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2014-12-08Merge from Chromium at DEPS revision 39.0.2171.95android-5.1.1_r5android-5.1.1_r28android-5.1.1_r22android-5.1.1_r17android-5.1.1_r12lollipop-mr1-wfc-releaselollipop-mr1-devBen Murdoch
This commit was generated by merge_to_master.py. Change-Id: Ie8322cf9b80cce0537ab2f0dda42eb304fe28f87
2014-12-08Update makefiles after merge of Chromium at 39.0.2171.95Ben Murdoch
This commit was generated by merge_from_chromium.py. Change-Id: I7d508dc2f6277f5e10d088d7b24cbd5a8af1ae03
2014-12-08Merge third_party/webrtc from ↵Ben Murdoch
https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at 33e74211aee74a1cde7968e646c59ba98337ea2a This commit was generated by merge_from_chromium.py. Change-Id: Ifdef209484f2547f7534ed54c2f52d7719c5a091
2014-12-02Merge 7729 "Build fix for MIPS Android Webview build."andrew@webrtc.org
> Build fix for MIPS Android Webview build. > > Excluding optimizations to support MIPS32R6 platform for Android Webview build (see also https://code.google.com/p/webrtc/source/detail?r=7580). > > R=andrew@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/32459004 TBR=ljubomir.papuga@gmail.com BUG=chromium:438351 Review URL: https://webrtc-codereview.appspot.com/27329004 git-svn-id: http://webrtc.googlecode.com/svn/branches/39/webrtc@7786 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02Merge 7343 "Changed mips_arch_variant variable value correspondi..."andrew@webrtc.org
> Changed mips_arch_variant variable value corresponding to Chromium code changes. > > 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>. TBR=ljubomir.papuga@gmail.com BUG=chromium:438351 Review URL: https://webrtc-codereview.appspot.com/32279004 git-svn-id: http://webrtc.googlecode.com/svn/branches/39/webrtc@7785 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02Merge 7580 "Build fix for MIPS32R6."andrew@webrtc.org
> Build fix for MIPS32R6. > > 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>. TBR=lpapuga@mips.com BUG=chromium:438351 Review URL: https://webrtc-codereview.appspot.com/31079004 git-svn-id: http://webrtc.googlecode.com/svn/branches/39/webrtc@7784 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29Merge from Chromium at DEPS revision 39.0.2171.44Ben Murdoch
This commit was generated by merge_to_master.py. Change-Id: I02f051e689b01973d19895b2b500af1e74b9e9ba
2014-10-29Update makefiles after merge of Chromium at 39.0.2171.44Ben Murdoch
This commit was generated by merge_from_chromium.py. Change-Id: I5c3566d504f0e795ef5518262dfd5148387a68d3
2014-10-29Merge third_party/webrtc from ↵Ben Murdoch
https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at 26a0c4c9568f9e616e9e9fa8652911ddd1f1f70a This commit was generated by merge_from_chromium.py. Change-Id: I3ebdc4c1beb90e1938d1646637e59ae4f3bc2b71
2014-10-27Merge r7418 to 39 branchtnakamura@webrtc.org
BUG=chromium:415935 TBR=xians Review URL: https://webrtc-codereview.appspot.com/29979004 git-svn-id: http://webrtc.googlecode.com/svn/branches/39/webrtc@7533 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14Merge from Chromium at DEPS revision 39.0.2171.26Torne (Richard Coles)
This commit was generated by merge_to_master.py. Change-Id: I90f1e1916435bed20b57f031468b55da8db2092b
2014-10-14Update makefiles after merge of Chromium at 39.0.2171.26Torne (Richard Coles)
This commit was generated by merge_from_chromium.py. Change-Id: Ieeacceb079ec9a3274cb535aa553687a7f48570c
2014-10-14Merge third_party/webrtc from ↵Torne (Richard Coles)
https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at 102e8974ff5b0bea642178e580076bf541841f9a This commit was generated by merge_from_chromium.py. Change-Id: Icb4e2ba2c733a7abd5b78be9be303d7742696695
2014-10-06Merge webrtc r7310 to M39.jiayl@webrtc.org
TBR=pbos@webrtc.org BUG=420753 Review URL: https://webrtc-codereview.appspot.com/29689004 git-svn-id: http://webrtc.googlecode.com/svn/branches/39/webrtc@7376 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-06Merge webrtc r7301 to M39 branch. jiayl@webrtc.org
TBR=juberti@webrtc.org BUG=414211 Review URL: https://webrtc-codereview.appspot.com/29679004 git-svn-id: http://webrtc.googlecode.com/svn/branches/39/webrtc@7375 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Create WebRTC branch 39 from trunk@7296tnakamura@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/branches/39/webrtc@7345 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Merge from Chromium at DEPS revision 267aeeb8d85cPrimiano Tucci
This commit was generated by merge_to_master.py. Change-Id: I64db0e7da49b895ed7379626b7d7abbe995d21f0
2014-09-25Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 53545bbfc47f2cddb7038395369a0dcd457c8b34 This commit was generated by merge_from_chromium.py. Change-Id: I9a04351d8f1bfde313d4565bd7f276b9684920d7
2014-09-25Revert r7049/r7123, which added unnecessary "u"s to "return 0"s.henrik.lundin@webrtc.org
