[SCM] WebKit Debian packaging branch, debian/experimental, updated. upstream/1.3.3-9427-gc2be6fc
crogers at google.com
crogers at google.com
Wed Dec 22 15:06:00 UTC 2010
The following commit has been merged in the debian/experimental branch:
commit a9cc96957ad3425d643313e62663b8a8aaddc8de
Author: crogers at google.com <crogers at google.com@268f45cc-cd09-0410-ab3c-d52691b4dbfc>
Date: Wed Oct 27 22:28:57 2010 +0000
2010-10-27 Chris Rogers <crogers at google.com>
Reviewed by Kenneth Russell.
Add AudioResamplerKernel files
https://bugs.webkit.org/show_bug.cgi?id=47624
No new tests since audio API is not yet implemented.
* platform/audio/AudioResamplerKernel.cpp: Added.
(WebCore::AudioResamplerKernel::AudioResamplerKernel):
(WebCore::AudioResamplerKernel::getSourcePointer):
(WebCore::AudioResamplerKernel::process):
(WebCore::AudioResamplerKernel::reset):
(WebCore::AudioResamplerKernel::rate):
* platform/audio/AudioResamplerKernel.h: Added.
git-svn-id: http://svn.webkit.org/repository/webkit/trunk@70719 268f45cc-cd09-0410-ab3c-d52691b4dbfc
diff --git a/WebCore/ChangeLog b/WebCore/ChangeLog
index a2039a2..336f9ae 100644
--- a/WebCore/ChangeLog
+++ b/WebCore/ChangeLog
@@ -1,3 +1,20 @@
+2010-10-27 Chris Rogers <crogers at google.com>
+
+ Reviewed by Kenneth Russell.
+
+ Add AudioResamplerKernel files
+ https://bugs.webkit.org/show_bug.cgi?id=47624
+
+ No new tests since audio API is not yet implemented.
+
+ * platform/audio/AudioResamplerKernel.cpp: Added.
+ (WebCore::AudioResamplerKernel::AudioResamplerKernel):
+ (WebCore::AudioResamplerKernel::getSourcePointer):
+ (WebCore::AudioResamplerKernel::process):
+ (WebCore::AudioResamplerKernel::reset):
+ (WebCore::AudioResamplerKernel::rate):
+ * platform/audio/AudioResamplerKernel.h: Added.
+
2010-10-27 Adam Barth <abarth at webkit.org>
Reviewed by Ojan Vafai.
diff --git a/WebCore/platform/audio/AudioResamplerKernel.cpp b/WebCore/platform/audio/AudioResamplerKernel.cpp
new file mode 100644
index 0000000..7b99997
--- /dev/null
+++ b/WebCore/platform/audio/AudioResamplerKernel.cpp
@@ -0,0 +1,143 @@
+/*
+ * Copyright (C) 2010, Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "config.h"
+
+#if ENABLE(WEB_AUDIO)
+
+#include "AudioResamplerKernel.h"
+
+#include "AudioResampler.h"
+#include <algorithm>
+
+using namespace std;
+
+namespace WebCore {
+
+const size_t AudioResamplerKernel::MaxFramesToProcess = 128;
+
+AudioResamplerKernel::AudioResamplerKernel(AudioResampler* resampler)
+ : m_resampler(resampler)
+ // The buffer size must be large enough to hold up to two extra sample frames for the linear interpolation.
+ , m_sourceBuffer(2 + static_cast<int>(MaxFramesToProcess * AudioResampler::MaxRate))
+ , m_virtualReadIndex(0.0)
+ , m_fillIndex(0)
+{
+ m_lastValues[0] = 0.0f;
+ m_lastValues[1] = 0.0f;
+}
+
+float* AudioResamplerKernel::getSourcePointer(size_t framesToProcess, size_t* numberOfSourceFramesNeededP)
+{
+ ASSERT(framesToProcess <= MaxFramesToProcess);
+
+ // Calculate the next "virtual" index. After process() is called, m_virtualReadIndex will equal this value.
+ double nextFractionalIndex = m_virtualReadIndex + framesToProcess * rate();
+
+ // Because we're linearly interpolating between the previous and next sample we need to round up so we include the next sample.
+ int endIndex = static_cast<int>(nextFractionalIndex + 1.0); // round up to next integer index
+
+ // Determine how many input frames we'll need.
+ // We need to fill the buffer up to and including endIndex (so add 1) but we've already buffered m_fillIndex frames from last time.
+ size_t framesNeeded = 1 + endIndex - m_fillIndex;
+ if (numberOfSourceFramesNeededP)
+ *numberOfSourceFramesNeededP = framesNeeded;
+
+ // Do bounds checking for the source buffer.
