@memberjunction/ng-conversations 5.43.0 → 5.44.0

package/dist/lib/services/realtime-audio-recorder.js

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+import { encodePcm16Wav, PeakAccumulator } from './realtime-pcm-wav';
+/**
+ * Browser-side audio recorder for a CLIENT-DIRECT realtime voice session.
+ *
+ * Mixes the user's microphone stream with the agent's remote-audio stream (when the driver
+ * exposes one — `OpenAIRealtimeClient.GetRemoteMediaStream()`) into a single mono **WAV** (16-bit
+ * PCM) blob via the Web Audio API. Rather than a `MediaRecorder` (which only emits webm/opus —
+ * header-less, no duration/cues, so HTTP-range seeking is unreliable), the mix is routed into an
+ * {@link AudioWorkletNode} (preferred) or a {@link ScriptProcessorNode} (fallback) that captures
+ * Float32 PCM frames. PCM is accumulated in memory and encoded as a seekable RIFF/WAVE file at
+ * {@link Stop}. WAV's header carries the exact duration and maps byte-offset → time linearly, so
+ * range seeking is exact and waveform peaks are computed cheaply during capture. The tradeoff is
+ * size (PCM WAV is larger than opus) — acceptable for these short session recordings.
+ *
+ * Robust + MVP by design — every step degrades gracefully:
+ * - No `AudioContext` (server-side render / unsupported env) → the recorder disables itself
+ *   ({@link IsRecording} stays `false`, {@link Stop} resolves `null`); the voice call proceeds.
+ * - No `AudioWorkletNode` AND no `ScriptProcessorNode` → disables itself rather than throwing.
+ * - No remote stream (non-WebRTC drivers, or the track hasn't landed yet) → records mic only.
+ * - Any construction/start error is logged and disables the recorder rather than throwing into
+ *   the session-start path.
+ *
+ * Not an Angular service: it owns transient per-session media plumbing the
+ * {@link RealtimeSessionService} creates and discards per call, so a plain class is the right
+ * shape (no DI, no singleton).
+ */
+export class RealtimeAudioRecorder {
+    /** The container/codec we now produce — a seekable mono 16-bit PCM WAV. */
+    static WavMimeType = 'audio/wav';
+    /** Target waveform-peak resolution (buckets) across the whole recording. */
+    static PeakBuckets = 600;
+    /** PCM frame size for the ScriptProcessor fallback (worklet uses its native 128-frame quantum). */
+    static ScriptProcessorBufferSize = 4096;
+    /** Inline AudioWorklet processor: forwards each 128-frame mono input block to the main thread. */
+    static WorkletProcessorSource = `
+        class MJPcmCaptureProcessor extends AudioWorkletProcessor {
+            process(inputs) {
+                const input = inputs[0];
+                if (input && input[0] && input[0].length > 0) {
+                    // Copy out of the reused worklet buffer before posting.
+                    this.port.postMessage(input[0].slice(0));
+                }
+                return true;
+            }
+        }
+        registerProcessor('mj-pcm-capture', MJPcmCaptureProcessor);
+    `;
+    audioContext = null;
+    destination = null;
+    workletNode = null;
+    scriptNode = null;
+    /** Captured mono Float32 PCM frames (in capture order); concatenated + encoded to WAV at Stop. */
+    pcmFrames = [];
+    /** Total captured sample count across {@link pcmFrames} (kept in step to avoid re-summing). */
+    totalSamples = 0;
+    /** Streaming waveform-peak accumulator — bounded regardless of recording length. */
+    peaks = new PeakAccumulator(RealtimeAudioRecorder.PeakBuckets);
+    /** Final normalized peaks, snapshotted at {@link Stop} so {@link GetPeaks} survives cleanup. */
+    finalPeaks = [];
+    recording = false;
+    /** True once the agent's remote stream has been wired into the mix (prevents double-attach). */
+    remoteAttached = false;
+    /** A remote stream handed to AttachRemoteStream before the audio graph was ready; wired at setup. */
+    pendingRemoteStream = null;
+    /** Sample index up to which crash-recovery shards have already been emitted (the segment cursor). */
+    flushedSampleCount = 0;
+    /** Wall-clock ms at recording start; anchors {@link NowOffsetMs}. 0 before {@link Start}. */
+    startedAtMs = 0;
+    /** Sample rate of the capturing {@link AudioContext}; 0 before {@link Start}. Needed to wrap shards. */
+    sampleRate = 0;
+    /** True while a recorder is actively capturing. */
+    get IsRecording() {
+        return this.recording;
+    }
+    /** Wall-clock ms when recording started (0 before {@link Start}). */
+    get StartedAtMs() {
+        return this.startedAtMs;
+    }
+    /**
+     * The container/codec the active recording uses. Always `'audio/wav'` once recording (empty
+     * string before {@link Start} / when disabled) — the consolidated upload and the WAV-wrapped
+     * crash-recovery shards are all 16-bit PCM WAV.
