@speech-sdk/core 0.5.2 → 0.6.0

package/README.md

@@ -117,6 +117,115 @@ Calling `streamSpeech()` on a model that doesn't declare the `"streaming"` featu
 
 Retries apply only to the initial request, until response headers arrive. Once bytes start flowing, mid-stream errors propagate to the `ReadableStream` consumer as a stream error and are not retried. Pass `maxRetries` (default `2`) and an `abortSignal` the same way as `generateSpeech()`.
 
+## Conversations
+
+`generateConversation()` produces a single multi-voice audio clip from an ordered array of turns. It picks the best path automatically:
+
+- **Native dialogue** — when every turn shares one model and that provider has a real multi-speaker dialogue endpoint, the SDK makes a single API call and returns the provider's natural mix. Works with **ElevenLabs v3**, **Google Gemini TTS** (exactly 2 voices), **Hume Octave**, **Fish Audio S2-Pro**, and **fal Dia**.
+- **Stitch fallback** — when turns span multiple providers, or the chosen model has no native dialogue endpoint, the SDK calls `generateSpeech()` per turn in parallel, normalizes each result to PCM, RMS-levels them so quieter providers don't get drowned out, inserts a configurable silence between turns, and returns a single WAV.
+
+```ts
+import { generateConversation } from "@speech-sdk/core/conversation";
+
+const result = await generateConversation({
+  turns: [
+    { model: "openai/tts-1", voice: "nova", text: "Hi, I'm hosted by OpenAI." },
+    { model: "elevenlabs/eleven_multilingual_v2", voice: "JBFqnCBsd6RMkjVDRZzb", text: "And I'm hosted by ElevenLabs." },
+    { model: "google/gemini-3.1-flash-tts-preview", voice: "Kore", text: "I'm Gemini three-point-one flash TTS." },
+    { model: "hume/octave-2", voice: "Kora", text: "And I'm Hume Octave. Thanks for listening." },
+  ],
+});
+
+result.audio.uint8Array;  // Uint8Array of one combined WAV
+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
+  gapMs?: number,                                 // silence between turns (stitch path), default 300
+  normalizeVolume?: boolean,                      // RMS-level stitched turns, default true
+  maxConcurrency?: number,                        // cap parallel generateSpeech calls, default 6
+  maxRetries?: number,                            // per-turn retries, default 2
+  apiKey?: string,
+  providerOptions?: Record<string, unknown>,      // forwarded to every provider; per-turn override available
+  abortSignal?: AbortSignal,
+  headers?: Record<string, string>,
+});
+
+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>;
+}
+```
+
+### 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:
+
+- 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.
+
+Pass `normalizeVolume: false` to skip the step entirely (zero work) when you want raw provider levels.
+
+### Errors
+
+Conversation-specific errors (importable 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 |
+| `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`) |
+
+### Native dialogue caps
+
+| Provider | Native dialogue model | Voice constraints |
+|---|---|---|
+| ElevenLabs | `eleven_v3` | 1–10 voices, ≤ 2,000 total chars |
+| Google | `gemini-2.5-flash-preview-tts`, `gemini-2.5-pro-preview-tts`, `gemini-3.1-flash-tts-preview` | **Exactly 2 voices** (API requirement) |
+| Hume | `octave-1`, `octave-2` | 1–4 voices |
+| 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.

package/dist/audio-utils.d.ts

@@ -0,0 +1,12 @@
+/**
+ * Parse a numeric parameter from a mediaType string (e.g. "audio/pcm;rate=24000").
+ * Returns undefined if missing or non-positive.
+ */
+export declare function parseMediaTypeParam(mediaType: string, name: string): number | undefined;
+/**
+ * Wrap raw 16-bit little-endian mono PCM bytes in a WAV container.
+ * Cross-platform (browser, Node, edge) via mediabunny's container ops —
+ * does not require the WebCodecs encoder.
+ */
+export declare function wrapPcm16Mono(pcm: Uint8Array, sampleRate: number): Promise<Uint8Array>;
+//# sourceMappingURL=audio-utils.d.ts.map

package/dist/audio-utils.d.ts.map

@@ -0,0 +1 @@
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package/dist/audio-utils.js

