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That's a different story, then. At kbps, you should perceive NO differences between ffdcaenc and the commercial encoders.
As for the M. Last edited by El Heggunte; 6th Nov at Reason: add info. Post 9 has been updated with a new binary.
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Replies: 17 Last Post: 11th Nov , Replies: 37 Last Post: 19th Oct , All times are GMT The time now is All rights reserved. Latest software updates. MediaInfoXP 2. The resulting context is an opaque value that should be passed to the other library functions as the first parameter.
The sample rate must be one of the following values: , , or those divided by 2 or 4. Only non-LFE channels have to be specified here. The approximate bitrate is specified in bits per second and may be rounded slightly up or down by the library. This may be useful if the loudness of the hiss resulting from mis-interpretation of the encoded stream as PCM must be reduced.
However, this also results in the reduction of the effective bitrate by DTS CDs are usually encoded with this option. This may be useful for writing the stream as a DVD sound track. This reduces distortions inherent in the filterbank design, but makes the filter output more sensitive to quantization errors introduced later in the encoder.
It makes no sense to use this option for streams with bitrate of 0. Namely, the library adds a 8-byte header to each frame and pads the frame with zeroes to achieve the same bitrate as a stereo bit PCM stream.
There is no way to find out the reason for the error. Now this value is always equal to The input buffer should contain interleaved signed bit samples. This is because it is unclear what functionality related to channel mapping should be exposed. Please mail your suggestions regarding this API to patrakov gmail.
If a non-NULL value is provided in the output parameter, the library encodes the final frame and puts it there. This may be useful because there is a sample latency inherent in the DTS filterbank, so the output frame gets the last portion of the PCM input submitted earlier. The returned value indicates the number of bytes written to the output buffer.
Currently there are no options to select bit encoding or change the endianness. This is a bug. Usage: dcaenc input. Some destinations require a specific bitrate to be specified. To create a CD-compatible DTS file from a multichannel file that needs to have the sample rate of Hz and either 16 or 32 bits per sample , run: dcaenc input. Total users , says I will keep this file, and 69 says I will remove it from my computer. I will keep this file, It's safe. I will remove this file, It's Dangerous file.
We are sorry for the inconvenience. Fix dcaenc. This program is not responding. Click OK to terminate the program. This file is missing or not found. Maybe better to only list with with no caveat for now?? Well, the open-source encoder itself supports those "overkilling bitrates", and so far I have not found a dca decoder that doesn't support them I didn't realize that the lower bitrates were possible with lower sampling rates.
I just knew that was lowest with 48k. I also wasn't aware that encoder allowed for higher than Well crud. SoX is failing me in rendering my w64 to wav to manipulate in dcaenc, as is every other program.
I'm assuming this is due to the wav file size being over 4GB. Thoughts on how to get around this? Worst case scenario I could split the audio track to two pieces, but previous posts seem to imply that there is a way for me to get around having to do this. On a probably unrelated note, SoX also is reporting the wrong length for w64 files.
The actual length is and SoX is reporting as There was a known issue with 24 bit depth and length reporting that was supposedly fixed in latest version. So I re-rendered w64 to 16 bits and now SoX reports length. Is it possible to create a dll also instead of only an exe? The command-line front-end internally just calls the encoder library.
Though in the current Visual Studio solution, the front-end and the library are not in separate projects yet - for sake of simplicity. Wouldn't be that hard to do though ;.
Wouldn't be that hard to do though ; That is good to hear :cool:. The only thing is that my C programming skills are limited and also compiling with Visual Studio I'm more a Delphi guy. So if anyone has the skills and wants to create the dll version then I am here to test it if needed. Just made few 2.
That is good to hear :cool:. It is now linked into the CLI encoder as a static library. I also added a third project to the solution, which builds dcaenc as a DLL file.
The DLL is still untested though ; You should be able to call dcaenc from your application just like it is done in 'main. For Delphi you will have to create your own "header".
Shouldn't be too hard to translate 'dcaenc. I suppose your AVR "expects" Thats possible. The main goal is to get as close to a source lossless as possible. I'll see if I can do anything here. Strange that Arcsoft is not touchy at all about bitrates and endians. No problem. I will translate the header to pas for delphi. To whom this may interest : what do you mean by open, VBR, Lossless? DLL removed. The open mode allows for bit rates not defined by the table.