+     */
+    get MimeType() {
+        return this.recording ? RealtimeAudioRecorder.WavMimeType : '';
+    }
+    /**
+     * The captured AudioContext sample rate (0 before {@link Start} / when disabled). Exposed so a
+     * recovery path can WAV-wrap the raw-PCM shards with the correct rate.
+     */
+    get SampleRate() {
+        return this.sampleRate;
+    }
+    /**
+     * Begins recording a mix of `micStream` and (when present) `remoteStream`. Best-effort:
+     * a missing `AudioContext` / no worklet+scriptprocessor support / any setup failure disables
+     * the recorder silently (logged) so the live call is never blocked.
+     *
+     * @param micStream The user's microphone capture stream (required).
+     * @param remoteStream The agent's remote-audio stream, or `null` for mic-only capture.
+     */
+    Start(micStream, remoteStream) {
+        if (this.recording) {
+            return;
+        }
+        if (typeof AudioContext === 'undefined') {
+            console.warn('[RealtimeAudioRecorder] AudioContext unavailable — recording disabled for this session.');
+            return;
+        }
+        // Mark recording + stamp t0 SYNCHRONOUSLY so a caller that checks IsRecording immediately after
+        // Start() (the session service does) sees `true`, even though the audio-graph setup
+        // (AudioContext.resume + worklet module load) completes asynchronously below. Without this the
+        // caller discards the recorder before it finishes starting and nothing is ever uploaded.
+        this.startedAtMs = Date.now();
+        this.finalPeaks = [];
+        this.recording = true;
+        // Fire-and-forget the async setup (it must `await audioContext.resume()` + worklet load); any
+        // failure disables the recorder rather than throwing into the session-start path.
+        void this.startMixedRecording(micStream, remoteStream).catch((error) => {
+            console.warn('[RealtimeAudioRecorder] Failed to start recording — disabling for this session:', error);
+            this.cleanup();
+        });
+    }
+    /**
+     * Wires the agent's remote-audio stream into the live mix once it arrives. The WebRTC
+     * `ontrack` that delivers the agent voice usually lands AFTER recording has already begun
+     * (mic-only), so the session attaches it here when it lands. Idempotent, and safe to call
+     * before the audio graph finishes its async setup (the stream is connected once ready).
+     */
+    AttachRemoteStream(stream) {
+        if (this.remoteAttached || stream.getAudioTracks().length === 0) {
+            return;
+        }
+        if (this.audioContext && this.captureNode()) {
+            this.connectStream(this.audioContext, stream);
+            this.remoteAttached = true;
+        }
+        else {
+            // Audio-graph setup still in flight (awaiting resume / worklet load) — connect at setup.
+            this.pendingRemoteStream = stream;
+        }
+    }
+    /**
+     * Returns a Blob of the raw PCM captured since the last call (a window of header-less,
+     * little-endian PCM16 bytes), advancing the segment cursor. `null` when nothing new. Used to
+     * upload crash-recovery shards mid-call. NOTE: shards are header-less raw PCM (NOT individually
+     * playable WAV) — recovery is concatenation in order, then WAV-wrapping with {@link SampleRate}.
+     * The canonical consolidated file is still authored as a single WAV from full memory at {@link Stop}.
+     */
+    SnapshotNewSegment() {
+        if (!this.recording || this.flushedSampleCount >= this.totalSamples) {
+            return null;
+        }
+        const fresh = this.collectSamples(this.flushedSampleCount, this.totalSamples);
+        this.flushedSampleCount = this.totalSamples;
+        if (fresh.length === 0) {
+            return null;
+        }
+        const pcm16 = this.floatToPcm16Bytes(fresh);
+        return new Blob([pcm16], { type: 'application/octet-stream' });
+    }
+    /** Ms elapsed since recording started (>= 0); 0 when not recording. */
+    NowOffsetMs() {
+        if (this.startedAtMs === 0) {
+            return 0;
+        }
+        return Math.max(0, Date.now() - this.startedAtMs);
+    }
+    /**
+     * The downsampled waveform peaks (each 0..1, max-abs per bucket, ~{@link RealtimeAudioRecorder.PeakBuckets}
+     * buckets across the whole recording). Computed incrementally during capture so it's cheap and
+     * bounded regardless of length. Returns `[]` before any audio / for a silent recording. Safe to
+     * call after {@link Stop} — the peaks are snapshotted there before {@link cleanup} clears state.
