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* pcm_volume: change old code to use format instead of bitsViliam Mateicka2009-12-031-1/+1
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* encoders: remove unnessesary pointers to const stringsViliam Mateicka2009-12-035-15/+5
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* httpd: use get_mime_type to determine encoder contentViliam Mateicka2009-12-031-7/+6
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* encoders: implement new get_mime_types methodViliam Mateicka2009-12-035-0/+45
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* encoder: add get_mime_type() method to determine content type by httpd ↵Viliam Mateicka2009-12-031-0/+17
| | | | output plugin
* pcm_mix: change old code to use format instead of bitsViliam Mateicka2009-12-031-1/+1
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* null_encoder: use pcm_bufferViliam Mateicka2009-12-031-10/+18
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* flac_encoder: add support for libFLAC < 1.1.3Viliam Mateicka2009-12-031-18/+47
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* compress: add config.hJeffrey Middleton2009-12-021-0/+19
| | | | | This includes some default values of #defined constants used in the code; it won't compile without it.
* audio_format: changed "bits" to "enum sample_format"Max Kellermann2009-12-0245-210/+506
| | | | | | This patch prepares support for floating point samples (and probably other formats). It changes the meaning of the "bits" attribute from a bit count to a symbolic value.
* compress: upgraded to AudioCompress 2.0J. Shagam2009-12-026-465/+233
| | | | | | | | Copied sources from http://beesbuzz.biz/code/audiocompress/AudioCompress-2.0.tar.gz [mk: created this patch under fluffy's name and fixed some gcc signed/unsigned comparison warnings]
* decoder/mpcdec: set 24 bit sample formatMax Kellermann2009-11-251-1/+1
| | | | | This fixes a regression due to a typo caused by "decoder: use audio_format_init_checked()".
* pcm_mix: implemented 32 bit supportMax Kellermann2009-11-191-0/+23
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* pcm_volume: implemented 32 bit supportMax Kellermann2009-11-192-0/+42
| | | | Support 32 bit samples with software mixer.
* Merged release 0.15.6 from branch 'v0.15.x'Max Kellermann2009-11-193-9/+28
|\ | | | | | | | | | | | | Conflicts: NEWS configure.ac
| * decoder/flac: fixed compiler warningMax Kellermann2009-11-191-3/+1
| | | | | | | | | | | | Removed the "vtrack" local variable (which triggered a gcc warning because it was after the newly introduced NULL check), and run strtol() on the original parameter.
| * decoder/flac: fixed NULL pointer dereference in CUE codeMax Kellermann2009-11-181-0/+2
| | | | | | | | The function flac_vtrack_tnum() was missing a strrchr()==NULL check.
| * id3: allow 4 MB RIFF/AIFF tagsMax Kellermann2009-11-151-1/+1
| | | | | | | | | | | | | | Allow RIFF/AIFF ID3 tags up to 4 MB (old limit was 256 kB). This might still be too small for some users, and when somebody complains, we might do something more clever (like streaming the data into libid3tag?).
| * decoder/ffmpeg: align the output bufferMax Kellermann2009-11-151-5/+24
| | | | | | | | | | | | On some platforms, libavcodec wants the output buffer aligned to 16 bytes (because it uses SSE/Altivec internally). It will segfault when you don't obey this rule.
* | cmdline: print out list of encoders in --version infoViliam Mateicka2009-11-173-0/+26
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* | encoder: let wave encoder to use pcm_buffer, pcm conversion code cleanupViliam Mateicka2009-11-171-29/+27
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* | encoder: introducing flac encoder pluginViliam Mateicka2009-11-172-0/+300
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* | output/openal: use audio_format_to_string()Max Kellermann2009-11-151-3/+3
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* | crossfade: use audio_format_valid() in assertionMax Kellermann2009-11-151-3/+1
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* | decoder/audio: eliminate the "bits" variableMax Kellermann2009-11-141-4/+1
| | | | | | | | | | Pass the audiofile_setup_sample_format() result to audio_format_init_checked().
* | decoder/audiofile: moved code to audiofile_setup_sample_format()Max Kellermann2009-11-141-10/+20
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* | decoder/modplug: count frame positionMax Kellermann2009-11-141-13/+11
| | | | | | | | | | Don't maintain the current time stamp in a floating point variable, because this is subject to rounding errors.
* | decoder/modplug: floating point division for song durationMax Kellermann2009-11-141-3/+1
| | | | | | | | More exact total time.
* | decoder/modplug: check ModPlug_Read() < 0Max Kellermann2009-11-141-3/+1
| | | | | | | | | | Negative return values are not documented here, but since the function prototype is signed, let's be sure.
* | decoder/mikmod: count frame positionMax Kellermann2009-11-141-8/+6
| | | | | | | | | | Don't maintain the current time stamp in a floating point variable, because this is subject to rounding errors.
* | decoder/mikmod: sample rate is configurableMax Kellermann2009-11-141-3/+12
| | | | | | | | The new option "sample_rate" sets the sample rate for libmikmod.
* | decoder/mikmod: set drv_name and drv_version from PACKAGE/VERSIONMax Kellermann2009-11-141-3/+3
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* | decoder/mikmod: no CamelCaseMax Kellermann2009-11-141-28/+34
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* | decoder/mikmod: removed the struct mod_DataMax Kellermann2009-11-141-14/+9
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* | decoder/mikmod: merged open()/close() into decode()Max Kellermann2009-11-141-31/+12
| | | | | | | | These functions are trivial, we don't need them separate.
* | decoder/mikmod: static mod_Data objectMax Kellermann2009-11-141-11/+9
| | | | | | | | Don't allocate this object, put it on the stack.
* | audio_format: added function audio_format_to_string()Max Kellermann2009-11-145-22/+80
| | | | | | | | | | Unified function for converting an audio_format object to a string, for log messages and for the "status" command.
* | decoder: use audio_format_init_checked()Max Kellermann2009-11-1414-85/+122
| | | | | | | | | | | | Let the audio_check library verify the audio format in all (relevant, i.e. non-hardcoded) plugins.
* | audio_check: checker functions for audio_format attributesMax Kellermann2009-11-143-12/+132
| | | | | | | | | | These functions are a wrapper for audio_valid_X(). On error, they return a GError object.
* | decoder/sidplay: correctly calculate floating point timeMax Kellermann2009-11-141-8/+11
| | | | | | | | | | Internally, use only the integer time. When needed, convert it to a floating point seconds value.
* | player_thread: corrected two assertions on "queued"Max Kellermann2009-11-141-2/+2
| | | | | | | | At this point, the function may be called from the SEEK handler.
* | player_thread: initialize chunk->times in silence generatorMax Kellermann2009-11-122-1/+5
| | | | | | | | | | | | | | | | | | | | When waiting for the decoder to provide more data, the player thread generates silence chunks if needed. However, it forgot to initialize the chunk.times attribute, which had now an undefined value. This patch sets it to -1.0, meaning "value is undefined". Add a ">= 0.0" check to audio_output_all_check(). This fixes spurious relative seeking errors, because sometimes, the "elapsed" value falls back to 0.0.
* | player_control: hold lock while reading statusMax Kellermann2009-11-121-1/+4
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* | added .#* to .gitignoreMax Kellermann2009-11-121-1/+0
| | | | | | | | Temporary editor files.
* | include config.h in all sourcesMax Kellermann2009-11-12199-63/+268
| | | | | | | | | | | | After we've been hit by Large File Support problems several times in the past week (which only occur on 32 bit platforms, which I don't have), this is yet another attempt to fix the issue.
* | decoder/vorbis: fixed gcc "signed" warningMax Kellermann2009-11-121-2/+2
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* | directory: include config.hMax Kellermann2009-11-111-0/+1
| | | | | | | | | | *sigh* another Large File breakage. ino_t/dev_t this time. We need to include config.h in directory.h to get this straight.
* | decoder/wavpack: allow more than 2 channelsMax Kellermann2009-11-111-3/+3
| | | | | | | | | | Remove the OPEN_2CH_MAX option. MPD's support for surround sound is still clunky, but we're working on it.
* | decoder/wavpack: activate 32 bit supportMax Kellermann2009-11-111-13/+7
| | | | | | | | | | | | | | | | | | | | MPD has been supporting 32 bit samples since version 0.15. This patch changes one check, and removes the 32->24 conversion code. Note that WavPack floating point samples have 32 bits, and MPD doesn't have a special check for floating point - therefore, this WavPack plugin still returns 24 bit integer samples as before (until we have float support in the MPD core).
* | decoder/vorbis: initialize before entering the loopMax Kellermann2009-11-111-21/+37
| | | | | | | | | | | | | | Call decoder_initialize() before entering the loop. We don't need to call ov_read() before ov_info(). When the stream number changes, check if the audio format is still the same.
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