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2017-02-23Leave only an empty top level OWNERS file.Chih-Hung Hsieh
We should not copy OWNERS files from upstream, or the owners should be registered in Gerrit Code Review. Bug: 33166666 Test: default build targets Change-Id: Ibfd47e643f03678bb65880653383adb84809169d
2016-02-11Suppress unused-parameter warnings.Chih-Hung Hsieh
Suppress warnings until upstream can fix them. BUG: 27074506 Change-Id: If7e6f190100fba025d25d2634d1c9a657cc24854
2016-01-15Merge upstream SHA 04cb763Alex Luebs
* git merge 04cb763 * See all upstream changes since the previous merge in branch aosp/upstream-master: git diff cb3f9bd..04cb763 * Modify webrtc/.gitignore to keep *.mk files. * Removed old files from *.mk files: - thread.cc - thread_posix.cc * Add new files to *.mk files: - event_tracer.cc * Android relevant upstream changes: - Make Beamforming dynamically settable for Android platform builds - Remove additional channel constraints when Beamforming is enabled in AudioProcessing - Use an explicit identifier in Config Change-Id: I384a4e8f6982c31c5bc70eef521bb2d90800b9a4
2016-01-13Convert channel counts to size_t.Peter Kasting
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1316523002 . Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-08Misc. small cleanups.pkasting
* Better param names * Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases. * Use arraysize() * Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers * reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead * Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition * Fix indenting * Use uint32_t for timestamps (matching how it's already a uint32_t in most places) * Spelling * RTC_CHECK_EQ(expected, actual) * Rewrap * Use .empty() * Be more pedantic about matching int/int32_t/ * Remove pointless consts on input parameters to functions * Add missing sanity checks All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first. BUG=none TEST=none Review URL: https://codereview.webrtc.org/1534193008 Cr-Commit-Position: refs/heads/master@{#11191}
2015-12-18iSAC: Remove unnecessary WEBRTC_LINUX define.kjellander
I can only find one use in iSAC codebase: https://code.google.com/p/chromium/codesearch#chromium/src/third_party/webrtc/modules/audio_coding/test/iSACTest.cc&l=19 It's the prime suspect for causing a compilation error for iOS failing to include linux/net.h which is being included in webrtc/voice_engine/voice_engine_defines.h NOTRY=True Review URL: https://codereview.webrtc.org/1539883002 Cr-Commit-Position: refs/heads/master@{#11089}
2015-12-16Disable warnings failing when using Clang on Windows.kjellander
This makes it possible to build WebRTC using Clang on Windows. Depends on https://codereview.webrtc.org/1524703006/ BUG=webrtc:5360, webrtc:5366 NOTRY=True Review URL: https://codereview.webrtc.org/1522223002 Cr-Commit-Position: refs/heads/master@{#11058}
2015-12-15Fix incorrect commentkwiberg
Review URL: https://codereview.webrtc.org/1524663004 Cr-Commit-Position: refs/heads/master@{#11036}
2015-12-08Add encode/decode time tracing to audio_coding.Peter Boström
Also removes virtual from VideoDecoder::Decode and updated mocks and tests accordingly to use VideoDecoder::DecodeInternal instead. BUG=webrtc:5167 R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1512483003 . Cr-Commit-Position: refs/heads/master@{#10935}
2015-11-25WIP: Changes after merge commit 'cb3f9bd'Chih-Hung Hsieh
Changes after "git merge cb3f9bd" * git mv old Android.mk from src/ to webrtc/ * Remove old unused files in src/*. * Modify webrtc/.gitignore to keep *.mk files. * Copy old files from master, lost in auto-merge. src/modules/audio_processing/test/unit_test.cc src/modules/audio_coding/codecs/isac/fix/test/{Android.mk,kenny.c} to webrtc, but most of the old test code do not compile with new webrtc API and are commented out. * Move src/modules/audio_processing/test/android/apmtest/jni/*.mk to webrtc/... but the Android.mk files does not work. Commented out its build target. * Changes to Android.mk files: * Change references of src/ to webrtc/. * Fix include path * Fix source file list, remove