@speech-sdk/core 0.6.0 → 0.6.2

package/README.md

@@ -42,6 +42,23 @@ result.audio.base64;      // string (lazy-computed)
 result.audio.mediaType;   // "audio/mpeg"
 ```
 
+### Volume normalization
+
+Pass `volumeDbfs` to RMS-normalize the output to an absolute target loudness (must be ≤ 0; lower is quieter; -20 is the broadcast/podcast voice convention with ~20 dB of peak headroom):
+
+```ts
+const result = await generateSpeech({
+  model: 'openai/gpt-4o-mini-tts',
+  text: 'Hello from speech-sdk!',
+  voice: 'alloy',
+  volumeDbfs: -20,
+});
+
+result.audio.mediaType;   // "audio/wav" — re-encoded after normalization
+```
+
+When `volumeDbfs` is set the SDK transparently asks the provider for its decodable PCM/WAV mode, normalizes the samples, and returns 16-bit mono WAV — so the response `mediaType` switches to `audio/wav` regardless of the provider's native default. Throws `VolumeAdjustmentUnsupportedError` if the provider has no decodable output mode.
+
 ## Streaming
 
 Use `streamSpeech()` instead of `generateSpeech()` to receive audio bytes incrementally as the provider produces them. The result's `audio` field is a standard `ReadableStream<Uint8Array>` that works in Node, Edge runtimes, and browsers.
@@ -142,43 +159,15 @@ result.audio.mediaType;   // "audio/wav"
 
 The return type is the standard `SpeechResult`, so it composes with everything else in the SDK.
 
-### Try it — listen to the difference
-
-The same four-provider conversation rendered two ways. The raw version exposes the natural mismatch between providers (Hume Octave is noticeably quieter than ElevenLabs or OpenAI); the normalized version (the default) levels every voice to a fixed −20 dBFS RMS target — the broadcast/podcast voice convention.
-
-| Sample | Audio |
-|---|---|
-| **Cross-provider stitch** (OpenAI + ElevenLabs) | <a href="./assets/audio/conversation/cross-provider-stitch.mp3">▶ Listen</a> |
-| **Four-provider stitch — raw** (`normalizeVolume: false`) | <a href="./assets/audio/conversation/four-providers-raw.mp3">▶ Listen</a> |
-| **Four-provider stitch — normalized** (default) | <a href="./assets/audio/conversation/four-providers-normalized.mp3">▶ Listen</a> |
-
-> The README renders these as inline audio players when viewed on GitHub. If your viewer doesn't support inline playback, click "Listen" to download the MP3.
-
-<details>
-<summary>Inline players</summary>
-
-Cross-provider stitch:
-
-<audio controls src="./assets/audio/conversation/cross-provider-stitch.mp3"></audio>
-
-Four-provider stitch — raw (no normalization):
-
-<audio controls src="./assets/audio/conversation/four-providers-raw.mp3"></audio>
-
-Four-provider stitch — normalized (default):
-
-<audio controls src="./assets/audio/conversation/four-providers-normalized.mp3"></audio>
-
-</details>
-
 ### Conversation options
 
 ```ts
 generateConversation({
   model?: string | ResolvedModel,                 // default model for all turns
-  turns: ConversationTurn[],                      // 1..N turns; up to 4 unique voices
+  turns: ConversationTurn[],                      // 1..N turns; any number of unique voices
   gapMs?: number,                                 // silence between turns (stitch path), default 300
-  normalizeVolume?: boolean,                      // RMS-level stitched turns, default true
+  normalizeVolume?: boolean,                      // RMS-level the output, default true
+  volumeDbfs?: number,                            // RMS target loudness in dBFS (≤0), default -20
   maxConcurrency?: number,                        // cap parallel generateSpeech calls, default 6
   maxRetries?: number,                            // per-turn retries, default 2
   apiKey?: string,
@@ -191,28 +180,36 @@ interface ConversationTurn {
   voice: Voice;                                   // required
   text: string;                                   // required, non-empty
   model?: string | ResolvedModel;                 // per-turn override of the top-level model
-  providerOptions?: Record<string, unknown>;
+  providerOptions?: Record<string, unknown>,      // stitch path only; see note below
 }
 ```
 
+Per-turn `providerOptions` are merged with the top-level `providerOptions` on the stitch path — each turn's underlying `generateSpeech()` call receives `{ ...topLevel, ...turn, ...stitchDefaults }`. On the native-dialogue path the provider renders the whole script in one API call, so per-turn overrides have no well-defined meaning; setting `providerOptions` on any turn throws `ConversationInputError`. Move the options to the top-level `providerOptions` (forwarded once to the dialogue call) instead.
