The config_audio_format used to contain the configured audio format,
which is copied to out_audio_format.  Let's convert the former to a
boolean, which indicates whether out_audio_format was already set.
This simplifies some code and saves a few bytes.

If a music_chunk does not contain any PCM data, then the "times" and
"bit_rate" attributes are undefined.

On 2009/03/17 Max Kellermann<max@duempel.org> wrote:

> There doesn't seem to be an "official" standard.  I'd say: search for
> TITLE[1] first (the most explicit form), then TITLE1, and finally fall
> back to TITLE.  This makes sure MPD supports every possible standard,
> without breaking.

I've also added some additional checks to make sure entry is long
enough.

The mixer state is defined as offline only if the associated stream is removed.
Signed-off-by: David Guibert <david.guibert@gmail.com>

when the mixer is closed,
- the mainloop is stopped.
- the context is disconnected.
- then the mainloop is freed.
Signed-off-by: David Guibert <david.guibert@gmail.com>

Check if encoder_plugin!=NULL, not encoder_plugin_get (which is a
function).

It's called "vorbis", not "ogg".

The cue sheet embedded in a flac file doen't contain any information
about track titles and similar.  There are three possibilities: Use an
external cue sheet that includes these information, use a tag CUESHEET
with a cue sheet including these information or use tags.  I think the
latter is the best option and is already used by other projects.

g_strerror() is more portable, and guarantees that the returned string
is UTF-8 encoded.

When printing the error message, MPD dereferences the NULL pointer to
print an error message if no audio_output section is present.

[mk: don't run pkg-config when shout is disabled]

Due to a race condition, httpd_client_out_event() could be called even
when its GLib event source was already removed.  Check that case.

When the httpd output is cancelled, it freed all pages, but didn't
remove them from the queue.  Call g_queue_clear() and remove the
write source id.

Copy all tags know to MPD to the vorbis_comment.

Allocate the vorbis_comment object when it's used.  It is not used
anymore in vorbis_encoder_tag().

Don't reinitialize the encoder with every tag.

Prepare the removal of vorbis_encoder.vc.

Use GLib the logging functions g_debug(), g_error() instead.

The function is unused.

This "upload" target uploads generated documentation to
http://www.musicpd.org/doc/, and works only for my user account.

Let's get rid of the "shout" plugin, and the awfully complicated
icecast daemon setup!  MPD can do better if it's doing the HTTP server
stuff on its own.  This new plugin has several advantages:

- easier to set up - only one daemon, no password settings, no mount
  settings
- MPD controls the encoder and thus already knows the packet
  boundaries - icecast has to parse them
- MPD doesn't bother to encode data while nobody is listening

This implementation is very experimental (no header parsing, ignores
request URI, no icy-metadata, ...).  It should be able to suport
several encoders in parallel in the future (with different bit rates,
different codec, ...), to make MPD the perfect streaming server.  Once
MPD gets multi-player support, we can even mount several different
radio stations on one server.

Converted the ogg_page attribute from the vorbis_encoder struct to a
local function of vorbis_encoder_read().  This simplifies some code,
because we don't need to check the page anymore before using it.

Add the "flush" flag, and defer the ogg_stream_flush() call.  Call
ogg_stream_pageout() or ogg_stream_flush() (depending on the "flush"
flag) in vorbis_encoder_read().  This prevents the ogg_page from
getting overwritten by consecutive ogg_stream_pageout() calls.

Moved code from listen_add_address() (listen.c) to socket_util.c.

It is a library which I have written years ago for other projects.
This library is licensed under BSD 2-clause, because it is very
generic.

Even if libsamplerate support is enabled, compile the fallback
resampler.  When the user specifies the option
"samplerate_converter=internal", it is chosen in favor of
libsamplerate.  This may help users with a weak FPU who don't want to
compile a custom MPD from source, because the fallback resampler does
not use floating point operations.