cartesia 2.0.11__tar.gz → 2.0.14__tar.gz

{cartesia-2.0.11 → cartesia-2.0.14}/PKG-INFO

@@ -1,6 +1,6 @@
 Metadata-Version: 2.1
 Name: cartesia
-Version: 2.0.11
+Version: 2.0.14
 Summary: 
 Requires-Python: >=3.8,<4.0
 Classifier: Intended Audience :: Developers
@@ -53,26 +53,36 @@ Instantiate and use the client with the following:
 
 ```python
 from cartesia import Cartesia
-from cartesia.tts import OutputFormat_Raw, TtsRequestIdSpecifier
 import os
 
 client = Cartesia(
-    api_key=os.getenv("CARTESIA_API_KEY"),
-)
-client.tts.bytes(
-    model_id="sonic-2",
-    transcript="Hello, world!",
-    voice={
-        "mode": "id",
-        "id": "694f9389-aac1-45b6-b726-9d9369183238",
-    },
-    language="en",
-    output_format={
-        "container": "raw",
-        "sample_rate": 44100,
-        "encoding": "pcm_f32le",
-    },
+    api_key=os.environ["CARTESIA_API_KEY"],
 )
+
+
+def main():
+    with open("sonic.wav", "wb") as f:
+        bytes_iter = client.tts.bytes(
+            model_id="sonic-3",
+            transcript="Hello, world!",
+            voice={
+                "mode": "id",
+                "id": "6ccbfb76-1fc6-48f7-b71d-91ac6298247b",
+            },
+            language="en",
+            output_format={
+                "container": "wav",
+                "sample_rate": 44100,
+                "encoding": "pcm_f32le",
+            },
+        )
+
+        for chunk in bytes_iter:
+            f.write(chunk)
+
+
+if __name__ == "__main__":
+    main()
 ```
 
 ## Async Client
@@ -81,31 +91,37 @@ The SDK also exports an `async` client so that you can make non-blocking calls t
 
 ```python
 import asyncio
-import os
-
 from cartesia import AsyncCartesia
-from cartesia.tts import OutputFormat_Raw, TtsRequestIdSpecifier
+import os
 
 client = AsyncCartesia(
-    api_key=os.getenv("CARTESIA_API_KEY"),
+    api_key=os.environ["CARTESIA_API_KEY"],
 )
 
-async def main() -> None:
-    async for output in client.tts.bytes(
-        model_id="sonic-2",
-        transcript="Hello, world!",
-        voice={"id": "694f9389-aac1-45b6-b726-9d9369183238"},
-        language="en",
-        output_format={
-            "container": "raw",
-            "sample_rate": 44100,
-            "encoding": "pcm_f32le",
-        },
-    ):
-        print(f"Received chunk of size: {len(output)}")
 
+async def main():
+    with open("sonic.wav", "wb") as f:
+        bytes_iter = client.tts.bytes(
+            model_id="sonic-3",
+            transcript="Hello, world!",
+            voice={
+                "mode": "id",
+                "id": "6ccbfb76-1fc6-48f7-b71d-91ac6298247b",
+            },
+            language="en",
+            output_format={
+                "container": "wav",
+                "sample_rate": 44100,
+                "encoding": "pcm_f32le",
+            },
+        )
+
+        async for chunk in bytes_iter:
+            f.write(chunk)
 
-asyncio.run(main())
+
+if __name__ == "__main__":
+    asyncio.run(main())
 ```
 