+
 ### Volume normalization
 
-When the stitch path runs, `normalizeVolume: true` (the default) RMS-normalizes each per-turn segment to a fixed **−20 dBFS** RMS target — the broadcast/podcast voice convention, with ~20 dB peak headroom so typical TTS speech doesn't clip after gain. The target is absolute, not relative, so:
+`normalizeVolume: true` (the default) RMS-normalizes the output to an absolute target loudness — broadcast/podcast voice convention — so two `generateConversation` calls produce comparable levels regardless of provider mix or content. The target defaults to **−20 dBFS** (~20 dB of peak headroom), and is configurable via `volumeDbfs` (must be ≤ 0; lower is quieter).
 
-- Two `generateConversation` calls produce comparable loudness even with completely different content — you can play them back-to-back without adjusting volume.
-- Each segment is normalized independently — no cross-segment dependency, just two O(N) passes over the int16 PCM samples per segment.
+```ts
+await generateConversation({
+  turns: [...],
+  volumeDbfs: -16,           // a touch louder than the default
+});
+```
 
-Pass `normalizeVolume: false` to skip the step entirely (zero work) when you want raw provider levels.
+Normalization runs on **both paths** — stitched multi-provider conversations and single-provider native dialogue. On the native path the SDK transparently asks the provider for its decodable PCM/WAV mode (via `getStitchOptions`), levels the result, and re-encodes as 16-bit mono WAV — so the response `mediaType` becomes `audio/wav` whenever normalization runs. If a native dialogue provider can't emit decodable audio, the request still succeeds but a `warning` is appended explaining that volume normalization was skipped.
+
+Pass `normalizeVolume: false` to skip normalization entirely (zero work) and keep the raw provider audio bytes and `mediaType` untouched.
 
 ### Errors
 
-Conversation-specific errors (importable from `@speech-sdk/core/conversation/errors`):
+Conversation-specific errors (re-exported from `@speech-sdk/core/conversation` alongside `generateConversation`, or importable on their own from `@speech-sdk/core/conversation/errors`):
 
 | Error | When |
 |---|---|
-| `ConversationInputError` | Validation failure — empty turns, blank text, more than 4 unique voices, or a turn missing a model |
+| `ConversationInputError` | Validation failure — empty turns, blank text, or a turn missing a model |
 | `DialogueConstraintError` | A native-dialogue provider was selected but the conversation violates its constraints (e.g. 3 voices on Gemini, which requires exactly 2) |
-| `StitchUnsupportedError` | The stitch path was selected but a chosen provider/model can't emit PCM/WAV (currently `unreal-speech`, `fal-ai`, `mistral`) |
+| `StitchUnsupportedError` | The stitch path was selected but a chosen provider/model can't emit PCM/WAV |
 
 ### Native dialogue caps
 
@@ -224,8 +221,6 @@ Conversation-specific errors (importable from `@speech-sdk/core/conversation/err
 | Fish Audio | `s2-pro` | 1–4 voices |
 | fal | `dia-tts` | 1–2 voices |
 
-Across the SDK, conversations are capped at **4 unique voices** total regardless of provider.
-
 ## Supported Providers
 
 Use `provider/model` strings. Passing just the provider name uses its default model.
@@ -240,7 +235,6 @@ Use `provider/model` strings. Passing just the provider name uses its default mo
 | [Inworld](https://docs.inworld.ai/tts) | `inworld` | `inworld-tts-1.5-max` | `INWORLD_API_KEY` | [API Reference](https://docs.inworld.ai/tts/api-reference) |
 | [Google (Gemini TTS)](https://docs.cloud.google.com/text-to-speech/docs/gemini-tts) | `google` | `gemini-2.5-flash-preview-tts` | `GOOGLE_API_KEY` | [API Reference](https://ai.google.dev/gemini-api/docs/text-generation) |
 | [Fish Audio](https://docs.fish.audio) | `fish-audio` | `s2-pro` | `FISH_AUDIO_API_KEY` | [API Reference](https://docs.fish.audio/developer-guide/core-features/text-to-speech) |
-| [Unreal Speech](https://docs.v8.unrealspeech.com) | `unreal-speech` | `default` | `UNREAL_SPEECH_API_KEY` | [API Reference](https://docs.v8.unrealspeech.com) |
 | [Murf](https://murf.ai/api/docs) | `murf` | `GEN2` | `MURF_API_KEY` | [API Reference](https://murf.ai/api/docs/api-reference/text-to-speech/generate) |
 | [Resemble](https://docs.resemble.ai) | `resemble` | `default` | `RESEMBLE_API_KEY` | [API Reference](https://docs.resemble.ai/api-reference/text-to-speech/synthesize) |
 | [fal](https://fal.ai/models) | `fal-ai` | *(user-specified)* | `FAL_API_KEY` | [API Reference](https://fal.ai/models) |
@@ -413,6 +407,12 @@ pnpm run typecheck              # type-check without emitting
 
 E2E tests hit real provider APIs. Set the relevant API key environment variables in a `.env` file or export them in your shell.
