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* Opus: update resume offset correctly while playing.Adam Sampson2016-09-07
| | | | | | | | | | | | | | | | | | | | 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
* Fix race conditions in parallel build.Frank Gevaerts2016-05-28
| | | | | | | | 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
* _BSD_SOURCE is deprecated, and we're supposed to use _DEFAULT_SOURCE now.Frank Gevaerts2016-03-18
| | | | Change-Id: Ia051bc758c8fe4002e222511fdc6be613cdd39e7
* mp3_enc.c: fix MP3 recording at 32 kHz sample rateCástor Muñoz2015-07-17
| | | | | | | Fixes a buffer overflow present when MP3 is encoded at 32000 Hz sample rate, affected bitrates are 320 and 256 kbps. Change-Id: I7634e70409be9d675d47be316a42630dd3147636
* iBasso DX50/DX90: Major code cleanup and reorganization.Udo Schläpfer2015-02-02
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* fix surround & pbe dsp crashChiwen Chang2015-01-30
| | | | | | check handle before clean up buffer in flush(). Change-Id: I36a130c45c9f5dce97aa723ef98922b6935ead75
* fix pbe/haas surround dsps: surround_enabled, redo flush functions.Chiwen Chang2015-01-22
| | | | | | | | 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
* Enable buflib and core_alloc for warble.Frank Gevaerts2015-01-19
| | | | | | | Most of the work comes from http://gerrit.rockbox.org/r/#/c/1088/ by Thomas Jarosch. Change-Id: Iaa673dad2388d1e44fc95ffaa14bafadc6158101
* three new DSPsChiwen Chang2015-01-19
| | | | | | | | | | | | | | | 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
* Limit more variables to file scopeThomas Jarosch2015-01-11
| | | | Change-Id: I30219d626316776eb73b4205d63376fa3dbc6361
* More standard conforming codec_realloc()Thomas Jarosch2014-12-27
| | | | | | | | - Leave original ptr untouched if allocation fails (bail out early) - Behave like malloc() in case ptr is NULL Change-Id: Ib854ca19bd0e069999b7780d2d9a533ece705add
* Fix warning in WMA Pro and remove a c++ comment.Michael Giacomelli2014-11-28
| | | | Change-Id: Id9b50c1fdeca4d87f158da717de8958330f027ef
* Fix FS#13009.Michael Giacomelli2014-11-28
| | | | | | | | | | | | | | 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
* fix yellowMarcin Bukat2014-09-22
| | | | Change-Id: Ie3aa9b208e3f4f17d4d02f11f69839e9b381217d
* metadata: Add cuesheet embedded in ape tags.nialv72014-09-22
| | | | | | | | 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>
* Fix last warning for WarbleMichael Sevakis2014-08-30
| | | | | | Unused result warnings will have to be dealt with separately. Change-Id: I00c45e28d4d43a5376745036e650ff8df576c2db
* Get the last errors I hope!Michael Sevakis2014-08-30
| | | | Change-Id: Ia285b95480cc9ac6494b745d80892c4b1b912341
* Fix warnings from 6ed0087Michael Sevakis2014-08-16
| | | | | | | | | | | | | | Forgot to (void) an unused parameter when priorityless. usb-drv-rl27xx.c was using a compound init to initialize a semaphore but the structure changed so that it is no longer correct. Use designated initializers to avoid having to complete all fields. Forgot to break compatibility on all plugins and codecs since the kernel objects are now different. Take care of that too and do the sort thing. Change-Id: Ie2ab8da152d40be0c69dc573ced8d697d94b0674
* libopus: use iram for more constantsNils Wallménius2014-07-13
| | | | | | | | Speeds up decoding of the 64 kbps test file by 2.59 MHz and the 128 kbps test file by 4.31 MHz on H300 (cf). Decoding the same files on c200 is sped up by 0.33 MHz and 0.55 MHz respectively. Change-Id: I0f9f9ef6a7293581cf45e3201b33c65504c95c81
* libopus: asm C_MUL for coldfireNils Wallménius2014-07-13
| | | | | | | | | | The recent merge of upstream changed the fft to use C_MUL which wasn't implemented in asm for coldfire. Speeds up decoding 64 kbps test file by 2.68 MHz and 128 kbps test file by 2.80 MHz on H300. Change-Id: I8b61fc0f9568d6350431e311a12e44fe4f60f72e
* Sync to upstream libopusNils Wallménius2014-07-13
