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Standardize assist pipelines on 10ms chunk size (#123024)

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* Make chunk size always 10ms

* Fix voip
This commit is contained in:
Michael Hansen 2024-08-01 15:39:17 -05:00 committed by GitHub
parent a3b5dcc21b
commit 80aa2c269b
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GPG Key ID: B5690EEEBB952194
9 changed files with 154 additions and 159 deletions
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8
homeassistant/components/assist_pipeline/__init__.py
View File

@ -16,6 +16,10 @@ from .const import (
DATA_LAST_WAKE_UP,
DOMAIN,
EVENT_RECORDING,
SAMPLE_CHANNELS,
SAMPLE_RATE,
SAMPLE_WIDTH,
SAMPLES_PER_CHUNK,
)
from .error import PipelineNotFound
from .pipeline import (
@ -53,6 +57,10 @@ __all__ = (
"PipelineNotFound",
"WakeWordSettings",
"EVENT_RECORDING",
"SAMPLES_PER_CHUNK",
"SAMPLE_RATE",
"SAMPLE_WIDTH",
"SAMPLE_CHANNELS",
)
CONFIG_SCHEMA = vol.Schema(

18
homeassistant/components/assist_pipeline/audio_enhancer.py
View File

@ -6,6 +6,8 @@ import logging
from pymicro_vad import MicroVad
from .const import BYTES_PER_CHUNK
_LOGGER = logging.getLogger(__name__)
@ -38,11 +40,6 @@ class AudioEnhancer(ABC):
def enhance_chunk(self, audio: bytes, timestamp_ms: int) -> EnhancedAudioChunk:
"""Enhance chunk of PCM audio @ 16Khz with 16-bit mono samples."""
@property
@abstractmethod
def samples_per_chunk(self) -> int | None:
"""Return number of samples per chunk or None if chunking isn't required."""
class MicroVadEnhancer(AudioEnhancer):
"""Audio enhancer that just runs microVAD."""
@ -61,22 +58,15 @@ class MicroVadEnhancer(AudioEnhancer):
_LOGGER.debug("Initialized microVAD with threshold=%s", self.threshold)
def enhance_chunk(self, audio: bytes, timestamp_ms: int) -> EnhancedAudioChunk:
"""Enhance chunk of PCM audio @ 16Khz with 16-bit mono samples."""
"""Enhance 10ms chunk of PCM audio @ 16Khz with 16-bit mono samples."""
is_speech: bool | None = None
if self.vad is not None:
# Run VAD
assert len(audio) == BYTES_PER_CHUNK
speech_prob = self.vad.Process10ms(audio)
is_speech = speech_prob > self.threshold
return EnhancedAudioChunk(
audio=audio, timestamp_ms=timestamp_ms, is_speech=is_speech
)
@property
def samples_per_chunk(self) -> int | None:
"""Return number of samples per chunk or None if chunking isn't required."""
if self.is_vad_enabled:
return 160 # 10ms
return None

4
homeassistant/components/assist_pipeline/const.py
View File

@ -19,4 +19,6 @@ EVENT_RECORDING = f"{DOMAIN}_recording"
SAMPLE_RATE = 16000 # hertz
SAMPLE_WIDTH = 2 # bytes
SAMPLE_CHANNELS = 1 # mono
SAMPLES_PER_CHUNK = 240 # 20 ms @ 16Khz
MS_PER_CHUNK = 10
SAMPLES_PER_CHUNK = SAMPLE_RATE // (1000 // MS_PER_CHUNK) # 10 ms @ 16Khz
BYTES_PER_CHUNK = SAMPLES_PER_CHUNK * SAMPLE_WIDTH * SAMPLE_CHANNELS # 16-bit