+     */
+    GetPeaks() {
+        return this.recording ? this.peaks.Normalize() : this.finalPeaks;
+    }
+    /**
+     * Stops recording and resolves the captured audio as a single mono 16-bit PCM **WAV**
+     * {@link Blob}, then releases the audio graph. Resolves `null` when nothing was recorded
+     * (recorder disabled, or no samples captured). Safe to call when not recording.
+     */
+    async Stop() {
+        if (!this.recording) {
+            this.cleanup();
+            return null;
+        }
+        // Detach the capture node so no further frames arrive while we assemble the file. Snapshot the
+        // peaks BEFORE cleanup so GetPeaks() still returns them for the post-Stop upload.
+        this.recording = false;
+        this.finalPeaks = this.peaks.Normalize();
+        const blob = this.encodeWavBlob();
+        this.cleanup();
+        return blob;
+    }
+    /** Wires up the Web Audio mix graph + PCM capture node (worklet preferred) and starts capture. */
+    async startMixedRecording(micStream, remoteStream) {
+        const audioContext = new AudioContext();
+        // CRITICAL: browsers create an AudioContext in the `suspended` state under the autoplay policy.
+        // Without resuming it, the mix graph never processes audio and we capture only silence. Starting
+        // a voice call is a user gesture, so resume() succeeds here.
+        if (audioContext.state === 'suspended') {
+            await audioContext.resume();
+        }
+        const destination = audioContext.createMediaStreamDestination();
+        // Publish the context + sample rate BEFORE wiring the remote so a concurrent AttachRemoteStream()
+        // (the agent track can land during the awaits above) sees a ready graph to connect to.
+        this.audioContext = audioContext;
+        this.destination = destination;
+        this.sampleRate = audioContext.sampleRate;
+        // Build the PCM capture node: AudioWorklet preferred, ScriptProcessor fallback. Either way the
+        // node both pulls audio from the mix AND drives the destination so the graph keeps processing.
+        await this.createCaptureNode(audioContext, destination);
+        this.connectStream(audioContext, micStream);
+        // Connect the agent's stream if it was passed now, or if AttachRemoteStream() stashed one while
+        // we were setting up. Either way, capture the agent voice — not just the mic.
+        const remote = remoteStream ?? this.pendingRemoteStream;
+        this.pendingRemoteStream = null;
+        if (remote && remote.getAudioTracks().length > 0) {
+            this.connectStream(audioContext, remote);
+            this.remoteAttached = true;
+        }
+    }
+    /**
+     * Creates the PCM capture node and inserts it into the graph: sources → captureNode → destination.
+     * Prefers an {@link AudioWorkletNode} (off-main-thread, no deprecation); falls back to a
+     * {@link ScriptProcessorNode} where worklets are unavailable. Throws when neither is supported
+     * (caught by {@link Start}, which then disables the recorder).
+     */
+    async createCaptureNode(audioContext, destination) {
+        if (await this.tryCreateWorkletNode(audioContext, destination)) {
+            return;
+        }
+        this.createScriptProcessorNode(audioContext, destination);
+    }
+    /** Attempts to build the AudioWorklet capture node; returns `false` (no throw) when unsupported. */
+    async tryCreateWorkletNode(audioContext, destination) {
+        if (!audioContext.audioWorklet || typeof AudioWorkletNode === 'undefined' || typeof Blob === 'undefined') {
+            return false;
+        }
+        try {
+            const moduleUrl = URL.createObjectURL(new Blob([RealtimeAudioRecorder.WorkletProcessorSource], { type: 'application/javascript' }));
+            try {
+                await audioContext.audioWorklet.addModule(moduleUrl);
+            }
+            finally {
+                URL.revokeObjectURL(moduleUrl);
+            }
+            const node = new AudioWorkletNode(audioContext, 'mj-pcm-capture', { numberOfInputs: 1, numberOfOutputs: 1 });
+            node.port.onmessage = (event) => this.captureFrame(event.data);
+            node.connect(destination);
+            this.workletNode = node;
+            return true;
+        }
+        catch (error) {
+            console.warn('[RealtimeAudioRecorder] AudioWorklet unavailable — falling back to ScriptProcessor:', error);
+            return false;
+        }
+    }
+    /** Builds the ScriptProcessor fallback capture node and inserts it into the graph. */
+    createScriptProcessorNode(audioContext, destination) {
+        if (typeof audioContext.createScriptProcessor !== 'function') {
+            throw new Error('Neither AudioWorklet nor ScriptProcessor is supported');
+        }
+        const node = audioContext.createScriptProcessor(RealtimeAudioRecorder.ScriptProcessorBufferSize, 1, 1);
+        node.onaudioprocess = (event) => {
+            // Copy out of the reused input buffer before retaining.
+            this.captureFrame(event.inputBuffer.getChannelData(0).slice(0));
+        };
+        // ScriptProcessor only fires onaudioprocess while connected to the destination.
