mtpadilla: consolidate streaming and non-streaming options into a single class with common API, with boolean switch variable added (streaming)
This commit is contained in:
@@ -16,7 +16,7 @@ from pipecat.pipeline.pipeline import Pipeline
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from pipecat.pipeline.runner import PipelineRunner
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from pipecat.pipeline.task import PipelineParams, PipelineTask
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from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
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from pipecat.services.inworld.tts import InworldHttpNonStreamingService, InworldHttpStreamingService
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from pipecat.services.inworld.tts import InworldTTSService
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from pipecat.services.openai.llm import OpenAILLMService
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from pipecat.services.openai.stt import OpenAISTTService
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from pipecat.transports.base_transport import BaseTransport, TransportParams
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@@ -58,30 +58,20 @@ async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_si
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prompt="Expect words related to dogs, such as breed names.",
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)
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streaming = True
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# Inworld TTS Service - Unified streaming and non-streaming
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# Set streaming=True for real-time audio, streaming=False for complete audio generation
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streaming = False # Toggle this to switch between modes
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if streaming:
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# Streaming TTS - Real-time audio generation as text is processed
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tts = InworldHttpStreamingService(
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api_key=os.getenv("INWORLD_API_KEY", ""),
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aiohttp_session=session,
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params=InworldHttpStreamingService.InputParams(
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voice_id="Ashley",
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model="inworld-tts-1",
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temperature=0.8,
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),
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)
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else:
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# Non-streaming TTS - Complete audio generation then playback
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tts = InworldHttpNonStreamingService(
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api_key=os.getenv("INWORLD_API_KEY", ""),
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aiohttp_session=session,
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params=InworldHttpNonStreamingService.InputParams(
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voice_id="Ashley",
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model="inworld-tts-1",
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temperature=0.8,
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),
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)
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tts = InworldTTSService(
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api_key=os.getenv("INWORLD_API_KEY", ""),
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aiohttp_session=session,
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streaming=streaming, # True: real-time chunks, False: complete audio then playback
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params=InworldTTSService.InputParams(
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voice_id="Ashley",
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model="inworld-tts-1",
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temperature=0.8,
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),
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)
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llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"))
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@@ -7,11 +7,12 @@
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"""Inworld AI Text-to-Speech Service Implementation.
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This module provides integration with Inworld AI's HTTP-based TTS API, enabling
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real-time text-to-speech synthesis with high-quality, natural-sounding voices.
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both streaming and non-streaming text-to-speech synthesis with high-quality,
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natural-sounding voices.
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Key Features:
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- HTTP streaming API support for low-latency audio generation
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- HTTP streaming and non-streaming API support for flexible audio generation
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- Multiple voice options (Ashley, Hades, etc.)
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- Automatic language detection from input text (no manual language setting required)
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- Real-time audio chunk processing with proper buffering
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@@ -20,8 +21,8 @@ Key Features:
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Technical Implementation:
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- Uses aiohttp for HTTP streaming connections
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- Implements JSON line-by-line parsing for streaming responses
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- Uses aiohttp for HTTP connections
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- Implements both JSON line-by-line parsing (streaming) and complete response (non-streaming)
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- Handles base64-encoded audio data with proper decoding
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- Manages audio continuity to prevent clicks and artifacts
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- Integrates with Pipecat's frame-based pipeline architecture
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@@ -29,15 +30,29 @@ Technical Implementation:
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Usage::
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async with aiohttp.ClientSession() as session:
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tts = InworldHttpStreamingService(
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# Streaming mode (default) - real-time audio generation
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tts = InworldTTSService(
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api_key=os.getenv("INWORLD_API_KEY"),
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aiohttp_session=session,
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params=InworldHttpStreamingService.InputParams(
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streaming=True, # Default
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params=InworldTTSService.InputParams(
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voice_id="Ashley",
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model="inworld-tts-1",
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temperature=0.8, # Optional: control synthesis variability (range: [0, 2])
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),
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)
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# Non-streaming mode - complete audio generation then playback
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tts = InworldTTSService(
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api_key=os.getenv("INWORLD_API_KEY"),
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aiohttp_session=session,
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streaming=False,
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params=InworldTTSService.InputParams(
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voice_id="Ashley",
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model="inworld-tts-1",
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temperature=0.8,
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),
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)
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"""
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import base64
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@@ -70,27 +85,30 @@ from pipecat.utils.text.skip_tags_aggregator import SkipTagsAggregator
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from pipecat.utils.tracing.service_decorators import traced_tts
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class InworldHttpStreamingService(TTSService):
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class InworldTTSService(TTSService):
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"""Inworld AI HTTP-based Text-to-Speech Service.
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This service integrates Inworld AI's high-quality TTS API with Pipecat's pipeline
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architecture. It provides real-time speech synthesis with natural-sounding voices
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and low-latency streaming audio delivery.
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This unified service integrates Inworld AI's high-quality TTS API with Pipecat's pipeline
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architecture. It supports both streaming and non-streaming modes, providing flexible
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speech synthesis with natural-sounding voices.
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Key Features:
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- Real-time HTTP streaming for minimal latency
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- **Streaming Mode**: Real-time HTTP streaming for minimal latency
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- **Non-Streaming Mode**: Complete audio synthesis then chunked playback
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- Multiple voice options (Ashley, Hades, etc.)
