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Backed out 2 changesets (bug 886618, bug 886657) for frequent OSX 10.7 mochitest-1 failures.

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Backed out changeset 9fd8b2adc693 (bug 886618)
Backed out changeset 376fe12f4de4 (bug 886657)
This commit is contained in:
Ryan VanderMeulen 2013-06-25 15:24:11 -04:00
parent 0517d938d7
commit 0406c5ea35
8 changed files with 40 additions and 127 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

35
content/media/webaudio/AudioBuffer.cpp
View File

@ -138,6 +138,30 @@ AudioBuffer::GetChannelData(JSContext* aJSContext, uint32_t aChannel,
return mJSChannels[aChannel];
}
bool
AudioBuffer::SetChannelDataFromArrayBufferContents(JSContext* aJSContext,
uint32_t aChannel,
void* aContents)
{
if (!RestoreJSChannelData(aJSContext)) {
return false;
}
MOZ_ASSERT(aChannel < NumberOfChannels());
JS::RootedObject arrayBuffer(aJSContext, JS_NewArrayBufferWithContents(aJSContext, aContents));
if (!arrayBuffer) {
return false;
}
mJSChannels[aChannel] = JS_NewFloat32ArrayWithBuffer(aJSContext, arrayBuffer,
0, -1);
if (!mJSChannels[aChannel]) {
return false;
}
MOZ_ASSERT(mLength == JS_GetTypedArrayLength(mJSChannels[aChannel]));
return true;
}
static already_AddRefed<ThreadSharedFloatArrayBufferList>
StealJSArrayDataIntoThreadSharedFloatArrayBufferList(JSContext* aJSContext,
const nsTArray<JSObject*>& aJSArrays)
@ -153,7 +177,8 @@ StealJSArrayDataIntoThreadSharedFloatArrayBufferList(JSContext* aJSContext,
&stolenData)) {
result->SetData(i, dataToFree, reinterpret_cast<float*>(stolenData));
} else {
return nullptr;
result->Clear();
return result.forget();
}
}
return result.forget();
@ -163,13 +188,7 @@ ThreadSharedFloatArrayBufferList*
AudioBuffer::GetThreadSharedChannelsForRate(JSContext* aJSContext)
{
if (!mSharedChannels) {
for (uint32_t i = 0; i < mJSChannels.Length(); ++i) {
if (mLength != JS_GetTypedArrayLength(mJSChannels[i])) {
// Probably one of the arrays was neutered
return nullptr;
}
}
// Steal JS data
mSharedChannels =
StealJSArrayDataIntoThreadSharedFloatArrayBufferList(aJSContext, mJSChannels);
}

7
content/media/webaudio/AudioBuffer.h
View File

@ -91,10 +91,15 @@ public:
/**
* Returns a ThreadSharedFloatArrayBufferList containing the sample data.
* Can return null if there is no data.
*/
ThreadSharedFloatArrayBufferList* GetThreadSharedChannelsForRate(JSContext* aContext);
// aContents should either come from JS_AllocateArrayBufferContents or
// JS_StealArrayBufferContents.
bool SetChannelDataFromArrayBufferContents(JSContext* aJSContext,
uint32_t aChannel,
void* aContents);
// This replaces the contents of the JS array for the given channel.
// This function needs to be called on an AudioBuffer which has not been
// handed off to the content yet, and right after the object has been

26
content/media/webaudio/AudioBufferSourceNode.cpp
View File

@ -41,12 +41,6 @@ NS_INTERFACE_MAP_END_INHERITING(AudioNode)
NS_IMPL_ADDREF_INHERITED(AudioBufferSourceNode, AudioNode)
NS_IMPL_RELEASE_INHERITED(AudioBufferSourceNode, AudioNode)
/**
* Media-thread playback engine for AudioBufferSourceNode.
* Nothing is played until a non-null buffer has been set (via
* AudioNodeStream::SetBuffer) and a non-zero duration has been set (via
* AudioNodeStream::SetInt32Parameter).
*/
class AudioBufferSourceNodeEngine : public AudioNodeEngine
{
public:
@ -359,9 +353,8 @@ public:
AudioChunk* aOutput,
bool* aFinished)
{
if (!mBuffer || !mDuration) {
if (!mBuffer)
return;
}
uint32_t channels = mBuffer->GetChannels();
if (!channels) {
@ -490,9 +483,7 @@ AudioBufferSourceNode::Start(double aWhen, double aOffset,
std::numeric_limits<double>::min();
SendOffsetAndDurationParametersToStream(ns, aOffset, duration);
} else {
// Remember our arguments so that we can use them once we have a buffer.
// We can't send these parameters now because we don't know the buffer
// sample rate.
// Remember our arguments so that we can use them once we have a buffer
mOffset = aOffset;
mDuration = aDuration.WasPassed() ?
aDuration.Value() :
@ -520,13 +511,11 @@ AudioBufferSourceNode::SendBufferParameterToStream(JSContext* aCx)
mBuffer->GetThreadSharedChannelsForRate(aCx);
ns->SetBuffer(data.forget());
ns->SetInt32Parameter(SAMPLE_RATE, rate);
if (mStartCalled) {
SendOffsetAndDurationParametersToStream(ns, mOffset, mDuration);
}
} else {
ns->SetBuffer(nullptr);
}
SendOffsetAndDurationParametersToStream(ns, mOffset, mDuration);
}
void
@ -534,11 +523,8 @@ AudioBufferSourceNode::SendOffsetAndDurationParametersToStream(AudioNodeStream*
double aOffset,
double aDuration)
{
NS_ASSERTION(mBuffer && mStartCalled,
"Only call this when we have a buffer and start() has been called");
float rate = mBuffer->SampleRate();
int32_t lengthSamples = mBuffer->Length();
float rate = mBuffer ? mBuffer->SampleRate() : Context()->SampleRate();
int32_t lengthSamples = mBuffer ? mBuffer->Length() : 0;
double length = double(lengthSamples) / rate;
double offset = std::max(0.0, aOffset);
double endOffset = aDuration == std::numeric_limits<double>::min() ?

