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2014-09-10Update makefiles after mergeandroid-wear-5.0.0_r1android-cts-5.0_r9android-cts-5.0_r8android-cts-5.0_r7android-cts-5.0_r6android-cts-5.0_r5android-cts-5.0_r4android-cts-5.0_r3android-5.0.2_r3android-5.0.2_r1android-5.0.1_r1android-5.0.0_r7android-5.0.0_r6android-5.0.0_r5.1android-5.0.0_r5android-5.0.0_r4android-5.0.0_r3android-5.0.0_r2android-5.0.0_r1lollipop-wear-releaselollipop-releaselollipop-devlollipop-cts-releaseTorne (Richard Coles)
Change-Id: I54fd8fa2b5737e4622f9edd92901c13a88159378
2014-08-06Merge from Chromium at DEPS revision 37.0.2062.68Bo Liu
This commit was generated by merge_to_master.py. Change-Id: I61c45b7edac14319b647e3ec56062b2fab6dbcef
2014-08-06Merge third_party/webrtc from ↵Bo Liu
https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at 14b8f01ca3c95e3f10a141f63c3250b38cf5433c This commit was generated by merge_from_chromium.py. Change-Id: Ia3dac47999c6fab4b3bb256de1db6582e9b630a0
2014-07-29Merge from Chromium at DEPS revision 37.0.2062.52Bo Liu
This commit was generated by merge_to_master.py. Change-Id: I09ef65c1efeffba799d62b31aa38782a2b4f979c
2014-07-29Merge third_party/webrtc from ↵Bo Liu
https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at 939b8c943d0198c81702beec3d7faa152b8a29a7 This commit was generated by merge_from_chromium.py. Change-Id: Ibf9d11a0241d98c988d88aef438438ce8bad60a8
2014-07-29Merge r6774 to branch 3.55.sprang@webrtc.org
See https://webrtc-codereview.appspot.com/13989004 R=asapersson@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18039004 git-svn-id: http://webrtc.googlecode.com/svn/branches/3.55/webrtc@6796 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-24Merge r6571 and r6572 to the 3.55 branch.stefan@webrtc.org
BUG=3527 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18909004 git-svn-id: http://webrtc.googlecode.com/svn/branches/3.55/webrtc@6775 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15Merge from Chromium at DEPS revision 37.0.2062.21Bo Liu
This commit was generated by merge_to_master.py. Change-Id: Ie7de57d4657b90fbbcb2617577a154b9492f6d1a
2014-07-15Update makefiles after merge of Chromium at 37.0.2062.21Bo Liu
This commit was generated by merge_from_chromium.py. Change-Id: I5419b760c2b33b2c1271554c8ed62f657c386d6c
2014-07-15Merge third_party/webrtc from ↵Bo Liu
https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at 1b4ed025b291d5d226ee45f1f1211becd39a3560 This commit was generated by merge_from_chromium.py. Change-Id: Id66890959e976958d482471269ed19c042e056d1
2014-07-10Merge r6544 to 3.55 branch.tnakamura@webrtc.org
BUG=3518 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14939004 git-svn-id: http://webrtc.googlecode.com/svn/branches/3.55/webrtc@6655 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01Merge from Chromium at DEPS revision 37.0.2062.10Torne (Richard Coles)
This commit was generated by merge_to_master.py. Change-Id: Ie88672208d018fa891f36c38df5431f6dfdc8de2
2014-07-01Merge third_party/webrtc from ↵Torne (Richard Coles)
https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at b5f5e9089dca995c3d4a6fe0c266d19b8a088b92 This commit was generated by merge_from_chromium.py. Change-Id: I2e4dc9b989394fa4a1948ec6d4975bcfe3b4876b
2014-06-25Create WebRTC 3.55 branch from trunk@6496tnakamura@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/branches/3.55/webrtc@6541 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-25Merge from Chromium at DEPS revision 278856Torne (Richard Coles)
This commit was generated by merge_to_master.py. Change-Id: I06a1e859eaeb2062e3242e0eceedd7f4796c7f00
2014-06-24Update makefiles after merge of Chromium at 278856Torne (Richard Coles)
This commit was generated by merge_from_chromium.py. Change-Id: I7ef3c4fb7c0faa15e95576fefd813d4377106dd7
2014-06-20Merge from Chromium at DEPS revision 278205Torne (Richard Coles)
This commit was generated by merge_to_master.py. Change-Id: I024c837265c4aa34ce7b896a9d3ad9d41767bae0
2014-06-19Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 68f4c7b51ec6434b302de9e97ee01f5ccdb48aa2 This commit was generated by merge_from_chromium.py. Change-Id: Iedf6d850648d6a5904340109e1f71ce52d44113b
2014-06-19Revert 6481 and 6482fgalligan@google.com
Revert 6482 "Update webrtc to fix unpack_lib expansion." Revert 6481 "Update generated asm offsets scripts." The roll has not been successful. Reverted based on the request of the committer. TBR=turaj Review URL: https://webrtc-codereview.appspot.com/17759004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6495 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19Maintain constantness of the input to iSAC-fix decoder, and prevent ↵turaj@webrtc.org
