From 0406c5ea35bcb8a86b8c661603a8f3e42b542a7f Mon Sep 17 00:00:00 2001
From: Ryan VanderMeulen <ryanvm@gmail.com>
Date: Tue, 25 Jun 2013 15:24:11 -0400
Subject: [PATCH] Backed out 2 changesets (bug 886618, bug 886657) for frequent
 OSX 10.7 mochitest-1 failures.

Backed out changeset 9fd8b2adc693 (bug 886618)
Backed out changeset 376fe12f4de4 (bug 886657)
---
 content/media/webaudio/AudioBuffer.cpp        | 35 ++++++++---
 content/media/webaudio/AudioBuffer.h          |  7 ++-
 .../media/webaudio/AudioBufferSourceNode.cpp  | 26 ++-------
 content/media/webaudio/ConvolverNode.cpp      |  2 +-
 content/media/webaudio/test/Makefile.in       |  3 -
 .../audioBufferSourceNodeNeutered_worker.js   |  3 -
 .../test_audioBufferSourceNodeNeutered.html   | 58 -------------------
 .../test_audioBufferSourceNodeNoStart.html    | 33 -----------
 8 files changed, 40 insertions(+), 127 deletions(-)
 delete mode 100644 content/media/webaudio/test/audioBufferSourceNodeNeutered_worker.js
 delete mode 100644 content/media/webaudio/test/test_audioBufferSourceNodeNeutered.html
 delete mode 100644 content/media/webaudio/test/test_audioBufferSourceNodeNoStart.html

diff --git a/content/media/webaudio/AudioBuffer.cpp b/content/media/webaudio/AudioBuffer.cpp
index 1d2c6c764e8..e7a712596f6 100644
--- a/content/media/webaudio/AudioBuffer.cpp
+++ b/content/media/webaudio/AudioBuffer.cpp
@@ -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);
   }
diff --git a/content/media/webaudio/AudioBuffer.h b/content/media/webaudio/AudioBuffer.h
index 9ca0f6d8511..f684a005228 100644
--- a/content/media/webaudio/AudioBuffer.h
+++ b/content/media/webaudio/AudioBuffer.h
@@ -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
diff --git a/content/media/webaudio/AudioBufferSourceNode.cpp b/content/media/webaudio/AudioBufferSourceNode.cpp
index 2cea4904fee..b65c2d88884 100644
--- a/content/media/webaudio/AudioBufferSourceNode.cpp
+++ b/content/media/webaudio/AudioBufferSourceNode.cpp
@@ -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() ?
diff --git a/content/media/webaudio/ConvolverNode.cpp b/content/media/webaudio/ConvolverNode.cpp
index df8280ea6f2..a84a78ec1ee 100644
--- a/content/media/webaudio/ConvolverNode.cpp
+++ b/content/media/webaudio/ConvolverNode.cpp
@@ -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.
diff --git a/content/media/webaudio/test/Makefile.in b/content/media/webaudio/test/Makefile.in
index 419fd057c36..7aeb580e7fa 100644
--- a/content/media/webaudio/test/Makefile.in
+++ b/content/media/webaudio/test/Makefile.in
@@ -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 \
diff --git a/content/media/webaudio/test/audioBufferSourceNodeNeutered_worker.js b/content/media/webaudio/test/audioBufferSourceNodeNeutered_worker.js
deleted file mode 100644
index 2a5a4bff89f..00000000000
--- a/content/media/webaudio/test/audioBufferSourceNodeNeutered_worker.js
+++ /dev/null
@@ -1,3 +0,0 @@
-onmessage = function (event) {
-  postMessage("Pong");
-};
diff --git a/content/media/webaudio/test/test_audioBufferSourceNodeNeutered.html b/content/media/webaudio/test/test_audioBufferSourceNodeNeutered.html
deleted file mode 100644
index da967dd8f01..00000000000
--- a/content/media/webaudio/test/test_audioBufferSourceNodeNeutered.html
+++ /dev/null
@@ -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>
diff --git a/content/media/webaudio/test/test_audioBufferSourceNodeNoStart.html b/content/media/webaudio/test/test_audioBufferSourceNodeNoStart.html
deleted file mode 100644
index 89340ade8b3..00000000000
--- a/content/media/webaudio/test/test_audioBufferSourceNodeNoStart.html
+++ /dev/null
@@ -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>