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- High-quality audio output (48kHz LINEAR16 PCM)
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- Automatic audio format handling and header stripping
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- Comprehensive error handling and recovery
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- Built-in performance metrics and monitoring
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- Unified interface for both modes
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Technical Architecture:
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- Uses aiohttp for non-blocking HTTP requests
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- Implements JSON line-by-line streaming protocol
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- Processes base64-encoded audio chunks in real-time
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- **Streaming**: Implements JSON line-by-line streaming protocol
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- **Non-Streaming**: Single HTTP POST with complete response
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- Processes base64-encoded audio chunks in real-time or batch
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- Manages audio continuity to prevent artifacts
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- Integrates with Pipecat's frame-based pipeline system
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@@ -100,31 +118,38 @@ class InworldHttpStreamingService(TTSService):
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- Models: inworld-tts-1 and other available models
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- Audio Formats: LINEAR16 PCM at various sample rates
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- Language Detection: Automatically inferred from input text (no explicit language setting required)
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- Mode Selection: streaming=True for real-time, streaming=False for complete synthesis
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Example Usage::
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async with aiohttp.ClientSession() as session:
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# Using default settings (Ashley voice, inworld-tts-1 model)
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tts = InworldHttpStreamingService(
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# Streaming mode (default) - Real-time audio generation
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tts_streaming = InworldTTSService(
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api_key=os.getenv("INWORLD_API_KEY"),
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aiohttp_session=session,
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streaming=True, # Default behavior
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params=InworldTTSService.InputParams(
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voice_id="Ashley",
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model="inworld-tts-1",
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temperature=0.8, # Add variability to speech synthesis (range: [0, 2])
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),
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)
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# Or with custom voice, model, and temperature via params
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params = InworldHttpStreamingService.InputParams(
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voice_id="Hades",
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model="inworld-tts-1-max",
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temperature=0.8, # Add variability to speech synthesis (range: [0, 2])
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)
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tts = InworldHttpStreamingService(
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# Non-streaming mode - Complete audio then playback
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tts_complete = InworldTTSService(
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api_key=os.getenv("INWORLD_API_KEY"),
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aiohttp_session=session,
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params=params,
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streaming=False,
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params=InworldTTSService.InputParams(
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voice_id="Hades",
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model="inworld-tts-1-max",
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temperature=0.8,
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),
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)
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"""
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class InputParams(BaseModel):
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"""Input parameters for Inworld HTTP TTS configuration.
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"""Input parameters for Inworld TTS configuration.
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Parameters:
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voice_id: Voice selection for speech synthesis (e.g., "Ashley", "Hades").
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@@ -146,23 +171,29 @@ class InworldHttpStreamingService(TTSService):
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*,
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api_key: str,
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aiohttp_session: aiohttp.ClientSession,
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base_url: str = "https://api.inworld.ai/tts/v1/voice:stream",
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streaming: bool = True,
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base_url: Optional[str] = None,
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sample_rate: Optional[int] = None,
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encoding: str = "LINEAR16",
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params: Optional[InputParams] = None,
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**kwargs,
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):
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"""Initialize the Inworld HTTP TTS service.
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"""Initialize the Inworld TTS service.
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Sets up the TTS service with Inworld AI's streaming API configuration.
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This constructor prepares all necessary parameters for real-time speech synthesis.
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Sets up the TTS service with Inworld AI's API configuration.
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This constructor prepares all necessary parameters for speech synthesis.
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Args:
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api_key: Inworld API key for authentication (base64-encoded from Inworld Portal).
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Get this from: Inworld Portal > Settings > API Keys > Runtime API Key
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aiohttp_session: Shared aiohttp session for HTTP requests. Must be provided
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for proper connection pooling and resource management.
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base_url: Base URL for Inworld HTTP API. Uses streaming endpoint by default.
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streaming: Whether to use streaming mode (True) or non-streaming mode (False).
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- True: Real-time audio chunks as they're generated (lower latency)
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- False: Complete audio file generated first, then chunked for playback (simpler)
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base_url: Base URL for Inworld HTTP API. If None, automatically selected based on streaming mode:
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- Streaming: "https://api.inworld.ai/tts/v1/voice:stream"
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- Non-streaming: "https://api.inworld.ai/tts/v1/voice"
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Should normally not be changed unless using a different environment.
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sample_rate: Audio sample rate in Hz. If None, uses default from StartFrame.
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Common values: 48000 (high quality), 24000 (good quality), 16000 (basic)
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@@ -185,11 +216,19 @@ class InworldHttpStreamingService(TTSService):
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super().__init__(sample_rate=sample_rate, **kwargs)
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# Use provided params or create default configuration
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params = params or InworldHttpStreamingService.InputParams()
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params = params or InworldTTSService.InputParams()
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# Store core configuration for API requests
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self._api_key = api_key # Authentication credentials
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self._session = aiohttp_session # HTTP session for requests
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self._streaming = streaming # Streaming mode selection
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# Set base URL based on streaming mode if not provided
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if base_url is None:
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if streaming:
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base_url = "https://api.inworld.ai/tts/v1/voice:stream" # Streaming endpoint
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else:
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base_url = "https://api.inworld.ai/tts/v1/voice" # Non-streaming endpoint
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self._base_url = base_url # API endpoint URL
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# Build settings dictionary that matches Inworld's API expectations
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@@ -216,12 +255,12 @@ class InworldHttpStreamingService(TTSService):
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"""Check if this service can generate processing metrics.
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Returns:
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True, as Inworld HTTP service supports metrics generation.
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True, as Inworld TTS service supports metrics generation.
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"""
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return True
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async def start(self, frame: StartFrame):
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"""Start the Inworld HTTP TTS service.
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"""Start the Inworld TTS service.
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Args:
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frame: The start frame containing initialization parameters.
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@@ -230,7 +269,7 @@ class InworldHttpStreamingService(TTSService):
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self._settings["audio_config"]["sample_rate_hertz"] = self.sample_rate
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async def stop(self, frame: EndFrame):
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"""Stop the Inworld HTTP TTS service.
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"""Stop the Inworld TTS service.
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Args:
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frame: The end frame.
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@@ -238,7 +277,7 @@ class InworldHttpStreamingService(TTSService):
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await super().stop(frame)
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async def cancel(self, frame: CancelFrame):
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"""Cancel the Inworld HTTP TTS service.
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"""Cancel the Inworld TTS service.
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Args:
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frame: The cancel frame.
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@@ -247,21 +286,30 @@ class InworldHttpStreamingService(TTSService):
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@traced_tts
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async def run_tts(self, text: str) -> AsyncGenerator[Frame, None]:
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"""Generate speech from text using Inworld's streaming HTTP API.
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"""Generate speech from text using Inworld's HTTP API.
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This is the core TTS processing function that:
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This is the core TTS processing function that adapts its behavior based on the streaming mode:
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**Streaming Mode (streaming=True)**:
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1. Sends text to Inworld's streaming TTS endpoint
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2. Receives JSON-streamed audio chunks in real-time
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3. Processes and cleans audio data (removes WAV headers, validates content)
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4. Yields audio frames for immediate playback in the pipeline
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**Non-Streaming Mode (streaming=False)**:
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1. Sends text to Inworld's non-streaming TTS endpoint
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2. Receives complete audio file as base64-encoded response
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3. Processes entire audio and chunks for playback
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4. Yields audio frames in manageable pieces
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Technical Details:
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- Uses HTTP streaming with JSON line-by-line responses
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- Each JSON line contains base64-encoded audio data
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- Implements buffering to handle partial JSON lines
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- **Streaming**: Uses HTTP streaming with JSON line-by-line responses
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- **Non-Streaming**: Single HTTP POST with complete JSON response
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- Each audio chunk contains base64-encoded audio data
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- Implements buffering to handle partial data (streaming mode)
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- Strips WAV headers to prevent audio artifacts/clicks
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- Provides real-time audio streaming for low latency
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- Provides optimized audio delivery for each mode
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Args:
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text: The text to synthesize into speech.
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@@ -272,7 +320,7 @@ class InworldHttpStreamingService(TTSService):
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Raises:
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ErrorFrame: If API errors occur or audio processing fails.
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"""
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logger.debug(f"{self}: Generating TTS [{text}]")
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logger.debug(f"{self}: Generating TTS [{text}] (streaming={self._streaming})")
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# ================================================================================
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# STEP 1: PREPARE API REQUEST
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@@ -302,7 +350,7 @@ class InworldHttpStreamingService(TTSService):
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try:
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# ================================================================================
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# STEP 2: INITIALIZE METRICS AND STREAMING
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# STEP 2: INITIALIZE METRICS AND PROCESSING
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# ================================================================================
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# Start measuring Time To First Byte (TTFB) for performance tracking
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await self.start_ttfb_metrics()
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@@ -312,10 +360,10 @@ class InworldHttpStreamingService(TTSService):
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yield TTSStartedFrame()
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# ================================================================================
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# STEP 3: MAKE HTTP STREAMING REQUEST
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# STEP 3: MAKE HTTP REQUEST (MODE-SPECIFIC)
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# ================================================================================
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# Use aiohttp's streaming POST to Inworld's streaming endpoint
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# The endpoint returns JSON lines with audio chunks as they're generated
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# Use aiohttp to make request to Inworld's endpoint
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# Behavior differs based on streaming mode
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async with self._session.post(
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self._base_url, json=payload, headers=headers
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) as response:
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@@ -330,115 +378,34 @@ class InworldHttpStreamingService(TTSService):
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return
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# ================================================================================
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# STEP 5: PROCESS STREAMING JSON RESPONSE
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# STEP 5: PROCESS RESPONSE (MODE-SPECIFIC)
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# ================================================================================
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# Inworld streams JSON lines where each line contains audio data
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# We need to buffer incoming data and process complete lines
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# Buffer to accumulate incoming text data
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# This handles cases where JSON lines are split across HTTP chunks
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buffer = ""
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# Read HTTP response in manageable chunks (1KB each)
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# This prevents memory issues with large responses
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async for chunk in response.content.iter_chunked(1024):
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if not chunk:
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continue
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# ============================================================================
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# STEP 6: BUFFER MANAGEMENT
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# ============================================================================
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# Decode binary chunk to text and add to our line buffer
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# Each chunk may contain partial JSON lines, so we need to accumulate
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buffer += chunk.decode("utf-8")
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# ============================================================================
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# STEP 7: LINE-BY-LINE JSON PROCESSING
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# ============================================================================
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# Process all complete lines in the buffer (lines ending with \n)
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# Leave partial lines in buffer for next iteration
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while "\n" in buffer:
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# Split on first newline, keeping remainder in buffer
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line, buffer = buffer.split("\n", 1)
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line_str = line.strip()
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# Skip empty lines (common in streaming responses)
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if not line_str:
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continue
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try:
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# ================================================================
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# STEP 8: PARSE JSON AND EXTRACT AUDIO
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# ================================================================
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# Parse the JSON line - should contain audio data
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chunk_data = json.loads(line_str)
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# Check if this line contains audio content
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# Inworld's response format: {"result": {"audioContent": "base64data"}}
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if "result" in chunk_data and "audioContent" in chunk_data["result"]:
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# Decode base64 audio data to binary
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audio_chunk = base64.b64decode(chunk_data["result"]["audioContent"])
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# ========================================================
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# STEP 9: AUDIO DATA VALIDATION
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# ========================================================
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# Skip empty audio chunks that could cause discontinuities
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# Empty chunks can create gaps or clicks in audio playback
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if not audio_chunk:
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continue
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# Start with the raw audio data
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audio_data = audio_chunk
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# ========================================================
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# STEP 10: WAV HEADER REMOVAL (CRITICAL FOR AUDIO QUALITY)
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# ========================================================
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# Each audio chunk may have its own WAV header (44 bytes)
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# These headers contain metadata and will sound like clicks if played
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# We must strip them from EVERY chunk, not just the first one
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if (
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len(audio_chunk) > 44 # Ensure chunk is large enough
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and audio_chunk.startswith(
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b"RIFF"
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) # Check for WAV header magic bytes
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):
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# Remove the 44-byte WAV header to get pure audio data
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audio_data = audio_chunk[44:]
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# ========================================================
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# STEP 11: YIELD AUDIO FRAME TO PIPELINE
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# ========================================================
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# Only yield frames with actual audio content
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# Empty frames can cause pipeline issues
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if len(audio_data) > 0:
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# Create Pipecat audio frame with processed audio data
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yield TTSAudioRawFrame(
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audio=audio_data, # Clean audio without headers
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||||
sample_rate=self.sample_rate, # Configured sample rate (48kHz)
|
||||
num_channels=1, # Mono audio
|
||||
)
|
||||
|
||||
except json.JSONDecodeError:
|
||||
# Ignore malformed JSON lines - streaming can have partial data
|
||||
# This is normal in HTTP streaming scenarios
|
||||
continue
|
||||
# Choose processing method based on streaming mode
|
||||
if self._streaming:
|
||||
# Stream processing: JSON line-by-line with real-time audio
|
||||
async for frame in self._process_streaming_response(response):
|
||||
yield frame
|
||||
else:
|
||||
# Non-stream processing: Complete JSON response with batch audio
|
||||
async for frame in self._process_non_streaming_response(response):
|
||||
yield frame
|
||||
|
||||
# ================================================================================
|
||||
# STEP 12: FINALIZE METRICS AND CLEANUP
|
||||
# STEP 6: FINALIZE METRICS AND CLEANUP
|
||||
# ================================================================================
|
||||
# Start usage metrics tracking after successful completion
|
||||
await self.start_tts_usage_metrics(text)
|
||||
|
||||
except Exception as e:
|
||||
# ================================================================================
|
||||
# STEP 13: ERROR HANDLING
|
||||
# STEP 7: ERROR HANDLING
|
||||
# ================================================================================
|
||||
# Log any unexpected errors and notify the pipeline
|
||||
logger.error(f"{self} exception: {e}")
|
||||
await self.push_error(ErrorFrame(f"Error generating TTS: {e}"))
|
||||
finally:
|
||||
# ================================================================================
|
||||
# STEP 14: CLEANUP AND COMPLETION
|
||||
# STEP 8: CLEANUP AND COMPLETION
|
||||
# ================================================================================
|
||||
# Always stop metrics tracking, even if errors occurred
|
||||
await self.stop_ttfb_metrics()
|
||||
@@ -447,363 +414,187 @@ class InworldHttpStreamingService(TTSService):
|
||||
# This allows downstream processors to finalize audio processing
|
||||
yield TTSStoppedFrame()
|
||||
|
||||
async def _process_streaming_response(
|
||||
self, response: aiohttp.ClientResponse
|
||||
) -> AsyncGenerator[Frame, None]:
|
||||
"""Process streaming JSON response with real-time audio chunks.
|
||||
|
||||
class InworldHttpNonStreamingService(TTSService):
|
||||
"""Inworld AI HTTP-based Text-to-Speech Service (Non-Streaming).
|
||||
|
||||
This service integrates with Inworld AI's non-streaming TTS API for simpler,
|
||||
complete audio synthesis. Suitable for use cases where streaming is not required
|
||||
and you prefer to receive the complete audio file at once.
|
||||
|
||||
Key Features:
|
||||
|
||||
- Simple HTTP request/response for complete audio synthesis
|
||||
- Same voice options as streaming version (Ashley, Hades, etc.)
|
||||
- High-quality audio output (48kHz LINEAR16 PCM)
|
||||
- Automatic language detection from input text
|
||||
- Support for temperature parameter for synthesis variability
|
||||
- Lower complexity compared to streaming implementation
|
||||
|
||||
Technical Architecture:
|
||||
|
||||
- Uses aiohttp for single HTTP POST request
|
||||
- Downloads complete audio as base64-encoded data
|
||||
- Processes entire audio file and chunks for playback
|
||||
- Integrates with Pipecat's frame-based pipeline system
|
||||
|
||||
Usage::
|
||||
|
||||
async with aiohttp.ClientSession() as session:
|
||||
# Using default settings (Ashley voice, inworld-tts-1 model)
|
||||
tts = InworldHttpNonStreamingService(
|
||||
api_key=os.getenv("INWORLD_API_KEY"),
|
||||
aiohttp_session=session,
|
||||
)
|
||||
|
||||
# Or with custom voice, model, and temperature
|
||||
params = InworldHttpNonStreamingService.InputParams(
|
||||
voice_id="Hades",
|
||||
model="inworld-tts-1-max",
|
||||
temperature=0.8, # Control synthesis variability (range: [0, 2])
|
||||
)
|
||||
tts = InworldHttpNonStreamingService(
|
||||
api_key=os.getenv("INWORLD_API_KEY"),
|
||||
aiohttp_session=session,
|
||||
params=params,
|
||||
)
|
||||
"""
|
||||
|
||||
class InputParams(BaseModel):
|
||||
"""Input parameters for Inworld non-streaming TTS configuration.
