mirror of
https://github.com/NousResearch/hermes-agent.git
synced 2026-07-30 19:09:28 +00:00
39975613b1
Second, deeper pass over tools/gateway/hermes_cli plus first pass over the trees wave 1 missed (acp, acp_adapter, skills, computer_use, docker, dashboard, conformance, monitoring, secret_sources, hermes_state, providers). Same rubric as wave 1 (AGENTS.md test policy); security, alternation/caching invariants, issue-number regressions, and E2E kept. Real test-quality fixes found and rooted out along the way: - tests/tools/test_command_guards.py made real auxiliary-LLM HTTPS calls (DEFAULT_CONFIG smart-approval leaked in) — pinned approval mode=manual via autouse fixture: 17.4s → 0.4s. - test_model_switch_custom_providers.py / test_user_providers_model_switch.py silently probed live provider catalogs (~2s/test) — stubbed cached_provider_model_ids/provider_model_ids/fetch_api_models. - test_telegram_noise_filter.py: 15-platform copy-paste matrix over shared gateway.run logic → 3 representative platforms (55s → 3.9s). - test_gateway_shutdown.py: stop()'s 5s interrupt-deadline loop spun on MagicMock agents — interrupt.side_effect now clears _running_agents (22s → 1.0s). - test_gateway_inactivity_timeout.py poll-harness timings shrunk 3-5x (24s → 1.1s); test_mcp_stability.py backoff/SIGTERM-grace sleeps patched (15.4s → 2.5s); test_async_delegation.py negative-drain wait 5s → 0.5s. - test_telegram_init_deadline.py: loop-block margin restored to 1.0s with rationale comment — the watchdog-dump assertion needs the loop blocked well past deadline+grace under parallel load (flaked once in the 40-worker verification run at a 0.2s margin). Verification: full hermetic suite via scripts/run_tests.sh — 2,438 files, 21,718 tests passed, 0 failed, 293.9s wall. Suite totals vs original baseline: 46,820 → 19,757 test functions (−57.8%), wall 583.5s → 293.9s (−50%), subprocess CPU 13,564s → 11,623s.
1596 lines
63 KiB
Python
1596 lines
63 KiB
Python
"""Tests for tools.voice_mode -- all mocked, no real microphone or API calls."""
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import os
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import struct
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import time
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import wave
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from pathlib import Path
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from unittest.mock import MagicMock, patch
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import pytest
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def _non_wsl_proc_version(real_open):
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"""Return an open() shim that makes host WSL detection deterministic."""
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def _fake_open(file, *args, **kwargs):
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if file == "/proc/version":
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from io import StringIO
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return StringIO("Linux test-kernel")
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return real_open(file, *args, **kwargs)
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return _fake_open
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# ============================================================================
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# Fixtures
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# ============================================================================
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@pytest.fixture
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def sample_wav(tmp_path):
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"""Create a minimal valid WAV file (1 second of silence at 16kHz)."""
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wav_path = tmp_path / "test.wav"
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n_frames = 16000 # 1 second at 16kHz
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silence = struct.pack(f"<{n_frames}h", *([0] * n_frames))
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with wave.open(str(wav_path), "wb") as wf:
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wf.setnchannels(1)
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wf.setsampwidth(2)
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wf.setframerate(16000)
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wf.writeframes(silence)
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return str(wav_path)
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@pytest.fixture
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def temp_voice_dir(tmp_path, monkeypatch):
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"""Redirect _TEMP_DIR to a temporary path."""
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voice_dir = tmp_path / "hermes_voice"
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voice_dir.mkdir()
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monkeypatch.setattr("tools.voice_mode._TEMP_DIR", str(voice_dir))
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return voice_dir
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@pytest.fixture
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def mock_sd(monkeypatch):
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"""Mock _import_audio to return (mock_sd, real_np) so lazy imports work."""
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mock = MagicMock()
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try:
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import numpy as real_np
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except ImportError:
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real_np = MagicMock()
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def _fake_import_audio():
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return mock, real_np
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monkeypatch.setattr("tools.voice_mode._import_audio", _fake_import_audio)
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monkeypatch.setattr("tools.voice_mode._audio_available", lambda: True)
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return mock
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class _FakeTime:
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"""Stand-in for the ``time`` module with a monotonic clock the test drives.
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Silence detection compares ``time.monotonic()`` deltas against thresholds
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of a few dozen milliseconds. Driving those deltas with real ``sleep()``
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calls only works when the platform clock is finer-grained than the margin
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the test leaves: ``time.monotonic()`` is ``GetTickCount64()`` (15.625 ms
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resolution) on Windows until CPython 3.13 moved it to
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``QueryPerformanceCounter()``, so a 60 ms sleep can legitimately measure
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as 46 ms and land under a 50 ms threshold. Advancing an explicit clock
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keeps the arithmetic exact on every platform.
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Everything other than ``monotonic`` delegates to the real module, so
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``time.sleep``/``time.strftime`` in the code under test keep working.
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"""
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def __init__(self, real_time, start: float = 1000.0) -> None:
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self._real = real_time
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self._now = start
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def monotonic(self) -> float:
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return self._now
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def advance(self, seconds: float) -> None:
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self._now += seconds
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def __getattr__(self, name):
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return getattr(self._real, name)
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@pytest.fixture
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def fake_clock(monkeypatch):
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"""Give voice_mode a hand-driven clock.
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Patches the name ``time`` inside ``tools.voice_mode`` rather than
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``time.monotonic`` itself -- ``voice_mode.time`` *is* the stdlib module,
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so setting the attribute on it would swap the clock out from under every
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other importer for the duration of the test.
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"""
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import time as real_time
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import tools.voice_mode as voice_mode
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clock = _FakeTime(real_time)
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monkeypatch.setattr(voice_mode, "time", clock)
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return clock
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# ============================================================================
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# detect_audio_environment — WSL / SSH / Docker detection
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# ============================================================================
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class TestPulseSocketReachable:
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def test_stale_socket_file_not_reachable(self, monkeypatch, tmp_path):
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"""A socket file with no listener should not count as reachable."""
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import socket as _socket
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sock_path = tmp_path / "pulse" / "native"
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sock_path.parent.mkdir(parents=True)
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# Create + bind, then close so the path is a stale socket file.
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s = _socket.socket(_socket.AF_UNIX, _socket.SOCK_STREAM)
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s.bind(str(sock_path))
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s.close()
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monkeypatch.delenv("PULSE_SERVER", raising=False)
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monkeypatch.delenv("PULSE_RUNTIME_PATH", raising=False)
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monkeypatch.setenv("XDG_RUNTIME_DIR", str(tmp_path))
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from tools.voice_mode import _pulse_socket_reachable
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assert _pulse_socket_reachable() is False
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def test_listening_socket_reachable_via_xdg_runtime(self, monkeypatch, tmp_path):
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"""A live PulseAudio-style socket under XDG_RUNTIME_DIR is reachable (#35622)."""
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import socket as _socket
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sock_path = tmp_path / "pulse" / "native"
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sock_path.parent.mkdir(parents=True)
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server = _socket.socket(_socket.AF_UNIX, _socket.SOCK_STREAM)
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server.bind(str(sock_path))
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server.listen(1)
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try:
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monkeypatch.delenv("PULSE_SERVER", raising=False)
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monkeypatch.delenv("PULSE_RUNTIME_PATH", raising=False)
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monkeypatch.setenv("XDG_RUNTIME_DIR", str(tmp_path))
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from tools.voice_mode import _pulse_socket_reachable
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assert _pulse_socket_reachable() is True
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finally:
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server.close()
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class TestDetectAudioEnvironment:
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def test_clean_environment_is_available(self, monkeypatch):
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"""No SSH, Docker, or WSL — should be available."""
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monkeypatch.delenv("SSH_CLIENT", raising=False)
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monkeypatch.delenv("SSH_TTY", raising=False)
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monkeypatch.delenv("SSH_CONNECTION", raising=False)
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monkeypatch.setattr("hermes_constants.is_container", lambda: False)
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monkeypatch.setattr("tools.voice_mode._import_audio",
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lambda: (MagicMock(), MagicMock()))
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monkeypatch.setattr("builtins.open", _non_wsl_proc_version(open))
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from tools.voice_mode import detect_audio_environment
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result = detect_audio_environment()
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assert result["available"] is True
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assert result["warnings"] == []
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def test_ssh_blocks_voice(self, monkeypatch):
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"""SSH environment without a reachable sound server should block voice mode."""
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monkeypatch.setenv("SSH_CLIENT", "1.2.3.4 54321 22")
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monkeypatch.delenv("PULSE_SERVER", raising=False)
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monkeypatch.delenv("PIPEWIRE_REMOTE", raising=False)
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monkeypatch.setattr("tools.voice_mode._pulse_socket_reachable", lambda: False)
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monkeypatch.setattr("tools.voice_mode._import_audio",
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lambda: (MagicMock(), MagicMock()))
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from tools.voice_mode import detect_audio_environment
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result = detect_audio_environment()
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assert result["available"] is False
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assert any("SSH" in w for w in result["warnings"])
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def test_ssh_with_pulse_server_allows_voice(self, monkeypatch):
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"""SSH with PULSE_SERVER set should NOT block voice mode (#35622)."""