old non-existing files, add new source files to resolve link errors. * Add new Android.mk files to build some new static libraries to link into current Android webrtc .so files. * Remove unnecessary LOCAL_SHARED_LIBRARIES in Android.mk files that build static libraries. * Remove old unnecessary clang workarounds like -Wno-tautological-pointer-compare -no-integrated-as * Fix include path of debug.pb.h in some source files. * Add -DWEBRTC_POSIX in android-webrtc.mk * Manually merge Android specific changes in src/typedefs.h to webrtc/typedefs.h * Fix trivial syntax error in scoped_ptr.h, calling static_assert. * Use -std=c++0x in webrtc/system_wrappers/source/Android.mk * #undef getchaar in spreadsort.hpp * Verified and not to carry old Android hacks from src/... to webrtc/... src/system_wrappers/source/android/cpu-features.c src/modules/interface/module.h src/modules/audio_coding/codecs/isac/fix/source/filters_neon.c src/system_wrappers/source/trace_posix.cc src/typedefs.h More pathes from Alex Luebs: * Use new unit test kenny.cc. Delete old kenny.cc. Comment out unessential code in kenny.cc to fix link error for now. * Replace old unit test files with new ones in webrtc/modules/audio_processing/Android.mk. Delete old audio_processing/test/unit_test.cc. * Fix compilation errors in webrtc/modules/audio_processing/test/audio_processing_unittest.cc Change-Id: I7bbf776eeb9dcfa21a82dd1f2dec378235cbbc3e
2015-11-18modules/audio_coding: Remove some codec include dirskjellander@webrtc.org
Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-16Adding stddef.h to opus_inst.h.minyue
This is to prevent size_t from undefined. This does not happen in current WebRTC since the sources that opus_inst.h gets used have proper definitions. But it would be good to add the definition in itself. Review URL: https://codereview.webrtc.org/1446093003 Cr-Commit-Position: refs/heads/master@{#10653}
2015-11-10Reland "Prevent Opus DTX from generating intermittent noise during silence"minyue
The original CL is reviewed at https://codereview.webrtc.org/1415173005/ A silly mistake was made at the last patch set, and the CL was reverted. This CL is to fix and reland it. BUG= Review URL: https://codereview.webrtc.org/1422213003 Cr-Commit-Position: refs/heads/master@{#10574}
2015-11-09Revert of Prevent Opus DTX from generating intermittent noise during silence ↵kjellander
(patchset #10 id:250001 of https://codereview.webrtc.org/1415173005/ ) Reason for revert: Breaks voe_auto_test on all three "large tests bots". https://build.chromium.org/p/client.webrtc/builders/Win32%20Release%20%5Blarge%20tests%5D/builds/5630/steps/voe_auto_test/logs/stdio https://build.chromium.org/p/client.webrtc/builders/Mac32%20Release%20%5Blarge%20tests%5D/builds/5599/steps/voe_auto_test/logs/stdio https://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/5645/steps/voe_auto_test/logs/stdio Notice these bots are no longer a part of the default trybot set, so they have to be run manually when working with code that affects their tests (they were removed as they queued up all the time in the CQ, and usually don't catch breakages). Original issue's description: > Prevent Opus DTX from generating intermittent noise during silence. > > Opus may have an internal error that causes this. Here we make a workaround by adding some small disturbance to the input signals to break a long sequence of zeros. > > BUG=webrtc:5127 > > Committed: https://crrev.com/f475add57eada116bc960fe2935876ec8c077977 > Cr-Commit-Position: refs/heads/master@{#10565} TBR=tina.legrand@webrtc.org,kwiberg@webrtc.org,solenberg@webrtc.org,minyue@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5127 Review URL: https://codereview.webrtc.org/1428613004 Cr-Commit-Position: refs/heads/master@{#10567}
2015-11-09Prevent Opus DTX from generating intermittent noise during silence.minyue
Opus may have an internal error that causes this. Here we make a workaround by adding some small disturbance to the input signals to break a long sequence of zeros. BUG=webrtc:5127 Review URL: https://codereview.webrtc.org/1415173005 Cr-Commit-Position: refs/heads/master@{#10565}