75
homeassistant/components/assist_pipeline/pipeline.py
View File

@ -51,11 +51,13 @@ from homeassistant.util.limited_size_dict import LimitedSizeDict
from .audio_enhancer import AudioEnhancer, EnhancedAudioChunk, MicroVadEnhancer
from .const import (
BYTES_PER_CHUNK,
CONF_DEBUG_RECORDING_DIR,
DATA_CONFIG,
DATA_LAST_WAKE_UP,
DATA_MIGRATIONS,
DOMAIN,
MS_PER_CHUNK,
SAMPLE_CHANNELS,
SAMPLE_RATE,
SAMPLE_WIDTH,
@ -502,9 +504,6 @@ class AudioSettings:
is_vad_enabled: bool = True
"""True if VAD is used to determine the end of the voice command."""
samples_per_chunk: int | None = None
"""Number of samples that will be in each audio chunk (None for no chunking)."""
silence_seconds: float = 0.5
"""Seconds of silence after voice command has ended."""
@ -525,11 +524,6 @@ class AudioSettings:
or (self.auto_gain_dbfs > 0)
)
@property
def is_chunking_enabled(self) -> bool:
"""True if chunk size is set."""
return self.samples_per_chunk is not None
@dataclass
class PipelineRun:
@ -566,7 +560,9 @@ class PipelineRun:
audio_enhancer: AudioEnhancer | None = None
"""VAD/noise suppression/auto gain"""
audio_chunking_buffer: AudioBuffer | None = None
audio_chunking_buffer: AudioBuffer = field(
default_factory=lambda: AudioBuffer(BYTES_PER_CHUNK)
)
"""Buffer used when splitting audio into chunks for audio processing"""
_device_id: str | None = None
@ -599,8 +595,6 @@ class PipelineRun:
self.audio_settings.is_vad_enabled,
)
self.audio_chunking_buffer = AudioBuffer(self.samples_per_chunk * SAMPLE_WIDTH)
def __eq__(self, other: object) -> bool:
"""Compare pipeline runs by id."""
if isinstance(other, PipelineRun):
@ -608,14 +602,6 @@ class PipelineRun:
return False
@property
def samples_per_chunk(self) -> int:
"""Return number of samples expected in each audio chunk."""
if self.audio_enhancer is not None:
return self.audio_enhancer.samples_per_chunk or SAMPLES_PER_CHUNK
return self.audio_settings.samples_per_chunk or SAMPLES_PER_CHUNK
@callback
def process_event(self, event: PipelineEvent) -> None:
"""Log an event and call listener."""
@ -728,7 +714,7 @@ class PipelineRun:
# after wake-word-detection.
num_audio_chunks_to_buffer = int(
(wake_word_settings.audio_seconds_to_buffer * SAMPLE_RATE)
/ self.samples_per_chunk
/ SAMPLES_PER_CHUNK
)
stt_audio_buffer: deque[EnhancedAudioChunk] | None = None
@ -1216,60 +1202,31 @@ class PipelineRun:
self.debug_recording_thread = None
async def process_volume_only(
self,
audio_stream: AsyncIterable[bytes],
sample_rate: int = SAMPLE_RATE,
sample_width: int = SAMPLE_WIDTH,
self, audio_stream: AsyncIterable[bytes]
) -> AsyncGenerator[EnhancedAudioChunk]:
"""Apply volume transformation only (no VAD/audio enhancements) with optional chunking."""
assert self.audio_chunking_buffer is not None
bytes_per_chunk = self.samples_per_chunk * sample_width
ms_per_sample = sample_rate // 1000
ms_per_chunk = self.samples_per_chunk // ms_per_sample
timestamp_ms = 0
async for chunk in audio_stream:
if self.audio_settings.volume_multiplier != 1.0:
chunk = _multiply_volume(chunk, self.audio_settings.volume_multiplier)
if self.audio_settings.is_chunking_enabled:
for sub_chunk in chunk_samples(
chunk, bytes_per_chunk, self.audio_chunking_buffer
):
yield EnhancedAudioChunk(
audio=sub_chunk,
timestamp_ms=timestamp_ms,
is_speech=None, # no VAD
)
timestamp_ms += ms_per_chunk
else:
# No chunking
for sub_chunk in chunk_samples(
chunk, BYTES_PER_CHUNK, self.audio_chunking_buffer
):
yield EnhancedAudioChunk(
audio=chunk,
audio=sub_chunk,
timestamp_ms=timestamp_ms,
is_speech=None, # no VAD
)
timestamp_ms += (len(chunk) // sample_width) // ms_per_sample
timestamp_ms += MS_PER_CHUNK
async def process_enhance_audio(
self,
audio_stream: AsyncIterable[bytes],
sample_rate: int = SAMPLE_RATE,
sample_width: int = SAMPLE_WIDTH,
self, audio_stream: AsyncIterable[bytes]
) -> AsyncGenerator[EnhancedAudioChunk]:
"""Split audio into 10 ms chunks and apply VAD/noise suppression/auto gain/volume transformation."""
"""Split audio into chunks and apply VAD/noise suppression/auto gain/volume transformation."""
assert self.audio_enhancer is not None
assert self.audio_enhancer.samples_per_chunk is not None
assert self.audio_chunking_buffer is not None
bytes_per_chunk = self.audio_enhancer.samples_per_chunk * sample_width
ms_per_sample = sample_rate // 1000
ms_per_chunk = (
self.audio_enhancer.samples_per_chunk // sample_width
) // ms_per_sample
timestamp_ms = 0
async for dirty_samples in audio_stream:
if self.audio_settings.volume_multiplier != 1.0:
# Static gain
@ -1279,10 +1236,10 @@ class PipelineRun:
# Split into chunks for audio enhancements/VAD
for dirty_chunk in chunk_samples(
dirty_samples, bytes_per_chunk, self.audio_chunking_buffer
dirty_samples, BYTES_PER_CHUNK, self.audio_chunking_buffer
):
yield self.audio_enhancer.enhance_chunk(dirty_chunk, timestamp_ms)
timestamp_ms += ms_per_chunk
timestamp_ms += MS_PER_CHUNK
def _multiply_volume(chunk: bytes, volume_multiplier: float) -> bytes:

