“Foshizzle” is a modified version of the Clang compiler which supports Ebonicode++ (an alternate C++ syntax). Foshizzle supports all the existing features of Clang, but with extensions to more fully express your gansta style. All the regular C++ syntax still works, but you can now substitute Ebonicode++ keywords and operators (see below).
Here’s a sample method that calculates factorials (Note: the semicolons can be replaced with a certain expletive):
int factorialz(int n)
{
int rezult be 1;
slongas (n bepimpin 0)
{
rezult be rezult dimes n;
dissin n;
}
putou rezult;
}
I can’t take all the credit. I was inspired by the Iowa State students that first created Ebonicode (webpage no longer available).
Here are my Clang modifications: ebpp-clang30rc4.patch
Build instructions are near the end of this post.
Continue reading »
Cyclic Redundancy Checks are a quick and convenient way to verify if a data set has changed. They are not meant to be “secure” like a MD5 or SHA-1, but they offer a very practical solution to most problems. CRC’s are also useful for verifying if 2 data sets are equal. Here’s a simple class I use that’s efficient. Honestly, I don’t remember what flavor of CRC32 it implements (sorry), but it has always worked for what I’ve needed. Cheers!
static const uint32_t kCrc32Table[256] = {
0x00000000, 0x77073096, 0xee0e612c, 0x990951ba,
0x076dc419, 0x706af48f, 0xe963a535, 0x9e6495a3,
0x0edb8832, 0x79dcb8a4, 0xe0d5e91e, 0x97d2d988,
0x09b64c2b, 0x7eb17cbd, 0xe7b82d07, 0x90bf1d91,
0x1db71064, 0x6ab020f2, 0xf3b97148, 0x84be41de,
0x1adad47d, 0x6ddde4eb, 0xf4d4b551, 0x83d385c7,
0x136c9856, 0x646ba8c0, 0xfd62f97a, 0x8a65c9ec,
0x14015c4f, 0x63066cd9, 0xfa0f3d63, 0x8d080df5,
0x3b6e20c8, 0x4c69105e, 0xd56041e4, 0xa2677172,
0x3c03e4d1, 0x4b04d447, 0xd20d85fd, 0xa50ab56b,
0x35b5a8fa, 0x42b2986c, 0xdbbbc9d6, 0xacbcf940,
0x32d86ce3, 0x45df5c75, 0xdcd60dcf, 0xabd13d59,
0x26d930ac, 0x51de003a, 0xc8d75180, 0xbfd06116,
0x21b4f4b5, 0x56b3c423, 0xcfba9599, 0xb8bda50f,
0x2802b89e, 0x5f058808, 0xc60cd9b2, 0xb10be924,
0x2f6f7c87, 0x58684c11, 0xc1611dab, 0xb6662d3d,
0x76dc4190, 0x01db7106, 0x98d220bc, 0xefd5102a, Continue reading »
Compiling Live555 Streaming Media with Visual Studio 2008 isn’t obvious. Using Cygwin or MinGW is just a pain and frankly unnecessary. There is no built-in support for VS solution files, because Windows support is a low priority for the Live555 community. This is evidenced by this forum response from Ross Finlayson:
>4. Is it possible to include a Visual Studio solution in the
>distrubution to make it more convenient for Windows developers to
>use live555?
>5. Is it possible for live555 to generate adequate makefiles for
>Windows systems with development environments newer than Visual
>Studio 2003?I have no current plans to change this. (These days, fewer and fewer people seem to be using Windows for development of system software.)
Regardless, Live555 works fine on Windows and is actually quite easy to build. Simply do the following:
- Open the ‘win32config’ file and change the
TOOLS32=...variable to your VS2008 install directory. For me, it’sTOOLS32=C:\Program Files\Microsoft Visual Studio 9.0\VC - In ‘win32config’, modify the
LINK_OPTS_0=...line frommsvcirt.libtomsvcrt.lib. This fixes the link error:
LINK : fatal error LNK1181: cannot open input file 'msvcirt.lib' - Open the Visual Studio command prompt.
- From the ‘live’ source directory, run
genWindowsMakefiles - Now you’re ready to build. Simply run the following commands:
cd liveMedia nmake /B -f liveMedia.mak cd ..\groupsock nmake /B -f groupsock.mak cd ..\UsageEnvironment nmake /B -f UsageEnvironment.mak cd ..\BasicUsageEnvironment nmake /B -f BasicUsageEnvironment.mak cd ..\testProgs nmake /B -f testProgs.mak cd ..\mediaServer nmake /B -f mediaServer.mak
That’s it. You should be good to go.