heap-buffer overflow. To save memory in iSAC-fix, decoder operated directly on the recieved bitstream. However, this breaks constantness of input when decoder performed in-place big to little Endian conversion. Furthermore, for bit-streams with odd lengths, this meant writing outside the memory. That is because the last byte will be shifted to the Most Significat Byte which might be outside the allocated memory. If we care about memory, the solution is to do a big-to-little Endian conversion everytime we read a Word16 from the bitstream. BUG=845,chrome:379458 R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6494 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19Adding an empty constructor implementation to the AudioSink classhenrik.lundin@webrtc.org
Turns out it was needed. TBR=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15849004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6493 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19Changes to tests and tools in audio_processing.bjornv@webrtc.org
- Disables ApmTest.EchoCancellationReportsCorrectDelays This test relys completely on the structure of how reported system delays are handled in AEC. In addition it assumes a fix setup of delay logging buffers. This test should be refactored. - Adds flag to turn off reported_delay in audioproc Now it is feasible to turn on and off the use of reported system delays. BUG=N/A R=aluebs@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6492 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19Ensure that the start bitrate can be set multiple times.stefan@webrtc.org
If the start bitrate is set twice, it will be set to the sum of the start bitrates of the currently registered bitrate observers, or left unchanged if the current estimate actually is greater than the sum. BUG=3503 R=henrik.lundin@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15839004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6491 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19Adding test::AudioSink interface and derived classeshenrik.lundin@webrtc.org
The AudioSink interface is supposed to be used by tests that produce audio output. Two implementation classes are also provided: OutputAudioFile: Writes the audio to a pcm file. AudioChecksum: Calculates the MD5 checksum of the audio. These will both be used in future changes. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6490 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19Fixes and re-enables tests disabled on Androidbjornv@webrtc.org
Several tests were disabled in r6325 and r6326. Also, see issue 3445. This CL fixes the remaining four of the audio_processing related ones. Affects the tests: - SystemDelayTest.CorrectDelayAfterStableBufferBuildUp - SystemDelayTest.CorrectDelayDuringDrift - SystemDelayTest.ShouldRecoverAfterGlitch - ApmTest.EchoCancellationReportsCorrectDelays The tests assumes reported delays are used, which now is explicitly set. BUG=3445 TESTED=trybots R=aluebs@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6489 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19Update makefiles after merge of Chromium at 278252Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I2e09453759ef2a9b23eb8b2cf0d92f70acc3ea89
2014-06-18Update webrtc to fix unpack_lib expansion.fgalligan@google.com
Add on fix for:https://webrtc-codereview.appspot.com/12789004/ *NOTE* This CL will break the Android bots as they are built in a Chromium checkout, which will pull in old libvpx DEPS. They will cycle to green when we roll libvpx into Chromium. We must do the rolls in this order because we have to land webrtc and libvpx at the same time into Chromium. BUG=377062 TBR=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6482 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18Update generated asm offsets scripts.fgalligan@google.com
Libvpx updated the unpack scripts to fix building dependencies. Roll libvpx 269083:278063 See https://codereview.chromium.org/295313002/ https://codereview.chromium.org/298063002/ https://codereview.chromium.org/305533008/ https://codereview.chromium.org/305703002/ https://codereview.chromium.org/298383003/ https://codereview.chromium.org/302863004/ https://codereview.chromium.org/320923003/ https://codereview.chromium.org/325313007/ https://codereview.chromium.org/346563002/ for the libvpx changes. See https://codereview.chromium.org/313243004/ for the WebView changes. *NOTE* This CL will break the Android bots as they are built in a Chromium checkout, which will pull in old libvpx DEPS. They will cycle to green when we roll libvpx into Chromium. We must do the rolls in this order because we have to land webrtc and libvpx at the same time into Chromium. BUG=377062 TBR=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12789004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6481 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18Neon version of FilterAdaptation()bjornv@webrtc.org
