[FFmpeg-devel] [PATCH] ALSA for libavdevice
Mon Dec 15 14:34:04 CET 2008
Le quintidi 25 frimaire, an CCXVII, Michael Niedermayer a ?crit?:
> you add 2 values, one being nanoseconds, one being samples. The timebase
> choosen is a multiple of neither.
Neither timebase can represent the value exactly, but both are accurate
enough so that it does not matter. Then the choice should be done, I think,
based on which one is more practical. And IMHO, a second-based unit is more
practical when it comes to integration with other parts of the software.
But I do not insist on it, and I will leave you the final word: if you just
tell me the solution you prefer, I will implement it. As far as I remember,
the possibilities are: nanoseconds, microseconds, samples, 128th of samples;
and the origin may be the Epoch or the initialization of the stream.
> Before you add them you round both incorrectly, correct being to the next
> closest integer.
You are perfectly right. For now I fixed that with a "+500" and a
> where can any localtime be passed to display?
> This localtime isnt ending up in an encoded file, is it? This would be a
> security issue
I have never actually seen the conversion done, but I have seen several live
streams where the timestamp looked suspiciously like an Unix time.
I do not see how it would be a security issue: a lot of things expose the
system time (the HTTP date header, IP timestamps) or the
creation/modification time of a file (headers in an archive). This is hardly
a confidential information.
> please explain me how av sync works if the samplerate is changed and thrown
An output format can not change the parameters of its streams, can it? That
means it must simply fail if the requested sample rate is not available. I
think I like this solution better anyway.
> NULL is not a good context, besides this line is likely not that usefull
This line is indeed redundant with the summary printed by ffmpeg. I removed
I will submit the new version of the patch as soon as I have your advice on
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