Diff
Modified: trunk/LayoutTests/ChangeLog (270293 => 270294)
--- trunk/LayoutTests/ChangeLog 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/LayoutTests/ChangeLog 2020-12-01 12:34:23 UTC (rev 270294)
@@ -1,5 +1,18 @@
2020-12-01 Youenn Fablet <[email protected]>
+ Allow RTCRtpScriptTransformer to request key frames
+ https://bugs.webkit.org/show_bug.cgi?id=219199
+
+ Reviewed by Eric Carlson.
+
+ * http/wpt/webrtc/audio-script-transform-expected.txt: Added.
+ * http/wpt/webrtc/audio-script-transform.html: Added.
+ * http/wpt/webrtc/video-script-transform-expected.txt: Added.
+ * http/wpt/webrtc/video-script-transform.html: Added.
+ * http/wpt/webrtc/context-transform.js: Added.
+
+2020-12-01 Youenn Fablet <[email protected]>
+
Add support for readable/writable to RTCRtpSFrameTransform
https://bugs.webkit.org/show_bug.cgi?id=219298
Added: trunk/LayoutTests/http/wpt/webrtc/audio-script-transform-expected.txt (0 => 270294)
--- trunk/LayoutTests/http/wpt/webrtc/audio-script-transform-expected.txt (rev 0)
+++ trunk/LayoutTests/http/wpt/webrtc/audio-script-transform-expected.txt 2020-12-01 12:34:23 UTC (rev 270294)
@@ -0,0 +1,5 @@
+
+
+PASS setup and check context
+PASS audio is playing
+
Added: trunk/LayoutTests/http/wpt/webrtc/audio-script-transform.html (0 => 270294)
--- trunk/LayoutTests/http/wpt/webrtc/audio-script-transform.html (rev 0)
+++ trunk/LayoutTests/http/wpt/webrtc/audio-script-transform.html 2020-12-01 12:34:23 UTC (rev 270294)
@@ -0,0 +1,66 @@
+<!doctype html>
+<html>
+ <head>
+ <meta charset="utf-8">
+ <script src=""
+ <script src=""
+ </head>
+ <body>
+ <video id="video" autoplay playsInline></video>
+ <script src=""
+ <script>
+function waitForMessage(port, data)
+{
+ let gotMessage;
+ const promise = new Promise((resolve, reject) => {
+ gotMessage = resolve;
+ setTimeout(() => { reject("did not get " + data) }, 5000);
+ });
+ port._onmessage_ = event => {
+ if (event.data ="" data)
+ gotMessage();
+ };
+ return promise;
+}
+
+let senderTransform, receiverTransform;
+let stream;
+promise_test(async (test) => {
+ worker = new Worker("context-transform.js");
+ const data = "" new Promise(resolve => worker._onmessage_ = (event) => resolve(event.data));
+ assert_equals(data, "registered");
+
+ const localStream = await navigator.mediaDevices.getUserMedia({audio: true});
+
+ let sender, receiver;
+ senderTransform = new RTCRtpScriptTransform(worker, 'MockRTCRtpTransform');
+ receiverTransform = new RTCRtpScriptTransform(worker, 'MockRTCRtpTransform');
+
+ promise1 = waitForMessage(senderTransform.port, "started audio sender");
+ promise2 = waitForMessage(receiverTransform.port, "started audio receiver");
+
+ stream = await new Promise((resolve, reject) => {
+ createConnections((firstConnection) => {
+ sender = firstConnection.addTrack(localStream.getAudioTracks()[0], localStream);
+ sender.transform = senderTransform;
+ }, (secondConnection) => {
+ secondConnection._ontrack_ = (trackEvent) => {
+ receiver = trackEvent.receiver;
+ receiver.transform = receiverTransform;
+ resolve(trackEvent.streams[0]);
+ };
+ });
+ setTimeout(() => reject("Test timed out"), 5000);
+ });
+
+ await promise1;
+ await promise2;
+}, "setup and check context");
+
+promise_test((test) => {
+ video.srcObject = stream;
+ return video.play();
+}, "audio is playing");
+ </script>
+ </body>
+</html>
Added: trunk/LayoutTests/http/wpt/webrtc/context-transform.js (0 => 270294)
--- trunk/LayoutTests/http/wpt/webrtc/context-transform.js (rev 0)
+++ trunk/LayoutTests/http/wpt/webrtc/context-transform.js 2020-12-01 12:34:23 UTC (rev 270294)
@@ -0,0 +1,43 @@
+class MockRTCRtpTransformer extends RTCRtpScriptTransformer {
+ constructor() {
+ super();
+ this.askKeyFrame = false;
+ this.port._onmessage_ = (event) => {
+ if (event.data ="" "startKeyFrames")
+ this.askKeyFrame = true;
+ else if (event.data ="" "endKeyFrames")
+ this.askKeyFrame = false;
+ };
+ }
+ start(readableStream, writableStream, context)
+ {
+ this.reader = readableStream.getReader();
+ this.writer = writableStream.getWriter();
+ this.process();
+ this.context = context;
+ this.port.postMessage("started " + context.mediaType + " " + context.side);
+ }
+
+ process()
+ {
+ this.reader.read().then(chunk => {
+ if (chunk.done)
+ return;
+
+ this.writer.write(chunk.value);
