Stream multi-channel audio to Amazon Transcribe using the Web Audio API | AWS Machine Learning Blog

Jun 10, 2025

Multi-channel transcription streaming is a feature of Amazon Transcribe that can be used in many cases with a web browser. Creating this stream source has it challenges, but with the JavaScript Web Audio API, you can connect and combine different audio sources like videos, audio files, or hardware like microphones to obtain transcripts.

In this post, we guide you through how to use two microphones as audio sources, merge them into a single dual-channel audio, perform the required encoding, and stream it to Amazon Transcribe. A Vue.js application source code is provided that requires two microphones connected to your browser. However, the versatility of this approach extends far beyond this use case—you can adapt it to accommodate a wide range of devices and audio sources.

With this approach, you can get transcripts for two sources in a single Amazon Transcribe session, offering cost savings and other benefits compared to using a separate session for each source.

For our use case, using a single-channel stream for two microphones and enabling Amazon Transcribe speaker label identification to identify the speakers might be enough, but there are a few considerations:

By using two audio sources with microphones, you can address these concerns by making sure each transcription is from a fixed input source. By assigning a device to a speaker, our application knows in advance which transcript to use. However, you might still encounter voice overlapping if two nearby microphones are picking up multiple voices. This can be mitigated by using directional microphones, volume management, and Amazon Transcribe word-level confidence scores.

The following diagram illustrates the solution workflow.

Application diagram for two microphones

We use two audio inputs with the Web Audio API. With this API, we can merge the two inputs, Mic A and Mic B, into a single audio data source, with the left channel representing Mic A and the right channel representing Mic B.

Then, we convert this audio source to PCM (Pulse-Code Modulation) audio. PCM is a common format for audio processing, and it’s one of the formats required by Amazon Transcribe for the audio input. Finally, we stream the PCM audio to Amazon Transcribe for transcription.

You should have the following prerequisites in place:

Complete the following steps to launch the application:

Application running on http://localhost:5173 with two connected microphones

In this section, we examine the important code pieces for the implementation:

The result object will include a ChannelId property that you can use to identify your microphone source, such as ch_0 and ch_1, respectively.

Audio Worklets can execute in a separate thread to provide very low-latency audio processing. The implementation and demo source code can be found in the public/worklets/recording-processor.js file.

For our case, we use the Worklet to perform two main tasks:

The general code structure to implement this is as follows:

You can pass custom options to this Worklet instance using the processorOptions attribute. In our demo, we set a maxFrameCount: (SAMPLE_RATE * 4) / 10 as a bitrate guide to determine when to emit a new message payload. A message is for example:

One of the most important sections is how to encode to PCM for two channels. Following the AWS documentation in the Amazon Transcribe API Reference, the AudioChunk is defined by: Duration (s) * Sample Rate (Hz) * Number of Channels * 2. For two channels, 1 second at 16000Hz is: 1 * 16000 * 2 * 2 = 64000 bytes. Our encoding function it should then look like this:

For more information how the audio data blocks are handled, see AudioWorkletProcessor: process() method. For more information on PCM format encoding, see Multimedia Programming Interface and Data Specifications 1.0.

In this post, we explored the implementation details of a web application that uses the browser’s Web Audio API and Amazon Transcribe streaming to enable real-time dual-channel transcription. By using the combination of AudioContext, ChannelMergerNode, and AudioWorklet, we were able to seamlessly process and encode the audio data from two microphones before sending it to Amazon Transcribe for transcription. The use of the AudioWorklet in particular allowed us to achieve low-latency audio processing, providing a smooth and responsive user experience.

You can build upon this demo to create more advanced real-time transcription applications that cater to a wide range of use cases, from meeting recordings to voice-controlled interfaces.

Try out the solution for yourself, and leave your feedback in the comments.

Jorge Lanzarotti is a Sr. Prototyping SA at Amazon Web Services (AWS) based on Tokyo, Japan. He helps customers in the public sector by creating innovative solutions to challenging problems.

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