Synchronize real-time audio API names

.gitignore

@@ -38,3 +38,6 @@ logs/
 
 # Local NuGet packages built from source
 local-packages/
+
+# JS SDK downloaded native core binaries
+sdk/js/foundry-local-core/

samples/cs/live-audio-transcription/Program.cs

@@ -52,7 +52,7 @@ await model.DownloadAsync(progress =>
 {
     try
     {
-        await foreach (var result in session.GetTranscriptionStream())
+        await foreach (var result in session.GetStream())
         {
             var text = result.Content?[0]?.Text;
             if (result.IsFinal)

samples/cs/live-audio-transcription/README.md

@@ -36,7 +36,7 @@ dotnet run -- --synth
 2. Creates a `LiveAudioTranscriptionSession` with 16kHz/16-bit/mono PCM settings
 3. Captures microphone audio via `NAudio.WaveInEvent` (or generates synthetic audio as fallback)
 4. Pushes PCM chunks to the SDK via `session.AppendAsync()` through a bounded channel for backpressure
-5. Reads transcription results via `await foreach (var result in session.GetTranscriptionStream())`
+5. Reads transcription results via `await foreach (var result in session.GetStream())`
 6. Access text via `result.Content[0].Text` (OpenAI Realtime ConversationItem pattern)
 
 ## API
@@ -54,7 +54,7 @@ await session.StartAsync();
 await session.AppendAsync(pcmBytes);
 
 // Read results
-await foreach (var result in session.GetTranscriptionStream())
+await foreach (var result in session.GetStream())
 {
     Console.WriteLine(result.Content[0].Text);       // transcribed text
     Console.WriteLine(result.Content[0].Transcript); // alias (OpenAI compat)

samples/js/live-audio-transcription/README.md

@@ -30,7 +30,7 @@ Speak into your microphone. Transcription appears in real-time. Press `Ctrl+C` t
 2. Creates a `LiveAudioTranscriptionSession` with 16kHz/16-bit/mono PCM settings
 3. Captures microphone audio via `naudiodon2` (or generates synthetic audio as fallback)
 4. Pushes PCM chunks to the SDK via `session.append()`
-5. Reads transcription results via `for await (const result of session.getTranscriptionStream())`
+5. Reads transcription results via `for await (const result of session.getStream())`
 6. Access text via `result.content[0].text` (OpenAI Realtime ConversationItem pattern)
 
 ## API
@@ -48,7 +48,7 @@ await session.start();
 await session.append(pcmBytes);
 
 // Read results
-for await (const result of session.getTranscriptionStream()) {
+for await (const result of session.getStream()) {
     console.log(result.content[0].text);       // transcribed text
     console.log(result.content[0].transcript); // alias (OpenAI compat)
     console.log(result.is_final);              // true for final results

samples/js/live-audio-transcription/app.js

@@ -15,7 +15,7 @@ console.log();
 // Initialize the Foundry Local SDK
 console.log('Initializing Foundry Local SDK...');
 const manager = FoundryLocalManager.create({
-    appName: 'foundry',
+    appName: 'foundry_local_samples',
     logLevel: 'info'
 });
 console.log('✓ SDK initialized');
@@ -55,7 +55,7 @@ console.log('✓ Session started');
 // Read transcription results in background
 const readPromise = (async () => {
     try {
-        for await (const result of session.getTranscriptionStream()) {
+        for await (const result of session.getStream()) {
             const text = result.content?.[0]?.text;
             if (!text) continue;
 

samples/python/live-audio-transcription/README.md

@@ -41,7 +41,7 @@ python src/app.py --synth
 2. Creates a `LiveAudioTranscriptionSession` with 16kHz/16-bit/mono PCM settings
 3. Captures microphone audio via `pyaudio` (or generates synthetic audio as fallback)
 4. Pushes PCM chunks to the SDK via `session.append()`
-5. Reads transcription results in a background thread via `for result in session.get_transcription_stream()`
+5. Reads transcription results in a background thread via `for result in session.get_stream()`
 6. Access text via `result.content[0].text` (OpenAI Realtime ConversationItem pattern)
 
