| Commit message (Collapse) | Author | Files | Lines |
|---|
|
|
|
|
|
Getting rid of CamelCase; not having typedefs also allows us to
forward-declare the structures.
|
|
|
|
This is generic enough to be used for various purposes. It will
only deconst their argument to work around various braindead
APIs without having to write a new wrapper each time we use one
of those braindead APIs. It does not cast nor do do anything
other than quietly remove the const qualifier for those
braindead APIs.
|
|
Previously we were using a naive randomization algorithm that
could shuffle already shuffled songs. Now we attempt to
correctly[1] implement the Fisher-Yates shuffle.
[1] Note: I absolutely suck at basic arithmetic, so there could
be off-by-one errors in here, too. I've added assertions in
swapSongs and swapOrder functions to more quickly detect them.
|
|
Otherwise we'd be writing to whatever directory that mpd is
running in.
|
|
|
|
And again, convert arguments to const.
|
|
The usual bunch of pointer arguments which should be const.
|
|
Use "unsigned int" whenever negative values are not meaningful. Use
size_t whenever we are going to describe buffer sizes.
|
|
Fix lots of "unused parameter" warnings in the OggFLAC decoder
plugin. Not sure if anybody uses it anymore, since newer libflac
obsoletes it.
|
|
Only include headers which are really needed.
|
|
|
|
Again, remove file descriptor parameters, which are not actually
used. These functions can also be converted to return void.
|
|
sample_size is a variable which is computed at compile time. Declare
it "static const", so the compiler can optimize it away.
|
|
Storing local configuration in global (static) variables is obviously
a bad idea. Move all those variables into the JackData struct,
including the locks.
|
|
|
|
There is only one caller of freeJackData() left: jack_finishDriver().
This function is called by the mpd core, and is called exactly once
for every successful jack_initDriver(). We do not need to clear
audioOutput->data, since this variable is invalidated anyway.
|
|
Over the lifetime of the jack AudioOutput object, we want a single
valid JackData object, so we can persistently store data there
(configuration etc.). Allocate JackData in jack_initDriver(). After
that, we can safely remove all audioOutput->data==NULL checks (and
replace them with assertions).
|
|
No need to destroy the JackData object when an error occurs, since
jack_finishDriver() already frees it. Only deinitialize the jack
library, introduce freeJackClient() for that, and move code from
freeJackData().
|
|
Prepare the next patch: make the "!jd" check independent of the
jd->client initialization. This way we can change the "jd"
initialization semantics later.
|
|
connect_jack() invokes freeJackData() in every error handler, although
its caller also invokes this function after a failure. We can save a
lot of lines in connect_jack() by removing these redundant
freeJackData() invocations.
|
|
Also introduce MUTEFRAME_NONE; previously, the code used "0".
|
|
|
|
The old code called can_seek() with the uninitialized pointer
"isp.is". Has this ever worked? Anyway, initialize "isp" first, then
call can_seek(&isp).
|
|
Move everything related to finding and initializing the WVC stream to
wavpack_open_wvc(). This greatly simplifies its error handling and
the function wavpack_streamdecode().
|
|
Return early when the player thread sent us a command. This saves one
level of indentation.
|
|
If the input stream is not seekable, the try_decode() function
consumes valuable data, which is not available to the decode()
function anymore. This means that the decode() function does not
parse the header correctly. Better skip the detection if we cannot
seek. Or implement better buffering, something like unread() or
buffered rewind().
|
|
The return value of audio_linear_dither() is always casted to
mpd_sint16. Returning long does not make sense, and consumed 8 bytes
on a 64 bit platform.
|
|
The output buffer always contains mpd_sint16; declaring it with that
type saves several casts.
|
|
The previous patch removed all loop specific dependencies from the
num_samples formula; we can now calculate it before entering the loop.
|
|
The output buffer is always flushed after being appended to, which
allows us to assume it is always empty. Always start writing at
outputBuffer, don't remember outputPtr.
|
|
The previous patch made mp3Read() flush the output buffer in every
iteration, which means we can eliminate the flush check after invoking
mp3Read().
|
|
Since we try to fill the buffer in every iteration, we assume that we
should flush the output buffer at the end of each iteration.
|
|
Fill the whole output buffer at a time by using dither_buffer()'s
ability to decode blocks. Calculate how many samples fit into the
output buffer before each invocation.
|
|
Simplifying loops for performance: why check dropSamplesAtEnd in every
iteration, when we could modify the loop boundary? The (writable)
variable samplesLeft can be eliminated; add a write-once variable
pcm_length instead, which is used for the loop condition.
|
|
The variable samplesPerFrame is used only in one single closure. Make
it local to this closure. The compiler will probably convert it to a
register anyway.
|
|
|
|
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.
|
|
Performance improvement by moving stuff out of a loop: skip part of
the first frame before entering the loop.
|
|
|
|
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.
|
|
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.
|
|
The AAC plugin sometimes does not check the length of available data
when checking for magic prefixes. Add length checks.
|
|
Eliminate some duplicated code by using fillAacBuffer().
|
|
Find AAC frames in the input and skip invalid data. This prepares AAC
streaming.
|
|
adts_check_frame() checks whether the buffer head is an AAC frame, and
returns the frame length.
|
|
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.
|
|
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.
|
