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a69f4899c6
webkit2gtk3/webkit2gtk3-CVE-2024-40776.patch
Dominique Leuenberger a69f4899c6 - Update to version 2.46.4 
Also fix a typo in a CVE ref, and remove some mistakenly-added bugs/CVEs that
don't affect Linux.

OBS-URL: https://build.opensuse.org/package/show/GNOME:Factory/webkit2gtk3?expand=0&rev=498
2024-11-28 09:07:48 +00:00

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From b951404ea74ae432312a83138f5c8945a0d09e1b Mon Sep 17 00:00:00 2001
From: Jean-Yves Avenard <jya@apple.com>
Date: Wed, 24 Apr 2024 19:01:06 -0700
Subject: [PATCH] Cherry-pick 272448.960@safari-7618-branch (b7ccdb65258e).
https://bugs.webkit.org/show_bug.cgi?id=273176
Always copy all audio channels to the AudioBus to guarantee data lifetime.
https://bugs.webkit.org/show_bug.cgi?id=273176
rdar://125166710
Reviewed by Chris Dumez.
Following 275262@main, a task is dispatched on the audio render thread.
This task dispatch takes a reference to the source and destination AudioBus
however when a MultiChannelResampler is in use, the source AudioBus may
contain a raw pointer to the resampled's AudioArray and the lifetime of
this object may be shorter than the AudioBus.
In 232182@main, a speed and memory optimisation was added by passed-in buffer
as memory for the first channel in the AudioBus.
We revert this change for now and copy all channels' data to the AudioBus.
Added test.
* LayoutTests/webaudio/crashtest/audioworklet-concurrent-resampler-crash-expected.txt: Added.
* LayoutTests/webaudio/crashtest/audioworklet-concurrent-resampler-crash.html: Added.
* Source/WebCore/platform/audio/MultiChannelResampler.cpp:
(WebCore::MultiChannelResampler::MultiChannelResampler):
(WebCore::MultiChannelResampler::provideInputForChannel):
* Source/WebCore/platform/audio/MultiChannelResampler.h:
Canonical link: https://commits.webkit.org/274313.332@webkitglib/2.44
---
...et-concurrent-resampler-crash-expected.txt | 1 +
...dioworklet-concurrent-resampler-crash.html | 44 +++++++++++++++++++
.../platform/audio/MultiChannelResampler.cpp | 23 ++--------
.../platform/audio/MultiChannelResampler.h | 2 -
4 files changed, 48 insertions(+), 22 deletions(-)
create mode 100644 LayoutTests/webaudio/crashtest/audioworklet-concurrent-resampler-crash-expected.txt
create mode 100644 LayoutTests/webaudio/crashtest/audioworklet-concurrent-resampler-crash.html
diff --git a/LayoutTests/webaudio/crashtest/audioworklet-concurrent-resampler-crash-expected.txt b/LayoutTests/webaudio/crashtest/audioworklet-concurrent-resampler-crash-expected.txt
new file mode 100644
index 000000000000..654ddf7f17ef
--- /dev/null
+++ b/LayoutTests/webaudio/crashtest/audioworklet-concurrent-resampler-crash-expected.txt
@@ -0,0 +1 @@
+This test passes if it does not crash.
