Diff
Modified: trunk/LayoutTests/ChangeLog (214043 => 214044)
--- trunk/LayoutTests/ChangeLog 2017-03-16 15:23:00 UTC (rev 214043)
+++ trunk/LayoutTests/ChangeLog 2017-03-16 16:09:50 UTC (rev 214044)
@@ -1,3 +1,23 @@
+2017-03-16 Youenn Fablet <you...@apple.com>
+
+ Improve WebRTC track enabled support
+ https://bugs.webkit.org/show_bug.cgi?id=169727
+
+ Reviewed by Alex Christensen.
+
+ * TestExpectations:
+ * webrtc/audio-peer-connection-webaudio.html:
+ * webrtc/peer-connection-audio-mute-expected.txt:
+ * webrtc/peer-connection-audio-mute.html:
+ * webrtc/peer-connection-audio-mute2-expected.txt: Added.
+ * webrtc/peer-connection-audio-mute2.html: Added.
+ * webrtc/peer-connection-remote-audio-mute-expected.txt: Added.
+ * webrtc/peer-connection-remote-audio-mute.html: Added.
+ * webrtc/video-mute-expected.txt:
+ * webrtc/video-mute.html:
+ * webrtc/video-remote-mute-expected.txt: Added.
+ * webrtc/video-remote-mute.html: Added.
+
2017-03-16 Manuel Rego Casasnovas <r...@igalia.com>
[css-grid] Crash on debug removing a positioned child
Modified: trunk/LayoutTests/TestExpectations (214043 => 214044)
--- trunk/LayoutTests/TestExpectations 2017-03-16 15:23:00 UTC (rev 214043)
+++ trunk/LayoutTests/TestExpectations 2017-03-16 16:09:50 UTC (rev 214044)
@@ -711,7 +711,10 @@
# GTK enables some of these tests on their TestExpectations file.
[ Release ] webrtc [ Skip ]
-[ Debug ] webrtc/audio-peer-connection-webaudio.html [ Failure ]
+[ Debug ] webrtc/peer-connection-audio-mute.html [ Pass Failure ]
+[ Debug ] webrtc/peer-connection-audio-mute2.html [ Pass Failure ]
+[ Debug ] webrtc/peer-connection-remote-audio-mute.html [ Pass Failure ]
+[ Debug ] webrtc/peer-connection-remote-audio-mute2.html [ Pass Failure ]
fast/mediastream/getUserMedia-webaudio.html [ Skip ]
fast/mediastream/RTCPeerConnection-AddRemoveStream.html [ Skip ]
fast/mediastream/RTCPeerConnection-closed-state.html [ Skip ]
Modified: trunk/LayoutTests/webrtc/audio-peer-connection-webaudio.html (214043 => 214044)
--- trunk/LayoutTests/webrtc/audio-peer-connection-webaudio.html 2017-03-16 15:23:00 UTC (rev 214043)
+++ trunk/LayoutTests/webrtc/audio-peer-connection-webaudio.html 2017-03-16 16:09:50 UTC (rev 214044)
@@ -11,7 +11,7 @@
if (window.testRunner)
testRunner.setUserMediaPermission(true);
- return navigator.mediaDevices.getUserMedia({audio: true}).then((stream) => {
+ return navigator.mediaDevices.getUserMedia({audio: true}).then((stream) => {
if (window.internals)
internals.useMockRTCPeerConnectionFactory("TwoRealPeerConnections");
return new Promise((resolve, reject) => {
@@ -21,14 +21,14 @@
secondConnection._onaddstream_ = (streamEvent) => { resolve(streamEvent.stream); };
});
setTimeout(() => reject("Test timed out"), 5000);
- }).then((stream) => {
- return analyseAudio(stream, 1000);
- }).then((results) => {
- assert_true(results.heardHum, "heard hum");
- assert_true(results.heardBip, "heard bip");
- assert_true(results.heardBop, "heard bop");
});
- });
+ }).then((remoteStream) => {
+ return analyseAudio(remoteStream, 1000);
+ }).then((results) => {
+ assert_true(results.heardHum, "heard hum");
+ assert_true(results.heardBip, "heard bip");
+ assert_true(results.heardBop, "heard bop");
+ });
}, "Basic audio playback through a peer connection");
</script>
</head>
Modified: trunk/LayoutTests/webrtc/peer-connection-audio-mute-expected.txt (214043 => 214044)
