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2014-11-05Update makefiles after merge of Chromium at 5a645aa13b82Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I038f8684aa804c94ca2c5175bdeaf605bf0611c5
2014-11-05Update makefiles after merge of Chromium at 2d0da5605d75Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I2b5db589b04e302cb1067fe730b81f3fb21b06bb
2014-11-04Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 9a8c28f10f329c5ce91e77057933e60224000627 This commit was generated by merge_from_chromium.py. Change-Id: If9e805f5024e1fcdd99127626811f9650e109b1d
2014-11-04Update makefiles after merge of Chromium at a99b7ad25d02Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I172fda810eb6cb37d17ba35571733f9eaeb9b230
2014-11-04Reworked paced sender queuesprang@webrtc.org
Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage. Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these. Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Update makefiles after merge of Chromium at 30ec995cdb2dAndroid Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I77678e9f2e5044a6457f21cada6ee13b75fbfb0c
2014-11-04Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at a1fd19c12e4efdf4b8a5f92323070443d50dc34e This commit was generated by merge_from_chromium.py. Change-Id: Ibd48eca2d93e6324a2e886e451f27307aab45e9b
2014-11-04Adds support for finch experiments to video_loopback.stefan@webrtc.org
Adds support for logging to stderr via -logs. Enables abs-send-time by default. R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23329004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7613 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Fix problem with late packets in NetEqhenrik.lundin@webrtc.org
Since r7255, it could happen that an old packet would block the decoding process until enough packet was received for the buffer to flush. This CL fixes that by: - Partially reverting r7255; - Remove recent old packets before taking a decision for GetAudio; - Remove all old packets after a packet has been extracted for decoding; - Adding tests for reordered packets. BUG=chrome:423985 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25079004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7612 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Delete VideoReceiveStream channels in destructor.pbos@webrtc.org
R=stefan@webrtc.org BUG=1667 Review URL: https://webrtc-codereview.appspot.com/31909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16kwiberg@webrtc.org
R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26039004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7610 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Remove the useless dummy state parameter to WebRtcG711_*kwiberg@webrtc.org
R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7609 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Remove the codec_type_ member from AudioDecoderkwiberg@webrtc.org
It isn't actually required, as evidenced by the comparative ease with which it can be removed. R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31939004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7606 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Enables AIMD control by default.stefan@webrtc.org
BUG=1788 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27019004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7604 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Improving error message from neteq_rtpplayhenrik.lundin@webrtc.org
If a packet with unknown RTP payload type is inserted, this CL will make sure that the error message is a little more detailed and gives a better understadning of what to do. BUG=2692 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7603 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Add format members to AudioConverter for DCHECKing.andrew@webrtc.org
And use a std::min. Post-commit fixes after: https://review.webrtc.org/30779004/ TBR=kwiberg Review URL: https://webrtc-codereview.appspot.com/25059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7600 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Update rate control parameter in vp9 test.marpan@webrtc.org
TBR=hellner@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7599 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Roll chromium_revision: 28d1981..d3db2ffmarpan@webrtc.org
Pick up the libvpx roll: https://codereview.chromium.org/674753002 Summary of changes (https://chromium.googlesource.com/chromium/src/+/28d1981..d3db2ff/DEPS): * third_party/android_tools 36bf7ac..ea50ccc * third_party/boringssl 7ea8481..751e889 * third_party/icu 8ac906f..d8b2a9d * third_party/libvpx efe9712..2e5ced5 * third_party/usrsctp/usrsctplib * tools/gyp 1990:1991 * tools/swarming_client a57d7db..bcb3bc3 Clang is not updated in this roll. Made the change getchar() --> getc(stdin) as seems like getchar() isn't supported on android anymore. (getchar() was causing the error: undefined reference to '__srget') Update rate control parameter in vp9 test. R=andrew@webrtc.org TBR=ajm@google.com Review URL: https://webrtc-codereview.appspot.com/23229004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7598 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Restore the void return type on WriteWavHeader.andrew@webrtc.org
Karl pointed out that the user can check the validity of the input parameters with CheckWavParameters prior to calling. TBR=kwiberg Review URL: https://webrtc-codereview.appspot.com/23339004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7597 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03replace inline assembly WebRtcNsx_AnalysisUpdate by intrinsics.andrew@webrtc.org
The modification only uses the unique part of the analysis_update function. Pass byte to byte conformance test on both ARMv7 and AArch64, and the single function performance is similar with original assembly version on different platforms. If not specified, the code is compiled by GCC 4.6. The result is the "X version / C version" ratio, and the less is better. | run 100k times | cortex-a7 | cortex-a9 | cortex-a15 | | use C as the base on each | (1.2Ghz) | (1.0Ghz) | (1.7Ghz) | | CPU target | | | | |----------------------------+-----------+-----------+------------| | Neon asm | 15.61% | 20.15% | 14.89% | | Neon inline asm (LLVM 3.4) | 25.98% | 33.96% | 18.18% | | Neon intrinsics (GCC 4.6) | 22.06% | 27.01% | 19.24% | | Neon intrinsics (GCC 4.8) | 17.28% | 18.23% | 18.55% | | Neon intrinsics (LLVM 3.4) | 21.02% | 19.98% | 16.76% | BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28849004 Patch from Zhongwei Yao <zhongwei.yao@arm.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7596 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Add Opus support to neteq_rtpplayhenrik.lundin@webrtc.org
