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2014-10-08audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>bjornv@webrtc.org
Replaced trivial shift macro with >>. The actual implementation of the macro is simply >>. Affected codecs: * ilbc * isac/fix BUG=3348,3353 TESTED=locally on linux and trybots R=henrik.lundin@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7396 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-08CHECK/DCHECK: Explicitly state whether the condition can have side effectskwiberg@webrtc.org
R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7394 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-08Change name of a NetEq internal member variablehenrik.lundin@webrtc.org
In the StatisticsCalculator class, the member last_report_timestamp_ was unfortunately named. It's now been changed to timestamps_since_last_report_, which describes it more accurately. R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7393 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07Explicitly unpoison FDs for MSan.pbos@webrtc.org
MSan doesn't handle inline assembly that's used by FD_ZERO causing a false positive. R=earthdok@chromium.org, henrike@webrtc.org BUG=chromium:344505 Review URL: https://webrtc-codereview.appspot.com/25799004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7388 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07Remove potential deadlock in WebRtcVideoEngine2.pbos@webrtc.org
Fixes lock-order inversions between capturer's SignalVideoFrame and WebRtcVideoSendStream. Additionally also removes all deadlock suppressions for WebRtcVideoEngine2. R=stefan@webrtc.org TBR=kjellander@webrtc.org BUG=1788,2999 Review URL: https://webrtc-codereview.appspot.com/26729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7386 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07Roll chromium_revision c264a05..fc668e2 (297113:298195)kjellander@webrtc.org
Mainly to pick up https://codereview.chromium.org/619723006/ to fix our MSan bot. Summary of changes (git diff c264a05..fc668e2 DEPS): * third_party/boringssl 01fe820..c7dd5f30 * third_party/usrsctp/usrsctplib 8975bd5..d5685d4 * tools/swarming_client 79940aee..33d442a Clang updated 216630:217949 (git diff c264a05..fc668e2 tools/clang/scripts/update.sh) This caused TSan v2 to hit an assert in libjingle_peerconnection_unittest and libjingle_media_unittest, which is why the dlclose call had to be disabled for now (webrtc:3895). BUG=3895 R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7385 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07Revert "Only configure the SSL library in one place."pbos@webrtc.org
This reverts commit r7378, which broke Windows compile targets elsewhere. R=kjellander@webrtc.org TBR=henrike@webrtc.org BUG=chromium:413497 Review URL: https://webrtc-codereview.appspot.com/28679004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7384 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07Isolate: Remove use of --ignore_broken_itemskjellander@webrtc.org
BUG=chromium:395700 R=jam@chromium.org Review URL: https://webrtc-codereview.appspot.com/30659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7383 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07Fix neteq_rtpplay so that empty SSRC is validhenrik.lundin@webrtc.org
In r7380, the command line flag --ssrc was added to neteq_rtpplay. However, it was not possible to omit that flag, since the validation did not accept an empty string. This CL fixes that. TBR=kwiberg@webrtc.org BUG=2692 Review URL: https://webrtc-codereview.appspot.com/24869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7382 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07Set NetEq playout mode through the Config structhenrik.lundin@webrtc.org
This change opens up the possibility to set the playout mode when creating the NetEq object. The old methods SetPlayoutMode and PlayoutMode are still available, but are deprecated. BUG=3520 R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7381 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07Add an SSRC filter to neteq_rtpplayhenrik.lundin@webrtc.org
This allows the user to set the desired SSRC if the input file contains multiple streams. BUG=2692 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30609004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7380 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07Prevent reading outside iSAC bitstream, if the stream is corrupted.turaj@webrtc.org
BUG=chrome_373312(#24) R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7379 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-06Only configure the SSL library in one place.henrike@webrtc.org
Build settings now use use_openssl in both Chromium and standalone builds. It moves all the platform-specific SSL-related build checks to be conditioned on this flag as appropriate. This is to avoid colliding with Chromium's transition away from NSS. BUG=chromium:413497 R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29559004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7378 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-06Mac: adds missing _DEBUG flag to mac debug builds.henrike@webrtc.org
BUG=3836 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7377 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-06Check on the existence of report directoryhoussainy@google.com
Reports will be written at rtcBot/test/reports/<report_name> R=andresp@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26689004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7372 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-03Wire up CPU adaptation in WebRtcVideoEngine2.pbos@webrtc.org
Includes clean-up work to be able to use the webrtc::Call::Config that's set up. This introduced a CallFactory to spawn a FakeCall with the config used and allowed removal of FakeWebRtcVideoChannel2. BUG=1788 R=mflodman@webrtc.org, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7370 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-02Moves xmllite's unittests to rtc_unittest.henrike@webrtc.org
BUG=3836 R=andrew@webrtc.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26669005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7369 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-02Let RtpFileSource use RtpFileReaderhenrik.lundin@webrtc.org
RtpFileSource used to implement it's own RTP dump file reader, but with this change it simply uses RtpFileReader. One benefit is that pcap files are now also supported. All NetEq test tools that use RtpFileSource are updated. BUG=2692 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22839004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7367 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-02audio_processing: Replaced WEBRTC_SPL_RSHIFT_U32 with >>bjornv@webrtc.org