--- trunk/LayoutTests/webrtc/peer-connection-audio-mute-expected.txt 2017-03-16 15:23:00 UTC (rev 214043)
+++ trunk/LayoutTests/webrtc/peer-connection-audio-mute-expected.txt 2017-03-16 16:09:50 UTC (rev 214044)
@@ -1,3 +1,3 @@
-FAIL Muting and unmuting an audio track assert_true: heard hum expected true got false
+PASS Muting a local audio track and making sure the remote track is silent
Modified: trunk/LayoutTests/webrtc/peer-connection-audio-mute.html (214043 => 214044)
--- trunk/LayoutTests/webrtc/peer-connection-audio-mute.html 2017-03-16 15:23:00 UTC (rev 214043)
+++ trunk/LayoutTests/webrtc/peer-connection-audio-mute.html 2017-03-16 16:09:50 UTC (rev 214044)
@@ -13,17 +13,19 @@
if (window.testRunner)
testRunner.setUserMediaPermission(true);
- return navigator.mediaDevices.getUserMedia({audio: true}).then((stream) => {
+ var localTrack;
+ return navigator.mediaDevices.getUserMedia({audio: true}).then((localStream) => {
if (window.internals)
internals.useMockRTCPeerConnectionFactory("TwoRealPeerConnections");
- var stream;
+ localTrack = localStream.getAudioTracks()[0];
+ var remoteStream;
return new Promise((resolve, reject) => {
createConnections((firstConnection) => {
- firstConnection.addStream(stream);
+ firstConnection.addStream(localStream);
}, (secondConnection) => {
secondConnection._onaddstream_ = (streamEvent) => {
- stream = streamEvent.stream;
+ remoteStream = streamEvent.stream;
resolve();
};
});
@@ -30,36 +32,19 @@
}).then(() => {
return waitFor(500);
}).then(() => {
- return analyseAudio(stream, 500).then((results) => {
- assert_true(results.heardHum, "heard hum");
- assert_true(results.heardBip, "heard bip");
- assert_true(results.heardBop, "heard bop");
+ return analyseAudio(remoteStream, 500).then((results) => {
+ assert_true(results.heardHum, "heard hum from remote enabled track");
});
}).then(() => {
- stream.getAudioTracks().forEach((track) => {
- track.enabled = false;
- });
+ localTrack.enabled = false;
return waitFor(500);
}).then(() => {
- return analyseAudio(stream, 500).then((results) => {
- assert_false(results.heardHum, "heard hum");
- assert_false(results.heardBip, "heard bip");
- assert_false(results.heardBop, "heard bop");
+ return analyseAudio(remoteStream, 500).then((results) => {
+ assert_false(results.heardHum, "not heard hum from remote disabled track");
});
- }).then(() => {
- stream.getAudioTracks().forEach((track) => {
- track.enabled = true;
- });
- return waitFor(500);
- }).then(() => {
- return analyseAudio(stream, 500).then((results) => {
- assert_true(results.heardHum, "heard hum");
- assert_true(results.heardBip, "heard bip");
- assert_true(results.heardBop, "heard bop");
- });
});
});
- }, "Muting and unmuting an audio track");
+ }, "Muting a local audio track and making sure the remote track is silent");
</script>
</body>
</html>
Added: trunk/LayoutTests/webrtc/peer-connection-audio-mute2-expected.txt (0 => 214044)
--- trunk/LayoutTests/webrtc/peer-connection-audio-mute2-expected.txt (rev 0)
+++ trunk/LayoutTests/webrtc/peer-connection-audio-mute2-expected.txt 2017-03-16 16:09:50 UTC (rev 214044)
@@ -0,0 +1,3 @@
+
+PASS Muting and unmuting a local audio track
+
Copied: trunk/LayoutTests/webrtc/peer-connection-audio-mute2.html (from rev 214043, trunk/LayoutTests/webrtc/peer-connection-audio-mute.html) (0 => 214044)
--- trunk/LayoutTests/webrtc/peer-connection-audio-mute2.html (rev 0)
+++ trunk/LayoutTests/webrtc/peer-connection-audio-mute2.html 2017-03-16 16:09:50 UTC (rev 214044)
