@memberjunction/ng-conversations 5.43.0 → 5.45.0

package/dist/lib/services/realtime-pcm-wav.d.ts

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+/**
+ * @fileoverview Pure, browser-and-node-safe helpers for the CLIENT-DIRECT realtime recorder:
+ * encoding accumulated Float32 PCM into a seekable 16-bit PCM **WAV** container and computing a
+ * bounded, downsampled waveform **peak** array at capture time.
+ *
+ * These functions are deliberately free of any browser API (`AudioContext`, `Blob`, …) so they can
+ * be unit-tested in a plain Node environment. {@link RealtimeAudioRecorder} owns the live audio
+ * graph and feeds the captured samples here.
+ *
+ * Why WAV (PCM) over webm/opus: a RIFF/WAVE header carries the exact duration and maps a byte
+ * offset linearly to a time offset, so HTTP-range seeking is exact (no client-side duration nudge),
+ * and peak extraction is trivial. The tradeoff is size — PCM WAV is larger than opus — which is
+ * acceptable for these short session recordings.
+ *
+ * @module @memberjunction/ng-conversations/realtime-pcm-wav
+ */
+/** Canonical WAV/PCM header size in bytes (RIFF + fmt + data chunk headers). */
+export declare const WAV_HEADER_BYTES = 44;
+/**
+ * Encodes mono Float32 PCM samples as a canonical single-chunk 16-bit PCM WAV (RIFF/WAVE).
+ * The returned `ArrayBuffer` is `WAV_HEADER_BYTES + samples.length * 2` bytes — a proper
+ * 44-byte header followed by little-endian PCM16 data, so it is seekable and self-describing.
+ *
+ * @param samples Mono Float32 PCM in `[-1, 1]` (downmixed by the caller).
+ * @param sampleRate Sample rate in Hz (from the capturing `AudioContext`).
+ * @returns A standalone `ArrayBuffer` containing the full WAV file.
+ */
+export declare function encodePcm16Wav(samples: Float32Array, sampleRate: number): ArrayBuffer;
+/**
+ * A bounded, streaming waveform-peak accumulator. It maintains a fixed number of buckets
+ * (`targetBuckets`) of max-absolute amplitude across the WHOLE recording regardless of length:
+ * once every bucket has absorbed `samplesPerBucket` samples, the buckets are halved (pairwise
+ * max-merged) and the per-bucket capacity doubles — so memory + cost stay O(targetBuckets) for an
+ * arbitrarily long call. {@link Normalize} returns the final 0..1 peaks.
+ */
+export declare class PeakAccumulator {
+    /** Target (max) number of buckets in the produced peak array. */
+    private readonly targetBuckets;
+    /** Current max-abs amplitude per bucket (length grows up to `2 * targetBuckets` before folding). */
+    private buckets;
+    /** How many samples each bucket currently absorbs before a new bucket opens. */
+    private samplesPerBucket;
+    /** Samples absorbed into the current (last) bucket so far. */
+    private samplesInCurrent;
+    /**
+     * @param targetBuckets Desired bucket count (waveform resolution). Clamped to >= 1.
+     *   ~600 gives a smooth overview waveform at trivial cost.
+     */
+    constructor(targetBuckets?: number);
+    /** Number of buckets currently held (for tests/inspection). */
+    get BucketCount(): number;
+    /** Absorbs a block of Float32 PCM samples, updating the running per-bucket max-abs. */
+    Push(samples: Float32Array): void;
+    /** Absorbs one sample into the current bucket, opening/folding buckets as capacity is reached. */
+    private pushSample;
+    /** Pairwise max-merges adjacent buckets, halving the count and doubling per-bucket capacity. */
+    private fold;
+    /**
+     * Produces the final normalized peak array (each value 0..1, divided by the global max).
+     * Trailing empty buckets are trimmed; an all-silent / empty recording yields `[]`.