22
homeassistant/components/voip/voip.py
View File

@ -21,7 +21,7 @@ from voip_utils import (
VoipDatagramProtocol,
)
from homeassistant.components import stt, tts
from homeassistant.components import assist_pipeline, stt, tts
from homeassistant.components.assist_pipeline import (
Pipeline,
PipelineEvent,
@ -331,15 +331,14 @@ class PipelineRtpDatagramProtocol(RtpDatagramProtocol):
async with asyncio.timeout(self.audio_timeout):
chunk = await self._audio_queue.get()
assert audio_enhancer.samples_per_chunk is not None
vad_buffer = AudioBuffer(audio_enhancer.samples_per_chunk * WIDTH)
vad_buffer = AudioBuffer(assist_pipeline.SAMPLES_PER_CHUNK * WIDTH)
while chunk:
chunk_buffer.append(chunk)
segmenter.process_with_vad(
chunk,
audio_enhancer.samples_per_chunk,
assist_pipeline.SAMPLES_PER_CHUNK,
lambda x: audio_enhancer.enhance_chunk(x, 0).is_speech is True,
vad_buffer,
)
@ -371,13 +370,12 @@ class PipelineRtpDatagramProtocol(RtpDatagramProtocol):
async with asyncio.timeout(self.audio_timeout):
chunk = await self._audio_queue.get()
assert audio_enhancer.samples_per_chunk is not None
vad_buffer = AudioBuffer(audio_enhancer.samples_per_chunk * WIDTH)
vad_buffer = AudioBuffer(assist_pipeline.SAMPLES_PER_CHUNK * WIDTH)
while chunk:
if not segmenter.process_with_vad(
chunk,
audio_enhancer.samples_per_chunk,
assist_pipeline.SAMPLES_PER_CHUNK,
lambda x: audio_enhancer.enhance_chunk(x, 0).is_speech is True,
vad_buffer,
):
@ -437,13 +435,13 @@ class PipelineRtpDatagramProtocol(RtpDatagramProtocol):
sample_channels = wav_file.getnchannels()
if (
(sample_rate != 16000)
or (sample_width != 2)
or (sample_channels != 1)
(sample_rate != RATE)
or (sample_width != WIDTH)
or (sample_channels != CHANNELS)
):
raise ValueError(
"Expected rate/width/channels as 16000/2/1,"
" got {sample_rate}/{sample_width}/{sample_channels}}"
f"Expected rate/width/channels as {RATE}/{WIDTH}/{CHANNELS},"
f" got {sample_rate}/{sample_width}/{sample_channels}"
)
audio_bytes = wav_file.readframes(wav_file.getnframes())

