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Null safety support #7

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Null safety support #7

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38 changes: 19 additions & 19 deletions lib/flutter_live.dart
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Original file line number Diff line number Diff line change
Expand Up @@ -110,17 +110,17 @@ class RealtimePlayer {
/// streamUrl: "webrtc://d.ossrs.net:11985/live/livestream"
class WebRTCUri {
/// The api server url for WebRTC streaming.
String api;
String? api;
/// The stream url to play or publish.
String streamUrl;
String? streamUrl;

/// Parse the url to WebRTC uri.
static WebRTCUri parse(String url) {
Uri uri = Uri.parse(url);

var schema = 'https'; // For native, default to HTTPS
if (uri.queryParameters.containsKey('schema')) {
schema = uri.queryParameters['schema'];
schema = uri.queryParameters['schema']!;
} else {
schema = 'https';
}
Expand All @@ -134,7 +134,7 @@ class WebRTCUri {

var api = '/rtc/v1/play/';
if (uri.queryParameters.containsKey('play')) {
api = uri.queryParameters['play'];
api = uri.queryParameters['play']!;
}

var apiParams = [];
Expand All @@ -159,8 +159,8 @@ class WebRTCUri {

/// A WebRTC player, using [flutter_webrtc](https://pub.dev/packages/flutter_webrtc)
class WebRTCPlayer {
webrtc.AddStreamCallback _onRemoteStream;
webrtc.RTCPeerConnection _pc;
webrtc.AddStreamCallback? _onRemoteStream;
webrtc.RTCPeerConnection? _pc;

/// When got a remote stream.
set onRemoteStream(webrtc.AddStreamCallback v) {
Expand All @@ -175,7 +175,7 @@ class WebRTCPlayer {
/// [url] must a path parsed by [WebRTCUri.parse] in https://github.com/rtcdn/rtcdn-draft
Future<void> play(String url) async {
if (_pc != null) {
await _pc.close();
await _pc!.close();
}

// Create the peer connection.
Expand All @@ -187,36 +187,36 @@ class WebRTCPlayer {
print('WebRTC: createPeerConnection done');

// Setup the peer connection.
_pc.onAddStream = (webrtc.MediaStream stream) {
_pc!.onAddStream = (webrtc.MediaStream stream) {
print('WebRTC: got stream ${stream.id}');
if (_onRemoteStream == null) {
print('Warning: Stream ${stream.id} is leak');
return;
}
_onRemoteStream(stream);
_onRemoteStream!(stream);
};

_pc.addTransceiver(
_pc!.addTransceiver(
kind: webrtc.RTCRtpMediaType.RTCRtpMediaTypeAudio,
init: webrtc.RTCRtpTransceiverInit(direction: webrtc.TransceiverDirection.RecvOnly),
);
_pc.addTransceiver(
_pc!.addTransceiver(
kind: webrtc.RTCRtpMediaType.RTCRtpMediaTypeVideo,
init: webrtc.RTCRtpTransceiverInit(direction: webrtc.TransceiverDirection.RecvOnly),
);
print('WebRTC: Setup PC done, A|V RecvOnly');

// Start SDP handshake.
webrtc.RTCSessionDescription offer = await _pc.createOffer({
webrtc.RTCSessionDescription offer = await _pc!.createOffer({
'mandatory': {'OfferToReceiveAudio': true, 'OfferToReceiveVideo': true},
});
await _pc.setLocalDescription(offer);
print('WebRTC: createOffer, ${offer.type} is ${offer.sdp.replaceAll('\n', '\\n').replaceAll('\r', '\\r')}');
await _pc!.setLocalDescription(offer);
print('WebRTC: createOffer, ${offer.type} is ${offer.sdp!.replaceAll('\n', '\\n').replaceAll('\r', '\\r')}');

webrtc.RTCSessionDescription answer = await _handshake(url, offer.sdp);
print('WebRTC: got ${answer.type} is ${answer.sdp.replaceAll('\n', '\\n').replaceAll('\r', '\\r')}');
webrtc.RTCSessionDescription answer = await _handshake(url, offer.sdp!);
print('WebRTC: got ${answer.type} is ${answer.sdp!.replaceAll('\n', '\\n').replaceAll('\r', '\\r')}');

await _pc.setRemoteDescription(answer);
await _pc!.setRemoteDescription(answer);
}

/// Handshake to exchange SDP, send offer and got answer.
Expand All @@ -238,7 +238,7 @@ class WebRTCPlayer {
// {api: "xxx", sdp: "offer", streamurl: "webrtc://d.ossrs.net:11985/live/livestream"}
// Response:
// {code: 0, sdp: "answer", sessionid: "007r51l7:X2Lv"}
HttpClientRequest req = await client.postUrl(Uri.parse(uri.api));
HttpClientRequest req = await client.postUrl(Uri.parse(uri.api!));
req.headers.set('Content-Type', 'application/json');
req.add(utf8.encode(json.encode({'api': uri.api, 'streamurl': uri.streamUrl, 'sdp': offer})));
print('WebRTC request: ${uri.api} offer=${offer.length}B');
Expand All @@ -261,7 +261,7 @@ class WebRTCPlayer {
/// Dispose the player.
void dispose() {
if (_pc != null) {
_pc.close();
_pc!.close();
}
}
}
Expand Down
8 changes: 6 additions & 2 deletions pubspec.yaml
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Original file line number Diff line number Diff line change
Expand Up @@ -4,15 +4,19 @@ description: Live streaming player, iOS+Android, RTMP/HTTP-FLV/HLS/WebRTC, by Fl
homepage: https://github.com/ossrs/flutter_live

environment:
sdk: ">=2.7.0 <3.0.0"
sdk: ">=2.12.0 <3.0.0"
flutter: ">=1.20.0 <2.0.0"

dependencies:
flutter:
sdk: flutter
fijkplayer: ^0.8.8
flutter_webrtc: ^0.6.3
camera_with_rtmp: ^0.3.2
camera_with_rtmp:
git:
url: https://github.com/babulpatel1309/flutter_rtmppublisher.git
ref: 963c0b82cd94f09f1e4387d33a89e3b067957c19


dev_dependencies:
flutter_test:
Expand Down