+     */
+    Normalize(): number[];
+}
+/**
+ * One-shot peak downsampling of a complete Float32 PCM buffer into `targetBuckets` normalized
+ * max-abs peaks (0..1). The streaming {@link PeakAccumulator} is the production path; this is the
+ * direct, allocation-light equivalent used where the whole buffer is already in hand (and in tests).
+ *
+ * @param samples The full mono Float32 PCM buffer.
+ * @param targetBuckets Desired bucket count (clamped to >= 1). Buckets beyond the sample count are
+ *   not produced — a short buffer yields at most `samples.length` peaks.
+ * @returns Normalized peaks; `[]` for an empty or all-silent buffer.
+ */
+export declare function downsamplePeaks(samples: Float32Array, targetBuckets?: number): number[];
+//# sourceMappingURL=realtime-pcm-wav.d.ts.map

package/dist/lib/services/realtime-pcm-wav.d.ts.map

@@ -0,0 +1 @@
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package/dist/lib/services/realtime-pcm-wav.js

@@ -0,0 +1,205 @@
+/**
+ * @fileoverview Pure, browser-and-node-safe helpers for the CLIENT-DIRECT realtime recorder:
+ * encoding accumulated Float32 PCM into a seekable 16-bit PCM **WAV** container and computing a
+ * bounded, downsampled waveform **peak** array at capture time.
+ *
+ * These functions are deliberately free of any browser API (`AudioContext`, `Blob`, …) so they can
+ * be unit-tested in a plain Node environment. {@link RealtimeAudioRecorder} owns the live audio
+ * graph and feeds the captured samples here.
+ *
+ * Why WAV (PCM) over webm/opus: a RIFF/WAVE header carries the exact duration and maps a byte
+ * offset linearly to a time offset, so HTTP-range seeking is exact (no client-side duration nudge),
+ * and peak extraction is trivial. The tradeoff is size — PCM WAV is larger than opus — which is
+ * acceptable for these short session recordings.
+ *
+ * @module @memberjunction/ng-conversations/realtime-pcm-wav
+ */
+/** Lowest value representable by signed 16-bit PCM. */
+const PCM16_MIN = -32768;
+/** Highest value representable by signed 16-bit PCM. */
+const PCM16_MAX = 32767;
+/** Canonical WAV/PCM header size in bytes (RIFF + fmt + data chunk headers). */
+export const WAV_HEADER_BYTES = 44;
+/**
+ * Converts one Float32 sample (nominally in `[-1, 1]`) to a clamped signed 16-bit integer.
+ * Out-of-range values (a hot mix can briefly exceed unity) are clamped, not wrapped.
+ */
+function floatToPcm16(sample) {
+    const clamped = sample < -1 ? -1 : sample > 1 ? 1 : sample;
+    // Asymmetric scaling: negative full-scale is -32768, positive is +32767.
+    const scaled = clamped < 0 ? clamped * -PCM16_MIN : clamped * PCM16_MAX;
+    const rounded = Math.round(scaled);
+    return rounded < PCM16_MIN ? PCM16_MIN : rounded > PCM16_MAX ? PCM16_MAX : rounded;
+}
+/**
+ * Encodes mono Float32 PCM samples as a canonical single-chunk 16-bit PCM WAV (RIFF/WAVE).
+ * The returned `ArrayBuffer` is `WAV_HEADER_BYTES + samples.length * 2` bytes — a proper
+ * 44-byte header followed by little-endian PCM16 data, so it is seekable and self-describing.
+ *
+ * @param samples Mono Float32 PCM in `[-1, 1]` (downmixed by the caller).
+ * @param sampleRate Sample rate in Hz (from the capturing `AudioContext`).
+ * @returns A standalone `ArrayBuffer` containing the full WAV file.
+ */
+export function encodePcm16Wav(samples, sampleRate) {
+    const numChannels = 1;
+    const bitsPerSample = 16;
+    const bytesPerSample = bitsPerSample / 8;
+    const blockAlign = numChannels * bytesPerSample;
+    const byteRate = sampleRate * blockAlign;
+    const dataSize = samples.length * bytesPerSample;
+    const buffer = new ArrayBuffer(WAV_HEADER_BYTES + dataSize);
+    const view = new DataView(buffer);
+    writeAscii(view, 0, 'RIFF');
+    view.setUint32(4, 36 + dataSize, true); // ChunkSize = 36 + Subchunk2Size
+    writeAscii(view, 8, 'WAVE');
+    writeAscii(view, 12, 'fmt ');
+    view.setUint32(16, 16, true); // Subchunk1Size (16 for PCM)
+    view.setUint16(20, 1, true); // AudioFormat = PCM
+    view.setUint16(22, numChannels, true);
+    view.setUint32(24, sampleRate, true);
+    view.setUint32(28, byteRate, true);
+    view.setUint16(32, blockAlign, true);
+    view.setUint16(34, bitsPerSample, true);
+    writeAscii(view, 36, 'data');
+    view.setUint32(40, dataSize, true);
+    let offset = WAV_HEADER_BYTES;
+    for (let i = 0; i < samples.length; i++) {
+        view.setInt16(offset, floatToPcm16(samples[i]), true);
+        offset += bytesPerSample;
+    }
+    return buffer;
+}
+/** Writes an ASCII string into a {@link DataView} byte-by-byte at `offset`. */
+function writeAscii(view, offset, text) {
+    for (let i = 0; i < text.length; i++) {
+        view.setUint8(offset + i, text.charCodeAt(i));
+    }
+}
+/**
+ * A bounded, streaming waveform-peak accumulator. It maintains a fixed number of buckets
+ * (`targetBuckets`) of max-absolute amplitude across the WHOLE recording regardless of length:
+ * once every bucket has absorbed `samplesPerBucket` samples, the buckets are halved (pairwise
+ * max-merged) and the per-bucket capacity doubles — so memory + cost stay O(targetBuckets) for an
+ * arbitrarily long call. {@link Normalize} returns the final 0..1 peaks.
+ */
+export class PeakAccumulator {
+    /** Target (max) number of buckets in the produced peak array. */
+    targetBuckets;
+    /** Current max-abs amplitude per bucket (length grows up to `2 * targetBuckets` before folding). */
+    buckets = [];
+    /** How many samples each bucket currently absorbs before a new bucket opens. */
+    samplesPerBucket = 1;
+    /** Samples absorbed into the current (last) bucket so far. */
+    samplesInCurrent = 0;
+    /**
+     * @param targetBuckets Desired bucket count (waveform resolution). Clamped to >= 1.
+     *   ~600 gives a smooth overview waveform at trivial cost.
+     */
+    constructor(targetBuckets = 600) {
+        this.targetBuckets = Math.max(1, Math.floor(targetBuckets));
+    }
+    /** Number of buckets currently held (for tests/inspection). */
+    get BucketCount() {
+        return this.buckets.length;
+    }
+    /** Absorbs a block of Float32 PCM samples, updating the running per-bucket max-abs. */
+    Push(samples) {
+        for (let i = 0; i < samples.length; i++) {
+            this.pushSample(samples[i]);
+        }
+    }
+    /** Absorbs one sample into the current bucket, opening/folding buckets as capacity is reached. */
+    pushSample(sample) {
+        if (this.buckets.length === 0) {
+            this.buckets.push(0);
+            this.samplesInCurrent = 0;
+        }
+        const abs = sample < 0 ? -sample : sample;
+        const last = this.buckets.length - 1;
+        if (abs > this.buckets[last]) {
+            this.buckets[last] = abs;
+        }
+        this.samplesInCurrent++;
+        if (this.samplesInCurrent >= this.samplesPerBucket) {
+            this.buckets.push(0);
+            this.samplesInCurrent = 0;
+            // Keep the bucket array bounded: once it reaches 2x the target, fold pairwise so the
+            // resolution stays under the cap while the per-bucket sample capacity doubles.
