0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
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07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use suffixed {uint,int}{8,16,32,64}_t types. Removes the use of uint8, etc. in favor of uint8_t. BUG=webrtc:5024 R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1362503003 . Cr-Commit-Position: refs/heads/master@{#10196}
/external/webrtc/talk/app/webrtc/test/fakeaudiocapturemodule.h
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c14f5ff60fb0c42c97702de112a9e8f1eccba574 |
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23-Sep-2015 |
henrika <henrika@webrtc.org> |
Improving support for Android Audio Effects in WebRTC. Now also supports AGC and NS effects and adds the possibility to override default settings. R=magjed@webrtc.org, pbos@webrtc.org, sophiechang@chromium.org TBR=perkj BUG=NONE Review URL: https://codereview.webrtc.org/1344563002 . Cr-Commit-Position: refs/heads/master@{#10030}
/external/webrtc/talk/app/webrtc/test/fakeaudiocapturemodule.h
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dce40cf804019a9898b6ab8d8262466b697c56e0 |
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24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Update a ton of audio code to use size_t more correctly and in general reduce 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}
/external/webrtc/talk/app/webrtc/test/fakeaudiocapturemodule.h
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ee8c6d327357ecd2e17edede8d15f6e3893409a8 |
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13-Aug-2015 |
deadbeef <deadbeef@webrtc.org> |
In PeerConnectionTestWrapper, put audio input on a separate thread. This will prevent it from blocking network input when it falls behind, which is happening when running with ThreadSanitizer. BUG=webrtc:4663 Review URL: https://codereview.webrtc.org/1236023010 Cr-Commit-Position: refs/heads/master@{#9707}
/external/webrtc/talk/app/webrtc/test/fakeaudiocapturemodule.h
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f045e4da43e671ae511aa1d9b6ef2968256a745d |
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11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Prepare to convert various types to size_t. This makes some behaviorally-invariant changes to make certain code that currently only works correctly with signed types work safely regardless of the signedness of the types in question. This is preparation for a future change that will convert a variety of types to size_t. There are also some formatting changes (e.g. converting "enum hack" usage to real consts) to make it simpler to just change "int" to "size_t" in the future to change the types of those constants. BUG=none R=andrew@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org TBR=ajm Review URL: https://codereview.webrtc.org/1174813003 Cr-Commit-Position: refs/heads/master@{#9413}
/external/webrtc/talk/app/webrtc/test/fakeaudiocapturemodule.h
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14665ff7d4024d07e58622f498b23fd980001871 |
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04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro Clang version changed 223108:230914 Details: https://chromium.googlesource.com/chromium/src/+/e144d30..6fdb142/tools/clang/scripts/update.sh Removes the OVERRIDE macro defined in: * webrtc/base/common.h * webrtc/typedefs.h The majority of the source changes were done by running this in src/: perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"` which converted all: virtual Foo() OVERRIDE functions to: Foo() override Then I manually edited: * talk/media/webrtc/fakewebrtccommon.h * webrtc/test/fake_common.h Remaining uses of OVERRIDE was fixed by search+replace. Manual edits were done to fix virtual destructors that were overriding inherited ones. Finally a build error related to the pure virtual definitions of Read, Write and Rewind in common_types.h required a bit of refactoring in: * webrtc/common_types.cc * webrtc/common_types.h * webrtc/system_wrappers/interface/file_wrapper.h * webrtc/system_wrappers/source/file_impl.cc This roll should make it possible for us to finally re-enable deadlock detection for TSan on the buildbots. BUG=4106 R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41069004 Cr-Commit-Position: refs/heads/master@{#8596} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/test/fakeaudiocapturemodule.h
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4161715e3f7e744bc9ef3d3ae437da1e8e4de38d |
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29-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Remove ChangeUniqueID. This fixes a two year old TODO of deleting dead code :) In cases where the _id or id_ member variable is being used for tracing, I changed the member to at least be const. It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them. BUG= R=henrika@webrtc.org, perkj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37849004 Cr-Commit-Position: refs/heads/master@{#8201} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/test/fakeaudiocapturemodule.h
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5f93d0a140515e3b8cdd1b9a4c6f5871144e5dee |
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20-Jan-2015 |
jlmiller@webrtc.org <jlmiller@webrtc.org> |
Update libjingle license statements at top of talk files for consistency BUG=2133 R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/39559004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/test/fakeaudiocapturemodule.h
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0b1534c52eab79372557a6d81aaf4dd9407f55d3 |
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15-Dec-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. This fixes a variety of MSVC warnings about value truncations when implicitly storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and removes the need for a number of explicit casts. This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack". BUG=chromium:81439 TEST=none R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/test/fakeaudiocapturemodule.h
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a954c07ee1c93175e6ebbeb20517b347474362ae |
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09-Dec-2014 |
henrika@webrtc.org <henrika@webrtc.org> |
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer BUG=4034 R=andrew@webrtc.org, perkj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/test/fakeaudiocapturemodule.h
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1972ff8a6e45f7ad3fb7e4ed51dc0135c72f6c9d |
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11-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE. This will make a subsequent change I intend to do safer, where I'll change the return type of one of the base Module functions, by breaking the compile if I miss any overrides. This also highlighted a number of unused functions (in many cases apparently virtual "overrides" of no-longer-existent base functions). I've removed some of these. This also highlighted several cases where "virtual" was used unnecessarily to mark a function that was only defined in one class. Removed "virtual" in those cases. BUG=none TEST=none R=andrew@webrtc.org, henrik.lundin@webrtc.org, mallinath@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7146 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/test/fakeaudiocapturemodule.h
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d4e598d57aed714a599444a7eab5e8fdde52a950 |
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29-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72097588-> 72159069 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/test/fakeaudiocapturemodule.h
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8804a29951bfeaf97a0964aa90ec69ac17820752 |
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23-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add CriticalSection to fakeaudiocapturemodule to protect the variables which will be accessed from process_thread_ and the main thread. TEST=try bots BUG=1205 R=henrike@webrtc.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5019 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/test/fakeaudiocapturemodule.h
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28e20752806a492f5a6a5d343c02f9556f39b1cd |
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10-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/test/fakeaudiocapturemodule.h
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