13
tests/components/assist_pipeline/conftest.py
View File

@ -11,6 +11,12 @@ import pytest
from homeassistant.components import stt, tts, wake_word
from homeassistant.components.assist_pipeline import DOMAIN, select as assist_select
from homeassistant.components.assist_pipeline.const import (
BYTES_PER_CHUNK,
SAMPLE_CHANNELS,
SAMPLE_RATE,
SAMPLE_WIDTH,
)
from homeassistant.components.assist_pipeline.pipeline import (
PipelineData,
PipelineStorageCollection,
@ -33,6 +39,8 @@ from tests.common import (
_TRANSCRIPT = "test transcript"
BYTES_ONE_SECOND = SAMPLE_RATE * SAMPLE_WIDTH * SAMPLE_CHANNELS
@pytest.fixture(autouse=True)
def mock_tts_cache_dir_autouse(mock_tts_cache_dir: Path) -> Path:
@ -462,3 +470,8 @@ def pipeline_data(hass: HomeAssistant, init_components) -> PipelineData:
def pipeline_storage(pipeline_data) -> PipelineStorageCollection:
"""Return pipeline storage collection."""
return pipeline_data.pipeline_store
def make_10ms_chunk(header: bytes) -> bytes:
"""Return 10ms of zeros with the given header."""
return header + bytes(BYTES_PER_CHUNK - len(header))

2
tests/components/assist_pipeline/snapshots/test_websocket.ambr
View File

@ -440,7 +440,7 @@
# ---
# name: test_device_capture_override.2
dict({
'audio': 'Y2h1bmsx',
'audio': 'Y2h1bmsxAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA=',
'channels': 1,
'rate': 16000,
'type': 'audio',

