Age | Commit message (Collapse) | Author |
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Change-Id: I710480a119e0a9b930a13184ed6571fd2dc1bd01
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Change-Id: I81d8f79f47f09528e2f7fa462e579350451c81f1
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Change-Id: Ief36c70b47ec25932651a146051a29224bdd2a0b
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Change-Id: Ie25b8ab90193e6bb580cd7c04f8c0ce281f7a301
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apparently we should be doing this anyway
mark4o> The packets overlap and may reuse state set by other recent packets,
so if you seek to a different position,
resetting the state helps to ensure that the subsequent
packets won't use the state set by the unrelated packets
that were processed before the seek.
remove stack bump WORKAROUND_FS13060
Change-Id: I1c14e23b1721a360b91e3e55202c1557aef0fcc6
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* Properly account for ID3v1 tags
* Play time computation fixes
* Add speech feedback
Patch by Igor Poretsky
Change-Id: Ia6df8fb171882a88527cfa9d3b76b705f09becdd
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Original patch by Stefan Waigand
Updated by Igor Poretsky
Change-Id: Icaf7beb8349ab90e21b94baee627c9412cb2b55d
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Change-Id: I9ec35e276e24ec7b5a2e1199d6264d9f2d5d9fc2
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opus requires the comment header to be a valid file our codec attemps to skip the comment data
in order to reduce the ram allocated originally it caused files with large album art to skip
the beginning of tracks my first attempt at fixing this then caused files with low bitrates
to do the same while fixing files with large album art
This patch should fix both although the initial start might be a bit slower but
this shouldn't cause too much of an issue
Change-Id: Ia1c3561347894cc45f24bb2659436914f8f03b43
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knocks off about .5 second from decode time not a big change but might help a bit on
devices that barely achieve realtime
Change-Id: If6e822b7273613c9449c102ce7dd3543bf975d37
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ogg_sync_reset() causes issues on the partial page boundary
due to the next page (already in buffer) being discarded
instead seek next page boundary past complete page
Change-Id: Ic05f188f489b015693d663f131e09cd92ad37ff7
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(resorting to an explicit cast this time)
Change-Id: Ib5fc7bcd9e573cd32fc4372003c6c5429e339652
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Change-Id: Ib83ce41582b18641badb389c3871e501c8be697f
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It was already mostly there.
Change-Id: I24ff278d9bf18a54be4b67c3075d5ebbe7947f65
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Change-Id: I63eef2c33bf3ea31a135cd6336882b600723f946
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Files with extension "aac" in ADTS or ADIF format are now playable.
Full credit goes to Igor Poretsky.
Change-Id: I413b34e15e5242fea60d3461966ae0984080f530
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* More tolerance to the file format variations.
* AC3 coded files in realaudio format are now playable
Full credit to Igor Poretsky
Change-Id: Id24e94bc00623e89fb8c80403efa92f69ab1e5d7
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(Patch by Igor Poretsky)
Change-Id: I0cdc2021b44f6cd6e76def190d9f04733b922454
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Switch to strrchr to find the extension
Change-Id: Id7ea01ecc2e0553f560308f8b0fc53bd33b023e5
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I'm pretty sure this was a false positive
Change-Id: I0ab375d1d844b3d468c24888c371f588052e1ce9
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In particular, this solves seeking glitches seen in ~6 hr mp3 files.
(Patch taken from Igor Poretsky's tree)
Change-Id: Id65b6726146b6d2d1a223e90b88e401d1b2d597a
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(Caused non-realtime playback on mips..)
Change-Id: I878229e16e31d49156f1ae71ab9c7bb627e4c17b
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On Classic, IRAM1 (second 128Kb of a total of 256KB available IRAM) is
slower than DRAM. Codecs that actually are using regions of IRAM1 runs
faster when DRAM is used, so IRAM1 is disabled and only IRAM0 remains
enabled: 48KB for core and 80KB for codecs/plugins.
The next test_codec results shows how decode time is decreased:
file boosted unboosted
*.ra ~1.5% ~0.5%
*.mpc ~21% ~4.5%
*.ogg ~0.5% ~0%
nero_he*.m4a ~8% ~1%
nero*.m4a ~25% ~7%
wmapro*.wma ~4.5% ~0%
wma*.wma ~25% ~7%
In addition there is a small power save when IRAM1 HW is disabled.