@@ -0,0 +1,53 @@
+import { BufferTarget, EncodedAudioPacketSource, EncodedPacket, Output, WavOutputFormat, } from "mediabunny";
+const PARAM_REGEX_CACHE = new Map();
+/**
+ * Parse a numeric parameter from a mediaType string (e.g. "audio/pcm;rate=24000").
+ * Returns undefined if missing or non-positive.
+ */
+export function parseMediaTypeParam(mediaType, name) {
+    let re = PARAM_REGEX_CACHE.get(name);
+    if (!re) {
+        // End boundary required: digits must be followed by ;, whitespace, or
+        // end-of-string. Rejects values like "rate=24000x".
+        re = new RegExp(`(?:^|;)\\s*${name}=(\\d+)(?=$|;|\\s)`, "i");
+        PARAM_REGEX_CACHE.set(name, re);
+    }
+    const match = mediaType.match(re);
+    if (!match) {
+        return undefined;
+    }
+    const value = Number(match[1]);
+    return Number.isFinite(value) && value > 0 ? value : undefined;
+}
+/**
+ * Wrap raw 16-bit little-endian mono PCM bytes in a WAV container.
+ * Cross-platform (browser, Node, edge) via mediabunny's container ops —
+ * does not require the WebCodecs encoder.
+ */
+export async function wrapPcm16Mono(pcm, sampleRate) {
+    const output = new Output({
+        format: new WavOutputFormat(),
+        target: new BufferTarget(),
+    });
+    const source = new EncodedAudioPacketSource("pcm-s16");
+    output.addAudioTrack(source);
+    await output.start();
+    // 2 bytes per sample, mono.
+    const numSamples = pcm.length / 2;
+    const durationSeconds = numSamples / sampleRate;
+    const packet = new EncodedPacket(pcm, "key", 0, durationSeconds, 0);
+    await source.add(packet, {
+        decoderConfig: {
+            codec: "pcm-s16",
+            numberOfChannels: 1,
+            sampleRate,
+        },
+    });
+    await output.finalize();
+    const buffer = output.target.buffer;
+    if (!buffer) {
+        throw new Error("audio-utils: WavOutputFormat produced no buffer");
+    }
+    return new Uint8Array(buffer);
+}
+//# sourceMappingURL=audio-utils.js.map

package/dist/audio-utils.js.map

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package/dist/conversation/dispatch.d.ts

@@ -0,0 +1,17 @@
+import type { ResolvedModel, Voice } from "../speech-provider.js";
+import type { ConversationTurn } from "./types.js";
+export type ConversationPath = {
+    kind: "native";
+    resolved: ResolvedModel<Voice>;
+} | {
+    kind: "stitch";
+    stitchOptionsPerTurn: readonly {
+        providerOptions: Record<string, unknown>;
+        mediaType: string;
+    }[];
+};
+export declare function chooseConversationPath(input: {
+    resolvedPerTurn: readonly ResolvedModel<Voice>[];
+    turns: readonly ConversationTurn<Voice>[];
+}): ConversationPath;
+//# sourceMappingURL=dispatch.d.ts.map

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

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package/dist/conversation/dispatch.js

@@ -0,0 +1,60 @@
+import { DialogueConstraintError, StitchUnsupportedError } from "./errors.js";
+import { newVoiceKeyContext, voiceKey } from "./validate.js";
+export function chooseConversationPath(input) {
+    const { resolvedPerTurn, turns } = input;
+    // Compare by provider instance reference, not just provider id, so two
+    // factories of the same provider with different apiKey/baseURL/fetch
+    // configs are not silently merged into one.
+    const first = resolvedPerTurn[0];
+    const allSame = resolvedPerTurn.every((r) => r.provider === first.provider && r.modelId === first.modelId);
+    if (allSame) {
+        const { provider, modelId } = first;
+        if (provider.generateDialogue && provider.dialogueCapabilities) {
+            const caps = provider.dialogueCapabilities(modelId);
+            if (caps) {
+                assertNativeConstraints({ provider, modelId, caps, turns });
+                return { kind: "native", resolved: first };
+            }
+        }
+    }
+    // Stitch path — every resolved (provider, modelId) must support getStitchOptions.
+    const stitchOptionsPerTurn = resolvedPerTurn.map((r) => {
+        const opts = r.provider.getStitchOptions?.(r.modelId);
+        if (!opts) {
+            throw new StitchUnsupportedError({
+                provider: r.provider.id,
+                model: r.modelId,
+            });
+        }
+        return opts;
+    });
+    return { kind: "stitch", stitchOptionsPerTurn };
+}
+function assertNativeConstraints(args) {
+    const { provider, modelId, caps, turns } = args;
+    const ctx = newVoiceKeyContext();
+    const unique = new Set(turns.map((t) => voiceKey(t.voice, ctx.refIds, ctx.refCounter))).size;
+    if (unique < caps.minVoices || unique > caps.maxVoices) {
+        const rule = caps.minVoices === caps.maxVoices
+            ? `exactly ${caps.minVoices} unique voices`
+            : `between ${caps.minVoices} and ${caps.maxVoices} unique voices`;
+        throw new DialogueConstraintError({
+            provider: provider.id,
+            model: modelId,
+            rule,
+            observed: `${unique} unique voices`,
+        });
+    }
+    if (caps.maxTotalChars != null) {
+        const total = turns.reduce((n, t) => n + t.text.length, 0);
+        if (total > caps.maxTotalChars) {
+            throw new DialogueConstraintError({
+                provider: provider.id,
+                model: modelId,
+                rule: `total characters <= ${caps.maxTotalChars}`,
+                observed: `${total} characters`,
+            });
+        }
+    }
+}
+//# sourceMappingURL=dispatch.js.map