 ## Exception Handling
@@ -129,7 +145,6 @@ The SDK supports streaming responses as well, returning a generator that you can
 
 ```python
 from cartesia import Cartesia
-from cartesia.tts import Controls, OutputFormat_RawParams, TtsRequestIdSpecifierParams
 import os
 
 def get_tts_chunks():
@@ -137,14 +152,11 @@ def get_tts_chunks():
         api_key=os.getenv("CARTESIA_API_KEY"),
     )
     response = client.tts.sse(
-        model_id="sonic-2",
+        model_id="sonic-3",
         transcript="Hello world!",
         voice={
+            "mode": "id",
             "id": "f9836c6e-a0bd-460e-9d3c-f7299fa60f94",
-            "experimental_controls": {
-                "speed": "normal",
-                "emotion": [],
-            },
         },
         language="en",
         output_format={
@@ -188,9 +200,9 @@ ws = client.tts.websocket()
 
 # Generate and stream audio using the websocket
 for output in ws.send(
-    model_id="sonic-2", # see: https://docs.cartesia.ai/getting-started/available-models
+    model_id="sonic-3", # see: https://docs.cartesia.ai/build-with-cartesia/tts-models
     transcript=transcript,
-    voice={"id": voice_id},
+    voice={"mode": "id", "id": voice_id},
     stream=True,
     output_format={
         "container": "raw",
@@ -252,7 +264,7 @@ ws.send("done")
 for result in ws.receive():
     if result['type'] == 'transcript':
         print(f"Transcription: {result['text']}")
-        
+
         # Handle word-level timestamps if available
         if 'words' in result and result['words']:
             print("Word-level timestamps:")
@@ -261,7 +273,7 @@ for result in ws.receive():
                 start = word_info['start']
                 end = word_info['end']
                 print(f"  '{word}': {start:.2f}s - {end:.2f}s")
-        
+
         if result['is_final']:
             print("Final result received")
     elif result['type'] == 'done':
@@ -286,7 +298,7 @@ async def streaming_stt_example():
     and demonstrates the new endpointing and word timestamp features.
     """
     client = AsyncCartesia(api_key=os.getenv("CARTESIA_API_KEY"))
-    
+
     try:
         # Create websocket connection with voice activity detection
         ws = await client.stt.websocket(
@@ -297,24 +309,24 @@ async def streaming_stt_example():
             min_volume=0.15,                 # Volume threshold for voice activity detection
             max_silence_duration_secs=0.3,   # Maximum silence duration before endpointing
         )
-        
+
         # Simulate streaming audio data (replace with your audio source)
         async def audio_stream():
             """Simulate real-time audio streaming - replace with actual audio capture"""
             # Load audio file for simulation
             with open("path/to/audio.wav", "rb") as f:
                 audio_data = f.read()
-            
+
             # Stream in 100ms chunks (realistic for real-time processing)
             chunk_size = int(16000 * 0.1 * 2)  # 100ms at 16kHz, 16-bit
-            
+
             for i in range(0, len(audio_data), chunk_size):
                 chunk = audio_data[i:i + chunk_size]
                 if chunk:
                     yield chunk
                     # Simulate real-time streaming delay
                     await asyncio.sleep(0.1)
-        
+
         # Send audio and receive results concurrently
         async def send_audio():
             """Send audio chunks to the STT websocket"""
@@ -324,31 +336,31 @@ async def streaming_stt_example():
                     print(f"Sent audio chunk of {len(chunk)} bytes")
                     # Small delay to simulate realtime applications
                     await asyncio.sleep(0.02)
-                
+
                 # Signal end of audio stream
                 await ws.send("finalize")
                 await ws.send("done")
                 print("Audio streaming completed")
-                
+
             except Exception as e:
                 print(f"Error sending audio: {e}")
-        
+
         async def receive_transcripts():
             """Receive and process transcription results with word timestamps"""
             full_transcript = ""
             all_word_timestamps = []
-            
+
             try:
                 async for result in ws.receive():
                     if result['type'] == 'transcript':
                         text = result['text']
                         is_final = result['is_final']
-                        
+
                         # Handle word-level timestamps
                         if 'words' in result and result['words']:
                             word_timestamps = result['words']
                             all_word_timestamps.extend(word_timestamps)
-                            
+
                             if is_final:
                                 print("Word-level timestamps:")
                                 for word_info in word_timestamps:
@@ -356,7 +368,7 @@ async def streaming_stt_example():
                                     start = word_info['start']
                                     end = word_info['end']
                                     print(f"  '{word}': {start:.2f}s - {end:.2f}s")
-                        
+
                         if is_final:
                             # Final result - this text won't change
                             full_transcript += text + " "
@@ -364,30 +376,30 @@ async def streaming_stt_example():
                         else:
                             # Partial result - may change as more audio is processed
                             print(f"PARTIAL: {text}")
-                            
+
                     elif result['type'] == 'done':
                         print("Transcription completed")
                         break
-                        
+
             except Exception as e:
                 print(f"Error receiving transcripts: {e}")
-            
+
             return full_transcript.strip(), all_word_timestamps
-        
+
         print("Starting streaming STT...")
-        
+
         # Use asyncio.gather to run audio sending and transcript receiving concurrently
         _, (final_transcript, word_timestamps) = await asyncio.gather(
             send_audio(),
             receive_transcripts()
         )
-        
+
         print(f"\nComplete transcript: {final_transcript}")
         print(f"Total words with timestamps: {len(word_timestamps)}")
-        
+
         # Clean up
         await ws.close()
-        
+
     except Exception as e:
         print(f"STT streaming error: {e}")
     finally:
@@ -442,7 +454,7 @@ import os
 
 async def transcribe_file():
     client = AsyncCartesia(api_key=os.getenv("CARTESIA_API_KEY"))
-    
+
     with open("path/to/audio.wav", "rb") as audio_file:
         response = await client.stt.transcribe(
             file=audio_file,
@@ -450,14 +462,14 @@ async def transcribe_file():
             language="en",
             timestamp_granularities=["word"],
         )
-    
+
     print(f"Transcribed text: {response.text}")
-    
+
     # Process word timestamps
     if response.words:
         for word_info in response.words:
             print(f"'{word_info.word}': {word_info.start:.2f}s - {word_info.end:.2f}s")
-    
+
     await client.close()
 
 asyncio.run(transcribe_file())
@@ -664,6 +676,3 @@ $ git commit --amend -m "manually regenerate from docs" # optional
 
 From https://github.com/cartesia-ai/docs click `Actions` then `Release Python SDK`. (Requires permissions.)
 
-
-
-

{cartesia-2.0.11 → cartesia-2.0.14}/README.md

@@ -21,26 +21,36 @@ Instantiate and use the client with the following:
 
