Changeset 275511 in webkit

Timestamp:

Apr 6, 2021 6:25:01 AM (3 years ago)

Author:

commit-queue@webkit.org

Message:

[GStreamer] Heap allocation exceptions
​https://bugs.webkit.org/show_bug.cgi?id=224188

Patch by Philippe Normand <​pnormand@igalia.com> on 2021-04-06
Reviewed by Xabier Rodriguez-Calvar.

The current GStreamer audio-related backends can't yet avoid heap allocations in the audio
thread, so we need exceptions for those cases.

• platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp:

(WebCore::AudioSourceProviderGStreamer::provideInput):

• platform/graphics/gstreamer/GstAllocatorFastMalloc.cpp:

(gstMemoryFastMallocNew):

• platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp:

(WebCore::RealtimeOutgoingAudioSourceLibWebRTC::audioSamplesAvailable):

Location:

trunk/Source/WebCore

Files:

• ChangeLog (modified) (1 diff)
• platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp (modified) (1 diff)
• platform/graphics/gstreamer/GstAllocatorFastMalloc.cpp (modified) (1 diff)
• platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp (modified) (1 diff)

trunk/Source/WebCore/ChangeLog

@@ +1 @@
+2021-04-06  Philippe Normand  <pnormand@igalia.com>
+
+        [GStreamer] Heap allocation exceptions
+        https://bugs.webkit.org/show_bug.cgi?id=224188
+
+        Reviewed by Xabier Rodriguez-Calvar.
+
+        The current GStreamer audio-related backends can't yet avoid heap allocations in the audio
+        thread, so we need exceptions for those cases.
+
+        * platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp:
+        (WebCore::AudioSourceProviderGStreamer::provideInput):
+        * platform/graphics/gstreamer/GstAllocatorFastMalloc.cpp:
+        (gstMemoryFastMallocNew):
+        * platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp:
+        (WebCore::RealtimeOutgoingAudioSourceLibWebRTC::audioSamplesAvailable):
+
 2021-04-06  Philippe Normand  <pnormand@igalia.com>
 

trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp

@@ -154 +154 @@
 {
     GST_TRACE("Fetching buffers from adapters");
-    auto locker = holdLock(m_adapterMutex);
+    auto locker = tryHoldLock(m_adapterMutex);
+    if (!locker)
+        return;
     for (auto& it : m_adapters)
         copyGStreamerBuffersToAudioChannel(it.value.get(), bus, it.key - 1, framesToProcess);

trunk/Source/WebCore/platform/graphics/gstreamer/GstAllocatorFastMalloc.cpp

@@ -45 +45 @@
     alignment |= gst_memory_alignment;
     ASSERT(!((alignment + 1) & alignment));
+
+    // GStreamer's allocator requires heap allocations.
+    DisableMallocRestrictionsForCurrentThreadScope disableMallocRestrictions;
 
     gsize headerSize = (sizeof(GstMemoryFastMalloc) + alignment) & ~alignment;

trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp

@@ -65 +65 @@
 void RealtimeOutgoingAudioSourceLibWebRTC::audioSamplesAvailable(const MediaTime&, const PlatformAudioData& audioData, const AudioStreamDescription& streamDescription, size_t /* sampleCount */)
 {
+    DisableMallocRestrictionsForCurrentThreadScope disableMallocRestrictions;
     auto data = static_cast<const GStreamerAudioData&>(audioData);
     auto desc = static_cast<const GStreamerAudioStreamDescription&>(streamDescription);