71
tests/components/assist_pipeline/test_init.py
View File

@ -13,6 +13,7 @@ from syrupy.assertion import SnapshotAssertion
from homeassistant.components import assist_pipeline, media_source, stt, tts
from homeassistant.components.assist_pipeline.const import (
BYTES_PER_CHUNK,
CONF_DEBUG_RECORDING_DIR,
DOMAIN,
)
@ -20,16 +21,16 @@ from homeassistant.core import Context, HomeAssistant
from homeassistant.setup import async_setup_component
from .conftest import (
BYTES_ONE_SECOND,
MockSttProvider,
MockSttProviderEntity,
MockTTSProvider,
MockWakeWordEntity,
make_10ms_chunk,
)
from tests.typing import ClientSessionGenerator, WebSocketGenerator
BYTES_ONE_SECOND = 16000 * 2
def process_events(events: list[assist_pipeline.PipelineEvent]) -> list[dict]:
"""Process events to remove dynamic values."""
@ -58,8 +59,8 @@ async def test_pipeline_from_audio_stream_auto(
events: list[assist_pipeline.PipelineEvent] = []
async def audio_data():
yield b"part1"
yield b"part2"
yield make_10ms_chunk(b"part1")
yield make_10ms_chunk(b"part2")
yield b""
await assist_pipeline.async_pipeline_from_audio_stream(
@ -79,7 +80,9 @@ async def test_pipeline_from_audio_stream_auto(
)
assert process_events(events) == snapshot
assert mock_stt_provider.received == [b"part1", b"part2"]
assert len(mock_stt_provider.received) == 2
assert mock_stt_provider.received[0].startswith(b"part1")
assert mock_stt_provider.received[1].startswith(b"part2")
async def test_pipeline_from_audio_stream_legacy(
@ -98,8 +101,8 @@ async def test_pipeline_from_audio_stream_legacy(
events: list[assist_pipeline.PipelineEvent] = []
async def audio_data():
yield b"part1"
yield b"part2"
yield make_10ms_chunk(b"part1")
yield make_10ms_chunk(b"part2")
yield b""
# Create a pipeline using an stt entity
@ -142,7 +145,9 @@ async def test_pipeline_from_audio_stream_legacy(
)
assert process_events(events) == snapshot
assert mock_stt_provider.received == [b"part1", b"part2"]
assert len(mock_stt_provider.received) == 2
assert mock_stt_provider.received[0].startswith(b"part1")
assert mock_stt_provider.received[1].startswith(b"part2")
async def test_pipeline_from_audio_stream_entity(
@ -161,8 +166,8 @@ async def test_pipeline_from_audio_stream_entity(
events: list[assist_pipeline.PipelineEvent] = []
async def audio_data():
yield b"part1"
yield b"part2"
yield make_10ms_chunk(b"part1")
yield make_10ms_chunk(b"part2")
yield b""
# Create a pipeline using an stt entity
@ -205,7 +210,9 @@ async def test_pipeline_from_audio_stream_entity(
)
assert process_events(events) == snapshot
assert mock_stt_provider_entity.received == [b"part1", b"part2"]
assert len(mock_stt_provider_entity.received) == 2
assert mock_stt_provider_entity.received[0].startswith(b"part1")
assert mock_stt_provider_entity.received[1].startswith(b"part2")
async def test_pipeline_from_audio_stream_no_stt(
@ -224,8 +231,8 @@ async def test_pipeline_from_audio_stream_no_stt(
events: list[assist_pipeline.PipelineEvent] = []
async def audio_data():
yield b"part1"
yield b"part2"
yield make_10ms_chunk(b"part1")
yield make_10ms_chunk(b"part2")
yield b""
# Create a pipeline without stt support
@ -285,8 +292,8 @@ async def test_pipeline_from_audio_stream_unknown_pipeline(
events: list[assist_pipeline.PipelineEvent] = []
async def audio_data():
yield b"part1"
yield b"part2"
yield make_10ms_chunk(b"part1")
yield make_10ms_chunk(b"part2")
yield b""
# Try to use the created pipeline
@ -327,7 +334,7 @@ async def test_pipeline_from_audio_stream_wake_word(
# [0, 2, ...]
wake_chunk_2 = bytes(it.islice(it.cycle(range(0, 256, 2)), BYTES_ONE_SECOND))
samples_per_chunk = 160
samples_per_chunk = 160 # 10ms @ 16Khz
bytes_per_chunk = samples_per_chunk * 2 # 16-bit
async def audio_data():
@ -343,8 +350,8 @@ async def test_pipeline_from_audio_stream_wake_word(
yield wake_chunk_2[i : i + bytes_per_chunk]
i += bytes_per_chunk
for chunk in (b"wake word!", b"part1", b"part2"):
yield chunk + bytes(bytes_per_chunk - len(chunk))
for header in (b"wake word!", b"part1", b"part2"):
yield make_10ms_chunk(header)
yield b""
@ -365,9 +372,7 @@ async def test_pipeline_from_audio_stream_wake_word(
wake_word_settings=assist_pipeline.WakeWordSettings(
audio_seconds_to_buffer=1.5
),
audio_settings=assist_pipeline.AudioSettings(
is_vad_enabled=False, samples_per_chunk=samples_per_chunk
),
audio_settings=assist_pipeline.AudioSettings(is_vad_enabled=False),
)
assert process_events(events) == snapshot
@ -408,13 +413,11 @@ async def test_pipeline_save_audio(
pipeline = assist_pipeline.async_get_pipeline(hass)
events: list[assist_pipeline.PipelineEvent] = []
# Pad out to an even number of bytes since these "samples" will be saved
# as 16-bit values.
async def audio_data():
yield b"wake word_"
yield make_10ms_chunk(b"wake word")
# queued audio
yield b"part1_"
yield b"part2_"
yield make_10ms_chunk(b"part1")
yield make_10ms_chunk(b"part2")
yield b""
await assist_pipeline.async_pipeline_from_audio_stream(
@ -457,12 +460,16 @@ async def test_pipeline_save_audio(
# Verify wake file
with wave.open(str(wake_file), "rb") as wake_wav:
wake_data = wake_wav.readframes(wake_wav.getnframes())
assert wake_data == b"wake word_"
assert wake_data.startswith(b"wake word")
# Verify stt file
with wave.open(str(stt_file), "rb") as stt_wav:
stt_data = stt_wav.readframes(stt_wav.getnframes())
assert stt_data == b"queued audiopart1_part2_"
assert stt_data.startswith(b"queued audio")
stt_data = stt_data[len(b"queued audio") :]
assert stt_data.startswith(b"part1")
stt_data = stt_data[BYTES_PER_CHUNK:]
assert stt_data.startswith(b"part2")
async def test_pipeline_saved_audio_with_device_id(
@ -645,10 +652,10 @@ async def test_wake_word_detection_aborted(
events: list[assist_pipeline.PipelineEvent] = []
async def audio_data():
yield b"silence!"
yield b"wake word!"
yield b"part1"
yield b"part2"
yield make_10ms_chunk(b"silence!")
yield make_10ms_chunk(b"wake word!")
yield make_10ms_chunk(b"part1")
yield make_10ms_chunk(b"part2")
yield b""
pipeline_store = pipeline_data.pipeline_store

