| /dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/blink/renderer/modules/peerconnection/ |
| H A D | rtc_rtp_sender.cc | 54 ReplaceTrackRequest(RTCRtpSender* sender, in ReplaceTrackRequest()
82 Member<RTCRtpSender> sender_;
111 Member<RTCRtpSender> sender_;
383 RTCRtpSender::RTCRtpSender(RTCPeerConnection* pc, in RTCRtpSender() function in blink::RTCRtpSender
410 MediaStreamTrack* RTCRtpSender::track() { in track()
414 RTCDtlsTransport* RTCRtpSender::transport() { in transport()
531 ScriptPromise RTCRtpSender::setParameters( in setParameters()
614 RTCDTMFSender* RTCRtpSender::dtmf() { in dtmf()
685 void RTCRtpSender::Trace(Visitor* visitor) { in Trace()
818 void RTCRtpSender::OnAudioFrameFromEncoder( in OnAudioFrameFromEncoder()
[all …]
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| H A D | rtc_rtp_transceiver.h | 25 class RTCRtpSender; variable
35 RTCRtpSender*,
40 RTCRtpSender* sender() const;
73 Member<RTCRtpSender> sender_;
|
| H A D | rtc_peer_connection.h | 76 class RTCRtpSender; variable
274 const HeapVector<Member<RTCRtpSender>>& getSenders() const;
279 RTCRtpSender* addTrack(MediaStreamTrack*, MediaStreamVector, ExceptionState&);
280 void removeTrack(RTCRtpSender*, ExceptionState&);
458 RTCRtpSender* FindSenderForTrackAndStream(MediaStreamTrack*, MediaStream*);
459 HeapVector<Member<RTCRtpSender>>::iterator FindSender(
476 RTCRtpSender* CreateOrUpdateSender(std::unique_ptr<RTCRtpSenderPlatform>,
591 HeapVector<Member<RTCRtpSender>> rtp_senders_;
|
| /dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/blink/renderer/modules/peerconnection/ |
| H A D | rtc_rtp_sender.cc | 62 ReplaceTrackRequest(RTCRtpSender* sender, in ReplaceTrackRequest()
90 Member<RTCRtpSender> sender_;
119 Member<RTCRtpSender> sender_;
397 RTCRtpSender::RTCRtpSender(RTCPeerConnection* pc, in RTCRtpSender() function in blink::RTCRtpSender
424 MediaStreamTrack* RTCRtpSender::track() { in track()
428 RTCDtlsTransport* RTCRtpSender::transport() { in transport()
432 RTCDtlsTransport* RTCRtpSender::rtcpTransport() { in rtcpTransport()
546 ScriptPromise RTCRtpSender::setParameters( in setParameters()
630 RTCDTMFSender* RTCRtpSender::dtmf() { in dtmf()
856 void RTCRtpSender::OnAudioFrameFromEncoder( in OnAudioFrameFromEncoder()
[all …]
|
| H A D | rtc_rtp_transceiver.h | 26 class RTCRtpSender; variable
37 RTCRtpSender*,
42 RTCRtpSender* sender() const;
76 Member<RTCRtpSender> sender_;
|
| H A D | rtc_peer_connection.h | 76 class RTCRtpSender; variable
271 const HeapVector<Member<RTCRtpSender>>& getSenders() const;
276 RTCRtpSender* addTrack(MediaStreamTrack*, MediaStreamVector, ExceptionState&);
277 void removeTrack(RTCRtpSender*, ExceptionState&);
468 RTCRtpSender* FindSenderForTrackAndStream(MediaStreamTrack*, MediaStream*);
469 HeapVector<Member<RTCRtpSender>>::iterator FindSender(
486 RTCRtpSender* CreateOrUpdateSender(std::unique_ptr<RTCRtpSenderPlatform>,
605 HeapVector<Member<RTCRtpSender>> rtp_senders_;
|
| /dports/devel/py-aiortc/aiortc-1.2.1/tests/ |
| H A D | test_rtcrtpsender.py | 17 from aiortc.rtcrtpsender import RTCRtpSender
56 capabilities = RTCRtpSender.getCapabilities("audio")
82 capabilities = RTCRtpSender.getCapabilities("video")
127 RTCRtpSender.getCapabilities("bogus")
130 sender = RTCRtpSender("audio", self.local_transport)
137 RTCRtpSender("audio", self.local_transport)
143 sender = RTCRtpSender(AudioStreamTrack(), self.local_transport)
151 sender = RTCRtpSender(VideoStreamTrack(), self.local_transport)
167 sender = RTCRtpSender(VideoStreamTrack(), self.local_transport)
