> For clean Markdown of any page, append `.md` to the page URL. > For a complete documentation index, see https://docs.sarvam.ai/llms.txt. > For full documentation content in one file, see https://docs.sarvam.ai/llms-full.txt. > For AI client integration (Claude Code, Cursor, etc.), connect to the MCP server at https://docs.sarvam.ai/_mcp/server. # How to set `output_audio_bitrate` > Control the quality and size of the synthesized audio output. The `output_audio_bitrate` parameter defines the **bitrate** of the audio stream in kilobits per second. It controls the **audio quality**, **file size**, and **streaming performance**. > **Note** > > This parameter is available only in the **WebSocket Streaming API**. It is not supported in the REST API. This parameter is optional. If not provided, the system uses the **default value `128k`**. ### Supported Bitrates | Value | Description | | ------ | --------------------------------------------- | | `32k` | Very low bitrate, smaller file, lower quality | | `64k` | Low bitrate, suitable for speech | | `96k` | Medium quality | | `128k` | Default – good balance of quality and size | | `192k` | High quality, larger audio files | --- ### **Example Streaming API code** ```python import asyncio import base64 from sarvamai import AsyncSarvamAI, AudioOutput import websockets async def tts_stream(): client = AsyncSarvamAI(api_subscription_key="YOUR_SARVAM_API_KEY") async with client.text_to_speech_streaming.connect(model="bulbul:v3") as ws: await ws.configure( language_code="hi-IN", speaker="shubh", output_audio_bitrate= "128k" ) print("Sent configuration") text = ( "भारत की संस्कृति विश्व की सबसे प्राचीन और समृद्ध संस्कृतियों में से एक है।" "यह विविधता, सहिष्णुता और परंपराओं का अद्भुत संगम है, " "जिसमें विभिन्न धर्म, भाषाएं, त्योहार, संगीत, नृत्य, वास्तुकला और जीवनशैली शामिल हैं।" ) await ws.convert(text) print("Sent text message") await ws.flush() print("Flushed buffer") chunk_count = 0 with open("output.mp3", "wb") as f: async for message in ws: if isinstance(message, AudioOutput): chunk_count += 1 audio_chunk = base64.b64decode(message.data.audio) f.write(audio_chunk) f.flush() print(f"All {chunk_count} chunks saved to output.mp3") print("Audio generation complete") if hasattr(ws, "_websocket") and not ws._websocket.closed: await ws._websocket.close() print("WebSocket connection closed.") if __name__ == "__main__": asyncio.run(tts_stream()) # --- Notebook/Colab usage --- # await tts_stream() ``` > Control the quality and size of the synthesized audio output.