[Scribe] Add audio format correctly and better typing

src/elevenlabs/realtime/connection.py

@@ -4,6 +4,9 @@
 import typing
 from enum import Enum
 
+if typing.TYPE_CHECKING:
+    from websockets.asyncio.client import ClientConnection
+
 
 class RealtimeEvents(str, Enum):
     """Events emitted by the RealtimeConnection"""
@@ -55,7 +58,7 @@ class RealtimeConnection:
         ```
     """
 
-    def __init__(self, websocket, current_sample_rate: int, ffmpeg_process: typing.Optional[subprocess.Popen] = None):
+    def __init__(self, websocket: "ClientConnection", current_sample_rate: int, ffmpeg_process: typing.Optional[subprocess.Popen] = None):
         self.websocket = websocket
         self.current_sample_rate = current_sample_rate
         self.ffmpeg_process = ffmpeg_process

src/elevenlabs/realtime/scribe.py

@@ -3,9 +3,12 @@
 import subprocess
 import typing
 from enum import Enum
+from typing import overload
+
+from typing_extensions import Required
 
 try:
-    import websockets
+    from websockets.asyncio.client import connect as websocket_connect
 except ImportError:
     raise ImportError(
         "The websockets package is required for realtime speech-to-text. "
@@ -17,10 +20,13 @@
 
 class AudioFormat(str, Enum):
     """Audio format options for realtime transcription"""
+    PCM_8000 = "pcm_8000"
     PCM_16000 = "pcm_16000"
     PCM_22050 = "pcm_22050"
     PCM_24000 = "pcm_24000"
     PCM_44100 = "pcm_44100"
+    PCM_48000 = "pcm_48000"
+    ULAW_8000 = "ulaw_8000"
 
 
 class CommitStrategy(str, Enum):
@@ -50,9 +56,9 @@ class RealtimeAudioOptions(typing.TypedDict, total=False):
         language_code: An ISO-639-1 or ISO-639-3 language_code corresponding to the language of the audio file. Can sometimes improve transcription performance if known beforehand.
         include_timestamps: Whether to receive the committed_transcript_with_timestamps event after committing the segment (optional, defaults to False)
     """
-    model_id: str
-    audio_format: AudioFormat
-    sample_rate: int
+    model_id: Required[str]
+    audio_format: Required[AudioFormat]
+    sample_rate: Required[int]
     commit_strategy: CommitStrategy
     vad_silence_threshold_secs: float
     vad_threshold: float
@@ -77,8 +83,8 @@ class RealtimeUrlOptions(typing.TypedDict, total=False):
         language_code: An ISO-639-1 or ISO-639-3 language_code corresponding to the language of the audio file. Can sometimes improve transcription performance if known beforehand.
         include_timestamps: Whether to receive the committed_transcript_with_timestamps event after committing the segment (optional, defaults to False)
     """
-    model_id: str
-    url: str
+    model_id: Required[str]
+    url: Required[str]
     commit_strategy: CommitStrategy
     vad_silence_threshold_secs: float
     vad_threshold: float
@@ -121,6 +127,18 @@ def __init__(self, api_key: str, base_url: str = "wss://api.elevenlabs.io"):
         self.api_key = api_key
         self.base_url = base_url
 
+    @overload
+    async def connect(
+        self,
+        options: RealtimeAudioOptions
+    ) -> RealtimeConnection: ...
+
+    @overload
+    async def connect(
+        self,
+        options: RealtimeUrlOptions
+    ) -> RealtimeConnection: ...
+
     async def connect(
         self,
         options: typing.Union[RealtimeAudioOptions, RealtimeUrlOptions]
@@ -185,8 +203,7 @@ async def _connect_audio(self, options: RealtimeAudioOptions) -> RealtimeConnect
         # Build WebSocket URL with query parameters
         ws_url = self._build_websocket_url(
             model_id=model_id,
-            encoding=audio_format.value,
-            sample_rate=sample_rate,
+            audio_format=audio_format.value,
             commit_strategy=commit_strategy.value,
             vad_silence_threshold_secs=vad_silence_threshold_secs,
             vad_threshold=vad_threshold,
@@ -197,7 +214,7 @@ async def _connect_audio(self, options: RealtimeAudioOptions) -> RealtimeConnect
         )
 
         # Connect to WebSocket
-        websocket = await websockets.connect(
+        websocket = await websocket_connect(
             ws_url,
             additional_headers={"xi-api-key": self.api_key}
         )
@@ -232,13 +249,12 @@ async def _connect_url(self, options: RealtimeUrlOptions) -> RealtimeConnection:
 
         # Default to 16kHz for URL streaming
         sample_rate = 16000
-        encoding = "pcm_16000"
+        audio_format = AudioFormat.PCM_16000
 
         # Build WebSocket URL
         ws_url = self._build_websocket_url(
             model_id=model_id,
-            encoding=encoding,
-            sample_rate=sample_rate,
+            audio_format=audio_format,
             commit_strategy=commit_strategy.value,
             vad_silence_threshold_secs=vad_silence_threshold_secs,
             vad_threshold=vad_threshold,
@@ -249,7 +265,7 @@ async def _connect_url(self, options: RealtimeUrlOptions) -> RealtimeConnection:
         )
 
         # Connect to WebSocket
-        websocket = await websockets.connect(
+        websocket = await websocket_connect(
             ws_url,
             additional_headers={"xi-api-key": self.api_key}
         )
@@ -341,8 +357,7 @@ async def _stream_ffmpeg_to_websocket(self, connection: RealtimeConnection) -> N
     def _build_websocket_url(
         self,
         model_id: str,
-        encoding: str,
-        sample_rate: int,
+        audio_format: str,
         commit_strategy: str,
         vad_silence_threshold_secs: typing.Optional[float] = None,
         vad_threshold: typing.Optional[float] = None,
@@ -358,8 +373,7 @@ def _build_websocket_url(
         # Build query parameters
         params = [
             f"model_id={model_id}",
-            f"encoding={encoding}",
-            f"sample_rate={sample_rate}",
+            f"audio_format={audio_format}",
             f"commit_strategy={commit_strategy}"
         ]