 
+Set `SPEECH_SDK_E2E_OUTPUT_DIR` to have the conversation e2e tests write their generated audio to disk (useful for sampling/comparing provider output):
+
+```bash
+SPEECH_SDK_E2E_OUTPUT_DIR=~/Downloads/convos pnpm run test:e2e
+```
+
 ## License
 
 MIT

package/dist/__tests__/e2e/_save-audio.d.ts

@@ -0,0 +1,37 @@
+import { generateConversation as _generateConversation } from "../../generate-conversation.js";
+import { generateSpeech as _generateSpeech } from "../../generate-speech.js";
+/**
+ * Write a test-generated audio file to `SPEECH_SDK_E2E_OUTPUT_DIR` if the env
+ * var is set. No-op otherwise, so normal CI runs don't produce artifacts.
+ * Usually you don't need to call this directly — use the `generateSpeech`,
+ * `generateConversation`, and `collectStreamAndSave` helpers exported from
+ * this module, which autosave using the current test name.
+ *
+ * Output layout: `$SPEECH_SDK_E2E_OUTPUT_DIR/<provider-file>/<test-slug>.<ext>`.
+ * If the same test saves multiple times, subsequent files are suffixed `-2`,
+ * `-3`, etc.
+ */
+export declare function maybeSaveAudio(name: string, audio: {
+    uint8Array: Uint8Array;
+    mediaType: string;
+}): Promise<void>;
+/**
+ * Drop-in replacement for `generateSpeech` that autosaves to
+ * `SPEECH_SDK_E2E_OUTPUT_DIR` using the current vitest test name.
+ */
+export declare const generateSpeech: typeof _generateSpeech;
+/**
+ * Drop-in replacement for `generateConversation` that autosaves to
+ * `SPEECH_SDK_E2E_OUTPUT_DIR` using the current vitest test name.
+ */
+export declare const generateConversation: typeof _generateConversation;
+/**
+ * Collects a streamed `streamSpeech` result into bytes AND autosaves them to
+ * `SPEECH_SDK_E2E_OUTPUT_DIR` using the current vitest test name. Use in place
+ * of `collectStream(result.audio)` in e2e tests.
+ */
+export declare function collectStreamAndSave(result: {
+    audio: ReadableStream<Uint8Array>;
+    mediaType: string;
+}): Promise<Uint8Array>;
+//# sourceMappingURL=_save-audio.d.ts.map

package/dist/__tests__/e2e/_save-audio.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"_save-audio.d.ts","sourceRoot":"","sources":["../../../src/__tests__/e2e/_save-audio.ts"],"names":[],"mappings":"AAGA,OAAO,EAAE,oBAAoB,IAAI,qBAAqB,EAAE,MAAM,gCAAgC,CAAC;AAC/F,OAAO,EAAE,cAAc,IAAI,eAAe,EAAE,MAAM,0BAA0B,CAAC;AAkF7E;;;;;;;;;;GAUG;AACH,wBAAsB,cAAc,CAClC,IAAI,EAAE,MAAM,EACZ,KAAK,EAAE;IAAE,UAAU,EAAE,UAAU,CAAC;IAAC,SAAS,EAAE,MAAM,CAAA;CAAE,GACnD,OAAO,CAAC,IAAI,CAAC,CAaf;AAOD;;;GAGG;AACH,eAAO,MAAM,cAAc,EAAE,OAAO,eAMR,CAAC;AAE7B;;;GAGG;AACH,eAAO,MAAM,oBAAoB,EAAE,OAAO,qBAMR,CAAC;AAEnC;;;;GAIG;AACH,wBAAsB,oBAAoB,CAAC,MAAM,EAAE;IACjD,KAAK,EAAE,cAAc,CAAC,UAAU,CAAC,CAAC;IAClC,SAAS,EAAE,MAAM,CAAC;CACnB,GAAG,OAAO,CAAC,UAAU,CAAC,CAOtB"}

package/dist/__tests__/e2e/_save-audio.js

@@ -0,0 +1,135 @@
+import { mkdir, writeFile } from "node:fs/promises";
+import { basename, join } from "node:path";
+import { expect } from "vitest";
+import { generateConversation as _generateConversation } from "../../generate-conversation.js";
+import { generateSpeech as _generateSpeech } from "../../generate-speech.js";
+import { collectStream } from "./_collect-stream.js";
+function extFor(mediaType) {
+    if (mediaType.includes("wav")) {
+        return "wav";
+    }
+    if (mediaType.includes("mpeg") || mediaType.includes("mp3")) {
+        return "mp3";
+    }
+    if (mediaType.includes("ogg")) {
+        return "ogg";
+    }
+    if (mediaType.includes("flac")) {
+        return "flac";
+    }
+    if (mediaType.includes("opus")) {
+        return "opus";
+    }
+    if (mediaType.includes("pcm")) {
+        return "pcm";
+    }
+    return "bin";
+}
+const NON_SLUG_CHARS = /[^a-zA-Z0-9._-]+/g;
+const LEADING_OR_TRAILING_UNDERSCORES = /^_+|_+$/g;
+const E2E_TEST_SUFFIX = /\.e2e\.test\.(ts|tsx|js|mjs)$/;