180 sender = RTCRtpSender(VideoStreamTrack(), self.local_transport)
[all …]
|
| /dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/ |
| H A D | RTCRtpSender.mm | 11 #import "RTCRtpSender+Private.h"
20 @implementation RTCRtpSender { implementation
35 RTCLogError(@"RTCRtpSender(%p): Failed to set parameters: %@", self,
51 RTCLogError(@"RTCRtpSender(%p): Failed to set track %@", self, track);
56 return [NSString stringWithFormat:@"RTCRtpSender {\n senderId: %@\n}",
70 RTCRtpSender *sender = (RTCRtpSender *)object;
89 RTCLogInfo(@"RTCRtpSender(%p): created sender: %@", self, self.description);
|
| /dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/webrtc/sdk/objc/api/peerconnection/ |
| H A D | RTCRtpSender.mm | 11 #import "RTCRtpSender+Private.h"
16 #import "RTCRtpSender+Native.h"
22 @implementation RTCRtpSender { implementation
40 RTCLogError(@"RTCRtpSender(%p): Failed to set parameters: %@", self,
56 RTCLogError(@"RTCRtpSender(%p): Failed to set track %@", self, track);
78 return [NSString stringWithFormat:@"RTCRtpSender {\n senderId: %@\n}",
92 RTCRtpSender *sender = (RTCRtpSender *)object;
124 RTCLogInfo(@"RTCRtpSender(%p): created sender: %@", self, self.description);
|
| H A D | RTCRtpSender.h | 21 @protocol RTCRtpSender <NSObject>
47 @interface RTCRtpSender : NSObject <RTCRtpSender>
|
| H A D | RTCPeerConnection.h | 24 @class RTCRtpSender;
182 @property(nonatomic, readonly) NSArray<RTCRtpSender *> *senders;
237 - (RTCRtpSender *)addTrack:(RTCMediaStreamTrack *)track streamIds:(NSArray<NSString *> *)streamIds;
246 - (BOOL)removeTrack:(RTCRtpSender *)sender;
320 - (RTCRtpSender *)senderWithKind:(NSString *)kind streamId:(NSString *)streamId;
349 - (void)statisticsForSender:(RTCRtpSender *)sender
|
| /dports/net-im/tg_owt/tg_owt-d578c76/src/sdk/objc/api/peerconnection/ |
| H A D | RTCRtpSender.mm | 11 #import "RTCRtpSender+Private.h"
16 #import "RTCRtpSender+Native.h"
22 @implementation RTC_OBJC_TYPE (RTCRtpSender) { category
40 RTCLogError(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): Failed to set parameters: %@", self, parameters);
56 RTCLogError(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): Failed to set track %@", self, track);
79 stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpSender) {\n senderId: %@\n}", self.senderId];
92 RTC_OBJC_TYPE(RTCRtpSender) *sender = (RTC_OBJC_TYPE(RTCRtpSender) *)object;
125 RTCLogInfo(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): created sender: %@", self, self.description);
|
| H A D | RTCRtpSender.h | 22 (RTCRtpSender)<NSObject>
48 @interface RTC_OBJC_TYPE (RTCRtpSender) : NSObject <RTC_OBJC_TYPE(RTCRtpSender)>
|
| H A D | RTCPeerConnection.h | 24 @class RTC_OBJC_TYPE(RTCRtpSender);
192 @property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpSender) *> *senders;
252 - (nullable RTC_OBJC_TYPE(RTCRtpSender) *)addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
262 - (BOOL)removeTrack:(RTC_OBJC_TYPE(RTCRtpSender) *)sender;
349 - (RTC_OBJC_TYPE(RTCRtpSender) *)senderWithKind : (NSString *)kind streamId
382 - (void)statisticsForSender:(RTC_OBJC_TYPE(RTCRtpSender) *)sender
|
| /dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/webrtc/sdk/objc/api/peerconnection/ |
| H A D | RTCRtpSender.mm | 11 #import "RTCRtpSender+Private.h"
16 #import "RTCRtpSender+Native.h"
22 @implementation RTC_OBJC_TYPE (RTCRtpSender) { category
40 RTCLogError(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): Failed to set parameters: %@", self, parameters);
