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Add go2rtc workaround for HA managed one until upstream fixes it (#130139)

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Robert Resch 2024-11-08 15:54:14 +01:00 • committed by GitHub
parent 6c7ac7a6ef
commit 51e691f832
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5 changed files with 270 additions and 37 deletions
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75
homeassistant/components/go2rtc/__init__.py
View file

@ -1,5 +1,8 @@
"""The go2rtc component."""
from __future__ import annotations
from dataclasses import dataclass
import logging
import shutil
@ -38,7 +41,13 @@ from homeassistant.helpers.typing import ConfigType
from homeassistant.util.hass_dict import HassKey
from homeassistant.util.package import is_docker_env
from .const import CONF_DEBUG_UI, DEBUG_UI_URL_MESSAGE, DOMAIN, HA_MANAGED_URL
from .const import (
CONF_DEBUG_UI,
DEBUG_UI_URL_MESSAGE,
DOMAIN,
HA_MANAGED_RTSP_PORT,
HA_MANAGED_URL,
)
from .server import Server
_LOGGER = logging.getLogger(__name__)
@ -85,13 +94,22 @@ CONFIG_SCHEMA = vol.Schema(
extra=vol.ALLOW_EXTRA,
)
_DATA_GO2RTC: HassKey[str] = HassKey(DOMAIN)
_DATA_GO2RTC: HassKey[Go2RtcData] = HassKey(DOMAIN)
_RETRYABLE_ERRORS = (ClientConnectionError, ServerConnectionError)
@dataclass(frozen=True)
class Go2RtcData:
"""Data for go2rtc."""
url: str
managed: bool
async def async_setup(hass: HomeAssistant, config: ConfigType) -> bool:
"""Set up WebRTC."""
url: str | None = None
managed = False
if DOMAIN not in config and DEFAULT_CONFIG_DOMAIN not in config:
await _remove_go2rtc_entries(hass)
return True
@ -126,8 +144,9 @@ async def async_setup(hass: HomeAssistant, config: ConfigType) -> bool:
hass.bus.async_listen(EVENT_HOMEASSISTANT_STOP, on_stop)
url = HA_MANAGED_URL
managed = True
hass.data[_DATA_GO2RTC] = url
hass.data[_DATA_GO2RTC] = Go2RtcData(url, managed)
discovery_flow.async_create_flow(
hass, DOMAIN, context={"source": SOURCE_SYSTEM}, data={}
)
@ -142,28 +161,32 @@ async def _remove_go2rtc_entries(hass: HomeAssistant) -> None:
async def async_setup_entry(hass: HomeAssistant, entry: ConfigEntry) -> bool:
"""Set up go2rtc from a config entry."""
url = hass.data[_DATA_GO2RTC]
data = hass.data[_DATA_GO2RTC]
# Validate the server URL
try:
client = Go2RtcRestClient(async_get_clientsession(hass), url)
client = Go2RtcRestClient(async_get_clientsession(hass), data.url)
await client.validate_server_version()
except Go2RtcClientError as err:
if isinstance(err.__cause__, _RETRYABLE_ERRORS):
raise ConfigEntryNotReady(
f"Could not connect to go2rtc instance on {url}"
f"Could not connect to go2rtc instance on {data.url}"
) from err
_LOGGER.warning("Could not connect to go2rtc instance on %s (%s)", url, err)
_LOGGER.warning(
"Could not connect to go2rtc instance on %s (%s)", data.url, err
)
return False
except Go2RtcVersionError as err:
raise ConfigEntryNotReady(
f"The go2rtc server version is not supported, {err}"
) from err
except Exception as err: # noqa: BLE001
_LOGGER.warning("Could not connect to go2rtc instance on %s (%s)", url, err)
_LOGGER.warning(
"Could not connect to go2rtc instance on %s (%s)", data.url, err
)
return False
provider = WebRTCProvider(hass, url)
provider = WebRTCProvider(hass, data)
async_register_webrtc_provider(hass, provider)
return True
@ -181,12 +204,12 @@ async def _get_binary(hass: HomeAssistant) -> str | None:
class WebRTCProvider(CameraWebRTCProvider):
"""WebRTC provider."""
def __init__(self, hass: HomeAssistant, url: str) -> None:
def __init__(self, hass: HomeAssistant, data: Go2RtcData) -> None:
"""Initialize the WebRTC provider."""
self._hass = hass
self._url = url
self._data = data
self._session = async_get_clientsession(hass)
self._rest_client = Go2RtcRestClient(self._session, url)
self._rest_client = Go2RtcRestClient(self._session, data.url)
self._sessions: dict[str, Go2RtcWsClient] = {}
@property
@ -208,7 +231,7 @@ class WebRTCProvider(CameraWebRTCProvider):
) -> None:
"""Handle the WebRTC offer and return the answer via the provided callback."""
self._sessions[session_id] = ws_client = Go2RtcWsClient(
self._session, self._url, source=camera.entity_id
self._session, self._data.url, source=camera.entity_id
)
if not (stream_source := await camera.stream_source()):
@ -219,8 +242,30 @@ class WebRTCProvider(CameraWebRTCProvider):
streams = await self._rest_client.streams.list()
if (stream := streams.get(camera.entity_id)) is None or not any(
stream_source == producer.url for producer in stream.producers
if self._data.managed:
# HA manages the go2rtc instance
stream_org_name = camera.entity_id + "_orginal"
stream_redirect_sources = [
f"rtsp://127.0.0.1:{HA_MANAGED_RTSP_PORT}/{stream_org_name}",
f"ffmpeg:{stream_org_name}#audio=opus",
]
if (
(stream_org := streams.get(stream_org_name)) is None
or not any(
stream_source == producer.url for producer in stream_org.producers
)
or (stream_redirect := streams.get(camera.entity_id)) is None
or stream_redirect_sources != [p.url for p in stream_redirect.producers]
):
await self._rest_client.streams.add(stream_org_name, stream_source)
await self._rest_client.streams.add(
camera.entity_id, stream_redirect_sources
)
# go2rtc instance is managed outside HA
elif (stream_org := streams.get(camera.entity_id)) is None or not any(
stream_source == producer.url for producer in stream_org.producers
):
await self._rest_client.streams.add(
camera.entity_id,