+function slugify(name) {
+    return name
+        .replace(NON_SLUG_CHARS, "_")
+        .replace(LEADING_OR_TRAILING_UNDERSCORES, "");
+}
+function resolveOutputDir() {
+    const dir = process.env.SPEECH_SDK_E2E_OUTPUT_DIR;
+    if (!dir) {
+        return null;
+    }
+    return dir.startsWith("~") ? join(process.env.HOME ?? "", dir.slice(1)) : dir;
+}
+function currentTestContext() {
+    // biome-ignore lint/suspicious/noMisplacedAssertion: reading vitest state, not asserting
+    const state = expect.getState();
+    return {
+        currentTestName: state.currentTestName,
+        testPath: state.testPath,
+    };
+}
+/**
+ * Derives the subdirectory for a given test file. e2e tests are named like
+ * `openai.e2e.test.ts` / `conversation-google.e2e.test.ts`; we strip the
+ * `.e2e.test.ts` suffix and use that as the per-provider bucket so a full run
+ * doesn't dump 100+ files into a single flat directory.
+ */
+function providerBucket(testPath) {
+    if (!testPath) {
+        return "unknown";
+    }
+    const base = basename(testPath).replace(E2E_TEST_SUFFIX, "");
+    return slugify(base) || "unknown";
+}
+// Counter keyed by `${bucket}/${slug}` so multiple generate/stream calls
+// within a single test don't overwrite each other. Vitest isolates modules
+// per file, so this resets per test file — collisions are only meaningful
+// within the same `it`.
+const callCounts = new Map();
+function nextFilename(bucket, slug, ext) {
+    const key = `${bucket}/${slug}`;
+    const n = (callCounts.get(key) ?? 0) + 1;
+    callCounts.set(key, n);
+    return n === 1 ? `${slug}.${ext}` : `${slug}-${n}.${ext}`;
+}
+/**
+ * Write a test-generated audio file to `SPEECH_SDK_E2E_OUTPUT_DIR` if the env
+ * var is set. No-op otherwise, so normal CI runs don't produce artifacts.
+ * Usually you don't need to call this directly — use the `generateSpeech`,
+ * `generateConversation`, and `collectStreamAndSave` helpers exported from
+ * this module, which autosave using the current test name.
+ *
+ * Output layout: `$SPEECH_SDK_E2E_OUTPUT_DIR/<provider-file>/<test-slug>.<ext>`.
+ * If the same test saves multiple times, subsequent files are suffixed `-2`,
+ * `-3`, etc.
+ */
+export async function maybeSaveAudio(name, audio) {
+    const dir = resolveOutputDir();
+    if (!dir) {
+        return;
+    }
+    const { testPath } = currentTestContext();
+    const bucket = providerBucket(testPath);
+    const bucketDir = join(dir, bucket);
+    await mkdir(bucketDir, { recursive: true });
+    const filename = nextFilename(bucket, slugify(name), extFor(audio.mediaType));
+    const file = join(bucketDir, filename);
+    await writeFile(file, audio.uint8Array);
+    console.log(`[maybeSaveAudio] wrote ${file}`);
+}
+function currentTestSlug() {
+    const { currentTestName } = currentTestContext();
+    return slugify(currentTestName ?? "unnamed") || "unnamed";
+}
+/**
+ * Drop-in replacement for `generateSpeech` that autosaves to
+ * `SPEECH_SDK_E2E_OUTPUT_DIR` using the current vitest test name.
+ */
+export const generateSpeech = (async (options) => {
+    const result = await _generateSpeech(options);
+    await maybeSaveAudio(currentTestSlug(), result.audio);
+    return result;
+});
+/**
+ * Drop-in replacement for `generateConversation` that autosaves to
+ * `SPEECH_SDK_E2E_OUTPUT_DIR` using the current vitest test name.
+ */
+export const generateConversation = (async (options) => {
+    const result = await _generateConversation(options);
+    await maybeSaveAudio(currentTestSlug(), result.audio);
+    return result;
+});
+/**
+ * Collects a streamed `streamSpeech` result into bytes AND autosaves them to
+ * `SPEECH_SDK_E2E_OUTPUT_DIR` using the current vitest test name. Use in place
+ * of `collectStream(result.audio)` in e2e tests.