package/dist/conversation/dispatch.js.map

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package/dist/conversation/errors.d.ts

@@ -0,0 +1,23 @@
+import { SpeechSDKError } from "../errors.js";
+export declare class ConversationInputError extends SpeechSDKError {
+    constructor(message: string);
+}
+export declare class DialogueConstraintError extends SpeechSDKError {
+    readonly provider: string;
+    readonly model: string;
+    constructor(options: {
+        provider: string;
+        model: string;
+        rule: string;
+        observed: string;
+    });
+}
+export declare class StitchUnsupportedError extends SpeechSDKError {
+    readonly provider: string;
+    readonly model: string;
+    constructor(options: {
+        provider: string;
+        model: string;
+    });
+}
+//# sourceMappingURL=errors.d.ts.map

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

@@ -0,0 +1 @@
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package/dist/conversation/errors.js

@@ -0,0 +1,28 @@
+import { SpeechSDKError } from "../errors.js";
+export class ConversationInputError extends SpeechSDKError {
+    constructor(message) {
+        super(message);
+        this.name = "ConversationInputError";
+    }
+}
+export class DialogueConstraintError extends SpeechSDKError {
+    provider;
+    model;
+    constructor(options) {
+        super(`${options.provider}/${options.model} native dialogue requires ${options.rule}; got ${options.observed}.`);
+        this.name = "DialogueConstraintError";
+        this.provider = options.provider;
+        this.model = options.model;
+    }
+}
+export class StitchUnsupportedError extends SpeechSDKError {
+    provider;
+    model;
+    constructor(options) {
+        super(`${options.provider}/${options.model} cannot be used in a stitched conversation: provider does not support PCM/WAV output for this model.`);
+        this.name = "StitchUnsupportedError";
+        this.provider = options.provider;
+        this.model = options.model;
+    }
+}
+//# sourceMappingURL=errors.js.map

package/dist/conversation/errors.js.map

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

@@ -0,0 +1,29 @@
+export interface Pcm16Segment {
+    readonly channels: number;
+    readonly pcm: Int16Array;
+    readonly sampleRate: number;
+}
+/** Decode a provider response to mono 16-bit PCM + its native sample rate. */
+export declare function decodeToPcm16(data: Uint8Array, mediaType: string): Pcm16Segment;
+/**
+ * RMS-normalize each segment to an absolute target amplitude. Each segment
+ * is processed independently — no cross-segment dependency — so:
+ *   - The output level is the same across runs regardless of input mix.
+ *   - Two `generateConversation` calls produce comparable loudness even
+ *     with completely different content.
+ *
+ * Silent segments pass through unchanged. Output is clamped to int16
+ * range, so a quiet segment with rare peaks may clip slightly when
+ * boosted; the default target leaves ~20 dB headroom to make this rare
+ * for typical TTS content.
+ */
+export declare function normalizeRms(segments: readonly Pcm16Segment[], targetRmsAmplitude?: number): Pcm16Segment[];
+/**
+ * Resample each segment to `targetSampleRate` mono, interleave with `gapMs`
+ * silence, and mux the result as a WAV file via mediabunny.
+ */
+export declare function concatPcmToWav(segments: readonly Pcm16Segment[], options: {
+    gapMs: number;
+    targetSampleRate: number;
+}): Promise<Uint8Array>;
+//# sourceMappingURL=pcm-concat.d.ts.map

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

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package/dist/conversation/pcm-concat.js