+ bool isGood = m_fillIndex < m_sourceBuffer.size() && m_fillIndex + framesNeeded <= m_sourceBuffer.size();
+ ASSERT(isGood);
+ if (!isGood)
+ return 0;
+
+ return m_sourceBuffer.data() + m_fillIndex;
+}
+
+void AudioResamplerKernel::process(float* destination, size_t framesToProcess)
+{
+ ASSERT(framesToProcess <= MaxFramesToProcess);
+
+ float* source = m_sourceBuffer.data();
+
+ double rate = this->rate();
+ rate = max(0.0, rate);
+ rate = min(AudioResampler::MaxRate, rate);
+
+ // Start out with the previous saved values (if any).
+ if (m_fillIndex > 0) {
+ source[0] = m_lastValues[0];
+ source[1] = m_lastValues[1];
+ }
+
+ // Make a local copy.
+ double virtualReadIndex = m_virtualReadIndex;
+
+ // Sanity check source buffer access.
+ ASSERT(framesToProcess > 0);
+ ASSERT(virtualReadIndex >= 0 && 1 + static_cast<unsigned>(virtualReadIndex + (framesToProcess - 1) * rate) < m_sourceBuffer.size());
+
+ // Do the linear interpolation.
+ int n = framesToProcess;
+ while (n--) {
+ unsigned readIndex = static_cast<unsigned>(virtualReadIndex);
+ double interpolationFactor = virtualReadIndex - readIndex;
+
+ double sample1 = source[readIndex];
+ double sample2 = source[readIndex + 1];
+
+ double sample = (1.0 - interpolationFactor) * sample1 + interpolationFactor * sample2;
+
+ *destination++ = static_cast<float>(sample);
+
+ virtualReadIndex += rate;
+ }
+
+ // Save the last two sample-frames which will later be used at the beginning of the source buffer the next time around.
+ int readIndex = static_cast<int>(virtualReadIndex);
+ m_lastValues[0] = source[readIndex];
+ m_lastValues[1] = source[readIndex + 1];
+ m_fillIndex = 2;
+
+ // Wrap the virtual read index back to the start of the buffer.
+ virtualReadIndex -= readIndex;
+
+ // Put local copy back into member variable.
+ m_virtualReadIndex = virtualReadIndex;
+}
+
+void AudioResamplerKernel::reset()
+{
+ m_virtualReadIndex = 0.0;
+ m_fillIndex = 0;
+ m_lastValues[0] = 0.0f;
+ m_lastValues[1] = 0.0f;
+}
+
+double AudioResamplerKernel::rate() const
+{
+ return m_resampler->rate();
+}
+
+} // namespace WebCore
+
+#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/platform/audio/AudioResamplerKernel.h b/WebCore/platform/audio/AudioResamplerKernel.h
new file mode 100644
index 0000000..99d877b
--- /dev/null
+++ b/WebCore/platform/audio/AudioResamplerKernel.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (C) 2010, Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef AudioResamplerKernel_h
+#define AudioResamplerKernel_h
+
+#include "AudioArray.h"
+
+namespace WebCore {
+
+class AudioResampler;
+
+// AudioResamplerKernel does resampling on a single mono channel.
+// It uses a simple linear interpolation for good performance.
+
+class AudioResamplerKernel {
+public:
+ AudioResamplerKernel(AudioResampler*);
+
+ // getSourcePointer() should be called each time before process() is called.
+ // Given a number of frames to process (for subsequent call to process()), it returns a pointer and numberOfSourceFramesNeeded
+ // where sample data should be copied. This sample data provides the input to the resampler when process() is called.
+ // framesToProcess must be less than or equal to MaxFramesToProcess.
+ float* getSourcePointer(size_t framesToProcess, size_t* numberOfSourceFramesNeeded);
+
+ // process() resamples framesToProcess frames from the source into destination.
+ // Each call to process() must be preceded by a call to getSourcePointer() so that source input may be supplied.
+ // framesToProcess must be less than or equal to MaxFramesToProcess.
+ void process(float* destination, size_t framesToProcess);
+
+ // Resets the processing state.
+ void reset();
+
+ static const size_t MaxFramesToProcess;
+
+private:
+ double rate() const;
+
+ AudioResampler* m_resampler;
+ AudioFloatArray m_sourceBuffer;
+
+ // This is a (floating point) read index on the input stream.
+ double m_virtualReadIndex;
+
+ // We need to have continuity from one call of process() to the next.
+ // m_lastValues stores the last two sample values from the last call to process().
+ // m_fillIndex represents how many buffered samples we have which can be as many as 2.
+ // For the first call to process() (or after reset()) there will be no buffered samples.
+ float m_lastValues[2];
+ unsigned m_fillIndex;
+};
+
+} // namespace WebCore
+
+#endif // AudioResamplerKernel_h
--
WebKit Debian packaging
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