+ */
+export async function collectStreamAndSave(result) {
+    const bytes = await collectStream(result.audio);
+    await maybeSaveAudio(currentTestSlug(), {
+        uint8Array: bytes,
+        mediaType: result.mediaType,
+    });
+    return bytes;
+}
+//# sourceMappingURL=_save-audio.js.map

package/dist/__tests__/e2e/_save-audio.js.map

@@ -0,0 +1 @@
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package/dist/conversation/pcm-concat.d.ts

@@ -5,6 +5,14 @@ export interface Pcm16Segment {
 }
 /** Decode a provider response to mono 16-bit PCM + its native sample rate. */
 export declare function decodeToPcm16(data: Uint8Array, mediaType: string): Pcm16Segment;
+/**
+ * Default RMS target: −20 dBFS — broadcast/podcast voice loudness convention
+ * with ~20 dB peak headroom. Comfortable to listen to and leaves room for
+ * typical TTS peaks not to clip.
+ */
+export declare const DEFAULT_VOLUME_DBFS = -20;
+/** Convert a dBFS level (≤ 0) to the equivalent int16 RMS amplitude. */
+export declare function dbfsToInt16Rms(dbfs: number): number;
 /**
  * RMS-normalize each segment to an absolute target amplitude. Each segment
  * is processed independently — no cross-segment dependency — so:

package/dist/conversation/pcm-concat.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"pcm-concat.d.ts","sourceRoot":"","sources":["../../src/conversation/pcm-concat.ts"],"names":[],"mappings":"AAEA,MAAM,WAAW,YAAY;IAC3B,QAAQ,CAAC,QAAQ,EAAE,MAAM,CAAC;IAC1B,QAAQ,CAAC,GAAG,EAAE,UAAU,CAAC;IACzB,QAAQ,CAAC,UAAU,EAAE,MAAM,CAAC;CAC7B;AAwED,8EAA8E;AAC9E,wBAAgB,aAAa,CAC3B,IAAI,EAAE,UAAU,EAChB,SAAS,EAAE,MAAM,GAChB,YAAY,CAqCd;AAkHD;;;;GAIG;AACH,eAAO,MAAM,mBAAmB,MAAM,CAAC;AAEvC,wEAAwE;AACxE,wBAAgB,cAAc,CAAC,IAAI,EAAE,MAAM,GAAG,MAAM,CAEnD;AAID;;;;;;;;;;;GAWG;AACH,wBAAgB,YAAY,CAC1B,QAAQ,EAAE,SAAS,YAAY,EAAE,EACjC,kBAAkB,SAA2B,GAC5C,YAAY,EAAE,CAQhB;AAED;;;GAGG;AACH,wBAAsB,cAAc,CAClC,QAAQ,EAAE,SAAS,YAAY,EAAE,EACjC,OAAO,EAAE;IAAE,KAAK,EAAE,MAAM,CAAC;IAAC,gBAAgB,EAAE,MAAM,CAAA;CAAE,GACnD,OAAO,CAAC,UAAU,CAAC,CA8BrB"}

package/dist/conversation/pcm-concat.js

@@ -27,6 +27,40 @@ function downmixToMono(interleaved, channels) {
     }
     return out;
 }
+/**
+ * View 32-bit little-endian float PCM bytes as a Float32Array. Reuses the
+ * existing buffer when 4-aligned; otherwise copies into a fresh, aligned
+ * buffer (Float32Array's view requires 4-byte alignment).
+ */
+function pcmBytesToFloat32(bytes) {
+    if (bytes.byteOffset % 4 === 0 && bytes.byteLength % 4 === 0) {
+        return new Float32Array(bytes.buffer, bytes.byteOffset, bytes.byteLength / 4);
+    }
+    const copy = new Uint8Array(bytes.byteLength);
+    copy.set(bytes);
+    return new Float32Array(copy.buffer);
+}
+const INT16_MAX = 32_767;
+const INT16_MIN = -32_768;
+/** Convert normalized [-1,1] float32 samples to int16 with clamping. */
+function float32ToInt16(f32) {
+    const out = new Int16Array(f32.length);
+    for (let i = 0; i < f32.length; i++) {
+        const s = f32[i];
+        if (s >= 1) {
+            out[i] = INT16_MAX;
+        }
+        else if (s <= -1) {
+            out[i] = INT16_MIN;
+        }
+        else {
+            out[i] = Math.round(s * INT16_MAX);
+        }
+    }
+    return out;
+}
+/** Match `encoding=<value>` (string) in a mediaType param list. */
+const ENCODING_PARAM_RE = /(?:^|;)\s*encoding=([a-z0-9_-]+)(?=$|;|\s)/i;
 /** Decode a provider response to mono 16-bit PCM + its native sample rate. */
 export function decodeToPcm16(data, mediaType) {
     const lower = mediaType.toLowerCase();
@@ -37,6 +71,15 @@ export function decodeToPcm16(data, mediaType) {
         // little-endian hosts before constructing the Int16Array.
