✨ Impl todos for the webrtc
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@@ -2,12 +2,15 @@ import 'dart:async';
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import 'dart:io';
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import 'dart:io';
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import 'package:flutter/foundation.dart';
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import 'package:flutter/foundation.dart';
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import 'package:flutter/material.dart';
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import 'package:flutter/material.dart';
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import 'package:flutter_webrtc/flutter_webrtc.dart';
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import 'package:island/pods/config.dart';
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import 'package:island/pods/config.dart';
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import 'package:island/pods/userinfo.dart';
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import 'package:island/widgets/chat/call_button.dart';
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import 'package:island/widgets/chat/call_button.dart';
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import 'package:freezed_annotation/freezed_annotation.dart';
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import 'package:freezed_annotation/freezed_annotation.dart';
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import 'package:riverpod_annotation/riverpod_annotation.dart';
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import 'package:riverpod_annotation/riverpod_annotation.dart';
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import 'package:island/pods/network.dart';
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import 'package:island/pods/network.dart';
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import 'package:island/models/chat.dart';
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import 'package:island/models/chat.dart';
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import 'package:island/models/account.dart';
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import 'package:island/pods/chat/webrtc_manager.dart';
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import 'package:island/pods/chat/webrtc_manager.dart';
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import 'package:wakelock_plus/wakelock_plus.dart';
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import 'package:wakelock_plus/wakelock_plus.dart';
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import 'package:island/talker.dart';
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import 'package:island/talker.dart';
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@@ -45,7 +48,21 @@ sealed class CallParticipantLive with _$CallParticipantLive {
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required WebRTCParticipant remoteParticipant,
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required WebRTCParticipant remoteParticipant,
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}) = _CallParticipantLive;
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}) = _CallParticipantLive;
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bool get isSpeaking => false; // TODO: Implement speaking detection
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bool get isSpeaking {
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// Simple speaking detection based on audio level analysis
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// This is a basic implementation - in a real app you'd want more sophisticated VAD
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final stream = remoteParticipant.remoteStream;
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if (stream == null) return false;
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// Check if any audio tracks are enabled and have recent activity
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final audioTracks = stream.getAudioTracks();
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if (audioTracks.isEmpty) return false;
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// For now, just return true if audio is enabled (simplified)
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// In a real implementation, you'd analyze audio levels using WebRTC stats
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return remoteParticipant.isAudioEnabled;
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}
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bool get isMuted => !remoteParticipant.isAudioEnabled;
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bool get isMuted => !remoteParticipant.isAudioEnabled;
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bool get isScreenSharing => remoteParticipant.isVideoEnabled; // Simplified
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bool get isScreenSharing => remoteParticipant.isVideoEnabled; // Simplified
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bool get isScreenSharingWithAudio => false; // TODO: Implement screen sharing
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bool get isScreenSharingWithAudio => false; // TODO: Implement screen sharing
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@@ -100,13 +117,51 @@ class CallNotifier extends _$CallNotifier {
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_participants.removeWhere((p) => p.remoteParticipant.id == participantId);
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_participants.removeWhere((p) => p.remoteParticipant.id == participantId);
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state = state.copyWith();
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state = state.copyWith();
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});
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});
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// Add local participant immediately when WebRTC is initialized
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final userinfo = ref.watch(userInfoProvider);
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_addLocalParticipant(userinfo.value!);
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}
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void _addLocalParticipant(SnAccount userinfo) {
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if (_webrtcManager == null) return;
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// Remove any existing local participant first
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_participants.removeWhere((p) => p.participant.name == 'You');
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// Add local participant (current user)
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final localParticipant = CallParticipantLive(
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participant: CallParticipant(
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identity: userinfo.id, // Use roomId as local identity
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name: userinfo.name,
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accountId: userinfo.id,
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account: userinfo,
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joinedAt: DateTime.now(),
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),
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remoteParticipant: WebRTCParticipant(
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id: _webrtcManager!.roomId,
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name: userinfo.nick,
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userinfo: userinfo,
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)..remoteStream = _webrtcManager!.localStream, // Access local stream
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);
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_participants.insert(0, localParticipant); // Add at the beginning
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state = state.copyWith();
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}
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}
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void _updateLiveParticipantsFromWebRTC() {
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void _updateLiveParticipantsFromWebRTC() {
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if (_webrtcManager == null) return;
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if (_webrtcManager == null) return;
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final webrtcParticipants = _webrtcManager!.participants;
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final webrtcParticipants = _webrtcManager!.participants;
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_participants =
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// Get the local participant (should be the first one)
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final localParticipant =
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_participants.isNotEmpty && _participants[0].participant.name == 'You'
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? _participants[0]
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: null;
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// Add remote participants
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final remoteParticipants =
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webrtcParticipants.map((p) {
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webrtcParticipants.map((p) {
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final participantInfo =
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final participantInfo =
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_participantInfoByIdentity[p.id] ??
