communitas-core 0.1.18

// Copyright (c) 2025 Saorsa Labs Limited
//
// This file is part of the Communitas P2P collaboration platform.
//
// Licensed under the GPL-3.0 license

//! WebRTC Service Integration
//!
//! Provides a high-level WebRTC service for Communitas that integrates
//! with the gossip overlay network for signaling and peer discovery.

use super::gossip_signaling::GossipSignalingTransport;
use super::identity::CommunitasIdentity;
use crate::gossip::GossipContext;
use anyhow::{Result, anyhow};
use saorsa_webrtc::types::{CallEvent, CallId, MediaConstraints};
use std::collections::HashMap;
use std::sync::Arc;
use tokio::sync::{RwLock, broadcast};
use tracing::{debug, info, warn};

/// Active call state
#[derive(Debug, Clone)]
pub struct CallState {
    /// Call ID
    pub call_id: CallId,
    /// Target peer identity
    pub target: CommunitasIdentity,
    /// Media constraints
    pub constraints: MediaConstraints,
    /// Is video currently enabled
    pub is_video_enabled: bool,
    /// Is audio currently enabled
    pub is_audio_enabled: bool,
    /// Is screen sharing active
    pub is_screen_sharing: bool,
}

/// Media device information
#[derive(Debug, Clone, serde::Serialize, serde::Deserialize)]
pub struct MediaDevice {
    /// Device ID
    pub device_id: String,
    /// Human-readable label
    pub label: String,
    /// Device kind (audioinput, audiooutput, videoinput)
    pub kind: String,
}

/// WebRTC service for Communitas
///
/// Manages WebRTC calls over the gossip overlay network, providing
/// voice, video, and screen sharing capabilities.
pub struct CommunitasWebRtcService {
    /// Signaling transport
    signaling: Arc<GossipSignalingTransport>,

    /// Local identity
    local_identity: CommunitasIdentity,

    /// Event broadcaster
    event_tx: broadcast::Sender<CallEvent<CommunitasIdentity>>,

    /// Active calls
    active_calls: Arc<RwLock<HashMap<CallId, CallState>>>,
}

impl CommunitasWebRtcService {
    /// Create a new WebRTC service
    ///
    /// # Arguments
    /// * `gossip` - The gossip context
    pub fn new(gossip: Arc<GossipContext>) -> Result<Self> {
        info!("Initializing Communitas WebRTC service");

        // Create signaling transport
        let signaling = Arc::new(GossipSignalingTransport::new(gossip.clone())?);

        // Get local identity
        let local_identity = CommunitasIdentity::new(gossip.four_words.clone())?;

        // Create event broadcaster
        let (event_tx, _) = broadcast::channel(100);

        // Initialize active calls tracking
        let active_calls = Arc::new(RwLock::new(HashMap::new()));

        Ok(Self {
            signaling,
            local_identity,
            event_tx,
            active_calls,
        })
    }

    /// Start the WebRTC service
    ///
    /// Subscribes to signaling messages and begins listening for incoming calls.
    pub async fn start(&self) -> Result<()> {
        info!("Starting WebRTC service for {}", self.local_identity);

        // Subscribe to signaling messages
        self.signaling.subscribe_to_signaling().await?;

        debug!("WebRTC service started successfully");

        Ok(())
    }

    /// Initiate a call to another peer
    ///
    /// # Arguments
    /// * `target_four_words` - Four-word address of the peer to call
    /// * `constraints` - Media constraints (audio, video, screen share)
    ///
    /// # Returns
    /// The call ID for the initiated call
    pub async fn initiate_call(
        &self,
        target_four_words: &str,
        constraints: MediaConstraints,
    ) -> Result<CallId> {
        info!(
            "Initiating call to {} with constraints: {:?}",
            target_four_words, constraints
        );

        // Create target identity
        let target = CommunitasIdentity::new(target_four_words.to_string())?;

        // Generate call ID
        let call_id = CallId::new();

        // Create call state
        let call_state = CallState {
            call_id,
            target: target.clone(),
            constraints: constraints.clone(),
            is_video_enabled: constraints.has_video(),
            is_audio_enabled: constraints.has_audio(),
            is_screen_sharing: false,
        };

        // Store call state
        {
            let mut calls = self.active_calls.write().await;
            calls.insert(call_id, call_state);
        }

        // TODO: Implement actual call initiation using saorsa-webrtc
        // This would involve:
        // 1. Creating an SDP offer
        // 2. Sending it via the signaling transport
        // 3. Waiting for the answer
        // 4. Establishing the QUIC connection
        // 5. Setting up media streams

        debug!("Created call {} to {}", call_id, target);

        // Emit call initiated event
        let event = CallEvent::CallInitiated {
            call_id,
            callee: target,
            constraints,
        };
        let _ = self.event_tx.send(event);

        Ok(call_id)
    }

    /// Accept an incoming call
    ///
    /// # Arguments
    /// * `call_id` - The ID of the call to accept
    pub async fn accept_call(&self, call_id: CallId) -> Result<()> {
        info!("Accepting call {}", call_id);

        // TODO: Implement actual call acceptance
        // This would involve:
        // 1. Creating an SDP answer
        // 2. Sending it via the signaling transport
        // 3. Establishing the QUIC connection
        // 4. Setting up media streams

        debug!("Call {} accepted", call_id);

        Ok(())
    }

    /// Reject an incoming call
    ///
    /// # Arguments
    /// * `call_id` - The ID of the call to reject
    pub async fn reject_call(&self, call_id: CallId) -> Result<()> {
        info!("Rejecting call {}", call_id);