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monkeypatch.setenv("SSH_CLIENT", "1.2.3.4 54321 22")
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monkeypatch.setenv("PULSE_SERVER", "unix:/run/user/1002/pulse/native")
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monkeypatch.delenv("PIPEWIRE_REMOTE", raising=False)
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monkeypatch.setattr("tools.voice_mode._import_audio",
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lambda: (MagicMock(), MagicMock()))
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monkeypatch.setattr("builtins.open", _non_wsl_proc_version(open))
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from tools.voice_mode import detect_audio_environment
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result = detect_audio_environment()
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assert result["available"] is True
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assert result["warnings"] == []
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assert any("SSH" in n for n in result.get("notices", []))
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def test_wsl_without_pulse_blocks_voice(self, monkeypatch, tmp_path):
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"""WSL without PULSE_SERVER should block voice mode."""
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monkeypatch.delenv("SSH_CLIENT", raising=False)
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monkeypatch.delenv("SSH_TTY", raising=False)
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monkeypatch.delenv("SSH_CONNECTION", raising=False)
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monkeypatch.delenv("PULSE_SERVER", raising=False)
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monkeypatch.setattr("tools.voice_mode._pulse_socket_reachable", lambda: False)
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monkeypatch.setattr("tools.voice_mode._import_audio",
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lambda: (MagicMock(), MagicMock()))
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proc_version = tmp_path / "proc_version"
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proc_version.write_text("Linux 5.15.0-microsoft-standard-WSL2")
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_real_open = open
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def _fake_open(f, *a, **kw):
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if f == "/proc/version":
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return _real_open(str(proc_version), *a, **kw)
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return _real_open(f, *a, **kw)
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with patch("builtins.open", side_effect=_fake_open):
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from tools.voice_mode import detect_audio_environment
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result = detect_audio_environment()
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assert result["available"] is False
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assert any("WSL" in w for w in result["warnings"])
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assert any("PulseAudio" in w for w in result["warnings"])
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def test_docker_with_pipewire_remote_and_no_devices_allows_voice(self, monkeypatch):
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"""PIPEWIRE_REMOTE should bypass empty PortAudio device lists in Docker."""
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monkeypatch.delenv("SSH_CLIENT", raising=False)
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monkeypatch.delenv("SSH_TTY", raising=False)
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monkeypatch.delenv("SSH_CONNECTION", raising=False)
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monkeypatch.delenv("PULSE_SERVER", raising=False)
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monkeypatch.setenv("PIPEWIRE_REMOTE", "/run/user/1000/pipewire-0")
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monkeypatch.setattr("hermes_constants.is_container", lambda: True)
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sd = MagicMock()
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sd.query_devices.return_value = []
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monkeypatch.setattr("tools.voice_mode._import_audio", lambda: (sd, MagicMock()))
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from tools.voice_mode import detect_audio_environment
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result = detect_audio_environment()
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assert result["available"] is True
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assert result["warnings"] == []
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assert any("host audio forwarding" in n.lower() for n in result.get("notices", []))
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def test_docker_without_audio_forwarding_blocks_voice(self, monkeypatch):
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"""Docker without PULSE_SERVER/PIPEWIRE_REMOTE keeps blocking voice mode."""
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monkeypatch.delenv("SSH_CLIENT", raising=False)
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monkeypatch.delenv("SSH_TTY", raising=False)
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monkeypatch.delenv("SSH_CONNECTION", raising=False)
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monkeypatch.delenv("PULSE_SERVER", raising=False)
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monkeypatch.delenv("PIPEWIRE_REMOTE", raising=False)
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monkeypatch.setattr("tools.voice_mode._pulse_socket_reachable", lambda: False)
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monkeypatch.setattr("hermes_constants.is_container", lambda: True)
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monkeypatch.setattr("tools.voice_mode._import_audio",
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lambda: (MagicMock(), MagicMock()))
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from tools.voice_mode import detect_audio_environment
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result = detect_audio_environment()
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assert result["available"] is False
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assert any("container" in w.lower() for w in result["warnings"])
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assert any("PULSE_SERVER" in w or "PIPEWIRE_REMOTE" in w for w in result["warnings"])
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# ============================================================================
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# check_voice_requirements
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# ============================================================================
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class TestCheckVoiceRequirements:
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def test_all_requirements_met(self, monkeypatch):
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monkeypatch.setattr("tools.voice_mode._audio_available", lambda: True)
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monkeypatch.setattr("tools.voice_mode.detect_audio_environment",
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lambda: {"available": True, "warnings": []})
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monkeypatch.setattr("tools.transcription_tools._get_provider", lambda cfg: "openai")
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from tools.voice_mode import check_voice_requirements
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result = check_voice_requirements()
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assert result["available"] is True
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assert result["audio_available"] is True
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assert result["stt_available"] is True
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assert result["missing_packages"] == []
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def test_plugin_stt_provider(self, monkeypatch):
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"""Plugin STT provider is recognized."""
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monkeypatch.setattr("tools.voice_mode._audio_available", lambda: True)
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monkeypatch.setattr("tools.voice_mode.detect_audio_environment",
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lambda: {"available": True, "warnings": []})
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monkeypatch.setattr(
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"tools.transcription_tools._load_stt_config",
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lambda: {"enabled": True, "provider": "my-plugin-stt"},
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)
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plugin_provider = MagicMock()
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plugin_provider.is_available.return_value = True
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monkeypatch.setattr(
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"agent.transcription_registry.get_provider",
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lambda p: plugin_provider if p == "my-plugin-stt" else None,
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)
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monkeypatch.setattr(
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"hermes_cli.plugins._ensure_plugins_discovered",
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lambda force=False: None,
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)
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from tools.voice_mode import check_voice_requirements
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result = check_voice_requirements()
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assert result["available"] is True
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assert result["stt_available"] is True
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assert "STT provider: OK (plugin: my-plugin-stt)" in result["details"]
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# ============================================================================
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# AudioRecorder
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# ============================================================================
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class TestCreateAudioRecorder:
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def test_termux_uses_termux_audio_recorder_when_api_present(self, monkeypatch):
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monkeypatch.setenv("TERMUX_VERSION", "0.118.3")
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monkeypatch.setenv("PREFIX", "/data/data/com.termux/files/usr")
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monkeypatch.setattr("tools.voice_mode._termux_microphone_command", lambda: "/data/data/com.termux/files/usr/bin/termux-microphone-record")
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monkeypatch.setattr("tools.voice_mode._termux_api_app_installed", lambda: True)
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from tools.voice_mode import create_audio_recorder, TermuxAudioRecorder
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recorder = create_audio_recorder()
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assert isinstance(recorder, TermuxAudioRecorder)
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assert recorder.supports_silence_autostop is False
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class TestTermuxAudioRecorder:
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def test_start_and_stop_use_termux_microphone_commands(self, monkeypatch, temp_voice_dir):
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command_calls = []
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output_path = Path(temp_voice_dir) / "recording_20260409_120000.aac"
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def fake_run(cmd, **kwargs):
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command_calls.append(cmd)
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if cmd[1] == "-f":
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Path(cmd[2]).write_bytes(b"aac-bytes")
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return MagicMock(returncode=0, stdout="", stderr="")
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monkeypatch.setenv("TERMUX_VERSION", "0.118.3")
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monkeypatch.setenv("PREFIX", "/data/data/com.termux/files/usr")
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monkeypatch.setattr("tools.voice_mode._termux_microphone_command", lambda: "/data/data/com.termux/files/usr/bin/termux-microphone-record")
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monkeypatch.setattr("tools.voice_mode._termux_api_app_installed", lambda: True)
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monkeypatch.setattr("tools.voice_mode.time.strftime", lambda fmt: "20260409_120000")
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monkeypatch.setattr("tools.voice_mode.subprocess.run", fake_run)
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from tools.voice_mode import TermuxAudioRecorder
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recorder = TermuxAudioRecorder()
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recorder.start()
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recorder._start_time = time.monotonic() - 1.0
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result = recorder.stop()
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assert result == str(output_path)
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assert command_calls[0][:2] == ["/data/data/com.termux/files/usr/bin/termux-microphone-record", "-f"]
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assert command_calls[1] == ["/data/data/com.termux/files/usr/bin/termux-microphone-record", "-q"]
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def test_cancel_removes_partial_termux_recording(self, monkeypatch, temp_voice_dir):
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output_path = Path(temp_voice_dir) / "recording_20260409_120000.aac"
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def fake_run(cmd, **kwargs):
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if cmd[1] == "-f":
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Path(cmd[2]).write_bytes(b"aac-bytes")
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return MagicMock(returncode=0, stdout="", stderr="")
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monkeypatch.setenv("TERMUX_VERSION", "0.118.3")
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monkeypatch.setenv("PREFIX", "/data/data/com.termux/files/usr")
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monkeypatch.setattr("tools.voice_mode._termux_microphone_command", lambda: "/data/data/com.termux/files/usr/bin/termux-microphone-record")
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monkeypatch.setattr("tools.voice_mode._termux_api_app_installed", lambda: True)
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monkeypatch.setattr("tools.voice_mode.time.strftime", lambda fmt: "20260409_120000")
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monkeypatch.setattr("tools.voice_mode.subprocess.run", fake_run)
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from tools.voice_mode import TermuxAudioRecorder
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recorder = TermuxAudioRecorder()
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recorder.start()
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recorder.cancel()
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assert output_path.exists() is False
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assert recorder.is_recording is False
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class TestAudioRecorder:
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def test_start_raises_without_audio_libs(self, monkeypatch):
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def _fail_import():
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raise ImportError("no sounddevice")
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monkeypatch.setattr("tools.voice_mode._import_audio", _fail_import)
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from tools.voice_mode import AudioRecorder
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recorder = AudioRecorder()
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with pytest.raises(RuntimeError, match="sounddevice and numpy"):
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recorder.start()
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def test_start_oserror_points_at_portaudio_not_pip(self, monkeypatch):
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"""OSError from _import_audio means PortAudio's shared library is
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missing — pip can't fix that. The error must point at the system
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package, not 'pip install sounddevice numpy' (#18432)."""
