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* require GLib 2.16Max Kellermann2012-07-101-1/+0
| | | | | GLib 2.16 was released more than 4 years ago. Let's remove some cruft from the glib_compat.h header, and avoid new cruft to it.
* output/raop: delete the RAOP pluginMax Kellermann2012-05-292-1093/+0
| | | | | | | | | This plugin is horrible code, I mean it. Last year, I tried hard to fix it, but I figured would take less time to do a full rewrite. Given that I don't even have any device that supports RAOP, I can't do that properly. After 16 months, nobody volunteered for fixing it. Hereby, I delete it, because having no RAOP plugin is better than having this mess. Sorry.
* output/alsa: multiply writei() result with out_frame_sizeMax Kellermann2012-04-231-1/+3
| | | | | | .. and not in_frame_size, because this relates to the frame size being sent to ALSA. pcm_export_source_size() will then turn it back into the in_frame_size scale.
* pcm_export: add _frame_size()Max Kellermann2012-04-231-3/+1
| | | | Move code from the ALSA output plugin.
* output/alsa: fix out_frame_size formula, multiply with channelsMax Kellermann2012-04-231-1/+3
| | | | | The hard-coded "3 bytes" was wrong because it ignored the number of channels.
* Merge branch 'v0.16.x'Max Kellermann2012-04-054-59/+55
|\ | | | | | | | | | | Conflicts: src/output/osx_plugin.c src/text_input_stream.c
| * encoder/vorbis: generate end-of-stream packet when playback endsMax Kellermann2012-04-052-2/+2
| | | | | | | | | | Add the encoder_plugin method end(). This is important for the recorder plugin.
| * output/jack: check for connection failure before starting playbackMax Kellermann2012-04-041-0/+3
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| * output/jack: workaround for libjack1 crash bugMax Kellermann2012-04-041-0/+13
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| * output/osx: use the fifo_buffer library instead of rolling ownMax Kellermann2012-03-281-56/+37
| | | | | | | | | | | | | | | | The existing buffer implementation has a major flaw: it is unable to re-fill the buffer until it has been consumed completely, leading to many occasions where the render callback needs to generate silence, just because the play() implementation was unable to append more data. The fifo_buffer library handles that well.
* | audio_format: remove SAMPLE_FORMAT_DSD_OVER_USBMax Kellermann2012-03-272-2/+0
| | | | | | | | | | | | | | DSD-over-USB should not be a MPD core format, because it is not a "natural" format; it is just a temnporary over-the-wire format. This format has been implemented in pcm_export, and does not need to be supported by pcm_convert.
* | output/alsa: support 32 bit DSD-over-USBMax Kellermann2012-03-271-4/+15
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* | pcm_export: implement 24 to 32 bit conversionMax Kellermann2012-03-272-2/+2
| | | | | | | | For 32 bit DSD-over-USB support.
* | output/alsa: use pcm_export for the DSD-over-USB conversionMax Kellermann2012-03-271-11/+10
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* | pcm_export: support DSD to DSD-over-USB conversionMax Kellermann2012-03-272-6/+12
| | | | | | | | Prepare for removing SAMPLE_FORMAT_DSD_OVER_USB.
* | output/alsa: move pcm_export_open() to callerMax Kellermann2012-03-271-11/+16
| | | | | | | | Give the caller more control, prepare for DSD-over-USB improvements.
* | output/alsa: more debug outputMax Kellermann2012-03-271-0/+8
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* | output/alsa: add option to enable DSD over USBMax Kellermann2012-03-221-1/+54
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* | output/alsa: split the frame_size attributeMax Kellermann2012-03-221-6/+18
| | | | | | | | Make it in_frame_size and out_frame_size, to account for packing.
* | audio_format: remove the packed S24 formatMax Kellermann2012-03-224-7/+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.
* | output/alsa: use pcm_export to pack 24 bit samplesMax Kellermann2012-03-221-15/+48
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* | output/oss: use pcm_export to pack 24 bit samplesMax Kellermann2012-03-221-10/+15
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* | pcm_export: add option "pack"Max Kellermann2012-03-222-1/+2
| | | | | | | | | | Converts padded 24 bit samples to packed 24 bit samples. Will replace the packed S24 sample format, which is not used internally.
* | output/oss: remember the real OSS formatMax Kellermann2012-03-221-5/+13
| | | | | | | | | | Improving oss_reopen() by using the very same value that was used initially.
* | output/alsa: simplify setup_format()Max Kellermann2012-03-221-7/+4
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* | output/alsa: don't pass audio_format to _try_format()Max Kellermann2012-03-221-16/+13
| | | | | | | | Let the caller configure the audio_format object.