        // TODO: Implement call rejection signaling

        debug!("Call {} rejected", call_id);

        let event = CallEvent::CallRejected { call_id };
        let _ = self.event_tx.send(event);

        Ok(())
    }

    /// End an active call
    ///
    /// # Arguments
    /// * `call_id` - The ID of the call to end
    pub async fn end_call(&self, call_id: CallId) -> Result<()> {
        info!("Ending call {}", call_id);

        // Remove call from active calls
        {
            let mut calls = self.active_calls.write().await;
            if calls.remove(&call_id).is_none() {
                warn!("Attempted to end non-existent call {}", call_id);
                return Err(anyhow!("Call not found"));
            }
        }

        // TODO: Implement call termination
        // This would involve:
        // 1. Sending call end signaling message
        // 2. Closing media streams
        // 3. Cleaning up QUIC connection

        debug!("Call {} ended", call_id);

        let event = CallEvent::CallEnded { call_id };
        let _ = self.event_tx.send(event);

        Ok(())
    }

    /// Enable or disable video in an active call
    ///
    /// # Arguments
    /// * `call_id` - The ID of the call
    /// * `enabled` - Whether to enable or disable video
    pub async fn set_video_enabled(&self, call_id: CallId, enabled: bool) -> Result<()> {
        info!("Setting video enabled={} for call {}", enabled, call_id);

        // Update call state
        {
            let mut calls = self.active_calls.write().await;
            let call = calls
                .get_mut(&call_id)
                .ok_or_else(|| anyhow!("Call not found"))?;

            call.is_video_enabled = enabled;
        }

        // TODO: Implement actual video track control
        // This would involve controlling the video MediaStreamTrack

        debug!(
            "Video {} for call {}",
            if enabled { "enabled" } else { "disabled" },
            call_id
        );

        Ok(())
    }

    /// Enable or disable audio in an active call
    ///
    /// # Arguments
    /// * `call_id` - The ID of the call
    /// * `enabled` - Whether to enable or disable audio (mute/unmute)
    pub async fn set_audio_enabled(&self, call_id: CallId, enabled: bool) -> Result<()> {
        info!("Setting audio enabled={} for call {}", enabled, call_id);

        // Update call state
        {
            let mut calls = self.active_calls.write().await;
            let call = calls
                .get_mut(&call_id)
                .ok_or_else(|| anyhow!("Call not found"))?;

            call.is_audio_enabled = enabled;
        }

        // TODO: Implement actual audio track control
        // This would involve controlling the audio MediaStreamTrack

        debug!(
            "Audio {} for call {}",
            if enabled { "enabled" } else { "disabled" },
            call_id
        );

        Ok(())
    }

    /// Start screen sharing in an active call
    ///
    /// # Arguments
    /// * `call_id` - The ID of the call
    pub async fn start_screen_share(&self, call_id: CallId) -> Result<()> {
        info!("Starting screen share for call {}", call_id);

        // Update call state
        {
            let mut calls = self.active_calls.write().await;
            let call = calls
                .get_mut(&call_id)
                .ok_or_else(|| anyhow!("Call not found"))?;

            call.is_screen_sharing = true;
        }

        // TODO: Implement actual screen share
        // This would involve adding a screen share MediaStreamTrack

        debug!("Screen share started for call {}", call_id);

        Ok(())
    }

    /// Stop screen sharing in an active call
    ///
    /// # Arguments
    /// * `call_id` - The ID of the call
    pub async fn stop_screen_share(&self, call_id: CallId) -> Result<()> {
        info!("Stopping screen share for call {}", call_id);

        // Update call state
        {
            let mut calls = self.active_calls.write().await;
            let call = calls
                .get_mut(&call_id)
                .ok_or_else(|| anyhow!("Call not found"))?;

            call.is_screen_sharing = false;
        }

        // TODO: Implement actual screen share stop
        // This would involve removing the screen share MediaStreamTrack

        debug!("Screen share stopped for call {}", call_id);

        Ok(())
    }

    /// Get available media devices
    ///
    /// # Returns
    /// List of available audio and video devices
    pub async fn get_media_devices(&self) -> Result<Vec<MediaDevice>> {
        info!("Getting media devices");

        // Media device enumeration is typically done on the client side
        // The backend service doesn't have access to browser media devices
        // This method exists for API consistency but returns empty list

        debug!("Media device enumeration should be done on the client side");

        Ok(Vec::new())
    }

    /// Subscribe to call events
    ///
    /// # Returns
    /// A broadcast receiver for call events
    pub fn subscribe_events(&self) -> broadcast::Receiver<CallEvent<CommunitasIdentity>> {
        self.event_tx.subscribe()
    }

    /// Get the local identity
    pub fn local_identity(&self) -> &CommunitasIdentity {
        &self.local_identity
    }
}

#[cfg(test)]
mod tests {
    use super::*;

    // Note: Full integration tests would require setting up a complete gossip context
    // For now, these are placeholders

    #[test]
    fn test_call_id_generation() {
        let id1 = CallId::new();
        let id2 = CallId::new();
        assert_ne!(id1, id2);
    }

    #[test]
    fn test_media_constraints() {
        let audio = MediaConstraints::audio_only();
        assert!(audio.has_audio());
        assert!(!audio.has_video());

        let video = MediaConstraints::video_call();
        assert!(video.has_audio());
        assert!(video.has_video());
    }
}