The AI Ear: enterprise-grade multi-modal AI audio listening system#1
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… system Co-authored-by: MASSIVEMAGNETICS <209589629+MASSIVEMAGNETICS@users.noreply.github.com>
Co-authored-by: MASSIVEMAGNETICS <209589629+MASSIVEMAGNETICS@users.noreply.github.com>
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LETS ADD FULL TECHNICAL DOCUMENTATION WITH USER GUIDES AND SMONE TESTS |
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Pull request overview
Introduces the initial end-to-end “AI Ear” implementation: an asyncio-based audio ingestion + concurrent analysis pipeline with typed Pydantic models, a temporal memory layer, and a FastAPI REST/WebSocket API, plus docs/examples/tests to exercise the stack.
Changes:
- Added core runtime modules: listener (mic/file), pipeline (fan-out + fusion + events), memory (rolling context), and shared Pydantic models/settings.
- Added analyzers for speech (Whisper), emotion (Transformers), environment (heuristics), and music (librosa/heuristics).
- Added FastAPI server endpoints, examples, and a full pytest suite; updated packaging/dependencies and README.
Reviewed changes
Copilot reviewed 29 out of 31 changed files in this pull request and generated 18 comments.
Show a summary per file
| File | Description |
|---|---|
| tests/test_pipeline.py | Tests pipeline lifecycle, fusion, callbacks, event detection, and tag derivation. |
| tests/test_models.py | Validates Pydantic model defaults, bounds, and serialization. |
| tests/test_memory.py | Tests result/event storage, eviction, and context summary synthesis. |
| tests/test_audio_utils.py | Tests numpy-only DSP utilities and signal generators. |
| tests/test_api.py | Tests REST endpoints and file upload analysis behavior. |
| tests/test_analyzers.py | Tests environment/music analyzers with synthetic audio. |
| tests/init.py | Declares tests as a package. |
| requirements.txt | Adds runtime + dev dependency list. |
| pyproject.toml | Defines project metadata, dependencies, and tooling (ruff/pytest/mypy). |
| examples/enterprise_integration.py | Demonstrates BYOM analyzer injection, alerting, LLM prompt building, and programmatic serving. |
| examples/basic_listening.py | Provides minimal demo for synthetic chunks and file ingestion. |
| ai_ear/utils/audio.py | Adds numpy-only audio feature extraction + test signal generators. |
| ai_ear/utils/init.py | Re-exports audio utilities. |
| ai_ear/core/pipeline.py | Implements concurrent analysis, fusion into AnalysisResult, event derivation, and stats. |
| ai_ear/core/models.py | Adds shared Pydantic models for chunks, results, and events. |
| ai_ear/core/memory.py | Implements rolling storage and LLM-ready context summaries. |
| ai_ear/core/listener.py | Implements mic capture via sounddevice and file ingestion via soundfile with optional resampling. |
| ai_ear/core/config.py | Adds pydantic-settings configuration with AIEAR_* env overrides. |
| ai_ear/core/init.py | Re-exports core components. |
| ai_ear/api/server.py | Implements REST endpoints + WebSocket streaming and pipeline lifecycle integration. |
| ai_ear/api/init.py | Exposes app factory. |
| ai_ear/analyzers/speech.py | Implements Whisper-based transcription in a thread executor. |
| ai_ear/analyzers/music.py | Implements librosa/heuristic music profiling (tempo/key/energy/genre hints). |
| ai_ear/analyzers/environment.py | Implements heuristic environment classification. |
| ai_ear/analyzers/emotion.py | Implements Transformers-based SER with fallback behavior. |
| ai_ear/analyzers/base.py | Defines analyzer lifecycle contract and typed partial result models. |
| ai_ear/analyzers/init.py | Re-exports analyzers and result types. |
| ai_ear/main.py | Adds CLI entry point for serving the API. |
| ai_ear/init.py | Adds package exports and version. |
| README.md | Replaces placeholder README with full architecture/usage documentation. |
| .gitignore | Adds Python, env, cache, and tool ignore patterns. |
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| async for chunk in chunks: | ||
| if not self._running: | ||
| break | ||
| asyncio.create_task(self._process_and_dispatch(chunk)) |
There was a problem hiding this comment.
process_stream() schedules a new task per incoming chunk without any bounding or awaiting, which can accumulate an unbounded number of pending tasks if input outpaces processing (even with the semaphore). Consider awaiting process(), using a bounded internal queue/worker pool, or tracking and applying backpressure/cancellation to avoid unbounded task growth.
| asyncio.create_task(self._process_and_dispatch(chunk)) | |
| await self._process_and_dispatch(chunk) |
| if chunk is None: | ||
| raise HTTPException(status_code=422, detail="Could not decode audio file") | ||
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| from ai_ear.core.models import AudioChunk |
There was a problem hiding this comment.
