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* fluidsynth: added "experimental" comments in the source fileMax Kellermann2009-04-251-0/+9
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* vorbis: move #define out of function to top of sourcesAvuton Olrich2009-04-021-1/+1
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* decoder: Rename all main decoder plugins functions to *decoder_plugin.Avuton Olrich2009-04-024-4/+4
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* flac: fixed "unused variable" warning without libcueMax Kellermann2009-04-011-2/+2
| | | | Move the declaration of "i" into the "for" loop.
* Tag subtracks according to "cuesheet" vorbis comment valueJochen Keil2009-03-311-4/+32
| | | | | | | Cuesheets are often saved as vorbis comment flac files (CUESHEET=.. case doesn't matter). We can parse this now and use the information to tag the subtracks (from the embedded cuesheets).
* track length is computed correctly nowJochen Keil2009-03-311-2/+1
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* free previously allocated flac metadata objectJochen Keil2009-03-311-0/+1
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* remove old commented codeJochen Keil2009-03-311-8/+0
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* Bugfix for time/offset in flac pluginJochen Keil2009-03-281-3/+3
| | | | | | Previous cast to float didn't have any effect because one value is uint and the other is a floating type but the number itself is even.. This caused some tracks to end before they were really at an end.
* mpcdec: changed plugin name to "mpcdec"Max Kellermann2009-03-271-3/+3
| | | | The "mpcdec" plugin is based on the libmpcdec library.
* mpcdec: support the new libmpcdec SV8 APIMax Kellermann2009-03-271-14/+90
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* mpcdec: no CamelCaseMax Kellermann2009-03-271-57/+65
| | | | Renamed variables and functions.
* 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.
* 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.
* faad: faad_decoder_decode() returns NULL, not falseMax Kellermann2009-03-141-1/+1
| | | | On failure, the function should return NULL, not a boolean.
* all: Update copyright header.Avuton Olrich2009-03-1319-127/+147
| | | | | | | | This updates the copyright header to all be the same, which is pretty much an update of where to mail request for a copy of the GPL and the years of the MPD project. This also puts all committers under 'The Music Player Project' umbrella. These entries should go individually in the AUTHORS file, for consistancy.
* Initial support for embedded cue sheets found in flac filesJochen Keil2009-03-093-6/+463
| | | | | | | | | So far only seekpoints are supported, so no proper tagging yet except for track number and track length. Tagging should be done by parsing the cue sheet which is often embedded as vorbis comment in flac files. Furthermore the pathname should be configurable like "%A - %t - %T", where %A means Artist, %t track number and %T Title or so.
* decoder_plugin: added method container_scan()Jochen Keil2009-03-091-0/+1
| | | | | [mk: fixed whitespace errors; use delete_song() instead of songvec_delete()]
* flac: explicitly check for STOP commandMax Kellermann2009-03-051-1/+2
| | | | | | After the decoder command was obtained, don't wait until libflac detects EOF (as a side effect), quit the decoder immediately. This check was missing completely.
* flac: check command after flac_process_single() failureMax Kellermann2009-03-051-3/+6
| | | | | | When the MPD core sends the decoder a command while flac_process_single() is executed, this function fails. Abort the decoder only if not seeking. This fixes a seeking bug.
* ffmpeg: fix version comparision for av_get_bits_per_sample_format() ↵Viliam Mateicka2009-03-031-1/+1
| | | | | | implemetation function was implemented in the version we are comparing to so there must be higher or equal
* ffmpeg: support for new metadata apiViliam Mateicka2009-03-031-0/+39
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* ffmpeg: use ffmpeg's sampleformat for output formatViliam Mateicka2009-03-021-1/+6
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* flac: parse stream tagsMax Kellermann2009-03-012-2/+19
| | | | | Parse the vorbis comments in libflac's metadata_callback and pass them as tag struct to the decoder API.
* audiofile: added 24 bit supportMax Kellermann2009-03-011-1/+8
| | | | | Don't hard code the "bits" parameter to 16. Try to use the input's sample format, if possible.
* audiofile: removed duplicate decoder_get_command() callsMax Kellermann2009-02-281-12/+16
| | | | | decoder_data() returns a decoder_command, no need to call decoder_get_command() twice after decoder_command().
* audiofile: refuse to play non-seekable filesMax Kellermann2009-02-281-0/+5
| | | | | | | | | If an input_stream is not seekable, libaudiofile fails to play at all: Audio File Library: unrecognized audio file format [error 0] Since we know in advance whether the input_stream is seekable, just refuse to play on a non-seekable stream.
* audiofile: no CamelCaseMax Kellermann2009-02-281-9/+9
| | | | Renamed several variables and a function.
* tag: no CamelCaseMax Kellermann2009-02-272-2/+2
| | | | Renamed numOfItems to num_items.
* ls: moved generic URI utilities to uri.cMax Kellermann2009-02-251-1/+1
| | | | | "ls" is a bad name for a library which parses URIs. We'll move the rest of the "ls" library later.
* ffmpeg: add all known ffmpeg extensions and mime-types.Avuton Olrich2009-02-231-18/+77
| | | | | | | | | | | | | After much research[1][2][3] this should be the majority of currently supported file extensions and mime-types for the currently supported ffmpeg formats. This list maybe incomplete, but it's more complete than anything else out there that I've been able to find. This list needs to be updated every now and again as the ffmpeg sources support more formats. 1. Sources 2. wiki.multimedia.cx 3. filext.com
* faad: use faacDecFrameInfo instead of NeAACDecFrameInfoMax Kellermann2009-02-191-1/+1
| | | | Don't use libfaad's internal type names.
* mp4ff: always free the mp4ff_read_sample() bufferMax Kellermann2009-02-181-2/+2
| | | | | When mp4ff_read_sample() returns a value bigger than zero, it guarantees that the buffer is set. Remove the check.
* mp4ff: don't allocate seek_table when input is not seekableMax Kellermann2009-02-181-2/+10
| | | | Don't waste any precious memory when the seek_table cannot be used.
* mp4ff: include cleanupMax Kellermann2009-02-181-4/+4
| | | | Don't include limits.h, use GLib constants instead.
* mp4ff: use faacDecInit2() to find the AAC trackMax Kellermann2009-02-181-49/+36
| | | | | | | | | Use faacDecInit2() instead of AudioSpecificConfig() to detect the AAC track in the MP4 file. This has a great advantage: it initializes the libfaad decoder, which the caller would normally do anyway - but now we can go without the AudioSpecificConfig() call. When decoder==NULL (called from mp4_tag_dup()), fall back to a mp4ff_get_track_type()==1 check, like other audio players do.
* mp4ff: moved code to mp4_faad_new()Max Kellermann2009-02-181-53/+65
| | | | | | Moved the libfaad decoder initialization to mp4_faad_new(), and also fill the audio_format struct there. This eliminates a little bit of complexity in mp4_decode().
* mp4ff: call decoder_initialized() after libfaad initializationMax Kellermann2009-02-181-23/+32
| | | | | Don't wait for the first frame to be decoded. We already have the sample rate and the channel count from faacDecInit2().
* mp4ff: merged mp4_load_tag() into mp4_tag_dup()Max Kellermann2009-02-181-7/+1
| | | | | The function mp4_load_tag() is used only once, and mp4_tag_dup() is a one-liner. Merge them.
* faad: variable cleanupMax Kellermann2009-02-171-9/+4
| | | | Make some variables more local, and eliminate superfluous ones.
* faad: added source code commentsMax Kellermann2009-02-171-1/+36
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* faad: faad_decoder_init() returns an audio_formatMax Kellermann2009-02-171-25/+23
| | | | | Instead of returning the sample rate and channel count as separate values, fill an audio_format struct.
* faad: call decoder_initialized() after libfaad initializationMax Kellermann2009-02-171-18/+27
| | | | | Don't wait for the first frame to be decoded. We already have the sample rate and the channel count from faacDecInit().
* faad: removed DECODE_COMMAND_SEEK checkMax Kellermann2009-02-171-2/+0
| | | | | The MPD core will never send a SEEK command to a decoder which has declared to be not seekable.
* faad: use the decoder_buffer libraryMax Kellermann2009-02-171-152/+155
| | | | | Replace this plugin's own buffer library with the new decoder_buffer library.
* faad: check the result of adts_find_frame()Max Kellermann2009-02-171-4/+2
| | | | | Instead of checking if the buffer is empty after adts_find_frame(), check adts_find_frame()'s return value. This is more robust.
* faad: added libfaad wrappersMax Kellermann2009-02-171-42/+61
| | | | | Moved libfaad API quirks to the wrapper functions faad_decoder_init() and faad_decoder_decode().
* faad: functions return duration, no float pointerMax Kellermann2009-02-171-15/+18
| | | | | Instead of writing the song duration into a float pointer, return it from the function.
* faad: removed length==NULL check in faad_song_duration()Max Kellermann2009-02-171-5/+1
| | | | There are no callers which pass NULL here.
* faad: fill buffer in adts_find_frame()Max Kellermann2009-02-171-4/+2
| | | | | | All callers of adts_find_frame() use faad_buffer_fill() before that. Move that faad_buffer_fill() call into adts_find_frame() instead. adts_find_frame() will get its own logic for on-demand filling.
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