I am building a proof-of-concept Android application that behaves like an “audio modem,” pushing and pulling live audio through a WebSocket connection with the absolute minimum possible latency. The primary scene is real-time audio streaming, so every design decision should favour tight buffering, fast reconnection and clean, uninterrupted playback. Here is what I need from you: • A lightweight Android app (Java or Kotlin) that opens a configurable WebSocket endpoint, captures audio We need an Android app that runs on a **Redmi 4A (Android 7)** device the app will capture audio from the headset mic via loopback cable and send it to our backend over **WebSocket**, using **Opus encoding**. Incoming audio from WebSocket must be decoded and played back into the headset output it is a **low-latency audio modem**. --- ## **Device & Platform Constraints** * Device: **Redmi 4A** * Android version: **7.x (Nougat)** * Audio routing will be enforced by our **Magisk module** (already handled by another developer) * SIM call audio is already forced to wired headset through 3.5mm TRRS loopback cable --- ## **What the App Must Do** ### **A. WebSocket Audio Transport** * Connect to backend via **secure WebSocket (wss://)** * Send **uplink audio**: captured mic → Opus → WebSocket * Receive **downlink audio**: WebSocket → Opus → AudioTrack * Must support: * auto reconnect * error reporting * configurable server URL * configurable device ID ### **B. Audio Capture & Playback** App will use **native Android audio APIs**, NOT WebRTC: * **Capture**: * `AudioRecord` * 8000 Hz, mono, 16-bit PCM * converted to Opus (frame size 20ms) * **Playback**: * `AudioTrack` * 8000 Hz, mono, 16-bit PCM (decoded from Opus) ### **C. Call State Driven** App must detect SIM call states using `TelephonyManager`: * When SIM enters **OFFHOOK**: * Connect WebSocket (if disconnected) * Start streaming audio uplink/downlink * When SIM returns to **IDLE**: * Stop streaming * Option to close or keep socket alive (configurable) ### **D. Codec** Use **Opus** (NOT WebRTC stack, NOT WebRTC AEC/NS/AGC) Codec profile: * Sample rate: **8000 Hz** * Channels: **mono** * Bitrate target: **12–16 kbps** * Frame size: **20 ms** * Transport: **binary frames over WebSocket** ### **E. Permissions** App needs: * `RECORD_AUDIO` * `MODIFY_AUDIO_SETTINGS` * `READ_PHONE_STATE` * `INTERNET` * `WAKE_LOCK` * (optional) `RECEIVE_BOOT_COMPLETED` for auto-start --- --- ## **Backend Compatibility** Backend will accept frames as: ``` [frame_header][opus_payload] ``` And will respond with similar frames. Exact header spec will be provided. --- ## **UI Requirements** Minimal UI: Main screen shows: * Device ID * WebSocket state: DISCONNECTED / CONNECTING / CONNECTED * Call state: IDLE / OFFHOOK * Last error / reconnect attempts Settings screen: * WebSocket URL * Device ID * Toggle: “Start on boot” * Toggle: “Disconnect when no call” * Toggle: “Auto reconnect” --- ## **Deliverables** Freelancer must deliver: 1. **Android Studio project** 2. **Signed APK (debug OK)** 3. **Source code on GitHub or zip** 4. **Simple README** with: * build steps * required permissions * run instructions 5. **Test video** demonstrating: * outgoing SIM call → audio uplink/downlink over WebSocket * incoming SIM call → same behavior * disconnect/reconnect handling * call end handling --- ## **Bonus Points (Not Required in v1)** If you can do these, indicate separately: * Background foreground service mode * Network status awareness * Audio buffer jitter smoothing * Automatic Opus bitrate adaptation * TLS pinning for WebSocket --- * Backend team that: * handles SIP/RTP/VoIP * handles AI voice * handles IVR logic So you do **front-end audio only**. --- ## **Required Skills** Developer **must have**: * Android native audio (`AudioRecord` + `AudioTrack`) * Opus codec integration (JNI or NDK) * WebSocket client experience (OkHttp / Java-WebSocket, etc.) * Telephony API (`PhoneStateListener`) * Java/Kotlin (either OK) * Good debugging/logging practice --- ## **Please Include in Proposal** * Past experience with audio apps (very important) * Whether you prefer Java or Kotlin * Choice of Opus implementation: * `libopus` via JNI (preferred), or * pure-Java Opus (acceptable if stable) --- ## **Project Type & Duration** * 1 developer * Estimated v1 timeline: **1 week * Possible follow-on work: * jitter buffer * dual SIM * encryption * monitoring * provisioning 100+ nodes