Second patch: rename the internal struct name. We will eventually
export this type as an opaque forward-declared struct later, so we
can pass a struct pointer instead of a file descriptor, which would
save us an expensive linear lookup.
I don't believe "interface" is a good name for something like
"connection by a client to MPD", let's call it "client". This is the
first patch in the series which changes the name, beginning with the
file name.
linux/list.h is a nice doubly linked list library - it is lightweight
and powerful at the same time. It will be useful later, when we begin
to allocate client structures dynamically. Import it, and strip out
all the stuff which we are not going to use.
It should be obvious in which thread or context a function is being
executed at runtime. The code which was left in decode.c is for the
decoder thread itself; give the file a better name.
This releases several include file dependencies. As a side effect,
"CHUNK_SIZE" isn't defined by decoder_api.h anymore, so we have to
define it directly in the plugins which need it. It just isn't worth
it to add it to the decoder plugin API.
The decoder plugins need this type, so export it in the public API.
This allows is to remove "decode.h" from "decoder_api.h", uncluttering
the API namespace some more.
Unfortunately, we have to pass the DecoderControl pointer to these
inline functions, because the global variable "dc" may not be
available here. This will be fixed later.
The decoder thread is responsible for resetting dc->command after a
command was executed. As a consequence, we can assume that
dc->command is already NONE after decoder_stop().
There is no unlocked caller of clearPlayerQueue(), and the functions
lockPlaylistInteraction() and unlockPlaylistInteraction() are trivial
- merge them.
Since playerPlay() already calls playerStop(), we can remove its
invocation of playerStop() from playPlaylistOrderNumber().
We can also make playerStop a static function.
All (indirect) callers of queueSong() ensure that the queue state is
BLANK, so there is no need to check it in queueSong() again. As a
side effect, queueSong() cannot fail anymore, and can return void.
Also, playlist_queueError and all its error handling can go away.
playerKill() was marked as deprecated, but it seems like a good idea
to do proper cleanup in all threads (e.g. for usable valgrind
results). Introduce the command "EXIT" which makes the player thread
exit cleanly.
playerWait() stops the player thread (twice!) and closes the output
device. It should be well enough to just send CLOSE_AUDIO, without
STOP.
This requires a tiny change to the player thread code: make it break
when CLOSE_AUDIO is sent.
It was possible for the decoder thread to go into an endless loop
(flac and oggflac decoders): when a "STOP" command arrived, the Read()
callback would return 0, but the EOF() callback returned false. Fix:
when decoder_get_command()!=NONE, return EOF==true.
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.
When we introduced decoder_read(), we added code which aborts the read
operation when a decoder command arrives. Several plugins however did
not expect that when they were converted to decoder_read(). Add
proper checks to the mp3 and flac decoder plugins.
The code said "decoder_command==STOP" because that was a conversion
from the old "dc->stop" test. As we can now check for all commands in
one test, we can simply rewrite that to decoder_command!=NONE.
This flag is used internally; it is set by decoder_seek_where(), and
indicates that the decoder plugin has begun the seek process. It is
used for the case that the decoder plugin has to read data during the
seek process. Before this patch, that was impossible, because
decoder_read() would refuse to read data unless dc->command is NONE.
This patch is kind of a dirty workaround, and needs to be redesigned
later.
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().
On our way to stabilize the decoder API, we will one day remove the
input stream functions. The most basic function, read() will be
provided by decoder_api.h with this patch. It already contains a loop
(still with manual polling), error/eof handling and decoder command
checks. This kind of code used to be duplicated in all decoder
plugins.
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 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.
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.
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.
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.
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.
The flag "ready" indicates whether the input stream is ready and it
has parsed all meta data. Previously, it was impossible for
decodeStart() to see the content type of HTTP input streams, because
at that time, the HTTP response wasn't parsed yet.
With the functions decoder_plugin_register() and
decoder_plugin_unregister(), decoder plugins can register a
"secondary" plugin, like the flac input plugin does this for
"oggflac".
