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* alsa_mixer: make "device" and "control" constMax Kellermann2009-03-261-6/+5
| | | | Use config_get_block_string() instead of config_dup_block_string().
* oss_mixer: call oss_find_mixer() in init() methodMax Kellermann2009-03-261-25/+26
| | | | Detect misconfiguration during MPD startup.
* oss_mixer: removed superfluous checksMax Kellermann2009-03-261-8/+8
| | | | | | 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.
* oss_mixer: use g_ascii_strncasecmp() instead of strncasecmp()Max Kellermann2009-03-261-1/+1
| | | | strncasecmp() is locale dependent, but we only need ASCII here.
* oss_mixer: pass default path to config_get_block_string()Max Kellermann2009-03-261-8/+5
| | | | | Instead of replacing NULL with the default path in the open() method, pass the default path to config_get_block_string().
* oss_mixer: make "device" and "control" constMax Kellermann2009-03-261-6/+5
| | | | Use config_get_block_string() instead of config_dup_block_string().
* mixer: don't reopen failed mixer automaticallyMax Kellermann2009-03-263-2/+13
| | | | | | 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.
* mixer_control: close mixer on failureMax Kellermann2009-03-263-5/+16
| | | | | When getting or setting the volume fails, the MPD core close the mixer. Moved the duplicated code from the mixer plugins.
* mixer: added flag "open"Max Kellermann2009-03-268-5/+61
| | | | | | | | | | | | 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.
* mixer_control: don't allow mixer==NULLMax Kellermann2009-03-261-9/+7
| | | | | | 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.
* output_plugin: replaced output_plugin.get_mixer() with mixer_pluginMax Kellermann2009-03-269-77/+44
| | | | | | The mixer core library is now responsible for creating and managing the mixer object. This removes duplicated code from the output plugins.
* player_thread: wait for decoder before seekingMax Kellermann2009-03-252-0/+12
| | | | | | | 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().
* playlist_state: start playing after restore is completeMax Kellermann2009-03-251-18/+17
| | | | | | | | | | | 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.
* output_all: synchronize playback with a notify objectMax Kellermann2009-03-252-2/+3
| | | | | | 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().
* output_all: moved code to audio_output_all_wait()Max Kellermann2009-03-253-10/+25
| | | | | | 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).
* output: protect audio_output.open with the mutexMax Kellermann2009-03-253-15/+33
| | | | | | 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.
* move printAllOutputPluginTypes to output_list.cViliam Mateicka2009-03-215-63/+18
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* output_control: close earlier on format mismatchMax Kellermann2009-03-201-3/+3
| | | | | To prevent a race condition, close the output thread before assigning the new audio format.
* output: convert audio_output.config_audio_format to booleanMax Kellermann2009-03-203-20/+17
| | | | | | | 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.
* 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).
* 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 Kellermann2009-03-162-0/+179
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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.
* 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.
* vorbis_encoder: added support for all MPD tag typesMax Kellermann2009-03-151-29/+4
| | | | 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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* httpd: new output plugin to replace "shout"Max Kellermann2009-03-155-0/+1097
| | | | | | | | | | | | | | | | | | | 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 Kellermann2009-03-152-0/+162
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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 Kellermann2009-03-142-0/+288
| | | | | | 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.
* pcm_resample: choose the fallback resampler at runtimeMax Kellermann2009-03-142-18/+34
| | | | | | | | | 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 Kellermann2009-03-144-29/+158
| | | | | | 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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