| Commit message (Collapse) | Author | Files | Lines | ||
|---|---|---|---|---|---|
| 2013-07-30 | tag: convert to C++ | Max Kellermann | 1 | -6/+6 | |
| 2013-07-30 | encoder_api: convert to C++ | Max Kellermann | 14 | -119/+94 | |
| 2013-07-30 | encoder/lame,twolame: convert to C++ | Max Kellermann | 4 | -104/+154 | |
| 2013-07-30 | encoder/wave: convert to C++ | Max Kellermann | 2 | -26/+58 | |
| 2013-07-30 | encoder/null: convert to C++ | Max Kellermann | 2 | -24/+54 | |
| 2013-07-30 | pcm_buffer: convert to C++ | Max Kellermann | 1 | -7/+5 | |
| 2013-07-29 | encoder/flac: convert to C++ | Max Kellermann | 2 | -22/+56 | |
| 2013-06-24 | fix overwriting bitrate with signal type | Matthias Larisch | 1 | -3/+3 | |
| I recently opened a bug: http://bugs.musicpd.org/view.php?id=3787 The main problem is that opus encoder config for signal overwrote bitrate setting. | |||||
| 2013-04-09 | pcm_*: move to src/pcm/ | Max Kellermann | 1 | -1/+1 | |
| 2013-01-30 | filter/chain, encoder: GLib include cleanup | Max Kellermann | 4 | -28/+27 | |
| 2013-01-30 | {encoder,output}_api.h: allow compiling as C++ | Max Kellermann | 2 | -8/+0 | |
| 2013-01-15 | encoder/{Vorbis,Opus}: use new/delete | Max Kellermann | 2 | -10/+14 | |
| 2013-01-15 | fifo_buffer: move to util/ | Max Kellermann | 3 | -6/+6 | |
| 2012-10-02 | {decoder,encoder}/flac: drop support for libFLAC 1.1 | Max Kellermann | 1 | -26/+5 | |
| 2012-10-02 | encoder/opus: implement lookahead | Max Kellermann | 1 | -2/+43 | |
| The "opusinfo" program complained about preskip value that was too small. This commit uses OPUS_GET_LOOKAHEAD to obtain the number of frames that shall be silence at the beginning. | |||||
| 2012-10-02 | encoder/opus: initialize the "granulepos" packet attribute | Max Kellermann | 1 | -1/+5 | |
| 2012-10-02 | encoder/{vorbis,opus}: merge code to new class OggStream | Max Kellermann | 3 | -89/+154 | |
| 2012-10-02 | encoder/vorbis: accept floating point input samples | Max Kellermann | 1 | -8/+9 | |
| Improves quality by not squeezing 32 bit samples down to 16 bit, and then back to 32 bit floating point. Reduces CPU usage by skipping a conversion step. | |||||
| 2012-10-02 | encoder/opus: call ogg_stream_flush() only in the last iteration | Max Kellermann | 1 | -4/+4 | |
| If there are multiple pages, the last partial page must be flushed. | |||||
| 2012-10-02 | encoder/opus: new encoder plugin for the Opus codec | Max Kellermann | 2 | -0/+442 | |
| 2012-10-02 | encoder/vorbis: use C++ compiler | Max Kellermann | 2 | -23/+51 | |
| 2012-10-02 | encoder/vorbis: make variables more local | Max Kellermann | 1 | -20/+12 | |
| 2012-04-05 | encoder/vorbis: generate end-of-stream packet when playback ends | Max Kellermann | 3 | -0/+3 | |
| Add the encoder_plugin method end(). This is important for the recorder plugin. | |||||
| 2012-04-04 | encoder/vorbis: generate end-of-stream packet before tag | Max Kellermann | 1 | -2/+0 | |
| Don't reset the ogg_stream_state object, because this discards the end-of-stream packet that was just added. | |||||
| 2012-03-22 | audio_format: remove the packed S24 format | Max Kellermann | 1 | -5/+0 | |
| For simplicity, the MPD core should not have to deal with packing. It is rarely used, and those plugins that need it should use the pcm_export library instead. | |||||
| 2011-11-28 | encoder/wave: support packed 24 bit samples | Max Kellermann | 1 | -0/+5 | |
| Convert to padded 24 bit samples, instead of falling back to 16 bit. | |||||
| 2011-11-28 | encoder/null: use fifo_buffer instead of pcm_buffer | Max Kellermann | 1 | -19/+15 | |
| This fixes a buffer corruption bug; pcm_buffer is not designed to be a persistent buffers, and will discard anything between two consecutive calls. | |||||
| 2011-11-28 | encoder/wave: use fifo_buffer instead of pcm_buffer | Max Kellermann | 1 | -19/+27 | |
| This fixes a buffer corruption bug; pcm_buffer is not designed to be a persistent buffers, and will discard anything between two consecutive calls. | |||||
| 2011-11-28 | encoder/flac: use fifo_buffer instead of pcm_buffer | Max Kellermann | 1 | -16/+20 | |
| This fixes a buffer corruption bug; pcm_buffer is not designed to be a persistent buffers, and will discard anything between two consecutive calls. | |||||
| 2011-07-20 | encoder_plugin: add method pre_tag() | Max Kellermann | 1 | -0/+11 | |