Change-Id: I102adee11458e82037f23076d5d5956e23235de8
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Change-Id: I26b51106c7b1c36a603fba6d521e917d79b5a95b
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Just use long so the compiler potentially doesn't complain about
use of other values not in the enum. It's also the type used
around the system for event ids.
Increase min codec API version.
No functional changes.
Change-Id: If4419b42912f5e4ef673adcdeb69313e503f94cc
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No functional changes.
Change-Id: If372023cb605389a203a635b700eca20685ad49b
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Change-Id: I9fba5b8cbf69d261a7ca1c66e080c08d2fc6d9db
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Most importantly is surround shouldn't operate in mono mode. Have it
watch and (de)activate itself on relevant format changes as it should.
Other changes to better handle buffer allocation failure.
PBE was set internally at 100 by default; SBZ.
Change-Id: I328e0b674e56751a255eae817d7892d685796b06
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Change-Id: I01dd320ac7f4641caaef62363556ca7527dbee19
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metadata.c does not need cuesheet.h, which in apps/ and has nothing to do with
rbcodec library.
Change-Id: I914a49e8c182f5c367d7db3479c2ff39565e5f07
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Change-Id: I124cf59c641c2e161cc147b031d9bef5ef773dfb
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On Windows 64-bit, the size of long is 32-bit, thus any pointer to long cast is
not valid. In any case, one should use intptr_t and ptrdiff_t when casting
to integers. This commit attempts to fix all instances reported by GCC.
When relevant, I replaced code by the macros PTR_ADD, ALIGN_UP from system.h
Change-Id: I2273b0e8465d3c4689824717ed5afa5ed238a2dc
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The mingw linker uses strlen() in some cases, and codeclib.c redefines it, that
leads to mingw runtime init to call into our strlen() and then ci->strlen() which
of course crashes. Apply the same fix as for malloc and friends: rename the symbol.
The codeclib.h include is necessary for normal builds.
Change-Id: Ifa85901a3e4a31cc0e10b4b905df348a239d5c99
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In DEBUG build, the codec API struct is consider with DEBUG flag in apps/
but without DEBUG flah in rbcodecs/, leading to unmatched structure and horrible
crashes in some cases (mostly encoders). I have no idea why the codecs Makefile
removes the DEBUG flag (maybe for performance reasons?) but it cannot be right.
Change-Id: Idb2c5f66741408ec2939624590fc39c4cf69fc2b
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The codec wasn't calling ci->set_offset() while decoding; as a result,
the saved offset in ci.id3->offset was only updated at the start of the
file and when seeking.
To reproduce the problem in the simulator or on a real device:
- Start playing an Opus file.
- Let it play until 15s, then turn the player off.
- Turn back on and resume playback. This'll resume correctly from 15s
(using time-based resume, I think, as the offset was 0?).
- Let it play until 30s, then turn the player off again.
- Turn back on and resume playback. This'll resume from 15s, based on
the initial position from last time, when it should resume from 30s.
I believe this will also fix FS#12799 ("Resuming opus file from bookmark
is not working correctly").
Change-Id: Iba67368e0029c968ef802693767e0722719bc38b
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ffmpeg_bitstream.c is included in libcodec, so there doesn't seem to
be any reason for individual codecs to also compile it (and clobber
any previous copy while they're at it, leading to broken builds)
Change-Id: I2bedc277ab109f44a6e8feb3d12ed01a720e00a6
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Change-Id: Ia051bc758c8fe4002e222511fdc6be613cdd39e7
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Fixes a buffer overflow present when MP3 is encoded at 32000 Hz sample
rate, affected bitrates are 320 and 256 kbps.
Change-Id: I7634e70409be9d675d47be316a42630dd3147636
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Reorganization
- Separated iBasso devices from PLATFORM_ANDROID. These are now standlone
hosted targets. Most device specific code is in the
firmware/target/hosted/ibasso directory.
- No dependency on Android SDK, only the Android NDK is needed.
32 bit Android NDK and Android API Level 16.
- Separate implementation for each device where feasible.
Code cleanup
- Rewrite of existing code, from simple reformat to complete reimplementation.
- New backlight interface, seperating backlight from touchscreen.
- Rewrite of device button handler, removing unneeded code and fixing memory
leaks.
- New Debug messages interface logging to Android adb logcat (DEBUGF, panicf,
logf).
- Rewrite of lcd device handler, removing unneeded code and fixing memory leaks.
- Rewrite of audiohw device handler/pcm interface, removing unneeded code and
fixing memory leaks, enabling 44.1/48kHz pthreaded playback.
- Rewrite of power and powermng, proper shutdown, using batterylog results
(see http://gerrit.rockbox.org/r/#/c/1047/).
- Rewrite of configure (Android NDK) and device specific config.
- Rewrite of the Android NDK specific Makefile.
Misc
- All plugins/games/demos activated.
- Update tinyalsa to latest from https://github.com/tinyalsa/tinyalsa.
Includes
- http://gerrit.rockbox.org/r/#/c/993/
- http://gerrit.rockbox.org/r/#/c/1010/
- http://gerrit.rockbox.org/r/#/c/1035/
Does not include http://gerrit.rockbox.org/r/#/c/1007/ due to new backlight
interface and new option for hold switch, touchscreen, physical button
interaction.
Rockbox needs the iBasso DX50/DX90 loader for startup, see
http://gerrit.rockbox.org/r/#/c/1099/
The loader expects Rockbox to be installed in /mnt/sdcard/.rockbox/. If
/mnt/sdcard/ is accessed as USB mass storage device, Rockbox will exit
gracefully and the loader will restart Rockbox on USB disconnect.
Tested on iBasso DX50.
Compiled (not tested) for iBasso DX90.
Compiled (not tested) for PLATFORM_ANDROID.
Change-Id: I5f5e22e68f5b4cf29c28e2b40b2c265f2beb7ab7
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check handle before clean up buffer in flush().
Change-Id: I36a130c45c9f5dce97aa723ef98922b6935ead75
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surround_enabled was never true, end up dsp_surround_flush didn't work; Thats why a cracking noise occurs in right channel when moving track positions.
redo pbe/surround flush in a much simpler way suits the current single buffer style.
Change-Id: I394054ddfb164b82c90b3dcf49df4442db87d8d2
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Most of the work comes from http://gerrit.rockbox.org/r/#/c/1088/
by Thomas Jarosch.
Change-Id: Iaa673dad2388d1e44fc95ffaa14bafadc6158101
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perceptual bass enhancement
- a bbe-ish group delay corrction with Biophonic EQ boost.
- precut
auditory fatigue reduction
-reduce signal in frequency that may trigger temporary threshold shift
haas surround
-frequency between f(x1) and f(x2) is always bypassed.
-can apply to side only.
Change-Id: Icb6355ce9b1c99bf2c58c9385c3c411c0ae209d3
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Change-Id: I30219d626316776eb73b4205d63376fa3dbc6361
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- Leave original ptr untouched if allocation fails
(bail out early)
- Behave like malloc() in case ptr is NULL
Change-Id: Ib854ca19bd0e069999b7780d2d9a533ece705add
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Change-Id: Id9b50c1fdeca4d87f158da717de8958330f027ef
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This file revealed several problems with our ASF parser:
1) The packet count in the ASF was actually a 64 bit value,
leading to overflow in very long files.
2) Seeking blindly trusted the bitrate listed in the ASF header
rather than computing it from the packet size and number of packets.
Fix these problems and fix a few minor issues.
Change-Id: Ie0f68734e6423e837757528ddb155f3bdcc979f3
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Change-Id: Ie3aa9b208e3f4f17d4d02f11f69839e9b381217d
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Change-Id: I5d9e731c3ea786fb910afbb0a5201fc68dcab9f9
Reviewed-on: http://gerrit.rockbox.org/965
Reviewed-by: Nick Peskett <rockbox@peskett.co.uk>
Tested: Nick Peskett <rockbox@peskett.co.uk>
Reviewed-by: Marcin Bukat <marcin.bukat@gmail.com>
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