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* update: added delete_name_in()Max Kellermann2008-10-131-1/+18
| | | | | delete_name_in() is similar to delete_path(), but it does not need to look up the parent directory.
* dirvec: dirvec_find() compares basenameMax Kellermann2008-10-131-1/+4
| | | | | | It is invalid to pass a path with the wrong dirname to dirvec_find(). To be able to find a subdirectory only by its basename, compare only the basename of both paths.
* directory: added directory_get_name()Max Kellermann2008-10-132-0/+13
| | | | directory_get_name() returns the base name of the directory.
* playlist: don't use isPlaylist() in deletePlaylist()Max Kellermann2008-10-133-22/+3
| | | | | | The only caller of deletePlaylist() appends PLAYLIST_FILE_SUFFIX, so we can be sure it's already there. We don't need to stat the file, since unlink() does all the checking.
* audio: don't clear input_audio_format on openAudioDevice(NULL)Max Kellermann2008-10-131-3/+1
| | | | | | | Commit 80a2c937 broke resume after pause: it cleared the input_audio_format when it attempted to simplify a complicated expression. Don't clear it, just assign input_audio_format if a new format was specified.
* directory: pass const pointers to inline functionsMax Kellermann2008-10-131-2/+2
| | | | | The inline functions directory_is_empty() and directory_get_path() don't modify the object - pass constant object pointers to them.
* songvec: avoid holding nr_lock during free(3)Eric Wong2008-10-131-3/+5
| | | | | | | We only need to lock sv->nr changes to prevent traversals ( why it's called "nr_lock"). free(3) is a "slow" function on my system; so we can avoid unnecessarily holding a lock long for longer than needed.
* update: allow music_root updates to be queuedEric Wong2008-10-131-3/+2
| | | | | | Previously only updates with subdirectories being specified could be queued. No harm in queueing full updates.
* directory: use mpd_sizeof_str_flex_array for path, tooEric Wong2008-10-132-7/+8
| | | | This way we avoid unnecessary heap allocations.
* tag_item: avoid wasting space when struct is unpackableEric Wong2008-10-132-2/+4
| | | | | Not all compilers support struct packing, and those that don't shouldn't be punished for it.
* song: song_get_url: fix "/" prefix for songs in music_rootEric Wong2008-10-131-1/+1
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* shout: make the protocol configurableAaron McEwan2008-10-121-2/+25
| | | | | Added configuration parameter "protocol" which lets the user choose from 3 shout protocols. This adds support for real shoutcast servers.
* shout: use strcmp() instead of strncasecmp()Max Kellermann2008-10-121-2/+2
| | | | | | Case insensitivity isn't helpful, and comparing only the first 3 bytes of a configured value may encourage users to supply wrong or misleading values.
* alsa: fall back to 16 bit outputMax Kellermann2008-10-121-0/+11
| | | | | | If the sample format isn't supported by the device (i.e. 24 bit on low-end sound chips), fall back to 16 bit output. There is code in pcm_utils.c which converts PCM data to 16 bit.
* pcm_utils: support any number of channels in pcm_sizeOfConvBuffer()Max Kellermann2008-10-121-11/+2
| | | | | When calculating the conversion buffer size, don't hard-code the formulas for only mono<->stereo.
* pcm_utils: support converting N channels to stereoMax Kellermann2008-10-121-0/+26
| | | | | | | Convert any number of channels to stereo. In fact, this isn't really stereo, it's rater mono blown up to stereo. This patch should only make it possible to play 5.1 files at all; "real" conversion to stereo should be implemented, but for now, this is better than nothing.
* pcm_utils: pass output channel count to pcm_convertChannels()Max Kellermann2008-10-121-35/+22
| | | | | | In order to be able to deal with non-trivial conversions, pcm_convertChannels() needs to know both the input and the output channel count. Simplify buffer allocation in that function.
* pcm_utils: moved code from pcm_convertChannels() to separate functionsMax Kellermann2008-10-121-20/+32
| | | | | | | Moved code from pcm_convertChannels() to pcm_convert_channels_1_to_2() and pcm_convert_channels_2_to_1(). Improved the quality of pcm_convert_channels_2_to_1() by calculating the arithmetic mean value of both samples.
* alsa: moved code to alsa_configure()Max Kellermann2008-10-121-13/+18
| | | | | Move code which loads configuration to alsa_configure(). This removes one indent level.
* pcm_utils: pass int16_t* where appropriateMax Kellermann2008-10-121-12/+15
| | | | | Pass int16_t pointers instead of char pointers to functions which can deal with 16 bit audio only.
* player: replaced buffered_before_play with buffering flagMax Kellermann2008-10-121-9/+10
| | | | | | buffered_before_play was copied to struct player because it was used to disable buffering when seeking. Instead of mainaining a copy of this number, move just the flag to the player struct.
* audio: renamed the audio_format variablesMax Kellermann2008-10-121-17/+13
| | | | | | Renamed audio_configFormat to configured_audio_format. Renamed audio_buffer.format to input_audio_format. Simplified its initialization in openAudioDevice().
* audio: removed enum ad_stateMax Kellermann2008-10-121-22/+14
| | | | | All what's left from ad_state is a simple boolean flag (ENABLE / DISABLE). Convert it to a C99 bool.
* audio: removed audio_buffer.bufferMax Kellermann2008-10-121-69/+2
| | | | | | audio.c maintained one of MPD's many layers of audio buffers. It was without any benefit, since playAudio() can simply send the source buffer directly to the audio output plugin.