100
tests/components/assist_pipeline/test_websocket.py
View File

@ -8,7 +8,12 @@ from unittest.mock import ANY, patch
import pytest
from syrupy.assertion import SnapshotAssertion
from homeassistant.components.assist_pipeline.const import DOMAIN
from homeassistant.components.assist_pipeline.const import (
DOMAIN,
SAMPLE_CHANNELS,
SAMPLE_RATE,
SAMPLE_WIDTH,
)
from homeassistant.components.assist_pipeline.pipeline import (
DeviceAudioQueue,
Pipeline,
@ -18,7 +23,13 @@ from homeassistant.core import HomeAssistant
from homeassistant.exceptions import HomeAssistantError
from homeassistant.helpers import device_registry as dr
from .conftest import MockWakeWordEntity, MockWakeWordEntity2
from .conftest import (
BYTES_ONE_SECOND,
BYTES_PER_CHUNK,
MockWakeWordEntity,
MockWakeWordEntity2,
make_10ms_chunk,
)
from tests.common import MockConfigEntry
from tests.typing import WebSocketGenerator
@ -205,7 +216,7 @@ async def test_audio_pipeline_with_wake_word_timeout(
"start_stage": "wake_word",
"end_stage": "tts",
"input": {
"sample_rate": 16000,
"sample_rate": SAMPLE_RATE,
"timeout": 1,
},
}
@ -229,7 +240,7 @@ async def test_audio_pipeline_with_wake_word_timeout(
events.append(msg["event"])
# 2 seconds of silence
await client.send_bytes(bytes([1]) + bytes(16000 * 2 * 2))
await client.send_bytes(bytes([1]) + bytes(2 * BYTES_ONE_SECOND))
# Time out error
msg = await client.receive_json()
@ -259,7 +270,7 @@ async def test_audio_pipeline_with_wake_word_no_timeout(
"type": "assist_pipeline/run",
"start_stage": "wake_word",
"end_stage": "tts",
"input": {"sample_rate": 16000, "timeout": 0, "no_vad": True},
"input": {"sample_rate": SAMPLE_RATE, "timeout": 0, "no_vad": True},
}
)
@ -282,9 +293,10 @@ async def test_audio_pipeline_with_wake_word_no_timeout(
events.append(msg["event"])
# "audio"
await client.send_bytes(bytes([handler_id]) + b"wake word")
await client.send_bytes(bytes([handler_id]) + make_10ms_chunk(b"wake word"))
msg = await client.receive_json()
async with asyncio.timeout(1):
msg = await client.receive_json()
assert msg["event"]["type"] == "wake_word-end"
assert msg["event"]["data"] == snapshot
events.append(msg["event"])
@ -365,7 +377,7 @@ async def test_audio_pipeline_no_wake_word_engine(
"start_stage": "wake_word",
"end_stage": "tts",
"input": {
"sample_rate": 16000,
"sample_rate": SAMPLE_RATE,
},
}
)
@ -402,7 +414,7 @@ async def test_audio_pipeline_no_wake_word_entity(
"start_stage": "wake_word",
"end_stage": "tts",
"input": {
"sample_rate": 16000,
"sample_rate": SAMPLE_RATE,
},
}
)
@ -1771,7 +1783,7 @@ async def test_audio_pipeline_with_enhancements(
"start_stage": "stt",
"end_stage": "tts",
"input": {
"sample_rate": 16000,
"sample_rate": SAMPLE_RATE,
# Enhancements
"noise_suppression_level": 2,
"auto_gain_dbfs": 15,
@ -1801,7 +1813,7 @@ async def test_audio_pipeline_with_enhancements(
# One second of silence.
# This will pass through the audio enhancement pipeline, but we don't test
# the actual output.
await client.send_bytes(bytes([handler_id]) + bytes(16000 * 2))
await client.send_bytes(bytes([handler_id]) + bytes(BYTES_ONE_SECOND))
# End of audio stream (handler id + empty payload)
await client.send_bytes(bytes([handler_id]))
@ -1871,7 +1883,7 @@ async def test_wake_word_cooldown_same_id(
"type": "assist_pipeline/run",
"start_stage": "wake_word",
"end_stage": "tts",
"input": {"sample_rate": 16000, "no_vad": True},
"input": {"sample_rate": SAMPLE_RATE, "no_vad": True},
}
)
@ -1880,7 +1892,7 @@ async def test_wake_word_cooldown_same_id(
"type": "assist_pipeline/run",
"start_stage": "wake_word",
"end_stage": "tts",
"input": {"sample_rate": 16000, "no_vad": True},
"input": {"sample_rate": SAMPLE_RATE, "no_vad": True},
}
)
@ -1914,8 +1926,8 @@ async def test_wake_word_cooldown_same_id(
assert msg["event"]["data"] == snapshot
# Wake both up at the same time
await client_1.send_bytes(bytes([handler_id_1]) + b"wake word")
await client_2.send_bytes(bytes([handler_id_2]) + b"wake word")
await client_1.send_bytes(bytes([handler_id_1]) + make_10ms_chunk(b"wake word"))
await client_2.send_bytes(bytes([handler_id_2]) + make_10ms_chunk(b"wake word"))
# Get response events
error_data: dict[str, Any] | None = None
@ -1954,7 +1966,7 @@ async def test_wake_word_cooldown_different_ids(
"type": "assist_pipeline/run",
"start_stage": "wake_word",
"end_stage": "tts",
"input": {"sample_rate": 16000, "no_vad": True},
"input": {"sample_rate": SAMPLE_RATE, "no_vad": True},
}
)
@ -1963,7 +1975,7 @@ async def test_wake_word_cooldown_different_ids(
"type": "assist_pipeline/run",
"start_stage": "wake_word",
"end_stage": "tts",
"input": {"sample_rate": 16000, "no_vad": True},
"input": {"sample_rate": SAMPLE_RATE, "no_vad": True},
}
)
@ -1997,8 +2009,8 @@ async def test_wake_word_cooldown_different_ids(
assert msg["event"]["data"] == snapshot
# Wake both up at the same time, but they will have different wake word ids
await client_1.send_bytes(bytes([handler_id_1]) + b"wake word")
await client_2.send_bytes(bytes([handler_id_2]) + b"wake word")
await client_1.send_bytes(bytes([handler_id_1]) + make_10ms_chunk(b"wake word"))
await client_2.send_bytes(bytes([handler_id_2]) + make_10ms_chunk(b"wake word"))
# Get response events
msg = await client_1.receive_json()
@ -2073,7 +2085,7 @@ async def test_wake_word_cooldown_different_entities(
"pipeline": pipeline_id_1,
"start_stage": "wake_word",
"end_stage": "tts",
"input": {"sample_rate": 16000, "no_vad": True},
"input": {"sample_rate": SAMPLE_RATE, "no_vad": True},
}
)
@ -2084,7 +2096,7 @@ async def test_wake_word_cooldown_different_entities(
"pipeline": pipeline_id_2,
"start_stage": "wake_word",
"end_stage": "tts",
"input": {"sample_rate": 16000, "no_vad": True},
"input": {"sample_rate": SAMPLE_RATE, "no_vad": True},
}
)
@ -2119,8 +2131,8 @@ async def test_wake_word_cooldown_different_entities(
# Wake both up at the same time.
# They will have the same wake word id, but different entities.
await client_1.send_bytes(bytes([handler_id_1]) + b"wake word")
await client_2.send_bytes(bytes([handler_id_2]) + b"wake word")
await client_1.send_bytes(bytes([handler_id_1]) + make_10ms_chunk(b"wake word"))
await client_2.send_bytes(bytes([handler_id_2]) + make_10ms_chunk(b"wake word"))
# Get response events
error_data: dict[str, Any] | None = None
@ -2158,7 +2170,11 @@ async def test_device_capture(
identifiers={("demo", "satellite-1234")},
)
audio_chunks = [b"chunk1", b"chunk2", b"chunk3"]
audio_chunks = [
make_10ms_chunk(b"chunk1"),
make_10ms_chunk(b"chunk2"),
make_10ms_chunk(b"chunk3"),
]
# Start capture
client_capture = await hass_ws_client(hass)
@ -2181,7 +2197,7 @@ async def test_device_capture(
"type": "assist_pipeline/run",
