Add live audio transcription streaming support to Foundry Local C# SDK by rui-ren · Pull Request #485 · microsoft/Foundry-Local

rui-ren · 2026-03-05T18:29:47Z

Here's the cleaned version:

Description:

Adds real-time audio streaming support to the Foundry Local C# SDK, enabling live microphone-to-text transcription via ONNX Runtime GenAI's StreamingProcessor API (Nemotron ASR).

The existing OpenAIAudioClient only supports file-based transcription. This PR introduces LiveAudioTranscriptionSession that accepts continuous PCM audio chunks (e.g., from a microphone) and returns partial/final transcription results as an async stream.

What's included

New files

src/OpenAI/LiveAudioTranscriptionClient.cs — Streaming session with StartAsync(), AppendAsync(), GetTranscriptionStream(), StopAsync()
src/OpenAI/LiveAudioTranscriptionTypes.cs — LiveAudioTranscriptionResponse (extends AudioCreateTranscriptionResponse) and CoreErrorResponse types
test/FoundryLocal.Tests/LiveAudioTranscriptionTests.cs — Unit tests for deserialization, settings, state guards

Modified files

src/OpenAI/AudioClient.cs — Added CreateLiveTranscriptionSession() factory method
src/Detail/ICoreInterop.cs — Added StreamingRequestBuffer struct, StartAudioStream, PushAudioData, StopAudioStream interface methods
src/Detail/CoreInterop.cs — Routes audio commands through existing execute_command / execute_command_with_binary native entry points
src/Detail/JsonSerializationContext.cs — Registered LiveAudioTranscriptionResponse for AOT compatibility
README.md — Added live audio transcription documentation

API surface

var audioClient = await model.GetAudioClientAsync();
var session = audioClient.CreateLiveTranscriptionSession();

session.Settings.SampleRate = 16000;
session.Settings.Channels = 1;
session.Settings.Language = "en";

await session.StartAsync();

// Push audio from microphone callback
await session.AppendAsync(pcmBytes);

// Read results as async stream
await foreach (var result in session.GetTranscriptionStream())
{
    Console.Write(result.Text);
}

await session.StopAsync();

Design highlights

Output type alignment — LiveAudioTranscriptionResponse extends AudioCreateTranscriptionResponse for consistent output format with file-based transcription
Internal push queue — Bounded Channel<T> serializes audio pushes from any thread (safe for mic callbacks) with backpressure
Fail-fast on errors — Push loop terminates immediately on any native error (no retry logic)
Settings freeze — Audio format settings are snapshot-copied at StartAsync() and immutable during the session
Cancellation-safe stop — StopAsync always calls native stop even if cancelled, preventing native session leaks
Dedicated session CTS — Push loop uses its own CancellationTokenSource, decoupled from the caller's token
Routes through existing exports — StartAudioStream and StopAudioStream route through execute_command; PushAudioData routes through execute_command_with_binary — no new native entry points required

Core integration (neutron-server)

The Core side (AudioStreamingSession.cs) uses StreamingProcessor + Generator + Tokenizer + TokenizerStream from onnxruntime-genai to perform real-time RNNT decoding. The native commands (audio_stream_start/push/stop) are handled as cases in NativeInterop.ExecuteCommandManaged / ExecuteCommandWithBinaryManaged.

Verified working

✅ SDK build succeeds (0 errors, 0 warnings)
✅ Unit tests for JSON deserialization, type inheritance, settings, state guards
✅ GenAI StreamingProcessor pipeline verified with WAV file (correct transcript)
✅ Core TranscribeChunk byte[] PCM path matches reference float[] path exactly
✅ Full E2E simulation: SDK Channel + JSON serialization + session management
✅ Live microphone test: real-time transcription through SDK → Core → GenAI

vercel · 2026-03-05T18:29:52Z

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Project	Deployment	Actions	Updated (UTC)
foundry-local	Error		Mar 28, 2026 3:05am

Copilot

Pull request overview

Adds a new C# SDK API for live/streaming audio transcription sessions (push PCM chunks, receive incremental/final text results) and includes a Windows microphone demo sample.