r7049 added some unnecessary casts ("return 0" -> "return static_cast<uint16_t>(0)"). r7123 converted these to "return 0u". The original impetus for this was to eliminate type conversion warnings. However, the 'u's are unnecessary; Visual Studio can return "0" from a function returning an unsigned value without producing a warning. The real reason for the original warnings was that the code was returning -1 from a function returning an unsigned value, which does need a cast; the -1s were eliminated before the above two revisions landed. Also reverse the order of some conditionals to prevent possible underflow. While the underflow wouldn't have changed the behavior of the code, it's easier to reason about the code when such underflow can't happen, and possibly safer against future modifications as well. BUG=3663 TEST=none R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22599004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7296 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25Revert r7049/r7123, which added unnecessary "u"s to "return 0"s.henrik.lundin@webrtc.org
r7049 added some unnecessary casts ("return 0" -> "return static_cast<uint16_t>(0)"). r7123 converted these to "return 0u". The original impetus for this was to eliminate type conversion warnings. However, the 'u's are unnecessary; Visual Studio can return "0" from a function returning an unsigned value without producing a warning. The real reason for the original warnings was that the code was returning -1 from a function returning an unsigned value, which does need a cast; the -1s were eliminated before the above two revisions landed. Also reverse the order of some conditionals to prevent possible underflow. While the underflow wouldn't have changed the behavior of the code, it's easier to reason about the code when such underflow can't happen, and possibly safer against future modifications as well. BUG=3663 TEST=none R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22599004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7296 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25Fix typo from RtpPacketizerH264.pbos@webrtc.org
BUG= R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27609004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7295 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25Fix typo from RtpPacketizerH264.pbos@webrtc.org
BUG= R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27609004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7295 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25Revert "Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup." (rev 7293).andresp@webrtc.org
Breaks windows bot as it was already showing on the try jobs on the BUG=crbug/414211 R=jiayl@webrtc.org,juberti@webrtc.org TBR=jiayl@webrtc.org,juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26599004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7294 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25Revert "Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup." (rev 7293).andresp@webrtc.org
Breaks windows bot as it was already showing on the try jobs on the BUG=crbug/414211 R=jiayl@webrtc.org,juberti@webrtc.org TBR=jiayl@webrtc.org,juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26599004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7294 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup.jiayl@webrtc.org
BUG=crbug/414211 R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22739004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7293 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup.jiayl@webrtc.org
BUG=crbug/414211 R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22739004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7293 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Enable render downmixing to mono in AudioProcessing.andrew@webrtc.org
In practice, we have been doing this since time immemorial, but have relied on the user to do the downmixing (first voice engine then Chromium). It's more logical for this burden to fall on AudioProcessing, however, who can be expected to know that this is a reasonable approach for AEC. Permitting two render channels results in running two AECs serially. Critically, in my recent change to have Chromium adopt the float interface: https://codereview.chromium.org/420603004 I removed the downmixing by Chromium, forgetting that we hadn't yet enabled this feature in AudioProcessing. This corrects that oversight. The change in paths hit by production users is very minor. As commented it required adding downmixing to the int16_t path to satisfy bit-exactness tests. For reference, find the ApmTest.Process errors here: https://paste.googleplex.com/6372007910309888 BUG=webrtc:3853 TESTED=listened to the files output from the Process test, and verified that they sound as expected: higher echo while the AEC is adapting, but afterwards very close. R=aluebs@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7292 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Enable render downmixing to mono in AudioProcessing.andrew@webrtc.org