1
homeassistant/components/go2rtc/const.py
View file

@ -6,3 +6,4 @@ CONF_DEBUG_UI = "debug_ui"
DEBUG_UI_URL_MESSAGE = "Url and debug_ui cannot be set at the same time."
HA_MANAGED_API_PORT = 11984
HA_MANAGED_URL = f"http://localhost:{HA_MANAGED_API_PORT}/"
HA_MANAGED_RTSP_PORT = 18554

15
homeassistant/components/go2rtc/server.py
View file

@ -12,7 +12,7 @@ from homeassistant.core import HomeAssistant
from homeassistant.exceptions import HomeAssistantError
from homeassistant.helpers.aiohttp_client import async_get_clientsession
from .const import HA_MANAGED_API_PORT, HA_MANAGED_URL
from .const import HA_MANAGED_API_PORT, HA_MANAGED_RTSP_PORT, HA_MANAGED_URL
_LOGGER = logging.getLogger(__name__)
_TERMINATE_TIMEOUT = 5
@ -24,15 +24,16 @@ _RESPAWN_COOLDOWN = 1
# Default configuration for HA
# - Api is listening only on localhost
# - Disable rtsp listener
# - Enable rtsp for localhost only as ffmpeg needs it
# - Clear default ice servers
_GO2RTC_CONFIG_FORMAT = r"""
_GO2RTC_CONFIG_FORMAT = r"""# This file is managed by Home Assistant
# Do not edit it manually
api:
listen: "{api_ip}:{api_port}"
rtsp:
# ffmpeg needs rtsp for opus audio transcoding
listen: "127.0.0.1:18554"
listen: "127.0.0.1:{rtsp_port}"
webrtc:
listen: ":18555/tcp"
@ -67,7 +68,9 @@ def _create_temp_file(api_ip: str) -> str:
with NamedTemporaryFile(prefix="go2rtc_", suffix=".yaml", delete=False) as file:
file.write(
_GO2RTC_CONFIG_FORMAT.format(
api_ip=api_ip, api_port=HA_MANAGED_API_PORT
api_ip=api_ip,
api_port=HA_MANAGED_API_PORT,
rtsp_port=HA_MANAGED_RTSP_PORT,
).encode()
)
return file.name