+            if (this.buckets.length >= this.targetBuckets * 2) {
+                this.fold();
+            }
+        }
+    }
+    /** Pairwise max-merges adjacent buckets, halving the count and doubling per-bucket capacity. */
+    fold() {
+        const folded = [];
+        for (let i = 0; i < this.buckets.length; i += 2) {
+            const a = this.buckets[i];
+            const b = i + 1 < this.buckets.length ? this.buckets[i + 1] : 0;
+            folded.push(a > b ? a : b);
+        }
+        this.buckets = folded;
+        this.samplesPerBucket *= 2;
+        this.samplesInCurrent = 0;
+    }
+    /**
+     * Produces the final normalized peak array (each value 0..1, divided by the global max).
+     * Trailing empty buckets are trimmed; an all-silent / empty recording yields `[]`.
+     */
+    Normalize() {
+        // Drop a trailing freshly-opened empty bucket that absorbed nothing.
+        const raw = this.buckets.length > 0 && this.samplesInCurrent === 0 && this.buckets.length > 1
+            ? this.buckets.slice(0, -1)
+            : this.buckets.slice();
+        let max = 0;
+        for (const v of raw) {
+            if (v > max) {
+                max = v;
+            }
+        }
+        if (max <= 0) {
+            return [];
+        }
+        return raw.map(v => v / max);
+    }
+}
+/**
+ * One-shot peak downsampling of a complete Float32 PCM buffer into `targetBuckets` normalized
+ * max-abs peaks (0..1). The streaming {@link PeakAccumulator} is the production path; this is the
+ * direct, allocation-light equivalent used where the whole buffer is already in hand (and in tests).
+ *
+ * @param samples The full mono Float32 PCM buffer.
+ * @param targetBuckets Desired bucket count (clamped to >= 1). Buckets beyond the sample count are
+ *   not produced — a short buffer yields at most `samples.length` peaks.
+ * @returns Normalized peaks; `[]` for an empty or all-silent buffer.
+ */
+export function downsamplePeaks(samples, targetBuckets = 600) {
+    const target = Math.max(1, Math.floor(targetBuckets));
+    if (samples.length === 0) {
+        return [];
+    }
+    const buckets = Math.min(target, samples.length);
+    const samplesPerBucket = samples.length / buckets;
+    const peaks = new Array(buckets).fill(0);
+    let globalMax = 0;
+    for (let b = 0; b < buckets; b++) {
+        const start = Math.floor(b * samplesPerBucket);
+        const end = b === buckets - 1 ? samples.length : Math.floor((b + 1) * samplesPerBucket);
+        let localMax = 0;
+        for (let i = start; i < end; i++) {
+            const abs = samples[i] < 0 ? -samples[i] : samples[i];
+            if (abs > localMax) {
+                localMax = abs;
+            }
+        }
+        peaks[b] = localMax;
+        if (localMax > globalMax) {
+            globalMax = localMax;
+        }
+    }
+    if (globalMax <= 0) {
+        return [];
+    }
+    return peaks.map(v => v / globalMax);
+}
+//# sourceMappingURL=realtime-pcm-wav.js.map

package/dist/lib/services/realtime-pcm-wav.js.map

@@ -0,0 +1 @@
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* @fileoverview Pure, browser-and-node-safe helpers for the CLIENT-DIRECT realtime recorder:\n * encoding accumulated Float32 PCM into a seekable 16-bit PCM **WAV** container and computing a\n * bounded, downsampled waveform **peak** array at capture time.\n *\n * These functions are deliberately free of any browser API (`AudioContext`, `Blob`, …) so they can\n * be unit-tested in a plain Node environment. {@link RealtimeAudioRecorder} owns the live audio\n * graph and feeds the captured samples here.