ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
|
04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames This changes the following module directories: * webrtc/modules/audio_conference_mixer/interface * webrtc/modules/interface * webrtc/modules/media_file/interface * webrtc/modules/rtp_rtcp/interface * webrtc/modules/utility/interface To avoid breaking downstream, I followed this recipe: 1. Copy the interface dir to a new sibling directory: include 2. Update the header guards in the include directory to match the style guide. 3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code. 4. Add a pragma warning in the header files in the interface dir. Example: #pragma message("WARNING: webrtc/modules/interface is DEPRECATED; " "use webrtc/modules/include") 5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S) 6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*) BUG=5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417683006 . Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
|
dce40cf804019a9898b6ab8d8262466b697c56e0 |
|
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/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
|
728d9037c016c01295177fa700fc7927f0bb80bb |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reformat existing code. There should be no functional effects. This includes changes like: * Attempt to break lines at better positions * Use "override" in more places, don't use "virtual" with it * Use {} where the body is more than one line * Make declaration and definition arg names match * Eliminate unused code * EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT) * Correct #include order * Use anonymous namespaces in preference to "static" for file-scoping * Eliminate unnecessary casts * Update reference code in comments of ARM assembly sources to match actual current C code * Fix indenting to be more style-guide compliant * Use arraysize() in more places * Use bool instead of int for "boolean" values (0/1) * Shorten and simplify code * Spaces around operators * 80 column limit * Use const more consistently * Space goes after '*' in type name, not before * Remove unnecessary return values * Use "(var == const)", not "(const == var)" * Spelling * Prefer true, typed constants to "enum hack" constants * Avoid "virtual" on non-overridden functions * ASSERT(x == y) -> ASSERT_EQ(y, x) BUG=none R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1172163004 Cr-Commit-Position: refs/heads/master@{#9420}
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
|
1cec3957b88cbab345535137329bd8f3f2a6b39e |
|
12-May-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. BUG=3206 R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6102 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
|
0a38432ea580f04ec0cbb8cd73f7bee84cc6547b |
|
04-Jun-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix error in mixing test for supported sample rates. With the switch to an arbitrary resampler, we now support these strange rates. TBR=turaj Review URL: https://webrtc-codereview.appspot.com/1604004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4158 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
|
956aa7e0874f2e08c335a82a2c32f400fac8b031 |
|
21-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include files from webrtc/.. paths in voice_engine/ BUG=1662 R=henrikg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1434005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4079 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
|
1b60ceb499ee35460886f2bfaecee1f47319f925 |
|
13-Dec-2012 |
roosa@google.com <roosa@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add GetAudioFrame API to VoiceEngine. Allows the caller to pull frames from a channel instead of sending them to the output mixer. BUG= Review URL: https://webrtc-codereview.appspot.com/973012 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3273 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
|
14b43beb7ce4440b30dcea31196de5b4a529cb6b |
|
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/ TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
|