A trivial macro that is replaced. Affected components: * AGC * NSx BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29599005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7366 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-02Remove mouse cursor capturer from the ScreenCapturer interfacesergeyu@chromium.org
Mouse can be captured using MouseCursorMonitor and all code in chromium already uses it instead of ScreenCapturer. R=jiayl@webrtc.org Committed: https://code.google.com/p/webrtc/source/detail?r=7363 Review URL: https://webrtc-codereview.appspot.com/31529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7365 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-02Revert "Remove mouse cursor capturer from the ScreenCapturer interface"sergeyu@chromium.org
This reverts commit 0adc4953512ee0a57cf7f3c0591b024c2316554a. It broke FYI bots TBR=sergeyu@chromium.org Review URL: https://webrtc-codereview.appspot.com/27649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7364 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-02Remove mouse cursor capturer from the ScreenCapturer interfacesergeyu@chromium.org
Mouse can be captured using MouseCursorMonitor and all code in chromium already uses it instead of ScreenCapturer. R=jiayl@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7363 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-01Add error trap for XFixesGetCursorImage()sergeyu@chromium.org
BUG=https://code.google.com/p/webrtc/issues/detail?id=3245 R=jiayl@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31519004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7362 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-01Import LappedTransform and friends.andrew@webrtc.org
Add code for doing block-based frequency domain processing. Developed and reviewed in isolation. Corresponding export CL: https://chromereviews.googleplex.com/95187013/ R=bercic@google.com, kjellander@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31539004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7359 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-01rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.henrike@webrtc.org
BUG=N/A R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7358 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-01Fix parallelizability in modules_tests.pbos@webrtc.org
R=henrik.lundin@webrtc.org BUG=3873 TEST=third_party/gtest-parallel/gtest-parallel -r 10 -w 64 out/Debug/modules_tests Review URL: https://webrtc-codereview.appspot.com/24799004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7354 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-01Reland "Remove DTMF status methods from Voice Engine" r7276henrik.lundin@webrtc.org
This reverts r7277. TBR=henrika@webrtc.org,pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29599004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7353 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30GN: Add common configs to tools and test.kjellander@webrtc.org
Similar changes as in https://review.webrtc.org/28589004/ were missed in https://review.webrtc.org/25569004/. This should fix the Chromium WebRTC FYI bots that currently are broken due to lack of include paths. BUG=3441 TBR=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7347 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30GN: Enable libvpx, add link target and convert some test targetskjellander@webrtc.org
Libvpx now supports GN and this CL turns on compiling it. I also introduced an executable target 'webrtc_tests' that depends on all in WeBRTC + tests in order to get a full linking step executed (since we've seen link problems for GN when rolling WebRTC into Chromium). I also converted a few test targets and made a GN file for third_party/gflags. BUG=3441 TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc dir to a workspace wit this CL applied. R=brettw@chromium.org TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Changed mips_arch_variant variable value corresponding to Chromium code changes.andrew@webrtc.org
Chromium commit URL: https://crrev.com/c8a5da7455b57b2399e4a69e8100c098d9870052 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23809004 Patch from Ljubomir Papuga <lpapuga@mips.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7343 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Revert 7337 "Reland 28629004: adding new AEC dump start interfac..."xians@webrtc.org
> Reland 28629004: adding new AEC dump start interface for chrome > > adding new AEC dump start interface for chrome. > > This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here: > http://msdn.microsoft.com/en-us/library/ms235460.aspx > > Chromium bug:crbug/415935 > TEST=bots > R=bjornv@webrtc.org, kwiberg@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/27639004 TBR=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22849004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7342 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Revert 7338 "Fixed the android build by making the interface pur..."xians@webrtc.org
> Fixed the android build by making the interface pure virtual. > > TBR=asapersson@webrtc.org, bjornv@webrtc.org, > > Review URL: https://webrtc-codereview.appspot.com/24789004 TBR=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30579004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7341 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Collecting stats every fixed time in webrtc_video_streaming.js testhoussainy@google.com
and prepare the format these collected stats to be plotted using one of external dev-tools. R=andresp@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7340 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Minor code change to fix some warnings in MIPS build.andrew@webrtc.org
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26619004 Patch from Ljubomir Papuga <lpapuga@mips.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7339 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Fixed the android build by making the interface pure virtual.xians@webrtc.org
TBR=asapersson@webrtc.org, bjornv@webrtc.org, Review URL: https://webrtc-codereview.appspot.com/24789004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7338 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Reland 28629004: adding new AEC dump start interface for chromexians@webrtc.org