+
+ if (this.context.mediaType === "video") {
+ if (chunk.value instanceof RTCEncodedVideoFrame)
+ this.port.postMessage("video frame " + chunk.value.type);
+
+ if (this.askKeyFrame)
+ this.context.requestKeyFrame();
+ }
+
+ this.process();
+ });
+ }
+};
+
+registerRTCRtpScriptTransformer("MockRTCRtpTransform", MockRTCRtpTransformer);
+self.postMessage("registered");
Added: trunk/LayoutTests/http/wpt/webrtc/video-script-transform-expected.txt (0 => 270294)
--- trunk/LayoutTests/http/wpt/webrtc/video-script-transform-expected.txt (rev 0)
+++ trunk/LayoutTests/http/wpt/webrtc/video-script-transform-expected.txt 2020-12-01 12:34:23 UTC (rev 270294)
@@ -0,0 +1,8 @@
+
+
+PASS setup
+PASS video is playing
+PASS video frame type
+PASS key frame requests from sender
+PASS key frame requests from receiver
+
Added: trunk/LayoutTests/http/wpt/webrtc/video-script-transform.html (0 => 270294)
--- trunk/LayoutTests/http/wpt/webrtc/video-script-transform.html (rev 0)
+++ trunk/LayoutTests/http/wpt/webrtc/video-script-transform.html 2020-12-01 12:34:23 UTC (rev 270294)
@@ -0,0 +1,87 @@
+<!doctype html>
+<html>
+ <head>
+ <meta charset="utf-8">
+ <script src=""
+ <script src=""
+ </head>
+ <body>
+ <video id="video" autoplay playsInline></video>
+ <script src=""
+ <script>
+function waitForMessage(port, data)
+{
+ let gotMessage;
+ const promise = new Promise((resolve, reject) => {
+ gotMessage = resolve;
+ setTimeout(() => { reject("did not get " + data) }, 5000);
+ });
+ port._onmessage_ = event => {
+ if (event.data ="" data)
+ gotMessage();
+ };
+ return promise;
+}
+
+let senderTransform, receiverTransform;
+let stream;
+promise_test(async (test) => {
+ worker = new Worker('context-transform.js');
+ const data = "" new Promise(resolve => worker._onmessage_ = (event) => resolve(event.data));
+ assert_equals(data, "registered");
+
+ const localStream = await navigator.mediaDevices.getUserMedia({video: true});
+
+ let sender, receiver;
+ senderTransform = new RTCRtpScriptTransform(worker, 'MockRTCRtpTransform');
+ receiverTransform = new RTCRtpScriptTransform(worker, 'MockRTCRtpTransform');
+
+ promise1 = waitForMessage(senderTransform.port, "started video sender");
+ promise2 = waitForMessage(receiverTransform.port, "started video receiver");
+
+ stream = await new Promise((resolve, reject) => {
+ createConnections((firstConnection) => {
+ sender = firstConnection.addTrack(localStream.getVideoTracks()[0], localStream);
+ sender.transform = senderTransform;
+ }, (secondConnection) => {
+ secondConnection._ontrack_ = (trackEvent) => {
+ receiver = trackEvent.receiver;
+ receiver.transform = receiverTransform;
+ resolve(trackEvent.streams[0]);
+ };
+ });
+ setTimeout(() => reject("Test timed out"), 5000);
+ });
+
+ await promise1;
+ await promise2;
+}, "setup");
+
+promise_test((test) => {
+ video.srcObject = stream;
+ return video.play();
+}, "video is playing");
+
+promise_test(async (test) => {
+ await waitForMessage(receiverTransform.port, "video frame delta");
+}, "video frame type");
+
+promise_test(async (test) => {
+ senderTransform.port.postMessage("startKeyFrames");
+ await waitForMessage(senderTransform.port, "video frame key");
+ await waitForMessage(receiverTransform.port, "video frame key");
+ senderTransform.port.postMessage("endKeyFrames");
+
+ await waitForMessage(receiverTransform.port, "video frame delta");
+}, "key frame requests from sender");
+
+promise_test(async (test) => {
+ receiverTransform.port.postMessage("startKeyFrames");
+ await waitForMessage(senderTransform.port, "video frame key");
+ await waitForMessage(receiverTransform.port, "video frame key");
+ receiverTransform.port.postMessage("endKeyFrames");
+
+}, "key frame requests from receiver");
+ </script>
+ </body>
+</html>
Modified: trunk/Source/ThirdParty/libwebrtc/ChangeLog (270293 => 270294)
--- trunk/Source/ThirdParty/libwebrtc/ChangeLog 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/ThirdParty/libwebrtc/ChangeLog 2020-12-01 12:34:23 UTC (rev 270294)
@@ -1,3 +1,29 @@
+2020-12-01 Youenn Fablet <[email protected]>
+
+ Allow RTCRtpScriptTransformer to request key frames
+ https://bugs.webkit.org/show_bug.cgi?id=219199
+
+ Reviewed by Eric Carlson.
+
+ Add GenerateKeyFrame to sender and receiver interface.
