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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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Some of these were found with http://www.samba.org/junkcode/#findstatic. Changes of note:
* The old MDCT has been removed.
* Makefile.test files that create test programs for libatrac, libcook, and libffmpegFLAC have been removed, as they don't work. My project will have a replacement that works with all codecs.
* I've tried not to remove anything useful. CLIP_TO_15 was removed from libtremor because there's another copy (also commented) in codeclib.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29945 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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support. As a result the memory consumption was drastically reduced. This allows to play several files with long duration -- especially on low memory targets. The change builds a lookup table from m4a's sample_to_chunk[] and chunk_offset[] and completely removes the allocation of the large tables chunk_offset[] and sample_byte_size[]. To be able to remove reading and allocating sample_byte_offset[] the aac and alac decoder now buffer a fixed amount of bytes for each frame. The generated lookup table is used for seek/resume and skipping bytes in empty chunks (aac decoder only). The precision for seek/resume is somewhat lower but still equals 0.5 sec for the worst case. Fixes FS#8923.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29745 a1c6a512-1295-4272-9138-f99709370657
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decoder does not need to use get_sample_info() to gather frame size or the number of consumed bytes.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29724 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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files only being playable on direct selection, but not if switched to via playback engine or skip.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@27939 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@27566 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@26038 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25169 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@12830 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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Also, use the 'standard' wait-for-metadata loop in the ALAC decoder.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11580 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@11188 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@9645 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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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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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8124 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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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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DSP_SET_SAMPLE_DEPTH function expects needs clarifying or changing - it seems to expect one less than the number of bits in cases where the depth is greater than the native depth (16 bits).
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7692 a1c6a512-1295-4272-9138-f99709370657
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always return up to GUARD_BUFSIZE bytes, even at the buffer wraparound point. This removes the need for the 32KB static input buffer.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7691 a1c6a512-1295-4272-9138-f99709370657
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be used by other codecs
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7682 a1c6a512-1295-4272-9138-f99709370657
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channel containing samples left-shifted to 28-bits (instead of 16-bit interleaved samples). 3) Remove the two 16KB internal predicterror_buffer arrays (we use the output arrays instead) 4) Small internal rearrangement of the code.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7667 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7631 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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improves responsiveness of UI during ALAC decoding
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7557 a1c6a512-1295-4272-9138-f99709370657
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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7547 a1c6a512-1295-4272-9138-f99709370657
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