[Webkit-unassigned] [Bug 69703] HRTF Database consolidation

[bugzilla-daemon at webkit.org]

https://bugs.webkit.org/show_bug.cgi?id=69703





--- Comment #14 from Chris Rogers <crogers at google.com>  2011-11-08 17:21:40 PST ---
(From update of attachment 113299)
View in context: https://bugs.webkit.org/attachment.cgi?id=113299&action=review

Hi Philippe, looks like we're getting really close.  The latest comments fix a problem where we were sample-rate converting the entire concatenated buffer.  Instead, it's more correct to extract the individual impulse-response at the same sample-rate as the audio file, then finally sample-rate convert that to the sample-rate we use for the audio hardware...

> Source/WebCore/platform/audio/HRTFElevation.cpp:54
> +static const size_t TotalNumberOfResponses = 240;

nit: you can remove the "static" keyword since it's implicit

> Source/WebCore/platform/audio/HRTFElevation.cpp:57
> +static const size_t ResponseByteSize = 256;

nit: remove the "static" keyword

Also, it appears that this should be called "ResponseFrameSize" instead of "ResponseByteSize" shouldn't it?

> Source/WebCore/platform/audio/HRTFElevation.cpp:60
> +static const float ResponseSampleRate = 44100;

nit: remove the "static" keyword

> Source/WebCore/platform/audio/HRTFElevation.cpp:92
> +static AudioBus* getConcatenatedImpulseResponsesForSubject(const String& subjectName, float sampleRate)

We don't need to pass in a "sampleRate" argument here since we need to load at "ResponseSampleRate"

> Source/WebCore/platform/audio/HRTFElevation.cpp:100
> +        OwnPtr<AudioBus> impulseResponse = AudioBus::loadPlatformResource(subjectName.utf8().data(), sampleRate);

I think a better name for "impulseResponse" would be "concatenatedImpulseResponses"

Instead of using "sampleRate" here, we need to load the concatenated impulse responses at the exact rate as stored in the audio resource -- which is "ResponseSampleRate"

> Source/WebCore/platform/audio/HRTFElevation.cpp:107
> +    size_t expectedLength = static_cast<size_t>(TotalNumberOfResponses * ResponseByteSize * (sampleRate / ResponseSampleRate));

Since we shouldn't be resampling here, this simplifies to:

TotalNumberOfResponses * ResponseFrameSize

in other words we shouldn't scale by (sampleRate / ResponseSampleRate)

> Source/WebCore/platform/audio/HRTFElevation.cpp:162
> +    AudioChannel* totalRightEarImpulseResponse = bus->channelByType(AudioBus::ChannelRight);

We can remove lines 161:162

> Source/WebCore/platform/audio/HRTFElevation.cpp:164
> +    unsigned stopFrame = startFrame + ResponseByteSize;

Keep lines 163:164

(except change ResponseByteSize -> ResponseFrameSize

> Source/WebCore/platform/audio/HRTFElevation.cpp:171
> +    rightEarImpulseResponse->copyFromRange(totalRightEarImpulseResponse, startFrame, stopFrame);

Replace lines 165:171 with:

OwnPtr<AudioBus> preSampleRateConvertedResponse = createBufferFromRange(bus, startFrame, stopFrame);
OwnPtr<AudioBus> response = createBySampleRateConverting(preSampleRateConvertedResponse.get(), false, sampleRate);
AudioChannel* leftEarImpulseResponse = response->channel(AudioBus::ChannelLeft);
AudioChannel* rightEarImpulseResponse = response->channel(AudioBus::ChannelRight);

-- 
Configure bugmail: https://bugs.webkit.org/userprefs.cgi?tab=email
------- You are receiving this mail because: -------
You are the assignee for the bug.