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def _fail_import():
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raise OSError("PortAudio library not found")
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monkeypatch.setattr("tools.voice_mode._import_audio", _fail_import)
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monkeypatch.setattr("tools.voice_mode._is_termux_environment", lambda: False)
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from tools.voice_mode import AudioRecorder
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recorder = AudioRecorder()
|
|
with pytest.raises(RuntimeError) as exc_info:
|
|
recorder.start()
|
|
msg = str(exc_info.value)
|
|
assert "PortAudio system library not found" in msg
|
|
assert "libportaudio2" in msg
|
|
assert "pip install" not in msg
|
|
|
|
def test_start_creates_and_starts_stream(self, mock_sd):
|
|
mock_stream = MagicMock()
|
|
mock_sd.InputStream.return_value = mock_stream
|
|
|
|
from tools.voice_mode import AudioRecorder
|
|
|
|
recorder = AudioRecorder()
|
|
recorder.start()
|
|
|
|
assert recorder.is_recording is True
|
|
mock_sd.InputStream.assert_called_once()
|
|
mock_stream.start.assert_called_once()
|
|
|
|
class TestAudioRecorderStop:
|
|
def test_stop_writes_wav_file(self, mock_sd, temp_voice_dir):
|
|
np = pytest.importorskip("numpy")
|
|
|
|
mock_stream = MagicMock()
|
|
mock_sd.InputStream.return_value = mock_stream
|
|
|
|
from tools.voice_mode import AudioRecorder, SAMPLE_RATE
|
|
|
|
recorder = AudioRecorder()
|
|
recorder.start()
|
|
|
|
# Simulate captured audio frames (1 second of loud audio above RMS threshold)
|
|
frame = np.full((SAMPLE_RATE, 1), 1000, dtype="int16")
|
|
recorder._frames = [frame]
|
|
recorder._peak_rms = 1000 # Peak RMS above threshold
|
|
|
|
wav_path = recorder.stop()
|
|
|
|
assert wav_path is not None
|
|
assert os.path.isfile(wav_path)
|
|
assert wav_path.endswith(".wav")
|
|
assert recorder.is_recording is False
|
|
|
|
# Verify it is a valid WAV
|
|
with wave.open(wav_path, "rb") as wf:
|
|
assert wf.getnchannels() == 1
|
|
assert wf.getsampwidth() == 2
|
|
assert wf.getframerate() == SAMPLE_RATE
|
|
|
|
def test_stop_returns_none_for_silent_recording(self, mock_sd, temp_voice_dir):
|
|
np = pytest.importorskip("numpy")
|
|
|
|
mock_stream = MagicMock()
|
|
mock_sd.InputStream.return_value = mock_stream
|
|
|
|
from tools.voice_mode import AudioRecorder, SAMPLE_RATE
|
|
|
|
recorder = AudioRecorder()
|
|
recorder.start()
|
|
|
|
# 1 second of near-silence (RMS well below threshold)
|
|
frame = np.full((SAMPLE_RATE, 1), 10, dtype="int16")
|
|
recorder._frames = [frame]
|
|
recorder._peak_rms = 10 # Peak RMS also below threshold
|
|
|
|
wav_path = recorder.stop()
|
|
assert wav_path is None
|
|
|
|
|
|
class TestAudioRecorderCancel:
|
|
def test_cancel_discards_frames(self, mock_sd):
|
|
mock_stream = MagicMock()
|
|
mock_sd.InputStream.return_value = mock_stream
|
|
|
|
from tools.voice_mode import AudioRecorder
|
|
|
|
recorder = AudioRecorder()
|
|
recorder.start()
|
|
recorder._frames = [MagicMock()] # simulate captured data
|
|
|
|
recorder.cancel()
|
|
|
|
assert recorder.is_recording is False
|
|
assert recorder._frames == []
|
|
# Stream is kept alive (persistent) — cancel() does NOT close it.
|
|
mock_stream.stop.assert_not_called()
|
|
mock_stream.close.assert_not_called()
|
|
|
|
# ============================================================================
|
|
# transcribe_recording
|
|
# ============================================================================
|
|
|
|
class TestTranscribeRecording:
|
|
def test_filters_whisper_hallucination(self):
|
|
mock_transcribe = MagicMock(return_value={
|
|
"success": True,
|
|
"transcript": "Thank you.",
|
|
})
|
|
|
|
with patch("tools.transcription_tools.transcribe_audio", mock_transcribe):
|
|
from tools.voice_mode import transcribe_recording
|
|
result = transcribe_recording("/tmp/test.wav")
|
|
|
|
assert result["success"] is True
|
|
assert result["transcript"] == ""
|
|
assert result["filtered"] is True
|
|
|
|
|
|
def test_other_error_does_not_trigger_chunk(self, tmp_path, monkeypatch):
|
|
"""Non-size errors from transcribe_audio are returned as-is."""
|
|
wav_path = tmp_path / "record.wav"
|
|
n_frames = 50000
|
|
audio = struct.pack(f"<{n_frames}h", *([1000] * n_frames))
|
|
with wave.open(str(wav_path), "wb") as wf:
|
|
wf.setnchannels(1)
|
|
wf.setsampwidth(2)
|
|
wf.setframerate(16000)
|
|
wf.writeframes(audio)
|
|
|
|
mock_transcribe = MagicMock(return_value={
|
|
"success": False,
|
|
"transcript": "",
|
|
"error": "STT is disabled in config.yaml",
|
|
})
|
|
|
|
with patch("tools.transcription_tools.transcribe_audio", mock_transcribe):
|
|
from tools.voice_mode import transcribe_recording
|
|
result = transcribe_recording(str(wav_path), model="base")
|
|
|
|
assert result["success"] is False
|
|
assert "STT is disabled" in result["error"]
|
|
mock_transcribe.assert_called_once()
|
|
|
|
|
|
class TestWhisperHallucinationFilter:
|
|
def test_known_hallucinations(self):
|
|
from tools.voice_mode import is_whisper_hallucination
|
|
|
|
assert is_whisper_hallucination("Thank you.") is True
|
|
assert is_whisper_hallucination("thank you") is True
|
|
assert is_whisper_hallucination("Thanks for watching.") is True
|
|
assert is_whisper_hallucination("Bye.") is True
|
|
assert is_whisper_hallucination(" Thank you. ") is True # with whitespace
|
|
assert is_whisper_hallucination("you") is True
|
|
|
|
def test_real_speech_not_filtered(self):
|
|
from tools.voice_mode import is_whisper_hallucination
|
|
|
|
assert is_whisper_hallucination("Hello, how are you?") is False
|
|
assert is_whisper_hallucination("Thank you for your help with the project.") is False
|
|
assert is_whisper_hallucination("Can you explain this code?") is False
|
|
|
|
|
|
# ============================================================================
|
|
# play_audio_file
|
|
# ============================================================================
|
|
|
|
class TestPlayAudioFile:
|
|
def test_play_wav_via_sounddevice(self, monkeypatch, sample_wav):
|
|
np = pytest.importorskip("numpy")
|
|
# Pin to a non-macOS platform: on macOS WAV output deliberately skips
|
|
# sounddevice (see TestMacOSAudioOutputPolicy), so this path is only
|
|
# exercised off Darwin.
|
|
monkeypatch.setattr("tools.voice_mode.platform.system", lambda: "Linux")
|
|
|
|
mock_sd_obj = MagicMock()
|
|
# Simulate stream completing immediately (get_stream().active = False)
|
|
mock_stream = MagicMock()
|
|
mock_stream.active = False
|
|
mock_sd_obj.get_stream.return_value = mock_stream
|
|
|
|
def _fake_import():
|
|
return mock_sd_obj, np
|
|
|
|
monkeypatch.setattr("tools.voice_mode._import_audio", _fake_import)
|
|
|
|
from tools.voice_mode import play_audio_file
|
|
|
|
result = play_audio_file(sample_wav)
|
|
|
|
assert result is True
|
|
mock_sd_obj.play.assert_called_once()
|
|
mock_sd_obj.stop.assert_called_once()
|
|
|
|
# ============================================================================
|
|
# macOS output policy (no sounddevice for OUTPUT -> avoids TCC prompt)
|
|
# ============================================================================
|
|
|
|
class TestMacOSAudioOutputPolicy:
|
|
def test_play_audio_file_skips_sounddevice_on_macos(self, monkeypatch, sample_wav):
|
|
"""On macOS, WAV playback must not import sounddevice; it routes to afplay."""