@@ -0,0 +1,57 @@
+<!DOCTYPE html>
+<html>
+<head>
+ <meta charset="utf-8">
+ <title>Testing local audio capture playback causes "playing" event to fire</title>
+ <script src=""
+ <script src=""
+</head>
+<body>
+ <script src =""
+ <script>
+ promise_test((test) => {
+ if (window.testRunner)
+ testRunner.setUserMediaPermission(true);
+
+ var localTrack;
+ return navigator.mediaDevices.getUserMedia({audio: true}).then((localStream) => {
+ if (window.internals)
+ internals.useMockRTCPeerConnectionFactory("TwoRealPeerConnections");
+
+ localTrack = localStream.getAudioTracks()[0];
+ var remoteStream;
+ return new Promise((resolve, reject) => {
+ createConnections((firstConnection) => {
+ firstConnection.addStream(localStream);
+ }, (secondConnection) => {
+ secondConnection._onaddstream_ = (streamEvent) => {
+ remoteStream = streamEvent.stream;
+ resolve();
+ };
+ });
+ }).then(() => {
+ return waitFor(500);
+ }).then(() => {
+ return analyseAudio(remoteStream, 500).then((results) => {
+ assert_true(results.heardHum, "heard hum from remote enabled track");
+ });
+ }).then(() => {
+ localTrack.enabled = false;
+ return waitFor(500);
+ }).then(() => {
+ return analyseAudio(remoteStream, 500).then((results) => {
+ assert_false(results.heardHum, "not heard hum from remote disabled track");
+ });
+ }).then(() => {
+ localTrack.enabled = true;
+ return waitFor(500);
+ }).then(() => {
+ return analyseAudio(remoteStream, 500).then((results) => {
+ assert_true(results.heardHum, "heard hum from remote reenabled track");
+ });
+ });
+ });
+ }, "Muting and unmuting a local audio track");
+ </script>
+</body>
+</html>
Added: trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute-expected.txt (0 => 214044)
--- trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute-expected.txt (rev 0)
+++ trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute-expected.txt 2017-03-16 16:09:50 UTC (rev 214044)
@@ -0,0 +1,3 @@
+
+PASS Muting an incoming audio track
+
Added: trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute.html (0 => 214044)
--- trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute.html (rev 0)
+++ trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute.html 2017-03-16 16:09:50 UTC (rev 214044)
@@ -0,0 +1,47 @@
+<!DOCTYPE html>
+<html>
+<head>
+ <meta charset="utf-8">
+ <title>Testing local audio capture playback causes "playing" event to fire</title>
+ <script src=""
+ <script src=""
+</head>
+<body>
+ <script src =""
+ <script>
+ promise_test((test) => {
+ if (window.testRunner)
+ testRunner.setUserMediaPermission(true);
+
+ return navigator.mediaDevices.getUserMedia({audio: true}).then((localStream) => {
+ if (window.internals)
+ internals.useMockRTCPeerConnectionFactory("TwoRealPeerConnections");
+
+ var remoteTrack;
+ var remoteStream;
+ return new Promise((resolve, reject) => {
+ createConnections((firstConnection) => {
+ firstConnection.addStream(localStream);
+ }, (secondConnection) => {
+ secondConnection._onaddstream_ = (streamEvent) => {
+ remoteStream = streamEvent.stream;
+ remoteTrack = remoteStream.getAudioTracks()[0];
+ resolve();
+ };
+ });
+ }).then(() => {
+ return analyseAudio(remoteStream, 500).then((results) => {
+ assert_true(results.heardHum, "heard hum from remote enabled track");
+ });
+ }).then(() => {
+ remoteTrack.enabled = false;
+ }).then(() => {
+ return analyseAudio(remoteStream, 500).then((results) => {