211
tests/components/go2rtc/test_init.py
View file

@ -3,7 +3,7 @@
from collections.abc import Callable, Generator
import logging
from typing import NamedTuple
from unittest.mock import AsyncMock, Mock, patch
from unittest.mock import AsyncMock, Mock, call, patch
from aiohttp.client_exceptions import ClientConnectionError, ServerConnectionError
from go2rtc_client import Stream
@ -296,7 +296,7 @@ async def _test_setup_and_signaling(
],
)
@pytest.mark.parametrize("has_go2rtc_entry", [True, False])
async def test_setup_go_binary(
async def test_setup_managed(
hass: HomeAssistant,
rest_client: AsyncMock,
ws_client: Mock,
@ -308,15 +308,131 @@ async def test_setup_go_binary(
config: ConfigType,
ui_enabled: bool,
) -> None:
"""Test the go2rtc config entry with binary."""
"""Test the go2rtc setup with managed go2rtc instance."""
assert (len(hass.config_entries.async_entries(DOMAIN)) == 1) == has_go2rtc_entry
camera = init_test_integration
def after_setup() -> None:
server.assert_called_once_with(hass, "/usr/bin/go2rtc", enable_ui=ui_enabled)
server_start.assert_called_once()
entity_id = camera.entity_id
stream_name_orginal = camera.entity_id + "_orginal"
assert camera.frontend_stream_type == StreamType.HLS
await _test_setup_and_signaling(
hass, rest_client, ws_client, config, after_setup, init_test_integration
assert await async_setup_component(hass, DOMAIN, config)
await hass.async_block_till_done(wait_background_tasks=True)
config_entries = hass.config_entries.async_entries(DOMAIN)
assert len(config_entries) == 1
assert config_entries[0].state == ConfigEntryState.LOADED
server.assert_called_once_with(hass, "/usr/bin/go2rtc", enable_ui=ui_enabled)
server_start.assert_called_once()
receive_message_callback = Mock(spec_set=WebRTCSendMessage)
async def test() -> None:
await camera.async_handle_async_webrtc_offer(
OFFER_SDP, "session_id", receive_message_callback
)
ws_client.send.assert_called_once_with(
WebRTCOffer(
OFFER_SDP,
camera.async_get_webrtc_client_configuration().configuration.ice_servers,
)
)
ws_client.subscribe.assert_called_once()
# Simulate the answer from the go2rtc server
callback = ws_client.subscribe.call_args[0][0]
callback(WebRTCAnswer(ANSWER_SDP))
receive_message_callback.assert_called_once_with(HAWebRTCAnswer(ANSWER_SDP))
await test()
stream_added_calls = [
call(stream_name_orginal, "rtsp://stream"),
call(
entity_id,
[
f"rtsp://127.0.0.1:18554/{stream_name_orginal}",
f"ffmpeg:{stream_name_orginal}#audio=opus",
],
),
]
assert rest_client.streams.add.call_args_list == stream_added_calls
# Stream original missing
rest_client.streams.add.reset_mock()
rest_client.streams.list.return_value = {
entity_id: Stream(
[
Producer(f"rtsp://127.0.0.1:18554/{stream_name_orginal}"),
Producer(f"ffmpeg:{stream_name_orginal}#audio=opus"),
]
)
}
receive_message_callback.reset_mock()
ws_client.reset_mock()
await test()
assert rest_client.streams.add.call_args_list == stream_added_calls
# Stream original source different
rest_client.streams.add.reset_mock()
rest_client.streams.list.return_value = {
stream_name_orginal: Stream([Producer("rtsp://different")]),
entity_id: Stream(
[
Producer(f"rtsp://127.0.0.1:18554/{stream_name_orginal}"),
Producer(f"ffmpeg:{stream_name_orginal}#audio=opus"),
]
),
}
receive_message_callback.reset_mock()
ws_client.reset_mock()
await test()
assert rest_client.streams.add.call_args_list == stream_added_calls
# Stream source different
rest_client.streams.add.reset_mock()
rest_client.streams.list.return_value = {
stream_name_orginal: Stream([Producer("rtsp://stream")]),
entity_id: Stream([Producer("rtsp://different")]),
}
receive_message_callback.reset_mock()
ws_client.reset_mock()
await test()
assert rest_client.streams.add.call_args_list == stream_added_calls
# If the stream is already added, the stream should not be added again.
rest_client.streams.add.reset_mock()
rest_client.streams.list.return_value = {
stream_name_orginal: Stream([Producer("rtsp://stream")]),
entity_id: Stream(
[
Producer(f"rtsp://127.0.0.1:18554/{stream_name_orginal}"),