Here’s a Lua function that returns the percentage of CPU used on Linux. The load includes time in user space and system space. In case you’re curious, I used `top` to get the load for a couple reasons:
- It’s installed by default on most distros (as opposed to the
sysstattools). - It gives the idle time in percentage normalized to 100% (as opposed to
/proc/statwhich requires knowledge of the jiffy’s interval, which can vary).
function cpu_load(interval)
interval = interval or 1
local f = io.popen('top -bp$$ -d ' .. interval .. ' -n 2')
local s = f:read('*all')
-- Find the idle times in the stdout output
local tokens = s:gmatch(',%s*([%d%.]+)%%%s*id')
-- We're interested in the 2nd 'top' idle time
local ii = 1
for idle in tokens do
if ii == 2 then
return 100.0 - tonumber(idle)
end
ii = ii + 1
end
end
You can call it from the Lua shell as:
> load = cpu_load()
> print(load)
43.7
The C99 spec requires that math.h should define the constant float value NAN. However, NAN isn’t defined in the Visual Studio version of math.h, so you have to define it yourself (VS only implements C89). It’s pretty straight-forward for 32-bit machines… here’s my implementation with cross-platform protection:
#ifdef WIN32
#ifndef NAN
static const unsigned long __nan[2] = {0xffffffff, 0x7fffffff};
#define NAN (*(const float *) __nan)
#endif
#endif
This code is adapted from an MSDN example.
Writing code that works on Windows, Linux, and Mac is frequently challenging. Socket programming is no exception. Modern versions of Linux and Mac have full implementations of the latest IPv6 socket API extensions defined in RFC 3493. Windows, however, has only a partial implementation of the original (deprecated) version, RFC 2553. This sounds worse than it is, but it’s something you have to consider.
Note: This post assumes you are already familiar with the socket extensions for IPv6 (RFC 3493).
Linux and Mac
Good news… they both fully support RFC 3493.
Windows
Windows IPv6 support varies based on which version you’re targeting. Microsoft started adding IPv6 in Windows 2000, and they’ve continued adding more of the socket extensions as time went on. Most of the core functionality is present in XP, and what’s missing is easily replaced by using Winsock calls directly (more on this later).
Windows gained IPv6 support while RFC 2553 was still the supported standard. Since then, it has been deprecated by RFC 3493. However, Microsoft doesn’t want to break existing code written against it’s API, so the older API lives on. The main impact of this is that sockaddr_in6 and sockaddr_storage are slightly different on Windows than Mac and Linux. The size of the structures across platforms is the same (the sa_family_t member was shortened), it’s just that the Windows structures don’t begin with the length member. For example:
// Linux and Mac
struct sockaddr_in6 {
uint8_t sin6_len; /* Added in RFC 3493 */
sa_family_t sin6_family;
...
};
struct sockaddr_storage {
uint8_t ss_len; /* Added in RFC 3493 */
sa_family_t ss_family;
...
};
// Windows
struct sockaddr_in6 {
sa_family_t sin6_family;
...
};
struct sockaddr_storage {
sa_family_t ss_family;
...
};I’ve never had a problem with this, because the size of sockaddr_in6 is easily determined (sizeof(sockaddr_in6)) and I always end up casting sockaddr_storage to the specific type (sockaddr_in or sockaddr_in6) based on ss_family.
Besides the data structure differences, it’s important to remember that Microsoft added IPv6 support over multiple versions. Support first appeared in Windows 2000, but more of the extensions have been added over time. Most of the core functionality was present in XP (including multicast), but not everything is implemented as of Windows 7. It’s annoying, but I will say that what’s missing is easily replaced by using Winsock calls directly.
Here’s the breakdown of IPv6 socket extensions by Windows version:
| Socket Extension | 2K | XP | Vista | 7 | Comments |
|---|---|---|---|---|---|
| if_indextoname() | x | x | GetAdaptersAddresses() for XP | ||
| if_nametoindex() | x | x | GetAdaptersAddresses() for XP | ||
| if_nameindex() | GetAdaptersAddresses() (XP, later) | ||||
| if_freenameindex() | |||||
| getaddrinfo() | x | x | x | x | |
| getnameinfo() | x | x | x | x | |
| freeaddrinfo() | x | x | x | x | |
| gai_strerror() | x | x | x | x | |
| inet_pton() | x | x | WSAStringToAddress() (2000, XP) | ||
| inet_ntop() | x | x | WSAAddressToString() (2000, XP) | ||
| All IN6_IS_ADDR_* macros | x | x | x | Ex: IN6_IS_ADDR_LOOPBACK() | |
| struct sockaddr_storage | x | x | x | ||
| Multicast support | x | x | x |
As you can see, you may still have to call Winsock directly depending on what version of Windows you are targeting. In my opinion, programming IPv6 on Windows is a lot easier if you only support XP and later, but I know that’s not always possible.