+ assert_false(results.heardHum, "not heard hum from remote disabled track");
+ });
+ });
+ });
+ }, "Muting an incoming audio track");
+ </script>
+</body>
+</html>
Added: trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute2-expected.txt (0 => 214044)
--- trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute2-expected.txt (rev 0)
+++ trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute2-expected.txt 2017-03-16 16:09:50 UTC (rev 214044)
@@ -0,0 +1,3 @@
+
+PASS Muting and unmuting an incoming audio track
+
Added: trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute2.html (0 => 214044)
--- trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute2.html (rev 0)
+++ trunk/LayoutTests/webrtc/peer-connection-remote-audio-mute2.html 2017-03-16 16:09:50 UTC (rev 214044)
@@ -0,0 +1,53 @@
+<!DOCTYPE html>
+<html>
+<head>
+ <meta charset="utf-8">
+ <title>Testing local audio capture playback causes "playing" event to fire</title>
+ <script src=""
+ <script src=""
+</head>
+<body>
+ <script src =""
+ <script>
+ promise_test((test) => {
+ if (window.testRunner)
+ testRunner.setUserMediaPermission(true);
+
+ return navigator.mediaDevices.getUserMedia({audio: true}).then((localStream) => {
+ if (window.internals)
+ internals.useMockRTCPeerConnectionFactory("TwoRealPeerConnections");
+
+ var remoteTrack;
+ var remoteStream;
+ return new Promise((resolve, reject) => {
+ createConnections((firstConnection) => {
+ firstConnection.addStream(localStream);
+ }, (secondConnection) => {
+ secondConnection._onaddstream_ = (streamEvent) => {
+ remoteStream = streamEvent.stream;
+ remoteTrack = remoteStream.getAudioTracks()[0];
+ resolve();
+ };
+ });
+ }).then(() => {
+ return analyseAudio(remoteStream, 500).then((results) => {
+ assert_true(results.heardHum, "heard hum from remote enabled track");
+ });
+ }).then(() => {
+ remoteTrack.enabled = false;
+ }).then(() => {
+ return analyseAudio(remoteStream, 500).then((results) => {
+ assert_false(results.heardHum, "not heard hum from remote disabled track");
+ });
+ }).then(() => {
+ remoteTrack.enabled = true;
+ }).then(() => {
+ return analyseAudio(remoteStream, 500).then((results) => {
+ assert_true(results.heardHum, "heard hum from remote reenabled track");
+ });
+ });
+ });
+ }, "Muting and unmuting an incoming audio track");
+ </script>
+</body>
+</html>
Modified: trunk/LayoutTests/webrtc/video-mute-expected.txt (214043 => 214044)
--- trunk/LayoutTests/webrtc/video-mute-expected.txt 2017-03-16 15:23:00 UTC (rev 214043)
+++ trunk/LayoutTests/webrtc/video-mute-expected.txt 2017-03-16 16:09:50 UTC (rev 214044)
@@ -1,4 +1,4 @@
-PASS Video muted/unmuted track
+PASS Outgoing muted/unmuted video track
Modified: trunk/LayoutTests/webrtc/video-mute.html (214043 => 214044)
--- trunk/LayoutTests/webrtc/video-mute.html 2017-03-16 15:23:00 UTC (rev 214043)
+++ trunk/LayoutTests/webrtc/video-mute.html 2017-03-16 16:09:50 UTC (rev 214044)
@@ -21,10 +21,11 @@
canvas.height = video.videoHeight;
canvas.getContext('2d').drawImage(video, 0, 0, canvas.width, canvas.height);
- imageData = canvas.getContext('2d').getImageData(10, 325, 250, 1);
+ imageData = canvas.getContext('2d').getImageData(0, 0, canvas.width, canvas.height);
data = ""
for (var cptr = 0; cptr < canvas.width * canvas.height; ++cptr) {
- if (data[4 * cptr] || data[4 * cptr + 1] || data[4 * cptr + 2])
+ // Approximatively black pixels.
