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You pay by the hour for each active inference instance (HostHrs), so cost tracks how long you run the model. The dimensions split into two modes. Batch mode runs on the ml.c4.2xlarge CPU instance for processing groups of audio. Real-time mode runs on GPU instances for live speech-to-text. Real-time offers two GPU families, each in sizes from xlarge up to 16xlarge. Larger instance sizes add compute capacity and bill at higher hourly rates. You choose the instance that fits your workload and scale by selecting a bigger size or running more instances.
Top-of-mind questions for buyers
What does one HostHrs unit cover, and how is it counted?
One HostHrs equals one hour of a single running inference instance of the chosen type. You are billed for each hour that instance is active. Running two instances of the same type doubles the hourly count. Time accrues while the instance runs, regardless of how much audio you process.
Am I charged when an inference instance is stopped or idle?
Software charges accrue per active instance-hour. A fully stopped instance stops accruing software charges. Underlying AWS infrastructure fees may still apply for stopped-but-allocated resources, but the listing meters running instance time only. To stop billing, shut the instance down.
Should I pick a batch or real-time instance, and how do they differ mechanically?
The ml.c4.2xlarge batch instance runs on CPU for processing groups of recorded audio. The real-time GPU instances handle live streaming speech-to-text. Batch suits offline transcription jobs. Real-time suits interactive voice applications needing low latency. You choose based on whether audio is pre-recorded or streamed live.
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An Amazon SageMaker model package is a pre-trained machine learning model ready to use without additional training. Use the model package to create a model on Amazon SageMaker for real-time inference or batch processing. Amazon SageMaker is a fully managed platform for building, training, and deploying machine learning models at scale.
Deploy the model on Amazon SageMaker AI using the following options:
Real-time inference
Deploy the model as an API endpoint for your applications. When you send data to the endpoint, SageMaker processes it and returns results by API response. The endpoint runs continuously until you delete it. You're billed for software and SageMaker infrastructure costs while the endpoint runs. AWS Marketplace models don't support Amazon SageMaker Asynchronous Inference. For more information, see Deploy models for real-time inference .
Batch transform
Deploy the model to process batches of data stored in Amazon Simple Storage Service (Amazon S3). SageMaker runs the job, processes your data, and returns results to Amazon S3. When complete, SageMaker stops the model. You're billed for software and SageMaker infrastructure costs only during the batch job. Duration depends on your model, instance type, and dataset size. AWS Marketplace models don't support Amazon SageMaker Asynchronous Inference. For more information, see Batch transform for inference with Amazon SageMaker AI .
Version release notes
Improve inference stability in long-running sessions (over 120 seconds)
Remove 300 seconds of session limit
Improve Japanese transcription accuracy and noise robustness
Add more sample rates and audio formats
Transcribe person's names in Kata-kana
Add keywords support
Add server-side turn detection (server VAD)
Additional details
Inputs
Outputs
Usage instructions
Sample notebooks
Inputs
Summary
Model input summary
This model accepts audio data via SageMaker bidirectional streaming using JSON events over HTTP/2.
Input Event Types:
transcription_session.update - Configure the session before sending audio
input_audio_format: Audio encoding format ("pcm16", "float32", "mulaw", or "opus")
input_audio_sample_rate: Sample rate in Hz (default: 24000)
input_audio_number_of_channels: Must be 1 (mono)
input_audio_transcription.language: Source language (ISO-639-1: "ja" or "en")
input_audio_transcription.target_language: Output language (ISO-639-1: "ja" or "en")
input_audio_transcription.kana: Transcribe person's name in Kata-kana (default: false)
input_audio_transcription.keywords: keyword list (default: null)
turn_detection: false (default) or a server VAD config object ({"type": "server_vad", "silence_duration_ms": 400}) to enable automatic turn detection
turn_detection.silence_duration_ms: Trailing silence (ms) that ends a turn (default: 400; tune per application/pipeline)
input_audio_buffer.append - Send audio chunks
audio: Base64-encoded audio bytes
event_id: Optional correlation ID for tracking (max 36 characters)
input_audio_buffer.commit - Signal end of audio stream
Supported Audio Formats:
pcm16: 16-bit signed integer PCM, little-endian, mono
float32: 32-bit floating point, little-endian, mono, range [-1.0, 1.0]
mulaw: G.711 μ-law
opus: Self-contained Ogg/Opus blob per append event
Recommended Settings:
Sample rate: 24000 Hz
Chunk duration: 80ms per event
Limitations for input type
- Maximum audio per event: 8 seconds (~1MB base64 encoded)
- Supported language: ja
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