In practice, we have been doing this since time immemorial, but have relied on the user to do the downmixing (first voice engine then Chromium). It's more logical for this burden to fall on AudioProcessing, however, who can be expected to know that this is a reasonable approach for AEC. Permitting two render channels results in running two AECs serially. Critically, in my recent change to have Chromium adopt the float interface: https://codereview.chromium.org/420603004 I removed the downmixing by Chromium, forgetting that we hadn't yet enabled this feature in AudioProcessing. This corrects that oversight. The change in paths hit by production users is very minor. As commented it required adding downmixing to the int16_t path to satisfy bit-exactness tests. For reference, find the ApmTest.Process errors here: https://paste.googleplex.com/6372007910309888 BUG=webrtc:3853 TESTED=listened to the files output from the Process test, and verified that they sound as expected: higher echo while the AEC is adapting, but afterwards very close. R=aluebs@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7292 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Add missing DesktopConfigurationMonitor Unlock in webrtc::ScreenCapturerMacjiayl@webrtc.org
BUG=3837 R=jiayl@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30469004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7291 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Add missing DesktopConfigurationMonitor Unlock in webrtc::ScreenCapturerMacjiayl@webrtc.org
BUG=3837 R=jiayl@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30469004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7291 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Fix a problem in Thread::Send.jiayl@webrtc.org
Previously if thread A->Send is called on thread B, B->ReceiveSends will be called, which enables an arbitrary thread to invoke calls on B while B is wait for A->Send to return. This caused mutliple problems like issue 3559, 3579. The fix is to limit B->ReceiveSends to only process requests from A. Also disallow the worker thread invoking other threads. BUG=3559 R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7290 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Fix a problem in Thread::Send.jiayl@webrtc.org
Previously if thread A->Send is called on thread B, B->ReceiveSends will be called, which enables an arbitrary thread to invoke calls on B while B is wait for A->Send to return. This caused mutliple problems like issue 3559, 3579. The fix is to limit B->ReceiveSends to only process requests from A. Also disallow the worker thread invoking other threads. BUG=3559 R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7290 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 291035ed1d8ec308ffbc81e9cd119e2f53f92f86 This commit was generated by merge_from_chromium.py. Change-Id: I8de5d3b4724dd14ebda167e51683b554ddb5e024
2014-09-24Call NS AnalyzeCaptureAudio before AECaluebs@webrtc.org
This attenuates the noise pumping generated from the NS adapting to the AEC comfort noise. When there is echo present the AEC suppresses it and adds comfort noise. This is underestimated on purpose to avoid adding more than the original background noise. The NS has to be called after the AEC, because every non-linear processing before it can ruin its performance. Therefore the noise estimation can adapt to this comfort noise, making it less aggressive and generating noise pumping. By putting the noise estimation analysis stage from the NS before the AEC, this effect can be avoided. This has been tested manually on recordings where noise pumping was present: Two long recordings done in our team by bjornv and kwiberg plus the most noisy (5) recordings in the QA set. On the other hand, one risk of doing this is to not adapt to the comfort noise and therefore suppress too much. As verified in the tested files, this is not a problem in practice. BUG=webrtc:3763 R=andrew@webrtc.org, bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24679004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7289 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Call NS AnalyzeCaptureAudio before AECaluebs@webrtc.org
This attenuates the noise pumping generated from the NS adapting to the AEC comfort noise. When there is echo present the AEC suppresses it and adds comfort noise. This is underestimated on purpose to avoid adding more than the original background noise. The NS has to be called after the AEC, because every non-linear processing before it can ruin its performance. Therefore the noise estimation can adapt to this comfort noise, making it less aggressive and generating noise pumping. By putting the noise estimation analysis stage from the NS before the AEC, this effect can be avoided. This has been tested manually on recordings where noise pumping was present: Two long recordings done in our team by bjornv and kwiberg plus the most noisy (5) recordings in the QA set. On the other hand, one risk of doing this is to not adapt to the comfort noise and therefore suppress too much. As verified in the tested files, this is not a problem in practice. BUG=webrtc:3763 R=andrew@webrtc.org, bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24679004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7289 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Reduce jitter delay for low fps streams.sprang@webrtc.org
Enabled by finch flag. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31389005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7288 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Reduce jitter delay for low fps streams.sprang@webrtc.org