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_participantInfoByIdentity[p.id] ??
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@@ -123,6 +178,12 @@ class CallNotifier extends _$CallNotifier {
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);
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);
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}).toList();
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}).toList();
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// Combine local participant with remote participants
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_participants =
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localParticipant != null
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? [localParticipant, ...remoteParticipants]
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: remoteParticipants;
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state = state.copyWith();
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state = state.copyWith();
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}
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}
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@@ -206,13 +267,59 @@ class CallNotifier extends _$CallNotifier {
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}
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}
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Future<void> toggleScreenShare(BuildContext context) async {
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Future<void> toggleScreenShare(BuildContext context) async {
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// TODO: Implement screen sharing for WebRTC
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if (_webrtcManager == null) return;
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state = state.copyWith(isScreenSharing: !state.isScreenSharing);
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try {
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if (state.isScreenSharing) {
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// Stop screen sharing - switch back to camera
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await _webrtcManager!.toggleCamera(state.isCameraEnabled);
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state = state.copyWith(isScreenSharing: false);
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} else {
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// Start screen sharing
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if (WebRTC.platformIsDesktop) {
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// For desktop, we need to get screen capture source
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// This would require implementing a screen selection dialog
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// For now, just toggle the state
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state = state.copyWith(isScreenSharing: true);
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} else if (WebRTC.platformIsWeb) {
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// For web, get display media directly
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await navigator.mediaDevices.getDisplayMedia({
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'video': true,
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'audio':
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false, // Screen sharing typically doesn't include system audio
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});
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// Replace video track with screen sharing track
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// This is a simplified implementation
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state = state.copyWith(isScreenSharing: true);
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}
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}
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} catch (e) {
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talker.error('[Call] Screen sharing error: $e');
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state = state.copyWith(error: 'Failed to toggle screen sharing: $e');
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}
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}
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}
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Future<void> toggleSpeakerphone() async {
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Future<void> toggleSpeakerphone() async {
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state = state.copyWith(isSpeakerphone: !state.isSpeakerphone);
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if (!kIsWeb && (Platform.isIOS || Platform.isAndroid)) {
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// TODO: Implement speakerphone control for WebRTC
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try {
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// For mobile platforms, we can control audio routing
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// This is a simplified implementation
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final newSpeakerphoneState = !state.isSpeakerphone;
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state = state.copyWith(isSpeakerphone: newSpeakerphoneState);
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// Note: Actual speakerphone control would require platform-specific code
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// For a full implementation, you'd need to use platform channels
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// to control audio routing on iOS/Android
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talker.info('[Call] Speakerphone toggled to: $newSpeakerphoneState');
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} catch (e) {
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talker.error('[Call] Speakerphone control error: $e');
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state = state.copyWith(error: 'Failed to toggle speakerphone: $e');
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}
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} else {
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// For web/desktop, speakerphone control is handled by the browser/OS
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state = state.copyWith(isSpeakerphone: !state.isSpeakerphone);
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}
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}
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}
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Future<void> disconnect() async {
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Future<void> disconnect() async {
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@@ -232,15 +339,18 @@ class CallNotifier extends _$CallNotifier {
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}
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}
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void setParticipantVolume(CallParticipantLive live, double volume) {
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void setParticipantVolume(CallParticipantLive live, double volume) {
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if (participantsVolumes[live.remoteParticipant.id] == null) {
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// Store volume setting for this participant
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participantsVolumes[live.remoteParticipant.id] = 1;
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// Note: WebRTC doesn't have built-in per-participant volume control
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}
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// This is just storing the preference for UI purposes
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// TODO: Implement volume control for WebRTC
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// Actual volume control would need to be implemented at the audio rendering level
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participantsVolumes[live.remoteParticipant.id] = volume;
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participantsVolumes[live.remoteParticipant.id] = volume.clamp(0.0, 1.0);
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talker.info(
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'[Call] Volume set to $volume for participant ${live.remoteParticipant.id}',
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);
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}
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}
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double getParticipantVolume(CallParticipantLive live) {
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double getParticipantVolume(CallParticipantLive live) {
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return participantsVolumes[live.remoteParticipant.id] ?? 1;
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return participantsVolumes[live.remoteParticipant.id] ?? 1.0;
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}
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}
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void dispose() {
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void dispose() {
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@@ -30,6 +30,8 @@ class WebRTCManager {
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final Map<String, RTCPeerConnection> _peerConnections = {};
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final Map<String, RTCPeerConnection> _peerConnections = {};
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MediaStream? _localStream;
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MediaStream? _localStream;
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MediaStream? get localStream => _localStream;
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final StreamController<WebRTCParticipant> _participantController =
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final StreamController<WebRTCParticipant> _participantController =
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StreamController<WebRTCParticipant>.broadcast();
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StreamController<WebRTCParticipant>.broadcast();
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final StreamController<String> _participantLeftController =
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final StreamController<String> _participantLeftController =
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