 ## API
@@ -59,7 +59,7 @@ session.start()
 session.append(pcm_bytes)
 
 # Read results (typically on a background thread)
-for result in session.get_transcription_stream():
+for result in session.get_stream():
     print(result.content[0].text)        # transcribed text
     print(result.content[0].transcript)  # alias (OpenAI compat)
     print(result.is_final)               # True for final results

samples/python/live-audio-transcription/src/app.py

@@ -53,7 +53,7 @@
 # --- Background thread reads transcription results (mirrors JS readPromise) ---
 
 def read_results():
-    for result in session.get_transcription_stream():
+    for result in session.get_stream():
         text = result.content[0].text if result.content else ""
         if result.is_final:
             print()

samples/rust/live-audio-transcription/src/main.rs

@@ -62,7 +62,7 @@ async fn main() -> Result<(), Box<dyn std::error::Error>> {
     println!("✓ Session started\n");
 
     // --- Background task reads transcription results (mirrors JS readPromise) ---
-    let mut stream = session.get_transcription_stream().await?;
+    let mut stream = session.get_stream().await?;
     let read_task = tokio::spawn(async move {
         while let Some(result) = stream.next().await {
             match result {

sdk/cpp/CMakeLists.txt

@@ -55,7 +55,7 @@ add_library(CppSdk STATIC
   src/openai_chat_client.cpp
   src/openai_audio_client.cpp
   src/openai_live_audio_types.cpp
-  src/openai_live_audio_client.cpp
+  src/openai_live_audio_session.cpp
   src/foundry_local_manager.cpp
 )
 

sdk/cpp/include/foundry_local.h

@@ -17,4 +17,4 @@
 #include "openai/openai_chat_client.h"
 #include "openai/openai_audio_client.h"
 #include "openai/openai_live_audio_types.h"
-#include "openai/openai_live_audio_client.h"
+#include "openai/openai_live_audio_session.h"

sdk/cpp/src/openai_audio_client.cpp

@@ -16,7 +16,7 @@
 #include "core_helpers.h"
 #include "logger.h"
 
-#include "openai/openai_live_audio_client.h"
+#include "openai/openai_live_audio_session.h"
 
 namespace foundry_local {
 

sdk/cpp/src/openai_live_audio_client.cpp → sdk/cpp/src/openai_live_audio_session.cpp

@@ -10,7 +10,7 @@
 
 #include <nlohmann/json.hpp>
 
-#include "openai/openai_live_audio_client.h"
+#include "openai/openai_live_audio_session.h"
 #include "openai/openai_live_audio_types.h"
 #include "foundry_local_internal_core.h"
 #include "foundry_local_exception.h"

sdk/cpp/test/live_audio_test.cpp

@@ -8,7 +8,7 @@
 #include "foundry_local_exception.h"
 
 #include "openai/openai_live_audio_types.h"
-#include "openai/openai_live_audio_client.h"
+#include "openai/openai_live_audio_session.h"
 
 #include <nlohmann/json.hpp>
 

sdk/cs/README.md

@@ -312,7 +312,7 @@ waveIn.DataAvailable += (sender, e) =>
 };
 
 // Read transcription results as they arrive
-await foreach (var result in session.GetTranscriptionStream())
+await foreach (var result in session.GetStream())
 {
     // result follows the OpenAI Realtime ConversationItem pattern:
     // - result.Content[0].Text       — incremental transcribed text (per chunk, not accumulated)
@@ -341,7 +341,7 @@ await session.StopAsync();
 |--------|-------------|
 | `StartAsync()` | Initialize the streaming session. Settings are frozen after this call. |
 | `AppendAsync(pcmData)` | Push a chunk of raw PCM audio. Thread-safe (bounded internal queue). |
-| `GetTranscriptionStream()` | Async enumerable of transcription results. |
+| `GetStream()` | Async enumerable of transcription results. |
 | `StopAsync()` | Signal end-of-audio, flush remaining audio, and clean up. |
 | `DisposeAsync()` | Calls `StopAsync` if needed. Use `await using` for automatic cleanup. |
 

sdk/cs/src/OpenAI/LiveAudioTranscriptionClient.cs → sdk/cs/src/OpenAI/LiveAudioSession.cs