+ if (data[4 * cptr] > 10 || data[4 * cptr + 1] > 10 || data[4 * cptr + 2] > 10)
return false;
}
return true;
@@ -35,35 +36,36 @@
if (window.testRunner)
testRunner.setUserMediaPermission(true);
- return navigator.mediaDevices.getUserMedia({ video: true}).then((stream) => {
+ return navigator.mediaDevices.getUserMedia({ video: true}).then((localStream) => {
return new Promise((resolve, reject) => {
if (window.internals)
internals.useMockRTCPeerConnectionFactory("TwoRealPeerConnections");
+ track = localStream.getVideoTracks()[0];
+
createConnections((firstConnection) => {
- firstConnection.addStream(stream);
+ firstConnection.addStream(localStream);
}, (secondConnection) => {
secondConnection._onaddstream_ = (streamEvent) => { resolve(streamEvent.stream); };
});
setTimeout(() => reject("Test timed out"), 5000);
});
- }).then((stream) => {
- video.srcObject = stream;
- track = stream.getVideoTracks()[0];
+ }).then((remoteStream) => {
+ video.srcObject = remoteStream;
return video.play();
}).then(() => {
- assert_false(isVideoBlack());
+ assert_false(isVideoBlack(), "track is enabled, video is not black");
}).then(() => {
track.enabled = false;
return waitFor(500);
}).then(() => {
- assert_true(isVideoBlack());
+ assert_true(isVideoBlack(), "track is disabled, video is black");
track.enabled = true;
return waitFor(500);
}).then(() => {
- assert_false(isVideoBlack());
+ assert_false(isVideoBlack(), "track is reenabled, video is not black");
});
-}, "Video muted/unmuted track");
+}, "Outgoing muted/unmuted video track");
</script>
</body>
</html>
Added: trunk/LayoutTests/webrtc/video-remote-mute-expected.txt (0 => 214044)
--- trunk/LayoutTests/webrtc/video-remote-mute-expected.txt (rev 0)
+++ trunk/LayoutTests/webrtc/video-remote-mute-expected.txt 2017-03-16 16:09:50 UTC (rev 214044)
@@ -0,0 +1,4 @@
+
+
+PASS Incoming muted/unmuted video track
+
Copied: trunk/LayoutTests/webrtc/video-remote-mute.html (from rev 214043, trunk/LayoutTests/webrtc/video-mute.html) (0 => 214044)
--- trunk/LayoutTests/webrtc/video-remote-mute.html (rev 0)
+++ trunk/LayoutTests/webrtc/video-remote-mute.html 2017-03-16 16:09:50 UTC (rev 214044)
@@ -0,0 +1,69 @@
+<!doctype html>
+<html>
+ <head>
+ <meta charset="utf-8">
+ <title>Testing basic video exchange from offerer to receiver</title>
+ <script src=""
+ <script src=""
+ </head>
+ <body>
+ <video id="video" autoplay=""></video>
+ <canvas id="canvas" width="640" height="480"></canvas>
+ <script src =""
+ <script>
+video = document.getElementById("video");
+canvas = document.getElementById("canvas");
+// FIXME: We should use tracks
+
+function isVideoBlack()
+{
+ canvas.width = video.videoWidth;
+ canvas.height = video.videoHeight;
+ canvas.getContext('2d').drawImage(video, 0, 0, canvas.width, canvas.height);
+
+ imageData = canvas.getContext('2d').getImageData(10, 325, 250, 1);
+ data = ""
