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annotate dictation/webrtc_test.py @ 279:b3b547563ec7

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Add Google connector service and agent wiki Implement the C/Seobeo Google Drive and Gmail connector with encrypted OAuth storage, Zenbu authentication, browser testing, AI tool discovery, chunked HTTP decoding, and Bazel coverage. Consolidate repository guidance into progressive wiki documentation and enforce arena-first allocation for new first-party C code. Co-authored-by: Copilot <[email protected]> Copilot-Session: 84c338fd-0939-4bb3-b7f3-1062eb213e5d
author MrJuneJune <me@mrjunejune.com>
date Mon, 17 Aug 2026 22:22:36 -0700
parents 78699f810817
children 49e9e591c9bb
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1 import asyncio
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2 import json
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3 import math
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4 from pathlib import Path
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5 import tempfile
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6 import unittest
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7 import wave
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8
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9 from aiortc import RTCPeerConnection, RTCSessionDescription
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10 from aiortc.contrib.media import MediaPlayer
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11 import numpy as np
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12
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13 from dictation.config import DictationConfig
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14 from dictation.server import DictationService, Offer
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15 from dictation.transcriber import Transcript
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16
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17
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18 class FakeTranscriber:
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19 async def warmup(self):
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20 return None
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21
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22 async def transcribe(self, samples, *, final):
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23 return Transcript(
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24 "final transcript" if final else "partial transcript",
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25 "en",
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26 0.99,
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27 )
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29
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30 def write_test_audio(path: str) -> None:
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31 sample_rate = 16000
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32 speech = np.array(
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33 [
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34 int(1600 * math.sin(2 * math.pi * 440 * index / sample_rate))
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35 for index in range(sample_rate * 2)
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36 ],
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37 dtype=np.int16,
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38 )
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39 silence = np.zeros(sample_rate, dtype=np.int16)
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40 with wave.open(path, "wb") as output:
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41 output.setnchannels(1)
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42 output.setsampwidth(2)
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43 output.setframerate(sample_rate)
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44 output.writeframes(np.concatenate((speech, silence)).tobytes())
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45
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46
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47 def test_config() -> DictationConfig:
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48 return DictationConfig(
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49 host="127.0.0.1",
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50 port=8090,
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51 model_dir=Path("/tmp/model"),
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52 compute_type="int8_float16",
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53 max_sessions=1,
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54 partial_interval_ms=500,
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55 silence_ms=200,
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56 max_utterance_seconds=5,
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57 speech_threshold=0.01,
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58 )
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59
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60
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61 class WebRtcTest(unittest.IsolatedAsyncioTestCase):
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62 async def test_audio_track_returns_transcript_events(self):
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63 service = DictationService(test_config(), FakeTranscriber())
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64 await service.start()
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65 client = RTCPeerConnection()
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66 channel = client.createDataChannel("transcripts")
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67 messages = []
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68 final_received = asyncio.Event()
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69 audio_file = tempfile.NamedTemporaryFile(suffix=".wav", delete=False)
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70 audio_file.close()
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71 write_test_audio(audio_file.name)
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72 player = MediaPlayer(audio_file.name)
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73
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74 @channel.on("message")
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75 def on_message(message):
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76 event = json.loads(message)
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77 messages.append(event)
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78 if event["type"] == "transcript.final":
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79 final_received.set()
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80
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81 client.addTrack(player.audio)
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82 offer = await client.createOffer()
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83 await client.setLocalDescription(offer)
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84 answer = await service.accept_offer(
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85 Offer(
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86 sdp=client.localDescription.sdp,
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87 type=client.localDescription.type,
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88 )
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89 )
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90 await client.setRemoteDescription(
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91 RTCSessionDescription(
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92 sdp=answer["sdp"],
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93 type=answer["type"],
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94 )
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95 )
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96
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97 try:
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98 try:
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99 await asyncio.wait_for(final_received.wait(), timeout=10)
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100 except TimeoutError as error:
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101 server_states = [
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102 {
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103 "connection": session.peer.connectionState,
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104 "ice": session.peer.iceConnectionState,
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105 "channel": (
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106 session.channel.readyState
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107 if session.channel is not None
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108 else None
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109 ),
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110 "audioTask": (
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111 "missing"
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112 if session.audio_task is None
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113 else (
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114 repr(session.audio_task.exception())
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115 if session.audio_task.done()
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116 and not session.audio_task.cancelled()
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117 else "running"
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118 )
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119 ),
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120 }
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121 for session in service.sessions.values()
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122 ]
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123 self.fail(
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124 "Timed out waiting for transcript: "
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125 f"client={client.connectionState}/"
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126 f"{client.iceConnectionState}, "
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127 f"server={server_states}, messages={messages}"
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128 )
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129 event_types = [message["type"] for message in messages]
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130 self.assertIn("ready", event_types)
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131 self.assertIn("speech.started", event_types)
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132 self.assertIn("transcript.partial", event_types)
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133 self.assertIn("transcript.final", event_types)
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134 finally:
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135 await client.close()
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136 await service.close()
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137 Path(audio_file.name).unlink(missing_ok=True)
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138 self.assertEqual(service.sessions, {})
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139
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140
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141 if __name__ == "__main__":
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142 unittest.main()