| Commit message (Collapse) | Author | Age | Files | Lines |
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Preparing for simplifying and thus speeding up the dithering code:
moved dithering to a separate function which contains a trivial loop.
With this patch, only one sample is dithered at a time, but the
following patches will allow us to dither a whole block at a time,
without complicated buffer length checks.
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Performance improvement by moving stuff out of a loop: skip part of
the first frame before entering the loop.
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Copy some code from aac_decode() to aac_stream_decode() and apply
necessary changes to allow streaming audio data. Both functions might
be merged later.
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initAacBuffer() should really only initialize the buffer; currently,
it also reads data from the input stream and parses the header. All
of the AAC buffer code should probably be moved to a separate library
anyway.
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The AAC plugin sometimes does not check the length of available data
when checking for magic prefixes. Add length checks.
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Eliminate some duplicated code by using fillAacBuffer().
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Find AAC frames in the input and skip invalid data. This prepares AAC
streaming.
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adts_check_frame() checks whether the buffer head is an AAC frame, and
returns the frame length.
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Shifting from the buffer queue is a common operation, and should be
provided as a separate function. Move code to aac_buffer_shift() and
add a bunch of assertions.
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When checking for EOF, we should not check whether the read request
has been fully satisified. The InputStream API does not guarantee
that readFromInputStream() always fills the whole buffer, if EOF is
not reached. Since there is the function inputStreamAtEOF() dedicated
for this purpose, we should use it for EOF checking after
readFromInputStream()==0.
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Fill the AacBuffer even when nothing has been consumed yet. The
function should not check for consumed data, but for free space at the
end of the buffer.
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Return instead of putting all the code into a if-closure. That saves
one level of indentation.
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adtsParse() always returns 1, and its caller does not use the return
value.
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Since we eliminated the parameters retFileread and retTagsize in all
callers, we can now safely remove it from the function prototype.
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Make the code more readable by hiding big formulas in an inline
function with a nice name.
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Don't use CPP macros when you can use C enum... this also allows
better type checking.
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We want to expose the AudioFormat structure to plugins; remove some
clutter by moving its declaration to a separate header file.
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Anonymous code blocks just to declare variables look ugly. Move the
variable declarations up and disband the code block.
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Similar to previous patch: eliminate one variable by using "break".
This also simplifies the code since we can remove one level of indent.
[ew: rewritten to match current API]
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"break" is so much easier than "eof=1; continue;", when "!eof" is the
loop condition.
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Include only headers which are really required. This speeds up
compilation and helps detect cross-layer accesses.
[ew: minor fixups to not break on new core]
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Also enable -Wunused-parameter - this forces us to add the gcc
"unused" attribute to a lot of parameters (mostly library callback
functions), but it's worth it during code refactorizations.
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Fix a "unused argument" warning, and several warnings regarding void
pointer calculation.
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Using struct iovec means having to cast iov_base everywhere
we want to do pointer arithmetic. Instead, just use rbvec
which can be safely casted to iovec whenever we use
the readv/writev functions.
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Fix a "signed/unsigned comparison warning", and several void pointer
math warnings.
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This avoids writing the metadata of a static song into
the URL of song; leading to confusing looking playlists.
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I considered calling it from metadata_pipe_recv() in
the past, but it's not necessary, so just inline it
again to simplify things.
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unK reported a bug in which explicitly calling "delete"
on each song would cause mpd to lock up. This is actually
triggered when the only song on the mpd playlist is deleted.
Additionally, add an extra assertion to ensure we play
a valid, non-NULL song in play_order_num().
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This way if we previously had a seek error, starting
to play a new song will immediately update the status
metadata.
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Hopefully this fixes a segfault I experienced inside
freeMpdTag earlier with the metadata_pipe. I could
not reproduce the segfault again, however.
Regardless, if multiple threads rely on this, we need to
atomically increment/decrement these counters.
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When we send metadata, there's a remote chance that our pipe is
full and our tag will be silently discarded. If that happens,
the readers will never have a chance to free the tag, so ensure
we free it before returning to the caller.
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I just forgot to reenable/reinitialize it after the core rewrite.
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This has been tested for both playback of streams and
outputting to streams, and seems to work fine with minimal
locking. This reuses the sequence number infrastructure
in OutputBuffer for synchronizing metadata payloads; so
(IMNSHO) should be much more understandable than various
flags being set here and there..
It could still use some cleanup and much testing, but
synchronization issues should be minimal.
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When deleting previous songs, we forgot to update the
playlist.queue value, causing syncPlaylistWithQueue to
trigger a false sync and screw with the playlist.current
pointer; causing the currentsong command to return
an incorrect song.
Thanks to unK to reporting this bug!
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When moving songs around, we forgot to update the
playlist.queue value, causing syncPlaylistWithQueue to
trigger a false sync and screw with the playlist.current
pointer; causing the currentsong command to return
an incorrect song.
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It's possible to calculate an impossibly small value that
we don't have a chance to xfade. Don't die if we can't
find the boundary to start crossfading on
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ob.xfade_time can be changed by the main process without
locking, so copy the float value into a local variable
and recheck the local variable for zero before
continuing.
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We don't assert on xfade_time > 0 inside any of the xfade
calculations since we have no lock around xfade_time.
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It sounds nasty and we didn't do it before the core-rewrite
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It's redundant, we already track that stuff elsewhere.
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This fixes the case where we wouldn't start playing a newly
added song if we're near the end of the playlist and done
decoding the last song (but still playing from the
buffer).
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buffer_before_play is a prebuffer; always respecting it
is almost as good as having no buffer at all. So we only
respect it when we haven't played anything.
Bugs that were a side effect of this also got fixed:
The player would not stop when we got to the end of the last
song on non-repeating playlists. The playlist would
continuously show the song in the last few seconds of
playback, and never move.
Having crossfade enabled would also amplify the above effect.
So, as a side effect, crossfade now correctly handles
end-of-playlist conditions, as well. It will fade out to
silence when we're at the end of a playlist.
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There are still some places where we try to call this
function without the playlist being stopped. It's really
harmless, to call it and just break out immediately, so
change the assertion.
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data->muteFrame won't necessarily get cleared when it
enters that block of code, so we don't signal the action
as complete until it is actually cleared.
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Hopefully this fixes the skipping problem Qball reports
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