 ```python
 from cartesia import Cartesia
-from cartesia.tts import OutputFormat_Raw, TtsRequestIdSpecifier
 import os
 
 client = Cartesia(
-    api_key=os.getenv("CARTESIA_API_KEY"),
-)
-client.tts.bytes(
-    model_id="sonic-2",
-    transcript="Hello, world!",
-    voice={
-        "mode": "id",
-        "id": "694f9389-aac1-45b6-b726-9d9369183238",
-    },
-    language="en",
-    output_format={
-        "container": "raw",
-        "sample_rate": 44100,
-        "encoding": "pcm_f32le",
-    },
+    api_key=os.environ["CARTESIA_API_KEY"],
 )
+
+
+def main():
+    with open("sonic.wav", "wb") as f:
+        bytes_iter = client.tts.bytes(
+            model_id="sonic-3",
+            transcript="Hello, world!",
+            voice={
+                "mode": "id",
+                "id": "6ccbfb76-1fc6-48f7-b71d-91ac6298247b",
+            },
+            language="en",
+            output_format={
+                "container": "wav",
+                "sample_rate": 44100,
+                "encoding": "pcm_f32le",
+            },
+        )
+
+        for chunk in bytes_iter:
+            f.write(chunk)
+
+
+if __name__ == "__main__":
+    main()
 ```
 
 ## Async Client
@@ -49,31 +59,37 @@ The SDK also exports an `async` client so that you can make non-blocking calls t
 
 ```python
 import asyncio
-import os
-
 from cartesia import AsyncCartesia
-from cartesia.tts import OutputFormat_Raw, TtsRequestIdSpecifier
+import os
 
 client = AsyncCartesia(
-    api_key=os.getenv("CARTESIA_API_KEY"),
+    api_key=os.environ["CARTESIA_API_KEY"],
 )
 
-async def main() -> None:
-    async for output in client.tts.bytes(
-        model_id="sonic-2",
-        transcript="Hello, world!",
-        voice={"id": "694f9389-aac1-45b6-b726-9d9369183238"},
-        language="en",
-        output_format={
-            "container": "raw",
-            "sample_rate": 44100,
-            "encoding": "pcm_f32le",
-        },
-    ):
-        print(f"Received chunk of size: {len(output)}")
 
+async def main():
+    with open("sonic.wav", "wb") as f:
+        bytes_iter = client.tts.bytes(
+            model_id="sonic-3",
+            transcript="Hello, world!",
+            voice={
+                "mode": "id",
+                "id": "6ccbfb76-1fc6-48f7-b71d-91ac6298247b",
+            },
+            language="en",
+            output_format={
+                "container": "wav",
+                "sample_rate": 44100,
+                "encoding": "pcm_f32le",
+            },
+        )
+
+        async for chunk in bytes_iter:
+            f.write(chunk)
 