Summary
Modern operating systems all support IPv6. However, for business reasons, Windows has a slightly older version of the socket API which requires special consideration. My goal was to enumerate those differences to help make the transition to IPv6 smoother. Writing cross-platform code can be a fun challenge at times, but it’s also a little tedious. Hopefully, this post helps ease the pain.
Here’s a simple function using getaddrinfo() that will take an IP address and return the address family (AF_INET for IPv4, AF_INET6 for IPv6, etc). I works on both Linux and Windows. This function will also accept hostnames and return the address family of the first address returned. You can disable this feature (and the corresponding DNS lookup) by passing the AI_NUMERICHOST flag.
// Returns the address family of an address or hostname.
// AF_INET, AF_INET6, or -1 on error.
int getaddrfamily(const char *addr)
{
struct addrinfo hint, *info =0;
memset(&hint, 0, sizeof(hint));
hint.ai_family = AF_UNSPEC;
// Uncomment this to disable DNS lookup
//hint.ai_flags = AI_NUMERICHOST;
int ret = getaddrinfo(addr, 0, &hint, &info);
if (ret)
return -1;
int result = info->ai_family;
freeaddrinfo(info);
return result;
}
See RFC 3493 for more information on the latest socket API for dealing with IPv6.
Recently, I was tasked with picking an AAC audio codec library for one of our products. There were several libraries I had to evaluate, and I needed some quantitative metrics for doing the comparison. I’m not what professionals call an “expert listener”, so I had to do the best with what I had. While creating my test plan, I noticed that more people seemed interested in how I was doing testing rather than the actual results. So I decided to share my approach to audio codec testing.
Note: This is intended to be a pragmatic guide for engineers evaluating codecs. It is not a comprehensive treatment of the subject. The goal is to give readers a solid overview and some practical ideas.
Get Familiar with Psychoacoustics
Psychoacoustics is the study of how humans perceive sound. As you might expect, we humans don’t process sound in a perfect, linear fashion. The physical shape of the ear, the transfer function of the Basilar membrane, and the psychological interpretation of the data all affect how we perceive sound (and by extension, how “good” an audio codec sounds to us).
I highly recommend you start by reading this excerpt from Surround Sound: Psychoacoustics Part 1, by Tomlinson Holman (he created THX for Lucasfilm).
Understand the Codec
Make sure you understand the codec you are testing; not necessarily the implementation, but what tools (i.e. methods) the codec uses for compression. Many codecs have different “profiles”, which describe what subset of available tools are used (e.g. AAC). You should also have some idea how each compression tool works and any short-comings it has. This will help guide you in selecting reference audio samples and knowing what artifacts to listen for.
For an introduction to modern audio compression, read Audio Coding: An Introduction to Data Compression Part 1, and Part 2 (discusses MP3 and AAC). I actually suggest buying the book “Introduction to Data Compression”, by Khalid Sayood.
Understand the API
Make sure you actually read the codec documentation and look at any available code samples. This step has more to do with due-diligence than anything, as I haven’t seen a codec API we couldn’t work with, but you need to do this. This will also help you scope the work required to get a working encoder/decoder for future steps (if your lucky, the sample code can be used).
Choose the Reference Audio Samples
An effective test requires multiple audio samples with different characteristics. There are many types of artifacts a codec can introduce, and your choice of audio samples will dictate how easy they are to detect. It’s also important to pick samples that reflect the actual types of sound the codec to have to deal with. For example, if the final system will primarily be encoding speech, then you should choose more speech-oriented references as opposed to music samples.
Some characteristics you might consider:
- Transients (snare drum): Sensitive to pre-echo and noise “smearing”.
- Tonal structure (clarinet, saxophone): Sensitive to noise and “roughness”.
- Natural speech (male and female voices of various languages): Sensitive to distortion and smearing of “attacks”.
- Complex sound (bag pipes): Stresses the codec.
- High bandwidth (bag pipes): Loss of high frequencies and program-modulated high frequency noise.
It is also possible to use synthetic sounds and sweeps, but this is only recommended for the automated objective tests below.
As a basic guideline, you need 10-25 second “raw” samples recorded at the highest sample rate your system needs to work with. It is vital that the samples you choose have never been compressed with a lossy codec (mp3, AAC, etc)… that would severally limit the quality of your test. For sample rate and size, I suggest 48kHz 16-bit PCM, but a lower rate/size makes sense if the final system is limited in this area. It also makes sense to use a sample rate of 44.1kHz, since many quality audio samples can be ripped losslessly from CD. Just keep in mind that the objective PEAQ test mentioned below requires 48kHz 16-bit PCM, so up-sampling may be required.