+        node.connect(destination);
+        this.scriptNode = node;
+    }
+    /** The active capture node (worklet or script processor), or `null` before setup. */
+    captureNode() {
+        return this.workletNode ?? this.scriptNode;
+    }
+    /** Absorbs one captured PCM frame: accumulate samples + update the bounded peak array. */
+    captureFrame(frame) {
+        if (!this.recording || frame.length === 0) {
+            return;
+        }
+        // Defensive copy — worklet/script buffers are reused by the engine.
+        const copy = frame.slice(0);
+        this.pcmFrames.push(copy);
+        this.totalSamples += copy.length;
+        this.peaks.Push(copy);
+    }
+    /** Routes one stream's audio tracks into the capture node (no-op when track-less / no node). */
+    connectStream(audioContext, stream) {
+        const node = this.captureNode();
+        if (!node || stream.getAudioTracks().length === 0) {
+            return;
+        }
+        const source = audioContext.createMediaStreamSource(stream);
+        source.connect(node);
+    }
+    /** Concatenates a half-open sample range [from, to) across the captured frames into one Float32Array. */
+    collectSamples(from, to) {
+        const out = new Float32Array(Math.max(0, to - from));
+        if (out.length === 0) {
+            return out;
+        }
+        let cursor = 0; // absolute sample index at the start of the current frame
+        let written = 0;
+        for (const frame of this.pcmFrames) {
+            const frameStart = cursor;
+            const frameEnd = cursor + frame.length;
+            cursor = frameEnd;
+            if (frameEnd <= from) {
+                continue; // entirely before the window
+            }
+            if (frameStart >= to) {
+                break; // entirely after the window
+            }
+            const sliceStart = Math.max(0, from - frameStart);
+            const sliceEnd = Math.min(frame.length, to - frameStart);
+            for (let i = sliceStart; i < sliceEnd; i++) {
+                out[written++] = frame[i];
+            }
+        }
+        return written === out.length ? out : out.subarray(0, written);
+    }
+    /**
+     * Converts a mono Float32 buffer to a little-endian PCM16 `ArrayBuffer` (for raw crash-recovery
+     * shards). Returns the backing `ArrayBuffer` directly so it's a valid `BlobPart`.
+     */
+    floatToPcm16Bytes(samples) {
+        const buffer = new ArrayBuffer(samples.length * 2);
+        const view = new DataView(buffer);
+        for (let i = 0; i < samples.length; i++) {
+            const s = samples[i] < -1 ? -1 : samples[i] > 1 ? 1 : samples[i];
+            view.setInt16(i * 2, s < 0 ? s * 0x8000 : s * 0x7fff, true);
+        }
+        return buffer;
+    }
+    /** Encodes all accumulated PCM into a single seekable WAV blob, or `null` when nothing captured. */
+    encodeWavBlob() {
+        if (this.totalSamples === 0 || this.sampleRate === 0) {
+            return null;
+        }
+        const all = this.collectSamples(0, this.totalSamples);
+        const wav = encodePcm16Wav(all, this.sampleRate);
+        return new Blob([wav], { type: RealtimeAudioRecorder.WavMimeType });
+    }
+    /** Releases the audio graph + capture node and resets state. Idempotent. */
+    cleanup() {
+        this.recording = false;
+        this.pcmFrames = [];
+        this.totalSamples = 0;
+        this.flushedSampleCount = 0;
+        this.sampleRate = 0;
+        this.peaks = new PeakAccumulator(RealtimeAudioRecorder.PeakBuckets);
+        this.remoteAttached = false;
+        this.pendingRemoteStream = null;
+        if (this.workletNode) {
+            this.workletNode.port.onmessage = null;
+            try {
+                this.workletNode.disconnect();
+            }
+            catch { /* already disconnected */ }
+            this.workletNode = null;
+        }
+        if (this.scriptNode) {
+            this.scriptNode.onaudioprocess = null;
+            try {
+                this.scriptNode.disconnect();
+            }
+            catch { /* already disconnected */ }
+            this.scriptNode = null;
+        }
+        this.destination = null;
+        if (this.audioContext) {
+            void this.audioContext.close().catch(() => { });
+            this.audioContext = null;
+        }
+    }
+}
+//# sourceMappingURL=realtime-audio-recorder.js.map

package/dist/lib/services/realtime-audio-recorder.js.map

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{ encodePcm16Wav, PeakAccumulator } from './realtime-pcm-wav';\n\n/**\n * Browser-side audio recorder for a CLIENT-DIRECT realtime voice session.\n *\n * Mixes the user's microphone stream with the agent's remote-audio stream (when the driver\n * exposes one — `OpenAIRealtimeClient.GetRemoteMediaStream()`) into a single mono **WAV** (16-bit\n * PCM) blob via the Web Audio API. Rather than a `MediaRecorder` (which only emits webm/opus —\n * header-less, no duration/cues, so HTTP-range seeking is unreliable), the mix is routed into an\n * {@link AudioWorkletNode} (preferred) or a {@link ScriptProcessorNode} (fallback) that captures\n * Float32 PCM frames. PCM is accumulated in memory and encoded as a seekable RIFF/WAVE file at\n * {@link Stop}. WAV's header carries the exact duration and maps byte-offset → time linearly, so\n * range seeking is exact and waveform peaks are computed cheaply during capture. The tradeoff is\n * size (PCM WAV is larger than opus) — acceptable for these short session recordings.\n *\n * Robust + MVP by design — every step degrades gracefully:\n * - No `AudioContext` (server-side render / unsupported env) → the recorder disables itself\n *   ({@link IsRecording} stays `false`, {@link Stop} resolves `null`); the voice call proceeds.\n * - No `AudioWorkletNode` AND no `ScriptProcessorNode` → disables itself rather than throwing.\n * - No remote stream (non-WebRTC drivers, or the track hasn't landed yet) → records mic only.\n * - Any construction/start error is logged and disables the recorder rather than throwing into\n *   the session-start path.\n *\n * Not an Angular service: it owns transient per-session media plumbing the\n * {@link RealtimeSessionService} creates and discards per call, so a plain class is the right\n * shape (no DI, no singleton).\n */\nexport class RealtimeAudioRecorder {\n    /** The container/codec we now produce — a seekable mono 16-bit PCM WAV. */\n    private static readonly WavMimeType = 'audio/wav';\n    /** Target waveform-peak resolution (buckets) across the whole recording. */\n    private static readonly PeakBuckets = 600;\n    /** PCM frame size for the ScriptProcessor fallback (worklet uses its native 128-frame quantum). */\n    private static readonly ScriptProcessorBufferSize = 4096;\n    /** Inline AudioWorklet processor: forwards each 128-frame mono input block to the main thread. */\n    private static readonly WorkletProcessorSource = `\n        class MJPcmCaptureProcessor extends AudioWorkletProcessor {\n            process(inputs) {\n                const input = inputs[0];\n                if (input && input[0] && input[0].length > 0) {\n                    // Copy out of the reused worklet buffer before posting.\n                    this.port.postMessage(input[0].slice(0));\n                }\n                return true;\n            }\n        }\n        registerProcessor('mj-pcm-capture', MJPcmCaptureProcessor);\n    `;\n\n    private audioContext: AudioContext | null = null;\n    private destination: MediaStreamAudioDestinationNode | null = null;\n    private workletNode: AudioWorkletNode | null = null;\n    private scriptNode: ScriptProcessorNode | null = null;\n    /** Captured mono Float32 PCM frames (in capture order); concatenated + encoded to WAV at Stop. */\n    private pcmFrames: Float32Array[] = [];\n    /** Total captured sample count across {@link pcmFrames} (kept in step to avoid re-summing). */\n    private totalSamples = 0;\n    /** Streaming waveform-peak accumulator — bounded regardless of recording length. */\n    private peaks = new PeakAccumulator(RealtimeAudioRecorder.PeakBuckets);\n    /** Final normalized peaks, snapshotted at {@link Stop} so {@link GetPeaks} survives cleanup. */\n    private finalPeaks: number[] = [];\n    private recording = false;\n\n    /** True once the agent's remote stream has been wired into the mix (prevents double-attach). */\n    private remoteAttached = false;\n    /** A remote stream handed to AttachRemoteStream before the audio graph was ready; wired at setup. */\n    private pendingRemoteStream: MediaStream | null = null;\n\n    /** Sample index up to which crash-recovery shards have already been emitted (the segment cursor). */\n    private flushedSampleCount = 0;\n\n    /** Wall-clock ms at recording start; anchors {@link NowOffsetMs}. 0 before {@link Start}. */\n    private startedAtMs = 0;\n\n    /** Sample rate of the capturing {@link AudioContext}; 0 before {@link Start}. Needed to wrap shards. */\n    private sampleRate = 0;\n\n    /** True while a recorder is actively capturing. */\n    public get IsRecording(): boolean {\n        return this.recording;\n    }\n\n    /** Wall-clock ms when recording started (0 before {@link Start}). */\n    public get StartedAtMs(): number {\n        return this.startedAtMs;\n    }\n\n    /**\n     * The container/codec the active recording uses. Always `'audio/wav'` once recording (empty\n     * string before {@link Start} / when disabled) — the consolidated upload and the WAV-wrapped\n     * crash-recovery shards are all 16-bit PCM WAV.\n     */\n    public get MimeType(): string {\n        return this.recording ? RealtimeAudioRecorder.WavMimeType : '';\n    }\n\n    /**\n     * The captured AudioContext sample rate (0 before {@link Start} / when disabled). Exposed so a\n     * recovery path can WAV-wrap the raw-PCM shards with the correct rate.