Enabled by finch flag. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31389005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7288 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Moved the filter calculation from analyze to process in ns_corealuebs@webrtc.org
It makes sense to have it there if the analyze and process methods are called in different stages. Tested over the entire QA set for bit exactness. BUG=webrtc:3811 R=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29549004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7287 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Moved the filter calculation from analyze to process in ns_corealuebs@webrtc.org
It makes sense to have it there if the analyze and process methods are called in different stages. Tested over the entire QA set for bit exactness. BUG=webrtc:3811 R=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29549004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7287 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24audioproc: Now also writes to output file in simulation modebjornv@webrtc.org
After changing to use wav as default file format no output was written in simulation mode. BUG=3359 TESTED=locally R=aluebs@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7286 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24audioproc: Now also writes to output file in simulation modebjornv@webrtc.org
After changing to use wav as default file format no output was written in simulation mode. BUG=3359 TESTED=locally R=aluebs@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7286 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_tkwiberg@webrtc.org
We have to fix both at once, since there's a macro that calls one of them or the other. BUG=909 R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org Committed: https://code.google.com/p/webrtc/source/detail?r=7266 Review URL: https://webrtc-codereview.appspot.com/19229004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7285 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_tkwiberg@webrtc.org
We have to fix both at once, since there's a macro that calls one of them or the other. BUG=909 R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org Committed: https://code.google.com/p/webrtc/source/detail?r=7266 Review URL: https://webrtc-codereview.appspot.com/19229004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7285 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Thread annotation of rtc::CriticalSection.pbos@webrtc.org
Effectively re-lands r5516 which was reverted because talk/-only checkouts existed. This now resides in webrtc/base/, so no talk/-only checkouts should be possible. This change also enables -Wthread-safety for talk/ and fixes a bug in talk/media/webrtc/webrtcvideoengine2.cc where a guarded variable was read without taking the corresponding lock. R=andresp@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/27569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7284 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Thread annotation of rtc::CriticalSection.pbos@webrtc.org
Effectively re-lands r5516 which was reverted because talk/-only checkouts existed. This now resides in webrtc/base/, so no talk/-only checkouts should be possible. This change also enables -Wthread-safety for talk/ and fixes a bug in talk/media/webrtc/webrtcvideoengine2.cc where a guarded variable was read without taking the corresponding lock. R=andresp@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/27569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7284 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Move thread_annotations.h to webrtc/base/.pbos@webrtc.org
R=andresp@webrtc.org, mflodman@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/27579004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Move thread_annotations.h to webrtc/base/.pbos@webrtc.org
R=andresp@webrtc.org, mflodman@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/27579004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23Use VPX_IMG_FMT_*/VPX_PLANE_* definesjohannkoenig@google.com
The compatibility layer has been removed upstream: https://gerrit.chromium.org/gerrit/gitweb?p=webm%2Flibvpx.git;a=commit;h=9cdaa3d72eade9ad162ef8f78a93bd8f85c6de10 BUG=webrtc:3839 R=marpan@google.com, marpan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7279 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23Use VPX_IMG_FMT_*/VPX_PLANE_* definesjohannkoenig@google.com
The compatibility layer has been removed upstream: https://gerrit.chromium.org/gerrit/gitweb?p=webm%2Flibvpx.git;a=commit;h=9cdaa3d72eade9ad162ef8f78a93bd8f85c6de10 BUG=webrtc:3839 R=marpan@google.com, marpan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7279 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23Revert "Remove DTMF status methods from Voice Engine" r7276henrik.lundin@webrtc.org
This change caused some trouble. TBR=henrika@webrtc.org,pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7277 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-28 19:34:57 +0000