The audio samples can be stored in whatever container format you want (raw, WAV, etc) as long as your codec test application can unpack it. This is important to keep in mind… you don’t want to accidentally run the WAV header through the codec (yes, I’ve done this). The container format is more of a practical issue, but it was worth mentioning.
Generate Various Test Samples and Observe CPU Load
This step is pretty straight-forward: wrap the codec in an application and encode the reference audio samples at different bit rates. You should choose bit rates that represent the full spectrum of bit rates that will be used in the final system. While you’re encoding, track the CPU usage on the codec and how many cores it’s using. You may even want to do a separate test running many encodes in parallel (this works nicely if the CPU usage is too low to measure accurately). Make sure to consider application overhead and disk I/O when making measurements.
After encoding, you need to decode back to raw PCM. Clearly label your files so you know what bit rate each one was encoded with. These decoded test samples are what we will be comparing to the original reference samples.
Do a Subjective Test
How you conduct your subjective testing will depend on several factors, such as time constraints, cost, and the required test precision. At the low end, you could simply listen to the test samples in a pair of headphones and judge the quality yourself. For a high precision test, you could do a full ITU BS.1116 test using “expert listeners” in a controlled environment. While these examples represent the extremes, there are many permutations that can give you the desired quality of results.
The most common subjective test is called a “double-blind triple-stimulus with hidden reference” test. The listener hears three samples (commonly labeled A, B, and C) for a period of 10 to 25 seconds. A is always the original reference sample. The next two samples, B and C, are randomly assigned either the test sample from the codec or the original reference sample played again (called the “hidden reference”). The listener must then rate the difference between B and A, and C and A, not knowing which one is the test sample. The grading scale is:
- 5.0 Imperceptible
- 4.0 Perceptible, but not annoying
- 3.0 Slightly annoying
- 2.0 Annoying
- 1.0 Very annoying
Ideally, you would conduct several tests and average the results together. If you do the listening test yourself, your results will be limited to your listening skills and understanding of audio codec artifacts. Here’s a summary of factors that affect the quality of your results:
- The quality of the listener.
- The choice of audio samples.
- The number and duration of the testing.
- The testing environment, including speaker/headphone quality, room design, and listener placement.
- The quality of randomization of sample order to remove any correlation between samples.
- Proper statistical analysis of the combined test results.
A proper subjective test is both expensive and time consuming. It’s important to find the right balance for your particular needs.
Do an Objective Test
Evaluating a codec objectively requires testing methods that correlate well to actual human perception. You can’t simply measure the distortion introduced by the codec using traditional measurements like Signal-to-Noise ratio (S/N) and Total-Harmonic-Distortion (THD), because they don’t correlate well to perceived audio quality. Some distortion is imperceptible to the human ear, and codecs take advantage of this to increase the compression ratio.
Fortunately, the ITU has standardized an objective audio test called PEAQ (BS.1387). The acronym stands for Perceptual Evaluation of Audio Quality. PEAQ uses software to model the entire human auditory system (including blood flow noise in the inner ear) to generate a set of metrics that are used to give a final “quality” score. The original reference signal is compared to a signal run through the codec, and the result is a real number between 0.0 and -4.0. The result is interpreted on the following scale:
- 0.0 = Imperceptible
- -1.0 = Perceptible but not annoying
- -2.0 = Slightly annoying
- -3 .0= Annoying
- -4.0 = Very annoying
Obviously, values closer to zero are better.
The test was developed by a similar group of audio experts that developed BS.1116 (mentioned above) and the results have been validated against a long list of subjective tests done using expert listeners.
There are several free and commercial software packages available for doing PEAQ tests. The best free package I’ve found is AFsp from the McGill Telecommunications and Signal Processing Lab. There’s also peaqb, but there’s a comment that it gives incorrect results. AFsp worked great in my tests and included some helpful tools like CompAudio and InfoAudio.
Summary
Hopefully this post has given you a good starting point and some practical ideas for testing audio codecs. My goal was to provide a pragmatic approach with different options depending on what your actual evaluation needs are. This is in no way a comprehensive treatment of the subject; only an overview. I highly suggest reading some of the books I referenced if you’d like a deeper treatment of the subject. Either way, I hope you found this post helpful.
A colleague of mine found a pretty cool profiling tool for Windows called Very Sleepy. It’s simple, straight-forward, free, and it works… everything I like in a tool (this is why I like the Sysinternals stuff so much). It’s also got call-graph and 64-bit support. Definitely worth checking out.
Some artists from Industrial Light & Magic (ILM) gave the closing keynote at the GPU Technology Conference (GTC) in 2009… it’ s well worth watching by itself (watch here). At GTC 2010, they presented a video talking about how the GPU and CUDA are helping to render effects faster. It’s a short video with lots of cool effects.
ct>
Here’s a copy of the video if it is ever removed (13MB, 3gp).
