Commit message (Collapse) | Author | Age | Files | Lines | |
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* | output/alsa: multiply writei() result with out_frame_size | Max Kellermann | 2012-04-23 | 1 | -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 Kellermann | 2012-04-23 | 1 | -3/+1 |
| | | | | Move code from the ALSA output plugin. | ||||
* | output/alsa: fix out_frame_size formula, multiply with channels | Max Kellermann | 2012-04-23 | 1 | -1/+3 |
| | | | | | The hard-coded "3 bytes" was wrong because it ignored the number of channels. | ||||
* | audio_format: remove SAMPLE_FORMAT_DSD_OVER_USB | Max Kellermann | 2012-03-27 | 1 | -1/+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-USB | Max Kellermann | 2012-03-27 | 1 | -4/+15 |
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* | pcm_export: implement 24 to 32 bit conversion | Max Kellermann | 2012-03-27 | 1 | -1/+1 |
| | | | | For 32 bit DSD-over-USB support. | ||||
* | output/alsa: use pcm_export for the DSD-over-USB conversion | Max Kellermann | 2012-03-27 | 1 | -11/+10 |
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* | pcm_export: support DSD to DSD-over-USB conversion | Max Kellermann | 2012-03-27 | 1 | -3/+5 |
| | | | | Prepare for removing SAMPLE_FORMAT_DSD_OVER_USB. | ||||
* | output/alsa: move pcm_export_open() to caller | Max Kellermann | 2012-03-27 | 1 | -11/+16 |
| | | | | Give the caller more control, prepare for DSD-over-USB improvements. | ||||
* | output/alsa: more debug output | Max Kellermann | 2012-03-27 | 1 | -0/+8 |
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* | output/alsa: add option to enable DSD over USB | Max Kellermann | 2012-03-22 | 1 | -1/+54 |
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* | output/alsa: split the frame_size attribute | Max Kellermann | 2012-03-22 | 1 | -6/+18 |
| | | | | Make it in_frame_size and out_frame_size, to account for packing. | ||||
* | audio_format: remove the packed S24 format | Max Kellermann | 2012-03-22 | 1 | -1/+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 samples | Max Kellermann | 2012-03-22 | 1 | -15/+48 |
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* | pcm_export: add option "pack" | Max Kellermann | 2012-03-22 | 1 | -1/+1 |
| | | | | | Converts padded 24 bit samples to packed 24 bit samples. Will replace the packed S24 sample format, which is not used internally. | ||||
* | output/alsa: simplify setup_format() | Max Kellermann | 2012-03-22 | 1 | -7/+4 |
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* | output/alsa: don't pass audio_format to _try_format() | Max Kellermann | 2012-03-22 | 1 | -16/+13 |
| | | | | Let the caller configure the audio_format object. | ||||
* | output/alsa: simplify alsa_output_try_format_both() | Max Kellermann | 2012-03-22 | 1 | -45/+18 |
| | | | | Merge three functions into one and call get_bitformat() only once. | ||||
* | output/{alsa,oss}: move endian code to new library pcm_export | Max Kellermann | 2012-03-21 | 1 | -31/+10 |
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* | audio_format: remove the reverse_endian attribute | Max Kellermann | 2012-03-21 | 1 | -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/alsa: always receive host byte order samples | Max Kellermann | 2012-03-21 | 1 | -3/+61 |
| | | | | Don't use audio_format.reverse_endian. | ||||
* | output/alsa: merge alsa_data_free() into destructor | Max Kellermann | 2012-03-21 | 1 | -8/+3 |
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* | audio_format: remove the format SAMPLE_FORMAT_DSD_LSBFIRST | Max Kellermann | 2012-03-21 | 1 | -1/+0 |
| | | | | | This format is unused since the DSDIFF decoder plugin now reverses the bit order. | ||||
* | audio_format: basic support for DSD-over-USB | Max Kellermann | 2012-03-19 | 1 | -0/+1 |
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* | audio_format: add DSD sample format | Max Kellermann | 2012-03-01 | 1 | -0/+2 |
| | | | | | Basic support for Direct Stream Digital. No conversion yet, and no decoder/output plugin support. | ||||
* | audio_format: basic support for floating point samples | Max Kellermann | 2011-10-20 | 1 | -0/+3 |
| | | | | | Support for conversion from float to 16, 24 and 32 bit integer samples. | ||||
* | Merge branch 'v0.16.x' | Max Kellermann | 2011-10-08 | 1 | -3/+6 |
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* | output_plugin: the plugin allocates the audio_output object | Max Kellermann | 2011-09-19 | 1 | -17/+24 |
| | | | | | | Pass audio_output objects around instead of void pointers. This will give some more control to the plugin, and prepares for non-blocking audio outputs. | ||||
* | output: rename plugin variables | Max Kellermann | 2011-09-17 | 1 | -1/+1 |
| | | | | Consistent naming. | ||||
* | output: per-plugin header | Max Kellermann | 2011-09-17 | 1 | -0/+1 |
| | | | | | Move the "extern" declarations from output_list.c, for more type safety. | ||||
* | output: rename plugin source files | Max Kellermann | 2011-09-17 | 1 | -0/+683 |