@@ -0,0 +1,200 @@
+import { parseMediaTypeParam, wrapPcm16Mono } from "../audio-utils.js";
+/**
+ * View 16-bit little-endian PCM bytes as an Int16Array. Reuses the existing
+ * buffer when `byteOffset` is 2-aligned; otherwise copies into a fresh,
+ * aligned buffer (Int16Array's buffer view requires 2-byte alignment).
+ */
+function pcmBytesToInt16(bytes) {
+    if (bytes.byteOffset % 2 === 0 && bytes.byteLength % 2 === 0) {
+        return new Int16Array(bytes.buffer, bytes.byteOffset, bytes.byteLength / 2);
+    }
+    const copy = new Uint8Array(bytes.byteLength);
+    copy.set(bytes);
+    return new Int16Array(copy.buffer);
+}
+function downmixToMono(interleaved, channels) {
+    if (channels === 1) {
+        return interleaved;
+    }
+    const frames = Math.floor(interleaved.length / channels);
+    const out = new Int16Array(frames);
+    for (let f = 0; f < frames; f++) {
+        let sum = 0;
+        for (let c = 0; c < channels; c++) {
+            sum += interleaved[f * channels + c];
+        }
+        out[f] = Math.round(sum / channels);
+    }
+    return out;
+}
+/** Decode a provider response to mono 16-bit PCM + its native sample rate. */
+export function decodeToPcm16(data, mediaType) {
+    const lower = mediaType.toLowerCase();
+    if (lower.startsWith("audio/pcm") || lower.startsWith("audio/x-pcm")) {
+        // NOTE: `audio/l16` (RFC 2586) is intentionally NOT handled here. The
+        // standard mandates network byte order (big-endian) but no provider in
+        // this SDK currently emits it. If support is added later, byte-swap on
+        // little-endian hosts before constructing the Int16Array.
+        const sampleRate = parseMediaTypeParam(mediaType, "rate") ?? 24_000;
+        const channels = parseMediaTypeParam(mediaType, "channels") ?? 1;
+        const interleaved = pcmBytesToInt16(data);
+        return {
+            pcm: downmixToMono(interleaved, channels),
+            sampleRate,
+            channels: 1,
+        };
+    }
+    if (lower.startsWith("audio/wav") || lower.startsWith("audio/x-wav")) {
+        return decodeWav(data);
+    }
+    throw new Error(`conversation.pcm-concat: unsupported stitch mediaType "${mediaType}". ` +
+        'getStitchOptions must return "audio/wav" or "audio/pcm;rate=..." so the stitch layer can concatenate without a compressed-audio decoder.');
+}
+function decodeWav(bytes) {
+    const view = new DataView(bytes.buffer, bytes.byteOffset, bytes.byteLength);
+    if (view.getUint32(0) !== 0x52_49_46_46 ||
+        view.getUint32(8) !== 0x57_41_56_45) {
+        throw new Error("conversation.pcm-concat: not a RIFF/WAVE file");
+    }
+    // Scan chunks for "fmt " and "data".
+    let offset = 12;
+    let sampleRate = 0;
+    let channels = 0;
+    let bitsPerSample = 0;
+    let audioFormat = 0;
+    let dataStart = -1;
+    let dataLen = 0;
+    while (offset + 8 <= bytes.byteLength) {
+        const chunkId = view.getUint32(offset);
+        const chunkSize = view.getUint32(offset + 4, true);
+        if (chunkId === 0x66_6d_74_20) {
+            audioFormat = view.getUint16(offset + 8, true);
+            channels = view.getUint16(offset + 10, true);
+            sampleRate = view.getUint32(offset + 12, true);
+            bitsPerSample = view.getUint16(offset + 22, true);
+        }
+        else if (chunkId === 0x64_61_74_61) {
+            dataStart = offset + 8;
+            dataLen = chunkSize;
+            break;
+        }
+        offset += 8 + chunkSize + (chunkSize % 2);
+    }
+    if (dataStart < 0 ||
+        sampleRate === 0 ||
+        bitsPerSample !== 16 ||
+        audioFormat !== 1) {
+        throw new Error(`conversation.pcm-concat: only 16-bit PCM WAV is supported (got audioFormat=${audioFormat}, bps=${bitsPerSample})`);
+    }
+    const payload = bytes.subarray(dataStart, dataStart + dataLen);
+    const interleaved = pcmBytesToInt16(payload);
+    return {
+        pcm: downmixToMono(interleaved, channels || 1),
+        sampleRate,
+        channels: 1,
+    };
+}
+/** Simple linear interpolation resampler for mono Int16 PCM. */
+function resamplePcm16LinearMono(input, fromRate, toRate) {
+    if (fromRate === toRate) {
+        return input;
+    }
+    const ratio = fromRate / toRate;
+    const outLen = Math.round(input.length / ratio);
+    const out = new Int16Array(outLen);
+    for (let i = 0; i < outLen; i++) {
+        const srcPos = i * ratio;
+        const i0 = Math.floor(srcPos);
+        const i1 = Math.min(i0 + 1, input.length - 1);
+        const frac = srcPos - i0;
+        out[i] = Math.round(input[i0] * (1 - frac) + input[i1] * frac);
+    }
+    return out;
+}
+function silencePcm16(ms, sampleRate) {
+    const samples = Math.round((ms / 1000) * sampleRate);
+    return new Int16Array(samples);
+}
+/** Root-mean-square amplitude of a PCM segment. */
+function rmsPcm16(pcm) {
+    if (pcm.length === 0) {
+        return 0;
+    }
+    let sumSq = 0;
+    for (const s of pcm) {
+        sumSq += s * s;
+    }
+    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;
+    }
+    if (value < INT16_MIN) {
+        return INT16_MIN;
+    }
+    return value;
+}
+/** Multiply each sample by `gain`, clamping to int16 range. */
+function scaleClamp(pcm, gain) {
+    const out = new Int16Array(pcm.length);
+    for (let i = 0; i < pcm.length; i++) {
+        out[i] = clampInt16(Math.round(pcm[i] * 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.
+ */
+const DEFAULT_TARGET_RMS_INT16 = 3277;
+/**
+ * RMS-normalize each segment to an absolute target amplitude. Each segment
+ * is processed independently — no cross-segment dependency — so:
+ *   - The output level is the same across runs regardless of input mix.
+ *   - Two `generateConversation` calls produce comparable loudness even
+ *     with completely different content.
+ *
+ * Silent segments pass through unchanged. Output is clamped to int16
+ * range, so a quiet segment with rare peaks may clip slightly when
+ * boosted; the default target leaves ~20 dB headroom to make this rare
+ * for typical TTS content.
+ */
+export function normalizeRms(segments, targetRmsAmplitude = DEFAULT_TARGET_RMS_INT16) {
+    return segments.map((s) => {
+        const segRms = rmsPcm16(s.pcm);
+        if (segRms === 0) {
+            return { ...s };
+        }
+        return { ...s, pcm: scaleClamp(s.pcm, targetRmsAmplitude / segRms) };
+    });
+}
+/**
+ * Resample each segment to `targetSampleRate` mono, interleave with `gapMs`
+ * silence, and mux the result as a WAV file via mediabunny.
+ */
+export async function concatPcmToWav(segments, options) {
+    const { gapMs, targetSampleRate } = options;
+    const resampled = [];
+    const gap = silencePcm16(gapMs, targetSampleRate);
+    for (let i = 0; i < segments.length; i++) {
+        const s = segments[i];
+        resampled.push(resamplePcm16LinearMono(s.pcm, s.sampleRate, targetSampleRate));
+        if (i < segments.length - 1 && gap.length > 0) {
+            resampled.push(gap);
+        }
+    }
+    const totalSamples = resampled.reduce((n, a) => n + a.length, 0);
+    const merged = new Int16Array(totalSamples);
+    let off = 0;
+    for (const a of resampled) {
+        merged.set(a, off);
+        off += a.length;
+    }
+    const mergedBytes = new Uint8Array(merged.buffer, merged.byteOffset, merged.byteLength);
+    return await wrapPcm16Mono(mergedBytes, targetSampleRate);
+}
+//# sourceMappingURL=pcm-concat.js.map