         const sampleRate = parseMediaTypeParam(mediaType, "rate") ?? 24_000;
         const channels = parseMediaTypeParam(mediaType, "channels") ?? 1;
+        const encoding = lower.match(ENCODING_PARAM_RE)?.[1];
+        if (encoding === "float32") {
+            const interleaved = float32ToInt16(pcmBytesToFloat32(data));
+            return {
+                pcm: downmixToMono(interleaved, channels),
+                sampleRate,
+                channels: 1,
+            };
+        }
         const interleaved = pcmBytesToInt16(data);
         return {
             pcm: downmixToMono(interleaved, channels),
@@ -48,7 +91,7 @@ export function decodeToPcm16(data, mediaType) {
         return decodeWav(data);
     }
     throw new Error(`conversation.pcm-concat: unsupported stitch mediaType "${mediaType}". ` +
-        'getStitchOptions must return "audio/wav" or "audio/pcm;rate=..." so the stitch layer can concatenate without a compressed-audio decoder.');
+        'getStitchOptions must return "audio/wav" or "audio/pcm;rate=...[;encoding=float32]" so the stitch layer can concatenate without a compressed-audio decoder.');
 }
 function decodeWav(bytes) {
     const view = new DataView(bytes.buffer, bytes.byteOffset, bytes.byteLength);
@@ -126,8 +169,6 @@ function rmsPcm16(pcm) {
     }
     return Math.sqrt(sumSq / pcm.length);
 }
-const INT16_MAX = 32_767;
-const INT16_MIN = -32_768;
 function clampInt16(value) {
     if (value > INT16_MAX) {
         return INT16_MAX;
@@ -146,11 +187,16 @@ function scaleClamp(pcm, gain) {
     return out;
 }
 /**
- * Default RMS target: −20 dBFS for int16 = round(32767 * 10^(-20/20)) = 3277.
- * Broadcast/podcast voice loudness convention with ~20 dB peak headroom —
- * comfortable to listen to, leaves room for typical TTS peaks not to clip.
+ * Default RMS target: −20 dBFS — broadcast/podcast voice loudness convention
+ * with ~20 dB peak headroom. Comfortable to listen to and leaves room for
+ * typical TTS peaks not to clip.
  */
-const DEFAULT_TARGET_RMS_INT16 = 3277;
+export const DEFAULT_VOLUME_DBFS = -20;
+/** Convert a dBFS level (≤ 0) to the equivalent int16 RMS amplitude. */
+export function dbfsToInt16Rms(dbfs) {
+    return Math.round(INT16_MAX * 10 ** (dbfs / 20));
+}
+const DEFAULT_TARGET_RMS_INT16 = dbfsToInt16Rms(DEFAULT_VOLUME_DBFS);
 /**
  * RMS-normalize each segment to an absolute target amplitude. Each segment
  * is processed independently — no cross-segment dependency — so:

package/dist/conversation/pcm-concat.js.map

@@ -1 +1 @@
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package/dist/conversation/stitch.d.ts

@@ -15,6 +15,7 @@ interface StitchInput<V extends Voice = Voice> {
     }[];
     readonly topLevelProviderOptions?: Record<string, unknown>;
     readonly turns: readonly ConversationTurn<V>[];
+    readonly volumeDbfs?: number;
 }
 interface StitchOutput {
     readonly audio: Uint8Array;

package/dist/conversation/stitch.d.ts.map

@@ -1 +1 @@
