| Commit message (Collapse) | Author | Age | Files | Lines |
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Detect misconfiguration during MPD startup.
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The MPD core guarantees that methods are always invoked in a
consistent state. This means we can remove lots of checks from the
volume methods.
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strncasecmp() is locale dependent, but we only need ASCII here.
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Instead of replacing NULL with the default path in the open() method,
pass the default path to config_get_block_string().
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Use config_get_block_string() instead of config_dup_block_string().
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If a (global) mixer has been closed due to failure, don't reopen it
with every volume get/set. Leave it closed until it is explicitly
opened.
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When getting or setting the volume fails, the MPD core close the
mixer. Moved the duplicated code from the mixer plugins.
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Remember if a mixer object is open or closed. Don't call open() again
if it is already open. This guarantees that the mixer plugin is
always called in a consistent state, and we will be able to remove
lots of checks from the implementations.
To support mixers which are automatically opened even if the audio
output is still closed (to set the volume before playback starts),
this patch also adds the "global" flag to the mixer_plugin struct.
Both ALSA and OSS set this flag, while PULSE does not.
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As a side effect, the previous patch added the mixer==NULL checks. It
is now illegal to call mixer functions with a NULL argument. Convert
the runtime checks to assertions.
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The mixer core library is now responsible for creating and managing
the mixer object. This removes duplicated code from the output
plugins.
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When the decoder initialization has not been completed yet, all calls
to dc_seek() will fail, because dc.seekable is not initialized yet.
Wait for the decoder to complete its initialization, i.e. until it has
called decoder_initialized().
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Don't start playback as soon as the "current" song is being loaded
from the state file. That is unclean, and leads to an obscure bug: in
repeat mode, when the song is started (which is yet the last song in
the list), the playlist code marked the very first song in the
playlist as "next" song, because the end of the playlist was wrapped.
It's easier to set up the playback after all songs have been loaded,
and after the random/repeat mode has been set.
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Use audio_output_client_notify instead of g_usleep(1ms) in
audio_output_all_wait() to synchronize with the output_thread. Signal
the audio_output_client_notify object in ao_play().
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Synchronization with the output thread will be implemented in
output_all.c, not in player_thread.c. Currently, that's just a simple
g_usleep(1ms).
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There was a deadlock between the output thread and the player thread:
when the output thread failed (and closed itself) while the player
thread worked with the audio_output object, MPD could crash.
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To prevent a race condition, close the output thread before assigning
the new audio format.
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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.
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If a music_chunk does not contain any PCM data, then the "times" and
"bit_rate" attributes are undefined.
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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.
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The mixer state is defined as offline only if the associated stream is removed.
Signed-off-by: David Guibert <david.guibert@gmail.com>
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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>
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Check if encoder_plugin!=NULL, not encoder_plugin_get (which is a
function).
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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.
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g_strerror() is more portable, and guarantees that the returned string
is UTF-8 encoded.
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When printing the error message, MPD dereferences the NULL pointer to
print an error message if no audio_output section is present.
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Due to a race condition, httpd_client_out_event() could be called even
when its GLib event source was already removed. Check that case.
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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.
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Copy all tags know to MPD to the vorbis_comment.
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Allocate the vorbis_comment object when it's used. It is not used
anymore in vorbis_encoder_tag().
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Don't reinitialize the encoder with every tag.
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Prepare the removal of vorbis_encoder.vc.
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Use GLib the logging functions g_debug(), g_error() instead.
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The function is unused.
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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.
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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.
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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.
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Moved code from listen_add_address() (listen.c) to socket_util.c.
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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.
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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.
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Added diversion functions to pcm_resample.c. These check which
resampler is enabled at compile time (libsamplerate or fallback).
This prepares the following patch.
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In NDEBUG, clear_tail_chunk() does not use its "chunk" parameter.
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The source output_all.c accesses music_chunk struct members, but did
not include chunk.h directly.
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The variable is private.
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