\n *\n * Why WAV (PCM) over webm/opus: a RIFF/WAVE header carries the exact duration and maps a byte\n * offset linearly to a time offset, so HTTP-range seeking is exact (no client-side duration nudge),\n * and peak extraction is trivial. The tradeoff is size — PCM WAV is larger than opus — which is\n * acceptable for these short session recordings.\n *\n * @module @memberjunction/ng-conversations/realtime-pcm-wav\n */\n\n/** Lowest value representable by signed 16-bit PCM. */\nconst PCM16_MIN = -32768;\n/** Highest value representable by signed 16-bit PCM. */\nconst PCM16_MAX = 32767;\n/** Canonical WAV/PCM header size in bytes (RIFF + fmt + data chunk headers). */\nexport const WAV_HEADER_BYTES = 44;\n\n/**\n * Converts one Float32 sample (nominally in `[-1, 1]`) to a clamped signed 16-bit integer.\n * Out-of-range values (a hot mix can briefly exceed unity) are clamped, not wrapped.\n */\nfunction floatToPcm16(sample: number): number {\n    const clamped = sample < -1 ? -1 : sample > 1 ? 1 : sample;\n    // Asymmetric scaling: negative full-scale is -32768, positive is +32767.\n    const scaled = clamped < 0 ? clamped * -PCM16_MIN : clamped * PCM16_MAX;\n    const rounded = Math.round(scaled);\n    return rounded < PCM16_MIN ? PCM16_MIN : rounded > PCM16_MAX ? PCM16_MAX : rounded;\n}\n\n/**\n * Encodes mono Float32 PCM samples as a canonical single-chunk 16-bit PCM WAV (RIFF/WAVE).\n * The returned `ArrayBuffer` is `WAV_HEADER_BYTES + samples.length * 2` bytes — a proper\n * 44-byte header followed by little-endian PCM16 data, so it is seekable and self-describing.\n *\n * @param samples Mono Float32 PCM in `[-1, 1]` (downmixed by the caller).\n * @param sampleRate Sample rate in Hz (from the capturing `AudioContext`).\n * @returns A standalone `ArrayBuffer` containing the full WAV file.\n */\nexport function encodePcm16Wav(samples: Float32Array, sampleRate: number): ArrayBuffer {\n    const numChannels = 1;\n    const bitsPerSample = 16;\n    const bytesPerSample = bitsPerSample / 8;\n    const blockAlign = numChannels * bytesPerSample;\n    const byteRate = sampleRate * blockAlign;\n    const dataSize = samples.length * bytesPerSample;\n\n    const buffer = new ArrayBuffer(WAV_HEADER_BYTES + dataSize);\n    const view = new DataView(buffer);\n\n    writeAscii(view, 0, 'RIFF');\n    view.setUint32(4, 36 + dataSize, true); // ChunkSize = 36 + Subchunk2Size\n    writeAscii(view, 8, 'WAVE');\n\n    writeAscii(view, 12, 'fmt ');\n    view.setUint32(16, 16, true); // Subchunk1Size (16 for PCM)\n    view.setUint16(20, 1, true); // AudioFormat = PCM\n    view.setUint16(22, numChannels, true);\n    view.setUint32(24, sampleRate, true);\n    view.setUint32(28, byteRate, true);\n    view.setUint16(32, blockAlign, true);\n    view.setUint16(34, bitsPerSample, true);\n\n    writeAscii(view, 36, 'data');\n    view.setUint32(40, dataSize, true);\n\n    let offset = WAV_HEADER_BYTES;\n    for (let i = 0; i < samples.length; i++) {\n        view.setInt16(offset, floatToPcm16(samples[i]), true);\n        offset += bytesPerSample;\n    }\n\n    return buffer;\n}\n\n/** Writes an ASCII string into a {@link DataView} byte-by-byte at `offset`. */\nfunction writeAscii(view: DataView, offset: number, text: string): void {\n    for (let i = 0; i < text.length; i++) {\n        view.setUint8(offset + i, text.charCodeAt(i));\n    }\n}\n\n/**\n * A bounded, streaming waveform-peak accumulator. It maintains a fixed number of buckets\n * (`targetBuckets`) of max-absolute amplitude across the WHOLE recording regardless of length:\n * once every bucket has absorbed `samplesPerBucket` samples, the buckets are halved (pairwise\n * max-merged) and the per-bucket capacity doubles — so memory + cost stay O(targetBuckets) for an\n * arbitrarily long call. {@link Normalize} returns the final 0..1 peaks.