|
|
monkeypatch.setattr("tools.voice_mode.platform.system", lambda: "Darwin")
|
|
|
|
def _forbidden_import():
|
|
raise AssertionError("sounddevice must not be imported for output on macOS")
|
|
|
|
monkeypatch.setattr("tools.voice_mode._import_audio", _forbidden_import)
|
|
|
|
popen_cmds = []
|
|
|
|
class _FakeProc:
|
|
returncode = 0
|
|
|
|
def wait(self, timeout=None):
|
|
return 0
|
|
|
|
def kill(self):
|
|
pass
|
|
|
|
def _fake_popen(cmd, **kwargs):
|
|
popen_cmds.append(cmd)
|
|
return _FakeProc()
|
|
|
|
monkeypatch.setattr("shutil.which", lambda exe: f"/usr/bin/{exe}")
|
|
monkeypatch.setattr("subprocess.Popen", _fake_popen)
|
|
|
|
from tools.voice_mode import play_audio_file
|
|
|
|
result = play_audio_file(sample_wav)
|
|
|
|
assert result is True
|
|
assert popen_cmds, "expected a system player to be invoked"
|
|
assert popen_cmds[0][0] == "afplay"
|
|
|
|
def test_play_beep_routes_through_afplay_on_macos(self, monkeypatch):
|
|
"""On macOS, beeps synthesize with numpy but play via the tempfile/afplay path."""
|
|
pytest.importorskip("numpy")
|
|
monkeypatch.setattr("tools.voice_mode.platform.system", lambda: "Darwin")
|
|
|
|
def _forbidden_import():
|
|
raise AssertionError("sounddevice must not be imported for beeps on macOS")
|
|
|
|
monkeypatch.setattr("tools.voice_mode._import_audio", _forbidden_import)
|
|
|
|
calls = []
|
|
monkeypatch.setattr(
|
|
"tools.voice_mode._play_int16_via_tempfile",
|
|
lambda audio, sample_rate: calls.append((len(audio), sample_rate)),
|
|
)
|
|
|
|
import tools.voice_mode as vm
|
|
|
|
vm.play_beep(frequency=880, count=1)
|
|
|
|
assert len(calls) == 1
|
|
n_samples, sample_rate = calls[0]
|
|
assert n_samples > 0
|
|
assert sample_rate == vm.SAMPLE_RATE
|
|
|
|
# ============================================================================
|
|
# cleanup_temp_recordings
|
|
# ============================================================================
|
|
|
|
class TestCleanupTempRecordings:
|
|
def test_old_files_deleted(self, temp_voice_dir):
|
|
# Create an "old" file
|
|
old_file = temp_voice_dir / "recording_20240101_000000.wav"
|
|
old_file.write_bytes(b"\x00" * 100)
|
|
# Set mtime to 2 hours ago
|
|
old_mtime = time.time() - 7200
|
|
os.utime(str(old_file), (old_mtime, old_mtime))
|
|
|
|
from tools.voice_mode import cleanup_temp_recordings
|
|
|
|
deleted = cleanup_temp_recordings(max_age_seconds=3600)
|
|
assert deleted == 1
|
|
assert not old_file.exists()
|
|
|
|
def test_recent_files_preserved(self, temp_voice_dir):
|
|
# Create a "recent" file
|
|
recent_file = temp_voice_dir / "recording_20260303_120000.wav"
|
|
recent_file.write_bytes(b"\x00" * 100)
|
|
|
|
from tools.voice_mode import cleanup_temp_recordings
|
|
|
|
deleted = cleanup_temp_recordings(max_age_seconds=3600)
|
|
assert deleted == 0
|
|
assert recent_file.exists()
|
|
|
|
# ============================================================================
|
|
# play_beep
|
|
# ============================================================================
|
|
|
|
class TestPlayBeep:
|
|
def test_beep_calls_sounddevice_play(self, mock_sd):
|
|
np = pytest.importorskip("numpy")
|
|
|
|
from tools.voice_mode import play_beep
|
|
|
|
# play_beep uses polling (get_stream) + sd.stop() instead of sd.wait()
|
|
mock_stream = MagicMock()
|
|
mock_stream.active = False
|
|
mock_sd.get_stream.return_value = mock_stream
|
|
|
|
play_beep(frequency=880, duration=0.1, count=1)
|
|
|
|
mock_sd.play.assert_called_once()
|
|
mock_sd.stop.assert_called()
|
|
# Verify audio data is int16 numpy array
|
|
audio_arg = mock_sd.play.call_args[0][0]
|
|
assert audio_arg.dtype == np.int16
|
|
assert len(audio_arg) > 0
|
|
|
|
# ============================================================================
|
|
# Silence detection
|
|
# ============================================================================
|
|
|
|
class TestSilenceDetection:
|
|
def test_silence_callback_fires_after_speech_then_silence(self, mock_sd, fake_clock):
|
|
np = pytest.importorskip("numpy")
|
|
import threading
|
|
|
|
mock_stream = MagicMock()
|
|
mock_sd.InputStream.return_value = mock_stream
|
|
|
|
from tools.voice_mode import AudioRecorder
|
|
|
|
recorder = AudioRecorder()
|
|
# Use very short durations for testing
|
|
recorder._silence_duration = 0.05
|
|
recorder._min_speech_duration = 0.05
|
|
|
|
fired = threading.Event()
|
|
|
|
def on_silence():
|
|
fired.set()
|
|
|
|
recorder.start(on_silence_stop=on_silence)
|
|
|
|
# Get the callback function from InputStream constructor
|
|
callback = mock_sd.InputStream.call_args.kwargs.get("callback")
|
|
if callback is None:
|
|
callback = mock_sd.InputStream.call_args[1]["callback"]
|
|
|
|
# Simulate sustained speech (multiple loud chunks to exceed min_speech_duration)
|
|
loud_frame = np.full((1600, 1), 5000, dtype="int16")
|
|
callback(loud_frame, 1600, None, None)
|
|
fake_clock.advance(0.06)
|
|
callback(loud_frame, 1600, None, None)
|
|
assert recorder._has_spoken is True
|
|
|
|
# Simulate silence
|
|
silent_frame = np.zeros((1600, 1), dtype="int16")
|
|
callback(silent_frame, 1600, None, None)
|
|
|
|
# Move past the silence duration, then send another silent frame
|
|
fake_clock.advance(0.06)
|
|
callback(silent_frame, 1600, None, None)
|
|
|
|
# The callback should have been fired (it runs on a real thread, so
|
|
# this wait is the one place real time is still involved)
|
|
assert fired.wait(timeout=5.0) is True
|
|
|
|
recorder.cancel()
|
|
|
|
def test_micro_pause_tolerance_during_speech(self, mock_sd, fake_clock):
|
|
"""Brief dips below threshold during speech should NOT reset speech tracking."""
|
|
np = pytest.importorskip("numpy")
|
|
import threading
|
|
|
|
mock_stream = MagicMock()
|
|
mock_sd.InputStream.return_value = mock_stream
|
|
|
|
from tools.voice_mode import AudioRecorder
|
|
|
|
recorder = AudioRecorder()
|
|
recorder._silence_duration = 0.05
|
|
recorder._min_speech_duration = 0.15
|
|
recorder._max_dip_tolerance = 0.1
|
|
|
|
fired = threading.Event()
|
|
recorder.start(on_silence_stop=lambda: fired.set())
|
|
|
|
callback = mock_sd.InputStream.call_args.kwargs.get("callback")
|
|
if callback is None:
|
|
callback = mock_sd.InputStream.call_args[1]["callback"]
|
|
|
|
loud_frame = np.full((1600, 1), 5000, dtype="int16")
|
|
quiet_frame = np.full((1600, 1), 50, dtype="int16")
|
|
|
|
# Speech chunk 1
|
|
callback(loud_frame, 1600, None, None)
|
|
fake_clock.advance(0.05)
|
|
# Brief micro-pause (dip < max_dip_tolerance)
|
|
callback(quiet_frame, 1600, None, None)
|
|
fake_clock.advance(0.05)
|
|
# Speech resumes -- speech_start should NOT have been reset
|
|
callback(loud_frame, 1600, None, None)
|
|
assert recorder._speech_start > 0, "Speech start should be preserved across brief dips"
|
|
fake_clock.advance(0.06)
|
|
# Another speech chunk to exceed min_speech_duration
|
|
callback(loud_frame, 1600, None, None)
|
|
assert recorder._has_spoken is True, "Speech should be confirmed after tolerating micro-pause"
|
|
|
|
recorder.cancel()
|
|
|
|
# ============================================================================
|
|
# Max recording length cap (voice.max_recording_seconds)
|
|
# ============================================================================
|
|
|
|
class TestMaxRecordingCap:
|
|
"""The hard cap must auto-stop through the real InputStream-callback
|
|
path — not just the predicate — and fire the one-shot callback exactly
|
|
once, independent of the silence-detection branches."""