Table 5. Speaking in general, for 48kHz, kbps per full-range channel should sound "transparent" to most people's ears but YMMV of course In other words, kbps has been overkill since the very-beginning :p Regarding Arcsoft, or even libavcodec, a software decoder is not required to save 1 cent per , manufactured units :rolleyes:. I will, only one question. How do I translate the following to Pascal? Should I just initialize a pointer? Yeah, I think the calling app doesn't need to know how the struct is defined internally - it might even change in future versions.
Speaking in general, for 48kHz, kbps per full-range channel should sound "transparent" to most people's ears but YMMV of course : I mostly agree with this. Under ideal listening conditions with the right song I could nitpick a bit higher maybe to kbps before I lose the ability to discern between the lossy and lossless version. The 48kHz is a big deal though in my opinion and an unfortunate problem with CD.
Increasing bitrate may yield a mathematical equivalence between I find there to be more value in sampling size than in each sample's description. Its like getting driving directions from somebody. Would you rather have very detailed directions but with a couple turns missing or simpler directions with a brief description of every turn? The number of letters used are mathematically equivalent.
I was just ripping some records today and contemplating the misconception that analogue lovers have about digital. Most of the richness that Vinyl collectors such as myself love has less to do with the digital format and more to do with dynamic range compression applied to digital in the mastering process. BTW, kbps is not the only "potentially-problematic" bitrate for dcaenc 1.
So, no need to specify this in the help. Anyone can edit the main. Lord Mulder is surely a very-busy person, and I could not convince him that my ideas are better than his :p I am not forcing anyone to download or to use my MinGW builds of dcaenc.
Selur: many many :thanks: for being less lazy than I :o :D P. Lord Mulder: couldn't those limits be calculated directly thru the dcaenc library and then conveniently output by a command-line switch??? WAV files, and then trying every integer "kbps" between 32 and :scared: Even with the help from a well-crafted batch-file or script, that would not be quite sane :. Well, the CLI front-end simply tries to initialize the dcaenc library with the given parameters.
So what exactly do you want to output on the manpage? Ooops, I have already found 1 incorrection in your list stereo 48kHz —??? I have been coordinating with Uklauer privately and via the bug tracker regarding SoX's length reporting with w64 files.
His response: Do you have a chance to compile and try a current development version from the git repository? If you can't try the development version, a Here the calculations: actual file length in bytes for 16 bit: 3. Apparently he can't compile for windows and neither can I :. Info I shared with him on my project: Here is a bit more feedback. I cannot seem to find a workaround on the length issue. The --ignore-length command doesn't change the outcome at all.
Using the following command line: sox -V4 --ignore-length "large file test. Source: - Output:large file test. Exiting parser. Input File : large file test. Sorry for over-complicating a bit with the piping into another program, but it is irrelevant. The problem is on the SoX end of things. Do I get this right?
You try to pipe a 'w64' file into dcaenc? I don't think its Wave parser will be able to deal with that. The only thing that the "-l" option does is ignoring the length field of the "data" chunk, as that field may not contain a valid value, e.
It seemed like it worked except for the fact that sox has issues in general with w The resulting DTS was the same length as what sox thinks the w64 is. This is a known problem with sox and w64 that is currently being worked on. Recommendations for getting this huge w64 file to dts? I know this is getting slightly off topic, but it is very relevant to dcaenc because many audio tracks will exceed the wav size limit.
If you're looking for a "one-line" solution, I'm afraid you won't find it anytime soon. So if I were you, I'd split that. W64 file into several mono. This is semi-OT and somewhat "late" :o , but also probably useful or interesting at least : Lord Mulder apparently thought I had used a "git clone" of Patrakov's original code.
Actually, I had used the tarball available on Patrakov's web site; that tarball was! Well the command line I posted was attempting to have sox convert the w64 to wav and then pipe that into dcaenc and thereby hopefully get around the 4GB limit of doing two separate steps by first converting to wav.
As long as SoX sends a valid Wave header with valid "fmt" and "data" chunk over the pipe this should work. You'll have to use the '-l' switch for dcaenc, of course. Anyone care to compile a development version of SoX for me?? Here you are
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