Changes:

Introduces LiveAudioTranscriptionSession + result/error types for streaming ASR over Core interop.
Extends Core interop to support audio stream start/push/stop (including binary payload routing).
Adds a samples/cs/LiveAudioTranscription demo project and updates the audio client factory API.

Reviewed changes

Copilot reviewed 12 out of 12 changed files in this pull request and generated 9 comments.

Show a summary per file

File	Description
sdk_v2/cs/test/FoundryLocal.Tests/Utils.cs	Replaced prior test utilities with ad-hoc top-level streaming harness code (currently breaks test build).
sdk_v2/cs/test/FoundryLocal.Tests/ModelTests.cs	Adds trailing blank lines (formatting noise).
sdk_v2/cs/src/OpenAI/LiveAudioTranscriptionTypes.cs	Adds `LiveAudioTranscriptionResult` and a structured Core error type.
sdk_v2/cs/src/OpenAI/LiveAudioTranscriptionClient.cs	Adds `LiveAudioTranscriptionSession` implementation (channels, retry, stop semantics).
sdk_v2/cs/src/OpenAI/AudioClient.cs	Adds `CreateLiveTranscriptionSession()` and removes the public file streaming transcription API.
sdk_v2/cs/src/Detail/JsonSerializationContext.cs	Registers new audio streaming types for source-gen JSON.
sdk_v2/cs/src/Detail/ICoreInterop.cs	Adds interop structs + methods for audio stream start/push/stop.
sdk_v2/cs/src/Detail/CoreInterop.cs	Implements binary command routing via `execute_command_with_binary` and start/stop routing via `execute_command`.
sdk_v2/cs/src/AssemblyInfo.cs	Adds `InternalsVisibleTo("AudioStreamTest")`.
samples/cs/LiveAudioTranscription/README.md	Documentation for the live transcription demo sample.
samples/cs/LiveAudioTranscription/Program.cs	Windows microphone demo using NAudio + new session API.
samples/cs/LiveAudioTranscription/LiveAudioTranscription.csproj	Adds sample project dependencies and references the SDK project (path currently incorrect).

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sdk/cs/src/Detail/JsonSerializationContext.cs