-asyncio.run(main())
+
+if __name__ == "__main__":
+    asyncio.run(main())
 ```
 
 ## Exception Handling
@@ -97,7 +113,6 @@ The SDK supports streaming responses as well, returning a generator that you can
 
 ```python
 from cartesia import Cartesia
-from cartesia.tts import Controls, OutputFormat_RawParams, TtsRequestIdSpecifierParams
 import os
 
 def get_tts_chunks():
@@ -105,14 +120,11 @@ def get_tts_chunks():
         api_key=os.getenv("CARTESIA_API_KEY"),
     )
     response = client.tts.sse(
-        model_id="sonic-2",
+        model_id="sonic-3",
         transcript="Hello world!",
         voice={
+            "mode": "id",
             "id": "f9836c6e-a0bd-460e-9d3c-f7299fa60f94",
-            "experimental_controls": {
-                "speed": "normal",
-                "emotion": [],
-            },
         },
         language="en",
         output_format={
@@ -156,9 +168,9 @@ ws = client.tts.websocket()
 
 # Generate and stream audio using the websocket
 for output in ws.send(
-    model_id="sonic-2", # see: https://docs.cartesia.ai/getting-started/available-models
+    model_id="sonic-3", # see: https://docs.cartesia.ai/build-with-cartesia/tts-models
     transcript=transcript,
-    voice={"id": voice_id},
+    voice={"mode": "id", "id": voice_id},
     stream=True,
     output_format={
         "container": "raw",
@@ -220,7 +232,7 @@ ws.send("done")
 for result in ws.receive():
     if result['type'] == 'transcript':
         print(f"Transcription: {result['text']}")
-        
+
         # Handle word-level timestamps if available
         if 'words' in result and result['words']:
             print("Word-level timestamps:")
@@ -229,7 +241,7 @@ for result in ws.receive():
                 start = word_info['start']
                 end = word_info['end']
                 print(f"  '{word}': {start:.2f}s - {end:.2f}s")
-        
+
         if result['is_final']:
             print("Final result received")
     elif result['type'] == 'done':
@@ -254,7 +266,7 @@ async def streaming_stt_example():
     and demonstrates the new endpointing and word timestamp features.
     """
     client = AsyncCartesia(api_key=os.getenv("CARTESIA_API_KEY"))
-    
+
     try:
         # Create websocket connection with voice activity detection
         ws = await client.stt.websocket(
@@ -265,24 +277,24 @@ async def streaming_stt_example():
             min_volume=0.15,                 # Volume threshold for voice activity detection
             max_silence_duration_secs=0.3,   # Maximum silence duration before endpointing
         )
-        
+
         # Simulate streaming audio data (replace with your audio source)
         async def audio_stream():
             """Simulate real-time audio streaming - replace with actual audio capture"""
             # Load audio file for simulation
             with open("path/to/audio.wav", "rb") as f:
                 audio_data = f.read()
-            
+
             # Stream in 100ms chunks (realistic for real-time processing)
             chunk_size = int(16000 * 0.1 * 2)  # 100ms at 16kHz, 16-bit
-            
+
             for i in range(0, len(audio_data), chunk_size):
                 chunk = audio_data[i:i + chunk_size]
                 if chunk:
                     yield chunk
                     # Simulate real-time streaming delay
                     await asyncio.sleep(0.1)
-        
+
         # Send audio and receive results concurrently
         async def send_audio():
             """Send audio chunks to the STT websocket"""
@@ -292,31 +304,31 @@ async def streaming_stt_example():
                     print(f"Sent audio chunk of {len(chunk)} bytes")
                     # Small delay to simulate realtime applications
                     await asyncio.sleep(0.02)
-                
+
                 # Signal end of audio stream
                 await ws.send("finalize")
                 await ws.send("done")
                 print("Audio streaming completed")
-                
+
             except Exception as e:
                 print(f"Error sending audio: {e}")
-        
+
         async def receive_transcripts():
             """Receive and process transcription results with word timestamps"""
             full_transcript = ""
             all_word_timestamps = []
-            
+
             try:
                 async for result in ws.receive():
                     if result['type'] == 'transcript':
                         text = result['text']
                         is_final = result['is_final']
-                        
+
                         # Handle word-level timestamps
                         if 'words' in result and result['words']:
                             word_timestamps = result['words']
                             all_word_timestamps.extend(word_timestamps)
-                            
+
                             if is_final:
                                 print("Word-level timestamps:")
                                 for word_info in word_timestamps:
@@ -324,7 +336,7 @@ async def streaming_stt_example():
                                     start = word_info['start']
                                     end = word_info['end']
                                     print(f"  '{word}': {start:.2f}s - {end:.2f}s")
-                        
+
                         if is_final:
                             # Final result - this text won't change
                             full_transcript += text + " "
@@ -332,30 +344,30 @@ async def streaming_stt_example():
                         else:
                             # Partial result - may change as more audio is processed
                             print(f"PARTIAL: {text}")
-                            
+
                     elif result['type'] == 'done':
                         print("Transcription completed")
                         break
-                        
+
             except Exception as e:
                 print(f"Error receiving transcripts: {e}")
-            
+
             return full_transcript.strip(), all_word_timestamps
-        
+
         print("Starting streaming STT...")
-        
+
         # Use asyncio.gather to run audio sending and transcript receiving concurrently
         _, (final_transcript, word_timestamps) = await asyncio.gather(
             send_audio(),
             receive_transcripts()
         )
-        
+
         print(f"\nComplete transcript: {final_transcript}")
         print(f"Total words with timestamps: {len(word_timestamps)}")
-        
+
         # Clean up
         await ws.close()
-        
+
     except Exception as e:
         print(f"STT streaming error: {e}")
     finally:
@@ -410,7 +422,7 @@ import os
 