2015-11-06Pass audio to AudioEncoder::Encode() in an ArrayViewkwiberg
Instead of in separate pointer and size arguments. Review URL: https://codereview.webrtc.org/1418423010 Cr-Commit-Position: refs/heads/master@{#10535}
2015-10-30Switch usage of _DEBUG macro to NDEBUG.tfarina
http://stackoverflow.com/a/29253284/5237416 BUG=None R=tommi@webrtc.org NOPRESUBMIT=true Review URL: https://codereview.webrtc.org/1429513004 Cr-Commit-Position: refs/heads/master@{#10468}
2015-10-29audio_coding: rename interface -> includeHenrik Kjellander
BUG=webrtc:5095 R=henrik.lundin@webrtc.org TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417173004 . Cr-Commit-Position: refs/heads/master@{#10444}
2015-10-28system_wrappers: rename interface -> includeHenrik Kjellander
BUG=webrtc:5095 R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1413333002 . Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-20Workaround for false positive -Wmaybe-uninitialized being triggered on some ↵msniegocki
compilers Some toolchains (in this case referring to a g++ 4.9, or "arm-linux- androideabi-g++ (GCC) 4.9 20140827 (prerelease)" according to my --version, from the Android NDK r10e-rc4 and potentially with custom patches; others may be affected as well) fail to prove that myVec in WebRtcIsac_CorrelateInterVec is never used uninitialized. This is likely due to the compiler thinking the assignment in line 468 might not happen. Changing the loop condition in line 466 to rowCntr < SOME_CONSTANT also helps, suggesting that the compiler can't infer that there are only 2 values interVecDim can have at that point, and neither of them are 0. Of course, this is not an acceptable fix, as it changes behaviour. This seems to be a compiler bug, or at least an issue with its heuristics. However, we can't really change toolchains at the moment, and ultimately this change improves support for certain older compilers. BUG= Review URL: https://codereview.webrtc.org/1406423004 Cr-Commit-Position: refs/heads/master@{#10337}
2015-10-14Delete full-band mode from the iSAC codechenrik.lundin
This mode is no longer used. BUG=4210 Review URL: https://codereview.webrtc.org/1392173004 Cr-Commit-Position: refs/heads/master@{#10275}
2015-09-23Returning correct duration estimate on Opus DTX packets.minyuel
Bug 4985 revealed two flaws 1. Opus duration estimate did not return correct length for DTX packets, 2. NetEq DoCodecInternalCng did not assign enough buffer. P.S. Generalizing problem 1, current NetEq decode function checks memory size by calling the duration estimate function. This is not ideal. A better way is to let codec's decode function to receive buffer size and return failure if it is not enough. This can be made in a separate CL. BUG=webrtc:4985 R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1334303005 . Cr-Commit-Position: refs/heads/master@{#10031}
2015-09-22Simple cleanups of AudioDecoder and AudioEncoder classeskwiberg
* Make sure they're all final and don't allow copying or assignment. * Get rid of the single-channel PCM decoder classes. * Move some includes from .h to .cc files where possible. BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1353803002 Cr-Commit-Position: refs/heads/master@{#10021}
2015-09-22Move AudioDecoderIsac* to its own filesKarl Wiberg
Currently, it's sitting in AudioEncoderIsac*'s files, which is less than obvious. This CL puts the encoder and decoder in separate files together with the C implementation; CLs are afoot to make it so for the other built-in codecs as well. BUG=webrtc:4557 R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1339253003 . Cr-Commit-Position: refs/heads/master@{#10018}
2015-09-22Move AudioDecoderPcm* next to AudioEncoderPcm*kwiberg
All AudioDecoder subclasses have historically lived in NetEq, but they fit better with the codec they wrap. BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1348613003 Cr-Commit-Position: refs/heads/master@{#10015}
2015-09-18Fix ChromeOS build (C99 break)henrikg
BUG=5016 TBR=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1354163002 Cr-Commit-Position: refs/heads/master@{#9992}
2015-09-17Move AudioDecoderG722 next to AudioEncoderG722kwiberg