|
||||
|
||||
Parameters:
|
||||
voice_id: Voice selection for speech synthesis (e.g., "Ashley", "Hades").
|
||||
model: TTS model to use (e.g., "inworld-tts-1", "inworld-tts-1-max").
|
||||
temperature: Voice temperature control for synthesis variability (e.g., 0.8).
|
||||
Valid range: [0, 2]. Higher values increase variability.
|
||||
|
||||
Note:
|
||||
Language is automatically inferred from the input text by Inworld's TTS models,
|
||||
so no explicit language parameter is required.
|
||||
"""
|
||||
|
||||
voice_id: Optional[str] = "Ashley" # defaults to the Ashley voice
|
||||
model: Optional[str] = "inworld-tts-1" # defaults to the inworld-tts-1 model
|
||||
temperature: Optional[float] = None # optional temperature control (range: [0, 2])
|
||||
|
||||
def __init__(
|
||||
self,
|
||||
*,
|
||||
api_key: str,
|
||||
aiohttp_session: Optional[aiohttp.ClientSession] = None,
|
||||
base_url: str = "https://api.inworld.ai/tts/v1/voice", # Non-streaming endpoint
|
||||
sample_rate: Optional[int] = None,
|
||||
encoding: str = "LINEAR16",
|
||||
params: Optional[InputParams] = None,
|
||||
**kwargs,
|
||||
):
|
||||
"""Initialize the Inworld non-streaming TTS service.
|
||||
|
||||
Sets up the TTS service with Inworld AI's non-streaming API configuration.
|
||||
This constructor prepares all necessary parameters for complete audio synthesis.
|
||||
This method handles Inworld's streaming endpoint response format:
|
||||
- JSON lines containing base64-encoded audio chunks
|
||||
- Real-time processing as data arrives
|
||||
- Line buffering to handle partial JSON data
|
||||
|
||||
Args:
|
||||
api_key: Inworld API key for authentication (base64-encoded from Inworld Portal).
|
||||
Get this from: Inworld Portal > Settings > API Keys > Runtime API Key
|
||||
aiohttp_session: Shared aiohttp session for HTTP requests. Must be provided
|
||||
for proper connection pooling and resource management.
|
||||
base_url: Base URL for Inworld non-streaming HTTP API. Uses non-streaming endpoint by default.
|
||||
Should normally not be changed unless using a different environment.
|
||||
sample_rate: Audio sample rate in Hz. If None, uses default from StartFrame.
|
||||
Common values: 48000 (high quality), 24000 (good quality), 16000 (basic)
|
||||
encoding: Audio encoding format. Supported options:
|
||||
- "LINEAR16" (default) - Uncompressed PCM, best quality
|
||||
- Other formats as supported by Inworld API
|
||||
params: Input parameters for voice and model configuration. Use this to specify:
|
||||
- voice_id: Voice selection ("Ashley", "Hades", etc.)
|
||||
- model: TTS model ("inworld-tts-1", "inworld-tts-1-max", etc.)
|
||||
- temperature: Voice temperature control for variability (range: [0, 2], e.g., 0.8, optional)
|
||||
If None, uses default values (Ashley voice, inworld-tts-1 model).
|
||||
Note: Language is automatically inferred from input text.
|
||||
**kwargs: Additional arguments passed to the parent TTSService class.
|
||||
|
||||
Note:
|
||||
The aiohttp_session parameter is required because Inworld's HTTP API
|
||||
benefits from connection reuse and proper async session management.
|
||||
"""
|
||||
# Initialize parent TTSService with audio configuration
|
||||
super().__init__(sample_rate=sample_rate, **kwargs)
|
||||
|
||||
# Use provided params or create default configuration
|
||||
params = params or InworldHttpNonStreamingService.InputParams()
|
||||
|
||||
# Store core configuration for API requests
|
||||
self._api_key = api_key # Authentication credentials
|
||||
self._session = aiohttp_session # HTTP session for requests (optional)
|
||||
self._base_url = base_url # API endpoint URL
|
||||
|
||||
# Build settings dictionary that matches Inworld's API expectations
|
||||
# This will be sent as JSON payload in the TTS request
|
||||
# Note: Language is automatically inferred from text by Inworld's models
|
||||
self._settings = {
|
||||
"voiceId": params.voice_id or "Ashley", # Voice selection from params
|
||||
"modelId": params.model or "inworld-tts-1", # TTS model selection from params
|
||||
"audio_config": { # Audio format configuration
|
||||
"audio_encoding": encoding, # Format: LINEAR16, MP3, etc.
|
||||
"sample_rate_hertz": 0, # Will be set in start() from parent service
|
||||
},
|
||||
}
|
||||
|
||||
# Add optional temperature parameter if provided (valid range: [0, 2])
|
||||
if params.temperature is not None:
|
||||
self._settings["temperature"] = params.temperature
|
||||
|
||||
# Register voice and model with parent service for metrics and tracking
|
||||
self.set_voice(params.voice_id or "Ashley") # Used for logging and metrics
|
||||
self.set_model_name(params.model or "inworld-tts-1") # Used for performance tracking
|
||||
|
||||
def can_generate_metrics(self) -> bool:
|
||||
"""Check if this service can generate processing metrics.
|
||||
|
||||
Returns:
|
||||
True, as Inworld non-streaming service supports metrics generation.
|
||||
"""
|
||||
return True
|
||||
|
||||
async def start(self, frame: StartFrame):
|
||||
"""Start the Inworld non-streaming TTS service.
|
||||
|
||||
Args:
|
||||
frame: The start frame containing initialization parameters.
