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Your overriden open and __xstat must not have any side-effects that can be seen by the running process.

No process expects open or __xstat to close and reopen the lowest numbered file descriptor, nor that it should be opened O_CLOEXEC, but this is indeed what syslog does if it finds that the logging socket has failed.

The solution is that you must call closelog after calling syslog to avoid any side-effects becoming visible to the process.

The failure scenario looks like this:

• xargs closes stdin.
• xargs calls open or stat.
• liboverride.so's logging calls syslog which opens a socket, and gets fd 0 as the socket fd.
• xargs calls fork.
• xargs calls dup2 to dup the right piped fd to stdin, and so overwrites fd 0 with the new stdin (the expectation is that nothing else could have opened fd 0)
• xargs is about to call execve but...
• xargs calls stat just before execve
• liboverride.so's logging calls syslog and the implementation detects the sendto has failed, closes fd 0, and reopens fd 0 as the socket fd with O_CLOEXEC and logs a message.
• xargs calls execve to run rev and the O_CLOEXEC socket fd, fd 0, is closed.
• rev expects fd 0 to be stdin, but it is closed and so fails to read from it and writes an error message to that effect on stdout (which is still valid).

When you write wrappers you must take care to avoid such side-effects. In this case it's relatively easy to use closelog, but that may not always be the case.

Depending on your version of xargs there may be more or less work done between the fork and exec and so it may work if liboverride.os's logging function is not called before the exec.

You are trying to append to the variable name with the value of x. You're close, but you need to use bracket notation rather than dot notation.

For example, if x is currently 5, you can get the value of strIngredient5 with drinkResponse.drinks[0]['strIngredient'+x] or drinkResponse.drinks[0][`strIngredient${x}`].

As Lain pointed out, you can also use Object.keys to enumerate all of the keys on the object, then filter for only the keys which start with strIngredient:

Use this

From what I see I assume your collision detection is just based on a single point of the player sprite - most likely it's center point. To make a more precise collision detection, we need to account for the corner points surrounding that object and the direction it's moving to. Additionally we need to check if we would collide with a solid object like a wall before we actually move it to that position. That way we can place it side-by-side to that object.

Consider this example:

Our player sprite is the red square and it's moving at 4 pixels per frame. A you can see the sprite and the blue wall are just 2 pixels apart! If we move to the right by 4 pixels, the bottom right hits the wall.

To get around that problem we need to check the sprite's top-right and bottom-right corners for a possible collision if we want to move to the right.

If we detect a collision - again, before we actually moved something on screen - we put the red hero next to the block instead.

Heres a sophisticated example. Give it focus by a mouse click and use the cursor keys to move around.

The error means that you attempt to use the function packet_destroy before you define it, so the compiler doesn't know how the function should be called.

You can fix this by either moving packet_destroy before any function that uses it, or by adding a declaration of the function before you use it.

I would say this is less of a question about raw audio representation and more a question of is there a lossless convolutional transformation to which I would say no

... as an aside there are plenty of transformations which are lossless (or nearly so) for example when you send audio into a Fourier Transform to convert it from the time domain into its frequency domain representation then perform a second transformation by sending the freq domain representation into an inverse Fourier Transform you will now have normal time domain audio which will match your original source input audio to an arbitrary level of precision ... I know this after writing a golang project which is given an input greyscale photograph which is parsed to synthesize the per pixel light intensity level information into a single channel audio signal (inverse Fourier Transform) which is then listened to (Fourier Transform) to synthesize an output photo which matches the input photo

If you are concerned with bit level accuracy (lossless) you should avoid using mp3 and use a lossless codec or just for starters use WAV format ... any audio CD uses WAV which is just the audio curve in PCM ... its just the points on the audio curve (samples for both channels) ... in your above step 2) if you just feed the audio curve directly into your neural net it will be given your lossless audio data ... the point of typical autoencoders is by definition a lossy transformation since it throws away bit level information

There are several challenges when using audio as input into a Neural Network

1) audio has the aspect of time so depending on what you need you may want to bulk up chunks of audio samples (to make a series of windows of samples) and feed each window as a unit of data into the NN or maybe not

2) As with images, audio has a massive number of data points ... that is each point on the raw audio curve was sampled upstream and now you have typically 44,100 samples per channel per second where semantic meaning is often the result of groupings of these samples ... for instance one spoken word is an aggregate notion easily involving thousands and possibly 10 of thousands of audio sample data points ... so its critical to properly create these windows of audio samples ... bundled into creation of a window of samples is design decision of how the next window will be created : does the next window contain some samples from previous window or are all the samples new ... is the number of audio samples in each window the same or does it vary

So open up the input audio file and read it into a buffer ... to confirm this buffer was created OK just write it out to a file then play back that file and verify its playing OK ... use free open source audio tool called Audacity to open up an audio file and view its audio curve

Consider:

You haven't set up paging, so your query always thinks its on page 1.

Also according to the WP Developer Resources , you shouldn't use query_posts- use WP_Query or pre_get_posts instead, so I'm using WP_Query below to do your 2nd query.

Note that I've assumed you are using this on a static homepage, but I've added the change required for a custom page in comments.

In below code with while loop you are using $row_all but when fetching value you are using $row. Change it, will work.