| Commit message (Collapse) | Author | Age |
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Change-Id: Id7f4717d51ed02d67cb9f9cb3c0ada4a81843f97
Reviewed-on: http://gerrit.rockbox.org/137
Reviewed-by: Nils Wallménius <nils@rockbox.org>
Tested-by: Nils Wallménius <nils@rockbox.org>
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buffer chunks.
* Samples and position indication is closely associated with audio data
instead of compensating by a latency constant. Alleviates problems with
using the elapsed as a track indicator where it could be off by several
steps.
* Timing is accurate throughout track even if resampling for pitch shift,
whereas before it updated during transition latency at the normal 1:1 rate.
* Simpler PCM buffer with a constant chunk size, no linked lists.
In converting crossfade, a minor change was made to not change the WPS until
the fade-in of the incoming track, whereas before it would change upon the
start of the fade-out of the outgoing track possibly having the WPS change
with far too much lead time.
Codec changes are to set elapsed times *before* writing next PCM frame because
time and position data last set are saved in the next committed PCM chunk.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30366 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29942 a1c6a512-1295-4272-9138-f99709370657
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possible a treatment of codec management, track change and metadata logic as possible while maintaining fairly narrow focus and not rewriting everything all at once. Please see the rockbox-dev mail archive on 2011-04-25 (Playback engine rework) for a more thorough manifest of what was addressed. Plugins and codecs become incompatible.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29785 a1c6a512-1295-4272-9138-f99709370657
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manner. Sort of a halfway patch; best would be to give them an internal copy of the current track information which lasts unaltered by playback until a track switch or unload.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29348 a1c6a512-1295-4272-9138-f99709370657
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Not only WAV but also Sun audio, SMAF, vox and WAV64 can resume.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25289 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25131 a1c6a512-1295-4272-9138-f99709370657
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- does not get dwAvgBytesPerSec
wave/aiff/smaf/wave64 codec
- corrects the problem that codec_main() returns invalid value.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25051 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25041 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25040 a1c6a512-1295-4272-9138-f99709370657
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aiif.c/wav.c: deletes #include <inttypes.h>.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24877 a1c6a512-1295-4272-9138-f99709370657
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- wav(RIFF) supports Microsoft ADPCM, Dialogic OKI ADPCM, YAMAHA ADPCM, Adobe SWF ADPCM.
- AIFF supports QuickTime IMA ADPCM.
- DVI ADPCM(IMA ADPCM) reworks.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24782 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24349 a1c6a512-1295-4272-9138-f99709370657
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libpcm.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24348 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24347 a1c6a512-1295-4272-9138-f99709370657
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PCM decoding by introducing libpcm, a library for decoding linear and non-uniform PCM independently of the container format.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24346 a1c6a512-1295-4272-9138-f99709370657
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Use a smaller PCM buffer on targets with 2MB or less ram.
(FS#9703)
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@19743 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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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@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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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@10710 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@10149 a1c6a512-1295-4272-9138-f99709370657
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it was left off. Mark Arigo helped with the Flac fix.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@10051 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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work before either.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9197 a1c6a512-1295-4272-9138-f99709370657
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task is removing use of interleaved audio). Could not test ADPCM due to
difficulty in finding files it could play, so tell me if it is broken.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9136 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8899 a1c6a512-1295-4272-9138-f99709370657
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facilitate 'on exit' functionality
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8374 a1c6a512-1295-4272-9138-f99709370657
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version numbering restarted for the new system. Uses the target ID from configure, so don't change that too often. * Fixed sim_plugin_load_ram() to truncate the tempfile. * Reduced plugin buffer size to 512KB for iriver and iPod.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8362 a1c6a512-1295-4272-9138-f99709370657
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plugins. Saves quite some disk space (and buffer space in case of codec changes during playback).
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8308 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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a seeking time.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7768 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7724 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7723 a1c6a512-1295-4272-9138-f99709370657
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api as "ci" instead of "rb"
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7722 a1c6a512-1295-4272-9138-f99709370657
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API change that wasn't implemented in all the codecs)
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7717 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7628 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7627 a1c6a512-1295-4272-9138-f99709370657
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lib/codeclib.[ch] and lib/xxx2wav.[ch] into just codeclib.[ch]. Deleted much of the unused code in the xxx2wav portion. All codecs should now only include codeclib.h, and whatever codec specific headers are needed.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7626 a1c6a512-1295-4272-9138-f99709370657
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includes.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7606 a1c6a512-1295-4272-9138-f99709370657
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letoh16, letoh32, htole16, htole32, betoh16, betoh32, htobe16 and htobe32
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7584 a1c6a512-1295-4272-9138-f99709370657
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