During my tests, it happened that data->position>newPosition. I have
not yet fully understood why this can happen; for now, replace this
with a run-time check.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7334 09075e82-0dd4-0310-85a5-a0d7c8717e4f
The patch "convert blocks until the buffer is full" did not update
data->chunk_length correctly: it added the number of samples, not the
number of bytes. Multiply that with bytes_per_channel
git-svn-id: https://svn.musicpd.org/mpd/trunk@7332 09075e82-0dd4-0310-85a5-a0d7c8717e4f
In the patch "special optimized case for 16bit stereo", the check for
"num_channels==2" was missing.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7331 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Not having to loop for every sample byte (depending on a variable
unknown at compile time) saves a lot of CPU cycles. We could consider
reimplementing this function with liboil...
git-svn-id: https://svn.musicpd.org/mpd/trunk@7330 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Read frame->header.channels once, and pass only this integer to
flac_convert().
git-svn-id: https://svn.musicpd.org/mpd/trunk@7329 09075e82-0dd4-0310-85a5-a0d7c8717e4f
flacWrite() is the only function which sets data->chunk_length. If we
flush the buffer before we return, we can assume that it is always
empty upon entering flacWrite().
git-svn-id: https://svn.musicpd.org/mpd/trunk@7328 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Move the inner loop which converts samples to flac_convert(). There
it is isolated and easier to optimize. This function does not have to
worry about buffer boundaries; the caller (i.e. flacWrite())
calculates how much is left and is responsible for flushing. That
saves a lot of superfluous range checks within the loop.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7327 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Check for flushing the chunk buffer only once per sample, before
iterating over channels and bytes. This saves another 5% CPU cycles.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7326 09075e82-0dd4-0310-85a5-a0d7c8717e4f
AudioFormat.bits is volatile, and to read it, 3 pointers had to be
deferenced. Calculate this value once. This speeds up this function
by 5%.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7325 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Try to only include headers which are really needed. We should
particularly check all "headers including other headers". The
long-term goal is to have a manageable, small API for plugins
(decoders, output) without so many mpd internals cluttering the
namespace.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7319 09075e82-0dd4-0310-85a5-a0d7c8717e4f
There is no danger that gcc will optimize access to OutputBufferChunk
properties, since decoder and player work in different chunk objects.
It is safe to remove "volatile" here.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7318 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Less global variables: at any invocation of decoder_sleep(), we have a
reference to the DecoderControl anyway, so we should pass it. This
costs less than having to call getPlayerData() in every tiny
function. Maybe some day we will be able to have multiple decoders at
the same time...
git-svn-id: https://svn.musicpd.org/mpd/trunk@7316 09075e82-0dd4-0310-85a5-a0d7c8717e4f
To do proper cleanup before exiting, we have to provide a destructor
for OutputBuffer. One day, valgrind will not complain about memory
leaks!
git-svn-id: https://svn.musicpd.org/mpd/trunk@7315 09075e82-0dd4-0310-85a5-a0d7c8717e4f
"end" is not being used anymore, since we moved most OutputBuffer
struct accesses into methods.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7314 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Don't be mean with integer sizes. Although we will probably never
have more than 32k buffered chunks, we should use 32 bit integers for
addressing them. We do not save very much (some of the saved space is
eaten by alignment anyway), but we save at least one assembler
instruction for converting short to int.
This change requires some more explicit casts, because gcc was less
picky when comparing short with a full int.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7313 09075e82-0dd4-0310-85a5-a0d7c8717e4f
First patch without camelCase ;)
output_buffer_skip() lets us eliminate advanceOutputBufferTo(), and
removes yet another external OutputBuffer struct access.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7312 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Second patch to make OutputBuffer self-contained: since OutputBuffer
now knows its own size, we do not need the global variable
"buffered_chunks" anymore.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7311 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Try to make OutputBuffer self-contained, without depending on a global
variable.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7310 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This is the first patch in a series which removes the shared memory,
and moves all the playerData objects into the normal libc heap.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7304 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Now that we do proper locking and signalling instead of continuous
polling for IPC, a deadlock was found: at the end of a song, the
player thread waits until the main thread sets pc->queueState from
PLAYER_QUEUE_DECODE to PLAYER_QUEUE_PLAY. He is never woken up, since
syncPlaylistWithQueue() does not activate the notification. I added
wakeup_player_nb() to setQueueState(), since the player must be
signalled whenever pc->queueState changes.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7303 09075e82-0dd4-0310-85a5-a0d7c8717e4f
needed for compatibility with older gcc. Also, threw in
some warnings for things I find offensive
(declaration-after-statement, shadow)
git-svn-id: https://svn.musicpd.org/mpd/trunk@7299 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Buffer sizes should be size_t. This is safe here, at least not
unsafer than without the patch. I have no idea why audioBufferSize
and audioBufferPos were explicitly declared as signed integer.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7296 09075e82-0dd4-0310-85a5-a0d7c8717e4f
When growing the audioOutput->convBuffer, we can use free()+malloc()
instead of realloc(), which saves a memcpy().
git-svn-id: https://svn.musicpd.org/mpd/trunk@7295 09075e82-0dd4-0310-85a5-a0d7c8717e4f
The audio output plugins should get a constant pointer, because they
must not modify the buffer. Since the size is a non-negative buffer
size in bytes, we should change its type to size_t.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7293 09075e82-0dd4-0310-85a5-a0d7c8717e4f
In my previous patch set, I forgot to change the
pcm_sizeOfConvBuffer() invocation in convertAudioFormat() to also use
size_t.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7292 09075e82-0dd4-0310-85a5-a0d7c8717e4f
calculateCrossFadeChunks() still returns int, although the caller uses
it as an unsigned value. Since the function body checks for negative
values, it is safe to cast to unsigned.
crossFade() takes signed parameters, although it callers pass unsigned
integers. Change declaration to unsigned.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7291 09075e82-0dd4-0310-85a5-a0d7c8717e4f
There were some const pointers missing in the previous const-cleanup
patch.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7290 09075e82-0dd4-0310-85a5-a0d7c8717e4f
libfaad wants uint32_t pointers. Passing a long pointer is bugged on
amd64.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7289 09075e82-0dd4-0310-85a5-a0d7c8717e4f
The patch "Start using song pointers in core data structures" removed
dc->utf8url, and the adaption for wavpack_plugin.c was missing.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7288 09075e82-0dd4-0310-85a5-a0d7c8717e4f