BUG=2416 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30859004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7595 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Add UMA metrics for the initial (after two seconds) packet loss, round-trip ↵stefan@webrtc.org
time and bandwidth estimate of a WebRTC call. BUG=crbug/425925 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7593 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Add stats for video:asapersson@webrtc.org
- number of sent/received RTCP NACK/FIR/PLI per minute - percentage of unique sent/received NACK requests - percentage of discarded/duplicated packets by the jitter buffer - permille of sent/received key frames BUG=crbug/419657 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7592 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Add more sanity checks to workaround the unidentified problem that ↵braveyao@webrtc.org
CaptureThread is still running while related resouces are destroyed already. BUG= TEST=auto test R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28859004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7590 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01Adjust/increase rate control thresold for a vp9 test.marpan@webrtc.org
TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25049004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7589 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01Add VP9 codec to VCM and vie_auto_test.marpan@webrtc.org
Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Update Android projects to API level 21.kjellander@webrtc.org
The update in https://webrtc-codereview.appspot.com/23309004 was not enough, so this updates to 21 instead. This is required in order to roll chromium_revision to keep up with Chrome, as third_party/android_tools have now dropped support for API level 20. Commands used: third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-21 --path webrtc/examples/android/opensl_loopback third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-21 --path webrtc/examples/android/media_demo/ third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-21 --path talk/examples/android/ Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when build/android/envsetup.sh is sourced. BUG= R=glaznev@webrtc.org, henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7587 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31replace inline assembly WebRtcNsx_SynthesisUpdateNeon by intrinsics.andrew@webrtc.org
The modification only uses the unique part of the synthesis_update function. Pass byte to byte conformance test both on ARMv7 and ARMv8, and the single function performance is similar with original assembly version on different platforms (if not specified, the code is compiled by GCC 4.6): | run 100k times | cortex-a7 | cortex-a9 | cortex-a15 | | use C as the base | (1.2Ghz) | (1.0Ghz) | (1.7Ghz) | | (the smaller the better) | | | | |----------------------------+-----------+-----------+------------| | C | 100% | 100% | 100% | | Neon asm | 15.93% | 17.01% | 12.50% | | Neon inline asm | 27.74% | 31.41% | 14.64% | | Neon intrinsics (GCC 4.8) | 17.84% | 14.10% | 13.84% | | Neon intrinsics (LLVM 3.4) | 16.63% | 14.01% | 12.98% | BUG=3580 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23159004 Patch from Zhongwei Yao <zhongwei.yao@arm.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7586 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Add a WavReader counterpart to WavWriter.andrew@webrtc.org
Don't bother with a C interface as we currently have no need to call this from C code. The first use will be in the audioproc tool. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7585 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Update makefiles after merge of Chromium at a41c404b1c7fAndroid Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Ifcab5d7c5bd698b1a0a72100960585183048352d
2014-10-31Update makefiles after merge of Chromium at b210e2d62956Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I3b99d06f861694a90ee0f32a97380e1c99cfaa07
2014-10-31Update all .isolate files for the new format.kjellander@webrtc.org
R=kjellander@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/27809004 Patch from Marc-Antoine Ruel <maruel@chromium.org>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Update Android projects to API level 20.kjellander@webrtc.org
This is required in order to roll chromium_revision to keep up with Chrome, as third_party/android_tools have now dropped support for API level 19. Commands used: third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-20 --path webrtc/examples/android/opensl_loopback third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-20 --path webrtc/examples/android/media_demo/ third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-20 --path talk/examples/android/ Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when build/android/envsetup.sh is sourced. BUG= R=glaznev@webrtc.org, henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23309004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7582 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Fix N7 camera aspect ratio.glaznev@webrtc.org
N7 video preview generates stretched output: https://code.google.com/p/android/issues/detail?id=70830. To workaround the problem set camera picture size in addition to video preview size with the same resolution. BUG=3971 R=braveyao@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7581 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Build fix for MIPS32R6.andrew@webrtc.org