2
content/media/webaudio/ConvolverNode.cpp
View File

@ -188,7 +188,7 @@ ConvolverNode::SetBuffer(JSContext* aCx, AudioBuffer* aBuffer, ErrorResult& aRv)
uint32_t length = mBuffer->Length();
nsRefPtr<ThreadSharedFloatArrayBufferList> data =
mBuffer->GetThreadSharedChannelsForRate(aCx);
if (data && length < WEBAUDIO_BLOCK_SIZE) {
if (length < WEBAUDIO_BLOCK_SIZE) {
// For very small impulse response buffers, we need to pad the
// buffer with 0 to make sure that the Reverb implementation
// has enough data to compute FFTs from.

3
content/media/webaudio/test/Makefile.in
View File

@ -11,7 +11,6 @@ relativesrcdir := @relativesrcdir@
include $(DEPTH)/config/autoconf.mk
MOCHITEST_FILES := \
audioBufferSourceNodeNeutered_worker.js \
webaudio.js \
layouttest-glue.js \
test_bug808374.html \
@ -45,8 +44,6 @@ MOCHITEST_FILES := \
test_audioBufferSourceNodeLoop.html \
test_audioBufferSourceNodeLoopStartEnd.html \
test_audioBufferSourceNodeLoopStartEndSame.html \
test_audioBufferSourceNodeNeutered.html \
test_audioBufferSourceNodeNoStart.html \
test_audioBufferSourceNodeNullBuffer.html \
test_badConnect.html \
test_biquadFilterNode.html \

3
content/media/webaudio/test/audioBufferSourceNodeNeutered_worker.js
View File

@ -1,3 +0,0 @@
onmessage = function (event) {
postMessage("Pong");
};

58
content/media/webaudio/test/test_audioBufferSourceNodeNeutered.html
View File

@ -1,58 +0,0 @@
<!DOCTYPE HTML>
<html>
<head>
<title>Test AudioBufferSourceNode when an AudioBuffer's getChanneData array is neutered</title>
<script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
<script type="text/javascript" src="webaudio.js"></script>
<link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
</head>
<body>
<pre id="test">
<script class="testbody" type="text/javascript">
function createGarbage() {
var s = [];
for (var i = 0; i < 10000000; ++i) {
s.push(i);
}
var sum = 0;
for (var i = 0; i < s.length; ++i) {
sum += s[i];
}
return sum;
}
var worker = new Worker("audioBufferSourceNodeNeutered_worker.js");
var gTest = {
length: 2048,
numberOfChannels: 1,
createGraph: function(context) {
var buffer = context.createBuffer(1, 10000000, context.sampleRate);
var data = buffer.getChannelData(0);
for (var i = 0; i < data.length; ++i) {
data[i] = (i%100)/100 - 0.5;
}
// Neuter the buffer now
var data = buffer.getChannelData(0).buffer;
worker.postMessage(data, [data]);
// Create garbage and GC to replace the buffer data with garbage
SpecialPowers.gc();
createGarbage();
SpecialPowers.gc();
var source = context.createBufferSource();
source.buffer = buffer;
source.start();
// This should play silence
return source;
},
};
runTest();
</script>
</pre>
</body>
</html>

33
content/media/webaudio/test/test_audioBufferSourceNodeNoStart.html
View File

@ -1,33 +0,0 @@
<!DOCTYPE HTML>
<html>
<head>
<title>Test AudioBufferSourceNode when start() is not called</title>
<script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
<script type="text/javascript" src="webaudio.js"></script>
<link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
</head>
<body>
<pre id="test">
<script class="testbody" type="text/javascript">
var gTest = {
length: 2048,
numberOfChannels: 1,
createGraph: function(context) {
var buffer = context.createBuffer(1, 2048, context.sampleRate);
var data = buffer.getChannelData(0);
for (var i = 0; i < data.length; ++i) {
data[i] = (i%100)/100 - 0.5;
}
var source = context.createBufferSource();
source.buffer = buffer;
return source;
},
};
runTest();
</script>
</pre>
</body>
</html>