|
|
|
|
def _get_stream_callback(self, mock_sd):
|
|
callback = mock_sd.InputStream.call_args.kwargs.get("callback")
|
|
if callback is None:
|
|
callback = mock_sd.InputStream.call_args[1]["callback"]
|
|
return callback
|
|
|
|
def test_cap_fires_one_shot_callback_during_continuous_speech(self, mock_sd, fake_clock):
|
|
np = pytest.importorskip("numpy")
|
|
import threading
|
|
|
|
mock_sd.InputStream.return_value = MagicMock()
|
|
|
|
from tools.voice_mode import AudioRecorder
|
|
|
|
recorder = AudioRecorder()
|
|
recorder._max_recording_seconds = 0.1
|
|
# Park the other auto-stop branches far away so only the cap can fire:
|
|
# loud frames keep the silence branch off, and max_wait covers the
|
|
# no-speech branch.
|
|
recorder._silence_duration = 60.0
|
|
recorder._max_wait = 60.0
|
|
|
|
fires = []
|
|
fired = threading.Event()
|
|
|
|
def on_stop():
|
|
fires.append(1)
|
|
fired.set()
|
|
|
|
recorder.start(on_silence_stop=on_stop)
|
|
callback = self._get_stream_callback(mock_sd)
|
|
|
|
loud_frame = np.full((1600, 1), 5000, dtype="int16")
|
|
callback(loud_frame, 1600, None, None)
|
|
assert not fired.is_set(), "cap must not fire before the limit elapses"
|
|
|
|
# Cross the cap while the user is STILL speaking — the silence branch
|
|
# can never fire here, so a hit proves the cap path.
|
|
fake_clock.advance(0.12)
|
|
callback(loud_frame, 1600, None, None)
|
|
assert fired.wait(timeout=5.0) is True
|
|
|
|
# One-shot: the handler cleared _on_silence_stop, further frames past
|
|
# the cap must not fire again — with the callback cleared, no further
|
|
# notifier thread is even spawned, so this needs no settle time.
|
|
assert recorder._on_silence_stop is None
|
|
callback(loud_frame, 1600, None, None)
|
|
assert len(fires) == 1
|
|
|
|
recorder.cancel()
|
|
|
|
def test_disabled_cap_never_fires_on_duration(self, mock_sd, fake_clock):
|
|
np = pytest.importorskip("numpy")
|
|
import threading
|
|
|
|
mock_sd.InputStream.return_value = MagicMock()
|
|
|
|
from tools.voice_mode import AudioRecorder
|
|
|
|
recorder = AudioRecorder()
|
|
recorder._max_recording_seconds = 0.0 # disabled (previous behaviour)
|
|
recorder._silence_duration = 60.0
|
|
recorder._max_wait = 60.0
|
|
|
|
fired = threading.Event()
|
|
recorder.start(on_silence_stop=lambda: fired.set())
|
|
callback = self._get_stream_callback(mock_sd)
|
|
|
|
loud_frame = np.full((1600, 1), 5000, dtype="int16")
|
|
callback(loud_frame, 1600, None, None)
|
|
fake_clock.advance(0.12)
|
|
callback(loud_frame, 1600, None, None)
|
|
|
|
assert fired.wait(timeout=0.1) is False
|
|
recorder.cancel()
|
|
|
|
|
|
# ============================================================================
|
|
# Playback interrupt
|
|
# ============================================================================
|
|
|
|
class TestPlaybackInterrupt:
|
|
"""Verify that TTS playback can be interrupted."""
|
|
|
|
def test_stop_playback_terminates_process(self):
|
|
from tools.voice_mode import stop_playback, _playback_lock
|
|
import tools.voice_mode as vm
|
|
|
|
mock_proc = MagicMock()
|
|
mock_proc.poll.return_value = None # process is running
|
|
|
|
with _playback_lock:
|
|
vm._active_playback = mock_proc
|
|
|
|
stop_playback()
|
|
|
|
mock_proc.terminate.assert_called_once()
|
|
|
|
with _playback_lock:
|
|
assert vm._active_playback is None
|
|
|
|
# ============================================================================
|
|
# Continuous mode flow
|
|
# ============================================================================
|
|
|
|
class TestContinuousModeFlow:
|
|
"""Verify continuous mode: auto-restart after transcription or silence."""
|
|
|
|
def test_continuous_restart_on_no_speech(self, mock_sd, temp_voice_dir):
|
|
np = pytest.importorskip("numpy")
|
|
|
|
mock_stream = MagicMock()
|
|
mock_sd.InputStream.return_value = mock_stream
|
|
|
|
from tools.voice_mode import AudioRecorder
|
|
|
|
recorder = AudioRecorder()
|
|
|
|
# First recording: only silence -> stop returns None
|
|
recorder.start()
|
|
callback = mock_sd.InputStream.call_args.kwargs.get("callback")
|
|
if callback is None:
|
|
callback = mock_sd.InputStream.call_args[1]["callback"]
|
|
|
|
for _ in range(10):
|
|
silence = np.full((1600, 1), 10, dtype="int16")
|
|
callback(silence, 1600, None, None)
|
|
|
|
wav_path = recorder.stop()
|
|
assert wav_path is None
|
|
|
|
# Simulate continuous mode restart
|
|
recorder.start()
|
|
assert recorder.is_recording is True
|
|
|
|
callback = mock_sd.InputStream.call_args.kwargs.get("callback")
|
|
if callback is None:
|
|
callback = mock_sd.InputStream.call_args[1]["callback"]
|
|
|
|
for _ in range(10):
|
|
speech = np.full((1600, 1), 5000, dtype="int16")
|
|
callback(speech, 1600, None, None)
|
|
|
|
wav_path = recorder.stop()
|
|
assert wav_path is not None
|
|
|
|
recorder.cancel()
|
|
|
|
# ============================================================================
|
|
# Audio level indicator
|
|
# ============================================================================
|
|
|
|
class TestAudioLevelIndicator:
|
|
"""Verify current_rms property updates in real-time for UI feedback."""
|
|
|
|
def test_peak_rms_tracks_maximum(self, mock_sd):
|
|
np = pytest.importorskip("numpy")
|
|
|
|
mock_stream = MagicMock()
|
|
mock_sd.InputStream.return_value = mock_stream
|
|
|
|
from tools.voice_mode import AudioRecorder
|
|
|
|
recorder = AudioRecorder()
|
|
recorder.start()
|
|
callback = mock_sd.InputStream.call_args.kwargs.get("callback")
|
|
if callback is None:
|
|
callback = mock_sd.InputStream.call_args[1]["callback"]
|
|
|
|
frames = [
|
|
np.full((1600, 1), 100, dtype="int16"),
|
|
np.full((1600, 1), 8000, dtype="int16"),
|
|
np.full((1600, 1), 500, dtype="int16"),
|
|
np.full((1600, 1), 3000, dtype="int16"),
|
|
]
|
|
for frame in frames:
|
|
callback(frame, 1600, None, None)
|
|
|
|
assert recorder._peak_rms == 8000
|
|
assert recorder.current_rms == 3000
|
|
|
|
recorder.cancel()
|
|
|
|
|
|
# ============================================================================
|
|
# Configurable silence parameters
|
|
# ============================================================================
|
|
|
|
class TestConfigurableSilenceParams:
|
|
"""Verify that silence detection params can be configured."""
|
|
|
|
def test_custom_threshold_and_duration(self, mock_sd, fake_clock):
|
|
np = pytest.importorskip("numpy")
|
|
|
|
mock_stream = MagicMock()
|
|
mock_sd.InputStream.return_value = mock_stream
|
|
|
|
from tools.voice_mode import AudioRecorder
|
|
import threading
|
|
|
|
recorder = AudioRecorder()
|
|
recorder._silence_threshold = 5000
|
|
recorder._silence_duration = 0.05
|
|
recorder._min_speech_duration = 0.05
|
|
|
|
fired = threading.Event()
|
|
recorder.start(on_silence_stop=lambda: fired.set())
|
|
callback = mock_sd.InputStream.call_args.kwargs.get("callback")
|
|
if callback is None:
|
|
callback = mock_sd.InputStream.call_args[1]["callback"]
|
|
|
|
# Audio at RMS 1000 -- below custom threshold (5000)
|
|
moderate = np.full((1600, 1), 1000, dtype="int16")
|
|
for _ in range(5):
|
|
callback(moderate, 1600, None, None)
|
|
fake_clock.advance(0.02)
|
|
|
|
assert recorder._has_spoken is False
|
|
assert fired.wait(timeout=0.2) is False
|
|
|
|
# Now send really loud audio (above 5000 threshold)
|
|
very_loud = np.full((1600, 1), 8000, dtype="int16")
|
|
callback(very_loud, 1600, None, None)
|
|
fake_clock.advance(0.06)
|
|
callback(very_loud, 1600, None, None)
|
|
assert recorder._has_spoken is True
|
|
|
|
recorder.cancel()
|
|
|
|
|
|
# ============================================================================
|
|
# Bugfix regression tests
|
|
# ============================================================================
|
|
|
|
|
|
class TestStreamLeakOnStartFailure:
|
|
"""Bug: stream.start() failure left stream unclosed."""