|
||||
"""
|
||||
await super().start(frame)
|
||||
self._settings["audio_config"]["sample_rate_hertz"] = self.sample_rate
|
||||
|
||||
async def stop(self, frame: EndFrame):
|
||||
"""Stop the Inworld non-streaming TTS service.
|
||||
|
||||
Args:
|
||||
frame: The end frame.
|
||||
"""
|
||||
await super().stop(frame)
|
||||
|
||||
async def cancel(self, frame: CancelFrame):
|
||||
"""Cancel the Inworld non-streaming TTS service.
|
||||
|
||||
Args:
|
||||
frame: The cancel frame.
|
||||
"""
|
||||
await super().cancel(frame)
|
||||
|
||||
@traced_tts
|
||||
async def run_tts(self, text: str) -> AsyncGenerator[Frame, None]:
|
||||
"""Generate speech from text using Inworld's non-streaming HTTP API.
|
||||
|
||||
This method sends text to Inworld's non-streaming TTS endpoint and receives
|
||||
the complete audio file as a base64-encoded response. The audio is then
|
||||
chunked and yielded for playback in the pipeline.
|
||||
|
||||
Args:
|
||||
text: The text to synthesize into speech.
|
||||
response: The aiohttp response object from streaming endpoint.
|
||||
|
||||
Yields:
|
||||
Frame: Audio frames containing the synthesized speech, plus control frames.
|
||||
|
||||
Raises:
|
||||
ErrorFrame: If API errors occur or audio processing fails.
|
||||
Frame: Audio frames as they're processed from the stream.
|
||||
"""
|
||||
logger.debug(f"{self}: Generating TTS [{text}]")
|
||||
# ================================================================================
|
||||
# STREAMING: PROCESS JSON LINE-BY-LINE RESPONSE
|
||||
# ================================================================================
|
||||
# Inworld streams JSON lines where each line contains audio data
|
||||
# We need to buffer incoming data and process complete lines
|
||||
|
||||
# Buffer to accumulate incoming text data
|
||||
# This handles cases where JSON lines are split across HTTP chunks
|
||||
buffer = ""
|
||||
|
||||
# Read HTTP response in manageable chunks (1KB each)
|
||||
# This prevents memory issues with large responses
|
||||
async for chunk in response.content.iter_chunked(1024):
|
||||
if not chunk:
|
||||
continue
|
||||
|
||||
# ============================================================================
|
||||
# BUFFER MANAGEMENT
|
||||
# ============================================================================
|
||||
# Decode binary chunk to text and add to our line buffer
|
||||
# Each chunk may contain partial JSON lines, so we need to accumulate
|
||||
buffer += chunk.decode("utf-8")
|
||||
|
||||
# ============================================================================
|
||||
# LINE-BY-LINE JSON PROCESSING
|
||||
# ============================================================================
|
||||
# Process all complete lines in the buffer (lines ending with \n)
|
||||
# Leave partial lines in buffer for next iteration
|
||||
while "\n" in buffer:
|
||||
# Split on first newline, keeping remainder in buffer
|
||||
line, buffer = buffer.split("\n", 1)
|
||||
line_str = line.strip()
|
||||
|
||||
# Skip empty lines (common in streaming responses)
|
||||
if not line_str:
|
||||
continue
|
||||
|
||||
try:
|
||||
# ================================================================
|
||||
# PARSE JSON AND EXTRACT AUDIO
|
||||
# ================================================================
|
||||
# Parse the JSON line - should contain audio data
|
||||
chunk_data = json.loads(line_str)
|
||||
|
||||
# Check if this line contains audio content
|
||||
# Inworld's response format: {"result": {"audioContent": "base64data"}}
|
||||
if "result" in chunk_data and "audioContent" in chunk_data["result"]:
|
||||
# Process the audio chunk
|
||||
async for frame in self._process_audio_chunk(
|
||||
base64.b64decode(chunk_data["result"]["audioContent"])
|
||||
):
|
||||
yield frame
|
||||
|
||||
except json.JSONDecodeError:
|
||||
# Ignore malformed JSON lines - streaming can have partial data
|
||||
# This is normal in HTTP streaming scenarios
|
||||
continue
|
||||
|
||||
async def _process_non_streaming_response(
|
||||
self, response: aiohttp.ClientResponse
|
||||
) -> AsyncGenerator[Frame, None]:
|
||||
"""Process complete JSON response with full audio content.
|
||||
|
||||
This method handles Inworld's non-streaming endpoint response format:
|
||||
- Single JSON response with complete base64-encoded audio
|
||||
- Full audio download then chunked playback
|
||||
- Simpler processing without line buffering
|
||||
|
||||
Args:
|
||||
response: The aiohttp response object from non-streaming endpoint.
|
||||
|
||||
Yields:
|
||||
Frame: Audio frames chunked from the complete audio.
|
||||
"""
|
||||
# ================================================================================
|
||||
# NON-STREAMING: PARSE COMPLETE JSON RESPONSE
|
||||
# ================================================================================
|
||||
# Parse the complete JSON response containing base64 audio data
|
||||
response_data = await response.json()
|
||||
|
||||
# ================================================================================
|
||||
# STEP 1: PREPARE API REQUEST
|
||||
# EXTRACT AND VALIDATE AUDIO CONTENT
|
||||
# ================================================================================
|
||||
# Build the JSON payload according to Inworld's non-streaming API specification
|
||||
# This matches the format shown in their documentation examples
|
||||
# Note: Language is automatically inferred from the input text by Inworld's models
|
||||
payload = {
|
||||
"text": text, # Text to synthesize
|
||||
"voiceId": self._settings["voiceId"], # Voice selection (Ashley, Hades, etc.)