diff --git a/LayoutTests/webaudio/crashtest/audioworklet-concurrent-resampler-crash.html b/LayoutTests/webaudio/crashtest/audioworklet-concurrent-resampler-crash.html
new file mode 100644
index 000000000000..b3ab181d4787
--- /dev/null
+++ b/LayoutTests/webaudio/crashtest/audioworklet-concurrent-resampler-crash.html
@@ -0,0 +1,44 @@
+<html>
+<head>
+ <script>
+ let worklet_source = `
+ class Processor extends AudioWorkletProcessor {
+ process(inputs, outputs, parameters) {
+ return true;
+ }
+ }
+ registerProcessor('P2', Processor);
+ `;
+
+ let blob = new Blob([worklet_source], { type: 'application/javascript' });
+ let worklet = URL.createObjectURL(blob);
+
+ var ctx = new AudioContext({ sampleRate: 44100});
+ const dest = ctx.destination;
+ dest.channelCountMode = "max";
+
+ async function main() {
+ await ctx.audioWorklet.addModule(worklet);
+ var script_processor = ctx.createScriptProcessor();
+ script_processor.onaudioprocess = function() {
+ dest.channelCount = 1;
+ audio_worklet.disconnect();
+ if (window.testRunner)
+ testRunner.notifyDone();
+ }
+ var audio_worklet = new AudioWorkletNode(ctx, "P2");
+ script_processor.connect(audio_worklet);
+ audio_worklet.connect(dest);
+ }
+ </script>
+</head>
+<body onload="main()">
+ <p>This test passes if it does not crash.</p>
+ <script>
+ if (window.testRunner) {
+ testRunner.waitUntilDone();
+ testRunner.dumpAsText();
+ }
+ </script>
+</body>
+</html>
diff --git a/Source/WebCore/platform/audio/MultiChannelResampler.cpp b/Source/WebCore/platform/audio/MultiChannelResampler.cpp
index e5a0cfc10caa..c44df274cbbc 100644
--- a/Source/WebCore/platform/audio/MultiChannelResampler.cpp
+++ b/Source/WebCore/platform/audio/MultiChannelResampler.cpp
@@ -42,19 +42,8 @@ namespace WebCore {
MultiChannelResampler::MultiChannelResampler(double scaleFactor, unsigned numberOfChannels, unsigned requestFrames, Function<void(AudioBus*, size_t framesToProcess)>&& provideInput)
: m_numberOfChannels(numberOfChannels)
, m_provideInput(WTFMove(provideInput))
- , m_multiChannelBus(AudioBus::create(numberOfChannels, requestFrames, false))
+ , m_multiChannelBus(AudioBus::create(numberOfChannels, requestFrames))
{
- // As an optimization, we will use the buffer passed to provideInputForChannel() as channel memory for the first channel so we
- // only need to allocate memory if there is more than one channel.
- if (numberOfChannels > 1) {
- m_channelsMemory = Vector<std::unique_ptr<AudioFloatArray>>(numberOfChannels - 1, [&](size_t i) {
- size_t channelIndex = i + 1;
- auto floatArray = makeUnique<AudioFloatArray>(requestFrames);
- m_multiChannelBus->setChannelMemory(channelIndex, floatArray->data(), requestFrames);
- return floatArray;
- });
- }
-
// Create each channel's resampler.
m_kernels = Vector<std::unique_ptr<SincResampler>>(numberOfChannels, [&](size_t channelIndex) {
return makeUnique<SincResampler>(scaleFactor, requestFrames, std::bind(&MultiChannelResampler::provideInputForChannel, this, std::placeholders::_1, std::placeholders::_2, channelIndex));
@@ -93,16 +82,10 @@ void MultiChannelResampler::process(AudioBus* destination, size_t framesToProces
void MultiChannelResampler::provideInputForChannel(std::span<float> buffer, size_t framesToProcess, unsigned channelIndex)
{
ASSERT(channelIndex < m_multiChannelBus->numberOfChannels());
- ASSERT(framesToProcess == m_multiChannelBus->length());
+ ASSERT(framesToProcess <= m_multiChannelBus->length());
- if (!channelIndex) {
- // As an optimization, we use the provided buffer as memory for the first channel in the AudioBus. This avoids
- // having to memcpy() for the first channel.
- RELEASE_ASSERT(framesToProcess <= buffer.size());
- m_multiChannelBus->setChannelMemory(0, buffer.data(), framesToProcess);
+ if (!channelIndex)
m_provideInput(m_multiChannelBus.get(), framesToProcess);
- return;
- }
// Copy the channel data from what we received from m_multiChannelProvider.
memcpySpan(buffer.subspan(0, framesToProcess), m_multiChannelBus->channel(channelIndex)->span().subspan(0, framesToProcess));
diff --git a/Source/WebCore/platform/audio/MultiChannelResampler.h b/Source/WebCore/platform/audio/MultiChannelResampler.h
index 25d43100b71f..214ee06567ac 100644
--- a/Source/WebCore/platform/audio/MultiChannelResampler.h
+++ b/Source/WebCore/platform/audio/MultiChannelResampler.h
@@ -29,7 +29,6 @@
#ifndef MultiChannelResampler_h
#define MultiChannelResampler_h
-#include "AudioArray.h"
#include <memory>
#include <wtf/Function.h>
#include <wtf/Vector.h>
@@ -62,7 +61,6 @@ private:
size_t m_outputFramesReady { 0 };
Function<void(AudioBus*, size_t framesToProcess)> m_provideInput;
RefPtr<AudioBus> m_multiChannelBus;
- Vector<std::unique_ptr<AudioFloatArray>> m_channelsMemory;
};
} // namespace WebCore
--
2.45.2
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