Age | Commit message (Collapse) | Author |
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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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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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implementation doesn't do what it claims any way
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15478 a1c6a512-1295-4272-9138-f99709370657
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This is _not_ a setting. This is a guessing tool used by either playback or buffering to serve its clients better. Use the REBUFFER_GUESS size in resume to help obviate pondlife's bug. This will also need to be used when FS8092 gets fixed correctly with a complete rebuffer for backward movements. It may also belong in buffering not playback, haven't decided for sure
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15475 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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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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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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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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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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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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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@10807 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9758 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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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9332 a1c6a512-1295-4272-9138-f99709370657
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menu code.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9298 a1c6a512-1295-4272-9138-f99709370657
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platforms.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9173 a1c6a512-1295-4272-9138-f99709370657
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band being modified.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8718 a1c6a512-1295-4272-9138-f99709370657
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* Linked list instead of static array buffer pointers
* Variable sized chunks
* Improved mix handling
* Reduction in duplicated code
* Reduced IRAM usage w/o sacrificing performance
* Converted to almost entirely unsigned math
* Add pause function to reduce pcm_* exposure to playback.
This WILL break playback on the iPod until linuxstb makes a followup commit.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8612 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8606 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8569 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8568 a1c6a512-1295-4272-9138-f99709370657
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to do both channels in one pass.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8099 a1c6a512-1295-4272-9138-f99709370657
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Removed CODEC_SET_FILEBUF_LIMIT setting; now playback.c determines how
to buffer the files.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7970 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7884 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7529 a1c6a512-1295-4272-9138-f99709370657
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Note that there is a small delay from leaving a setting until the change
can be heard (due to audio data buffering).
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7234 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7174 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7036 a1c6a512-1295-4272-9138-f99709370657
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codecs (currently works corrently only with mp3's, somebody should fix
that).
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6877 a1c6a512-1295-4272-9138-f99709370657
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