samples/cs/LiveAudioTranscription/LiveAudioTranscription.csproj

sdk/cs/test/FoundryLocal.Tests/ModelTests.cs

sdk_v2/cs/test/FoundryLocal.Tests/Utils.cs

sdk/cs/src/OpenAI/AudioClient.cs

sdk/cs/src/OpenAI/LiveAudioTranscriptionClient.cs

sdk_v2/cs/src/OpenAI/LiveAudioTranscriptionClient.cs

sdk_v2/cs/src/AssemblyInfo.cs

sdk_v2/cs/test/FoundryLocal.Tests/Utils.cs

samples/cs/LiveAudioTranscription/LiveAudioTranscription.csproj

…g-support-sdk # Conflicts: # sdk/js/test/openai/chatClient.test.ts

samples/cs/GettingStarted/src/LiveAudioTranscriptionExample/Program.cs

sdk/cs/src/OpenAI/LiveAudioTranscriptionClient.cs

sdk/cs/src/Microsoft.AI.Foundry.Local.csproj

…ionItem pattern (#561) ### Description Redesigns `LiveAudioTranscriptionResponse` to follow the OpenAI Realtime API's `ConversationItem` shape, enabling forward compatibility with a future WebSocket-based architecture. **Motivation:** - Customers using OpenAI's Realtime API access transcription via `result.content[0].transcript` - By adopting this pattern now, customers who write `result.Content[0].Text` won't need to change their code when we migrate to WebSocket transport - Aligns with the team's plan to move toward OpenAI Realtime API compatibility **Before:** ```csharp // Extended AudioCreateTranscriptionResponse from Betalgo await foreach (var result in session.GetTranscriptionStream()) { Console.Write(result.Text); // inherited from base bool final = result.IsFinal; // custom field var segments = result.Segments; // inherited from base } ``` **After:** ```csharp // Own type shaped like OpenAI Realtime ConversationItem await foreach (var result in session.GetTranscriptionStream()) { Console.Write(result.Content[0].Text); // ConversationItem pattern Console.Write(result.Content[0].Transcript); // alias for Text (Realtime compat) bool final = result.IsFinal; double? start = result.StartTime; } ``` **Changes:** | File | Change | |------|--------| | LiveAudioTranscriptionTypes.cs | Removed `AudioCreateTranscriptionResponse` inheritance. New standalone `LiveAudioTranscriptionResponse` with `Content` list + new `TranscriptionContentPart` type | | LiveAudioTranscriptionClient.cs | Updated text checks: `.Text` → `.Content?[0]?.Text` | | JsonSerializationContext.cs | Registered `TranscriptionContentPart`, removed `AudioCreateTranscriptionResponse.Segment` | | LiveAudioTranscriptionTests.cs | Updated assertions to match new type shape | | Program.cs (sample) | Updated result reading to `result.Content?[0]?.Text` | | README.md | Updated docs and output type table | **Key design decisions:** - `TranscriptionContentPart` has both `Text` and `Transcript` (set to the same value) for maximum compatibility with both Whisper and Realtime API patterns - `StartTime`/`EndTime` are top-level on the response (not nested in Segments) — simpler access, maps to Realtime's `audio_start_ms`/`audio_end_ms` - No dependency on Betalgo's `ConversationItem` — we own the type to avoid carrying unused chat/tool-calling fields - `LiveAudioTranscriptionRaw` (Core JSON deserialization) is unchanged — this is purely an SDK presentation change, no Core/neutron-server impact **No breaking changes to:** Core API, native interop, audio pipeline, session lifecycle --------- Co-authored-by: ruiren_microsoft <ruiren@microsoft.com>

kunal-vaishnavi · 2026-03-28T03:13:15Z

sdk/cs/README.md

+
+For real-time microphone-to-text transcription, use `CreateLiveTranscriptionSession()`. Audio is pushed as raw PCM chunks and transcription results stream back as an `IAsyncEnumerable`.
+
+The streaming result type (`LiveAudioTranscriptionResponse`) extends `AudioCreateTranscriptionResponse` from the Betalgo OpenAI SDK, so it's compatible with the file-based transcription output format while adding streaming-specific fields.


This line needs to be updated as it does not extend that class anymore.

kunal-vaishnavi · 2026-03-28T03:19:43Z

sdk/js/test/openai/chatClient.test.ts

            let response = await client.completeChat(messages, tools);

-            // Check that a tool call was generated
+            // Check response is valid


These unit tests are specifically for tool calling. Ex:

Foundry-Local/sdk/js/test/openai/chatClient.test.ts

Lines 185 to 224 in b247611

it('should perform tool calling chat completion (non-streaming)', async function() {

this.timeout(20000);

const manager = getTestManager();

const catalog = manager.catalog;

const cachedModels = await catalog.getCachedModels();

expect(cachedModels.length).to.be.greaterThan(0);

const cachedVariant = cachedModels.find(m => m.alias === TEST_MODEL_ALIAS);

expect(cachedVariant).to.not.be.undefined;

const model = await catalog.getModel(TEST_MODEL_ALIAS);

expect(model).to.not.be.undefined;

if (!cachedVariant) return;

model.selectVariant(cachedVariant);

await model.load();

try {

const client = model.createChatClient();

client.settings.maxTokens = 500;

client.settings.temperature = 0.0;

client.settings.toolChoice = { type: 'required' }; // Force the model to make a tool call

// Prepare messages and tools

const messages: any[] = [

{ role: 'system', content: 'You are a helpful AI assistant. If necessary, you can use any provided tools to answer the question.' },

{ role: 'user', content: 'What is the answer to 7 multiplied by 6?' }

];

const tools: any[] = [getMultiplyTool()];

// Start the conversation

let response = await client.completeChat(messages, tools);

// Check that a tool call was generated

expect(response).to.not.be.undefined;

expect(response.choices).to.be.an('array').with.length.greaterThan(0);

expect(response.choices[0].finish_reason).to.equal('tool_calls');

expect(response.choices[0].message).to.not.be.undefined;

expect(response.choices[0].message.tool_calls).to.be.an('array').with.length.greaterThan(0);

These tests should not be modified. If they are failing on the PR, then there is a true failure somewhere that needs to be resolved. These tests are already passing on the main branch.

kunal-vaishnavi · 2026-03-28T03:20:07Z

sdk/js/test/openai/chatClient.test.ts

-                const content = chunk.choices?.[0]?.message?.content ?? chunk.choices?.[0]?.delta?.content;
-                if (content) {
-                    fullResponse += content;
+            // The model may either call the tool or respond directly.