This local import of AudioChunk is unused and will fail Ruff F401. It can be removed since chunk is already produced by _load_audio_bytes().
| from ai_ear.core.models import AudioChunk |
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| if sr != target_sr: | ||
| try: | ||
| import librosa # type: ignore[import-untyped] | ||
| mono = librosa.resample(mono, orig_sr=sr, target_sr=target_sr) | ||
| except ImportError: | ||
| pass # best-effort | ||
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| from ai_ear.core.models import AudioChunk | ||
| return AudioChunk(samples=mono, sample_rate=target_sr, source_id=source_id) |
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If the uploaded audio sample rate (sr) differs from target_sr and librosa is unavailable, the code falls back to "pass" (no resample) but still returns an AudioChunk labeled with sample_rate=target_sr. This mislabels the audio and can break downstream analysers expecting the declared sample rate. If resampling can't be performed, either return sample_rate=sr or fail with a clear error (e.g., require librosa for resampling).
| if sr != target_sr: | |
| try: | |
| import librosa # type: ignore[import-untyped] | |
| mono = librosa.resample(mono, orig_sr=sr, target_sr=target_sr) | |
| except ImportError: | |
| pass # best-effort | |
| from ai_ear.core.models import AudioChunk | |
| return AudioChunk(samples=mono, sample_rate=target_sr, source_id=source_id) | |
| effective_sr = sr | |
| if sr != target_sr: | |
| try: | |
| import librosa # type: ignore[import-untyped] | |
| mono = librosa.resample(mono, orig_sr=sr, target_sr=target_sr) | |
| effective_sr = target_sr | |
| except ImportError: | |
| # Best-effort: keep original sample rate if resampling is unavailable | |
| log.warning( | |
| "librosa is not installed; returning audio at original sample rate %d " | |
| "instead of requested %d", | |
| sr, | |
| target_sr, | |
| ) | |
| from ai_ear.core.models import AudioChunk | |
| return AudioChunk(samples=mono, sample_rate=effective_sr, source_id=source_id) |
| self._model = None | ||
| self._executor.shutdown(wait=False) |
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unload() shuts down the ThreadPoolExecutor. If the pipeline/analyzer instance is later started again, subsequent load()/analyse() calls will fail because run_in_executor cannot schedule work on a shutdown executor. Either recreate the executor on load(), or avoid shutting it down unless the analyzer instance will never be reused.
| self._model = None | |
| self._executor.shutdown(wait=False) | |
| # Release the loaded model but keep the executor alive so this | |
| # analyzer instance can be safely reused with future load()/analyse() calls. | |
| self._model = None |
| import asyncio | ||
| import io | ||
| import struct | ||
| import time | ||
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| import numpy as np | ||
| import pytest | ||
| from fastapi.testclient import TestClient | ||
| from httpx import AsyncClient, ASGITransport |
There was a problem hiding this comment.
Several imports in this test module are unused (asyncio, struct, AsyncClient, ASGITransport). With Ruff lint selecting F rules, these will fail F401. Remove unused imports or add coverage that actually uses the async httpx client if intended.
| import asyncio | |
| import io | |
| import struct | |
| import time | |
| import numpy as np | |
| import pytest | |
| from fastapi.testclient import TestClient | |
| from httpx import AsyncClient, ASGITransport | |
| import io | |
| import time | |
| import numpy as np | |
| import pytest | |
| from fastapi.testclient import TestClient |
| import asyncio | ||
|
|
There was a problem hiding this comment.
asyncio is imported but unused in this test module. With Ruff F rules enabled, this will fail lint (F401). Remove the unused import.
| import asyncio |
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| import asyncio | ||
| import time | ||
| import uuid |
There was a problem hiding this comment.
uuid is imported but not used in this module, which will fail Ruff F401. Please remove the unused import (or use it if you intended to attach unique IDs to chunks).
| import uuid |
| * **Back-pressure aware** – if the result queue fills up, older results are | ||
| silently dropped rather than blocking the audio thread. |
There was a problem hiding this comment.