| | | | | | | | | | | | | | | | | | | | | | Sync to commit bb4b6885a139644cf3ac14e7deda9f633ec2d93c This brings in a bunch of optimizations to decode speed and memory usage. Allocations are switched from using the pseudostack to using the real stack. Enabled hacks to reduce stack usage. This should fix crashes on sansa clip, although some files will not play due to failing allocations in the codec buffer. Speeds up decoding of the following test files: H300 (cf) C200 (arm7tdmi) ipod classic (arm9e) 16 kbps (silk) 14.28 MHz 4.00 MHz 2.61 MHz 64 kbps (celt) 4.09 MHz 8.08 MHz 6.24 MHz 128 kbps (celt) 1.93 MHz 8.83 MHz 6.53 MHz Change-Id: I851733a8a5824b61feb363a173091bc7e6629b58
* libmad: Use 32bit unsigned for requantize table.Thomas Martitz2014-04-15
| | | | | | | | | | Implicit promotion of integer literals to unsigned long introduced a subtle bug on 64-bit systems due to weird sign extensions (leads to audible glitches in a few files). The table is originally designed for unsigned 32bit integers, and it works with those so use them. As a consequence the lookup table size is halved as well. Change-Id: I35d878d6df03300387f0e403e0f3c3bdc73eea00
* Implement time-based resume and playback start.Michael Sevakis2014-03-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | This complements offset-based resume and playback start funcionality. The implementation is global on both HWCODEC and SWCODEC. Basically, if either the specified elapsed or offset are non-zero, it indicates a mid-track resume. To resume by time only, set elapsed to nonzero and offset to zero. To resume by offset only, set offset to nonzero and elapsed to zero. Which one the codec uses and which has priority is up to the codec; however, using an elapsed time covers more cases: * Codecs not able to use an offset such as VGM or other atomic formats * Starting playback at a nonzero elapsed time from a source that contains no offset, such as a cuesheet The change re-versions pretty much everything from tagcache to nvram. Change-Id: Ic7aebb24e99a03ae99585c5e236eba960d163f38 Reviewed-on: http://gerrit.rockbox.org/516 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested: Michael Sevakis <jethead71@rockbox.org>
* Fix more reds.Thomas Martitz2014-03-03
| | | | Change-Id: I4b58dda0953b7f9799238c32b78037b0a5403c04
* Fix various reds. Some includes needed fixup.Thomas Martitz2014-03-03
| | | | Change-Id: I4327740bae17054131feb917abdd58846c451988
* Proposed fix for FS#12878: Zero-length embedded album art prevents mp3 playbackJack Whitham2013-12-23
| | | | | | | | | see http://www.rockbox.org/tracker/task/12878 Change-Id: Ib4233c06e18d1d193dfb9e73e745ca5d174e40b2 Reviewed-on: http://gerrit.rockbox.org/507 Reviewed-by: Michael Giacomelli <giac2000@hotmail.com> Reviewed-by: Thomas Martitz <kugel@rockbox.org>
* Sync libopus to upstream release 1.1Nils Wallménius2013-12-16
| | | | Change-Id: I9fea7460fc33f60faff961b3389dd97b5191463c
* DSP Compressor: Sidechain, Exponential Atk/RlsRyan Billing2013-12-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is an improvement to the current compressor which I have added to my own Sansa Fuze V2 build. I am submitting here in case others find it interesting. Features added to the existing compressor: Attack, Look-ahead, Sidechain Filtering. Exponential attack and release characteristic response. Benefits from adding missing features: Attack: Preserve perceived "brightness" of tone by letting onset transients come through at a higher level than the rest of the compressed program material. Look-ahead: With Attack comes clipping on the leading several cycles of a transient onset. With look-ahead function, this can be pre-emptively mitigated with a slower gain change (less distortion). Look-ahead limiting is implemented to prevent clipping while keeping gain change ramp to an interval near 3ms instead of instant attack. The existing compressor implementation distorts the leading edge of a transient by causing instant gain change, resulting in log() distortion. This sounds "woofy" to me. Exponential Attack/Release: eMore natural sounding. On