+ Pipe the request to the necessary underlying objects implementing this as follows:
+ - Ask encoder to generate a key frame for sender transforms
+ - Send a FIR network request from receiver to sender for receiver transforms
+
+ * Source/webrtc/api/rtp_receiver_interface.h:
+ * Source/webrtc/api/rtp_sender_interface.h:
+ * Source/webrtc/call/video_send_stream.h:
+ * Source/webrtc/media/base/media_channel.h:
+ (cricket::MediaChannel::GenerateKeyFrame):
+ * Source/webrtc/media/engine/webrtc_video_engine.cc:
+ * Source/webrtc/media/engine/webrtc_video_engine.h:
+ * Source/webrtc/pc/rtp_sender.cc:
+ * Source/webrtc/pc/rtp_sender.h:
+ * Source/webrtc/pc/video_rtp_receiver.cc:
+ * Source/webrtc/pc/video_rtp_receiver.h:
+ * Source/webrtc/video/video_send_stream.cc:
+ * Source/webrtc/video/video_send_stream.h:
+
2020-11-30 Youenn Fablet <[email protected]>
Introduce an experimental flag specific to VP9 profile 2
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_receiver_interface.h (270293 => 270294)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_receiver_interface.h 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_receiver_interface.h 2020-12-01 12:34:23 UTC (rev 270294)
@@ -113,6 +113,9 @@
// logic.
virtual void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
+#if defined(WEBRTC_WEBKIT_BUILD)
+ virtual void GenerateKeyFrame() { }
+#endif
protected:
~RtpReceiverInterface() override = default;
@@ -142,6 +145,9 @@
PROXY_METHOD1(void,
SetDepacketizerToDecoderFrameTransformer,
rtc::scoped_refptr<FrameTransformerInterface>)
+#if defined(WEBRTC_WEBKIT_BUILD)
+PROXY_METHOD0(void, GenerateKeyFrame)
+#endif
END_PROXY_MAP()
} // namespace webrtc
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_sender_interface.h (270293 => 270294)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_sender_interface.h 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_sender_interface.h 2020-12-01 12:34:23 UTC (rev 270294)
@@ -96,6 +96,9 @@
virtual void SetEncoderToPacketizerFrameTransformer(
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
+#if defined(WEBRTC_WEBKIT_BUILD)
+ virtual void GenerateKeyFrame() { }
+#endif
protected:
~RtpSenderInterface() override = default;
@@ -126,6 +129,9 @@
PROXY_METHOD1(void,
SetEncoderToPacketizerFrameTransformer,
rtc::scoped_refptr<FrameTransformerInterface>)
+#if defined(WEBRTC_WEBKIT_BUILD)
+PROXY_METHOD0(void, GenerateKeyFrame)
+#endif
END_PROXY_MAP()
} // namespace webrtc
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.h (270293 => 270294)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.h 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.h 2020-12-01 12:34:23 UTC (rev 270294)
@@ -238,6 +238,10 @@
virtual Stats GetStats() = 0;
+#if defined(WEBRTC_WEBKIT_BUILD)
+ virtual void GenerateKeyFrame() = 0;
+#endif
+
protected:
virtual ~VideoSendStream() {}
};
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtc_video_engine.cc (270293 => 270294)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtc_video_engine.cc 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtc_video_engine.cc 2020-12-01 12:34:23 UTC (rev 270294)
@@ -2644,6 +2644,14 @@
RecreateWebRtcStream();
}
+#if defined(WEBRTC_WEBKIT_BUILD)
+void WebRtcVideoChannel::WebRtcVideoSendStream::GenerateKeyFrame() {
+ RTC_DCHECK_RUN_ON(&thread_checker_);
+ if (stream_ != NULL)
+ stream_->GenerateKeyFrame();
+}
+#endif
+
void WebRtcVideoChannel::WebRtcVideoSendStream::RecreateWebRtcStream() {
RTC_DCHECK_RUN_ON(&thread_checker_);
if (stream_ != NULL) {
@@ -3316,11 +3324,20 @@
WebRtcVideoReceiveStream* stream = FindReceiveStream(ssrc);
if (stream) {
stream->GenerateKeyFrame();
- } else {
- RTC_LOG(LS_ERROR)
- << "Absent receive stream; ignoring key frame generation for ssrc "
- << ssrc;
+ return;
}
+#if defined(WEBRTC_WEBKIT_BUILD)
+ if (ssrc != 0) {
+ auto matching_stream = send_streams_.find(ssrc);
+ if (matching_stream != send_streams_.end()) {
+ matching_stream->second->GenerateKeyFrame();
+ return;
+ }
+ }
+#endif
+ RTC_LOG(LS_ERROR)
+ << "Absent receive stream; ignoring key frame generation for ssrc "
+ << ssrc;
}
void WebRtcVideoChannel::SetEncoderToPacketizerFrameTransformer(
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtc_video_engine.h (270293 => 270294)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtc_video_engine.h 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtc_video_engine.h 2020-12-01 12:34:23 UTC (rev 270294)
@@ -359,6 +359,9 @@
void SetEncoderToPacketizerFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface>
frame_transformer);
+#if defined(WEBRTC_WEBKIT_BUILD)
+ void GenerateKeyFrame();
+#endif
private:
// Parameters needed to reconstruct the underlying stream.