The module docstring claims the pipeline is "back-pressure aware" and drops results when a "result queue" fills up, but there is no result queue or drop logic implemented in this file. Either implement the described bounded queue/drop policy or adjust the docstring to match the current semaphore-based concurrency limiting behavior.
| * **Back-pressure aware** – if the result queue fills up, older results are | |
| silently dropped rather than blocking the audio thread. | |
| * **Concurrency-limited** – an internal semaphore caps the number of in-flight | |
| analyses, naturally applying back-pressure without unbounded queuing. |
| import asyncio | ||
| import hashlib | ||
| import logging | ||
| import time | ||
| import uuid | ||
| from collections.abc import Callable, Coroutine | ||
| from typing import Any | ||
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| from ai_ear.core.models import ( | ||
| AnalysisResult, | ||
| AudioChunk, | ||
| AuralEvent, | ||
| AuralEventType, | ||
| EnvironmentLabel, | ||
| MusicProfile, | ||
| SpeechSegment, | ||
| ) |
There was a problem hiding this comment.
There are several unused imports here (e.g., uuid, MusicProfile, SpeechSegment), which will fail Ruff's F401 checks per pyproject.toml. Please remove unused imports or use them.
| if self._loop is not None: | ||
| self._loop.call_soon_threadsafe(self._queue.put_nowait, chunk) | ||
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There was a problem hiding this comment.
call_soon_threadsafe(self._queue.put_nowait, chunk) can raise asyncio.QueueFull in the event loop thread if the queue is at maxsize, which will surface as an unhandled exception. If the intent is backpressure/drop behavior, wrap put_nowait in a small helper that catches QueueFull and drops/logs the chunk.
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@copilot apply changes based on the comments in this thread |
Co-authored-by: MASSIVEMAGNETICS <209589629+MASSIVEMAGNETICS@users.noreply.github.com>
Applied all changes from the review thread in commit
Added in commit Full technical documentation (
22 new smoke/integration tests (
Total: 119 tests passing |
|
@copilot apply changes based on the comments in this thread |
All changes from that review thread were already applied in commit |
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| async def _dispatch(result: AnalysisResult) -> None: | ||
| with suppress(Exception): | ||
| await websocket.send_json(result.model_dump()) |
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Scope WebSocket dispatch to the originating source
The stream() handler subscribes _dispatch to the pipeline’s global result bus, and _dispatch forwards every AnalysisResult to this socket without checking result.source_id. In multi-client runs (or while /analyse requests are processed), one connection will receive other sources’ results, which mixes streams and can leak another client’s audio-derived data; filter on the connection’s source_id before sending.
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All changes from that review thread were already applied in commit |
AI that truly hears — not just transcribes. Every audio window is concurrently analysed for speech, emotion, acoustic environment, and music, fused into a typed result with a temporal memory layer that produces LLM-ready context summaries.
Core architecture
AudioListener— asyncio-native mic capture (sounddevice) + file ingestion; C audio thread bridged to event loop viacall_soon_threadsafewithQueueFulldrop-and-log protectionAudioPipeline— fans outAudioChunkto all analysers viaasyncio.gather; dispatchesAnalysisResult+AuralEventto registered callbacks; detects state transitions (speech started/ended, music onset, env change, alarm); event state tracked per source_id to eliminate cross-talk between concurrent sources;on_result()/on_event()return unsubscribe callables for leak-free teardown;add_analyzer()schedulesload()automatically when the pipeline is already runningAuralMemory— bounded rolling store withcontext_summary()returning a structured dict ready for LLM system-prompt injection, plus transcript, event log, and acoustic fingerprint registryAudioChunk,AnalysisResult,SpeechSegment,EmotionProfile,EnvironmentSnapshot,MusicProfile,AuralEventSettings— fullpydantic-settingsconfig, all values overridable viaAIEAR_*env varsAnalysers (all pluggable via
BaseAnalyzer)SpeechAnalyzerEmotionAnalyzerEnvironmentAnalyzerMusicAnalyzerAll analysers use nullable
ThreadPoolExecutorinstances that are safely recreated onload()afterunload(), enabling full pipeline restart.Enterprise API
FastAPI REST + WebSocket server:
POST /analyse,GET /memory/context,GET /memory/transcript,GET /memory/events,GET /pipeline/stats,WS /stream(raw PCM float32 frames → streaming JSON results). WebSocket per-connection callbacks are automatically unregistered on disconnect.BYOM example
LLM context injection
Documentation
Full technical documentation in
docs/:architecture.mduser_guide.mdapi_reference.mdconfiguration.mdAIEAR_*env var with types, defaults, and examplesextending.mddeployment.mdTests
119 tests passing, including a dedicated
tests/test_smoke.pywith end-to-end smoke tests covering the full pipeline, memory, event generation, REST API, and file ingestion.Original prompt
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