A lot of the preparation was needed (and done in previous
months) in making update thread-safe, but here it is.
This was the first thing I made work inside a thread when I
started mpd-uclinux many years ago, and also the last thing I've
done in mainline mpd to work inside a thread, go figure.
pthreads with our existing signal blocking/handling is broken,
for now just sleep a bit in the child to prevent the CHLD handler
from being called too early. Also, improve error reporting when
handling SIGCHLD by storing the status to be called in the main
task (which can be logged, since we can't do logging inside the
sig handler).
Our linked-list implementation is wasteful and the
SongList isn't modified enough to benefit from being a linked
list. So use a more compact array of song pointers which
saves ~200K on a library with ~9K songs (on x86-32).
It hasn't been used in many years
commit 3a89afdd80
Author: Warren Dukes <warren.dukes@gmail.com>
Date: Sat Nov 20 20:28:32 2004 +0000
remove --update-db option
(SVN r2719)
This allows us to avoid the nasty repetition in strncmp(foo,
bar, strlen(foo)). We'll miss out on the compiler optimizing
strlen() into sizeof() - 1 for string literals for this; but we
don't use this it for performance-critical functions anyways...
This should save a few thousand ops. Not worth it to malloc
for such a small (3-words on 32-bit ARM and x86) structures.
Signed-off-by: Eric Wong <normalperson@yhbt.net>
The function decodeFirstFrame() allocates memory based on data from
the mp3 header. This can make the buffer size allocation overflow, or
lead to a DoS attack with a very large buffer. Cap this buffer at 8
million frames, which should really be enough for reasonable files.
The assertion on "!client_is_expired(client)" was wrong, because
writing the command response may cause the client to become expired.
Replace that assertion with a check.
A crafted mp4 file could cause an integer overflow in mp4_decode
function in src/inputPlugins/mp4_plugin.c. mp4ff_num_samples()
function returns some tainted value. sizeof(float) * numSamples is an
integer overflow operation if numSamples is too huge, so xmalloc will
allocate a small memory region. I constructe a mp4 file, and use
faad2 to open the file. mp4ff_num_samples() returns -1. So I think mpd
bears from the same problem.
Since the buffer size is known at compile time, we can save an
indirection by declaring it as a char array instead of a pointer.
That saves an extra allocation, and we can calculate with the
compile-time constant sizeof(data) instead of the attribute "max_len".
Shout encoder plugins are known at compile time. There is no reason
to use a complex data structure as "List" to manage them at runtime -
just put the pointers into a static array.
[mk: moved this patch after "Refactor and cleanup of shout Ogg and MP3
audio outputs". The original commit message follows, although it is
outdated:]
Creation of shout_mp3 audio output plugin. Basically I just copied the
existing shout plugin and replaced ogg with lame. Uses lame for mp3
encoding. Next step is to pull common functionality out of each shout
plugin and share it between them.
Configuration options for "shout_mp3" are the same as for "shout".
I've perhaps gone a bit overboard, but here's the current rundown:
Both Ogg and MP3 use the "shout" audio output plugin. The shout audio
output plugin itself has two new plugins, one for the Ogg encoder,
and another for the MP3 (LAME) encoder.
Configuration for an Ogg stream doesn't change. For an MP3 stream,
configuration is the same as Ogg, with two exceptions. First, you must
specify the optional "encoding" parameter, which should be set to "mp3".
See mpd.conf(5) for more details. Second, the "quality" parameter is
reversed for LAME, such that 1 is high quality for LAME, whereas 10 is
high quality for Ogg.
I've decomposed the code so that all libshout related operations
are done in audioOutput_shout.c, all Ogg specific functions are in
audioOutput_shout_ogg.c, and of course then all LAME specific functions
are handled in audioOutput_shout_mp3.c.
To develop encoder plugins for the shout audio output plugin, I basically
just mimicked the plugin system used for audio outputs. This might be
overkill, but hopefully if anyone ever wants to support some other sort
of stream, like maybe AAC, FLAC, or WMA (hey it could happen), they will
hopefully be all set.
