Title: [237036] trunk/Source/WebCore
- Revision
- 237036
- Author
- [email protected]
- Date
- 2018-10-11 10:29:30 -0700 (Thu, 11 Oct 2018)
Log Message
[GStreamer] Fix race condition in GStreamerVideoDecoder
https://bugs.webkit.org/show_bug.cgi?id=190470
The GStreamerVideoDecoder.m_dtsPtsMap filed is accessed from
the main thread and some GStreamer streaming thread, make sure
to protect its access.
And use WTF::StdMap instead of std::map.
Patch by Thibault Saunier <[email protected]> on 2018-10-11
Reviewed by Philippe Normand.
Manually tested and a random crash is gone.
* platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp:
(WebCore::GStreamerVideoDecoder::newSampleCallback):
Modified Paths
Diff
Modified: trunk/Source/WebCore/ChangeLog (237035 => 237036)
--- trunk/Source/WebCore/ChangeLog 2018-10-11 17:25:18 UTC (rev 237035)
+++ trunk/Source/WebCore/ChangeLog 2018-10-11 17:29:30 UTC (rev 237036)
@@ -1,3 +1,21 @@
+2018-10-11 Thibault Saunier <[email protected]>
+
+ [GStreamer] Fix race condition in GStreamerVideoDecoder
+ https://bugs.webkit.org/show_bug.cgi?id=190470
+
+ The GStreamerVideoDecoder.m_dtsPtsMap filed is accessed from
+ the main thread and some GStreamer streaming thread, make sure
+ to protect its access.
+
+ And use WTF::StdMap instead of std::map.
+
+ Reviewed by Philippe Normand.
+
+ Manually tested and a random crash is gone.
+
+ * platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp:
+ (WebCore::GStreamerVideoDecoder::newSampleCallback):
+
2018-10-11 Enrique Ocaña González <[email protected]>
[GStreamer][MSE] Fix height calculation for streams with source aspect ratio
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp (237035 => 237036)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp 2018-10-11 17:25:18 UTC (rev 237035)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp 2018-10-11 17:29:30 UTC (rev 237036)
@@ -34,6 +34,8 @@
#include <gst/app/gstappsrc.h>
#include <gst/video/video.h>
#include <mutex>
+#include <wtf/Lock.h>
+#include <wtf/StdMap.h>
#include <wtf/glib/RunLoopSourcePriority.h>
#include <wtf/text/WTFString.h>
@@ -163,7 +165,10 @@
inputImage._size));
GST_BUFFER_DTS(buffer.get()) = (static_cast<guint64>(inputImage._timeStamp) * GST_MSECOND) - m_firstBufferDts;
GST_BUFFER_PTS(buffer.get()) = (static_cast<guint64>(renderTimeMs) * GST_MSECOND) - m_firstBufferPts;
- m_dtsPtsMap[GST_BUFFER_PTS(buffer.get())] = inputImage._timeStamp;
+ {
+ auto locker = holdLock(m_bufferMapLock);
+ m_dtsPtsMap[GST_BUFFER_PTS(buffer.get())] = inputImage._timeStamp;
+ }
GST_LOG_OBJECT(pipeline(), "%ld Decoding: %" GST_PTR_FORMAT, renderTimeMs, buffer.get());
auto sample = adoptGRef(gst_sample_new(buffer.get(), GetCapsForFrame(inputImage), nullptr, nullptr));
@@ -223,10 +228,13 @@
auto sample = gst_app_sink_pull_sample(GST_APP_SINK(sink));
auto buffer = gst_sample_get_buffer(sample);
- // Make sure that the frame.timestamp == previsouly input_frame._timeStamp
- // as it is required by the VideoDecoder baseclass.
- GST_BUFFER_DTS(buffer) = m_dtsPtsMap[GST_BUFFER_PTS(buffer)];
- m_dtsPtsMap.erase(GST_BUFFER_PTS(buffer));
+ {
+ auto locker = holdLock(m_bufferMapLock);
+ // Make sure that the frame.timestamp == previsouly input_frame._timeStamp
+ // as it is required by the VideoDecoder baseclass.
+ GST_BUFFER_DTS(buffer) = m_dtsPtsMap[GST_BUFFER_PTS(buffer)];
+ m_dtsPtsMap.erase(GST_BUFFER_PTS(buffer));
+ }
auto frame(LibWebRTCVideoFrameFromGStreamerSample(sample, webrtc::kVideoRotation_0));
GST_BUFFER_DTS(buffer) = GST_CLOCK_TIME_NONE;
GST_LOG_OBJECT(pipeline(), "Output decoded frame! %d -> %" GST_PTR_FORMAT,
@@ -258,7 +266,8 @@
GstVideoInfo m_info;
webrtc::DecodedImageCallback* m_imageReadyCb;
- std::map<GstClockTime, GstClockTime> m_dtsPtsMap;
+ Lock m_bufferMapLock;
+ StdMap<GstClockTime, GstClockTime> m_dtsPtsMap;
GstClockTime m_firstBufferPts;
GstClockTime m_firstBufferDts;
};
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