\n */\nexport class PeakAccumulator {\n    /** Target (max) number of buckets in the produced peak array. */\n    private readonly targetBuckets: number;\n    /** Current max-abs amplitude per bucket (length grows up to `2 * targetBuckets` before folding). */\n    private buckets: number[] = [];\n    /** How many samples each bucket currently absorbs before a new bucket opens. */\n    private samplesPerBucket = 1;\n    /** Samples absorbed into the current (last) bucket so far. */\n    private samplesInCurrent = 0;\n\n    /**\n     * @param targetBuckets Desired bucket count (waveform resolution). Clamped to >= 1.\n     *   ~600 gives a smooth overview waveform at trivial cost.\n     */\n    constructor(targetBuckets = 600) {\n        this.targetBuckets = Math.max(1, Math.floor(targetBuckets));\n    }\n\n    /** Number of buckets currently held (for tests/inspection). */\n    public get BucketCount(): number {\n        return this.buckets.length;\n    }\n\n    /** Absorbs a block of Float32 PCM samples, updating the running per-bucket max-abs. */\n    public Push(samples: Float32Array): void {\n        for (let i = 0; i < samples.length; i++) {\n            this.pushSample(samples[i]);\n        }\n    }\n\n    /** Absorbs one sample into the current bucket, opening/folding buckets as capacity is reached. */\n    private pushSample(sample: number): void {\n        if (this.buckets.length === 0) {\n            this.buckets.push(0);\n            this.samplesInCurrent = 0;\n        }\n        const abs = sample < 0 ? -sample : sample;\n        const last = this.buckets.length - 1;\n        if (abs > this.buckets[last]) {\n            this.buckets[last] = abs;\n        }\n        this.samplesInCurrent++;\n        if (this.samplesInCurrent >= this.samplesPerBucket) {\n            this.buckets.push(0);\n            this.samplesInCurrent = 0;\n            // Keep the bucket array bounded: once it reaches 2x the target, fold pairwise so the\n            // resolution stays under the cap while the per-bucket sample capacity doubles.\n            if (this.buckets.length >= this.targetBuckets * 2) {\n                this.fold();\n            }\n        }\n    }\n\n    /** Pairwise max-merges adjacent buckets, halving the count and doubling per-bucket capacity. */\n    private fold(): void {\n        const folded: number[] = [];\n        for (let i = 0; i < this.buckets.length; i += 2) {\n            const a = this.buckets[i];\n            const b = i + 1 < this.buckets.length ? this.buckets[i + 1] : 0;\n            folded.push(a > b ? a : b);\n        }\n        this.buckets = folded;\n        this.samplesPerBucket *= 2;\n        this.samplesInCurrent = 0;\n    }\n\n    /**\n     * Produces the final normalized peak array (each value 0..1, divided by the global max).\n     * Trailing empty buckets are trimmed; an all-silent / empty recording yields `[]`.\n     */\n    public Normalize(): number[] {\n        // Drop a trailing freshly-opened empty bucket that absorbed nothing.\n        const raw = this.buckets.length > 0 && this.samplesInCurrent === 0 && this.buckets.length > 1\n            ? this.buckets.slice(0, -1)\n            : this.buckets.slice();\n        let max = 0;\n        for (const v of raw) {\n            if (v > max) {\n                max = v;\n            }\n        }\n        if (max <= 0) {\n            return [];\n        }\n        return raw.map(v => v / max);\n    }\n}\n\n/**\n * One-shot peak downsampling of a complete Float32 PCM buffer into `targetBuckets` normalized\n * max-abs peaks (0..1). The streaming {@link PeakAccumulator} is the production path; this is the\n * direct, allocation-light equivalent used where the whole buffer is already in hand (and in tests).