get webrtc adm into rust by xianshijing-lk · Pull Request #1037 · livekit/rust-sdks

xianshijing-lk · 2026-04-22T19:49:23Z

Summary

This PR implements Platform Audio support for the LiveKit Rust SDK, enabling WebRTC's built-in audio device handling with microphone capture and speaker playout. The implementation introduces a handle-based PlatformAudio API that coexists with the existing NativeAudioSource for manual audio pushing.

Key Features

Two Audio Source Types:
- RtcAudioSource::Native (default): Manual audio push via NativeAudioSource for TTS, file streaming, agents
- RtcAudioSource::Device: WebRTC handles mic capture & speaker playout with echo cancellation (AEC)
Handle-based API: Create PlatformAudio instances that enable ADM recording; drop to release
Reference Counting: Multiple PlatformAudio instances share the same underlying ADM
Device Enumeration & Selection: List and select recording/playout devices
Hot-swap Device Switching: switch_recording_device() / switch_playout_device() for changing devices during active sessions
Audio Processing Configuration: AEC, AGC, NS with hardware/software preference
WebRTC Patching: external_audio_source.patch prevents audio mixing conflicts between device and manual sources

Design Document

See docs/ADM_PROXY_DESIGN.md for full architecture details including:

Recording gate pattern
WebRTC patching explanation
FFI API documentation

API Overview

use livekit::prelude::*;

// Create PlatformAudio instance (enables ADM recording)
let audio = PlatformAudio::new()?;

// Enumerate and select devices
for i in 0..audio.recording_devices() as u16 {
println!("Mic [{}]: {}", i, audio.recording_device_name(i));
}
audio.set_recording_device(0)?;

// Connect and publish
let (room, _) = Room::connect(&url, &token, RoomOptions::default()).await?;
let track = LocalAudioTrack::create_audio_track("mic", audio.rtc_source());
room.local_participant().publish_track(LocalTrack::Audio(track), opts).await?;

// Cleanup - just drop the handle
room.close().await?;
drop(audio); // ADM recording disabled when all handles released

Testing

Run Standalone Tests (no LiveKit server required)

Set custom WebRTC build path

export LK_CUSTOM_WEBRTC="/path/to/webrtc-sys/libwebrtc/mac-arm64-debug"

Run standalone PlatformAudio tests

cargo test -p livekit --test platform_audio_test test_platform_audio_standalone -- --nocapture

Run FFI request handler tests

cargo test -p livekit-ffi requests::tests -- --nocapture

Run E2E Integration Tests (requires LiveKit server)

Start a local LiveKit server first, then:

LIVEKIT_URL=ws://localhost:7880
LIVEKIT_API_KEY=devkey
LIVEKIT_API_SECRET=secret
cargo test -p livekit --test platform_audio_test --features __lk-e2e-test -- --nocapture

Test Coverage
Category │ Tests │ Description │

Standalone - Creation │ 1 │ PlatformAudio creation, device enumeration
Standalone - Ref Counting │ 1 │ Clone, sharing, drop behavior
Standalone - Device Selection │ 1 │ Set devices, invalid index handling
Standalone - Processing │ 1 │ AEC/AGC/NS configuration, hardware availability
Standalone - Reset │ 1 │ reset_platform_audio() function
Standalone - Lifecycle │ 1 │ Full create→configure→use→release cycle
FFI - Handlers │ 6 │ NewPlatformAudio, GetDevices, SetDevice, handle lifecycle
E2E - Room Connection │ 4+ │ Platform audio with room, two participants, device switching

All tests handle missing audio devices gracefully (CI-friendly).
Run the Example

List Audio Devices

cargo run -p basic_room -- --list-devices

Connect with Platform Audio (microphone capture)

LIVEKIT_URL=wss://your-server.livekit.cloud
LIVEKIT_API_KEY=your-key
LIVEKIT_API_SECRET=your-secret
cargo run -p basic_room -- --platform-audio

Connect with File Audio

cargo run -p basic_room -- --file path/to/audio.raw

Connect with Both Platform Audio and File

cargo run -p basic_room -- --platform-audio-and-file path/to/audio.raw

WebRTC Build Requirements

The external_audio_source.patch must be applied to WebRTC. The patch is automatically applied by all platform build scripts:

build_macos.sh
build_ios.sh
build_linux.sh
build_android.sh
build_windows.cmd

For local development, set LK_CUSTOM_WEBRTC to point to your patched WebRTC build.

