summaryrefslogtreecommitdiff
AgeCommit message (Collapse)Author
2014-09-30Merge from Chromium at DEPS revision 267aeeb8d85candroid-cts-5.1_r9android-cts-5.1_r8android-cts-5.1_r7android-cts-5.1_r6android-cts-5.1_r5android-cts-5.1_r4android-cts-5.1_r3android-cts-5.1_r28android-cts-5.1_r27android-cts-5.1_r26android-cts-5.1_r25android-cts-5.1_r24android-cts-5.1_r23android-cts-5.1_r22android-cts-5.1_r21android-cts-5.1_r20android-cts-5.1_r2android-cts-5.1_r19android-cts-5.1_r18android-cts-5.1_r17android-cts-5.1_r16android-cts-5.1_r15android-cts-5.1_r14android-cts-5.1_r13android-cts-5.1_r10android-cts-5.1_r1android-5.1.1_r9android-5.1.1_r8android-5.1.1_r7android-5.1.1_r6android-5.1.1_r5android-5.1.1_r4android-5.1.1_r38android-5.1.1_r37android-5.1.1_r36android-5.1.1_r35android-5.1.1_r34android-5.1.1_r33android-5.1.1_r30android-5.1.1_r3android-5.1.1_r29android-5.1.1_r28android-5.1.1_r26android-5.1.1_r25android-5.1.1_r24android-5.1.1_r23android-5.1.1_r22android-5.1.1_r20android-5.1.1_r2android-5.1.1_r19android-5.1.1_r18android-5.1.1_r17android-5.1.1_r16android-5.1.1_r15android-5.1.1_r14android-5.1.1_r13android-5.1.1_r12android-5.1.1_r10android-5.1.1_r1android-5.1.0_r5android-5.1.0_r4android-5.1.0_r3android-5.1.0_r1lollipop-mr1-wfc-releaselollipop-mr1-releaselollipop-mr1-fi-releaselollipop-mr1-devlollipop-mr1-cts-releasePrimiano Tucci
This commit was generated by merge_to_master.py. Change-Id: I3cccc8f04ad0036aecdb7eefe316a059ebcefaf9
2014-09-25Merge third_party/libjingle/source/talk from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 40539b82d5a2c9bcf23d078e997ce0368160f5a3 This commit was generated by merge_from_chromium.py. Change-Id: I01a4e13d3b66df627fcc0992da660f11cfe85646
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/talk@7290 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24Merge third_party/libjingle/source/talk from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 47740f2c26aea1b3b7830abdcba063a12a61d009 This commit was generated by merge_from_chromium.py. Change-Id: I716b47abf872803a836ee46221e5d100dfbc3984
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/talk@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/talk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23Change Android video renderer to maintain video aspectglaznev@webrtc.org
ratio when displaying camera or decoded video frames. - R=tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7282 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23Switch HW video decoder to output byte buffers if videoglaznev@webrtc.org
renderer EGL context is not provided by app. R=tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24699004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7281 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23(Auto)update libjingle 76169599-> 76176062buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7280 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23Enable ipv6 by default for webrtc under a Finch experiment.guoweis@webrtc.org
Reapply 23529005 after fixing the build break issue (Chromium:582133002) Committed: https://code.google.com/p/webrtc/source/detail?r=7253 Review URL: https://webrtc-codereview.appspot.com/23529005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7278 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/talk@7277 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23Remove DTMF status methods from Voice Enginehenrik.lundin@webrtc.org
These methods are not used. R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24689004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7276 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23Skeleton for registering external encoders/decoders.pbos@webrtc.org
R=pthatcher@webrtc.org BUG=1788 Review URL: https://webrtc-codereview.appspot.com/31429005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7270 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22Remove engine-level SetOptions.pbos@webrtc.org
Already removed in WebRtcVideoEngine. R=andresp@webrtc.org BUG=1788 Review URL: https://webrtc-codereview.appspot.com/29549005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7265 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22Remove Get/SetNetEQPlayoutMode APIshenrik.lundin@webrtc.org
These are not used anymore. R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7262 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19Reapply 23529005 after fixing the build break issue (Chromium:582133002)guoweis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23529005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7253 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19(Auto)update libjingle 75925673-> 75926712buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7252 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19(Auto)update libjingle 75924589-> 75925673buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7251 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19(Auto)update libjingle 75922684-> 75924589buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7250 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19Fix HW video decoder crash on some Android KK devices.glaznev@webrtc.org
Remove direct access to decoder Java output buffer memory when HW decoder is configured to decode to surface. - R=tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30459005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7249 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19Fix the libjingle_media_unittest failure in Windows build by modifying ↵thorcarpenter@google.com
libjingle_tests.gyp and sctpdataengine_unittests.cc instead of ssladapter.cc. R=harryjin@google.com, pthatcher@webrtc.org, tpsiaki@google.com Review URL: https://webrtc-codereview.appspot.com/22699004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7245 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19Fixing compilation failure in peerconnection_jni.cc with WEBRTC_CHROMIUM_BUILD.glaznev@webrtc.org