|
|
|
|
def test_stream_closed_on_start_failure(self, mock_sd):
|
|
mock_stream = MagicMock()
|
|
mock_stream.start.side_effect = OSError("Audio device busy")
|
|
mock_sd.InputStream.return_value = mock_stream
|
|
|
|
from tools.voice_mode import AudioRecorder
|
|
recorder = AudioRecorder()
|
|
|
|
with pytest.raises(RuntimeError, match="Failed to open audio input stream"):
|
|
recorder._ensure_stream()
|
|
|
|
mock_stream.close.assert_called_once()
|
|
|
|
|
|
# ============================================================================
|
|
# listen_for_speech — VAD barge-in monitor
|
|
# ============================================================================
|
|
|
|
class _FakeInputStream:
|
|
"""Context-manager InputStream serving a fixed sequence of RMS levels."""
|
|
|
|
def __init__(self, np, levels):
|
|
self._np = np
|
|
self._levels = list(levels)
|
|
self.reads = 0
|
|
|
|
def __enter__(self):
|
|
return self
|
|
|
|
def __exit__(self, *args):
|
|
return False
|
|
|
|
def read(self, frames):
|
|
level = self._levels[min(self.reads, len(self._levels) - 1)]
|
|
self.reads += 1
|
|
return self._np.full((frames, 1), level, dtype=self._np.int16), False
|
|
|
|
|
|
class TestListenForSpeech:
|
|
"""listen_for_speech: calibration → sustained-speech trigger → barge-in."""
|
|
|
|
CALIB_BLOCKS = 14 # 400ms / 30ms
|
|
TRIP_BLOCKS = 10 # 300ms / 30ms
|
|
|
|
def _run(self, mock_sd, levels, should_stop=None, **kwargs):
|
|
np = pytest.importorskip("numpy")
|
|
stream = _FakeInputStream(np, levels)
|
|
mock_sd.InputStream.return_value = stream
|
|
from tools.voice_mode import listen_for_speech
|
|
stops = iter([False] * 200 + [True] * 10_000)
|
|
return listen_for_speech(should_stop or (lambda: next(stops)), **kwargs), stream
|
|
|
|
def test_sustained_speech_triggers(self, mock_sd):
|
|
levels = [0] * self.CALIB_BLOCKS + [5000] * 50
|
|
heard, _ = self._run(mock_sd, levels)
|
|
assert heard is True
|
|
|
|
|
|
def test_returns_false_when_audio_unavailable(self, monkeypatch):
|
|
monkeypatch.setattr("tools.voice_mode._import_audio", MagicMock(side_effect=OSError("no audio")))
|
|
from tools.voice_mode import listen_for_speech
|
|
assert listen_for_speech(lambda: False) is False
|
|
|
|
def test_quiet_then_loud_playback_does_not_trip(self, mock_sd):
|
|
"""TTS that starts quiet and gets louder must NOT trip barge-in.
|
|
|
|
This is the core regression: a one-shot calibration freezes the
|
|
floor from the quiet opening, then louder TTS exceeds the stale
|
|
floor and false-triggers. The rolling window keeps the floor
|
|
current so the louder passage is absorbed into the floor.
|
|
"""
|
|
levels = [100] * self.CALIB_BLOCKS + [200] * 30 + [500] * 30 + [1000] * 30
|
|
heard, _ = self._run(mock_sd, levels)
|
|
assert heard is False
|
|
|
|
def test_silence_calibration_does_not_false_trip_on_tts(self, mock_sd):
|
|
"""Calibration during an inter-sentence gap must NOT false-trip.
|
|
|
|
If the grace period ends during a pause between TTS sentences, the
|
|
calibration window samples near-silence. Without the min_floor clamp,
|
|
min_floor locks near zero, the trigger drops to 400 RMS (SILENCE_RMS_THRESHOLD
|
|
* 2), and the next TTS sentence at 800 RMS exceeds it — those blocks are
|
|
excluded from the rolling window (rms >= trigger), the floor freezes, and
|
|
after sustained_ms the VAD false-triggers and cuts playback mid-sentence.
|
|
|
|
With the clamp, min_floor stays at SILENCE_RMS_THRESHOLD * 2 = 400, the
|
|
trigger is max(400, 400 * 8.0) = 3200, and 800-RMS TTS stays below it and
|
|
feeds the rolling floor. No false trip.
|
|
"""
|
|
# calibration_ms=800 → CALIB_BLOCKS = 800/30 ≈ 26 blocks of silence
|
|
# Then TTS resumes at 800 RMS — must NOT trip (below 3200 trigger).
|
|
calib = 800 // 30
|
|
levels = [0] * calib + [800] * 100
|
|
heard, _ = self._run(
|
|
mock_sd, levels,
|
|
sustained_ms=1000,
|
|
calibration_ms=800,
|
|
)
|
|
assert heard is False
|
|
|
|
|
|
class TestListenForSpeechCapture:
|
|
"""capture=True: the barge monitor records the interruption with pre-roll,
|
|
so the utterance is complete from its first syllable — nothing is lost
|
|
between detection and a recorder restart."""
|
|
|
|
CALIB_BLOCKS = 14 # 400ms / 30ms
|
|
LOUD_BLOCKS = 30 # speech: trips after 10, keeps talking
|
|
BLOCK = 480 # 16000 * 0.03
|
|
|
|
def _run(self, mock_sd, monkeypatch, levels, should_stop=None, **kwargs):
|
|
np = pytest.importorskip("numpy")
|
|
stream = _FakeInputStream(np, levels)
|
|
mock_sd.InputStream.return_value = stream
|
|
written = {}
|
|
monkeypatch.setattr(
|
|
"tools.voice_mode.AudioRecorder._write_wav",
|
|
staticmethod(lambda audio: written.update(audio=audio) or "/tmp/barge.wav"),
|
|
)
|
|
from tools.voice_mode import listen_for_speech
|
|
stops = iter([False] * 200 + [True] * 10_000)
|
|
path = listen_for_speech(
|
|
should_stop or (lambda: next(stops)), capture=True, **kwargs
|
|
)
|
|
return path, written.get("audio"), stream
|
|
|
|
def test_captured_utterance_includes_speech_onset(self, mock_sd, monkeypatch):
|
|
"""Every loud block — including the ones BEFORE detection tripped —
|
|
must land in the WAV. That pre-roll is the whole point."""
|
|
triggered = []
|
|
levels = [0] * self.CALIB_BLOCKS + [5000] * self.LOUD_BLOCKS + [0] * 500
|
|
path, audio, _ = self._run(
|
|
mock_sd, monkeypatch, levels,
|
|
should_stop=lambda: False,
|
|
on_trigger=lambda: triggered.append(True),
|
|
)
|
|
assert path == "/tmp/barge.wav"
|
|
assert triggered == [True]
|
|
assert int((audio == 5000).sum()) == self.LOUD_BLOCKS * self.BLOCK
|
|
|
|
def test_no_trip_returns_none(self, mock_sd, monkeypatch):
|
|
triggered = []
|
|
path, audio, _ = self._run(
|
|
mock_sd, monkeypatch, [0] * 500,
|
|
on_trigger=lambda: triggered.append(True),
|
|
)
|
|
assert path is None
|
|
assert audio is None
|
|
assert triggered == []
|
|
|
|
class TestFullDuplexListen:
|
|
"""full_duplex_listen: one agent-turn listener spanning generation and
|
|
playback — pre-playback calibration, phase-aware trigger, grace window."""
|
|
|
|
CALIB = 15 # 450ms / 30ms — pre-playback quiet calibration
|
|
TRIP = 10 # 300ms / 30ms window; trip needs >=8 above
|
|
GRACE = 16 # 500ms / 30ms
|
|
BLOCK = 480 # 30ms at 16 kHz
|
|
|
|
def _run(self, mock_sd, monkeypatch, levels, playing_from=None,
|
|
playing_until=None, should_stop=None, on_trigger=None, **kwargs):
|
|
np = pytest.importorskip("numpy")
|
|
stream = _FakeInputStream(np, levels)
|
|
mock_sd.InputStream.return_value = stream
|
|
written = {}
|
|
monkeypatch.setattr(
|
|
"tools.voice_mode.AudioRecorder._write_wav",
|
|
staticmethod(lambda audio: written.update(audio=audio) or "/tmp/fd.wav"),
|
|
)
|
|
from tools.voice_mode import full_duplex_listen
|
|
|
|
def is_playing():
|
|
if playing_from is None:
|
|
return False
|
|
if stream.reads < playing_from:
|
|
return False
|
|
if playing_until is not None and stream.reads >= playing_until:
|
|
return False
|
|
return True
|
|
|
|
stops = iter([False] * len(levels) + [True] * 10_000)
|
|
path = full_duplex_listen(
|
|
should_stop or (lambda: next(stops)),
|
|
is_playing=is_playing,
|
|
on_trigger=on_trigger,
|
|
**kwargs,
|
|
)
|
|
return path, written.get("audio"), stream
|
|
|
|
def test_generation_phase_speech_trips_and_captures(self, mock_sd, monkeypatch):
|
|
"""Speech while the LLM generates (no playback) trips with
|
|
phase='generation' and the utterance is captured with pre-roll."""