-{"version":3,"file":"stitch.d.ts","sourceRoot":"","sources":["../../src/conversation/stitch.ts"],"names":[],"mappings":"AACA,OAAO,KAAK,EAAE,aAAa,EAAE,KAAK,EAAE,MAAM,uBAAuB,CAAC;AAElE,OAAO,KAAK,EAAE,gBAAgB,EAAE,MAAM,YAAY,CAAC;AAEnD,UAAU,WAAW,CAAC,CAAC,SAAS,KAAK,GAAG,KAAK;IAC3C,QAAQ,CAAC,WAAW,CAAC,EAAE,WAAW,CAAC;IACnC,QAAQ,CAAC,MAAM,CAAC,EAAE,MAAM,CAAC;IACzB,QAAQ,CAAC,KAAK,EAAE,MAAM,CAAC;IACvB,QAAQ,CAAC,OAAO,CAAC,EAAE,MAAM,CAAC,MAAM,EAAE,MAAM,CAAC,CAAC;IAC1C,QAAQ,CAAC,cAAc,EAAE,MAAM,CAAC;IAChC,QAAQ,CAAC,UAAU,EAAE,MAAM,CAAC;IAC5B,QAAQ,CAAC,eAAe,EAAE,OAAO,CAAC;IAClC,QAAQ,CAAC,eAAe,EAAE,SAAS,aAAa,CAAC,CAAC,CAAC,EAAE,CAAC;IACtD,QAAQ,CAAC,oBAAoB,EAAE,SAAS;QACtC,eAAe,EAAE,MAAM,CAAC,MAAM,EAAE,OAAO,CAAC,CAAC;QACzC,SAAS,EAAE,MAAM,CAAC;KACnB,EAAE,CAAC;IACJ,QAAQ,CAAC,uBAAuB,CAAC,EAAE,MAAM,CAAC,MAAM,EAAE,OAAO,CAAC,CAAC;IAC3D,QAAQ,CAAC,KAAK,EAAE,SAAS,gBAAgB,CAAC,CAAC,CAAC,EAAE,CAAC;CAChD;AAED,UAAU,YAAY;IACpB,QAAQ,CAAC,KAAK,EAAE,UAAU,CAAC;IAC3B,QAAQ,CAAC,SAAS,EAAE,MAAM,CAAC;IAC3B,QAAQ,CAAC,QAAQ,EAAE;QACjB,QAAQ,CAAC,UAAU,EAAE,MAAM,CAAC;QAC5B,QAAQ,CAAC,SAAS,EAAE,MAAM,CAAC;QAC3B,QAAQ,CAAC,eAAe,CAAC,EAAE,MAAM,CAAC;KACnC,CAAC;IACF,QAAQ,CAAC,uBAAuB,EAAE,SAAS,CACvC,MAAM,CAAC,MAAM,EAAE,OAAO,CAAC,GACvB,SAAS,CACZ,EAAE,CAAC;IACJ,QAAQ,CAAC,QAAQ,EAAE,SAAS,MAAM,EAAE,CAAC;CACtC;AA+BD,wBAAsB,SAAS,CAAC,CAAC,SAAS,KAAK,EAC7C,KAAK,EAAE,WAAW,CAAC,CAAC,CAAC,GACpB,OAAO,CAAC,YAAY,CAAC,CA4EvB"}
+{"version":3,"file":"stitch.d.ts","sourceRoot":"","sources":["../../src/conversation/stitch.ts"],"names":[],"mappings":"AACA,OAAO,KAAK,EAAE,aAAa,EAAE,KAAK,EAAE,MAAM,uBAAuB,CAAC;AAOlE,OAAO,KAAK,EAAE,gBAAgB,EAAE,MAAM,YAAY,CAAC;AAEnD,UAAU,WAAW,CAAC,CAAC,SAAS,KAAK,GAAG,KAAK;IAC3C,QAAQ,CAAC,WAAW,CAAC,EAAE,WAAW,CAAC;IACnC,QAAQ,CAAC,MAAM,CAAC,EAAE,MAAM,CAAC;IACzB,QAAQ,CAAC,KAAK,EAAE,MAAM,CAAC;IACvB,QAAQ,CAAC,OAAO,CAAC,EAAE,MAAM,CAAC,MAAM,EAAE,MAAM,CAAC,CAAC;IAC1C,QAAQ,CAAC,cAAc,EAAE,MAAM,CAAC;IAChC,QAAQ,CAAC,UAAU,EAAE,MAAM,CAAC;IAC5B,QAAQ,CAAC,eAAe,EAAE,OAAO,CAAC;IAClC,QAAQ,CAAC,eAAe,EAAE,SAAS,aAAa,CAAC,CAAC,CAAC,EAAE,CAAC;IACtD,QAAQ,CAAC,oBAAoB,EAAE,SAAS;QACtC,eAAe,EAAE,MAAM,CAAC,MAAM,EAAE,OAAO,CAAC,CAAC;QACzC,SAAS,EAAE,MAAM,CAAC;KACnB,EAAE,CAAC;IACJ,QAAQ,CAAC,uBAAuB,CAAC,EAAE,MAAM,CAAC,MAAM,EAAE,OAAO,CAAC,CAAC;IAC3D,QAAQ,CAAC,KAAK,EAAE,SAAS,gBAAgB,CAAC,CAAC,CAAC,EAAE,CAAC;IAC/C,QAAQ,CAAC,UAAU,CAAC,EAAE,MAAM,CAAC;CAC9B;AAED,UAAU,YAAY;IACpB,QAAQ,CAAC,KAAK,EAAE,UAAU,CAAC;IAC3B,QAAQ,CAAC,SAAS,EAAE,MAAM,CAAC;IAC3B,QAAQ,CAAC,QAAQ,EAAE;QACjB,QAAQ,CAAC,UAAU,EAAE,MAAM,CAAC;QAC5B,QAAQ,CAAC,SAAS,EAAE,MAAM,CAAC;QAC3B,QAAQ,CAAC,eAAe,CAAC,EAAE,MAAM,CAAC;KACnC,CAAC;IACF,QAAQ,CAAC,uBAAuB,EAAE,SAAS,CACvC,MAAM,CAAC,MAAM,EAAE,OAAO,CAAC,GACvB,SAAS,CACZ,EAAE,CAAC;IACJ,QAAQ,CAAC,QAAQ,EAAE,SAAS,MAAM,EAAE,CAAC;CACtC;AA+BD,wBAAsB,SAAS,CAAC,CAAC,SAAS,KAAK,EAC7C,KAAK,EAAE,WAAW,CAAC,CAAC,CAAC,GACpB,OAAO,CAAC,YAAY,CAAC,CA+EvB"}

package/dist/conversation/stitch.js

@@ -1,5 +1,5 @@
 import { generateSpeech } from "../generate-speech.js";
-import { concatPcmToWav, decodeToPcm16, normalizeRms } from "./pcm-concat.js";
+import { concatPcmToWav, dbfsToInt16Rms, decodeToPcm16, normalizeRms, } from "./pcm-concat.js";
 const TARGET_SAMPLE_RATE = 24_000;
 /**
  * Run `worker(items[i], i)` for each item, capping in-flight executions at
@@ -50,7 +50,7 @@ export async function runStitch(input) {
     });
     const segments = perTurn.map((p) => p.segment);
     const leveledSegments = input.normalizeVolume
-        ? normalizeRms(segments)
+        ? normalizeRms(segments, input.volumeDbfs == null ? undefined : dbfsToInt16Rms(input.volumeDbfs))
         : segments;
     const audio = await concatPcmToWav(leveledSegments, {
         gapMs: input.gapMs,

package/dist/conversation/stitch.js.map

@@ -1 +1 @@