The Ogg encoder is slightly less optimal under this configuration.
It used to send shout data directly out of its ogg_page structures. Now,
in the interest of encapsulation, it copies the data from its ogg_page
structures into a buffer provided by the shout audio output plugin (see
audioOutput_shout_ogg.c, line 77.) I suspect the performance impact
is negligible.
As for metadata, I'm pretty sure they'll both work. I wrote up a test
scaffold that would create a fake tag, and tell the plugin to send it
out to the stream every few seconds. It seemed to work fine. Of course,
if something does break, I'll be glad to fix it.
Lastly, I've renamed lots of things into snake_case, in keeping with
normalperson's wishes in that regard.
[mk: moved the MP3 patch after this one. Splitted this patch into
several parts; the others were already applied before this one. Fixed
a bunch GCC warnings and wrong whitespace modifications. Made it
compile with mpd-mk by adapting to its prototypes]
Support sending metadata to a shout server using shout_metadata_new()
and shout_metadata_add(). The Ogg Vorbis encoder does not support
this currently.
[mk: this patch was separated from Eric's patch "Refactor and cleanup
of shout Ogg and MP3 audio outputs", I added a description]
Preparing the merge of Eric Wollesen's patch "Refactor and cleanup of
shout Ogg and MP3 audio outputs": we declare one of the struct types
here, to make the merge smoother.
The Ogg encoder is slightly less optimal under this configuration. It
used to send shout data directly out of its ogg_page structures. Now,
in the interest of encapsulation, it copies the data from its ogg_page
structures into a buffer provided by the shout audio output plugin
(see audioOutput_shout_ogg.c, line 77.) I suspect the performance
impact is negligible.
[mk: this patch and its description was separated from Eric's patch
"Refactor and cleanup of shout Ogg and MP3 audio outputs"]
Begin dividing audioOutput_shout.c: move everything OGG Vorbis related
to audioOutput_shout_ogg.c. The header audioOutput_shout.h has to
keep its dependency on vorbis/vorbisenc.h, because it needs the vorbis
encoder types.
For this patch, we have to export several internal functions with
generic names to the ABI; these will be removed later when the encoder
plugin patches are merged.
Remove unused code which is in comments. Remove that comment about
"stolen code", since the plugin has changed much, and it isn't obvious
which parts are derived.
If the output device is already open, it may have modified
outAudioFormat; in this case, outAudioFormat is still valid, and does
not need an overwrite.
As long as the device isn't open, both attributes are not used. Since
they will both be initialized in audio_output_open(), we do not need
the initialization in audio_output_init().
Storing pointers to immutable audio_format structs isn't worth it,
because the struct itself isn't much larger than the pointer. Since
the shout plugin requires the user to configure a fixed audio format,
we can simply copy it in myShout_initDriver().
Save one allocation, since the whole audio_format struct is nearly the
same size as the pointer to it. Check audio_format_defined(af)
instead of af!=NULL.
free(NULL) isn't explicitly forbidden, but isn't exactly good style.
Check the rare case that the audio buffer isn't initialized yet in
closeAudioDevice(). In this case, we also don't have to call
flushAudioBuffer().
To make openAudioDevice() smaller and more readable, move code to a
static function. Also don't use realloc(), since the old value of the
buffer isn't needed anymore, saving a memcpy().
There are too many static variables in audio.c - organize all
properties of the audio buffer in a struct. The current audio format
is also a property of the buffer, since it describes the buffer's
data format.
audio_format_clear() sets an audio_format struct to an cleared
(undefined) state, which is both faster and smaller than memset(0).
audio_format_defined() checks if the audio_format struct actually has
a defined value (i.e. non-zero). Both can be used to avoid pointers
to audio_format, replacing the "NULL" value with an "undefined"
audio_format.
Since the caller chain doesn't care about the return value (except for
COMMAND_RETURN_KILL, COMMAND_RETURN_CLOSE), just return 0 if there is
nothing special. This saves one local variable initialization, and
one access to it.
Also remove one unreachable "return 1" from client_read().