Symbol LogcatTraceContext not defined. Submitting on behalf of serya@. Dup of https://webrtc-codereview.appspot.com/29529004/ TEST=Build target libjingle_peerconnection_javalib with applied CL https://codereview.chromium.org/551793003/ BUG=https://crbug.com/383418 R=serya@chromium.org Review URL: https://webrtc-codereview.appspot.com/28529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7244 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19Config struct for VideoEncoder.pbos@webrtc.org
Used for config parameters in common between multiple codecs as well as the encoder-specific pointer. In particular this contains content mode (realtime video vs. screenshare). BUG=1788 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16319004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7239 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19Merge third_party/libjingle/source/talk from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/talk.git at b25f2cd3bd9b8444d2a1d48ca26e2721b42c78e1 This commit was generated by merge_from_chromium.py. Change-Id: I3c55437f08741cdd874f85672c334aaf12ab2fd2
2014-09-19(Auto)update libjingle 75875619-> 75878731buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7235 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19(Auto)update libjingle 75865376-> 75875619buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7234 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19(Auto)update libjingle 75854833-> 75865376buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7233 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18(Auto)update libjingle 75854418-> 75854833buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7232 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18(Auto)update libjingle 75852725-> 75853560buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7231 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18A few fixes to avoid crash in HW codec on device orientation change.glaznev@webrtc.org
- Fix video encoder Reset() function to avoid setting codec resolution to zero. - Follow SW codec implementation and do not crash when frame with the resolution different from the encoder resolution arrives. Instead wait for at least 3 frames with new resolution and re-initialize the codec. HW codec reset may take much longer than SW codec, so these 3 frames threshold avoids resetting codec when outstanding camera frame captured from previous device orientation arrives. - Plus some minor changes to make encoder reset/release implementation closer to decoder implementation. BUG= R=tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30439004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7230 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18Revert maximum video codec resolution on Android back to 720p again.glaznev@webrtc.org
Some low end Android devices still have problems with 1080p support. BUG=3757 R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7228 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18(Auto)update libjingle 75818332-> 75837294buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7227 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18Avoid writing a double/float to a string to avoid a crash.jiayl@webrtc.org
BUG=crbug/367223 R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7225 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18Expose VP8/H264 defaults through video_encoder.h.pbos@webrtc.org
Reduces code duplication quite a bit, these identical defaults were set in quite a few different places. R=mflodman@webrtc.org, stefan@webrtc.org BUG=3070 Review URL: https://webrtc-codereview.appspot.com/19299004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7220 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18Split video_render_module implementation into default and internal ↵andresp@webrtc.org
implementation. Targets must now link with implementation of their choice instead of at "gyp"-time. Targets linking with libjingle_media: - internal implementation when build_with_chromium=0, default otherwise. Targets linking with default render implementation: - video_engine_tests - video_loopback - video_replay - anything dependent on webrtc_test_common Targets linking with internal render implementation: - vie_auto_test - video_render_tests - libwebrtcdemo-jni - video_engine_core_unittests GN changes: - Not many since there is almost no test definitions. Work-around for chromium: - Until chromium has updated libpeerconnection to link with video_capture_impl and video_render_impl, webrtc target automatically depends on it. This should fix the FYI bots and not require a webrtc roll to fix. Re-enable android tests by reverting 7026 (some tests left disabled). TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in. BUG=3770 R=kjellander@webrtc.org, pbos@webrtc.org TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7217 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17Implemented Network::GetBestIP() selection logic as following.guoweis@webrtc.org
1) return the first global temporary and non-deprecrated ones. 2) if #1 not available, return global one. 3) if #2 not available, use ULA ipv6 as last resort. ULA stands for unique local address. They are only useful in a private WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address BUG=3808 At this point, rule #3 actually won't happen at current implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway. R=jiayl@webrtc.org Committed: https://code.google.com/p/webrtc/source/detail?r=7200 Committed: https://code.google.com/p/webrtc/source/detail?r=7201 Review URL: https://webrtc-codereview.appspot.com/31369004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7216 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17Enable HW video decoding on Qualcomm devices.glaznev@webrtc.org