* output_buffer: converted ob_is_empty() to inlineMax Kellermann2008-10-122-6/+4
| | | | | The function ob_is_empty() is called very often. It's worth it to convert it to an inline function.
* player: added commands QUEUE and CANCELMax Kellermann2008-10-124-155/+74
| | | | | | QUEUE adds a new song to the player's queue. CANCEL clears the queue. These two commands replace the old and complex queueState and queueLockState code.
* playlist: call clearPlayerQueue() only if song is queuedMax Kellermann2008-10-121-23/+15
| | | | | | Simplify and merge several if clauses before the clearPlayerQueue() invocation. Call clearPlayerQueue() only if a song is actually queued; add an assertion for that in clearPlayerQueue().
* player: added struct playerMax Kellermann2008-10-121-48/+75
| | | | | The player struct holds the local variables which used to be passed to all those helper functions in player_thread.c.
* player: removed set_current_song()Max Kellermann2008-10-111-13/+5
| | | | | set_current_song() is an internal function and consists of only one assignment. Eliminate it.
* player: removed player_control.fileTimeMax Kellermann2008-10-113-3/+2
| | | | | | | This variable is superfluous, it is only used to copy its value to player_control.totalTime. Since the original source of this value (song->tag->time) will still be available at this point, we can safely remove fileTime.
* alsa: don't override libasound's buffer_time and period_timeMax Kellermann2008-10-111-20/+19
| | | | | ALSA does a good job measuring its buffer_time and period_time. Don't override its defaults, unless the user demands it.
* alsa: re-enable blocking modeMax Kellermann2008-10-111-15/+2
| | | | | | | | Revert e4f5d6bd "re-enable-nonblocking, but sleep if busy". Non-blocking mode with manual sleeping doesn't help at all (by the way, the patch should have used snd_pcm_wait() instead of my_usleep()). ALSA knows much more about the hardware quirks, so we just let it do the job.
* oss: fix opening default OSS deviceMax Kellermann2008-10-111-1/+1
| | | | | | Leftover from the output API changes: oss_open_default() was changed to return a void*, but it still returned "0" to report success. Report the OssData pointer instead.
* player: don't wake up decoder after every frameMax Kellermann2008-10-101-1/+7
| | | | | | | The decoder was woken up after each chunk which had been played. That caused a lot of superfluous context switches. Wake up the decoder only when a certain amount of the buffer has been consumed. This formula is somewhat arbitrary, and has to be proven experimentally.
* mp3: dither an arbitrary number of channelsMax Kellermann2008-10-101-6/+3
| | | | | | | | The mp3 plugin did not use the MAD_NCHANNELS() value correctly: when a stream was not stereo, it was assumed to be mono, although the correct number was passed to MPD. libmad doesn't support more than 2 channels, but this change allows gcc to optimize its inlining strategy.
* mp3: hard-code dithering to 16 bitsMax Kellermann2008-10-101-8/+6
| | | | | | The dithering function audio_linear_dither() worked for signed 16 bits only anyway, having a variable "bits" just disables important gcc optimizations.
* player: added player_get_audio_format()Max Kellermann2008-10-103-22/+8
| | | | | player_get_audio_format() replaces getPlayerSampleRate(), getPlayerBits(), getPlayerChannels().
* player: added player_control.audio_formatMax Kellermann2008-10-104-17/+12
| | | | This replaces the attributes bits, channels, sampleRate.
* mvp: unsinged integersMax Kellermann2008-10-101-4/+5
| | | | | | Fix some gcc warnings by using unsigned where appropriate. Declare numfrequencies as "const", and replaced the magic number 12 with a sizeof.
* mvp: adapt to the new output device APIMax Kellermann2008-10-101-20/+21
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* mvp: missing includesMax Kellermann2008-10-101-0/+5
| | | | | Again, a plugin which was disabled in my test environment and suffered from compilation errors.
* oss: added OssData.audio_formatMax Kellermann2008-10-101-19/+20
| | | | This replaces the separate properties channels, sampleRate, bits.
* audio_format: added audio_format_frame_size()Max Kellermann2008-10-105-7/+12
| | | | | | | A frame contains one sample per channel, thus it is sample_size * channels. This patch includes some cleanup for various locations where the sample size for 24 bit audio was still 3 bytes (instead of 4).
* audio_format: renamed sampleRate to sample_rateMax Kellermann2008-10-1027-97/+96
| | | | | The last bit of CamelCase in audio_format.h. Additionally, rename a bunch of local variables.
* audio_format: unsigned integersMax Kellermann2008-10-106-15/+15
| | | | "bits" and "channels" cannot be negative.
* update: replaced update_return with global "modified" flagMax Kellermann2008-10-091-65/+41
| | | | | | | There is only once update thread at a time. Make the "modified" flag global and remove the return values of most functions. Propagating an error is only useful for updateDirectory(), since updateInDirectory() will delete failed subdirectories.
* update: make the variable "progress" staticMax Kellermann2008-10-091-1/+1
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* update: don't print debug message when song was not modifiedMax Kellermann2008-10-091-4/+5
| | | | | When a song file was not modified, MPD printed the debug message "not a directory or music", because the first "if" branch did not return.
* update: fix memory leak in directory_update_init()Max Kellermann2008-10-091-1/+4
| | | | | When the update queue is full, directory_update_init() did not free the path argument.
* update: make the job id unsignedMax Kellermann2008-10-094-9/+13
| | | | Since the return value cannot be -1 anymore, we can make it unsigned.
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