Age | Commit message (Collapse) | Author |
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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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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@13249 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12819 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12817 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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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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plugins. Currently, in case of plugins using IRAM bss is cleared twice,
once in the loader, once in PLUGIN_IRAM_INIT. For codecs, bss is cleared only
during codec initialization. Also, removed double variables in codecs
storing a pointer to codec_api.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11606 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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audio file. SSND chunk block_size and offset are 32-bit values, not 16-bit; this bug would probably never even matter since most sound data isn't block aligned.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11596 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@9758 a1c6a512-1295-4272-9138-f99709370657
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variable in most places. Should help with problems people have had with GUI vs. playback sync.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9670 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9230 a1c6a512-1295-4272-9138-f99709370657
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task is removing use of interleaved audio). Fixed broken handling of 8
bit files.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9135 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8525 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8524 a1c6a512-1295-4272-9138-f99709370657
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