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".
"decoder plugin" is a better name than "input plugin", since the
plugin does not actually do the input - InputStream does. Also don't
use typedef, so we can forward-declare it if required.
PlayerControl.command replaces the old attributes play, stop, pause,
closeAudio, lockQueue, unlockQueue, seek. The main thread waits for
each command synchronously, so there can only be one command enabled
at a time anyway.
The wavpack decoder plugin implements a hack, and it needs the song
URL for that. This API (and the hack) should be revised later, but
add that function for now.