The performance gain on a Nexus 7 reported by audioproc is ~5.2%. The output is bit exact. Measured total of 15% speed gain on N7 compared to C. R=bjornv@webrtc.org, cd@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17699004 Patch from Scott LaVarnway <slavarnw@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6480 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18Update PacketSource and RtpFileSourcehenrik.lundin@webrtc.org
The NextPacket method should now return NULL when the end of the source was reached. In the RtpFileSource, this means that when the end of file is reached, NULL is returned. Also, when an RTCP packet is encountered, the next packet will be read from file immediately. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20699004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6479 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18Revert "Restore ptypes.txt file"henrik.lundin@webrtc.org
This reverts r6460. It turns out the file was no longer needed after all. BUG=2996 TBR=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6478 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17Revert 6473 "Update generated asm offsets scripts."turaj@webrtc.org
The roll has not been successful. Reverted based on the request of the committer. > Update generated asm offsets scripts. > > Libvpx updated the unpack scripts to fix building dependencies. > > Roll libvpx 269083:277778 > See https://codereview.chromium.org/295313002/ > https://codereview.chromium.org/298063002/ > https://codereview.chromium.org/305533008/ > https://codereview.chromium.org/305703002/ > https://codereview.chromium.org/298383003/ > https://codereview.chromium.org/302863004/ > https://codereview.chromium.org/320923003/ > https://codereview.chromium.org/325313007/ > for the libvpx changes. > > See https://codereview.chromium.org/313243004/ > for the WebView changes. > > *NOTE* This CL will break the Android bots as they are built in a > Chromium checkout, which will pull in old libvpx DEPS. They will > cycle to green when we roll libvpx into Chromium. We must do the > rolls in this order because we have to land webrtc and libvpx at > the same time into Chromium. > > BUG=377062 > TBR=andrew@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/15809004 TBR=fgalligan@google.com Review URL: https://webrtc-codereview.appspot.com/18589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6475 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17Update generated asm offsets scripts.fgalligan@google.com
Libvpx updated the unpack scripts to fix building dependencies. Roll libvpx 269083:277778 See https://codereview.chromium.org/295313002/ https://codereview.chromium.org/298063002/ https://codereview.chromium.org/305533008/ https://codereview.chromium.org/305703002/ https://codereview.chromium.org/298383003/ https://codereview.chromium.org/302863004/ https://codereview.chromium.org/320923003/ https://codereview.chromium.org/325313007/ for the libvpx changes. See https://codereview.chromium.org/313243004/ for the WebView changes. *NOTE* This CL will break the Android bots as they are built in a Chromium checkout, which will pull in old libvpx DEPS. They will cycle to green when we roll libvpx into Chromium. We must do the rolls in this order because we have to land webrtc and libvpx at the same time into Chromium. BUG=377062 TBR=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6473 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17Add round-robin selection of send stream to pad on.stefan@webrtc.org
BUG=1812 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21669004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17Add high perf mode to VP8niklas.enbom@webrtc.org
R=marpan@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14709004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6470 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17base: Renaming + conforming: post commit review changes for ↵henrike@webrtc.org
https://webrtc-codereview.appspot.com/17699005/ BUG=N/A R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20709004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6467 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17Rebase webrtc/base with r6464 version of talk/base:henrike@webrtc.org