\n     */\n    public get SampleRate(): number {\n        return this.sampleRate;\n    }\n\n    /**\n     * Begins recording a mix of `micStream` and (when present) `remoteStream`. Best-effort:\n     * a missing `AudioContext` / no worklet+scriptprocessor support / any setup failure disables\n     * the recorder silently (logged) so the live call is never blocked.\n     *\n     * @param micStream The user's microphone capture stream (required).\n     * @param remoteStream The agent's remote-audio stream, or `null` for mic-only capture.\n     */\n    public Start(micStream: MediaStream, remoteStream: MediaStream | null): void {\n        if (this.recording) {\n            return;\n        }\n        if (typeof AudioContext === 'undefined') {\n            console.warn('[RealtimeAudioRecorder] AudioContext unavailable — recording disabled for this session.');\n            return;\n        }\n        // Mark recording + stamp t0 SYNCHRONOUSLY so a caller that checks IsRecording immediately after\n        // Start() (the session service does) sees `true`, even though the audio-graph setup\n        // (AudioContext.resume + worklet module load) completes asynchronously below. Without this the\n        // caller discards the recorder before it finishes starting and nothing is ever uploaded.\n        this.startedAtMs = Date.now();\n        this.finalPeaks = [];\n        this.recording = true;\n        // Fire-and-forget the async setup (it must `await audioContext.resume()` + worklet load); any\n        // failure disables the recorder rather than throwing into the session-start path.\n        void this.startMixedRecording(micStream, remoteStream).catch((error) => {\n            console.warn('[RealtimeAudioRecorder] Failed to start recording — disabling for this session:', error);\n            this.cleanup();\n        });\n    }\n\n    /**\n     * Wires the agent's remote-audio stream into the live mix once it arrives. The WebRTC\n     * `ontrack` that delivers the agent voice usually lands AFTER recording has already begun\n     * (mic-only), so the session attaches it here when it lands. Idempotent, and safe to call\n     * before the audio graph finishes its async setup (the stream is connected once ready).\n     */\n    public AttachRemoteStream(stream: MediaStream): void {\n        if (this.remoteAttached || stream.getAudioTracks().length === 0) {\n            return;\n        }\n        if (this.audioContext && this.captureNode()) {\n            this.connectStream(this.audioContext, stream);\n            this.remoteAttached = true;\n        } else {\n            // Audio-graph setup still in flight (awaiting resume / worklet load) — connect at setup.\n            this.pendingRemoteStream = stream;\n        }\n    }\n\n    /**\n     * Returns a Blob of the raw PCM captured since the last call (a window of header-less,\n     * little-endian PCM16 bytes), advancing the segment cursor. `null` when nothing new. Used to\n     * upload crash-recovery shards mid-call. NOTE: shards are header-less raw PCM (NOT individually\n     * playable WAV) — recovery is concatenation in order, then WAV-wrapping with {@link SampleRate}.\n     * The canonical consolidated file is still authored as a single WAV from full memory at {@link Stop}.\n     */\n    public SnapshotNewSegment(): Blob | null {\n        if (!this.recording || this.flushedSampleCount >= this.totalSamples) {\n            return null;\n        }\n        const fresh = this.collectSamples(this.flushedSampleCount, this.totalSamples);\n        this.flushedSampleCount = this.totalSamples;\n        if (fresh.length === 0) {\n            return null;\n        }\n        const pcm16 = this.floatToPcm16Bytes(fresh);\n        return new Blob([pcm16], { type: 'application/octet-stream' });\n    }\n\n    /** Ms elapsed since recording started (>= 0); 0 when not recording. */\n    public NowOffsetMs(): number {\n        if (this.startedAtMs === 0) {\n            return 0;\n        }\n        return Math.max(0, Date.now() - this.startedAtMs);\n    }\n\n    /**\n     * The downsampled waveform peaks (each 0..1, max-abs per bucket, ~{@link RealtimeAudioRecorder.PeakBuckets}\n     * buckets across the whole recording). Computed incrementally during capture so it's cheap and\n     * bounded regardless of length. Returns `[]` before any audio / for a silent recording. Safe to\n     * call after {@link Stop} — the peaks are snapshotted there before {@link cleanup} clears state.\n     */\n    public GetPeaks(): number[] {\n        return this.recording ? this.peaks.Normalize() : this.finalPeaks;\n    }\n\n    /**\n     * Stops recording and resolves the captured audio as a single mono 16-bit PCM **WAV**\n     * {@link Blob}, then releases the audio graph. Resolves `null` when nothing was recorded\n     * (recorder disabled, or no samples captured). Safe to call when not recording.