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* output/{recorder,shout}: call encoder_read() in a loopMax Kellermann2012-10-022-16/+21
| | | | 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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* output/recorder, test/*: invoke encoder_read() after _open()Max Kellermann2012-10-021-0/+7
| | | | | 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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* output/jack: implement method delay()Max Kellermann2012-08-141-4/+11
| | | | Eliminate the g_usleep() call.
* output/pulse: implement method delay()Max Kellermann2012-08-141-7/+21
| | | | Reduce command latency while paused.
* output/pulse: simplify _wait_stream()Max Kellermann2012-08-141-55/+16
| | | | One large loop and only one pa_stream_get_state() call.
* output/httpd: move delay from _pause() to _delay()Max Kellermann2012-08-141-1/+5
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* output/httpd: fix throttling bug after resuming playbackMax Kellermann2012-08-141-0/+8
| | | | | | Reset the timer when paused and no client is connected. This fixes Mantis ticket 0003527.
* output/httpd: move code to _has_clients()Max Kellermann2012-08-141-11/+27
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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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