Same comment as here

kunal-vaishnavi · 2026-03-28T03:20:31Z

sdk/rust/tests/integration/chat_client_test.rs

-                "arguments": tool_call.function.arguments,
+        .is_some_and(|tc| !tc.is_empty());
+
+    if has_tool_calls {


Same comment as here

kunal-vaishnavi · 2026-03-28T03:20:42Z

sdk/rust/tests/integration/chat_client_test.rs

-                "name": tool_call_name,
-                "arguments": tool_call_args
+    // The model may either call the tool or respond directly.
+    if !tool_call_name.is_empty() {


Same comment as here

kunal-vaishnavi · 2026-03-28T03:24:21Z

sdk/cs/test/FoundryLocal.Tests/LiveAudioTranscriptionTests.cs

+internal sealed class LiveAudioTranscriptionTests
+{
+    // --- LiveAudioTranscriptionResponse.FromJson tests ---
+


Can we add an E2E test here that takes in pre-saved bytes, appends them to a session, transcribes the result, and checks the result object's attributes?

kunal-vaishnavi · 2026-03-28T03:26:21Z

sdk/rust/build.rs

    let is_linux = rid.starts_with("linux");

-    let core_version = if nightly {
-        resolve_latest_version("Microsoft.AI.Foundry.Local.Core", ORT_NIGHTLY_FEED)


Why do we need to make changes to the Rust SDK for getting a package version? I believe the only changes needed in this file are the ORT GenAI version used and the feed URLs for any packages. The remaining changes can be undone.

support audio streaming-csharp

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rui-ren changed the title ~~Add real-time audio streaming support (Microphone ASR) - c#~~ Add live audio transcription streaming support to Foundry Local C# SDK Mar 13, 2026

fix: add missing using directives for EnumeratorCancellation and Channel

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		For real-time microphone-to-text transcription, use `CreateLiveTranscriptionSession()`. Audio is pushed as raw PCM chunks and transcription results stream back as an `IAsyncEnumerable`.

		The streaming result type (`LiveAudioTranscriptionResponse`) extends `AudioCreateTranscriptionResponse` from the Betalgo OpenAI SDK, so it's compatible with the file-based transcription output format while adding streaming-specific fields.

	it('should perform tool calling chat completion (non-streaming)', async function() {
	this.timeout(20000);
	const manager = getTestManager();
	const catalog = manager.catalog;

	const cachedModels = await catalog.getCachedModels();
	expect(cachedModels.length).to.be.greaterThan(0);

	const cachedVariant = cachedModels.find(m => m.alias === TEST_MODEL_ALIAS);
	expect(cachedVariant).to.not.be.undefined;

	const model = await catalog.getModel(TEST_MODEL_ALIAS);
	expect(model).to.not.be.undefined;
	if (!cachedVariant) return;

	model.selectVariant(cachedVariant);
	await model.load();

	try {
	const client = model.createChatClient();
	client.settings.maxTokens = 500;
	client.settings.temperature = 0.0;
	client.settings.toolChoice = { type: 'required' }; // Force the model to make a tool call

	// Prepare messages and tools
	const messages: any[] = [
	{ role: 'system', content: 'You are a helpful AI assistant. If necessary, you can use any provided tools to answer the question.' },
	{ role: 'user', content: 'What is the answer to 7 multiplied by 6?' }
	];
	const tools: any[] = [getMultiplyTool()];

	// Start the conversation
	let response = await client.completeChat(messages, tools);

	// Check that a tool call was generated
	expect(response).to.not.be.undefined;
	expect(response.choices).to.be.an('array').with.length.greaterThan(0);
	expect(response.choices[0].finish_reason).to.equal('tool_calls');
	expect(response.choices[0].message).to.not.be.undefined;
	expect(response.choices[0].message.tool_calls).to.be.an('array').with.length.greaterThan(0);

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Description:

What's included

API surface

Design highlights

Core integration (neutron-server)

Verified working

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rui-ren commented Mar 5, 2026 •

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