|
|
phases = []
|
|
levels = [100] * self.CALIB + [5000] * 30 + [0] * 500
|
|
path, audio, _ = self._run(
|
|
mock_sd, monkeypatch, levels,
|
|
on_trigger=lambda phase: phases.append(phase),
|
|
)
|
|
assert path == "/tmp/fd.wav"
|
|
assert phases == ["generation"]
|
|
assert int((audio == 5000).sum()) > 0 # speech onset in the capture
|
|
|
|
def test_playback_bleed_alone_does_not_trip(self, mock_sd, monkeypatch):
|
|
"""Speaker bleed (~1000 RMS) during playback stays below the playback
|
|
trigger clamp — the old monitor's self-calibration deafness class."""
|
|
phases = []
|
|
levels = [100] * self.CALIB + [1000] * 300
|
|
path, _, _ = self._run(
|
|
mock_sd, monkeypatch, levels,
|
|
playing_from=self.CALIB,
|
|
on_trigger=lambda phase: phases.append(phase),
|
|
)
|
|
assert path is None
|
|
assert phases == []
|
|
|
|
def test_speech_over_bleed_trips_in_playback_phase(self, mock_sd, monkeypatch):
|
|
"""Real speech (5000 RMS) over playback trips with phase='playback'
|
|
even though playback bleed was present — the trigger comes from the
|
|
PRE-playback quiet floor, never from bleed self-calibration."""
|
|
phases = []
|
|
# quiet calib → playback bleed (past grace) → user speaks over it
|
|
levels = (
|
|
[100] * self.CALIB
|
|
+ [1000] * (self.GRACE + 20)
|
|
+ [5000] * 30
|
|
+ [1000] * 200
|
|
)
|
|
path, _, _ = self._run(
|
|
mock_sd, monkeypatch, levels,
|
|
playing_from=self.CALIB,
|
|
on_trigger=lambda phase: phases.append(phase),
|
|
)
|
|
assert path == "/tmp/fd.wav"
|
|
assert phases == ["playback"]
|
|
|
|
def test_grace_window_suppresses_playback_onset(self, mock_sd, monkeypatch):
|
|
"""Loud blocks inside the 0.5s grace right after playback starts are
|
|
suppressed (onset transient), but speech after grace still trips."""
|
|
phases = []
|
|
# loud transient fully inside grace, then quiet bleed, then speech
|
|
levels = (
|
|
[100] * self.CALIB
|
|
+ [5000] * 8 # onset transient (inside 16-block grace)
|
|
+ [800] * 40
|
|
+ [5000] * 30
|
|
+ [0] * 200
|
|
)
|
|
|
|
def trig(phase):
|
|
phases.append(phase)
|
|
|
|
path, _, stream = self._run(
|
|
mock_sd, monkeypatch, levels,
|
|
playing_from=self.CALIB,
|
|
on_trigger=trig,
|
|
)
|
|
assert path == "/tmp/fd.wav"
|
|
assert phases == ["playback"]
|
|
# Trip must come from the post-grace speech, not the onset transient:
|
|
# by the time capture starts, we're past calib+transient+bleed blocks.
|
|
assert stream.reads > self.CALIB + 8 + 40
|
|
|
|
class TestGetBeepVolume:
|
|
"""Issue #55908: beep amplitude must come from config.yaml, with safe fallback."""
|
|
|
|
def _get(self):
|
|
from tools.voice_mode import _get_beep_volume
|
|
return _get_beep_volume()
|
|
|
|
@pytest.mark.parametrize("config,expected", [
|
|
({"voice": {}}, 0.3), # unset -> default
|
|
({"voice": {"beep_volume": 0.0}}, 0.0), # boundary, honored
|
|
({"voice": {"beep_volume": 1.5}}, 0.3), # out of range -> default
|
|
({"voice": {"beep_volume": "0.7"}}, 0.7), # numeric string coerced
|
|
({"voice": {"beep_volume": True}}, 0.3), # bool is not a volume
|
|
])
|
|
def test_config_value_resolution(self, config, expected):
|
|
with patch("hermes_cli.config.load_config", return_value=config):
|
|
assert self._get() == expected
|
|
|
|
def test_load_config_exception_falls_back(self):
|
|
with patch("hermes_cli.config.load_config",
|
|
side_effect=RuntimeError("broken config")):
|
|
assert self._get() == 0.3
|
|
|
|
# ============================================================================
|
|
# Device-native input sample rate — mics that reject 16 kHz capture
|
|
# ============================================================================
|
|
|
|
class TestDefaultInputSamplerate:
|
|
def test_uses_device_default_rate(self):
|
|
from tools.voice_mode import _default_input_samplerate
|
|
|
|
sd = MagicMock()
|
|
sd.query_devices.return_value = {"default_samplerate": 44100.0}
|
|
assert _default_input_samplerate(sd) == 44100
|
|
|
|
|
|
def test_wav_written_at_capture_rate(self, mock_sd, temp_voice_dir):
|
|
np = pytest.importorskip("numpy")
|
|
|
|
mock_sd.query_devices.return_value = {"default_samplerate": 48000.0}
|
|
mock_stream = MagicMock()
|
|
mock_sd.InputStream.return_value = mock_stream
|
|
|
|
from tools.voice_mode import AudioRecorder
|
|
|
|
recorder = AudioRecorder()
|
|
recorder.start()
|
|
|
|
# 1 second of loud audio at the device rate (above RMS threshold)
|
|
frame = np.full((48000, 1), 1000, dtype="int16")
|
|
recorder._frames = [frame]
|
|
recorder._peak_rms = 1000
|
|
|
|
wav_path = recorder.stop()
|
|
|
|
assert wav_path is not None
|
|
with wave.open(wav_path, "rb") as wf:
|
|
assert wf.getframerate() == 48000
|
|
|
|
|
|
class TestWSL2PowerShellFallback:
|
|
"""Regression tests for WSL2 PowerShell TTS fallback (issue #17608).
|
|
|
|
On WSL2 without a PulseAudio bridge, ffplay/aplay have no audio device.
|
|
play_audio_file() should insert a PowerShell-based player at the front
|
|
of the player list when powershell.exe and ffmpeg are available.
|
|
"""
|
|
|
|
def _fake_check_output(self, responses):
|
|
"""Build a subprocess.check_output side_effect from a list of responses."""
|
|
it = iter(responses)
|
|
def _side_effect(cmd, **kwargs):
|
|
return next(it)
|
|
return _side_effect
|
|
|
|
def test_powershell_pipeline_preserves_real_exit_status(self, sample_wav):
|
|
"""Regression (review of #63768): the shell pipeline must preserve
|
|
the (ffmpeg && powershell) exit status past the unconditional
|
|
cleanup, so a real conversion/playback failure falls through to the
|
|
next player instead of being masked by rm -f's always-zero exit."""
|
|
from unittest.mock import patch, MagicMock
|
|
from tools import voice_mode as vm
|
|
|
|
captured_cmds = []
|
|
|
|
def _capture_popen(cmd, **kw):
|
|
captured_cmds.append(list(cmd))
|
|
m = MagicMock()
|
|
# Simulate the PowerShell pipeline failing (nonzero rc), and
|
|
# the fallback ffplay succeeding.
|
|
if cmd[0] in ("/bin/sh", "sh"):
|
|
m.returncode = 1
|
|
else:
|
|
m.returncode = 0
|
|
m.wait = MagicMock(return_value=m.returncode)
|
|
return m
|
|
|
|
with patch("tools.voice_mode._is_wsl2_env", return_value=True), \
|
|
patch("tools.voice_mode._import_audio", side_effect=ImportError), \
|
|
patch("tools.voice_mode.shutil.which",
|
|
side_effect=lambda x: f"/bin/{x}" if x in ("powershell.exe", "ffmpeg", "ffplay", "sh") else (x if x.startswith("/") else None)), \
|
|
patch("tools.voice_mode.subprocess.check_output",
|
|
side_effect=self._fake_check_output([
|
|
b"C:/Temp\r\n",
|
|
b"/mnt/c/Temp\n",
|
|
b"C:/Temp/hermes.wav\n",
|
|
])), \
|
|
patch("tools.voice_mode.subprocess.Popen", side_effect=_capture_popen):
|
|
result = vm.play_audio_file(str(sample_wav))
|
|
|
|
assert result is True, "Must fall through to ffplay and succeed"
|
|
assert len(captured_cmds) == 2, (
|
|
f"Expected sh pipeline to be tried and fail, then ffplay to be "
|
|
f"tried: {captured_cmds}"
|
|
)
|
|
assert captured_cmds[0][0] in ("/bin/sh", "sh")
|
|
assert captured_cmds[1][0] == "ffplay"
|
|
# The subshell command must capture and re-exit with $rc, not rely
|
|
# on rm -f's exit status.
|
|
sh_script = captured_cmds[0][2]
|
|
assert "rc=$?" in sh_script and "exit $rc" in sh_script, (
|
|
"Shell pipeline must preserve the real exit status past cleanup: " + sh_script
|
|
)
|
|
|
|
def test_wsl2_unique_temp_filename(self, monkeypatch, tmp_path, sample_wav):
|
|
"""Two concurrent calls must use different temp WAV filenames."""
