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* player: don't send partial frames of silenceMax Kellermann2008-10-231-1/+7
| | | | | | Another partial frame fix: the silence buffer was 1020 bytes, which had room for 127.5 24 bit stereo frames. Don't send the partial last frame in this case.
* pcm_utils: added 24 bit conversion functionsMax Kellermann2008-10-231-0/+108
| | | | | | 24 bit output is as important as 16 bit output. Provide a pcm_convert() implementation which can convert to 24 bit with as little quality loss as possible.
* pcm_utils: generic pcm_convert_size() implementationMax Kellermann2008-10-231-21/+2
| | | | | | The old pcm_convert_size() ignored most of the destination format, e.g. it did not check its sample size, and assumed it is 16 bit. Simplify and universalize it by using audio_format_frame_size().
* pcm_utils: moved code to pcm_convert_16()Max Kellermann2008-10-231-21/+39
| | | | | pcm_convert() converted only to 16 bit. To be able to support other sample sizes, move that 16 bit specific code to a separate function.
* pcm_channels: added 24 bit implementationsMax Kellermann2008-10-232-0/+84
| | | | | The 24 bit implementation is mostly copy'n'paste of the 16 bit version, except that the data type is int32_t instead of int16_t.
* pcm_utils: moved channel conversion functions to pcm_channels.cMax Kellermann2008-10-234-82/+138
| | | | Separate code from pcm_utils.c to keep it small and simple.
* pcm_resample: implemented 24 bit resamplingMax Kellermann2008-10-233-0/+94
| | | | | | | Similar to pcm_resample_16(), implement pcm_resample_24(). The 24 bit implementation is very similar, but it uses src_int_to_float_array() instead of src_short_to_float_array() before sending data to libsamplerate.
* pcm_resample: moved code to pcm_resample_set()Max Kellermann2008-10-231-29/+40
| | | | | A future patch will implement a 24 bit resampler. To unify code, move code which can be shared to a separate function.
* pcm_resample: eliminated "sample" local variablesMax Kellermann2008-10-231-9/+3
| | | | | Copy from source to destination buffer directly, don't use the temporary variables "lsample" and "rsample".
* pcm_resample: don't resample partial samplesMax Kellermann2008-10-232-0/+9
| | | | | Added assertions which ensure that there are no partial samples in the source buffer.
* pcm_resample: don't hard-code sample sizeMax Kellermann2008-10-232-4/+4
| | | | | | Use sizeof(sample) instead of hard-coding "2". Although we're in 16 bit right now, this will make code sharing easier when we support other sample sizes.
* pcm_utils: moved code to pcm_resample.cMax Kellermann2008-10-237-176/+306
| | | | | | Separate the resampling code from the rest of pcm_utils.c. Create two sub-libraries: pcm_resample_libsamplerate.c and pcm_resample_fallback.c.
* command: fix boolean value parserMax Kellermann2008-10-231-1/+1
| | | | | Due to a logic error, no value was valid for the boolean value parser. Replace "||" with "&&".
* mp3: send 24 bit PCM dataMax Kellermann2008-10-231-63/+24
| | | | | | | libmad produces samples of more than 24 bit. Rounding that down to 16 bits using dithering makes those people lose quality who have a 24 bit capable sound device. Send 24 bit PCM data, and let the receiver decide whether to apply 16 bit dithering.
* mp3: use sizeof(sample) instead of hard-coded "2"Max Kellermann2008-10-231-2/+3
| | | | | We are going to convert the code to 24 bit; don't hard-code a sample size of 2 bytes.
* pcm_dither: added generic 24 to 16 bit ditheringMax Kellermann2008-10-235-8/+139
| | | | | Copied and adapted code from the mp3 decoder plugin. This library now replaces the old and low-quality function pcm_convert_24_to_16().
* audio: allow 24 and 8 bit outputMax Kellermann2008-10-231-4/+2
| | | | | | | I added 24 bit support a while ago, but it wasn't possible to force 24 bit output. Add 24 and 8 bit to the list of allowed sample sizes. Although 8 bit audio isn't as widely used as 24 bit, there is no reason to exclude it.
* output_buffer: don't split framesMax Kellermann2008-10-233-1/+16
| | | | | | | Splitting a frame between two buffer chunks causes distortion in the output. MPD used to assume that the chunk size 1020 would never cause splitted frames, but that isn't the case for 24 bit stereo (127.5 frames), and even less for files with even more channels.
* stored_playlist: don't map files outside the databaseMax Kellermann2008-10-231-2/+3
| | | | | | Don't attempt to map paths which are already absolute with map_song_fs(): check with song_in_database() instead of song_is_file().
* stored_playlist: emit idle event on deleteMax Kellermann2008-10-231-0/+1
| | | | The "rm" command did not send notifications to idle clients. Add it.
* stored_playlist: renamed and moved spl_delete() to stored_playlist.cMax Kellermann2008-10-235-17/+19
| | | | The function deletePlaylist() shouldn't be in playlist.c.
* stored_playlist: spl_load() returns GPtrArrayMax Kellermann2008-10-233-146/+75
| | | | | Don't use our deprecated linked list library, use GLib's GPtrArray instead.
* command: check over/underflows in check_int()Max Kellermann2008-10-231-2/+13
| | | | | The "long" result of strtol() was implicitly casted down to a 32 bit integer. Add some range checking instead.
* command: added check_unsigned() / check_bool()Max Kellermann2008-10-231-13/+57
| | | | | | | Many command arguments must not be negative; add a separate parser/checker function for that. For the same reason, add check_bool(). This eliminates two strange special cases handlers from check_int().
* stored_playlist: unsigned index argumentsMax Kellermann2008-10-232-6/+6
| | | | | | Pass index arguments as unsigned integers. They must not be negative, and even if some caller accidently passes -1, it won't pass the bound checks (since it's now 2**32-1).
* playlist: unsigned integersMax Kellermann2008-10-233-150/+131
| | | | | | There are some integers which have a "magic" -1 value which means "undefined" or "nothing". All others can be converted to unsigned, since they must not contain a negative number.
* command: use the bool datatype instead of intMax Kellermann2008-10-222-51/+54
| | | | | Instead of returning 0 for success and -1 for failure, return true or false. This seems more natural.
* command: converted COMMAND_RETURN_* macros to enumMax Kellermann2008-10-222-168/+173
| | | | | | | | Also add names for "error" and "ok". I don't like passing anonymous integer codes around. This is not yet complete: lots of functions (e.g. in playlist.c) follow the same convention of -1/0, and these have to be adapted, too.
* command: replaced "goto" with "break"Max Kellermann2008-10-221-2/+2
| | | | http://xkcd.com/292/
* command: no CamelCaseMax Kellermann2008-10-224-238/+260
| | | | Eliminate CamelCase in all public and static functions.
* command: removed CommandHandlerFunction typedefMax Kellermann2008-10-221-3/+1
| | | | | The typedef CommandHandlerFunction is only used once. Move its type into the command struct.
* ack: converted ACK_ERROR_* macros to enumMax Kellermann2008-10-223-16/+18
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* stored_playlist: send timestampsMax Kellermann2008-10-221-0/+8
| | | | | Send last modification timestamps to the client. This allows the client to see when another client modifies a stored playlist.
* command: added command "listplaylists"Max Kellermann2008-10-221-0/+17
| | | | | "listplaylists" returns a list of all stored playlists. This command seems more elaborate than listing them below "lsinfo".
* command: added print_spl_list()Max Kellermann2008-10-223-22/+18
| | | | | The function print_spl_list() replaces the old function lsPlaylists() from ls.c.
* stored_playlist: added spl_list()Max Kellermann2008-10-223-73/+106
| | | | | | spl_list() provides an interface for enumerating all stored playlists. This separates the internal playlist logic from the protocol specific function lsPlaylists().
* stored_playlist: spl_append_uri() returns enum playlist_resultMax Kellermann2008-10-222-5/+4
| | | | | The return value of spl_append_uri() was somewhat buggy: some branches returned ACK_* values, and some an enum playlist_result. Unify this.
* stored_playlist: de-CamelCase moved functionMax Kellermann2008-10-223-6/+6
| | | | | Rename addToStoredPlaylist() to spl_append_uri(), and remove the clearStoredPlaylist() macro.
* stored_playlist: moved functions from playlist.cMax Kellermann2008-10-224-32/+26
| | | | | | The two functions clearStoredPlaylist() and addToStoredPlaylist() don't belong into playlist.c. clearStoredPlaylist() was a wrapper for spl_clear(), and is converted into a CPP macro for now.
* stored_playlist: no CamelCaseMax Kellermann2008-10-225-34/+39
| | | | Renamed all public functions, prefix is "spl_".
* renamed storedPlaylist.c to stored_playlist.cMax Kellermann2008-10-226-8/+8
| | | | No CamelCase in file names.
* command: added "commands" array instead of registering manuallyMax Kellermann2008-10-221-193/+145
| | | | | | The list of commands is known at compile time. Instead of creating a linked list on startup, we can just register all commands in a static sorted array.
* command: make command pointers constantMax Kellermann2008-10-221-12/+13
| | | | | | The command pointers which are passed around aren't being modified - in fact, no command pointer must be modified once it has been added to the commandList.
* command: renamed CommandEntry to struct commandMax Kellermann2008-10-221-12/+10
| | | | No CamelCase and no struct typedefs.
* pcm_utils: added pcm_convert_init()Max Kellermann2008-10-214-2/+9
| | | | | | Instead of manually calling memset(0) on the pcm_convert_state struct, client code should use a library function from pcm_utils.c. This way, we can change the semantics of the struct easily.
* pcm_utils: no CamelCaseMax Kellermann2008-10-215-88/+89
| | | | Renamed all functions which were still in CamelCase.
* pcm_utils: renamed ConvState to struct pcm_convert_stateMax Kellermann2008-10-215-9/+12
| | | | No CamelCase, and a struct instead of a typedef.
* ffmpeg: don't pass pointer as hexadecimal stringMax Kellermann2008-10-211-20/+27
| | | | | | | | | | | | | | Casting a pointer to some sort of integer and formatting it into a string isn't valid. A pointer derived from this hex string won't work reliably. Since ffmpeg doesn't provide a nice API for passing our pointer, we have to think of a different hack: ffmpeg passes the exact URL pointer to mpdurl_open(), and we can make this string part of a struct. This reduces the problem to casting the string back to the struct. This is still a workaround, but this is "sort of portable", unless the ffmpeg people start messing with the URL pointer (which would be valid according to the API definition).
* ffmpeg: detect which ffmpeg headers should be includedMax Kellermann2008-10-211-0/+6
| | | | | | | | | Since ffmpeg svn r12865, you have to include libavcodec/avcodec.h instead of avcodec.h. This cannot be checked at compile time, instead we have to add a check to configure.ac. Viliam's original ffmpeg plugin was based on the newer ffmpeg library, while my Debian installation had the older version. My attempt to correct his include statements wasn't correct after all.
* update: fix multiple deletes from *vec iteratorsEric Wong2008-10-212-2/+10
| | | | | | | | | | | {song,dir}vec_for_each each failed to gracefully handle deleted files when iterating through. While we were thread-safe, we were not safe within the calling thread. If a callback we passed caused sv->nr to shring, our index would still increment; causing files to stay in the database. A way to test this is to remove 10 or so contiguous songs from a >10 song directory.
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