@@ -37,7 +37,7 @@ public sealed class LiveAudioTranscriptionSession : IAsyncDisposable
     private bool _started;
     private bool _stopped;
 
-    // Output channel: native callback writes, user reads via GetTranscriptionStream
+    // Output channel: native callback writes, user reads via GetStream
     private Channel<LiveAudioTranscriptionResponse>? _outputChannel;
 
     // Internal push queue: user writes audio chunks, background loop drains to native core.
@@ -90,7 +90,7 @@ internal LiveAudioTranscriptionSession(string modelId)
 
     /// <summary>
     /// Start a real-time audio streaming session.
-    /// Must be called before <see cref="AppendAsync"/> or <see cref="GetTranscriptionStream"/>.
+    /// Must be called before <see cref="AppendAsync"/> or <see cref="GetStream"/>.
     /// Settings are frozen after this call.
     /// </summary>
     /// <param name="ct">Cancellation token.</param>
@@ -249,7 +249,7 @@ private async Task PushLoopAsync(CancellationToken ct)
     /// </summary>
     /// <param name="ct">Cancellation token.</param>
     /// <returns>Async enumerable of transcription results.</returns>
-    public async IAsyncEnumerable<LiveAudioTranscriptionResponse> GetTranscriptionStream(
+    public async IAsyncEnumerable<LiveAudioTranscriptionResponse> GetStream(
         [EnumeratorCancellation] CancellationToken ct = default)
     {
         if (_outputChannel == null)
@@ -266,7 +266,7 @@ public async IAsyncEnumerable<LiveAudioTranscriptionResponse> GetTranscriptionSt
     /// <summary>
     /// Signal end-of-audio and stop the streaming session.
     /// Any remaining buffered audio in the push queue will be drained to native core first.
-    /// Final results are delivered through <see cref="GetTranscriptionStream"/> before it completes.
+    /// Final results are delivered through <see cref="GetStream"/> before it completes.
     /// </summary>
     /// <param name="ct">Cancellation token.</param>
     public async Task StopAsync(CancellationToken ct = default)

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

@@ -154,14 +154,14 @@ public async Task AppendAsync_BeforeStart_Throws()
 
     [Test]
     [SkipUnlessIntegration]
-    public async Task GetTranscriptionStream_BeforeStart_Throws()
+    public async Task GetStream_BeforeStart_Throws()
     {
         await using var session = new LiveAudioTranscriptionSession("test-model");
 
         FoundryLocalException? caught = null;
         try
         {
-            await foreach (var _ in session.GetTranscriptionStream())
+            await foreach (var _ in session.GetStream())
             {
                 // should not reach here
             }
@@ -212,7 +212,7 @@ public async Task LiveStreaming_E2E_WithSyntheticPCM_ReturnsValidResponse()
             var results = new List<LiveAudioTranscriptionResponse>();
             var readTask = Task.Run(async () =>
             {
-                await foreach (var result in session.GetTranscriptionStream())
+                await foreach (var result in session.GetStream())
                 {
                     results.Add(result);
                 }

sdk/js/src/detail/model.ts

@@ -3,7 +3,7 @@ import { ChatClient } from '../openai/chatClient.js';
 import { AudioClient } from '../openai/audioClient.js';
 import { EmbeddingClient } from '../openai/embeddingClient.js';
 import { ResponsesClient } from '../openai/responsesClient.js';
-import { LiveAudioTranscriptionSession } from '../openai/liveAudioTranscriptionClient.js';
+import { LiveAudioTranscriptionSession } from '../openai/liveAudioSession.js';
 import { IModel } from '../imodel.js';
 import { ModelInfo } from '../types.js';
 

