| Commit message (Collapse) | Author | Age | Files | Lines |
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We need the decoder object at several places in the AAC plugin. Add
it to mp3DecodeData, so we don't have to pass it around in every
function.
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We need the decoder object at several places in the mp3 plugin. Add
it to mp3DecodeData, so we don't have to pass it around in every
function.
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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.
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The output buffer always contains mpd_sint16; declaring it with that
type saves several casts.
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The previous patch removed all loop specific dependencies from the
num_samples formula; we can now calculate it before entering the loop.
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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.
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The previous patch made mp3Read() flush the output buffer in every
iteration, which means we can eliminate the flush check after invoking
mp3Read().
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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.
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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.
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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.
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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.
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cmd has already been checked before, it cannot have changed meanwhile
because it is a local variable.
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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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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".
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InputPlugin to decoder_plugin, and no camelCase.
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Since inputPlugin.c manages the list of registered decoders, we should
rename the source file.
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"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.
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(Ab)use the decoder_command enumeration, which has nearly the same
values and the same meaning.
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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.
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dc->seekable is already set by decodeStart().
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Since we want to hide mpd internals from the decoder plugins, the
plugins should not check dc->state whether they have already called
decoder_initialized(). Use a local variable to track that.
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Provide access to seeking for the decoder plugins; they have to know
where to seek, and they need a way to tell us that seeking has failed.
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Some decoder commands are implemented in the decoder plugins, thus
they need to have an API call to signal that their current command has
been finished. Let them use the new decoder_command_finished()
instead of the internal dc_command_finished().
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Another big patch which hides internal mpd APIs from decoder plugins:
decoder plugins regularly poll dc->command; expose it with a
decoder_api.h function.
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InputPlugin is the API which is implemented by a decoder plugin. This
belongs to the public API/ABI, so move it to decoder_api.h. It will
later be renamed to something like "decoder_plugin".
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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.
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"break" is so much easier than "eof=1; continue;", when "!eof" is the
loop condition.
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Since we have merged dc->stop, dc->seek into one variable, we don't
have to check both conditions at a time; we can replace "!stop &&
!seek" with "none".
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Similar to the previous patch: pass total_time instead of manipulating
dc->totalTime directly.
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dc->audioFormat is set once by the decoder plugins before invoking
decoder_initialized(); hide dc->audioFormat and let the decoder pass
an AudioFormat pointer to decoder_initialized().
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We are now beginning to remove direct structure accesses from the
decoder plugins. decoder_clear() and decoder_flush() mask two very
common buffer functions.
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Moved all of the player-waiting code to decoder_data(), to make
OutputBuffer more generic.
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