+ for (var cptr = 0; cptr < canvas.width * canvas.height; ++cptr) {
+ if (data[4 * cptr] || data[4 * cptr + 1] || data[4 * cptr + 2])
+ return false;
+ }
+ return true;
+}
+
+var track;
+promise_test((test) => {
+ if (window.testRunner)
+ testRunner.setUserMediaPermission(true);
+
+ return navigator.mediaDevices.getUserMedia({ video: true}).then((localStream) => {
+ return new Promise((resolve, reject) => {
+ if (window.internals)
+ internals.useMockRTCPeerConnectionFactory("TwoRealPeerConnections");
+
+ createConnections((firstConnection) => {
+ firstConnection.addStream(localStream);
+ }, (secondConnection) => {
+ secondConnection._onaddstream_ = (streamEvent) => { resolve(streamEvent.stream); };
+ });
+ setTimeout(() => reject("Test timed out"), 5000);
+ });
+ }).then((remoteStream) => {
+ video.srcObject = remoteStream;
+ track = remoteStream.getVideoTracks()[0];
+ return video.play();
+ }).then(() => {
+ assert_false(isVideoBlack());
+ }).then(() => {
+ track.enabled = false;
+ return waitFor(500);
+ }).then(() => {
+ assert_true(isVideoBlack());
+ track.enabled = true;
+ return waitFor(500);
+ }).then(() => {
+ assert_false(isVideoBlack());
+ });
+}, "Incoming muted/unmuted video track");
+ </script>
+ </body>
+</html>
Modified: trunk/Source/WebCore/ChangeLog (214043 => 214044)
--- trunk/Source/WebCore/ChangeLog 2017-03-16 15:23:00 UTC (rev 214043)
+++ trunk/Source/WebCore/ChangeLog 2017-03-16 16:09:50 UTC (rev 214044)
@@ -1,5 +1,34 @@
2017-03-16 Youenn Fablet <you...@apple.com>
+ Improve WebRTC track enabled support
+ https://bugs.webkit.org/show_bug.cgi?id=169727
+
+ Reviewed by Alex Christensen.
+
+ Tests: webrtc/peer-connection-audio-mute2.html
+ webrtc/peer-connection-remote-audio-mute.html
+ webrtc/video-remote-mute.html
+
+ Making sure muted/disabled sources produce silence/black frames.
+ For outgoing audio/video sources, this should be done by the actual a/v providers.
+ We keep this filtering here until we are sure they implement that.
+
+ * platform/audio/mac/AudioSampleDataSource.mm:
+ (WebCore::AudioSampleDataSource::pullAvalaibleSamplesAsChunks): Ensuring disabled audio tracks send silence.
+ Used for outgoing webrtc tracks.
+ * platform/mediastream/mac/MockRealtimeAudioSourceMac.mm:
+ (WebCore::MockRealtimeAudioSourceMac::render): Ditto.
+ * platform/mediastream/mac/RealtimeIncomingAudioSource.cpp:
+ (WebCore::RealtimeIncomingAudioSource::OnData): Ditto.
+ * platform/mediastream/mac/RealtimeIncomingVideoSource.cpp:
+ (WebCore::RealtimeIncomingVideoSource::pixelBufferFromVideoFrame): Generating black frames if muted.
+ (WebCore::RealtimeIncomingVideoSource::OnFrame):
+ * platform/mediastream/mac/RealtimeIncomingVideoSource.h:
+ * platform/mediastream/mac/RealtimeOutgoingVideoSource.cpp:
+ (WebCore::RealtimeOutgoingVideoSource::videoSampleAvailable): Ensuring we quit after sending black frame.
