ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
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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/tools/agc/test_utils.cc
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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/webrtc/tools/agc/test_utils.cc
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a33f05e8d7f293b5984b3cd7695eadefd16dcaba |
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29-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Re-land "Remove <(webrtc_root) from source file entries." Changes differing from https://webrtc-codereview.appspot.com/37859004: * I put the include_tests==1 stuff of audio_coding.gypi in its own audio_coding_tests.gypi file, including the Android and isolate targets which were incorrectly located in the previous CL * I moved the bwe utilities in remote_bitrate_estimator.gypi into include_tests==1 since they depend on test.gyp after I cleaned up the duplicated inclusion of rtp_file_reader.cc R=stefan@webrtc.org TBR=tina.legrand@webrtc.org TESTED=Passing gyp and compile using: webrtc/build/gyp_webrtc -Dinclude_tests=1 webrtc/build/gyp_webrtc -Dinclude_tests=0 I also setup a Chromium checkout with my checkout mounted in third_party/webrtc and ran build/gyp_chromium successfully. BUG=4185 Review URL: https://webrtc-codereview.appspot.com/33159004 Cr-Commit-Position: refs/heads/master@{#8205} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8205 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/tools/agc/test_utils.cc
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1ece0cbbec63d4fc14e3a6121d3d828c250fc20f |
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29-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Revert "Remove <(webrtc_root) from source file entries." And the follow-up fix in r8198 that was not sufficient. Reason: breaks Chromium bots runhooks (GYP). I will have to try some more to make sure I don't include test code, since include_tests==0 in Chromium. TBR=andrew@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37039004 Cr-Commit-Position: refs/heads/master@{#8200} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8200 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/tools/agc/test_utils.cc
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2d2a1f9f056bb552e725b70b863d31cbee5ef7d8 |
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29-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Remove <(webrtc_root) from source file entries. This required to move the AGC tools source files into webrtc/tools and create a new agc_test_utils target. Since audio_codec_speed_tests.gypi referenced sources above, the best approach I could come up with was to add an audio_coding.gypi file at a higher level and move the targets in there (+ the includes from modules.gyp which is an improvement IMO). I also added a PRESUBMIT.py check to prevent new source entries being added with <(webrtc_root) in the path. BUG=4185 R=andrew@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37859004 Cr-Commit-Position: refs/heads/master@{#8197} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8197 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/tools/agc/test_utils.cc
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