All AudioDecoder subclasses have historically lived in NetEq, but they fit better with the codec they wrap. BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1346993002 Cr-Commit-Position: refs/heads/master@{#9966}
2015-09-17Add RTC_ prefix to (D)CHECKs and related macros.henrikg
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition. Alternative solutions: * Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable. * Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce. * Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable. * Changes in Chromium for this is obviously not an option. BUG=chromium:468375 NOTRY=true Review URL: https://codereview.webrtc.org/1335923002 Cr-Commit-Position: refs/heads/master@{#9964}
2015-09-17Move AudioDecoderPcm16B next to AudioEncoderPcm16BKarl Wiberg
All AudioDecoder subclasses have historically lived in NetEq, but they fit better with the codec they wrap. BUG=webrtc:4557 R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1348113002 . Cr-Commit-Position: refs/heads/master@{#9963}
2015-09-17Move AudioDecoderIlbc next to AudioEncoderIlbckwiberg
All AudioDecoder subclasses have historically lived in NetEq, but they fit better with the codec they wrap. BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1348053002 Cr-Commit-Position: refs/heads/master@{#9961}
2015-09-16Add RTC_ prefix to contructormagic macros.henrikg
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition. * DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN * DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN * DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS Related CL: https://codereview.webrtc.org/1335923002/ BUG=chromium:468375 NOTRY=true Review URL: https://codereview.webrtc.org/1345433002 Cr-Commit-Position: refs/heads/master@{#9953}
2015-09-15Move AudioDecoderOpus next to AudioEncoderOpusKarl Wiberg
All AudioDecoder subclasses have historically lived in NetEq, but they fit better with the codec they wrap. BUG=webrtc:4557 R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1342933005 . Cr-Commit-Position: refs/heads/master@{#9944}
2015-09-15Android: Enable C99 mode instead of C89 (default).Henrik Kjellander
BUG=webrtc:4960 TESTED=Built locally using GYP and GN for Android. R=andrew@webrtc.org, brettw@chromium.org Review URL: https://codereview.webrtc.org/1321193003 . Cr-Commit-Position: refs/heads/master@{#9937}
2015-09-10MockAudioEncoder: Use a dedicated marker method for test expectationskwiberg
This makes the sequence of expected calls easier to read. Also, we can save one line and get rid of a gmock warning by expecting the MockAudioEncoder object to be destroyed at the end of the test instead of making a final marker call. Review URL: https://codereview.webrtc.org/1331793003 Cr-Commit-Position: refs/heads/master@{#9916}
2015-09-09AudioEncoderOpusTest.PacketLossRateOptimized: Fix bug and make prettierkwiberg
Fix bug 4981, which caused the second half (decreasing loss rates) to not test anything. In the process, the test is changed slightly to make it less dependent on the exact rounding behavior of doubles (by not testing exactly at the the points where the effective loss rate goes through a step---just very very close). A bunch of symbolic constants are also replaced with easy-to-read literal numbers. BUG=4981 Review URL: https://codereview.webrtc.org/1316673010 Cr-Commit-Position: refs/heads/master@{#9908}
2015-09-09Merge two files with AudioEncoderOpus testskwiberg
Merge the contents of audio_encoder_mutable_opus_test.cc into audio_encoder_opus_unittest.cc, since they're now both testing AudioEncoderOpus. (While preparing this CL, I noted a bug in the PacketLossRateOptimized test. This CL leaves that test essentially unchanged; I've posted bug 4981 about the problem.) Review URL: https://codereview.webrtc.org/1319713004 Cr-Commit-Position: refs/heads/master@{#9906}
2015-09-09Remove AudioEncoder methods SetMaxBitrate and SetMaxPayloadSizekwiberg
And the corresponding ACM methods SetISACMaxRate and SetISACMaxPayloadSize. They were only used in tests. Review URL: https://codereview.webrtc.org/1311533010 Cr-Commit-Position: refs/heads/master@{#9903}