cd webrtc/base svn diff -r 6463:6464 http://webrtc.googlecode.com/svn/trunk/talk/base > 6464.diff patch -p0 -i 6464.diff BUG=3379 TBR=phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12749005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6466 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."minyue@webrtc.org
> Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling. > > TEST=passed_all_trybots > R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/16619005 TBR=minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6462 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17Initial GN work for WebRTCkjellander@webrtc.org
This CL makes it possible to build the 'webrtc_base' target using GN. The majority of our GYP stuff in webrtc/build/common.gypi has been translated into the configs of webrtc/BUILD.gn. The webrtc/base/base.gyp file is translated into webrtc/base/BUILD.gn. This CL depends on https://codereview.chromium.org/322373002/ for the jsoncpp BUILD.gn file and the ssl config. To build inside Chromium, https://codereview.chromium.org/321313006/ needs to be landed first. BUG=webrtc:3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true" && ninja -C out/Default I also ran: gn gen out/Default --args="build_with_chromium=false have_dbus_glib=true" but it fails to compile: something is probably wrong with with pkg-config for that. For Chromium, I symlinked src/third_party/webrtc to the webrtc subfolder of the WebRTC checkout and applied the following patches: https://codereview.chromium.org/322373002 (for jsoncpp and ssl config) https://codereview.chromium.org/321313006 (enable building WebRTC) Then I built successfully using: gn gen out/Default && ninja -C out/Default webrtc_base R=brettw@chromium.org TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17669004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6461 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17Restore ptypes.txt filehenrik.lundin@webrtc.org
The file was lost when the neteq folders where moved and renamed. BUG=2996 TBR=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6460 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17Updated W3C getusermedia tests to the latest version of the spec.phoglund@webrtc.org
BUG=webrtc:3455 R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6459 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the ↵minyue@webrtc.org
48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling. TEST=passed_all_trybots R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16619005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6458 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16Makes it possible to prevent some third party libraries (jsoncpp and ↵henrike@webrtc.org
openssl) from being linked. This makes it possible to link webrtc with external implementations of those libraries in case the project depending on webrtc requires another version of those libraries. BUG=3379 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17699005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6455 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16Update makefiles after merge of Chromium at 277521Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I5e2487610f17d8a9d3ac7705201232981d75caf3
2014-06-16Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 847dfa535730a30d57cf26d788d31070b70a02af This commit was generated by merge_from_chromium.py. Change-Id: Id1e94a534a8e364431bcb714b54729e7a410664d
2014-06-16Update makefiles after merge of Chromium at 277428Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I35c59b36614d836accbb543178393a6c061586f1
2014-06-16Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 5fcef2b6df45ceab39ee96a616ab0a4d3c63b83a This commit was generated by merge_from_chromium.py. Change-Id: I58be5a5957c0a6b1be9beac86538af8d38058e9e
2014-06-16Add max limit of number for overuses. When limit is reached always apply the ↵asapersson@webrtc.org
rampup delay. BUG=1577 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6451 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class.asapersson@webrtc.org
BUG=2450 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19559004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6449 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16Remove ivinnichenko from webrtc/test/OWNERSkjellander@webrtc.org
Apparently, We're doing a poor job of cleaning out really old OWNERS. R=phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20669004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6447 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-29 14:33:41 +0000