adding new AEC dump start interface for chrome. This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here: http://msdn.microsoft.com/en-us/library/ms235460.aspx Chromium bug:crbug/415935 TEST=bots R=bjornv@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7337 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest.henrike@webrtc.org
BUG=N/A R=andrew@webrtc.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31499004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7336 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Revert 7334 "adding new AEC dump start interface for chrome."xians@webrtc.org
> adding new AEC dump start interface for chrome. > > This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here: > http://msdn.microsoft.com/en-us/library/ms235460.aspx > > Chromium bug:crbug/415935 > TEST=bots > R=bjornv@webrtc.org, kwiberg@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/28629004 TBR=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7335 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30adding new AEC dump start interface for chrome.xians@webrtc.org
This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here: http://msdn.microsoft.com/en-us/library/ms235460.aspx Chromium bug:crbug/415935 TEST=bots R=bjornv@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7334 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Minor modifications to test::RtpFileReaderhenrik.lundin@webrtc.org
Adding original_length to the Packet struct. This is populated with the plen value from the RTP dump file. In the case of reading a pcap file, original_length will be equal to length. Also increasing the maximum packet size to 3500 bytes. This is to accomodate some test files that contain PCM16b audio encoding. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28609004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7333 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30audio_processing/aecm: Added help function for calculating log of energybjornv@webrtc.org
The same operation of calculating log of the energy was executed four times. This CL adds a help function LogOfEnergyInQ8() for that. BUG=N/A TESTED=locally on linux and trybots R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7331 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30audio_processing: Removed usage of macro WEBRTC_SPL_MULbjornv@webrtc.org
WEBRTC_SPL_MUL is a trivial multiplication after casting to int32_t. This is already taken care of by the compiler which makes the macro unnecessary. Affected components: * AGC * NSx BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7330 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30audio_processing: Replaced trivial macro WEBRTC_SPL_LSHIFT_W32 with <<bjornv@webrtc.org
Affected components: * AECM * AGC * HPF * NSx BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7329 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Revert 7327 "Update isolate.gypi files + link to isolate_driver.py"kjellander@webrtc.org
Breaks debug compilation (didn't run all trybots when testing this). > Update isolate.gypi files + link to isolate_driver.py > > This updates the isolate.gypi copies we're forced to > maintain in our code repo to Chromium revision c264a05. > > Since isolated testing is now using a new launch script > in tools: isolate_driver.py, that is added to our links > script. > > BUG=395700 > TESTED=Ran one of our tests with: > ninja -C out/Release tools_unittests_run > tools/isolate_driver.py run --isolated out/Release/tools_unittests.isolated --isolate webrtc/tools/tools_unittests.isolate > > R=henrika@webrtc.org, jam@chromium.org > > Review URL: https://webrtc-codereview.appspot.com/26649004 TBR=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7328 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Update isolate.gypi files + link to isolate_driver.pykjellander@webrtc.org
This updates the isolate.gypi copies we're forced to maintain in our code repo to Chromium revision c264a05. Since isolated testing is now using a new launch script in tools: isolate_driver.py, that is added to our links script. BUG=395700 TESTED=Ran one of our tests with: ninja -C out/Release tools_unittests_run tools/isolate_driver.py run --isolated out/Release/tools_unittests.isolated --isolate webrtc/tools/tools_unittests.isolate R=henrika@webrtc.org, jam@chromium.org Review URL: https://webrtc-codereview.appspot.com/26649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7327 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29Reland disallowing blocking calls on the worker thread.jiayl@webrtc.org
This fixed the issue that invoking the call when the thread is not started. BUG=3559 R=juberti@webrtc.org, thorcarpenter@google.com Review URL: https://webrtc-codereview.appspot.com/24769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7325 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29Set thread scheduling parameters inside the new thread.henrike@webrtc.org
This makes it possible to restrict threads from modifying scheduling parameters of another thread in the Chrome Linux sandbox. BUG= R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28539004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7324 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29Fix parallel test execution for tools, testsupport and metrics tests.kjellander@webrtc.org
BUG=2600 TESTED=Passing tests using: python third_party/gtest-parallel/gtest-parallel -w 10 -r 20 out/Release/test_support_unittests python third_party/gtest-parallel/gtest-parallel -w 10 -r 20 out/Release/tools_unittests python third_party/gtest-parallel/gtest-parallel -w 10 -r 20 out/Release/video_engine_tests R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31479004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7322 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29audio_processing: Replaced macro WEBRTC_SPL_LSHIFT_W16 with <<bjornv@webrtc.org
A trivial macro that serves no purpose. Affected components are: * audio_processing/nsx * audio_processing/agc * audio_processing/aecm * common_audio/LpcToReflCoef BUG=3348,3353 TESTED=locally on linux R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22539004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7321 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29common_audio refactoring: Removed macro WEBRTC_SPL_LSHIFT_U32bjornv@webrtc.org
The macro is a trivial shift operator including a cast before shift. There is no guard against negative shifts. Replaced with << at place and added casts when necessary. Affects both fixed and float point versions of iSAC BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27369004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7320 4adac7df-926f-26a2-2b94-8c16560cd09d
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