|
||||
"modelId": self._settings["modelId"], # TTS model (inworld-tts-1)
|
||||
"audio_config": self._settings["audio_config"], # Audio format settings
|
||||
}
|
||||
# Extract the base64-encoded audio content from response
|
||||
if "audioContent" not in response_data:
|
||||
logger.error("No audioContent in Inworld API response")
|
||||
await self.push_error(ErrorFrame("No audioContent in response"))
|
||||
return
|
||||
|
||||
# Add optional temperature parameter if configured (valid range: [0, 2])
|
||||
if "temperature" in self._settings:
|
||||
payload["temperature"] = self._settings["temperature"]
|
||||
# ================================================================================
|
||||
# DECODE AND PROCESS COMPLETE AUDIO DATA
|
||||
# ================================================================================
|
||||
# Decode the base64 audio data to binary
|
||||
audio_data = base64.b64decode(response_data["audioContent"])
|
||||
|
||||
# Set up HTTP headers for authentication and content type
|
||||
# Inworld requires Basic auth with base64-encoded API key
|
||||
headers = {
|
||||
"Authorization": f"Basic {self._api_key}", # Base64 API key from Inworld Portal
|
||||
"Content-Type": "application/json", # JSON request body
|
||||
}
|
||||
# Strip WAV header if present (Inworld may include WAV header)
|
||||
# This prevents audio clicks and ensures clean audio playback
|
||||
if len(audio_data) > 44 and audio_data.startswith(b"RIFF"):
|
||||
audio_data = audio_data[44:]
|
||||
|
||||
try:
|
||||
# ================================================================================
|
||||
# STEP 2: INITIALIZE METRICS AND STREAMING
|
||||
# ================================================================================
|
||||
# Start measuring Time To First Byte (TTFB) for performance tracking
|
||||
await self.start_ttfb_metrics()
|
||||
# ================================================================================
|
||||
# CHUNK AND YIELD COMPLETE AUDIO FOR PLAYBACK
|
||||
# ================================================================================
|
||||
# Chunk the complete audio for streaming playback
|
||||
# This allows the pipeline to process audio in manageable pieces
|
||||
CHUNK_SIZE = self.chunk_size
|
||||
|
||||
# Signal to the pipeline that TTS generation has started
|
||||
# This allows downstream processors to prepare for incoming audio
|
||||
yield TTSStartedFrame()
|
||||
for i in range(0, len(audio_data), CHUNK_SIZE):
|
||||
chunk = audio_data[i : i + CHUNK_SIZE]
|
||||
if len(chunk) > 0:
|
||||
await self.stop_ttfb_metrics()
|
||||
yield TTSAudioRawFrame(
|
||||
audio=chunk,
|
||||
sample_rate=self.sample_rate,
|
||||
num_channels=1,
|
||||
)
|
||||
|
||||
# ================================================================================
|
||||
# STEP 3: MAKE HTTP NON-STREAMING REQUEST
|
||||
# ================================================================================
|
||||
# Make single HTTP POST request to Inworld's non-streaming endpoint
|
||||
# This endpoint returns complete audio as base64-encoded data
|
||||
# Create session if none was provided
|
||||
if self._session:
|
||||
session = self._session
|
||||
else:
|
||||
session = aiohttp.ClientSession()
|
||||
async def _process_audio_chunk(self, audio_chunk: bytes) -> AsyncGenerator[Frame, None]:
|
||||
"""Process a single audio chunk (common logic for both modes).
|
||||
|
||||
async with (
|
||||
session
|
||||
if not self._session
|
||||
else session.post(
|
||||
self._base_url, json=payload, headers=headers
|
||||
) as context_or_response
|
||||
):
|
||||
if self._session:
|
||||
response = context_or_response
|
||||
else:
|
||||
async with context_or_response.post(
|
||||
self._base_url, json=payload, headers=headers
|
||||
) as response:
|
||||
# ================================================================
|
||||
# STEP 4: HANDLE HTTP ERRORS
|
||||
# ================================================================
|
||||
# Check for API errors (expired keys, invalid requests, etc.)
|
||||
if response.status != 200:
|
||||
error_text = await response.text()
|
||||
logger.error(f"Inworld API error: {error_text}")
|
||||
await self.push_error(ErrorFrame(f"Inworld API error: {error_text}"))
|
||||
return
|
||||
This method handles audio chunk processing that's common to both streaming
|
||||
and non-streaming modes:
|
||||
- WAV header removal
|
||||
- Audio validation
|
||||
- Frame creation and yielding
|
||||
|
||||
# ================================================================
|
||||
# STEP 5: PARSE COMPLETE JSON RESPONSE
|
||||
# ================================================================
|
||||
# Parse the complete JSON response containing base64 audio data
|
||||
response_data = await response.json()
|
||||
Args:
|
||||
audio_chunk: Raw audio data bytes to process.
|
||||
|
||||
# ================================================================
|
||||
# STEP 6: EXTRACT AND VALIDATE AUDIO CONTENT
|
||||
# ================================================================
|
||||
# Extract the base64-encoded audio content from response
|
||||
if "audioContent" not in response_data:
|
||||
logger.error("No audioContent in Inworld API response")
|
||||
await self.push_error(ErrorFrame("No audioContent in response"))
|
||||
return
|
||||
Yields:
|
||||
Frame: Audio frame if chunk contains valid audio data.