attack, this is a true straight line of 10dB per attack interval. Release is a little different, however, sounds natural as an analog compressor. Sidechain Filtering: Mild high-pass filter reduces response to low frequency onsets. For example, a hard kick drum is less likely to make the whole of the program material appear to fade in and out. Combined with a moderate attack time, such a transient will ride through with minimal audible artifact. Overall these changes make dynamic music sound more "open", more natural. The goal of a compressor is to make dyanamic music sound louder without necessarily sounding as though it has been compressed. I believe these changes come closer to this goal. Enjoy. If not, I am enjoying it Change-Id: I664eace546c364b815b4dc9ed4a72849231a0eb2 Reviewed-on: http://gerrit.rockbox.org/626 Tested: Purling Nayuki <cyq.yzfl@gmail.com> Reviewed-by: Michael Giacomelli <giac2000@hotmail.com>
* Add support for flac embeded album art.Albert Song2013-12-13
| | | | | | Change-Id: I077768f7d80b57976f9a7278b640ef67cf4f2af2 Reviewed-on: http://gerrit.rockbox.org/694 Reviewed-by: Thomas Martitz <kugel@rockbox.org>
* flac: fix seeking.Andrew Ryabinin2013-11-18
| | | | | | | | | | | | | | | | | | | | | | | As comment in code states: "It is possible for our seek to land in the middle of audio data that looks exactly like a frame header from a future version of an encoder. When that happens, frame_sync() will return false. But there is a remote possibility that it is properly synced at such a "future-codec frame", so to make sure, we wait to see several "unparseable" errors in a row before bailing out." Currently we wait for 10 "unparseable" errors. libFLAC waits for 20. But I've got a valid flac+cue, wherein switching to certain track gave me 24 "unparsaeable" errors. Therefore I increased unparseable_count to 30. Change-Id: I4e97a5385c729adf3d5075d41ea312622c69e548 Reviewed-on: http://gerrit.rockbox.org/658 Reviewed-by: Michael Giacomelli <giac2000@hotmail.com> Reviewed-by: Boris Gjenero <boris.gjenero@gmail.com> Tested-by: Andrew Ryabinin <ryabinin.a.a@gmail.com> Reviewed-by: Andrew Ryabinin <ryabinin.a.a@gmail.com>
* Add missing #include statements.Kevin Zheng2013-10-20
| | | | | | | | | | | | | | Although Linux accepts several implicit definitions of SEEK_END found in stdio.h, the compiler on FreeBSD won't. Rockbox compilation will fail without stdio.h included. There is a precedent for including this header, see lib/rbcodec/codecs/libtremor/ivorbisfile.h. Change-Id: I58510101b59a354cd6601cb3f323f385a824d2e8 Reviewed-on: http://gerrit.rockbox.org/639 Tested-by: Kevin Zheng <kevinz5000@gmail.com> Reviewed-by: Frank Gevaerts <frank@gevaerts.be>
* Simulator - encoders can now be loadedLorenzo Miori2013-10-05
| | | | | | | | | This enables the encoders - i.e. to record audio - to be loaded also on the simulator. Change-Id: I54fdbeb75b89023c0d7824a34cf76301c02c3150 Reviewed-on: http://gerrit.rockbox.org/632 Reviewed-by: Thomas Martitz <kugel@rockbox.org>
* Opus: delete two files that were moved into a subdirNils Wallménius2013-09-01
| | | | Change-Id: I54ef0dfd57fbb493ad38855767a8f5e724e5bc65
* Fix warning from a35c1b3Nils Wallménius2013-09-01
| | | | Change-Id: I0e9b2c265a6a2355dc39b1696df4c8f266d9a638
* Opus: Coldfire inline asm for comb_filter_constNils Wallménius2013-09-01
| | | | | | Speeds up decoding a 64kbps test file by 2.6MHz Change-Id: Ibeb30f37cc00a4a6f65b64851009753f40e06fc1
* Add cf asm inline for multiplication commonly used in silk.Nils Wallménius2013-08-31
| | | | | | Speeds up decoding a 16kbps test file by 4.9MHz on h300. Change-Id: I8c25431c98dfa9a1c3806a84055e0847eb77a9f1
* Put two hot silk arrays on real stack (iram)Nils Wallménius2013-08-31
| | | | | | Speeds up decoding of 16kbps test file by 16.7MHz on H300. Change-Id: I39c90e3b423ae8e2ee5c2b88c5dcec8d48807f77
* Silence spurious warningNils Wallménius2013-08-31
| | | | Change-Id: I856c722e959314c0a86e9c0a3a31cb824ddb41cc
* Sync opus codec to upstream gitNils Wallménius2013-08-31