sdk/js/src/detail/modelVariant.ts

@@ -4,7 +4,7 @@ import { ModelInfo } from '../types.js';
 import { ChatClient } from '../openai/chatClient.js';
 import { AudioClient } from '../openai/audioClient.js';
 import { EmbeddingClient } from '../openai/embeddingClient.js';
-import { LiveAudioTranscriptionSession } from '../openai/liveAudioTranscriptionClient.js';
+import { LiveAudioTranscriptionSession } from '../openai/liveAudioSession.js';
 import { ResponsesClient } from '../openai/responsesClient.js';
 import { IModel } from '../imodel.js';
 

sdk/js/src/imodel.ts

@@ -1,7 +1,6 @@
 import { ChatClient } from './openai/chatClient.js';
 import { AudioClient } from './openai/audioClient.js';
 import { EmbeddingClient } from './openai/embeddingClient.js';
-import { LiveAudioTranscriptionSession } from './openai/liveAudioTranscriptionClient.js';
 import { ResponsesClient } from './openai/responsesClient.js';
 import { ModelInfo } from './types.js';
 
@@ -28,12 +27,6 @@ export interface IModel {
     createAudioClient(): AudioClient;
     createEmbeddingClient(): EmbeddingClient;
 
-    /**
-     * Creates a LiveAudioTranscriptionSession for real-time audio streaming ASR.
-     * The model must be loaded before calling this method.
-     * @returns A LiveAudioTranscriptionSession instance.
-     */
-    createLiveTranscriptionSession(): LiveAudioTranscriptionSession;
     /**
      * Creates a ResponsesClient for interacting with the model via the Responses API.
      * Unlike createChatClient/createAudioClient (which use FFI), the Responses API

sdk/js/src/index.ts

@@ -9,8 +9,8 @@ export type { IModel } from './imodel.js';
 export { ChatClient, ChatClientSettings } from './openai/chatClient.js';
 export { AudioClient, AudioClientSettings } from './openai/audioClient.js';
 export { EmbeddingClient } from './openai/embeddingClient.js';
-export { LiveAudioTranscriptionSession, LiveAudioTranscriptionOptions } from './openai/liveAudioTranscriptionClient.js';
-export type { LiveAudioTranscriptionResponse, TranscriptionContentPart } from './openai/liveAudioTranscriptionTypes.js';
+export { LiveAudioTranscriptionSession, LiveAudioTranscriptionOptions } from './openai/liveAudioSession.js';
+export type { LiveAudioTranscriptionResponse, TranscriptionContentPart } from './openai/liveAudioTypes.js';
 export { ResponsesClient, ResponsesClientSettings, getOutputText } from './openai/responsesClient.js';
 export { ModelLoadManager } from './detail/modelLoadManager.js';
 /** @internal */

sdk/js/src/openai/audioClient.ts

@@ -1,5 +1,5 @@
 import { CoreInterop } from '../detail/coreInterop.js';
-import { LiveAudioTranscriptionSession } from './liveAudioTranscriptionClient.js';
+import { LiveAudioTranscriptionSession } from './liveAudioSession.js';
 
 export class AudioClientSettings {
     language?: string;

sdk/js/src/openai/liveAudioTranscriptionClient.ts → sdk/js/src/openai/liveAudioSession.ts

@@ -1,5 +1,5 @@
 import { CoreInterop } from '../detail/coreInterop.js';
-import { LiveAudioTranscriptionResponse, parseTranscriptionResult, tryParseCoreError } from './liveAudioTranscriptionTypes.js';
+import { LiveAudioTranscriptionResponse, parseTranscriptionResult, tryParseCoreError } from './liveAudioTypes.js';
 
 /**
  * Audio format settings for a streaming session.
@@ -191,7 +191,7 @@ export class LiveAudioTranscriptionSession {
 
