refactor(examples): modify examples for new V6 architecture #647
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Pull request overview
This PR refactors example files to align with the new V6 SDK architecture by updating method names, reorganizing file numbering to follow a structured scheme (multiples of 10 by feature area), and replacing incomplete examples with production-ready versions that demonstrate real-time audio streaming patterns.
Changes:
- Updated WebSocket method calls across examples to use new V6 API conventions (
send_text,send_media,send_settings, etc.) - Reorganized example file numbering to group features in ranges (01-09 for Auth, 10-19 for Transcription, etc.)
- Replaced basic WebSocket examples with comprehensive versions that include actual audio streaming implementation
Reviewed changes
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Show a summary per file
| File | Description |
|---|---|
| examples/README.md | Updated file numbering scheme and reorganized examples by feature area |
| examples/30-voice-agent.py | Updated method calls to use new V6 API (send_settings, send_media) |
| examples/26-transcription-live-websocket-v2.py | Removed incomplete Listen V2 example lacking audio streaming |
| examples/21-text-to-speech-streaming.py | Updated TTS WebSocket methods to V6 API (send_text, send_flush, send_close) |
| examples/14-transcription-live-websocket-v2.py | Added complete Listen V2 example with real-time audio streaming and turn detection |
| examples/13-transcription-live-websocket.py | Added complete Listen V1 example with audio streaming and finalization |
| examples/07-transcription-live-websocket.py | Removed incomplete Listen V1 example |
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…-ready patterns Reorganize all example files with a more scalable numbering system organized by feature area: - 01-09: Authentication - 10-19: Transcription (Listen) - 20-29: Text-to-Speech (Speak) - 30-39: Voice Agent - 40-49: Text Intelligence (Read) - 50-59: Management API - 60-69: On-Premises - 70-79: Configuration & Advanced Changes: - Renamed all examples to follow new numbering scheme - Updated WebSocket examples (13, 14) with production-ready streaming patterns - Removed artificial delays that don't reflect real usage - Simplified to straightforward file streaming approach - Added clear async implementation examples in comments - Updated README.md to reflect new organization The new numbering makes it easier to add future examples without renumbering existing ones.
Remove trailing whitespace and format code consistently in WebSocket streaming examples.
Update all WebSocket examples to use the correct method names: Listen V1/V2: - send_media() instead of send_listen_v_1_media() or send_listen_v_2_media() Speak V1: - send_text() instead of send_speak_v_1_text() - send_flush() instead of send_speak_v_1_flush() - send_close() instead of send_speak_v_1_close() Agent V1: - send_settings() instead of send_agent_v_1_settings() - send_media() instead of send_agent_v_1_media() Updated in examples: - 13-transcription-live-websocket.py - 14-transcription-live-websocket-v2.py - 21-text-to-speech-streaming.py - 30-voice-agent.py
…xamples - Add chunk delay calculation to simulate microphone audio streaming - Refactor audio sending into background thread functions - Align v2 example chunking behavior with v1 example - Improve async examples with proper streaming delays
- Add minimal connection example (30-voice-agent-connection.py) - Add comprehensive interaction example (31-voice-agent-interaction.py) - Follow proper order: connect → register listeners → wait for Welcome → send settings → wait for SettingsApplied → send user message - Use valid Anthropic model (claude-sonnet-4-20250514) - Save binary audio data to output file similar to TTS examples
- Use Path.resolve() for output paths in TTS examples - Add minor formatting improvements and comments
…se structure - Access results from response.results object instead of direct attributes - Properly access sentiments, summary, topics, and intents from nested structure - Add example output for segments and improve formatting
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This pull request updates and improves several real-time audio transcription and streaming examples, primarily by enhancing WebSocket usage, updating code to use newer client APIs, and reorganizing example files for clarity. The changes include replacing older synchronous examples with more complete and production-ready versions, introducing a new advanced Listen V2 example, and updating method calls to match the latest client library conventions.
Transcription Examples Modernization
examples/13-transcription-live-websocket.pythat streams audio chunks from a file in real time, simulating microphone input, and uses updated event handling and media transmission methods. The example also demonstrates both synchronous and asynchronous usage patterns.examples/07-transcription-live-websocket.py.Advanced Listen V2 Example
examples/14-transcription-live-websocket-v2.py, demonstrating advanced conversational speech recognition with Listen V2, including contextual turn detection and strict requirements for audio format and streaming.examples/26-transcription-live-websocket-v2.py, which lacked audio streaming and contextual turn handling.API Method Updates Across Streaming Examples
examples/21-text-to-speech-streaming.py, previouslyexamples/11-text-to-speech-streaming.py) to use the latest client API conventions (send_text,send_flush,send_close) for both synchronous and asynchronous usage. [1] [2]examples/30-voice-agent.py, previouslyexamples/09-voice-agent.py) to use new API methods (send_settings,send_media) and updated async usage accordingly. [1] [2]