Known Limitations

   Limitation      │                                  Description

Process-global │ Audio configuration affects all rooms in the process
Device indices │ May change on hot-plug; match by name for persistence
Single device track │ One device audio track per ADM (use NativeAudioSource for additional streams)

…o a room and thus failing the audio mode switching

…different local audio tracks

github-actions · 2026-04-27T23:37:24Z

Changeset

The following package versions will be affected by this PR:

Package	Bump
`libwebrtc`	`patch`
`livekit`	`patch`
`livekit-ffi`	`patch`
`webrtc-sys`	`patch`

…hub.com/livekit/rust-sdks into sxian/CLT-2765/bring-webrtc-adm-to-rust

MaxHeimbrock · 2026-05-07T13:40:51Z

+- Text-to-speech (TTS) audio
+- Audio from files or network streams
+- Testing without audio hardware
+


This is the original audio input right? Existing Unity clients who want to keep the "Unity" style microphone management would also use this.

MaxHeimbrock · 2026-05-07T13:43:19Z

+
+### Hybrid Approach
+
+You can combine both approaches - use `PlatformAudio` for automatic speaker playback while also creating `NativeAudioStream` for audio processing/analysis:


Is this also possible from a Unity client? As we discussed, for the lip sync animation Unity clients might want read access to the audio data, but still want output through the platform audio.

Yes, that is possible with Unity client for such lip sync animation + mic / speaker audio via PlatformAudio

MaxHeimbrock · 2026-05-07T13:45:25Z

+// Set recording device
+message SetRecordingDeviceRequest {
+  uint64 platform_audio_handle = 1;
+  uint32 index = 2;
+}
+
+message SetRecordingDeviceResponse {
+  optional string error = 1;
+}
+
+// Set playout device
+message SetPlayoutDeviceRequest {
+  uint64 platform_audio_handle = 1;
+  uint32 index = 2;
+}


How does it handle switching the device at runtime?

webrtc handles the runtime device switching automatically.

If the current device is removed during a call, it will automatically switch over to "default device", and the call should still be on-going.

MaxHeimbrock · 2026-05-07T14:08:43Z

+**Suitable for:**
+- Server-side agents
+- Text-to-speech (TTS) audio
+- Audio from files or network streams


Or screen share audio right?

xianshijing-lk · 2026-05-12T21:45:38Z

Hi @ladvoc , I think I addressed your comments, PTAL.

ladvoc

This is looking great, just a few more comments.

ladvoc · 2026-05-12T22:34:01Z

+    /// ```
+    ///
+    /// [`recording_device_guid`]: Self::recording_device_guid
+    pub fn set_recording_device_by_guid(&self, guid: &str) -> AudioResult<()> {


issue: It looks like these methods that don't accept new types have been left here for FFI support. A better pattern would be to remove them and add a constructor on the new types to initialize from the raw value (explicitly marked as unchecked, only to be used for FFI):

impl PlayoutDeviceId { #[doc(hidden)] pub fn from_unchcked_guid(guid: &str) → Self { Self(guid.to_string()) } }

Then FFI can still construct as needed and use the existing set/switch methods:

match ffi_audio.audio.set_recording_device(PlayoutDeviceId::from_unchecked_guid(&req.device_id)) { // ... }

ladvoc

LGTM ✅

ladvoc · 2026-05-14T00:22:55Z

+    }
+
+    // =========================================================================
+    // Audio Processing (AEC, AGC, NS)


nitpick: Prefer grouping related methods into separate impl blocks. impl blocks can also have doc comments attached to them, and each block will get it's own section in the generated docs to make it easier for developers to find things. Preview generated docs locally with cargo doc --no-deps from within the livekit crate. Example:

/// Audio Processing (AEC, AGC, NS) impl PlatformAudio { // ... }

ladvoc · 2026-05-14T00:27:03Z

+/// // Reset handle references
+/// reset_platform_audio();
+/// ```
+pub fn reset_platform_audio() {


suggestion (non-blocking): Maybe make this an associated function (no &self) on PlatformAudio to avoid the extra import for the user:

impl PlatformAudio { pub fn reset() { // ... } }

User can invoke with PlatformAudio::reset().

ladvoc · 2026-05-14T00:34:46Z

Last thing I just noticed: livekit-ffi is missing from the changeset.

cloudwebrtc

lgtm!

cloudwebrtc · 2026-05-14T00:19:33Z

  std::shared_ptr<RtcRuntime> rtc_runtime() const { return rtc_runtime_; }

+  // Device enumeration
+  int16_t playout_devices() const;


Should we move the following method to the audio_device.rs file? And add a PeerConnectionFactory::audio_device() -> SharedPtr<AudioDevice> pointing to (AudioDevice/AdmProxy)?

This isn't a strong requirement, but it will make the webrtc-sys codebase look clearer.

hmm, I got where this comment comes from.

I refactor it into a audio_device_controller.cpp and audio_device_controller.r, and move those device handling code to such audio_device_controller so that the peer connection factory will be cleaner.
It is a larger refactoring than what I initially expected it to be.

The layers now look like this:

PlatformAudio

LkRuntime

libwebrtc::PeerConnectionFactory

webrtc-sys::PeerConnectionFactory

webrtc-sys::AudioDeviceController

AdmProxy

What each layer does:

PlatformAudio

public Rust API for device enumeration, selection, switching, and platform audio lifecycle

LkRuntime

internal LiveKit runtime wrapper

forwards platform-audio operations to libwebrtc

libwebrtc::PeerConnectionFactory

Rust wrapper over the native bridge

still owns the main sys factory handle

now also caches one sys audio-device controller handle

webrtc-sys::PeerConnectionFactory

native C++ factory object

owns the real shared audio-device state via AdmProxy

creates and caches one AudioDeviceController

webrtc-sys::AudioDeviceController

focused facade for audio-device/ADM operations only

no independent audio state of its own

AdmProxy

the actual implementation behind device enumeration, selection, playout/recording control, and platform ADM activation

How AdmProxy is shared:

webrtc-sys::PeerConnectionFactory constructs AdmProxy once in its constructor

it stores that in adm_proxy_

it also constructs one AudioDeviceController with the same adm_proxy_

both hold references to the same AdmProxy

So the ownership/reference picture is:

PeerConnectionFactory

owns adm_proxy_

owns cached audio_device_

AudioDeviceController

holds a scoped_refptr

cloudwebrtc · 2026-05-14T00:30:07Z

+  // Create synthetic ADM for synthetic mode.
+  // This pumps the WebRTC audio pipeline without platform audio,
+  // allowing FFI callbacks to receive decoded remote audio.
+  synthetic_adm_ = webrtc::make_ref_counted<AudioDevice>(env_);


Should we rename AudioDevice to DummyAudioDevice/dummy_audio_device.cpp? (maybe SyntheticAudioDevice)

Good idea, SyntheticAudioDevice is clearer than AudioDevice.

Done with renaming it.

xianshijing-lk · 2026-05-14T02:28:12Z

Last thing I just noticed: livekit-ffi is missing from the changeset.

Done with adding livekit-ffi to the changeset and also added
FFI platform audio support: Expose platform audio device enumeration and selection through livekit-ffi

xianshijing-lk · 2026-05-08T22:36:38Z

@@ -136,9 +136,19 @@ export declare enum AudioStreamType {
 */
 export declare enum AudioSourceType {


@ladvoc , I change the audio_frame proto in this PR, should I commit this pb.d.ts change? not sure why the node proto code is in the rust SDK

@lukasIO moved the Node bindings to this repo a month or two ago (I don't remember the exact reasons). But yes, the generated Node bindings get committed. However, they are automatically generated by CI so you shouldn't have to worry about generating/committing manually.

xianshijing-lk · 2026-05-13T23:45:11Z

+    /// ```
+    ///
+    /// [`recording_device_guid`]: Self::recording_device_guid
+    pub fn set_recording_device_by_guid(&self, guid: &str) -> AudioResult<()> {


Good idea, done with addressing the comment.

xianshijing-lk · 2026-05-14T02:31:13Z

+  // Create synthetic ADM for synthetic mode.
+  // This pumps the WebRTC audio pipeline without platform audio,
+  // allowing FFI callbacks to receive decoded remote audio.
+  synthetic_adm_ = webrtc::make_ref_counted<AudioDevice>(env_);


Good idea, SyntheticAudioDevice is clearer than AudioDevice.

Done with renaming it.

xianshijing-lk · 2026-05-14T02:34:20Z

  std::shared_ptr<RtcRuntime> rtc_runtime() const { return rtc_runtime_; }

+  // Device enumeration
+  int16_t playout_devices() const;


hmm, I got where this comment comes from.

I refactor it into a audio_device_controller.cpp and audio_device_controller.r, and move those device handling code to such audio_device_controller so that the peer connection factory will be cleaner.
It is a larger refactoring than what I initially expected it to be.

The layers now look like this:

PlatformAudio

LkRuntime

libwebrtc::PeerConnectionFactory

webrtc-sys::PeerConnectionFactory

webrtc-sys::AudioDeviceController

AdmProxy

What each layer does:

PlatformAudio

public Rust API for device enumeration, selection, switching, and platform audio lifecycle

LkRuntime

internal LiveKit runtime wrapper

forwards platform-audio operations to libwebrtc

libwebrtc::PeerConnectionFactory

Rust wrapper over the native bridge

still owns the main sys factory handle

now also caches one sys audio-device controller handle

webrtc-sys::PeerConnectionFactory

native C++ factory object

owns the real shared audio-device state via AdmProxy

creates and caches one AudioDeviceController

webrtc-sys::AudioDeviceController

focused facade for audio-device/ADM operations only

no independent audio state of its own

AdmProxy

the actual implementation behind device enumeration, selection, playout/recording control, and platform ADM activation

How AdmProxy is shared:

webrtc-sys::PeerConnectionFactory constructs AdmProxy once in its constructor

it stores that in adm_proxy_

it also constructs one AudioDeviceController with the same adm_proxy_

both hold references to the same AdmProxy

So the ownership/reference picture is:

PeerConnectionFactory

owns adm_proxy_

owns cached audio_device_

AudioDeviceController

holds a scoped_refptr

xianshijing-lk · 2026-05-14T20:26:53Z

Landing now

@cloudwebrtc

> [!IMPORTANT] > Merging this pull request will create these releases # livekit 0.7.40 (2026-05-14) ## Fixes - feat: add scalability mode for AV1/VP9. - #1076 (@cloudwebrtc) - Add `LIVEKIT_PREFERRED_HW_ENCODER` to prefer `nvenc` or `vaapi` hardware video encoding when both are available. - Relocate unrelated types out of `livekit-protocol` ### Get WebRTC ADM into Rust - #1037 (@xianshijing-lk) This PR introduces platform audio device management via WebRTC's Audio Device Module (ADM). ### Features - **ADM Proxy**: New `AdmProxy` class that switches between Dummy ADM (synthetic mode) and Platform ADM (real audio I/O) - **PlatformAudio API**: High-level Rust API for microphone capture and speaker playout with AEC/AGC/NS - **Device enumeration**: List and select recording/playout devices by index or GUID - **Mode switching**: Seamlessly switch between synthetic mode (FFI callbacks) and platform mode (native speakers) while audio is active - **FFI platform audio support**: Expose platform audio device enumeration and selection through `livekit-ffi` - **Audio processing**: Configure echo cancellation, noise suppression, and auto gain control with platform-specific defaults (hardware on iOS, software elsewhere) ### Audio Modes | Mode | Recording | Playout | Use Case | |------|-----------|---------|----------| | Synthetic | NativeAudioSource | Dummy ADM + FFI | Unity audio, agents | | Platform | Platform ADM mic | Platform ADM speakers | VoIP with AEC | ### API ```rust // Create PlatformAudio for microphone/speaker access let audio = PlatformAudio::new()?; // Enumerate and select devices for i in 0..audio.recording_devices() as u16 { println!("Mic {}: {}", i, audio.recording_device_name(i)); } audio.set_recording_device(0)?; // Create audio track for publishing let track = LocalAudioTrack::create_audio_track("mic", audio.rtc_source()); ``` # livekit-api 0.4.22 (2026-05-14) ## Fixes - Proper client SDK - #1081 (@stephen-derosa) # livekit-protocol 0.7.7 (2026-05-14) ## Features - Relocate unrelated types out of `livekit-protocol` # livekit-ffi 0.12.57 (2026-05-14) ## Fixes - feat: add scalability mode for AV1/VP9. - #1076 (@cloudwebrtc) - Add `LIVEKIT_PREFERRED_HW_ENCODER` to prefer `nvenc` or `vaapi` hardware video encoding when both are available. - Reword audio filter logs to be less confusing - #1092 (@1egoman) ### Get WebRTC ADM into Rust - #1037 (@xianshijing-lk) This PR introduces platform audio device management via WebRTC's Audio Device Module (ADM). ### Features - **ADM Proxy**: New `AdmProxy` class that switches between Dummy ADM (synthetic mode) and Platform ADM (real audio I/O) - **PlatformAudio API**: High-level Rust API for microphone capture and speaker playout with AEC/AGC/NS - **Device enumeration**: List and select recording/playout devices by index or GUID - **Mode switching**: Seamlessly switch between synthetic mode (FFI callbacks) and platform mode (native speakers) while audio is active - **FFI platform audio support**: Expose platform audio device enumeration and selection through `livekit-ffi` - **Audio processing**: Configure echo cancellation, noise suppression, and auto gain control with platform-specific defaults (hardware on iOS, software elsewhere) ### Audio Modes | Mode | Recording | Playout | Use Case | |------|-----------|---------|----------| | Synthetic | NativeAudioSource | Dummy ADM + FFI | Unity audio, agents | | Platform | Platform ADM mic | Platform ADM speakers | VoIP with AEC | ### API ```rust // Create PlatformAudio for microphone/speaker access let audio = PlatformAudio::new()?; // Enumerate and select devices for i in 0..audio.recording_devices() as u16 { println!