\n     */\n    public async Stop(): Promise<Blob | null> {\n        if (!this.recording) {\n            this.cleanup();\n            return null;\n        }\n        // Detach the capture node so no further frames arrive while we assemble the file. Snapshot the\n        // peaks BEFORE cleanup so GetPeaks() still returns them for the post-Stop upload.\n        this.recording = false;\n        this.finalPeaks = this.peaks.Normalize();\n        const blob = this.encodeWavBlob();\n        this.cleanup();\n        return blob;\n    }\n\n    /** Wires up the Web Audio mix graph + PCM capture node (worklet preferred) and starts capture. */\n    private async startMixedRecording(micStream: MediaStream, remoteStream: MediaStream | null): Promise<void> {\n        const audioContext = new AudioContext();\n        // CRITICAL: browsers create an AudioContext in the `suspended` state under the autoplay policy.\n        // Without resuming it, the mix graph never processes audio and we capture only silence. Starting\n        // a voice call is a user gesture, so resume() succeeds here.\n        if (audioContext.state === 'suspended') {\n            await audioContext.resume();\n        }\n        const destination = audioContext.createMediaStreamDestination();\n        // Publish the context + sample rate BEFORE wiring the remote so a concurrent AttachRemoteStream()\n        // (the agent track can land during the awaits above) sees a ready graph to connect to.\n        this.audioContext = audioContext;\n        this.destination = destination;\n        this.sampleRate = audioContext.sampleRate;\n\n        // Build the PCM capture node: AudioWorklet preferred, ScriptProcessor fallback. Either way the\n        // node both pulls audio from the mix AND drives the destination so the graph keeps processing.\n        await this.createCaptureNode(audioContext, destination);\n\n        this.connectStream(audioContext, micStream);\n        // Connect the agent's stream if it was passed now, or if AttachRemoteStream() stashed one while\n        // we were setting up. Either way, capture the agent voice — not just the mic.\n        const remote = remoteStream ?? this.pendingRemoteStream;\n        this.pendingRemoteStream = null;\n        if (remote && remote.getAudioTracks().length > 0) {\n            this.connectStream(audioContext, remote);\n            this.remoteAttached = true;\n        }\n    }\n\n    /**\n     * Creates the PCM capture node and inserts it into the graph: sources → captureNode → destination.\n     * Prefers an {@link AudioWorkletNode} (off-main-thread, no deprecation); falls back to a\n     * {@link ScriptProcessorNode} where worklets are unavailable. Throws when neither is supported\n     * (caught by {@link Start}, which then disables the recorder).\n     */\n    private async createCaptureNode(audioContext: AudioContext, destination: MediaStreamAudioDestinationNode): Promise<void> {\n        if (await this.tryCreateWorkletNode(audioContext, destination)) {\n            return;\n        }\n        this.createScriptProcessorNode(audioContext, destination);\n    }\n\n    /** Attempts to build the AudioWorklet capture node; returns `false` (no throw) when unsupported. */\n    private async tryCreateWorkletNode(audioContext: AudioContext, destination: MediaStreamAudioDestinationNode): Promise<boolean> {\n        if (!audioContext.audioWorklet || typeof AudioWorkletNode === 'undefined' || typeof Blob === 'undefined') {\n            return false;\n        }\n        try {\n            const moduleUrl = URL.createObjectURL(\n                new Blob([RealtimeAudioRecorder.WorkletProcessorSource], { type: 'application/javascript' }),\n            );\n            try {\n                await audioContext.audioWorklet.addModule(moduleUrl);\n            } finally {\n                URL.revokeObjectURL(moduleUrl);\n            }\n            const node = new AudioWorkletNode(audioContext, 'mj-pcm-capture', { numberOfInputs: 1, numberOfOutputs: 1 });\n            node.port.onmessage = (event: MessageEvent<Float32Array>) => this.captureFrame(event.data);\n            node.connect(destination);\n            this.workletNode = node;\n            return true;\n        } catch (error) {\n            console.warn('[RealtimeAudioRecorder] AudioWorklet unavailable — falling back to ScriptProcessor:', error);\n            return false;\n        }\n    }\n\n    /** Builds the ScriptProcessor fallback capture node and inserts it into the graph. */\n    private createScriptProcessorNode(audioContext: AudioContext, destination: MediaStreamAudioDestinationNode): void {\n        if (typeof audioContext.createScriptProcessor !== 'function') {\n            throw new Error('Neither AudioWorklet nor ScriptProcessor is supported');\n        }\n        const node = audioContext.createScriptProcessor(RealtimeAudioRecorder.ScriptProcessorBufferSize, 1, 1);\n        node.onaudioprocess = (event: AudioProcessingEvent) => {\n            // Copy out of the reused input buffer before retaining.