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtp_sender.cc (270293 => 270294)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtp_sender.cc 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtp_sender.cc 2020-12-01 12:34:23 UTC (rev 270294)
@@ -647,4 +647,15 @@
});
}
+#if defined(WEBRTC_WEBKIT_BUILD)
+void VideoRtpSender::GenerateKeyFrame()
+{
+ if (video_media_channel() && ssrc_ && !stopped_) {
+ worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
+ video_media_channel()->GenerateKeyFrame(ssrc_);
+ });
+ }
+}
+#endif
+
} // namespace webrtc
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtp_sender.h (270293 => 270294)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtp_sender.h 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtp_sender.h 2020-12-01 12:34:23 UTC (rev 270294)
@@ -348,6 +348,10 @@
// Hook to allow custom logic when tracks are attached.
void AttachTrack() override;
+#if defined(WEBRTC_WEBKIT_BUILD)
+ void GenerateKeyFrame() override;
+#endif
+
private:
cricket::VideoMediaChannel* video_media_channel() {
return static_cast<cricket::VideoMediaChannel*>(media_channel_);
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/video_rtp_receiver.cc (270293 => 270294)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/video_rtp_receiver.cc 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/video_rtp_receiver.cc 2020-12-01 12:34:23 UTC (rev 270294)
@@ -115,6 +115,18 @@
});
}
+#if defined(WEBRTC_WEBKIT_BUILD)
+void VideoRtpReceiver::GenerateKeyFrame()
+{
+ worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
+ RTC_DCHECK_RUN_ON(worker_thread_);
+ if (media_channel_ && !stopped_) {
+ media_channel_->GenerateKeyFrame(ssrc_.value_or(0));
+ }
+ });
+}
+#endif
+
void VideoRtpReceiver::Stop() {
// TODO(deadbeef): Need to do more here to fully stop receiving packets.
if (stopped_) {
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/video_rtp_receiver.h (270293 => 270294)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/video_rtp_receiver.h 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/video_rtp_receiver.h 2020-12-01 12:34:23 UTC (rev 270294)
@@ -88,6 +88,9 @@
void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) override;
+#if defined(WEBRTC_WEBKIT_BUILD)
+ void GenerateKeyFrame() override;
+#endif
// RtpReceiverInternal implementation.
void Stop() override;
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.cc (270293 => 270294)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.cc 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.cc 2020-12-01 12:34:23 UTC (rev 270294)
@@ -240,5 +240,14 @@
send_stream_->DeliverRtcp(packet, length);
}
+#if defined(WEBRTC_WEBKIT_BUILD)
+void VideoSendStream::GenerateKeyFrame()
+{
+ RTC_DCHECK_RUN_ON(&thread_checker_);
+ if (video_stream_encoder_)
+ video_stream_encoder_->SendKeyFrame();
+}
+#endif
+
} // namespace internal
} // namespace webrtc
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.h (270293 => 270294)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.h 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.h 2020-12-01 12:34:23 UTC (rev 270294)
@@ -86,6 +86,9 @@
void ReconfigureVideoEncoder(VideoEncoderConfig) override;
Stats GetStats() override;
+#if defined(WEBRTC_WEBKIT_BUILD)
+ void GenerateKeyFrame() override;
+#endif
void StopPermanentlyAndGetRtpStates(RtpStateMap* rtp_state_map,
RtpPayloadStateMap* payload_state_map);
Modified: trunk/Source/WebCore/CMakeLists.txt (270293 => 270294)
--- trunk/Source/WebCore/CMakeLists.txt 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/CMakeLists.txt 2020-12-01 12:34:23 UTC (rev 270294)
@@ -410,6 +410,7 @@
Modules/mediastream/RTCRtpScriptTransformProvider.idl
Modules/mediastream/RTCRtpScriptTransformer.idl
Modules/mediastream/RTCRtpScriptTransformerConstructor.idl
+ Modules/mediastream/RTCRtpScriptTransformerContext.idl
Modules/mediastream/RTCRtpSendParameters.idl
Modules/mediastream/RTCRtpSender+Transform.idl
Modules/mediastream/RTCRtpSender.idl
Modified: trunk/Source/WebCore/ChangeLog (270293 => 270294)
--- trunk/Source/WebCore/ChangeLog 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/ChangeLog 2020-12-01 12:34:23 UTC (rev 270294)
@@ -1,3 +1,53 @@
+2020-12-01 Youenn Fablet <[email protected]>
+
+ Allow RTCRtpScriptTransformer to request key frames
+ https://bugs.webkit.org/show_bug.cgi?id=219199
+ <rdar://problem/71763010>
+
+ Reviewed by Eric Carlson.
+
+ Expose RTCRtpScriptTransformerContext to RTCRtpScriptTransformer as part of third parameter of the 'start' callback.
+ This context exposes some states (sender or receiver transform, audio or video transform).
+ It also exposes the ability to request key frames for video transforms.
+
+ Make RTCRtpTransformableFrame ref counted so that we can keep it even though the frame was piped to the writable stream.