("Mic {}: {}", i, audio.recording_device_name(i)); } audio.set_recording_device(0)?; // Create audio track for publishing let track = LocalAudioTrack::create_audio_track("mic", audio.rtc_source()); ``` # libwebrtc 0.3.33 (2026-05-14) ## Fixes - feat: add scalability mode for AV1/VP9. - #1076 (@cloudwebrtc) - Add `LIVEKIT_PREFERRED_HW_ENCODER` to prefer `nvenc` or `vaapi` hardware video encoding when both are available. - Relocate unrelated types out of `livekit-protocol` ### Get WebRTC ADM into Rust - #1037 (@xianshijing-lk) This PR introduces platform audio device management via WebRTC's Audio Device Module (ADM). ### Features - **ADM Proxy**: New `AdmProxy` class that switches between Dummy ADM (synthetic mode) and Platform ADM (real audio I/O) - **PlatformAudio API**: High-level Rust API for microphone capture and speaker playout with AEC/AGC/NS - **Device enumeration**: List and select recording/playout devices by index or GUID - **Mode switching**: Seamlessly switch between synthetic mode (FFI callbacks) and platform mode (native speakers) while audio is active - **FFI platform audio support**: Expose platform audio device enumeration and selection through `livekit-ffi` - **Audio processing**: Configure echo cancellation, noise suppression, and auto gain control with platform-specific defaults (hardware on iOS, software elsewhere) ### Audio Modes | Mode | Recording | Playout | Use Case | |------|-----------|---------|----------| | Synthetic | NativeAudioSource | Dummy ADM + FFI | Unity audio, agents | | Platform | Platform ADM mic | Platform ADM speakers | VoIP with AEC | ### API ```rust // Create PlatformAudio for microphone/speaker access let audio = PlatformAudio::new()?; // Enumerate and select devices for i in 0..audio.recording_devices() as u16 { println!("Mic {}: {}", i, audio.recording_device_name(i)); } audio.set_recording_device(0)?; // Create audio track for publishing let track = LocalAudioTrack::create_audio_track("mic", audio.rtc_source()); ``` # webrtc-sys 0.3.31 (2026-05-14) ## Fixes - chore: bump libwebrtc version to webrtc-51ef663 - Add `LIVEKIT_PREFERRED_HW_ENCODER` to prefer `nvenc` or `vaapi` hardware video encoding when both are available. ### fix: fix LICENSE.md generation in webrtc build scripts - Add fix_license_json_parsing.patch to handle GN warnings in JSON output - Enable add_licenses.patch for iOS and Android builds (was commented out) - Restore LICENSE.md copy in iOS build script (regression from #1053) The license generation script was failing because `gn desc --format=json` outputs warnings before the JSON when certain build args trigger deprecation notices. The new patch strips non-JSON content before parsing. ### Get WebRTC ADM into Rust - #1037 (@xianshijing-lk) This PR introduces platform audio device management via WebRTC's Audio Device Module (ADM). ### Features - **ADM Proxy**: New `AdmProxy` class that switches between Dummy ADM (synthetic mode) and Platform ADM (real audio I/O) - **PlatformAudio API**: High-level Rust API for microphone capture and speaker playout with AEC/AGC/NS - **Device enumeration**: List and select recording/playout devices by index or GUID - **Mode switching**: Seamlessly switch between synthetic mode (FFI callbacks) and platform mode (native speakers) while audio is active - **FFI platform audio support**: Expose platform audio device enumeration and selection through `livekit-ffi` - **Audio processing**: Configure echo cancellation, noise suppression, and auto gain control with platform-specific defaults (hardware on iOS, software elsewhere) ### Audio Modes | Mode | Recording | Playout | Use Case | |------|-----------|---------|----------| | Synthetic | NativeAudioSource | Dummy ADM + FFI | Unity audio, agents | | Platform | Platform ADM mic | Platform ADM speakers | VoIP with AEC | ### API ```rust // Create PlatformAudio for microphone/speaker access let audio = PlatformAudio::new()?; // Enumerate and select devices for i in 0..audio.recording_devices() as u16 { println!("Mic {}: {}", i, audio.recording_device_name(i)); } audio.set_recording_device(0)?; // Create audio track for publishing let track = LocalAudioTrack::create_audio_track("mic", audio.rtc_source()); ``` Co-authored-by: knope-bot[bot] <152252888+knope-bot[bot]@users.noreply.github.com>