\n            this.captureFrame(event.inputBuffer.getChannelData(0).slice(0));\n        };\n        // ScriptProcessor only fires onaudioprocess while connected to the destination.\n        node.connect(destination);\n        this.scriptNode = node;\n    }\n\n    /** The active capture node (worklet or script processor), or `null` before setup. */\n    private captureNode(): AudioNode | null {\n        return this.workletNode ?? this.scriptNode;\n    }\n\n    /** Absorbs one captured PCM frame: accumulate samples + update the bounded peak array. */\n    private captureFrame(frame: Float32Array): void {\n        if (!this.recording || frame.length === 0) {\n            return;\n        }\n        // Defensive copy — worklet/script buffers are reused by the engine.\n        const copy = frame.slice(0);\n        this.pcmFrames.push(copy);\n        this.totalSamples += copy.length;\n        this.peaks.Push(copy);\n    }\n\n    /** Routes one stream's audio tracks into the capture node (no-op when track-less / no node). */\n    private connectStream(audioContext: AudioContext, stream: MediaStream): void {\n        const node = this.captureNode();\n        if (!node || stream.getAudioTracks().length === 0) {\n            return;\n        }\n        const source = audioContext.createMediaStreamSource(stream);\n        source.connect(node);\n    }\n\n    /** Concatenates a half-open sample range [from, to) across the captured frames into one Float32Array. */\n    private collectSamples(from: number, to: number): Float32Array {\n        const out = new Float32Array(Math.max(0, to - from));\n        if (out.length === 0) {\n            return out;\n        }\n        let cursor = 0; // absolute sample index at the start of the current frame\n        let written = 0;\n        for (const frame of this.pcmFrames) {\n            const frameStart = cursor;\n            const frameEnd = cursor + frame.length;\n            cursor = frameEnd;\n            if (frameEnd <= from) {\n                continue; // entirely before the window\n            }\n            if (frameStart >= to) {\n                break; // entirely after the window\n            }\n            const sliceStart = Math.max(0, from - frameStart);\n            const sliceEnd = Math.min(frame.length, to - frameStart);\n            for (let i = sliceStart; i < sliceEnd; i++) {\n                out[written++] = frame[i];\n            }\n        }\n        return written === out.length ? out : out.subarray(0, written);\n    }\n\n    /**\n     * Converts a mono Float32 buffer to a little-endian PCM16 `ArrayBuffer` (for raw crash-recovery\n     * shards). Returns the backing `ArrayBuffer` directly so it's a valid `BlobPart`.\n     */\n    private floatToPcm16Bytes(samples: Float32Array): ArrayBuffer {\n        const buffer = new ArrayBuffer(samples.length * 2);\n        const view = new DataView(buffer);\n        for (let i = 0; i < samples.length; i++) {\n            const s = samples[i] < -1 ? -1 : samples[i] > 1 ? 1 : samples[i];\n            view.setInt16(i * 2, s < 0 ? s * 0x8000 : s * 0x7fff, true);\n        }\n        return buffer;\n    }\n\n    /** Encodes all accumulated PCM into a single seekable WAV blob, or `null` when nothing captured. */\n    private encodeWavBlob(): Blob | null {\n        if (this.totalSamples === 0 || this.sampleRate === 0) {\n            return null;\n        }\n        const all = this.collectSamples(0, this.totalSamples);\n        const wav = encodePcm16Wav(all, this.sampleRate);\n        return new Blob([wav], { type: RealtimeAudioRecorder.WavMimeType });\n    }\n\n    /** Releases the audio graph + capture node and resets state. Idempotent. */\n    private cleanup(): void {\n        this.recording = false;\n        this.pcmFrames = [];\n        this.totalSamples = 0;\n        this.flushedSampleCount = 0;\n        this.sampleRate = 0;\n        this.peaks = new PeakAccumulator(RealtimeAudioRecorder.PeakBuckets);\n        this.remoteAttached = false;\n        this.pendingRemoteStream = null;\n        if (this.workletNode) {\n            this.workletNode.port.onmessage = null;\n            try { this.workletNode.disconnect(); } catch { /* already disconnected */ }\n            this.workletNode = null;\n        }\n        if (this.scriptNode) {\n            this.scriptNode.onaudioprocess = null;\n            try { this.scriptNode.disconnect(); } catch { /* already disconnected */ }\n            this.scriptNode = null;\n        }\n        this.destination = null;\n        if (this.audioContext) {\n            void this.audioContext.close().catch(() => { /* already closed */ });\n            this.audioContext = null;\n        }\n    }\n}\n"]}

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

@@ -0,0 +1,75 @@
+/**
+ * @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

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