Age | Commit message (Collapse) | Author |
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now).
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@18449 a1c6a512-1295-4272-9138-f99709370657
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by Nicolas Pitre. Shouldn't affect playback unless it's explicitly enabled, but let me know if it does. Currently has a dedicated setting, but maybe inclusion of the code will inspire someone to integrate this with the pitch screen...
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@18446 a1c6a512-1295-4272-9138-f99709370657
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to Tomer Shalev for the investigation work and the patch.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@17976 a1c6a512-1295-4272-9138-f99709370657
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later. We still need to hunt down snippets used that are not. 1324 modified
files...
http://www.rockbox.org/mail/archive/rockbox-dev-archive-2008-06/0060.shtml
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@17847 a1c6a512-1295-4272-9138-f99709370657
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they are in sync. Made some local functions static.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@17312 a1c6a512-1295-4272-9138-f99709370657
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assembler routines for the gain function there now.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@17040 a1c6a512-1295-4272-9138-f99709370657
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macros which really aren't needed since all performance sensitive target DSP code should be assembler anyway.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@17037 a1c6a512-1295-4272-9138-f99709370657
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threads. Do at least one per buffer full as well (or things can stick a bit on initial input triggers).
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@16721 a1c6a512-1295-4272-9138-f99709370657
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channels_process_sound_chan_mono(), channels_process_sound_chan_karaoke(), sample_output_mono() and sample_output_stereo(). By measurement the speed up is ~75% for the first three functions and ~40% for sample_output_stereo(). Additionally avoid calling yield() to often in dsp.c -- it is now limited to once per tick.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@16717 a1c6a512-1295-4272-9138-f99709370657
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reasons, change that and kill some identical formatting functions while we're at it. NOTE TO CROSSFEED USERS: this will completely garble your crossfeed settings, write them down if you want to keep them.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15895 a1c6a512-1295-4272-9138-f99709370657
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shaping error being calculated after clipping instead of before.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15873 a1c6a512-1295-4272-9138-f99709370657
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codec swapping and build speex voice decoding directly into the core binary.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15668 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15370 a1c6a512-1295-4272-9138-f99709370657
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bug would only affect people using WMA. Also, add a small comment in an unrelated place.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15369 a1c6a512-1295-4272-9138-f99709370657
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for you golden-eared people.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@14514 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@13369 a1c6a512-1295-4272-9138-f99709370657
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Hunt down and remove the remaining ones in the recording system as well.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12955 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12929 a1c6a512-1295-4272-9138-f99709370657
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optimizing as seems reasonably possible by jumping through some hoops to avoid stalls. Further boost reduction will just be fractional points if taken to extremes-- not worth it. Wrap up the ASM for awhile.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12905 a1c6a512-1295-4272-9138-f99709370657
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SWCODEC stereo width and channel configuration instead of the old more spread out #ifdef based approach. Rename the DSP functions involved for more consistent naming.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12677 a1c6a512-1295-4272-9138-f99709370657
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crossfeed filter with it. Crossfeed _should_ still sound the same, so please do tell if it doesn't.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12674 a1c6a512-1295-4272-9138-f99709370657
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not limited to FLAC actually. Crossfade doesn't like being hand the larger chunks that the dsp was now able to process when not resampling. The real fix is making it not matter.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12521 a1c6a512-1295-4272-9138-f99709370657
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duplicated into right channel in pcm buffer.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12509 a1c6a512-1295-4272-9138-f99709370657
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negative gain.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12508 a1c6a512-1295-4272-9138-f99709370657
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SWITCHPARAM cruft as it's no longer needed.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12507 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12506 a1c6a512-1295-4272-9138-f99709370657
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differing parameters between ARM and Coldfire halfway clean. Hopefully those differences can be reconciled soon. A tiny bit of C optimizing for karaoke channel mode.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12505 a1c6a512-1295-4272-9138-f99709370657
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entered. Make all codecs set the replay gain or else formats that do not have replaygain will not set the gain back to default if a file with gain applied proceeded them.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12498 a1c6a512-1295-4272-9138-f99709370657
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functionality in hardware (currently only X5). They can also be used on any other SWCODEC target by adding #define HAVE_SW_TONE_CONTROLS in the relevant config-*.h file. Also remove some now unneeded zero checks when using get_replaygain_int(). Comments on sound quality are welcome as some parameters can still be fine-tuned.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12489 a1c6a512-1295-4272-9138-f99709370657
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some usless stuff. Some assembly routines for Coldfire with speed in mind over size for the outputs but the channel modes remain compact. Miscellaneous coldfire asm updates to accomodate the changes. Codec API structure version has to increase so do a full update.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12472 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12441 a1c6a512-1295-4272-9138-f99709370657
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Assembly resampling for Coldfire. Word has it ARM will get that soon.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12399 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12361 a1c6a512-1295-4272-9138-f99709370657
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and have the proper includes included etc
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12360 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12336 a1c6a512-1295-4272-9138-f99709370657
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ones. Add config.h to dsp.c so that these macros actually get used, and also do some minor nitpicks to the resampler while I'm at it.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12322 a1c6a512-1295-4272-9138-f99709370657
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and dsp_configure and stop all the silly type casting of intergral types to pointers to set dsp configuration and watermarks. Shouldn't have any effect on already compiled codecs at all. Will fix any important patches in the tracker so they compile.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12259 a1c6a512-1295-4272-9138-f99709370657
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clicked because of size inaccuracies returned by DSP. Fix by simplifying audio system to use per-channel sample count from codec to pcm buffer. 2) Downsampling affected by 1) and was often starting passed the end of the data when not enough was available to generate an output sample. Fix by clamping input range to last sample in buffer and using the last sample value in the buffer. A perfect fix will require a double buffering scheme on the resampler to sufficient data during small data transients on both ends at all times of the down ratio on input and the up ratio on output.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12218 a1c6a512-1295-4272-9138-f99709370657
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unneeded fsqrt function to plugin fixed point library in case it'll be needed. Move all fixed point helper macros to dsp.h. Added FRACMUL_SHL macro to facilitate high-precision shifting of 64 bit multiplies and remove rounding from macsr in main thread to make this work as intended.
Tested quite thorougly, but as always, be careful with your ears.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12203 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11737 a1c6a512-1295-4272-9138-f99709370657
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the song. Forgot to reinit the resampler delta when adding DSP_FLUSH on my last change. Bah.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11621 a1c6a512-1295-4272-9138-f99709370657
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switching the DSP frequency and not resetting the resampler at track boundaries. Will make sure DSP is correctly flushed at dicontinuities but don't hear any problems currently.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11600 a1c6a512-1295-4272-9138-f99709370657
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other than 44.1kHz to not procede. It created a dependency of setting sample depth before frequency and aiff.codec does it the other way.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11595 a1c6a512-1295-4272-9138-f99709370657
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if album gain requested but not available). Share the mode decision logic with playback code and simplify the %rg tag handling.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11532 a1c6a512-1295-4272-9138-f99709370657
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already in Rockbox, and make it a user option instead of a codec-controlled option. The majority of people probably will not even hear any difference with this enabled, but feedback is welcome. Save your settings!
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11368 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9711 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9705 a1c6a512-1295-4272-9138-f99709370657
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new options is appreciated. Thanks to Dan Everton for the settings/GUI
code.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9609 a1c6a512-1295-4272-9138-f99709370657
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mp3 metadata parser.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9438 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9332 a1c6a512-1295-4272-9138-f99709370657
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