|
|
from unittest.mock import patch, MagicMock
|
|
from tools import voice_mode as vm
|
|
|
|
filenames = []
|
|
|
|
def _capture_check_output(cmd, **kwargs):
|
|
cmd_str = " ".join(str(c) for c in cmd)
|
|
if "TEMP" in cmd_str:
|
|
return b"C:\\Temp\r\n"
|
|
if "wslpath" in cmd_str and "-u" in cmd_str:
|
|
return b"/mnt/c/Temp\n"
|
|
if "wslpath" in cmd_str and "-w" in cmd_str:
|
|
wsl_path = cmd[-1] if isinstance(cmd[-1], str) else cmd[-1].decode()
|
|
filenames.append(wsl_path.split("/")[-1])
|
|
return f"C:\\Temp\\{wsl_path.split('/')[-1]}\n".encode()
|
|
return b""
|
|
|
|
def _fake_open(path, *args, **kwargs):
|
|
if str(path) == "/proc/version":
|
|
import io
|
|
return io.StringIO("Linux Microsoft WSL2")
|
|
return open(path, *args, **kwargs)
|
|
|
|
with patch("builtins.open", side_effect=_fake_open), \
|
|
patch("shutil.which", side_effect=lambda x: f"/bin/{x}" if x in ("powershell.exe", "ffmpeg", "ffplay") else None), \
|
|
patch("subprocess.check_output", side_effect=_capture_check_output), \
|
|
patch("subprocess.Popen", return_value=MagicMock(returncode=0, wait=lambda **k: 0)), \
|
|
patch("tools.voice_mode._playback_lock"), \
|
|
patch("tools.voice_mode._active_playback", None):
|
|
vm.play_audio_file(str(sample_wav))
|
|
vm.play_audio_file(str(sample_wav))
|
|
|
|
# Regression (review of #63768): the original test made this
|
|
# assertion conditional on len(filenames) >= 2, so a broken
|
|
# (zero-captured) run passed trivially. Require exactly two.
|
|
assert len(filenames) == 2, (
|
|
f"Expected exactly 2 captured temp filenames from 2 calls, got "
|
|
f"{len(filenames)}: {filenames}"
|
|
)
|
|
assert filenames[0] != filenames[1], (
|
|
"Concurrent TTS calls must use unique temp WAV filenames"
|
|
)
|
|
|
|
def test_non_wsl_skips_powershell_fallback(self, monkeypatch, sample_wav):
|
|
"""On non-WSL Linux, the PowerShell player must not be inserted."""
|
|
from unittest.mock import patch, MagicMock
|
|
from tools import voice_mode as vm
|
|
|
|
captured_players = []
|
|
|
|
def _capture_popen(cmd, **kw):
|
|
captured_players.append(cmd)
|
|
m = MagicMock()
|
|
m.returncode = 0
|
|
m.wait.return_value = 0
|
|
return m
|
|
|
|
def _fake_open(path, *args, **kwargs):
|
|
if str(path) == "/proc/version":
|
|
import io
|
|
return io.StringIO("Linux version 5.15.0-generic #72-Ubuntu")
|
|
return open(path, *args, **kwargs)
|
|
|
|
with patch("builtins.open", side_effect=_fake_open), \
|
|
patch("tools.voice_mode._import_audio", side_effect=ImportError), \
|
|
patch("shutil.which", side_effect=lambda x: f"/bin/{x}" if x in ("ffplay", "aplay") else None), \
|
|
patch("subprocess.Popen", side_effect=_capture_popen), \
|
|
patch("tools.voice_mode._playback_lock"), \
|
|
patch("tools.voice_mode._active_playback", None):
|
|
vm.play_audio_file(str(sample_wav))
|
|
|
|
assert captured_players, "No players were tried"
|
|
for cmd in captured_players:
|
|
assert not (cmd[0] == "sh" and "powershell" in " ".join(str(c) for c in cmd)), (
|
|
"PowerShell player must not appear on non-WSL Linux"
|
|
)
|
|
|
|
|
|
class TestWSLAudioEnvironmentGate:
|
|
"""Regression tests (review of #63768) for detect_audio_environment()'s
|
|
WSL gate: when the PowerShell TTS fallback is viable, voice mode must
|
|
not be hard-blocked, but the recording/STT PulseAudio-bridge guidance
|
|
must still be surfaced (as a non-blocking notice)."""
|
|
|
|
def _fake_open_wsl(self, path, *args, **kwargs):
|
|
if str(path) == "/proc/version":
|
|
import io
|
|
return io.StringIO("Linux version 5.15 Microsoft Standard WSL2")
|
|
return open(path, *args, **kwargs)
|
|
|
|
def test_wsl_no_pulse_but_powershell_available_not_hard_blocked(self, monkeypatch):
|
|
from unittest.mock import patch
|
|
from tools import voice_mode as vm
|
|
|
|
monkeypatch.delenv("PULSE_SERVER", raising=False)
|
|
monkeypatch.delenv("PIPEWIRE_REMOTE", raising=False)
|
|
for _ssh_var in ("SSH_CLIENT", "SSH_TTY", "SSH_CONNECTION"):
|
|
monkeypatch.delenv(_ssh_var, raising=False)
|
|
monkeypatch.setattr("tools.voice_mode._import_audio",
|
|
lambda: (MagicMock(), MagicMock()))
|
|
with patch("builtins.open", side_effect=self._fake_open_wsl), \
|
|
patch("tools.voice_mode._wsl_powershell_tts_available", return_value=True), \
|
|
patch("tools.voice_mode._pulse_socket_reachable", return_value=False), \
|
|
patch("hermes_constants.is_container", return_value=False):
|
|
result = vm.detect_audio_environment()
|
|
|
|
assert result["available"] is True, (
|
|
"PowerShell TTS fallback must keep voice mode enabled even "
|
|
"without a PulseAudio bridge: " + str(result["warnings"])
|
|
)
|
|
assert any("PowerShell" in n or "Media.SoundPlayer" in n for n in result["notices"]), (
|
|
"The PowerShell fallback path must be mentioned in notices"
|
|
)
|
|
assert any("recording" in n.lower() or "PulseAudio" in n for n in result["notices"]), (
|
|
"The recording/STT PulseAudio caveat must still be surfaced"
|
|
)
|
|
|
|
def test_wsl_no_pulse_no_powershell_still_blocked(self, monkeypatch):
|
|
from unittest.mock import patch
|
|
from tools import voice_mode as vm
|
|
|
|
monkeypatch.delenv("PULSE_SERVER", raising=False)
|
|
monkeypatch.delenv("PIPEWIRE_REMOTE", raising=False)
|
|
for _ssh_var in ("SSH_CLIENT", "SSH_TTY", "SSH_CONNECTION"):
|
|
monkeypatch.delenv(_ssh_var, raising=False)
|
|
monkeypatch.setattr("tools.voice_mode._import_audio",
|
|
lambda: (MagicMock(), MagicMock()))
|
|
with patch("builtins.open", side_effect=self._fake_open_wsl), \
|
|
patch("tools.voice_mode._wsl_powershell_tts_available", return_value=False), \
|
|
patch("tools.voice_mode._pulse_socket_reachable", return_value=False), \
|
|
patch("hermes_constants.is_container", return_value=False):
|
|
result = vm.detect_audio_environment()
|
|
|
|
assert result["available"] is False, (
|
|
"Without PulseAudio AND without the PowerShell fallback, WSL "
|
|
"must still be hard-blocked as before"
|
|
)
|
|
|
|
def test_wsl_with_pulse_server_unaffected(self, monkeypatch):
|
|
"""PULSE_SERVER already configured: existing behavior unchanged."""
|
|
from unittest.mock import patch
|
|
from tools import voice_mode as vm
|
|
|
|
monkeypatch.setenv("PULSE_SERVER", "unix:/mnt/wslg/PulseServer")
|
|
for _ssh_var in ("SSH_CLIENT", "SSH_TTY", "SSH_CONNECTION"):
|
|
monkeypatch.delenv(_ssh_var, raising=False)
|
|
monkeypatch.setattr("tools.voice_mode._import_audio",
|
|
lambda: (MagicMock(), MagicMock()))
|
|
with patch("builtins.open", side_effect=self._fake_open_wsl), \
|
|
patch("hermes_constants.is_container", return_value=False):
|
|
result = vm.detect_audio_environment()
|
|
|
|
assert result["available"] is True
|
|
# Merged with #37346: any forwarded sound server (PULSE_SERVER or
|
|
# PIPEWIRE_REMOTE) yields the shared reachable-sound-server notice.
|
|
assert any(
|
|
"PulseAudio" in n and "WSL" in n for n in result["notices"]
|
|
)
|