Parallel decoding and encoding problem is fixed now (b/16353967), so it is possible to start using Qualcomm VP8 HW decoder. Bitrate overshoots should be fixed as well. BUG= R=tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7215 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17talk/p2p/base: removed unused variable "port_"henrike@webrtc.org
BUG=N/A R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7212 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17Split video_capture_module specific implementation (external vs internal ↵andresp@webrtc.org
capture) into its own targets. Dependencies must link directly with the desired one. Targets linking with libjingle_media: - internal implementation when build_with_chromium=0, default otherwise. Targets linking with default/external capture implementation: - anything dependent on webrtc_test_common - anything dependent on video_engine_core Targets linking with internal capture implementation: - vie_auto_test - anything dependent on webrtc_test_renderer GN changes: - Not many since there is almost no test definitions. TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in. BUG=3768 R=glaznev@webrtc.org TBR=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7209 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17Split video engine android initialization into each internal module ↵andresp@webrtc.org
initialization. This is to later on allow targets to pick at link time if to include the external or internal implementation. In order to do that the video_engine cannot compile different based on which option is picked later on. BUG=3768,3770 R=glaznev@webrtc.org, stefan@webrtc.org TBR=henrike@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7208 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""pbos@webrtc.org
Re-lands r7114 after landing r7204 to adress the compile error causing the rollback in r7151. BUG=3070 TBR=henrikg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7207 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17(Auto)update libjingle 75683337-> 75695882buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7206 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17Java VideoRenderer class may be backed by two different nativeglaznev@webrtc.org
classes depending on type of rendering. Fix crash in AppRtcDemo by calling correct destructor on exit. BUG= R=braveyao@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23699004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16Implemented Network::GetBestIP() selection logic as following.guoweis@webrtc.org
1) return the first global temporary and non-deprecrated ones. 2) if #1 not available, return global one. 3) if #2 not available, use ULA ipv6 as last resort. ULA stands for unique local address. They are only useful in a private WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address BUG=3808 At this point, rule #3 actually won't happen at current implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway. R=jiayl@webrtc.org Committed: https://code.google.com/p/webrtc/source/detail?r=7200 Review URL: https://webrtc-codereview.appspot.com/31369004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7201 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16Implemented Network::GetBestIP() selection logic as following.guoweis@webrtc.org
1) return the first global temporary and non-deprecrated ones. 2) if #1 not available, return global one. 3) if #2 not available, use ULA ipv6 as last resort. ULA stands for unique local address. They are only useful in a private WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address BUG=3808 At this point, rule #3 actually won't happen at current implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway. R=jiayl@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31369004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7200 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16Recreate VideoStreams when setting resolution.pbos@webrtc.org
Instead of just changing resolution on the last stream streams are reallocated to make sure that all streams are updated to match the new input resolution. R=pthatcher@webrtc.org BUG=1788 Review URL: https://webrtc-codereview.appspot.com/29469004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7197 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16Add pbos@webrtc.org (myself) to talk/media/webrtc/.pbos@webrtc.org
Allows easier reviews of webrtcvideoengine2.cc and landing the new video API on shorter review cycles. R=pthatcher@webrtc.org BUG=1788 Review URL: https://webrtc-codereview.appspot.com/30409004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7196 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16(Auto)update libjingle 75610402-> 75610402buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7194 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16Merge third_party/libjingle/source/talk from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 1f59bcb2ae6b867fb2f52ff4654b137f98b30536 This commit was generated by merge_from_chromium.py. Change-Id: Ic6f021ebece4c960a234811c11205565441e01dc
2014-09-16Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..."kjellander@webrtc.org
Breaks Chrome build and prevents rolling WebRTC into Chrome DEPS. > Enable ipv6 by default for webrtc under a Finch experiment. > > BUG=413437 (chromium) > https://code.google.com/p/chromium/issues/detail?id=413437 > > Review URL: https://webrtc-codereview.appspot.com/23529005 TBR=guoweis@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30399004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7190 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-26 01:59:23 +0000