+ Make LibWebRTCRtpTransformableFrame able to cope with a nullptr internal rtc frame.
+
+ Add support for RTCEncodedVideoFrame.type.
+
+ Tests: http/wpt/webrtc/audio-script-transform.html
+ http/wpt/webrtc/video-script-transform.html
+
+ * CMakeLists.txt:
+ * DerivedSources-input.xcfilelist:
+ * DerivedSources-output.xcfilelist:
+ * DerivedSources.make:
+ * Modules/mediastream/RTCEncodedVideoFrame.cpp:
+ (WebCore::RTCEncodedVideoFrame::RTCEncodedVideoFrame):
+ * Modules/mediastream/RTCEncodedVideoFrame.h:
+ (WebCore::RTCEncodedVideoFrame::type const):
+ * Modules/mediastream/RTCEncodedVideoFrame.idl:
+ * Modules/mediastream/RTCRtpScriptTransformerContext.h: Added.
+ (WebCore::RTCRtpScriptTransformerContext::side const):
+ (WebCore::RTCRtpScriptTransformerContext::mediaType const):
+ (WebCore::RTCRtpScriptTransformerContext::create):
+ (WebCore::RTCRtpScriptTransformerContext::RTCRtpScriptTransformerContext):
+ (WebCore::RTCRtpScriptTransformerContext::requestKeyFrame):
+ * Modules/mediastream/RTCRtpScriptTransformerContext.idl: Added.
+ * Modules/mediastream/RTCRtpTransformBackend.h:
+ * Modules/mediastream/RTCRtpTransformableFrame.h:
+ * Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverTransformBackend.cpp:
+ (WebCore::LibWebRTCRtpReceiverTransformBackend::requestKeyFrame):
+ * Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverTransformBackend.h:
+ * Modules/mediastream/libwebrtc/LibWebRTCRtpSenderTransformBackend.cpp:
+ (WebCore::LibWebRTCRtpSenderTransformBackend::requestKeyFrame):
+ * Modules/mediastream/libwebrtc/LibWebRTCRtpSenderTransformBackend.h:
+ * Modules/mediastream/libwebrtc/LibWebRTCRtpTransformableFrame.cpp:
+ (WebCore::LibWebRTCRtpTransformableFrame::isKeyFrame const):
+ * Modules/mediastream/libwebrtc/LibWebRTCRtpTransformableFrame.h:
+ * Sources.txt:
+ * WebCore.xcodeproj/project.pbxproj:
+
2020-12-01 Rob Buis <[email protected]>
Remove m_gridItemsIndexesMap
Modified: trunk/Source/WebCore/DerivedSources-input.xcfilelist (270293 => 270294)
--- trunk/Source/WebCore/DerivedSources-input.xcfilelist 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/DerivedSources-input.xcfilelist 2020-12-01 12:34:23 UTC (rev 270294)
@@ -244,6 +244,7 @@
$(PROJECT_DIR)/Modules/mediastream/RTCRtpScriptTransformProvider.idl
$(PROJECT_DIR)/Modules/mediastream/RTCRtpScriptTransformer.idl
$(PROJECT_DIR)/Modules/mediastream/RTCRtpScriptTransformerConstructor.idl
+$(PROJECT_DIR)/Modules/mediastream/RTCRtpScriptTransformerContext.idl
$(PROJECT_DIR)/Modules/mediastream/RTCRtpSendParameters.idl
$(PROJECT_DIR)/Modules/mediastream/RTCRtpSender+Transform.idl
$(PROJECT_DIR)/Modules/mediastream/RTCRtpSender.idl
Modified: trunk/Source/WebCore/DerivedSources-output.xcfilelist (270293 => 270294)
--- trunk/Source/WebCore/DerivedSources-output.xcfilelist 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/DerivedSources-output.xcfilelist 2020-12-01 12:34:23 UTC (rev 270294)
@@ -1717,6 +1717,8 @@
$(BUILT_PRODUCTS_DIR)/DerivedSources/WebCore/JSRTCRtpScriptTransformer.h
$(BUILT_PRODUCTS_DIR)/DerivedSources/WebCore/JSRTCRtpScriptTransformerConstructor.cpp
$(BUILT_PRODUCTS_DIR)/DerivedSources/WebCore/JSRTCRtpScriptTransformerConstructor.h
+$(BUILT_PRODUCTS_DIR)/DerivedSources/WebCore/JSRTCRtpScriptTransformerContext.cpp
+$(BUILT_PRODUCTS_DIR)/DerivedSources/WebCore/JSRTCRtpScriptTransformerContext.h
$(BUILT_PRODUCTS_DIR)/DerivedSources/WebCore/JSRTCRtpSendParameters.cpp
$(BUILT_PRODUCTS_DIR)/DerivedSources/WebCore/JSRTCRtpSendParameters.h
$(BUILT_PRODUCTS_DIR)/DerivedSources/WebCore/JSRTCRtpSender+Transform.cpp
Modified: trunk/Source/WebCore/DerivedSources.make (270293 => 270294)
--- trunk/Source/WebCore/DerivedSources.make 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/DerivedSources.make 2020-12-01 12:34:23 UTC (rev 270294)
@@ -271,6 +271,7 @@
$(WebCore)/Modules/mediastream/RTCRtpScriptTransformProvider.idl \
$(WebCore)/Modules/mediastream/RTCRtpScriptTransformer.idl \
$(WebCore)/Modules/mediastream/RTCRtpScriptTransformerConstructor.idl \
+ $(WebCore)/Modules/mediastream/RTCRtpScriptTransformerContext.idl \
$(WebCore)/Modules/mediastream/RTCRtpSynchronizationSource.idl \
$(WebCore)/Modules/mediastream/RTCRtpTransceiver.idl \
$(WebCore)/Modules/mediastream/RTCRtpTransceiverDirection.idl \
Modified: trunk/Source/WebCore/Modules/mediastream/RTCEncodedVideoFrame.cpp (270293 => 270294)
--- trunk/Source/WebCore/Modules/mediastream/RTCEncodedVideoFrame.cpp 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/Modules/mediastream/RTCEncodedVideoFrame.cpp 2020-12-01 12:34:23 UTC (rev 270294)
@@ -32,6 +32,7 @@
RTCEncodedVideoFrame::RTCEncodedVideoFrame(Ref<RTCRtpTransformableFrame>&& frame)
: RTCEncodedFrame(WTFMove(frame))
+ , m_type(m_frame->isKeyFrame() ? Type::Key : Type::Delta)
{
}
Modified: trunk/Source/WebCore/Modules/mediastream/RTCEncodedVideoFrame.h (270293 => 270294)
--- trunk/Source/WebCore/Modules/mediastream/RTCEncodedVideoFrame.h 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/Modules/mediastream/RTCEncodedVideoFrame.h 2020-12-01 12:34:23 UTC (rev 270294)
@@ -36,8 +36,13 @@
static Ref<RTCEncodedVideoFrame> create(Ref<RTCRtpTransformableFrame>&& frame) { return adoptRef(*new RTCEncodedVideoFrame(WTFMove(frame))); }
~RTCEncodedVideoFrame();
+ enum class Type { Empty, Key, Delta };
+ Type type() const { return m_type; }
+
private:
explicit RTCEncodedVideoFrame(Ref<RTCRtpTransformableFrame>&&);
+
+ Type m_type;
};
} // namespace WebCore
Modified: trunk/Source/WebCore/Modules/mediastream/RTCEncodedVideoFrame.idl (270293 => 270294)
--- trunk/Source/WebCore/Modules/mediastream/RTCEncodedVideoFrame.idl 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/Modules/mediastream/RTCEncodedVideoFrame.idl 2020-12-01 12:34:23 UTC (rev 270294)
@@ -23,12 +23,14 @@
* THE POSSIBILITY OF SUCH DAMAGE.
*/
+enum RTCEncodedVideoFrameType { "empty", "key", "delta" };
+
[
Conditional=WEB_RTC,
Exposed=DedicatedWorker,
ImplementationLacksVTable,
] interface RTCEncodedVideoFrame {
- // readonly attribute RTCEncodedVideoFrameType type;
+ readonly attribute RTCEncodedVideoFrameType type;
// readonly attribute unsigned long long timestamp;
attribute ArrayBuffer data;
// RTCVideoFrameMetadata getMetadata();
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpScriptTransformer.cpp (270293 => 270294)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpScriptTransformer.cpp 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpScriptTransformer.cpp 2020-12-01 12:34:23 UTC (rev 270294)
@@ -32,6 +32,7 @@
#include "JSCallbackData.h"
#include "JSRTCEncodedAudioFrame.h"
#include "JSRTCEncodedVideoFrame.h"
+#include "JSRTCRtpScriptTransformerContext.h"
#include "RTCRtpTransformableFrame.h"
#include "ReadableStream.h"
#include "ReadableStreamSource.h"
@@ -102,6 +103,7 @@
JSC::MarkedArgumentBuffer args;
args.append(toJSNewlyCreated(&globalObject, &globalObject, readableStream.releaseReturnValue()));
args.append(toJSNewlyCreated(&globalObject, &globalObject, writableStream.releaseReturnValue()));
+ args.append(toJSNewlyCreated(&globalObject, &globalObject, RTCRtpScriptTransformerContext::create(makeRef(backend))));
NakedPtr<JSC::Exception> returnedException;
callback->invokeCallback(JSC::jsUndefined(), args, JSCallbackData::CallbackType::Object, JSC::Identifier::fromString(vm, "start"), returnedException);
Copied: trunk/Source/WebCore/Modules/mediastream/RTCRtpScriptTransformerContext.h (from rev 270293, trunk/Source/WebCore/Modules/mediastream/RTCEncodedVideoFrame.h) (0 => 270294)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpScriptTransformerContext.h (rev 0)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpScriptTransformerContext.h 2020-12-01 12:34:23 UTC (rev 270294)
@@ -0,0 +1,80 @@
+/*
+ * Copyright (C) 2020 Apple Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+ * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
+ * OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#pragma once
+
+#if ENABLE(WEB_RTC)
+
+#include "RTCRtpTransformBackend.h"
+#include <wtf/Ref.h>
+
+namespace WebCore {
+
+class RTCRtpTransformBackend;
+
+class RTCRtpScriptTransformerContext
+ : public RefCounted<RTCRtpScriptTransformerContext> {
+public:
+ static Ref<RTCRtpScriptTransformerContext> create(Ref<RTCRtpTransformBackend>&&);
+ ~RTCRtpScriptTransformerContext() = default;
+
+ using Side = RTCRtpTransformBackend::Side;
+ Side side() const { return m_backend->side(); }
+
+ using MediaType = RTCRtpTransformBackend::MediaType;
+ MediaType mediaType() const { return m_backend->mediaType(); }
+
+ ExceptionOr<void> requestKeyFrame();
+
+private:
+ explicit RTCRtpScriptTransformerContext(Ref<RTCRtpTransformBackend>&&);
+
+ Ref<RTCRtpTransformBackend> m_backend;
+};
+
+inline Ref<RTCRtpScriptTransformerContext> RTCRtpScriptTransformerContext::create(Ref<RTCRtpTransformBackend>&& backend)
+{
+ return adoptRef(*new RTCRtpScriptTransformerContext(WTFMove(backend)));
+}
+
+inline RTCRtpScriptTransformerContext::RTCRtpScriptTransformerContext(Ref<RTCRtpTransformBackend>&& backend)
+ : m_backend(WTFMove(backend))
+{
+}
+
+inline ExceptionOr<void> RTCRtpScriptTransformerContext::requestKeyFrame()
+{
+ switch (mediaType()) {
+ case MediaType::Audio:
+ return Exception { InvalidStateError, "Cannot request key frame on audio sender"_s };
+ case MediaType::Video:
+ m_backend->requestKeyFrame();
+ return { };
+ }
+}
+
+} // namespace WebCore
+
+#endif // ENABLE(WEB_RTC)
Copied: trunk/Source/WebCore/Modules/mediastream/RTCRtpScriptTransformerContext.idl (from rev 270293, trunk/Source/WebCore/Modules/mediastream/RTCEncodedVideoFrame.idl) (0 => 270294)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpScriptTransformerContext.idl (rev 0)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpScriptTransformerContext.idl 2020-12-01 12:34:23 UTC (rev 270294)
@@ -0,0 +1,38 @@
+/*
+ * Copyright (C) 2020 Apple Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS''
+ * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
+ * THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+ * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF
+ * THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+enum RTCRtpScriptTransformerContextSide { "receiver", "sender" };
+enum RTCRtpScriptTransformerContextMediaType { "audio", "video" };
+
+[
+ Conditional=WEB_RTC,
+ Exposed=DedicatedWorker,
+ ImplementationLacksVTable,
+] interface RTCRtpScriptTransformerContext {
+ readonly attribute RTCRtpScriptTransformerContextSide side;
+ readonly attribute RTCRtpScriptTransformerContextMediaType mediaType;
+
+ [MayThrowException] undefined requestKeyFrame();
+};
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpTransformBackend.h (270293 => 270294)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpTransformBackend.h 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpTransformBackend.h 2020-12-01 12:34:23 UTC (rev 270294)
@@ -46,6 +46,8 @@
enum class Side { Receiver, Sender };
virtual Side side() const = 0;
+
+ virtual void requestKeyFrame() = 0;
};
} // namespace WebCore
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpTransformableFrame.h (270293 => 270294)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpTransformableFrame.h 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpTransformableFrame.h 2020-12-01 12:34:23 UTC (rev 270294)
@@ -40,6 +40,8 @@
};
virtual Data data() const = 0;
virtual void setData(Data) = 0;
+
+ virtual bool isKeyFrame() const = 0;
};
} // namespace WebCore
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverTransformBackend.cpp (270293 => 270294)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverTransformBackend.cpp 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverTransformBackend.cpp 2020-12-01 12:34:23 UTC (rev 270294)
@@ -52,6 +52,12 @@
m_rtcReceiver->SetDepacketizerToDecoderFrameTransformer(this);
}
+void LibWebRTCRtpReceiverTransformBackend::requestKeyFrame()
+{
+ ASSERT(mediaType() == MediaType::Video);
+ m_rtcReceiver->GenerateKeyFrame();
+}
+
} // namespace WebCore
#endif // ENABLE(WEB_RTC) && USE(LIBWEBRTC)
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverTransformBackend.h (270293 => 270294)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverTransformBackend.h 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverTransformBackend.h 2020-12-01 12:34:23 UTC (rev 270294)
@@ -53,6 +53,7 @@
// RTCRtpTransformBackend
void setTransformableFrameCallback(Callback&&) final;
+ void requestKeyFrame() final;
rtc::scoped_refptr<webrtc::RtpReceiverInterface> m_rtcReceiver;
};
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderTransformBackend.cpp (270293 => 270294)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderTransformBackend.cpp 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderTransformBackend.cpp 2020-12-01 12:34:23 UTC (rev 270294)
@@ -52,6 +52,12 @@
m_rtcSender->SetEncoderToPacketizerFrameTransformer(this);
}
+void LibWebRTCRtpSenderTransformBackend::requestKeyFrame()
+{
+ ASSERT(mediaType() == MediaType::Video);
+ m_rtcSender->GenerateKeyFrame();
+}
+
} // namespace WebCore
#endif // ENABLE(WEB_RTC) && USE(LIBWEBRTC)
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderTransformBackend.h (270293 => 270294)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderTransformBackend.h 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderTransformBackend.h 2020-12-01 12:34:23 UTC (rev 270294)
@@ -56,6 +56,7 @@
// RTCRtpTransformBackend
void setTransformableFrameCallback(Callback&&) final;
+ void requestKeyFrame() final;
rtc::scoped_refptr<webrtc::RtpSenderInterface> m_rtcSender;
};
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransformableFrame.cpp (270293 => 270294)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransformableFrame.cpp 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransformableFrame.cpp 2020-12-01 12:34:23 UTC (rev 270294)
@@ -65,6 +65,13 @@
m_rtcFrame->SetData({ data.data, data.size });
}
+bool LibWebRTCRtpTransformableFrame::isKeyFrame() const
+{
+ ASSERT(m_rtcFrame);
+ auto* videoFrame = static_cast<webrtc::TransformableVideoFrameInterface*>(m_rtcFrame.get());
+ return videoFrame && videoFrame->IsKeyFrame();
+}
+
} // namespace WebCore
#endif // ENABLE(WEB_RTC) && USE(LIBWEBRTC)
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransformableFrame.h (270293 => 270294)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransformableFrame.h 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransformableFrame.h 2020-12-01 12:34:23 UTC (rev 270294)
@@ -46,8 +46,10 @@
private:
explicit LibWebRTCRtpTransformableFrame(std::unique_ptr<webrtc::TransformableFrameInterface>&&);
+ // RTCRtpTransformableFrame
Data data() const final;
void setData(Data) final;
+ bool isKeyFrame() const final;
std::unique_ptr<webrtc::TransformableFrameInterface> m_rtcFrame;
};
Modified: trunk/Source/WebCore/Sources.txt (270293 => 270294)
--- trunk/Source/WebCore/Sources.txt 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/Sources.txt 2020-12-01 12:34:23 UTC (rev 270294)
@@ -3347,6 +3347,7 @@
JSRTCRtpScriptTransform.cpp
JSRTCRtpScriptTransformer.cpp
JSRTCRtpScriptTransformerConstructor.cpp
+JSRTCRtpScriptTransformerContext.cpp
JSRTCRtpSendParameters.cpp
JSRTCRtpSender.cpp
JSRTCRtpSynchronizationSource.cpp
Modified: trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj (270293 => 270294)
--- trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2020-12-01 11:31:23 UTC (rev 270293)
+++ trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2020-12-01 12:34:23 UTC (rev 270294)
@@ -7795,6 +7795,9 @@
419E6EC42566D211002B5010 /* RTCPeerConnectionIceErrorEvent.idl */ = {isa = PBXFileReference; lastKnownFileType = text; path = RTCPeerConnectionIceErrorEvent.idl; sourceTree = "<group>"; };
419E6EC62566D213002B5010 /* RTCPeerConnectionIceErrorEvent.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = RTCPeerConnectionIceErrorEvent.cpp; sourceTree = "<group>"; };
419E6EC72566D214002B5010 /* RTCPeerConnectionIceErrorEvent.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = RTCPeerConnectionIceErrorEvent.h; sourceTree = "<group>"; };
+ 419E6ED22567BF69002B5010 /* RTCRtpScriptTransformerContext.idl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = RTCRtpScriptTransformerContext.idl; sourceTree = "<group>"; };
+ 419E6ED52567BF6A002B5010 /* RTCRtpScriptTransformerContext.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RTCRtpScriptTransformerContext.h; sourceTree = "<group>"; };
+ 419E6ED82567D8D9002B5010 /* LibWebRTCRtpTransformableFrame.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = LibWebRTCRtpTransformableFrame.cpp; path = libwebrtc/LibWebRTCRtpTransformableFrame.cpp; sourceTree = "<group>"; };
419FAFAD1ABABCD5005B3572 /* ReadableStreamBYOBReader.idl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = ReadableStreamBYOBReader.idl; sourceTree = "<group>"; };
419FAFAD1ABABCD5005B828B /* ReadableStreamDefaultReader.idl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = ReadableStreamDefaultReader.idl; sourceTree = "<group>"; };
41A023ED1A39DB7900F722CF /* ReadableStream.idl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = ReadableStream.idl; sourceTree = "<group>"; };
@@ -17186,6 +17189,8 @@
414189F12562DF5600363417 /* RTCRtpScriptTransformer.idl */,
414189F62562E67A00363417 /* RTCRtpScriptTransformerConstructor.h */,
414189F72562E67A00363417 /* RTCRtpScriptTransformerConstructor.idl */,
+ 419E6ED52567BF6A002B5010 /* RTCRtpScriptTransformerContext.h */,
+ 419E6ED22567BF69002B5010 /* RTCRtpScriptTransformerContext.idl */,
414189EF2562DF0C00363417 /* RTCRtpScriptTransformProvider.idl */,
413C8B262552EEA900E65055 /* RTCRtpSender+Transform.idl */,
5E2C43591BCEE30D0001E2BC /* RTCRtpSender.cpp */,
