Compiling livekit-protocol v0.1.0 (/home/gbuzogany/src/client-sdk-rust/livekit-protocol)
error[E0432]: unresolved import `futures_util::sink`
 --> livekit-protocol/src/observer.rs:1:19
  |
1 | use futures_util::sink::Sink;
  |                   ^^^^ could not find `sink` in `futures_util`

For more information about this error, try `rustc --explain E0432`.
error: could not compile `livekit-protocol` due to previous error

has anybody seen anything like this before?

Q: What's the easiest way to find out which codec was used to encode a remote video track?

The

client-sdk-rust

doesn't seem to be able to receive H264 streams 🤔. If I change

videoCodec

and

backupCodec

to

h264

in the client (browser), the

client-sdk-rust

connecting to the same room just keeps re-negotiating forever