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

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

@@ -0,0 +1,32 @@
+import type { ResolvedModel, Voice } from "../speech-provider.js";
+import type { ConversationTurn } from "./types.js";
+interface StitchInput<V extends Voice = Voice> {
+    readonly abortSignal?: AbortSignal;
+    readonly apiKey?: string;
+    readonly gapMs: number;
+    readonly headers?: Record<string, string>;
+    readonly maxConcurrency: number;
+    readonly maxRetries: number;
+    readonly normalizeVolume: boolean;
+    readonly resolvedPerTurn: readonly ResolvedModel<V>[];
+    readonly stitchOptionsPerTurn: readonly {
+        providerOptions: Record<string, unknown>;
+        mediaType: string;
+    }[];
+    readonly topLevelProviderOptions?: Record<string, unknown>;
+    readonly turns: readonly ConversationTurn<V>[];
+}
+interface StitchOutput {
+    readonly audio: Uint8Array;
+    readonly mediaType: string;
+    readonly metadata: {
+        readonly inputChars: number;
+        readonly latencyMs: number;
+        readonly audioDurationMs?: number;
+    };
+    readonly providerMetadataPerTurn: readonly (Record<string, unknown> | undefined)[];
+    readonly warnings: readonly string[];
+}
+export declare function runStitch<V extends Voice>(input: StitchInput<V>): Promise<StitchOutput>;
+export {};
+//# sourceMappingURL=stitch.d.ts.map

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

@@ -0,0 +1 @@
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