{"id":13607752,"url":"https://github.com/rviscarra/webrtc-speech-to-text","last_synced_at":"2025-04-12T14:31:07.298Z","repository":{"id":96353188,"uuid":"192868079","full_name":"rviscarra/webrtc-speech-to-text","owner":"rviscarra","description":"Speech transcription on the browser using WebRTC and Google Speech","archived":false,"fork":false,"pushed_at":"2019-06-24T06:44:52.000Z","size":282,"stargazers_count":121,"open_issues_count":3,"forks_count":24,"subscribers_count":6,"default_branch":"master","last_synced_at":"2024-11-07T13:37:26.285Z","etag":null,"topics":[],"latest_commit_sha":null,"homepage":null,"language":"Go","has_issues":true,"has_wiki":null,"has_pages":null,"mirror_url":null,"source_name":null,"license":"mit","status":null,"scm":"git","pull_requests_enabled":true,"icon_url":"https://github.com/rviscarra.png","metadata":{"files":{"readme":"README.md","changelog":null,"contributing":null,"funding":null,"license":"LICENSE","code_of_conduct":null,"threat_model":null,"audit":null,"citation":null,"codeowners":null,"security":null,"support":null,"governance":null,"roadmap":null,"authors":null}},"created_at":"2019-06-20T07:07:51.000Z","updated_at":"2024-10-19T07:47:52.000Z","dependencies_parsed_at":null,"dependency_job_id":"41399771-b1c1-48ea-96fd-34943a86b8b6","html_url":"https://github.com/rviscarra/webrtc-speech-to-text","commit_stats":null,"previous_names":[],"tags_count":1,"template":false,"template_full_name":null,"repository_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/rviscarra%2Fwebrtc-speech-to-text","tags_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/rviscarra%2Fwebrtc-speech-to-text/tags","releases_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/rviscarra%2Fwebrtc-speech-to-text/releases","manifests_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/rviscarra%2Fwebrtc-speech-to-text/manifests","owner_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/owners/rviscarra","download_url":"https://codeload.github.com/rviscarra/webrtc-speech-to-text/tar.gz/refs/heads/master","host":{"name":"GitHub","url":"https://github.com","kind":"github","repositories_count":248581078,"owners_count":21128099,"icon_url":"https://github.com/github.png","version":null,"created_at":"2022-05-30T11:31:42.601Z","updated_at":"2022-07-04T15:15:14.044Z","host_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub","repositories_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories","repository_names_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repository_names","owners_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/owners"}},"keywords":[],"created_at":"2024-08-01T19:01:21.303Z","updated_at":"2025-04-12T14:31:06.989Z","avatar_url":"https://github.com/rviscarra.png","language":"Go","funding_links":[],"categories":["Media API"],"sub_categories":[],"readme":"## WebRTC speech to text server\n\n### Dependencies\n\nThe speech to text server only depends on [Go 1.12](https://golang.org/doc/install).\n\n### Disclaimer\n\n**This project is only a proof of concept and SHOULDN'T be deployed on a production \nenvironment as it lacks even the most basic security measures.**\n\n### Build the project\n\nThe project includes a Makefile, to build execute:\n\n```bash\nmake\n```\n\nThis should generate a `transcribe-server` binary.\n\n### Running the server\n\nThe server receives the following flags through the command line:\n\n`--google.cred` (Required)\n\nPath to a Google Cloud credential's file. The project must have Google Speech activated.\n\n`--http.port` (Optional) \n\nSpecifies the port where the HTTP server should listen, by default the port 9000 is used.\n\n`--stun.server` (Optional)\n\nAllows to speficy a different [STUN](https://es.wikipedia.org/wiki/STUN) server, by default a Google STUN server is used.\n\n### Demo page\n\nThe demo works on Chrome 75, Firefox 67 and Safari 12.1.1\n\n![Demo screenshot](docs/demo.png)\n\nTo run the demo execute the server and navigate to `http://localhost:9000`. \n\nAfter pressing the **Start** button a dialog asking for permission to access the microphone should appear. \nAfter grating access a WebRTC connection is made to the local server, where audio data is decoded and streamed \nto Google Speech (streaming recognition).\n\nSay something and press the **Stop** button, the results (if any) should appear on screen.\n\n### Architecture\n\n![Architecture and data flow](docs/architecture.png)\n\n### Coming soon\n\n- Unit tests.\n- Be able to specify the desired language.\n- Support for interim results.\n\n### License\n\nMIT - see [LICENSE](LICENSE) for the full text.","project_url":"https://awesome.ecosyste.ms/api/v1/projects/github.com%2Frviscarra%2Fwebrtc-speech-to-text","html_url":"https://awesome.ecosyste.ms/projects/github.com%2Frviscarra%2Fwebrtc-speech-to-text","lists_url":"https://awesome.ecosyste.ms/api/v1/projects/github.com%2Frviscarra%2Fwebrtc-speech-to-text/lists"}