xianshijing-lk added 3 commits April 22, 2026 12:47

get webrtc adm into rust

7e9d580

fix copyright and build

905f6ee

fix the tests where that some of them don't know if it is connected t…

7565045

…o a room and thus failing the audio mode switching

xianshijing-lk requested review from MaxHeimbrock, cloudwebrtc, ladvoc and reenboog April 22, 2026 22:50

ladvoc reviewed Apr 24, 2026

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Comment thread libwebrtc/src/audio_source.rs Outdated

Comment thread livekit/src/rtc_engine/lk_runtime.rs Outdated

Comment thread livekit/src/rtc_engine/lk_runtime.rs Outdated

Comment thread livekit/tests/audio_manager_test.rs Outdated

Switch over to PlatformAudio that supports different AudioSource for …

a2ac932

…different local audio tracks

xianshijing-lk added 2 commits April 27, 2026 16:45

cargo fmt

086935b

added unit tests to the new reqeusts.rs function

0ba69b0

xianshijing-lk force-pushed the sxian/CLT-2765/bring-webrtc-adm-to-rust branch from faf99f6 to 0ba69b0 Compare April 27, 2026 23:54

github-actions Bot and others added 12 commits April 27, 2026 23:54

generated protobuf

924d638

refactor some code and try integrating with Unity

64aadf9

Merge branch 'sxian/CLT-2765/bring-webrtc-adm-to-rust' of https://git…

95d1af1

…hub.com/livekit/rust-sdks into sxian/CLT-2765/bring-webrtc-adm-to-rust

generated protobuf

adb5375

update with the latest changes that make ffi work

7b4389c

Merge branch 'sxian/CLT-2765/bring-webrtc-adm-to-rust' of https://git…

0fdc716

…hub.com/livekit/rust-sdks into sxian/CLT-2765/bring-webrtc-adm-to-rust

generated protobuf

4949561

WebRTC build improvements (to be moved to separate PR)

fe98b5e

Merge branch 'sxian/CLT-2765/bring-webrtc-adm-to-rust' of https://git…

30b1a6b

…hub.com/livekit/rust-sdks into sxian/CLT-2765/bring-webrtc-adm-to-rust

fix the patch

8ecfc49

adding more ffi features to control mute / unmute of recording

11437ee

generated protobuf

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MaxHeimbrock mentioned this pull request May 6, 2026

Integrate the PlatformAudio to unity livekit/client-sdk-unity#268

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xianshijing-lk requested review from 1egoman, alan-george-lk and lukasIO as code owners May 12, 2026 21:44

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xianshijing-lk added 2 commits May 13, 2026 11:56

Merge branch 'main' into sxian/CLT-2765/bring-webrtc-adm-to-rust

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addressed the rest comments

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address the comments

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xianshijing-lk force-pushed the sxian/CLT-2765/bring-webrtc-adm-to-rust branch from 62cc378 to 5ae4d00 Compare May 14, 2026 04:51

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xianshijing-lk commented May 14, 2026

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xianshijing-lk added 8 commits May 14, 2026 08:21

Merge branch 'main' into sxian/CLT-2765/bring-webrtc-adm-to-rust

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fix the tests

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tri fixing the windows build due to tests

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another fix to the unit tests on Linux and Windows

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cargo fmt

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another try to fix the linux and windows bots

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a new way to get a serialized error in Rust for windows tests

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another try to workaround the MSVC exception what() issue on debug mode

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xianshijing-lk merged commit 038c7b9 into main May 14, 2026
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xianshijing-lk deleted the sxian/CLT-2765/bring-webrtc-adm-to-rust branch May 14, 2026 20:27

knope-bot Bot mentioned this pull request May 14, 2026

Release packages #1085

Merged


		### Hybrid Approach

		You can combine both approaches - use `PlatformAudio` for automatic speaker playback while also creating `NativeAudioStream` for audio processing/analysis:

		@@ -136,9 +136,19 @@ export declare enum AudioStreamType {
		*/
		export declare enum AudioSourceType {

Conversation

xianshijing-lk commented Apr 22, 2026 • edited Loading Uh oh! There was an error while loading. Please reload this page.

Uh oh!

Set custom WebRTC build path

Run standalone PlatformAudio tests

Run FFI request handler tests

Start a local LiveKit server first, then:

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github-actions Bot commented Apr 27, 2026 • edited Loading Uh oh! There was an error while loading. Please reload this page.

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MaxHeimbrock May 7, 2026 • edited Loading Uh oh! There was an error while loading. Please reload this page.

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xianshijing-lk commented May 12, 2026

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