Producer(f"ffmpeg:{stream_name_orginal}#audio=opus"),
]
),
}
receive_message_callback.reset_mock()
ws_client.reset_mock()
await test()
rest_client.streams.add.assert_not_called()
assert isinstance(camera._webrtc_provider, WebRTCProvider)
# Set stream source to None and provider should be skipped
rest_client.streams.list.return_value = {}
receive_message_callback.reset_mock()
camera.set_stream_source(None)
await camera.async_handle_async_webrtc_offer(
OFFER_SDP, "session_id", receive_message_callback
)
receive_message_callback.assert_called_once_with(
WebRTCError("go2rtc_webrtc_offer_failed", "Camera has no stream source")
)
await hass.async_stop()
@ -332,7 +448,7 @@ async def test_setup_go_binary(
],
)
@pytest.mark.parametrize("has_go2rtc_entry", [True, False])
async def test_setup_go(
async def test_setup_self_hosted(
hass: HomeAssistant,
rest_client: AsyncMock,
ws_client: Mock,
@ -342,16 +458,83 @@ async def test_setup_go(
mock_is_docker_env: Mock,
has_go2rtc_entry: bool,
) -> None:
"""Test the go2rtc config entry without binary."""
"""Test the go2rtc with selfhosted go2rtc instance."""
assert (len(hass.config_entries.async_entries(DOMAIN)) == 1) == has_go2rtc_entry
config = {DOMAIN: {CONF_URL: "http://localhost:1984/"}}
camera = init_test_integration
def after_setup() -> None:
server.assert_not_called()
entity_id = camera.entity_id
assert camera.frontend_stream_type == StreamType.HLS
await _test_setup_and_signaling(
hass, rest_client, ws_client, config, after_setup, init_test_integration
assert await async_setup_component(hass, DOMAIN, config)
await hass.async_block_till_done(wait_background_tasks=True)
config_entries = hass.config_entries.async_entries(DOMAIN)
assert len(config_entries) == 1
assert config_entries[0].state == ConfigEntryState.LOADED
server.assert_not_called()
receive_message_callback = Mock(spec_set=WebRTCSendMessage)
async def test() -> None:
await camera.async_handle_async_webrtc_offer(
OFFER_SDP, "session_id", receive_message_callback
)
ws_client.send.assert_called_once_with(
WebRTCOffer(
OFFER_SDP,
camera.async_get_webrtc_client_configuration().configuration.ice_servers,
)
)
ws_client.subscribe.assert_called_once()
# Simulate the answer from the go2rtc server
callback = ws_client.subscribe.call_args[0][0]
callback(WebRTCAnswer(ANSWER_SDP))
receive_message_callback.assert_called_once_with(HAWebRTCAnswer(ANSWER_SDP))
await test()
rest_client.streams.add.assert_called_once_with(
entity_id, ["rtsp://stream", f"ffmpeg:{camera.entity_id}#audio=opus"]
)
# Stream exists but the source is different
rest_client.streams.add.reset_mock()
rest_client.streams.list.return_value = {
entity_id: Stream([Producer("rtsp://different")])
}
receive_message_callback.reset_mock()
ws_client.reset_mock()
await test()
rest_client.streams.add.assert_called_once_with(
entity_id, ["rtsp://stream", f"ffmpeg:{camera.entity_id}#audio=opus"]
)
# If the stream is already added, the stream should not be added again.
rest_client.streams.add.reset_mock()
rest_client.streams.list.return_value = {
entity_id: Stream([Producer("rtsp://stream")])
}
receive_message_callback.reset_mock()
ws_client.reset_mock()
await test()
rest_client.streams.add.assert_not_called()
assert isinstance(camera._webrtc_provider, WebRTCProvider)
# Set stream source to None and provider should be skipped
rest_client.streams.list.return_value = {}
receive_message_callback.reset_mock()
camera.set_stream_source(None)
await camera.async_handle_async_webrtc_offer(
OFFER_SDP, "session_id", receive_message_callback
)
receive_message_callback.assert_called_once_with(
WebRTCError("go2rtc_webrtc_offer_failed", "Camera has no stream source")
)
mock_get_binary.assert_not_called()

5
tests/components/go2rtc/test_server.py
View file

@ -105,12 +105,13 @@ async def test_server_run_success(
# Verify that the config file was written
mock_tempfile.write.assert_called_once_with(
f"""
f"""# This file is managed by Home Assistant
# Do not edit it manually
api:
listen: "{api_ip}:11984"
rtsp:
# ffmpeg needs rtsp for opus audio transcoding
listen: "127.0.0.1:18554"
webrtc: