diff options
Diffstat (limited to 'content')
| -rw-r--r-- | content/browser/media/webrtc_browsertest.cc | 24 | ||||
| -rw-r--r-- | content/renderer/media/media_stream_center.cc | 19 | ||||
| -rw-r--r-- | content/renderer/media/media_stream_center.h | 8 | ||||
| -rw-r--r-- | content/renderer/media/media_stream_dependency_factory.cc | 146 | ||||
| -rw-r--r-- | content/renderer/media/media_stream_dependency_factory.h | 8 | ||||
| -rw-r--r-- | content/renderer/media/media_stream_dependency_factory_unittest.cc | 59 | ||||
| -rw-r--r-- | content/renderer/media/mock_media_stream_dependency_factory.cc | 16 | ||||
| -rw-r--r-- | content/test/data/media/getusermedia.html | 76 | ||||
| -rw-r--r-- | content/test/data/media/getusermedia_and_stop.html | 18 | ||||
| -rw-r--r-- | content/test/data/media/peerconnection-call.html | 98 | ||||
| -rw-r--r-- | content/test/data/media/webrtc_test_utilities.js | 80 |
11 files changed, 379 insertions, 173 deletions
diff --git a/content/browser/media/webrtc_browsertest.cc b/content/browser/media/webrtc_browsertest.cc index fd6c169..fe52307 100644 --- a/content/browser/media/webrtc_browsertest.cc +++ b/content/browser/media/webrtc_browsertest.cc @@ -47,7 +47,7 @@ class WebrtcBrowserTest: public ContentBrowserTest { // see that the success callback is called. If the error callback is called or // none of the callbacks are called the tests will simply time out and fail. IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetVideoStreamAndStop) { - GURL url(test_server()->GetURL("files/media/getusermedia_and_stop.html")); + GURL url(test_server()->GetURL("files/media/getusermedia.html")); NavigateToURL(shell(), url); EXPECT_TRUE(ExecuteJavascript("getUserMedia({video: true});")); @@ -56,7 +56,7 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetVideoStreamAndStop) { } IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetAudioAndVideoStreamAndStop) { - GURL url(test_server()->GetURL("files/media/getusermedia_and_stop.html")); + GURL url(test_server()->GetURL("files/media/getusermedia.html")); NavigateToURL(shell(), url); EXPECT_TRUE(ExecuteJavascript("getUserMedia({video: true, audio: true});")); @@ -64,6 +64,15 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetAudioAndVideoStreamAndStop) { ExpectTitle("OK"); } +IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, GetAudioAndVideoStreamAndClone) { + GURL url(test_server()->GetURL("files/media/getusermedia.html")); + NavigateToURL(shell(), url); + + EXPECT_TRUE(ExecuteJavascript("getUserMediaAndClone();")); + + ExpectTitle("OK"); +} + // These tests will make a complete PeerConnection-based call and verify that // video is playing for the call. IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CanSetupVideoCall) { @@ -147,5 +156,16 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, ExpectTitle("OK"); } +// This test will make a PeerConnection-based call and send a new Video +// MediaStream that has been created based on a MediaStream created with +// getUserMedia. +IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CallWithNewVideoMediaStream) { + GURL url(test_server()->GetURL("files/media/peerconnection-call.html")); + NavigateToURL(shell(), url); + + EXPECT_TRUE(ExecuteJavascript("callWithNewVideoMediaStream();")); + ExpectTitle("OK"); +} + } // namespace content diff --git a/content/renderer/media/media_stream_center.cc b/content/renderer/media/media_stream_center.cc index 29f2363..14b0327 100644 --- a/content/renderer/media/media_stream_center.cc +++ b/content/renderer/media/media_stream_center.cc @@ -32,7 +32,6 @@ static webrtc::MediaStreamInterface* GetNativeMediaStream( return extra_data->stream(); } - static webrtc::MediaStreamTrackInterface* GetNativeMediaStreamTrack( const WebKit::WebMediaStream& stream, const WebKit::WebMediaStreamTrack& component) { @@ -99,4 +98,22 @@ void MediaStreamCenter::didCreateMediaStream( rtc_factory_->CreateNativeLocalMediaStream(&stream); } +bool MediaStreamCenter::didAddMediaStreamTrack( + const WebKit::WebMediaStream& stream, + const WebKit::WebMediaStreamTrack& track) { + if (!rtc_factory_) + return false; + + return rtc_factory_->AddNativeMediaStreamTrack(stream, track); +} + +bool MediaStreamCenter::didRemoveMediaStreamTrack( + const WebKit::WebMediaStream& stream, + const WebKit::WebMediaStreamTrack& track) { + if (!rtc_factory_) + return false; + + return rtc_factory_->RemoveNativeMediaStreamTrack(stream, track); +} + } // namespace content diff --git a/content/renderer/media/media_stream_center.h b/content/renderer/media/media_stream_center.h index bd125b5..de5f2e1 100644 --- a/content/renderer/media/media_stream_center.h +++ b/content/renderer/media/media_stream_center.h @@ -42,6 +42,14 @@ class CONTENT_EXPORT MediaStreamCenter virtual void didCreateMediaStream( WebKit::WebMediaStream& stream) OVERRIDE; + virtual bool didAddMediaStreamTrack( + const WebKit::WebMediaStream& stream, + const WebKit::WebMediaStreamTrack& track) OVERRIDE; + + virtual bool didRemoveMediaStreamTrack( + const WebKit::WebMediaStream& stream, + const WebKit::WebMediaStreamTrack& track) OVERRIDE; + private: // |rtc_factory_| is a weak pointer and is owned by the RenderThreadImpl. // It is valid as long as RenderThreadImpl exist. diff --git a/content/renderer/media/media_stream_dependency_factory.cc b/content/renderer/media/media_stream_dependency_factory.cc index 4e24b58..1a27722 100644 --- a/content/renderer/media/media_stream_dependency_factory.cc +++ b/content/renderer/media/media_stream_dependency_factory.cc @@ -66,7 +66,7 @@ class WebAudioConstraints : public RTCMediaConstraints { } } - virtual ~WebAudioConstraints() {}; + virtual ~WebAudioConstraints() {} }; class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface { @@ -309,92 +309,118 @@ void MediaStreamDependencyFactory::CreateNativeLocalMediaStream( WebKit::WebMediaStream* description) { DVLOG(1) << "MediaStreamDependencyFactory::CreateNativeLocalMediaStream()"; if (!EnsurePeerConnectionFactory()) { - DVLOG(1) << "EnsurePeerConnectionFactory() failed!"; - return; + DVLOG(1) << "EnsurePeerConnectionFactory() failed!"; + return; } std::string label = UTF16ToUTF8(description->label()); scoped_refptr<webrtc::MediaStreamInterface> native_stream = CreateLocalMediaStream(label); + MediaStreamExtraData* extra_data = new MediaStreamExtraData(native_stream, + true); + description->setExtraData(extra_data); // Add audio tracks. WebKit::WebVector<WebKit::WebMediaStreamTrack> audio_tracks; - description->audioSources(audio_tracks); - - bool start_stream = false; + description->audioTracks(audio_tracks); for (size_t i = 0; i < audio_tracks.size(); ++i) { - WebKit::WebMediaStreamSource source = audio_tracks[i].source(); + AddNativeMediaStreamTrack(*description, audio_tracks[i]); + } - // See if we're adding a WebAudio MediaStream. + // Add video tracks. + WebKit::WebVector<WebKit::WebMediaStreamTrack> video_tracks; + description->videoTracks(video_tracks); + for (size_t i = 0; i < video_tracks.size(); ++i) { + AddNativeMediaStreamTrack(*description, video_tracks[i]); + } +} + +void MediaStreamDependencyFactory::CreateNativeLocalMediaStream( + WebKit::WebMediaStream* description, + const MediaStreamExtraData::StreamStopCallback& stream_stop) { + CreateNativeLocalMediaStream(description); + + MediaStreamExtraData* extra_data = + static_cast<MediaStreamExtraData*>(description->extraData()); + extra_data->SetLocalStreamStopCallback(stream_stop); +} + +bool MediaStreamDependencyFactory::AddNativeMediaStreamTrack( + const WebKit::WebMediaStream& stream, + const WebKit::WebMediaStreamTrack& track) { + MediaStreamExtraData* extra_data = + static_cast<MediaStreamExtraData*>(stream.extraData()); + webrtc::MediaStreamInterface* native_stream = extra_data->stream(); + DCHECK(native_stream); + + WebKit::WebMediaStreamSource source = track.source(); + MediaStreamSourceExtraData* source_data = + static_cast<MediaStreamSourceExtraData*>(source.extraData()); + + if (!source_data) { if (source.requiresAudioConsumer()) { + // We're adding a WebAudio MediaStream. // TODO(crogers, xians): In reality we should be able to send a unique - // audio stream to each PeerConnection separately. But currently WebRTC - // is only able to handle a global audio stream sent to ALL peers. - - // Create a special source where default WebAudio constraints are used. - if (!CreateWebAudioSource(&source)) { + // audio stream to each PeerConnection separately. + // Currently WebRTC is only able to handle a global audio stream sent to + // ALL peers. Create a special source where default WebAudio constraints + // are used. + if (CreateWebAudioSource(&source)) { + source_data = + static_cast<MediaStreamSourceExtraData*>(source.extraData()); + } else { LOG(ERROR) << "Failed to create WebAudio source"; - continue; + return false; } - } - - MediaStreamSourceExtraData* source_data = - static_cast<MediaStreamSourceExtraData*>(source.extraData()); - - if (!source_data) { + } else { // TODO(perkj): Implement support for sources from // remote MediaStreams. NOTIMPLEMENTED(); - continue; + return false; } - - scoped_refptr<webrtc::AudioTrackInterface> audio_track( - CreateLocalAudioTrack(UTF16ToUTF8(audio_tracks[i].id()), - source_data->local_audio_source())); - native_stream->AddTrack(audio_track); - audio_track->set_enabled(audio_tracks[i].isEnabled()); - start_stream = true; } - if (start_stream && GetWebRtcAudioDevice()) { - WebRtcAudioCapturer* capturer = GetWebRtcAudioDevice()->capturer(); - capturer->Start(); - } + WebKit::WebMediaStreamSource::Type type = track.source().type(); + DCHECK(type == WebKit::WebMediaStreamSource::TypeAudio || + type == WebKit::WebMediaStreamSource::TypeVideo); - // Add video tracks. - WebKit::WebVector<WebKit::WebMediaStreamTrack> video_tracks; - description->videoSources(video_tracks); - for (size_t i = 0; i < video_tracks.size(); ++i) { - const WebKit::WebMediaStreamSource& source = video_tracks[i].source(); - MediaStreamSourceExtraData* source_data = - static_cast<MediaStreamSourceExtraData*>(source.extraData()); - if (!source_data || !source_data->video_source()) { - // TODO(perkj): Implement support for sources from remote MediaStreams. - NOTIMPLEMENTED(); - continue; + std::string track_id = UTF16ToUTF8(track.id()); + if (source.type() == WebKit::WebMediaStreamSource::TypeAudio) { + // TODO(henrika,xians): Refactor how an audio track is created to harmonize + // with video tracks. + scoped_refptr<webrtc::AudioTrackInterface> audio_track( + CreateLocalAudioTrack(track_id, source_data->local_audio_source())); + audio_track->set_enabled(track.isEnabled()); + if (GetWebRtcAudioDevice()) { + WebRtcAudioCapturer* capturer = GetWebRtcAudioDevice()->capturer(); + if (!capturer->is_recording()) + capturer->Start(); } - + return native_stream->AddTrack(audio_track); + } else { scoped_refptr<webrtc::VideoTrackInterface> video_track( - CreateLocalVideoTrack(UTF16ToUTF8(video_tracks[i].id()), - source_data->video_source())); - - native_stream->AddTrack(video_track); - video_track->set_enabled(video_tracks[i].isEnabled()); + CreateLocalVideoTrack(track_id, source_data->video_source())); + video_track->set_enabled(track.isEnabled()); + return native_stream->AddTrack(video_track); } - - MediaStreamExtraData* extra_data = new MediaStreamExtraData(native_stream, - true); - description->setExtraData(extra_data); } -void MediaStreamDependencyFactory::CreateNativeLocalMediaStream( - WebKit::WebMediaStream* description, - const MediaStreamExtraData::StreamStopCallback& stream_stop) { - CreateNativeLocalMediaStream(description); - +bool MediaStreamDependencyFactory::RemoveNativeMediaStreamTrack( + const WebKit::WebMediaStream& stream, + const WebKit::WebMediaStreamTrack& track) { MediaStreamExtraData* extra_data = - static_cast<MediaStreamExtraData*>(description->extraData()); - extra_data->SetLocalStreamStopCallback(stream_stop); + static_cast<MediaStreamExtraData*>(stream.extraData()); + webrtc::MediaStreamInterface* native_stream = extra_data->stream(); + DCHECK(native_stream); + + WebKit::WebMediaStreamSource::Type type = track.source().type(); + DCHECK(type == WebKit::WebMediaStreamSource::TypeAudio || + type == WebKit::WebMediaStreamSource::TypeVideo); + + std::string track_id = UTF16ToUTF8(track.id()); + return type == WebKit::WebMediaStreamSource::TypeAudio ? + native_stream->RemoveTrack(native_stream->FindAudioTrack(track_id)) : + native_stream->RemoveTrack(native_stream->FindVideoTrack(track_id)); } bool MediaStreamDependencyFactory::CreatePeerConnectionFactory() { diff --git a/content/renderer/media/media_stream_dependency_factory.h b/content/renderer/media/media_stream_dependency_factory.h index 27e4f4a..2bdd8ff 100644 --- a/content/renderer/media/media_stream_dependency_factory.h +++ b/content/renderer/media/media_stream_dependency_factory.h @@ -89,6 +89,14 @@ class CONTENT_EXPORT MediaStreamDependencyFactory WebKit::WebMediaStream* description, const MediaStreamExtraData::StreamStopCallback& stream_stop); + // Adds a libjingle representation of a MediaStreamTrack to |stream| based + // on the source of |track|. + bool AddNativeMediaStreamTrack(const WebKit::WebMediaStream& stream, + const WebKit::WebMediaStreamTrack& track); + + bool RemoveNativeMediaStreamTrack(const WebKit::WebMediaStream& stream, + const WebKit::WebMediaStreamTrack& track); + // Asks the libjingle PeerConnection factory to create a libjingle // PeerConnection object. // The PeerConnection object is owned by PeerConnectionHandler. diff --git a/content/renderer/media/media_stream_dependency_factory_unittest.cc b/content/renderer/media/media_stream_dependency_factory_unittest.cc index a44f134..879c425 100644 --- a/content/renderer/media/media_stream_dependency_factory_unittest.cc +++ b/content/renderer/media/media_stream_dependency_factory_unittest.cc @@ -49,10 +49,9 @@ class MediaStreamDependencyFactoryTest : public ::testing::Test { dependency_factory_.reset(new MockMediaStreamDependencyFactory()); } - WebKit::WebMediaStream CreateWebKitMediaStream(bool audio, - bool video) { + WebKit::WebMediaStream CreateWebKitMediaStream(bool audio, bool video) { WebKit::WebVector<WebKit::WebMediaStreamSource> audio_sources( - audio ? static_cast<size_t>(1) : 0); + audio ? static_cast<size_t>(1) : 0); WebKit::WebVector<WebKit::WebMediaStreamSource> video_sources( video ? static_cast<size_t>(1) : 0); @@ -108,6 +107,17 @@ class MediaStreamDependencyFactoryTest : public ::testing::Test { EXPECT_TRUE(observer.description() == descriptor); } + void VerifyMediaStream(const WebKit::WebMediaStream& stream_desc, + size_t num_audio_tracks, + size_t num_video_tracks) { + content::MediaStreamExtraData* extra_data = + static_cast<content::MediaStreamExtraData*>(stream_desc.extraData()); + ASSERT_TRUE(extra_data && extra_data->stream()); + EXPECT_TRUE(extra_data->is_local()); + EXPECT_EQ(num_audio_tracks, extra_data->stream()->GetAudioTracks().size()); + EXPECT_EQ(num_video_tracks, extra_data->stream()->GetVideoTracks().size()); + } + protected: scoped_ptr<MockMediaStreamDependencyFactory> dependency_factory_; }; @@ -124,13 +134,7 @@ TEST_F(MediaStreamDependencyFactoryTest, CreateNativeMediaStream) { CreateNativeSources(&stream_desc); dependency_factory_->CreateNativeLocalMediaStream(&stream_desc); - - content::MediaStreamExtraData* extra_data = - static_cast<content::MediaStreamExtraData*>(stream_desc.extraData()); - ASSERT_TRUE(extra_data && extra_data->stream()); - EXPECT_TRUE(extra_data->is_local()); - EXPECT_EQ(1u, extra_data->stream()->GetAudioTracks().size()); - EXPECT_EQ(1u, extra_data->stream()->GetVideoTracks().size()); + VerifyMediaStream(stream_desc, 1, 1); } // Test that we don't crash if a MediaStream is created in WebKit with unknown @@ -153,12 +157,35 @@ TEST_F(MediaStreamDependencyFactoryTest, CreateNativeMediaStreamWithoutSource) { EXPECT_TRUE(dependency_factory_->EnsurePeerConnectionFactory()); dependency_factory_->CreateNativeLocalMediaStream(&stream_desc); - MediaStreamExtraData* extra_data = static_cast<MediaStreamExtraData*>( - stream_desc.extraData()); - ASSERT_TRUE(extra_data && extra_data->stream()); - EXPECT_TRUE(extra_data->is_local()); - EXPECT_EQ(0u, extra_data->stream()->GetVideoTracks().size()); - EXPECT_EQ(0u, extra_data->stream()->GetAudioTracks().size()); + VerifyMediaStream(stream_desc, 0, 0); +} + +TEST_F(MediaStreamDependencyFactoryTest, AddAndRemoveNativeTrack) { + WebKit::WebMediaStream stream_desc = CreateWebKitMediaStream(true, true); + CreateNativeSources(&stream_desc); + + dependency_factory_->CreateNativeLocalMediaStream(&stream_desc); + VerifyMediaStream(stream_desc, 1, 1); + + WebKit::WebVector<WebKit::WebMediaStreamTrack> audio_tracks; + stream_desc.audioTracks(audio_tracks); + EXPECT_TRUE(dependency_factory_->RemoveNativeMediaStreamTrack( + stream_desc, audio_tracks[0])); + VerifyMediaStream(stream_desc, 0, 1); + + EXPECT_TRUE(dependency_factory_->AddNativeMediaStreamTrack( + stream_desc, audio_tracks[0])); + VerifyMediaStream(stream_desc, 1, 1); + + WebKit::WebVector<WebKit::WebMediaStreamTrack> video_tracks; + stream_desc.videoTracks(video_tracks); + EXPECT_TRUE(dependency_factory_->RemoveNativeMediaStreamTrack( + stream_desc, video_tracks[0])); + VerifyMediaStream(stream_desc, 1, 0); + + EXPECT_TRUE(dependency_factory_->AddNativeMediaStreamTrack( + stream_desc, video_tracks[0])); + VerifyMediaStream(stream_desc, 1, 1); } } // namespace content diff --git a/content/renderer/media/mock_media_stream_dependency_factory.cc b/content/renderer/media/mock_media_stream_dependency_factory.cc index f2656c0..3920a71 100644 --- a/content/renderer/media/mock_media_stream_dependency_factory.cc +++ b/content/renderer/media/mock_media_stream_dependency_factory.cc @@ -50,12 +50,20 @@ class MockMediaStream : public webrtc::MediaStreamInterface { return true; } virtual bool RemoveTrack(AudioTrackInterface* track) OVERRIDE { - NOTIMPLEMENTED(); - return false; + AudioTrackVector::iterator it = FindTrack(&audio_track_vector_, + track->id()); + if (it == audio_track_vector_.end()) + return false; + audio_track_vector_.erase(it); + return true; } virtual bool RemoveTrack(VideoTrackInterface* track) OVERRIDE { - NOTIMPLEMENTED(); - return false; + VideoTrackVector::iterator it = FindTrack(&video_track_vector_, + track->id()); + if (it == video_track_vector_.end()) + return false; + video_track_vector_.erase(it); + return true; } virtual std::string label() const OVERRIDE { return label_; } virtual AudioTrackVector GetAudioTracks() OVERRIDE { diff --git a/content/test/data/media/getusermedia.html b/content/test/data/media/getusermedia.html new file mode 100644 index 0000000..429dd66 --- /dev/null +++ b/content/test/data/media/getusermedia.html @@ -0,0 +1,76 @@ +<html> +<head> + <script type="text/javascript" src="webrtc_test_utilities.js"></script> + <script type="text/javascript"> + $ = function(id) { + return document.getElementById(id); + }; + + var gLocalStream = null; + + setAllEventsOccuredHandler(function() { + gLocalStream.stop(); + document.title = 'OK'; + }); + + // This test that a MediaStream can be created and a local preview + // rendered. + function getUserMedia(constraints) { + navigator.webkitGetUserMedia(constraints, displayAndWaitForVideo, + failedCallback); + } + + // This test that a MediaStream can be cloned and that the clone can + // be rendered. + function getUserMediaAndClone() { + navigator.webkitGetUserMedia({video: true, audio: true}, + createAndRenderClone, failedCallback); + } + + function failedCallback(error) { + document.title = 'GetUserMedia call failed with code ' + error.code; + } + + function displayAndWaitForVideo(stream) { + gLocalStream = stream; + var localStreamUrl = webkitURL.createObjectURL(stream); + $('local-view').src = localStreamUrl; + waitForVideo('local-view'); + } + + function createAndRenderClone(stream) { + gLocalStream = stream; + // TODO(perkj): --use-fake-device-for-media-stream do not currently + // work with audio devices and not all bots has a microphone. + new_stream = new webkitMediaStream(); + new_stream.addTrack(stream.getVideoTracks()[0]); + expectEquals(new_stream.getVideoTracks().length, 1); + if (stream.getAudioTracks().length > 0) { + new_stream.addTrack(stream.getAudioTracks()[0]); + expectEquals(new_stream.getAudioTracks().length, 1); + new_stream.removeTrack(new_stream.getAudioTracks()[0]); + expectEquals(new_stream.getAudioTracks().length, 0); + } + + var newStreamUrl = webkitURL.createObjectURL(new_stream); + $('local-view').src = newStreamUrl; + waitForVideo('local-view'); + } + + </script> +</head> +<body> + <table border="0"> + <tr> + <td>Local Preview</td> + </tr> + <tr> + <td><video width="320" height="240" id="local-view" + autoplay="autoplay"></video></td> + <!-- Canvases are named after their corresponding video elements. --> + <td><canvas width="320" height="240" id="local-view-canvas" + style="display:none"></canvas></td> + </tr> + </table> +</body> +</html>
\ No newline at end of file diff --git a/content/test/data/media/getusermedia_and_stop.html b/content/test/data/media/getusermedia_and_stop.html deleted file mode 100644 index f1c43be..0000000 --- a/content/test/data/media/getusermedia_and_stop.html +++ /dev/null @@ -1,18 +0,0 @@ -<html> -<head> - <script type="text/javascript"> - function getUserMedia(constraints) { - navigator.webkitGetUserMedia(constraints, okCallback, failedCallback); - } - - function failedCallback(error) { - document.title = 'GetUserMedia call failed with code ' + error.code; - } - - function okCallback(stream) { - stream.stop(); - document.title = 'OK'; - } - </script> -</head> -</html>
\ No newline at end of file diff --git a/content/test/data/media/peerconnection-call.html b/content/test/data/media/peerconnection-call.html index 7581b72..39b01b5 100644 --- a/content/test/data/media/peerconnection-call.html +++ b/content/test/data/media/peerconnection-call.html @@ -1,28 +1,22 @@ <html> <head> + <script type="text/javascript" src="webrtc_test_utilities.js"></script> <script type="text/javascript"> $ = function(id) { return document.getElementById(id); }; - // These must match with how the video and canvas tags are declared in html. - const VIDEO_TAG_WIDTH = 320; - const VIDEO_TAG_HEIGHT = 240; - var gFirstConnection = null; var gSecondConnection = null; var gTestWithoutMsidAndBundle = false; - // Number of test events to occur before the test pass. When the test pass, - // the document title change to OK. - var gNumberOfExpectedEvents = 0; - - // Number of events that currently have occured. - var gNumberOfEvents = 0; - var gLocalStream = null; var gSentTones = ''; + setAllEventsOccuredHandler(function() { + document.title = 'OK'; + }); + // Test that we can setup call with an audio and video track. function call(constraints) { createConnections(null); @@ -109,6 +103,16 @@ // Do the DTMF test after we have received video. detectVideoIn('remote-view-2', onCallEstablished); } + + // Test call with a new Video MediaStream that has been created based on a + // stream generated by getUserMedia. + function callWithNewVideoMediaStream() { + createConnections(null); + navigator.webkitGetUserMedia({audio:true, video:true}, + createNewVideoStreamAndAddToBothConnections, printGetUserMediaError); + waitForVideo('remote-view-1'); + waitForVideo('remote-view-2'); + } // This function is used for setting up a test that: // 1. Creates a data channel on |gFirstConnection| and sends data to @@ -155,7 +159,7 @@ secondDataChannel.send(sendDataString); } } - } + } function onToneChange(tone) { gSentTones += tone.tone; @@ -203,6 +207,15 @@ gFirstConnection.addStream(localStream); negotiate(); } + + // Called if getUserMedia succeeds when we want to send a modified + // MediaStream. A new MediaStream is created and the video track from + // |localStream| is added. + function createNewVideoStreamAndAddToBothConnections(localStream) { + var new_stream = new webkitMediaStream(); + new_stream.addTrack(localStream.getVideoTracks()[0]); + addStreamToBothConnectionsAndNegotiate(new_stream); + } function negotiate() { gFirstConnection.createOffer(onOfferCreated); @@ -275,66 +288,7 @@ var remoteVideo = $(target); remoteVideo.src = remoteStreamUrl; } - - // TODO(phoglund): perhaps use the video detector in chrome/test/data/webrtc/? - function detectVideoIn(videoElementName, callback) { - var width = VIDEO_TAG_WIDTH; - var height = VIDEO_TAG_HEIGHT; - var videoElement = $(videoElementName); - var canvas = $(videoElementName + '-canvas'); - var waitVideo = setInterval(function() { - var context = canvas.getContext('2d'); - context.drawImage(videoElement, 0, 0, width, height); - var pixels = context.getImageData(0, 0, width, height).data; - - if (isVideoPlaying(pixels, width, height)) { - clearInterval(waitVideo); - callback(); - } - }, 100); - } - - function waitForVideo(videoElement) { - document.title = 'Waiting for video...'; - addExpectedEvent(); - detectVideoIn(videoElement, function () { eventOccured(); }); - } - - // This very basic video verification algorithm will be satisfied if any - // pixels are nonzero in a small sample area in the middle. It relies on the - // assumption that a video element with null source just presents zeroes. - function isVideoPlaying(pixels, width, height) { - // Sample somewhere near the middle of the image. - var middle = width * height / 2; - for (var i = 0; i < 20; i++) { - if (pixels[middle + i] > 0) { - return true; - } - } - return false; - } - - - // This function matches |left| and |right| and throws an exception if the - // values don't match. - function expectEquals(left, right) { - if (left != right) { - var s = "expectEquals failed left: " + left + " right: " + right; - document.title = s; - throw s; - } - } - - function addExpectedEvent() { - ++gNumberOfExpectedEvents; - } - - function eventOccured() { - ++gNumberOfEvents; - if (gNumberOfEvents == gNumberOfExpectedEvents) { - document.title = 'OK'; - } - } + </script> </head> <body> diff --git a/content/test/data/media/webrtc_test_utilities.js b/content/test/data/media/webrtc_test_utilities.js new file mode 100644 index 0000000..dae549f --- /dev/null +++ b/content/test/data/media/webrtc_test_utilities.js @@ -0,0 +1,80 @@ +// Copyright (c) 2012 The Chromium Authors. All rights reserved. +// Use of this source code is governed by a BSD-style license that can be +// found in the LICENSE file. + +// These must match with how the video and canvas tags are declared in html. +const VIDEO_TAG_WIDTH = 320; +const VIDEO_TAG_HEIGHT = 240; + +// Number of test events to occur before the test pass. When the test pass, +// the function gAllEventsOccured is called. +var gNumberOfExpectedEvents = 0; + +// Number of events that currently have occurred. +var gNumberOfEvents = 0; + +var gAllEventsOccured = function () {}; + +// Use this function to set a function that will be called once all expected +// events has occurred. +function setAllEventsOccuredHandler(handler) { + gAllEventsOccured = handler; +} + +function detectVideoIn(videoElementName, callback) { + var width = VIDEO_TAG_WIDTH; + var height = VIDEO_TAG_HEIGHT; + var videoElement = $(videoElementName); + var canvas = $(videoElementName + '-canvas'); + var waitVideo = setInterval(function() { + var context = canvas.getContext('2d'); + context.drawImage(videoElement, 0, 0, width, height); + var pixels = context.getImageData(0, 0, width, height).data; + + if (isVideoPlaying(pixels, width, height)) { + clearInterval(waitVideo); + callback(); + } + }, 100); +} + +function waitForVideo(videoElement) { + document.title = 'Waiting for video...'; + addExpectedEvent(); + detectVideoIn(videoElement, function () { eventOccured(); }); +} + +function addExpectedEvent() { + ++gNumberOfExpectedEvents; +} + +function eventOccured() { + ++gNumberOfEvents; + if (gNumberOfEvents == gNumberOfExpectedEvents) { + gAllEventsOccured(); + } +} + +// This very basic video verification algorithm will be satisfied if any +// pixels are nonzero in a small sample area in the middle. It relies on the +// assumption that a video element with null source just presents zeroes. +function isVideoPlaying(pixels, width, height) { + // Sample somewhere near the middle of the image. + var middle = width * height / 2; + for (var i = 0; i < 20; i++) { + if (pixels[middle + i] > 0) { + return true; + } + } + return false; +} + +// This function matches |left| and |right| and throws an exception if the +// values don't match. +function expectEquals(left, right) { + if (left != right) { + var s = "expectEquals failed left: " + left + " right: " + right; + document.title = s; + throw s; + } +}
\ No newline at end of file |