* | output/alsa: simplify alsa_output_try_format_both()Max Kellermann2012-03-221-45/+18
| | | | | | | | Merge three functions into one and call get_bitformat() only once.
* | output/oss: move code to oss_probe_sample_format()Max Kellermann2012-03-211-34/+59
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* | output/{alsa,oss}: move endian code to new library pcm_exportMax Kellermann2012-03-212-61/+23
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* | audio_format: remove the reverse_endian attributeMax Kellermann2012-03-211-3/+1
| | | | | | | | | | | | Eliminate support for reverse endian samples from the MPD core. This moves a lot of complexity to the plugins that really need it (only ALSA and CDIO currently).
* | output/oss: always receive host byte order samplesMax Kellermann2012-03-211-7/+68
| | | | | | | | Don't use audio_format.reverse_endian.
* | output/alsa: always receive host byte order samplesMax Kellermann2012-03-211-3/+61
| | | | | | | | Don't use audio_format.reverse_endian.
* | output/alsa: merge alsa_data_free() into destructorMax Kellermann2012-03-211-8/+3
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* | Fix the build on OSXRich Healey2012-03-211-0/+1
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* | audio_format: remove the format SAMPLE_FORMAT_DSD_LSBFIRSTMax Kellermann2012-03-212-2/+0
| | | | | | | | | | This format is unused since the DSDIFF decoder plugin now reverses the bit order.
* | audio_format: basic support for DSD-over-USBMax Kellermann2012-03-192-0/+2
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* | use g_strerror() instead of strerror()Max Kellermann2012-03-064-19/+19
| | | | | | | | Make sure we get a UTF-8 encoded string.
* | raop_output: fix raop_session inbalanceKurt Van Dijck2012-03-011-2/+8
| | | | | | | | | | | | | | | | raop_session_free must be called from raop_output_finish, not from raop_output_remove. In raop_output_remove, do close the ntp_server & control port. Signed-off-by: Kurt Van Dijck <kurt.van.dijck@skynet.be>
* | audio_format: add DSD sample formatMax Kellermann2012-03-012-0/+4
| | | | | | | | | | Basic support for Direct Stream Digital. No conversion yet, and no decoder/output plugin support.
* | Merge branch 'v0.16.x'Max Kellermann2012-02-131-5/+1
|\| | | | | | | | | | | | | | | Conflicts: NEWS configure.ac src/decoder/ffmpeg_decoder_plugin.c test/read_tags.c
| * output/winmm: remove pointless NULL checkMax Kellermann2012-02-131-5/+1
| | | | | | | | pcm_buffer_get() cannot ever return NULL.
* | output/osx: fix memory leak after AudioUnitSetProperty() failureMax Kellermann2012-01-041-0/+1
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* | output/osx: implement 32 bit playbackMax Kellermann2011-12-241-2/+6
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* | output/osx: allocate the device in enable()Max Kellermann2011-12-241-102/+116
| | | | | | | | | | Keep the device open as long as the output is enabled, but initialize it only when playback starts.
* | Merge branch 'v0.16.x'Max Kellermann2011-12-241-6/+3
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| * output/osx: clear render buffer when there's not enough dataMax Kellermann2011-12-241-2/+3
| | | | | | | | | | | | When we don't have enough data, generate some silence, hoping the input buffer will fill soon. Reducing the render buffer size is not legal.
| * output/osx: remove sleep call from render callbackMax Kellermann2011-12-241-4/+0
| | | | | | | | | | Blocking inside the render callback is forbidden, and this sleep call didn't make any sense.
* | output/openal: improve synchronizationMax Kellermann2011-12-131-13/+16
| | | | | | | | | | | | | | | | | | This plugin's use of the "Timer" library was wrong; it added the same amount of virtual data in every iteration in _play(), but did not actually play something. This created an artificial, but useless, delay. This patch implements the method _cancel(), and implements hard-coded sleep values. This is only slightly better, but does not attempt to look sane.
* | output/openal: move code to inline functionsMax Kellermann2011-12-131-9/+25
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* | output/openal: use alGetSourcei(AL_BUFFER) to force-unqueue buffersMax Kellermann2011-12-131-14/+4
| | | | | | | | | | | | | | | | The implementation of cancel() did not work well: you cannot use alSourceUnqueueBuffers() to unqueue queued buffers, and our function openal_unqueue_buffers() left the OpenAL library in a rather undefined state; nothing was supposed to be queued, but the "filled" variable was not reset.
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