+
+2017-03-16 Youenn Fablet <you...@apple.com>
+
LibWebRTC outgoing source should be thread safe refcounted
https://bugs.webkit.org/show_bug.cgi?id=169726
Modified: trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.mm (214043 => 214044)
--- trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.mm 2017-03-16 15:23:00 UTC (rev 214043)
+++ trunk/Source/WebCore/platform/audio/mac/AudioSampleDataSource.mm 2017-03-16 16:09:50 UTC (rev 214044)
@@ -311,6 +311,16 @@
timeStamp = startFrame;
startFrame = timeStamp;
+
+ if (m_muted) {
+ AudioSampleBufferList::zeroABL(buffer, sampleCountPerChunk * m_outputDescription->bytesPerFrame());
+ while (endFrame - startFrame >= sampleCountPerChunk) {
+ consumeFilledBuffer();
+ startFrame += sampleCountPerChunk;
+ }
+ return true;
+ }
+
while (endFrame - startFrame >= sampleCountPerChunk) {
if (m_ringBuffer->fetch(&buffer, sampleCountPerChunk, startFrame, CARingBuffer::Copy))
return false;
Modified: trunk/Source/WebCore/platform/mediastream/mac/MockRealtimeAudioSourceMac.mm (214043 => 214044)
--- trunk/Source/WebCore/platform/mediastream/mac/MockRealtimeAudioSourceMac.mm 2017-03-16 15:23:00 UTC (rev 214043)
+++ trunk/Source/WebCore/platform/mediastream/mac/MockRealtimeAudioSourceMac.mm 2017-03-16 16:09:50 UTC (rev 214044)
@@ -147,9 +147,6 @@
void MockRealtimeAudioSourceMac::render(double delta)
{
- if (m_muted || !m_enabled)
- return;
-
if (!m_audioBufferList)
reconfigure();
@@ -162,8 +159,11 @@
uint32_t bipBopCount = std::min(frameCount, bipBopRemain);
for (auto& audioBuffer : m_audioBufferList->buffers()) {
audioBuffer.mDataByteSize = frameCount * m_streamFormat.mBytesPerFrame;
- memcpy(audioBuffer.mData, &m_bipBopBuffer[bipBopStart], sizeof(Float32) * bipBopCount);
- addHum(HumVolume, HumFrequency, m_sampleRate, m_samplesRendered, static_cast<float*>(audioBuffer.mData), bipBopCount);
+ if (!m_muted && m_enabled) {
+ memcpy(audioBuffer.mData, &m_bipBopBuffer[bipBopStart], sizeof(Float32) * bipBopCount);
+ addHum(HumVolume, HumFrequency, m_sampleRate, m_samplesRendered, static_cast<float*>(audioBuffer.mData), bipBopCount);
+ } else
+ memset(audioBuffer.mData, 0, sizeof(Float32) * bipBopCount);
}
emitSampleBuffers(bipBopCount);
m_samplesRendered += bipBopCount;
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.cpp (214043 => 214044)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.cpp 2017-03-16 15:23:00 UTC (rev 214043)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingAudioSource.cpp 2017-03-16 16:09:50 UTC (rev 214044)
@@ -95,13 +95,17 @@
m_audioSourceProvider->prepare(&m_streamFormat);
}
+ // FIXME: We should not need to do the extra memory allocation and copy.
+ // Instead, we should be able to directly pass audioData pointer.
WebAudioBufferList audioBufferList { CAAudioStreamDescription(m_streamFormat), WTF::safeCast<uint32_t>(numberOfFrames) };
audioBufferList.buffer(0)->mDataByteSize = numberOfChannels * numberOfFrames * bitsPerSample / 8;
audioBufferList.buffer(0)->mNumberChannels = numberOfChannels;
- // FIXME: We should not need to do the extra memory allocation and copy.
- // Instead, we should be able to directly pass audioData pointer.