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package/dist/conversation/types.d.ts

@@ -14,18 +14,26 @@ export interface GenerateConversationOptions<V extends Voice = Voice> {
     readonly maxRetries?: number;
     readonly model?: string | ResolvedModel<V>;
     /**
-     * RMS-normalize each per-turn segment to an absolute target level
-     * (-20 dBFS, the broadcast/podcast voice standard). Every call to
-     * generateConversation produces output at the same loudness regardless
-     * of which providers or content are used, so two separate conversations
-     * can be played back-to-back without the listener adjusting volume.
-     * Roughly two O(N) passes over the int16 PCM samples — cheap. Pass
-     * `false` to skip the step entirely (~zero work) and keep the raw
-     * provider levels. Stitch path only; native single-call dialogue
-     * providers control their own mix. Default: true.
+     * RMS-normalize the output audio to an absolute target level (see
+     * `volumeDbfs` for the level itself, default -20 dBFS — the broadcast /
+     * podcast voice standard). Every call to generateConversation produces
+     * output at the same loudness regardless of which providers or content
+     * are used, so two separate conversations can be played back-to-back
+     * without the listener adjusting volume. Roughly two O(N) passes over
+     * the int16 PCM samples — cheap. Pass `false` to skip the step entirely
+     * (~zero work) and keep the raw provider levels. Applied on both the
+     * stitch and native dialogue paths, provided the chosen provider
+     * exposes a decodable PCM/WAV mode via `getStitchOptions`. Default: true.
      */
     readonly normalizeVolume?: boolean;
     readonly providerOptions?: Record<string, unknown>;
     readonly turns: readonly ConversationTurn<V>[];
+    /**
+     * Target loudness in dBFS for `normalizeVolume`. Must be ≤ 0 (0 dBFS is
+     * the int16 ceiling). Lower values are quieter — -20 leaves ~20 dB of
+     * peak headroom so typical TTS speech doesn't clip after gain. Ignored
+     * when `normalizeVolume` is `false`. Default: -20.
+     */
+    readonly volumeDbfs?: number;
 }
 //# sourceMappingURL=types.d.ts.map

package/dist/conversation/types.d.ts.map

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package/dist/conversation/validate.d.ts.map

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package/dist/conversation/validate.js

@@ -1,5 +1,4 @@
 import { ConversationInputError } from "./errors.js";
-const MAX_UNIQUE_VOICES = 4;
 /**
  * Stable key for a voice so we can count unique voices across turns within
  * one call. String voices and URL voices use their value; binary
@@ -41,10 +40,5 @@ export function validateConversationInput(options) {
             throw new ConversationInputError(`turns[${i}]: model must be set, either at top-level or on the turn.`);
         }
     }
-    const ctx = newVoiceKeyContext();
-    const uniqueVoices = new Set(options.turns.map((t) => voiceKey(t.voice, ctx.refIds, ctx.refCounter)));
-    if (uniqueVoices.size > MAX_UNIQUE_VOICES) {
-        throw new ConversationInputError(`generateConversation accepts at most 4 unique voices; got ${uniqueVoices.size}.`);
-    }
 }
 //# sourceMappingURL=validate.js.map

package/dist/conversation/validate.js.map

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