|
||||
"""
|
||||
# ========================================================
|
||||
# AUDIO DATA VALIDATION
|
||||
# ========================================================
|
||||
# Skip empty audio chunks that could cause discontinuities
|
||||
# Empty chunks can create gaps or clicks in audio playback
|
||||
if not audio_chunk:
|
||||
return
|
||||
|
||||
# ================================================================
|
||||
# STEP 7: DECODE AND PROCESS AUDIO DATA
|
||||
# ================================================================
|
||||
# Decode the base64 audio data to binary
|
||||
audio_data = base64.b64decode(response_data["audioContent"])
|
||||
# Start with the raw audio data
|
||||
audio_data = audio_chunk
|
||||
|
||||
# Strip WAV header if present (Inworld may include WAV header)
|
||||
# This prevents audio clicks and ensures clean audio playback
|
||||
if len(audio_data) > 44 and audio_data.startswith(b"RIFF"):
|
||||
audio_data = audio_data[44:]
|
||||
# ========================================================
|
||||
# WAV HEADER REMOVAL (CRITICAL FOR AUDIO QUALITY)
|
||||
# ========================================================
|
||||
# Each audio chunk may have its own WAV header (44 bytes)
|
||||
# These headers contain metadata and will sound like clicks if played
|
||||
# We must strip them from EVERY chunk, not just the first one
|
||||
if (
|
||||
len(audio_chunk) > 44 # Ensure chunk is large enough
|
||||
and audio_chunk.startswith(b"RIFF") # Check for WAV header magic bytes
|
||||
):
|
||||
# Remove the 44-byte WAV header to get pure audio data
|
||||
audio_data = audio_chunk[44:]
|
||||
|
||||
# ================================================================
|
||||
# STEP 8: START USAGE METRICS TRACKING
|
||||
# ================================================================
|
||||
await self.start_tts_usage_metrics(text)
|
||||
|
||||
# ================================================================
|
||||
# STEP 9: CHUNK AND YIELD AUDIO FOR PLAYBACK
|
||||
# ================================================================
|
||||
# Chunk the complete audio for streaming playback
|
||||
# This allows the pipeline to process audio in manageable pieces
|
||||
CHUNK_SIZE = self.chunk_size
|
||||
|
||||
for i in range(0, len(audio_data), CHUNK_SIZE):
|
||||
chunk = audio_data[i : i + CHUNK_SIZE]
|
||||
if len(chunk) > 0:
|
||||
await self.stop_ttfb_metrics()
|
||||
yield TTSAudioRawFrame(
|
||||
audio=chunk,
|
||||
sample_rate=self.sample_rate,
|
||||
num_channels=1,
|
||||
)
|
||||
|
||||
if self._session:
|
||||
# Handle HTTP errors
|
||||
if response.status != 200:
|
||||
error_text = await response.text()
|
||||
logger.error(f"Inworld API error: {error_text}")
|
||||
await self.push_error(ErrorFrame(f"Inworld API error: {error_text}"))
|
||||
return
|
||||
|
||||
# Parse the complete JSON response
|
||||
response_data = await response.json()
|
||||
|
||||
# Extract the base64-encoded audio content
|
||||
if "audioContent" not in response_data:
|
||||
logger.error("No audioContent in Inworld API response")
|
||||
await self.push_error(ErrorFrame("No audioContent in response"))
|
||||
return
|
||||
|
||||
# Decode the base64 audio data
|
||||
audio_data = base64.b64decode(response_data["audioContent"])
|
||||
|
||||
# Strip WAV header if present (Inworld may include WAV header)
|
||||
if len(audio_data) > 44 and audio_data.startswith(b"RIFF"):
|
||||
audio_data = audio_data[44:]
|
||||
|
||||
await self.start_tts_usage_metrics(text)
|
||||
|
||||
# Chunk the complete audio for streaming playback
|
||||
CHUNK_SIZE = self.chunk_size
|
||||
|
||||
for i in range(0, len(audio_data), CHUNK_SIZE):
|
||||
chunk = audio_data[i : i + CHUNK_SIZE]
|
||||
if len(chunk) > 0:
|
||||
await self.stop_ttfb_metrics()
|
||||
yield TTSAudioRawFrame(
|
||||
audio=chunk,
|
||||
sample_rate=self.sample_rate,
|
||||
num_channels=1,
|
||||
)
|
||||
|
||||
except Exception as e:
|
||||
# ================================================================================
|
||||
# STEP 10: ERROR HANDLING
|
||||
# ================================================================================
|
||||
# Log any unexpected errors and notify the pipeline
|
||||
logger.error(f"{self} exception: {e}")
|
||||
await self.push_error(ErrorFrame(f"Error generating TTS: {e}"))
|
||||
finally:
|
||||
# ================================================================================
|
||||
# STEP 11: CLEANUP AND COMPLETION
|
||||
# ================================================================================
|
||||
# Always stop metrics tracking, even if errors occurred
|
||||
await self.stop_ttfb_metrics()
|
||||
|
||||
# Signal to pipeline that TTS generation is complete
|
||||
# This allows downstream processors to finalize audio processing
|
||||
yield TTSStoppedFrame()
|
||||
# ========================================================
|
||||
# YIELD AUDIO FRAME TO PIPELINE
|
||||
# ========================================================
|
||||
# Only yield frames with actual audio content
|
||||
# Empty frames can cause pipeline issues
|
||||
if len(audio_data) > 0:
|
||||
# Create Pipecat audio frame with processed audio data
|
||||
yield TTSAudioRawFrame(
|
||||
audio=audio_data, # Clean audio without headers
|
||||
sample_rate=self.sample_rate, # Configured sample rate (48kHz)
|
||||
num_channels=1, # Mono audio
|
||||
)
|
||||
|
||||
Reference in New Issue
Block a user