- memcpy(audioBufferList.buffer(0)->mData, audioData, audioBufferList.buffer(0)->mDataByteSize);
+ if (muted() || !enabled())
+ memset(audioBufferList.buffer(0)->mData, 0, audioBufferList.buffer(0)->mDataByteSize);
+ else
+ memcpy(audioBufferList.buffer(0)->mData, audioData, audioBufferList.buffer(0)->mDataByteSize);
+
audioSamplesAvailable(mediaTime, audioBufferList, CAAudioStreamDescription(m_streamFormat), numberOfFrames);
}
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingVideoSource.cpp (214043 => 214044)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingVideoSource.cpp 2017-03-16 15:23:00 UTC (rev 214043)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingVideoSource.cpp 2017-03-16 16:09:50 UTC (rev 214044)
@@ -92,13 +92,40 @@
m_videoTrack->RemoveSink(this);
}
+CVPixelBufferRef RealtimeIncomingVideoSource::pixelBufferFromVideoFrame(const webrtc::VideoFrame& frame)
+{
+ if (muted() || !enabled()) {
+ if (!m_blackFrame || m_blackFrameWidth != frame.width() || m_blackFrameHeight != frame.height()) {
+ CVPixelBufferRef pixelBuffer = nullptr;
+ auto status = CVPixelBufferCreate(kCFAllocatorDefault, frame.width(), frame.height(), kCVPixelFormatType_420YpCbCr8Planar, nullptr, &pixelBuffer);
+ ASSERT_UNUSED(status, status == noErr);
+
+ m_blackFrame = pixelBuffer;
+ m_blackFrameWidth = frame.width();
+ m_blackFrameHeight = frame.height();
+
+ status = CVPixelBufferLockBaseAddress(pixelBuffer, 0);
+ ASSERT(status == noErr);
+ void* data = ""
+ size_t yLength = frame.width() * frame.height();
+ memset(data, 0, yLength);
+ memset(static_cast<uint8_t*>(data) + yLength, 128, yLength / 2);
+
+ status = CVPixelBufferUnlockBaseAddress(pixelBuffer, 0);
+ ASSERT(!status);
+ }
+ return m_blackFrame.get();
+ }
+ auto buffer = frame.video_frame_buffer();
+ return static_cast<CVPixelBufferRef>(buffer->native_handle());
+}
+
void RealtimeIncomingVideoSource::OnFrame(const webrtc::VideoFrame& frame)
{
if (!m_isProducingData)
return;
- auto buffer = frame.video_frame_buffer();
- CVPixelBufferRef pixelBuffer = static_cast<CVPixelBufferRef>(buffer->native_handle());
+ auto pixelBuffer = pixelBufferFromVideoFrame(frame);
// FIXME: Convert timing information from VideoFrame to CMSampleTimingInfo.
// For the moment, we will pretend that frames should be rendered asap.
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingVideoSource.h (214043 => 214044)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingVideoSource.h 2017-03-16 15:23:00 UTC (rev 214043)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeIncomingVideoSource.h 2017-03-16 16:09:50 UTC (rev 214044)
@@ -70,6 +70,8 @@
// rtc::VideoSinkInterface
void OnFrame(const webrtc::VideoFrame&) final;
+ CVPixelBufferRef pixelBufferFromVideoFrame(const webrtc::VideoFrame&);
+
RefPtr<Image> m_currentImage;
RealtimeMediaSourceSettings m_currentSettings;
RealtimeMediaSourceSupportedConstraints m_supportedConstraints;
@@ -79,6 +81,9 @@
rtc::scoped_refptr<webrtc::VideoTrackInterface> m_videoTrack;
RetainPtr<CMSampleBufferRef> m_buffer;
PixelBufferConformerCV m_conformer;
+ RetainPtr<CVPixelBufferRef> m_blackFrame;
+ int m_blackFrameWidth { 0 };
+ int m_blackFrameHeight { 0 };
};
} // namespace WebCore
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSource.cpp (214043 => 214044)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSource.cpp 2017-03-16 15:23:00 UTC (rev 214043)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSource.cpp 2017-03-16 16:09:50 UTC (rev 214044)
@@ -92,6 +92,7 @@
auto blackBuffer = m_bufferPool.CreateBuffer(settings.width(), settings.height());
blackBuffer->SetToBlack();
sendFrame(WTFMove(blackBuffer));
+ return;
}
ASSERT(sample.platformSample().type == PlatformSample::CMSampleBufferType);