Exclude MIPS optimizations for MIPS32R6 build since some of the instructions are not supported. This is temporary fix, until the MIPS32R6 code is added. R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25989004 Patch from Ljubomir Papuga <lpapuga@mips.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7580 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Fix a name collision with Android libc++andrew@webrtc.org
The Android libc++ has a symbol called '_P' This CL renames a property called _P in webrtc. BUG=chromium:427718 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30009004 Patch from Fabrice de Gans-Riberi <fdegans@chromium.org>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7579 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 1a02faa335e7d8076b5cf8dd9a584e72669b0c8e This commit was generated by merge_from_chromium.py. Change-Id: I6a99300dc0ba646ed641a90356f632d782c83dbe
2014-10-31Implement conference-mode temporal-layer screencast.pbos@webrtc.org
Renames VideoStream::temporal_layers to temporal_layer_thresholds_bps to convey that it contains thresholds needed to ramp up between them (1 threshold -> 2 temporal layers, etc.). R=mflodman@webrtc.org, stefan@webrtc.org BUG=1788,1667 Review URL: https://webrtc-codereview.appspot.com/23269004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7578 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Configure A/V sync in WebRtcVideoEngine2.pbos@webrtc.org
Sets up A/V sync for the first video receive channel with the default voice channel. This is only done when conference mode is disabled to preserve existing behavior. Ideally we'd know which voice channel to sync with here. R=mflodman@webrtc.org, stefan@webrtc.org BUG=1788 Review URL: https://webrtc-codereview.appspot.com/23249004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7577 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Simplify bwe tests.stefan@webrtc.org
R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7576 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Revert "Revert part of r7561, "Refactor audio conversion functions.""andrew@webrtc.org
This restores the conversion changes to AudioProcessing originally added in r7561, with minor alterations to ensure it passes all tests. TBR=kwiberg Review URL: https://webrtc-codereview.appspot.com/28899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7574 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31arm64 iOS build.tkchin@webrtc.org
Allows successful build of arm64 libraries using GYP_DEFINES="OS=ios target_arch=arm64 target_subarch=arm64". Note that not all libraries will be NEON optimized (eg common_audio), however most importantly libvpx will be. WEBRTC_ARCH_ARM needs to be defined so that libvpx doesn't post-process, which is significantly detrimental to performance. BUG=3898 R=kjellander@webrtc.org, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7573 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30Add 15 fps support for Android devices with missing 15 fpsglaznev@webrtc.org
camera mode. Some latest Android devices support only 30 fps for front camera, but HW VP8 encoder performance is not enough for 720p 30 fps encoding. Add 15 fps support for these devices by allowing frame drop in Android camera wrapper. BUG= R=tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7571 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30Creating a C++ wrapper class for VADhenrik.lundin@webrtc.org
Also adding a mock. This work is part of an ongoing effort to encapsulate encoders in AudioEncoder classes. The CNG encoder will also be implemented as an AudioEncoder class, and will also contain a VAD C++ wrapper. BUG=3926 R=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27839004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7570 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30Revert part of r7561, "Refactor audio conversion functions."kwiberg@webrtc.org
Specifically, revert this part: "Remove hacks in AudioBuffer intended to maintain bit-exactness with the float path. The conversions etc. are now all natural, and instead we enforce close but not bit-exact output between the two paths." But keep the conversion function rename, since that doesn't seem to be causing problems. R=tina.legrand@webrtc.org, bjornv@webrtc.org TBR=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7569 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30Refactor audio conversion functions.andrew@webrtc.org
Use a consistent naming scheme that can be understood at the callsite without having to refer to documentation. Remove hacks in AudioBuffer intended to maintain bit-exactness with the float path. The conversions etc. are now all natural, and instead we enforce close but not bit-exact output between the two paths. Output of ApmTest.Process: https://paste.googleplex.com/5931055831842816 R=aluebs@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13049004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7561 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29Use external VideoDecoders in VideoReceiveStream.pbos@webrtc.org
Removes direct VideoCodec use from the new API, exposes VideoDecoders through webrtc/video_decoder.h similar to VideoEncoders. Also includes some preparation for wiring up external decoders in WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they were allocated internally or externally. Additionally addresses a data race in VideoReceiver that was exposed with this change. R=mflodman@webrtc.org, stefan@webrtc.org TBR=pthatcher@webrtc.org BUG=2854,1667 Review URL: https://webrtc-codereview.appspot.com/27829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29Add stats for duplicate sent and received NACK requests.asapersson@webrtc.org
R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27799004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7559 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29common_audio: Removed macro WEBRTC_SPL_RSHIFT_W32bjornv@webrtc.org
Replaces the trivial macro WEBRTC_SPL_RSHIFT_W32 with >> at various places in common_audio and removes it. BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7558 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29Remove unused code in overuse detector.asapersson@webrtc.org
R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31579004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7557 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-01 15:39:39 +0000