| In the "vorbis" plugin, this is a copy of the old flush() method, while flush() gets a lot of code remove, it just sets the "flush" flag and nothing else. It doesn't start a new stream now, which should fix a few problems in some players. | |||||
| 2011-03-31 | fix common misspellings | Jonathan Neuschäfer | 1 | -1/+1 | |
| These fixes were mostly generated with `codespell' [0] and manually reviewed. [0] http://git.profusion.mobi/cgit.cgi/lucas/codespell/ | |||||
| 2011-03-16 | encoder/vorbis: reset the Ogg stream after flush | Max Kellermann | 1 | -0/+2 | |
| Without the ogg_stream_reset() call, the "e_o_s" flag never gets reset, and libogg writes EOS packets over and over. | |||||
| 2011-03-16 | encoder/vorbis: reset the Ogg stream after flush | Max Kellermann | 1 | -0/+2 | |
| Without the ogg_stream_reset() call, the "e_o_s" flag never gets reset, and libogg writes EOS packets over and over. | |||||
| 2011-02-09 | general: whitespace cleanup | Thomas Jansen | 2 | -5/+5 | |
| Remove trailing whitespace found by this command: find -name '*.[ch]' | xargs grep "[[:space:]]$" | |||||
| 2011-01-29 | copyright year 2011 | Max Kellermann | 6 | -6/+6 | |
| 2011-01-07 | encoder/lame: explicitly configure the output sample rate | Max Kellermann | 1 | -0/+7 | |
| When you don't explicitly set an output sample rate, liblame tries to guess an output sample rate from the input sample rate. You would think that this "guessing" consists of just setting both equal, but that is not the case. For 44.1kHz at 96kbit/s, liblame chooses 32kHz. This patch explicitly configures the output sample rate, to stop the bad guessing. | |||||
| 2010-09-25 | eliminate g_error() usage | Thomas Jansen | 1 | -1/+2 | |
| Replaced all occurrences of g_error() with MPD_ERROR() located in a new header file 'mpd_error.h'. This macro uses g_critical() to print the error message and then exits gracefully in contrast to g_error() which would internally call abort() to produce a core dump. The macro name is distinctive and allows to find all places with dubious error handling. The long-term goal is to get rid of MPD_ERROR() altogether. To facilitate the eventual removal of this macro it was added in a new header file rather than to an existing header file. This fixes #2995 and #3007. | |||||
| 2009-12-31 | Update copyright notices. | Avuton Olrich | 6 | -6/+6 | |
| 2009-12-15 | encoder/flac: fix write callback prototype for libFLAC 1.1.2 | Max Kellermann | 1 | -1/+7 | |
| 2009-12-03 | encoders: remove unnessesary pointers to const strings | Viliam Mateicka | 5 | -15/+5 | |
| 2009-12-03 | encoders: implement new get_mime_types method | Viliam Mateicka | 5 | -0/+45 | |
| 2009-12-03 | null_encoder: use pcm_buffer | Viliam Mateicka | 1 | -10/+18 | |
| 2009-12-03 | flac_encoder: add support for libFLAC < 1.1.3 | Viliam Mateicka | 1 | -18/+47 | |
| 2009-12-02 | audio_format: changed "bits" to "enum sample_format" | Max Kellermann | 5 | -18/+61 | |
| 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. | |||||
| 2009-11-17 | encoder: let wave encoder to use pcm_buffer, pcm conversion code cleanup | Viliam Mateicka | 1 | -29/+27 | |
| 2009-11-17 | encoder: introducing flac encoder plugin | Viliam Mateicka | 1 | -0/+296 | |
| 2009-11-12 | include config.h in all sources | Max Kellermann | 5 | -0/+5 | |
| 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. | |||||
| 2009-11-10 | wave_encoder: new encoder for streaming PCM wave files. | Viliam Mateicka | 1 | -0/+241 | |
| When using wave encoder with httpd audio output mpd can input this stream via http and audiofile decoder. This for example opens simple way to configure lossless audio streaming port(like jack or pulseaudio does but without overhead). Another possibility can be using it for gathering raw data for visualization plugins (If sync issue will be resolved) | |||||
| 2009-11-08 | encoder/null: removed empty close() method | Max Kellermann | 1 | -6/+0 | |
| That's an optional method. | |||||
| 2009-11-08 | encoder/null: removed unused audio_format attribute | Max Kellermann | 1 | -5/+2 | |
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index : mpd | |
| clone of git://git.musicpd.org/master/mpd.git | Alexander Sulfrian |
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