56 RTCLogError(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): Failed to set track %@", self, track);
79 stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpSender) {\n senderId: %@\n}", self.senderId];
92 RTC_OBJC_TYPE(RTCRtpSender) *sender = (RTC_OBJC_TYPE(RTCRtpSender) *)object;
125 RTCLogInfo(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): created sender: %@", self, self.description);
|
| H A D | RTCRtpSender.h | 22 (RTCRtpSender)<NSObject>
48 @interface RTC_OBJC_TYPE (RTCRtpSender) : NSObject <RTC_OBJC_TYPE(RTCRtpSender)>
|
| H A D | RTCPeerConnection.h | 24 @class RTC_OBJC_TYPE(RTCRtpSender);
186 @property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpSender) *> *senders;
241 - (RTC_OBJC_TYPE(RTCRtpSender) *)addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
251 - (BOOL)removeTrack:(RTC_OBJC_TYPE(RTCRtpSender) *)sender;
330 - (RTC_OBJC_TYPE(RTCRtpSender) *)senderWithKind : (NSString *)kind streamId
363 - (void)statisticsForSender:(RTC_OBJC_TYPE(RTCRtpSender) *)sender
|
| /dports/lang/spidermonkey60/firefox-60.9.0/testing/web-platform/meta/webrtc/ |
| H A D | RTCRtpSender-getCapabilities.html.ini | 1 [RTCRtpSender-getCapabilities.html]
2 [RTCRtpSender.getCapabilities('audio') should return RTCRtpCapabilities dictionary]
5 [RTCRtpSender.getCapabilities('video') should return RTCRtpCapabilities dictionary]
|
| H A D | RTCRtpReceiver-getCapabilities.html.ini | 2 [RTCRtpSender.getCapabilities('audio') should return RTCRtpCapabilities dictionary]
5 [RTCRtpSender.getCapabilities('video') should return RTCRtpCapabilities dictionary]
|
| /dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/ |
| H A D | RTCRtpSender.h | 20 @protocol RTCRtpSender <NSObject>
40 @interface RTCRtpSender : NSObject <RTCRtpSender>
|
| /dports/lang/spidermonkey78/firefox-78.9.0/testing/web-platform/meta/webrtc/ |
| H A D | RTCRtpSender-transport.https.html.ini | 1 [RTCRtpSender-transport.https.html]
4 [RTCRtpSender/receiver.transport has a value when connected]
|
| /dports/www/firefox-esr/firefox-91.8.0/testing/web-platform/meta/webrtc-encoded-transform/ |
| H A D | RTCPeerConnection-insertable-streams-worker.https.html.ini | 2 …[RTCRtpSender readable stream transferred to a Worker and the Worker sends an RTCEncodedAudioFrame…
5 …[RTCRtpSender readable stream transferred to a Worker and the Worker sends an RTCEncodedVideoFrame…
|
| /dports/www/firefox/firefox-99.0/testing/web-platform/meta/webrtc-encoded-transform/ |
| H A D | RTCPeerConnection-insertable-streams-worker.https.html.ini | 2 …[RTCRtpSender readable stream transferred to a Worker and the Worker sends an RTCEncodedAudioFrame…
5 …[RTCRtpSender readable stream transferred to a Worker and the Worker sends an RTCEncodedVideoFrame…
|
| H A D | idlharness.https.window.js.ini | 98 [RTCRtpSender interface: attribute transform]
101 …[RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit prope…
140 [RTCRtpSender interface: operation generateKeyFrame(optional sequence<DOMString>)]
143 …[RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit prope…
146 …[RTCRtpSender interface: calling generateKeyFrame(optional sequence<DOMString>) on new RTCPeerConn…
|
| /dports/mail/thunderbird/thunderbird-91.8.0/testing/web-platform/meta/webrtc-encoded-transform/ |
| H A D | RTCPeerConnection-insertable-streams-worker.https.html.ini | 2 …[RTCRtpSender readable stream transferred to a Worker and the Worker sends an RTCEncodedAudioFrame…
5 …[RTCRtpSender readable stream transferred to a Worker and the Worker sends an RTCEncodedVideoFrame…
|