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* page: added page_new_concat()Max Kellermann2009-03-172-0/+20
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* player_thread: ignore time stamp on empty chunksMax Kellermann2009-03-171-3/+6
| | | | | If a music_chunk does not contain any PCM data, then the "times" and "bit_rate" attributes are undefined.
* flac/cue: added support for TITLE[n] commentsMario Lenz2009-03-171-2/+9
| | | | | | | | | | | | 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.
* pulse_mixer: keep mixer attached when another pulse sink is closed.David Guibert2009-03-171-1/+2
| | | | | | The mixer state is defined as offline only if the associated stream is removed. Signed-off-by: David Guibert <david.guibert@gmail.com>
* pulse_mixer: closeDavid Guibert2009-03-171-1/+14
| | | | | | | | | 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>
* httpd: fixed encoder plugin checkJeffrey Middleton2009-03-171-1/+1
| | | | | Check if encoder_plugin!=NULL, not encoder_plugin_get (which is a function).
* httpd: correct the encoder name in the example mpd.confJeffrey Middleton2009-03-171-3/+3
| | | | It's called "vorbis", not "ogg".
* flac: get CUE track titles from additional FLAC commentsMario Lenz2009-03-164-23/+32
| | | | | | | | 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.
* socket_util: use g_strerror() instead of strerror()Max Kellermann2009-03-161-1/+1
| | | | | g_strerror() is more portable, and guarantees that the returned string is UTF-8 encoded.
* solaris: new audio output plugin for Solaris /dev/audioMax Kellermann2009-03-166-0/+237
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* output_all: check param!=NULL before accessing itMax Kellermann2009-03-161-2/+7
| | | | | When printing the error message, MPD dereferences the NULL pointer to print an error message if no audio_output section is present.
* configure: Merge lame*encoder to lameAvuton Olrich2009-03-161-17/+5
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* configure: don't warn just because we don't have an optional library.Avuton Olrich2009-03-161-3/+2
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* configure: error out against streaming output, not encoder.Avuton Olrich2009-03-161-2/+2
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* configure: Always run pkg-config for shout, define AC_DEFINE if enabledAvuton Olrich2009-03-161-11/+4
| | | | [mk: don't run pkg-config when shout is disabled]
* configure: add shout autoconf argument.Avuton Olrich2009-03-151-0/+5
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* configure: Fail if trying to build with streaming output and no encoderAvuton Olrich2009-03-151-0/+6
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* configure: Move encoders to their own report section.Avuton Olrich2009-03-151-18/+24
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* configure: replace all instances shout*[mp3|ogg] with [lame|oggvorbis]*encoderAvuton Olrich2009-03-151-26/+26
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* httpd_output: check client->write_source_id in handlerMax Kellermann2009-03-151-0/+7
| | | | | Due to a race condition, httpd_client_out_event() could be called even when its GLib event source was already removed. Check that case.
* httpd_output: clear the client's page queue on cancelMax Kellermann2009-03-152-2/+13
| | | | | | 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.
* configure.ac: Enable ENCODER if httpd_output, even if SHOUT is disabled.Avuton Olrich2009-03-151-1/+1
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* vorbis_encoder: added support for all MPD tag typesMax Kellermann2009-03-152-29/+6
| | | | Copy all tags know to MPD to the vorbis_comment.
* vorbis_encoder: removed vorbis_encoder.vcMax Kellermann2009-03-151-4/+6
| | | | | Allocate the vorbis_comment object when it's used. It is not used anymore in vorbis_encoder_tag().
* vorbis_encoder: use vorbis_commentheader_out() in the tag() methodMax Kellermann2009-03-151-6/+15
| | | | Don't reinitialize the encoder with every tag.
* vorbis_encoder: pass vorbis_comment to tag helper functionsMax Kellermann2009-03-151-9/+9
| | | | Prepare the removal of vorbis_encoder.vc.
* log: removed DEBUG() and FATAL()Max Kellermann2009-03-159-49/+22
| | | | Use GLib the logging functions g_debug(), g_error() instead.
* dbUtils: removed printSavedMemoryFromFilenames()Max Kellermann2009-03-152-37/+0
| | | | The function is unused.
* sig_handlers: include signal.h instead of sys/signal.hMax Kellermann2009-03-151-1/+1
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* Makefile.am: added documentation upload targetMax Kellermann2009-03-151-0/+8
| | | | | This "upload" target uploads generated documentation to http://www.musicpd.org/doc/, and works only for my user account.
* httpd: new output plugin to replace "shout"Max Kellermann2009-03-159-0/+1207
| | | | | | | | | | | | | | | | | | | 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.
* page: new library for reference counted buffersMax Kellermann2009-03-153-0/+164
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* vorbis_encoder: make ogg_page a local variableMax Kellermann2009-03-151-24/+13
| | | | | | 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.
* vorbis_encoder: fill the ogg_page in the read() methodMax Kellermann2009-03-151-4/+23
| | | | | | | 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.
* socket_util: added socket_bind_listen()Max Kellermann2009-03-143-41/+85
| | | | Moved code from listen_add_address() (listen.c) to socket_util.c.
* client: use the new fifo_buffer libraryMax Kellermann2009-03-141-48/+41
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* fifo_buffer: added buffering libraryMax Kellermann2009-03-143-0/+290
| | | | | | 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.
* doc: added "audio outputs" section to the user's manualMax Kellermann2009-03-141-4/+277
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* pcm_resample: choose the fallback resampler at runtimeMax Kellermann2009-03-145-20/+41
| | | | | | | | | 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.
* pcm_resample: renamed implementation functionsMax Kellermann2009-03-145-29/+159
| | | | | | Added diversion functions to pcm_resample.c. These check which resampler is enabled at compile time (libsamplerate or fallback). This prepares the following patch.
* output_all: added missing "unused" attributeMax Kellermann2009-03-141-1/+1
| | | | In NDEBUG, clear_tail_chunk() does not use its "chunk" parameter.
* output_all: include chunk.hMax Kellermann2009-03-141-0/+1
| | | | | The source output_all.c accesses music_chunk struct members, but did not include chunk.h directly.
* pcm_volume: use #ifdef instead of #if __i386__Max Kellermann2009-03-141-2/+2
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* tag_pool: make "slots" staticMax Kellermann2009-03-141-1/+1
| | | | The variable is private.
* pcm_resample: return NULL on failureMax Kellermann2009-03-141-4/+4
| | | | Changed "0" to "NULL".
* pcm_convert: return NULL on failureMax Kellermann2009-03-141-1/+1
| | | | Changed "0" to "NULL".
* command: return COMMAND_RETURN_OK in handle_addid()Max Kellermann2009-03-141-1/+1
| | | | | | At the last line of handle_addid(), the playlist_result value has already been evaluated. Don't return this variable, it's the wrong type.
* command: handle the addToPlaylist() result properlyMax Kellermann2009-03-141-1/+2
| | | | | addToPlaylist() has a "enum playlist_result" return value. Convert that to "enum command_return" properly.
* faad: faad_decoder_decode() returns NULL, not falseMax Kellermann2009-03-141-1/+1
| | | | On failure, the function should return NULL, not a boolean.
* Makefile.am: don't let sparse scan C++ sourcesMax Kellermann2009-03-141-1/+1
| | | | sparse cannot deal with C++.
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