\n *\n * @param samples The full mono Float32 PCM buffer.\n * @param targetBuckets Desired bucket count (clamped to >= 1). Buckets beyond the sample count are\n *   not produced — a short buffer yields at most `samples.length` peaks.\n * @returns Normalized peaks; `[]` for an empty or all-silent buffer.\n */\nexport function downsamplePeaks(samples: Float32Array, targetBuckets = 600): number[] {\n    const target = Math.max(1, Math.floor(targetBuckets));\n    if (samples.length === 0) {\n        return [];\n    }\n    const buckets = Math.min(target, samples.length);\n    const samplesPerBucket = samples.length / buckets;\n    const peaks = new Array<number>(buckets).fill(0);\n    let globalMax = 0;\n    for (let b = 0; b < buckets; b++) {\n        const start = Math.floor(b * samplesPerBucket);\n        const end = b === buckets - 1 ? samples.length : Math.floor((b + 1) * samplesPerBucket);\n        let localMax = 0;\n        for (let i = start; i < end; i++) {\n            const abs = samples[i] < 0 ? -samples[i] : samples[i];\n            if (abs > localMax) {\n                localMax = abs;\n            }\n        }\n        peaks[b] = localMax;\n        if (localMax > globalMax) {\n            globalMax = localMax;\n        }\n    }\n    if (globalMax <= 0) {\n        return [];\n    }\n    return peaks.map(v => v / globalMax);\n}\n"]}

package/dist/lib/services/realtime-session-review.service.d.ts

@@ -8,14 +8,31 @@ import * as i0 from "@angular/core";
  * legacy row was closed before the column existed.
  */
 export type RealtimeReviewCloseReason = 'Error' | 'Explicit' | 'Janitor' | 'Shutdown';
-/** One historical caption turn of a reviewed session (a persisted `MJ: Conversation Details` row). */
+/**
+ * One historical caption turn of a reviewed session (a persisted `MJ: Conversation Details` row).
+ *
+ * Carries BOTH the thread-rendering fields (`Role`/`Text`/`At`) AND the raw fields the time-aligned
+ * evidence player reads (`ID`, `Message`, `__mj_CreatedAt`, `UtteranceStartMs`, `UtteranceEndMs`) so
+ * a review's turns can be handed straight to {@link RealtimeEvidencePlaybackComponent} for the
+ * Recording tab — this shape structurally satisfies that player's `EvidencePlaybackTurn` input.
+ */
 export interface RealtimeSessionReviewTurn {
+    /** `MJ: Conversation Details.ID` — the player's `@for` track key. */
+    ID: string;
     /** Who spoke the turn. Persisted `AI` rows map to `Assistant`. */
     Role: 'User' | 'Assistant';
     /** The transcript text. */
     Text: string;
+    /** The raw transcript text (mirrors {@link Text}) — the player reads `Message`. */
+    Message: string | null;
+    /** When the turn was persisted (`__mj_CreatedAt`), or null when unknown. Aliases {@link At}. */
+    __mj_CreatedAt: Date | null;
     /** When the turn was persisted (`__mj_CreatedAt`), or null when unknown. */
     At: Date | null;
+    /** Precise media-relative start (ms from recording t0), or null when the driver supplied no timing. */
+    UtteranceStartMs: number | null;
+    /** Precise media-relative end (ms from recording t0), or null. */
+    UtteranceEndMs: number | null;
 }
 /**
  * A compact preview of one DELEGATED agent run the reviewed session spawned (an
@@ -108,6 +125,16 @@ export interface RealtimeSessionReview {
     StartedAt: Date | null;
     LastActiveAt: Date | null;
     ClosedAt: Date | null;
+    /**
+     * The `MJ: Files` id of the reviewed session's recorded audio, or null when nothing was recorded.