     /**
      * Start a real-time audio streaming session.
-     * Must be called before append() or getTranscriptionStream().
+     * Must be called before append() or getStream().
      * Settings are frozen after this call.
      */
     public async start(): Promise<void> {
@@ -319,17 +319,17 @@ export class LiveAudioTranscriptionSession {
      *
      * Usage:
      * ```ts
-     * for await (const result of client.getTranscriptionStream()) {
+     * for await (const result of client.getStream()) {
      *     console.log(result.content[0].text);
      * }
      * ```
      */
-    public async *getTranscriptionStream(): AsyncGenerator<LiveAudioTranscriptionResponse> {
+    public async *getStream(): AsyncGenerator<LiveAudioTranscriptionResponse> {
         if (!this.outputQueue) {
             throw new Error('No active streaming session. Call start() first.');
         }
         if (this.streamConsumed) {
-            throw new Error('getTranscriptionStream() can only be called once per session. The output stream has already been consumed.');
+            throw new Error('getStream() can only be called once per session. The output stream has already been consumed.');
         }
         this.streamConsumed = true;
 
@@ -341,7 +341,7 @@ export class LiveAudioTranscriptionSession {
     /**
      * Signal end-of-audio and stop the streaming session.
      * Any remaining buffered audio in the push queue will be drained to native core first.
-     * Final results are delivered through getTranscriptionStream() before it completes.
+     * Final results are delivered through getStream() before it completes.
      */
     public async stop(): Promise<void> {
         if (!this.started || this.stopped) {

sdk/js/test/openai/liveAudioTranscription.test.ts

@@ -1,7 +1,7 @@
 import { describe, it } from 'mocha';
 import { expect } from 'chai';
-import { parseTranscriptionResult, tryParseCoreError } from '../../src/openai/liveAudioTranscriptionTypes.js';
-import { LiveAudioTranscriptionOptions } from '../../src/openai/liveAudioTranscriptionClient.js';
+import { parseTranscriptionResult, tryParseCoreError } from '../../src/openai/liveAudioTypes.js';
+import { LiveAudioTranscriptionOptions } from '../../src/openai/liveAudioSession.js';
 import { getTestManager } from '../testUtils.js';
 
 describe('Live Audio Transcription Types', () => {
@@ -160,7 +160,7 @@ describe('Live Audio Transcription Types', () => {
                 // Collect results in background (must start before pushing audio)
                 const results: any[] = [];
                 const readPromise = (async () => {
-                    for await (const result of session.getTranscriptionStream()) {
+                    for await (const result of session.getStream()) {
                         results.push(result);
                     }
                 })();

sdk/python/src/openai/__init__.py

@@ -7,8 +7,8 @@
 from .chat_client import ChatClient, ChatClientSettings
 from .audio_client import AudioClient
 from .embedding_client import EmbeddingClient
-from .live_audio_transcription_client import LiveAudioTranscriptionSession
-from .live_audio_transcription_types import (
+from .live_audio_session import LiveAudioTranscriptionSession
+from .live_audio_types import (
     CoreErrorResponse,
     LiveAudioTranscriptionOptions,
     LiveAudioTranscriptionResponse,

sdk/python/src/openai/audio_client.py

@@ -14,7 +14,7 @@
 
 from ..detail.core_interop import CoreInterop, InteropRequest
 from ..exception import FoundryLocalException
-from .live_audio_transcription_client import LiveAudioTranscriptionSession
+from .live_audio_session import LiveAudioTranscriptionSession
 
 logger = logging.getLogger(__name__)
 
@@ -76,7 +76,7 @@ def create_live_transcription_session(self) -> LiveAudioTranscriptionSession:
                     session.settings.sample_rate = 16000
                     session.start()
                     session.append(pcm_bytes)
-                    for result in session.get_transcription_stream():
+                    for result in session.get_stream():
                         print(result.content[0].text)
         """
         return LiveAudioTranscriptionSession(self.model_id, self._core_interop)