"start_stage": "stt",
"end_stage": "stt",
"input": {"sample_rate": 16000, "no_vad": True},
"input": {"sample_rate": SAMPLE_RATE, "no_vad": True},
"device_id": satellite_device.id,
}
)
@ -2232,9 +2248,9 @@ async def test_device_capture(
# Verify audio chunks
for i, audio_chunk in enumerate(audio_chunks):
assert events[i]["type"] == "audio"
assert events[i]["rate"] == 16000
assert events[i]["width"] == 2
assert events[i]["channels"] == 1
assert events[i]["rate"] == SAMPLE_RATE
assert events[i]["width"] == SAMPLE_WIDTH
assert events[i]["channels"] == SAMPLE_CHANNELS
# Audio is base64 encoded
assert events[i]["audio"] == base64.b64encode(audio_chunk).decode("ascii")
@ -2259,7 +2275,11 @@ async def test_device_capture_override(
identifiers={("demo", "satellite-1234")},
)
audio_chunks = [b"chunk1", b"chunk2", b"chunk3"]
audio_chunks = [
make_10ms_chunk(b"chunk1"),
make_10ms_chunk(b"chunk2"),
make_10ms_chunk(b"chunk3"),
]
# Start first capture
client_capture_1 = await hass_ws_client(hass)
@ -2282,7 +2302,7 @@ async def test_device_capture_override(
"type": "assist_pipeline/run",
"start_stage": "stt",
"end_stage": "stt",
"input": {"sample_rate": 16000, "no_vad": True},
"input": {"sample_rate": SAMPLE_RATE, "no_vad": True},
"device_id": satellite_device.id,
}
)
@ -2365,9 +2385,9 @@ async def test_device_capture_override(
# Verify all but first audio chunk
for i, audio_chunk in enumerate(audio_chunks[1:]):
assert events[i]["type"] == "audio"
assert events[i]["rate"] == 16000
assert events[i]["width"] == 2
assert events[i]["channels"] == 1
assert events[i]["rate"] == SAMPLE_RATE
assert events[i]["width"] == SAMPLE_WIDTH
assert events[i]["channels"] == SAMPLE_CHANNELS
# Audio is base64 encoded
assert events[i]["audio"] == base64.b64encode(audio_chunk).decode("ascii")
@ -2427,7 +2447,7 @@ async def test_device_capture_queue_full(
"type": "assist_pipeline/run",
"start_stage": "stt",
"end_stage": "stt",
"input": {"sample_rate": 16000, "no_vad": True},
"input": {"sample_rate": SAMPLE_RATE, "no_vad": True},
"device_id": satellite_device.id,
}
)
@ -2448,8 +2468,8 @@ async def test_device_capture_queue_full(
assert msg["event"]["type"] == "stt-start"
assert msg["event"]["data"] == snapshot
# Single sample will "overflow" the queue
await client_pipeline.send_bytes(bytes([handler_id, 0, 0]))
# Single chunk will "overflow" the queue
await client_pipeline.send_bytes(bytes([handler_id]) + bytes(BYTES_PER_CHUNK))
# End of audio stream
await client_pipeline.send_bytes(bytes([handler_id]))
@ -2557,7 +2577,7 @@ async def test_stt_cooldown_same_id(
"start_stage": "stt",
"end_stage": "tts",
"input": {
"sample_rate": 16000,
"sample_rate": SAMPLE_RATE,
"wake_word_phrase": "ok_nabu",
},
}
@ -2569,7 +2589,7 @@ async def test_stt_cooldown_same_id(
"start_stage": "stt",
"end_stage": "tts",
"input": {
"sample_rate": 16000,
"sample_rate": SAMPLE_RATE,
"wake_word_phrase": "ok_nabu",
},
}
@ -2628,7 +2648,7 @@ async def test_stt_cooldown_different_ids(
"start_stage": "stt",
"end_stage": "tts",
"input": {
"sample_rate": 16000,
"sample_rate": SAMPLE_RATE,
"wake_word_phrase": "ok_nabu",
},
}
@ -2640,7 +2660,7 @@ async def test_stt_cooldown_different_ids(
"start_stage": "stt",
"end_stage": "tts",
"input": {
"sample_rate": 16000,
"sample_rate": SAMPLE_RATE,
"wake_word_phrase": "hey_jarvis",
},
}