+     * When set, review mode resolves a signed download URL and shows a Recording tab with the
+     * time-aligned evidence player. From the PRIMARY (reviewed) leg only.
+     */
+    RecordingFileID: string | null;
+    /** Recording alignment origin (t0), used by the playback transcript sync. From the primary leg. */
+    RecordingStartedAt: Date | null;
+    /** What was captured (`Audio` / `AudioVideo`), or null. From the primary leg. */
+    RecordingMedia: string | null;
     /** Chronological caption turns (oldest first) — ALL legs of the chain, flattened. */
     Turns: RealtimeSessionReviewTurn[];
     /** Delegated run previews (oldest first, all legs), minus each leg's co-agent observability run. */

package/dist/lib/services/realtime-session-review.service.d.ts.map

@@ -1 +1 @@
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package/dist/lib/services/realtime-session-review.service.js

@@ -33,8 +33,8 @@ export function MergeChainChannelStates(legChannelStates) {
     }
     return [...byName.values()];
 }
-const SESSION_FIELDS = ['ID', 'AgentID', 'Agent', 'Status', 'ConversationID', 'Config', 'LastSessionID', 'LastActiveAt', 'ClosedAt', 'CloseReason', '__mj_CreatedAt'];
-const DETAIL_FIELDS = ['ID', 'Role', 'Message', 'HiddenToUser', '__mj_CreatedAt'];
+const SESSION_FIELDS = ['ID', 'AgentID', 'Agent', 'Status', 'ConversationID', 'Config', 'LastSessionID', 'LastActiveAt', 'ClosedAt', 'CloseReason', 'RecordingFileID', 'RecordingStartedAt', 'RecordingMedia', '__mj_CreatedAt'];
+const DETAIL_FIELDS = ['ID', 'Role', 'Message', 'HiddenToUser', 'UserID', 'Status', 'UtteranceStartMs', 'UtteranceEndMs', '__mj_CreatedAt'];
 const RUN_FIELDS = ['ID', 'AgentID', 'Agent', 'Status', 'Success', 'Message', 'ErrorMessage', 'FinalStep', 'StartedAt', 'CompletedAt'];
 const CHANNEL_FIELDS = ['ID', 'Channel', 'Config'];
 const JUNCTION_FIELDS = ['ID', 'ConversationDetailID', 'ArtifactVersionID'];
@@ -326,6 +326,9 @@ export class RealtimeSessionReviewService {
             StartedAt: this.toDate(session.__mj_CreatedAt),
             LastActiveAt: this.toDate(session.LastActiveAt),
             ClosedAt: this.toDate(session.ClosedAt),
+            RecordingFileID: session.RecordingFileID ?? null,
+            RecordingStartedAt: this.toDate(session.RecordingStartedAt),
+            RecordingMedia: session.RecordingMedia ?? null,
             Turns: legs.flatMap(l => l.Turns),
             DelegatedRuns: legs.flatMap(l => l.DelegatedRuns),
             // Multi-leg: newest leg with a SAVED state wins per channel — a final leg that
@@ -355,10 +358,16 @@ export class RealtimeSessionReviewService {
             if (row.HiddenToUser || text.length === 0 || (row.Role !== 'User' && row.Role !== 'AI')) {
                 continue;
             }
+            const createdAt = this.toDate(row.__mj_CreatedAt);
             turns.push({
+                ID: row.ID,
                 Role: row.Role === 'AI' ? 'Assistant' : 'User',
                 Text: text,
-                At: this.toDate(row.__mj_CreatedAt)
+                Message: text,
+                __mj_CreatedAt: createdAt,
+                At: createdAt,
+                UtteranceStartMs: row.UtteranceStartMs ?? null,
+                UtteranceEndMs: row.UtteranceEndMs ?? null
             });
         }
         return turns;