| | | | | | | | | | | | | | | | | | | | | | | Sync opus codec to upstream commit 02fed471a4568852d6618e041c4f2af0d7730ee2 (August 30 2013) This brings in a lot of optimizations but also makes the diff between our codec and the upstream much smaller as most of our optimizations have been upstreamed or supeceded. Speedups across the board for CELT mode files: 64kbps 128kbps H300 9.82MHz 15.48MHz c200 4.86MHz 9.63MHz fuze v1 10.32MHz 15.92MHz For the silk mode test file (16kbps) arm targets get a speedup of about 2MHz while the H300 is 7.8MHz slower, likely because it's now using the pseudostack more rather than the real stack which is in iram. Patches to get around that are upcomming. Change-Id: Ifecf963e461c51ac42e09dac1e91bc4bc3b12fa3
* Change CODECFLAGS to a "simply-expanded" var to give the individualNils Wallménius2013-08-23
| | | | | | | | | | | | codec makefiles larger freedom in what they can do to it. Use this in libopus to prepend the libopus searchpaths to CODECFLAGS so that its internal config.h will be picked up before our global one. This avoids having to do a s/config.h/opus_config.h/ when syncing which will be handy soon. Change-Id: I018d729aa0c8300fa3149f22a5a8c5668b339dfa Reviewed-on: http://gerrit.rockbox.org/496 Reviewed-by: Nils Wallménius <nils@rockbox.org>
* Fix FS#12889 : Audible pop right after setting Repeat/ShuffleMichael Sevakis2013-08-16
| | | | | | | | | The quickscreen calls settings_apply() and the crossfeed code wasn't checking that the right crossfeed was set before updating the filter for the custom setting, which was overwriting the Meier crossfeed data (custom and Meier share the same data space). Change-Id: Ifaa2f46fe062d4497681a2dd0d5068ec906c96a3
* mp3_enc: Fix early snafu with stream finish on COPMichael Sevakis2013-07-11
| | | | | | | Distractions make logic fail. It only needs one more loop and should not trigger further compression cycles after not feeding more data. Change-Id: Ie0dbb34af92e0ca5718480dd4ab4719a141717ff
* Multithread compressing encoders on multicore targets.Michael Sevakis2013-07-09
| | | | | | | | | | | | | | | | For mp3_enc, split encoding duties between COP and CPU. For wavpack_enc, simply run the encoding on COP (splitting that one needs more consideration) which keeps the it and the UI from running on the same core. As a result, at least they are now useable on PP at "normal" sample rates. mp3_enc in all this gets an extensive renovation and some optimizations for speed, to reduce IRAM requirements and remove unneeded stuff. Change-Id: I215578dbe36f14e516b05a5ca70880eb01ca0ec2
* Enable setting of global output samplerate on certain targets.Michael Sevakis2013-07-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Replaces the NATIVE_FREQUENCY constant with a configurable frequency. The user may select 48000Hz if the hardware supports it. The default is still 44100Hz and the minimum is 44100Hz. The setting is located in the playback settings, under "Frequency". "Frequency" was duplicated in english.lang for now to avoid having to fix every .lang file for the moment and throwing everything out of sync because of the new play_frequency feature in features.txt. The next cleanup should combine it with the one included for recording and generalize the ID label. If the hardware doesn't support 48000Hz, no setting will be available. On particular hardware where very high rates are practical and desireable, the upper bound can be extended by patching. The PCM mixer can be configured to play at the full hardware frequency range. The DSP core can configure to the hardware minimum up to the maximum playback setting (some buffers must be reserved according to the maximum rate). If only 44100Hz is supported or possible on a given target for playback, using the DSP and mixer at other samperates is possible if the hardware offers them. Change-Id: I6023cf0c0baa8bc6292b6919b4dd3618a6a25622 Reviewed-on: http://gerrit.rockbox.org/479 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
* Update software recording engine to latest codec interface.Michael Sevakis2013-06-30
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Basically, just give it a good rewrite. Software codec recording can be implemented in a more straightforward and simple manner and made more robust through the better codec control now available. Encoded audio buffer uses a packed format instead of fixed-size chunks and uses smaller data headers leading to more efficient usage. The greatest benefit is with a VBR format like wavpack which needs to request a maximum size but only actually ends up committing part of that request. No guard buffers are used for either PCM or encoded audio. PCM is read into the codec's provided buffer and mono conversion done at that time in the core if required. Any highly-specialized sample conversion is still done within the codec itself, such as 32-bit (wavpack) or interleaved mono (mp3). There is no longer a separate filename array. All metadata goes onto the main encoded audio buffer, eliminating any predermined file limit on the buffer as well as not wasting the space for unused path queue slots. The core and codec interface is less awkward and a bit more sensible. Some less useful interface features were removed. Threads are kept on narrow code paths ie. the audio thread never calls encoding functions and the codec thread never calls file functions as before. Codecs no longer call file functions directly. Writes are buffered in the core and data written to storage in larger chunks to speed up flushing of data. In fact, codecs are no longer aware of the stream being a file at all and have no access to the fd. SPDIF frequency detection no longer requires a restart of recording or plugging the source before entering the screen. It will poll for changes and update when stopped or prerecording (which does discard now-invalid prerecorded data). I've seen to it that writing a proper header on full disk works when the format makes it reasonably practical to do so. Other cases may have incorrect data sizes but sample info will be in tact. File left that way may play anyway. mp3_enc.codec acquires the ability to write 'Info' headers with LAME tags to make it gapless (bonus). Change-Id: I670685166d5eb32ef58ef317f50b8af766ceb653 Reviewed-on: http://gerrit.rockbox.org/493 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
* Fix some whitespace in files changed in following commit.Michael Sevakis2013-06-30
| | | | | | | Change-Id: Ie3f43e43076e0dcae9a10f1b0b9e4698b398acee Reviewed-on: http://gerrit.rockbox.org/492 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
* Remove old EQ presets.Michael Giacomelli2013-06-21
| | | | | | | | | | | The old presets never made sense for Rockbox's EQ. They were apparently copied from some other software. We have a parametric EQ, that means that EQ bands can be made wider or narrower. Putting two identical bands side by side just wastes battery life and adds rounding error. Replacement presets are on gerrit but they need more work. In the mean time, users should probably not be using these. Change-Id: I85213100129fafd3ac0fa1a9438cb4d651bb94cb
* Silence some (harmless) warnings.Frank Gevaerts2013-06-16
| | | | Change-Id: I8d1278b8cfaa376d2ad5a99dd552dc980c66e1da
* Fix id3v2 album art if more than one image is present.Dominik Riebeling2013-06-15
| | | | | | | | | | | | | Rockbox only uses the first album art image (APIC / PIC frame) found in id3v2 tags. When a file contains more than one image the second one is ignored but the parsealbumart() callback overwrites the already set data. This causes the metadata structure to contain an invalid pointer to the image data, resulting in no image shown. Make parsealbumart() aware of this and skip parsing when an albumart image has already been found. Fixes FS#12870. Change-Id: Id8164f319cd5e1ee868b581f8f4ad3ea69c17f77
* Missed removing a couple unwanted includes in previous commit.Michael Sevakis2013-05-27
| | | | | | Get those too. Change-Id: Id2a39afe7a61d6ec0cea38633b94fe1b7122204f
generated by cgit v1.1 at 2026-04-25 13:26:12 +0000