2015-09-09Change return type of AudioEncoder::SetMaxPlaybackRate to voidkwiberg
There's no point in returning a status code, since the max playback rate is only a suggestion that the encoder is free to disregard. Review URL: https://codereview.webrtc.org/1332573003 Cr-Commit-Position: refs/heads/master@{#9900}
2015-09-09Turn webrtc::Vad into a pure virtual interfacekwiberg
Review URL: https://codereview.webrtc.org/1317243005 Cr-Commit-Position: refs/heads/master@{#9899}
2015-09-08Fold AudioEncoderMutable into AudioEncoderkwiberg
It makes more sense to combine the two interfaces, since there wasn't a clear line separating them. The result is a combined interface with just over a dozen methods, half of which need to be implemented by every subclass, while the other half have sensible (and trivial) default implementations and are implemented only by the few subclasses that need non-default behavior. Review URL: https://codereview.webrtc.org/1322973004 Cr-Commit-Position: refs/heads/master@{#9894}
2015-09-08copy-red: Fill an rtc::Buffer with bytes the easy waykwiberg
The easy way also happens to be more efficient if we have to reallocate, but that's a minor concern here. Review URL: https://codereview.webrtc.org/1327053002 Cr-Commit-Position: refs/heads/master@{#9876}
2015-08-31Revert "Avoiding size_t in MIPS."Niklas Enbom
This reverts commit 32e2f461b13c530d34f9c434e7e76da6ff3eda83. BUG=526716 TBR=minyue@webrtc.org Review URL: https://codereview.webrtc.org/1314873004 . Cr-Commit-Position: refs/heads/master@{#9824}
2015-08-31Avoiding size_t in MIPS.minyuel
TBR=pkasting@chromium.org BUG=526716 Review URL: https://codereview.webrtc.org/1310553008 . Cr-Commit-Position: refs/heads/master@{#9823}
2015-08-29Convert some more things to size_t.Peter Kasting
These changes stem from requests by Andrew on https://codereview.webrtc.org/1228823002/ to eliminate some "return -1"s and change to using asserts plus returning size_ts. I then also converted the relevant connected bits. This also cleans up a bunch of style issues, e.g. no spaces around operators. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, henrik.lundin@webrtc.org, niklas.enbom@webrtc.org Review URL: https://codereview.webrtc.org/1305983003 . Cr-Commit-Position: refs/heads/master@{#9813}
2015-08-27Rename local variable to avoid shadowingkwiberg
See comment here: https://codereview.webrtc.org/1208993010/diff/180001/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h#newcode189 TBR=minyue@webrtc.org Review URL: https://codereview.webrtc.org/1315333003 Cr-Commit-Position: refs/heads/master@{#9800}
2015-08-27AudioDecoder: Replace Init() with Reset()Karl Wiberg
The Init() method was previously used to initialize and reset decoders, and returned an error code. The new Reset() method is used for reset only; the constructor is now responsible for fully initializing the AudioDecoder. Reset() doesn't return an error code; it turned out that none of the functions it ended up calling could actually fail, so this CL removes their error return codes as well. R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1319683002 . Cr-Commit-Position: refs/heads/master@{#9798}
2015-08-26Fix MIPS compile.Peter Kasting
BUG=chromium:524885 TEST=none TBR=turaj Review URL: https://codereview.webrtc.org/1316843003 . Cr-Commit-Position: refs/heads/master@{#9790}
2015-08-24Update a ton of audio code to use size_t more correctly and in general reducePeter Kasting
use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24iSAC: Make separate AudioEncoder and AudioDecoder objectskwiberg
The only shared state is now the bandwidth estimation info. This reduces the amount and complexity of the locking substantially. Review URL: https://codereview.webrtc.org/1208993010 Cr-Commit-Position: refs/heads/master@{#9762}
2015-08-14Remove WebRtcIsac_Highpass_float().pkasting
This function is unreferenced and not even declared in a header file. Split from https://codereview.webrtc.org/1228793004/ . BUG=none TEST=none Review URL: https://codereview.webrtc.org/1296513002 Cr-Commit-Position: refs/heads/master@{#9716}
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