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Audio files must be encoded and packaged in formats that balance quality, size, and compatibility. Consistent encoding parameters ensure accurate recognition and low latency across both synchronous and asynchronous workflows. The API supports both containerized audio formats (such as Ogg and WebM) as well as raw PCM audio streams.
Please ensure your audio files conform to the specifications listed below. Let us know if you need help with audio formatting or API request configuration.

Supported Audio Formats

Container-based Audio

The following audio containers and their associated codecs are supported by the Corti API:
This parameter is optional but recommended
The audioFormat parameter can be defined in transcribe and streams configuration to declare the audio format the speech to text system should expect in the incoming audio stream.For container formats (audio/ogg, audio/webm), you can optionally specify a codec parameter. Allowed codecs are opus and vorbis.Examples:
WAV files are supported for upload to the /recordings endpoint, but raw PCM audio should follow approach outlined below.

Raw Audio

Raw pulse code modulation (PCM) audio is supported when rate, channels, and bits parameters are defined in configuration.
This parameter is required for use with raw PCM audio
The audioFormat parameter can be defined in transcribe and streams configuration to declare the audio format the speech to text system should expect in the incoming audio stream.For raw audio (audio/pcm), the parameters rate, channels, and bits must be defined.Examples:
When using Raw PCM audio, 16-bit little-endian mono at 16 kHz is recommended.

Audio streaming recommendations

Sample rate of 16 kHz

Captures the full range of human speech frequencies, with higher rates offering negligible recognition benefit but increasing computational cost

Audio chunk size of 250 milliseconds

Optimal speed to support both dictation and AI scribing workflows, with sending much smaller chunks more frequently can degrade recognition accuracy without improving latency

Stream at real-time speed

Audio should be streamed at or near real-time speed. Streaming audio faster than real time is not recommended and may cause buffering issues, degraded results, or stream termination. Pace audio chunks according to their actual audio duration.

Microphone Configuration

Dictation

Ambient Conversation

Maintain average loudness around –12 dBFS RMS with peaks near –3 dBFS for optimal speech-to-text normalization.

Channel Configuration

Choosing the right channel configuration ensures accurate transcription, speaker separation, and diarization across different use cases.
Mono input supports transcription with diarization; however, speaker separation may be unreliable when there is not clear turn-taking in the dialogue.Multichannel input (two audio channels, one per participant, in telehealth workflow) provides opportunity for improved speaker separation and labeling.

Streams Endpoint

Transcripts Endpoint

Additional Notes

  • Enabling diarization is typically only required on mono audio.
  • Mono audio with diarization disabled will produce transcripts with one channel (-1), whereas diarized-mono transcripts will have two channels (0, 1).
  • For multichannel audio, each channel should capture only one speaker’s microphone feed in order to avoid cross-talk or echo between channels.
  • Keep all channels aligned in time; do not trim or delay audio streams independently.
  • Ensure each channel contains only one participant’s feed to avoid duplicated transcript content.
  • Recommended capture format is 16-bit / 16 kHz PCM

Please contact us if you need more information about supported audio formats or are having issues processing an audio file.Additional references and resources: