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| * decoder/ffmpeg: support planar audioMax Kellermann2012-10-051-0/+6
| | | | | | | | Implements Mantis feature request 3582.
| * decoder/ffmpeg: fix playback of planar PCM dataMax Kellermann2012-10-051-8/+13
| | | | | | | | | | Interleaving was completely wrong. This code was never used, so it didn't have an effect.
| * decoder/ffmpeg: move code to copy_interleave_frame2()Max Kellermann2012-10-051-5/+14
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| * decoder/ffmpeg: ignore negative time stampsMax Kellermann2012-10-051-1/+1
| | | | | | | | Works around assertion failure in decoder_timestamp().
| * decoder/ffmpeg: show unsupported sample format nameMax Kellermann2012-10-051-2/+13
| | | | | | | | Use av_get_sample_fmt_string() to obtain a human-readable string.
| * decoder/ffmpeg: pass AVSampleFormat to ffmpeg_sample_format()Max Kellermann2012-10-051-4/+9
| | | | | | | | API simplification.
| * decoder/ffmpeg: remove duplicate sample format error messageMax Kellermann2012-10-051-1/+6
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* | decoder/flac: support FLAC files inside archivesMax Kellermann2012-10-044-2/+215
| | | | | | | | | | | | Implement the "scan_stream" method that can read tags from any input_stream object. This requires a FLAC__IOCallbacks implementation based on the input_stream API.
* | decoder/flac: move flac_scan_file2() to main plugin fileMax Kellermann2012-10-043-20/+9
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* | input/file: use errno_quark()Max Kellermann2012-10-041-11/+6
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* | PlaylistFile, client_file, tag_id3: don't use g_file_error_quark()Max Kellermann2012-10-045-8/+57
| | | | | | | | | | | | | | | | | | g_file_error_quark() is meant to be used with the GFileError enum which does not correspond with errno, but must be converted with g_file_error_from_errno(). At the same time, this removes g_strerror() use for g_file_error_quark().
* | decoder/flac: move callbacks to class FLACInputMax Kellermann2012-10-025-110/+242
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* | decoder/flac: add ctor/dtor to struct flac_dataMax Kellermann2012-10-023-34/+17
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* | decoder/flac: add method FLACMetadataChain::Scan()Max Kellermann2012-10-023-12/+12
| | | | | | | | Merge common code.
* | decoder/flac: don't use FLAC__Metadata_SimpleIteratorMax Kellermann2012-10-021-32/+8
| | | | | | | | Use the new FLACMetadataIterator C++ class, which is more powerful.
* | decoder/flac: add C++ libFLAC wrappersMax Kellermann2012-10-022-14/+71
| | | | | | | | | | Not using libFLAC++ because this library adds a lot of overhead due to virtual method calls. This new class library is zero-overhead.
* | decoder/flac: eliminate the obsolete "track number" codeMax Kellermann2012-10-024-41/+18
| | | | | | | | This has been deprecated by the "embcue" playlist plugin.
* | Merge branch 'v0.17.x'Max Kellermann2012-10-021-3/+7
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| * OSX: Set mDataByteSize correctly on AudioBuffers during render.Gregory Smith2012-10-021-3/+7
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* | decoder/flac: remove unused function flac_tag_load()Max Kellermann2012-10-022-17/+0
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* | decoder/flac: use C++ compilerMax Kellermann2012-10-029-62/+110
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* | {decoder,encoder}/flac: drop support for libFLAC 1.1Max Kellermann2012-10-023-194/+10
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* | encoder/opus: implement lookaheadMax Kellermann2012-10-021-2/+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.
* | encoder/opus: initialize the "granulepos" packet attributeMax Kellermann2012-10-021-1/+5
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* | encoder/{vorbis,opus}: merge code to new class OggStreamMax Kellermann2012-10-023-89/+154
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* | encoder/vorbis: accept floating point input samplesMax Kellermann2012-10-021-8/+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.
* | pcm_channels: support floating point samplesMax Kellermann2012-10-023-62/+105
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* | encoder/opus: call ogg_stream_flush() only in the last iterationMax Kellermann2012-10-021-4/+4
| | | | | | | | If there are multiple pages, the last partial page must be flushed.
* | encoder/opus: new encoder plugin for the Opus codecMax Kellermann2012-10-023-0/+446
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* | output/shout: move code to my_shout_configure()Max Kellermann2012-10-021-31/+40
| | | | | | | | Eliminate the evil goto.
* | encoder/vorbis: use C++ compilerMax Kellermann2012-10-023-25/+53
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* | Merge branch 'v0.17.x'Max Kellermann2012-10-025-143/+119
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| * output/{recorder,shout}: call encoder_read() in a loopMax Kellermann2012-10-023-16/+23
| | | | | | | | This is necessary for Ogg packets that span more than one page.
| * output/recorder: move code to _write_to_file()Max Kellermann2012-10-021-19/+31
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| * output/recorder: fix write() error checkMax Kellermann2012-10-021-3/+3
| | | | | | | | We can only check for negative values if the variable is signed.
| * output/recorder: make variables more localMax Kellermann2012-10-021-16/+12
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| * output/httpd: make variables more localMax Kellermann2012-10-021-31/+16
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| * encoder/vorbis: make variables more localMax Kellermann2012-10-021-20/+12
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| * output/recorder, test/*: invoke encoder_read() after _open()Max Kellermann2012-10-022-0/+11
| | | | | | | | | | Make sure the file header gets written at the beginning, before _write() gets called.
| * output/shout: eliminate struct shout_bufferMax Kellermann2012-10-021-7/+3
| | | | | | | | Move the raw buffer to struct shout_data.
| * output/shout: remove shout_buffer.lenMax Kellermann2012-10-021-9/+4
| | | | | | | | Make it a local variable instead.
| * output/shout: fix memory leak in error handlerMax Kellermann2012-10-021-2/+7
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| * output/shout: make variables more localMax Kellermann2012-10-021-49/+26
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* | main: use C++ compilerMax Kellermann2012-09-2819-35/+36
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* | PlaylistFile: use std::list instead of GPtrArrayMax Kellermann2012-09-286-159/+114
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* | stored_playlist, playlist_save: use C++ compilerMax Kellermann2012-09-2812-32/+49
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* | stored_playlist: make variables more localMax Kellermann2012-09-281-46/+25
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* | output: new option "tags" may be used to disable sending tagsMax Kellermann2012-09-253-1/+9
| | | | | | | | Implements Mantis ticket 0003340.
* | decoder/adplug: new decoder pluginMax Kellermann2012-09-253-0/+177
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* | decoder/vorbis: skip 16 bit quantisation, provide float samplesSimon Hosie2012-09-251-0/+44
| | | | | | | | | | | | | | | | | | | | Internally the vorbis (non-Tremor) decoder is working in floating point, and it's not really necessary to cut the output back to 16-bit if the soundcard or OS supports higher resolution. The decoder can be trivially modified to bypass its internal quantisation and produce floating-point output, and a separate quantisation can be used as appropriate to the platform.
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