 async def transcribe_file():
     client = AsyncCartesia(api_key=os.getenv("CARTESIA_API_KEY"))
-    
+
     with open("path/to/audio.wav", "rb") as audio_file:
         response = await client.stt.transcribe(
             file=audio_file,
@@ -418,14 +430,14 @@ async def transcribe_file():
             language="en",
             timestamp_granularities=["word"],
         )
-    
+
     print(f"Transcribed text: {response.text}")
-    
+
     # Process word timestamps
     if response.words:
         for word_info in response.words:
             print(f"'{word_info.word}': {word_info.start:.2f}s - {word_info.end:.2f}s")
-    
+
     await client.close()
 
 asyncio.run(transcribe_file())
@@ -631,6 +643,3 @@ $ git commit --amend -m "manually regenerate from docs" # optional
 ### Automatically generating new SDK releases
 
 From https://github.com/cartesia-ai/docs click `Actions` then `Release Python SDK`. (Requires permissions.)
-
-
-

{cartesia-2.0.11 → cartesia-2.0.14}/pyproject.toml

@@ -3,7 +3,7 @@ name = "cartesia"
 
 [tool.poetry]
 name = "cartesia"
-version = "2.0.11"
+version = "2.0.14"
 description = ""
 readme = "README.md"
 authors = []

{cartesia-2.0.11 → cartesia-2.0.14}/src/cartesia/__init__.py

@@ -52,7 +52,10 @@ from .tts import (
     Controls,
     ControlsParams,
     Emotion,
+    EmotionDeprecated,
     FlushId,
+    GenerationConfig,
+    GenerationConfigParams,
     GenerationRequest,
     GenerationRequestParams,
     ModelSpeed,
@@ -211,6 +214,7 @@ __all__ = [
     "EmbeddingSpecifier",
     "EmbeddingSpecifierParams",
     "Emotion",
+    "EmotionDeprecated",
     "ErrorMessage",
     "ErrorMessageParams",
     "FilePurpose",
@@ -219,6 +223,8 @@ __all__ = [
     "FlushId",
     "Gender",
     "GenderPresentation",
+    "GenerationConfig",
+    "GenerationConfigParams",
     "GenerationRequest",
     "GenerationRequestParams",
     "GetVoicesResponse",

{cartesia-2.0.11 → cartesia-2.0.14}/src/cartesia/core/client_wrapper.py

@@ -16,7 +16,7 @@ class BaseClientWrapper:
         headers: typing.Dict[str, str] = {
             "X-Fern-Language": "Python",
             "X-Fern-SDK-Name": "cartesia",
-            "X-Fern-SDK-Version": "2.0.11",
+            "X-Fern-SDK-Version": "2.0.14",
         }
         headers["X-API-Key"] = self.api_key
         headers["Cartesia-Version"] = "2024-11-13"