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2014-11-12Merge from Chromium at DEPS revision 03655fd3f6d7HEADwebview-m40_r4webview-m40_r3webview-m40_r2webview-m40_r1android-m-preview-2android-m-preview-1android-m-previewub-webview-m40-releasemaster-soongmastermainTorne (Richard Coles)
This commit was generated by merge_to_master.py. Change-Id: Ie97de41dee6631b70dd07c00db5bf3ad4dfe8e14
2014-11-12Merge third_party/libjingle/source/talk from ↵Torne (Richard Coles)
https://chromium.googlesource.com/external/webrtc/trunk/talk.git at f97800413f157c911aefbf5be167dbd4806a2323 This commit was generated by merge_from_chromium.py. Change-Id: Ie174f5f2a901346a28e1887ee81dcd22b2fc6dff
2014-11-07Reapply "Advertise G722 as 8 kHz rather than 16 kHz""henrik.lundin@webrtc.org
This reverts r7653 and relands r7645. The reason for the original revert was that G722 disappeared from the SDP offer. This is now fixed. Also, a unit test was updated compared with the original change. BUG=3951 TBR=pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7662 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-07Change dummy address to use 0.0.0.0 instead of ::perkj@webrtc.org
This is to not break compatiblity with FF. https://code.google.com/p/chromium/issues/detail?id=430333 TBR=pthatcher@webrtc.org, juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31979004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7661 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-07Prevent a lot of VideoSendStream reconfigures.pbos@webrtc.org
Checking whether we're setting the same configuration or not. Experimentally this brings down underlying reconfigures from ~20 to about 4-5. R=stefan@webrtc.org BUG=1788 Review URL: https://webrtc-codereview.appspot.com/30909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-07Refactor webrtcvideoengines to have the default list of supported codecs ↵andresp@webrtc.org
being generated in runtime. This will allow to plugin VP9 based on a field trial. R=pbos@webrtc.org, pbos, pthatcher Review URL: https://webrtc-codereview.appspot.com/27949004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7658 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Reland Volume buttons in AppRTCDemo should affect output audio volume (part I).henrika@webrtc.org
Second attempt to land https://webrtc-codereview.appspot.com/32399004/ TBR=perkj@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/30919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7657 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Revert "Advertise G722 as 8 kHz rather than 16 kHz"henrik.lundin@webrtc.org
This reverts r7645. TBR=pthatcher@webrtc.org BUG=3951 Review URL: https://webrtc-codereview.appspot.com/24199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7653 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06(Auto)update libjingle 79326895-> 79329222buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7652 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Volume buttons in AppRTCDemo should affect output audio volume.henrika@webrtc.org
BUG=3279 R=glaznev@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32399004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7651 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Remove deprecated PeerConnection APIs.perkj@webrtc.org
Removes PeerConnectionObserver::OnError. Removes MediaConstraints argument to PeerConnection::AddStream. None of these have ever been implemented and have been removed from the spec. R=tommi@chromium.org Review URL: https://webrtc-codereview.appspot.com/24189004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7650 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Removing unused method GetDefaultVideoEncoderConfig.andresp@webrtc.org
R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7649 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Merge from Chromium at DEPS revision db3f05efe0f9Torne (Richard Coles)
This commit was generated by merge_to_master.py. Change-Id: Ied615f07e5617f89603f29f94846edd58c32a3a9
2014-11-06Merge third_party/libjingle/source/talk from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 7d974c11e23898cd59838c79751b96c45b09ec4b This commit was generated by merge_from_chromium.py. Change-Id: If28216ffbc16e927cf326c567601f482dd2bf36b
2014-11-06(Auto)update libjingle 79285346-> 79320771buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7647 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse ↵mcasas@webrtc.org
adaptation. Also removed some unused "summary" ListPreference fields. The looks of it can be found in [1] (lowest row). [1] https://drive.google.com/file/d/0By6DR2QIwc_ZQm9TMW5YVEpsMWc/view?usp=sharing R=glaznev@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27939004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7646 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06Advertise G722 as 8 kHz rather than 16 kHzhenrik.lundin@webrtc.org
G722 is a 16 kHz (wideband) speech codec, but a "bug" in the RFC has it listed as 8 kHz. This means that the codec should be advertised as 8 kHz in SDP messages. This change fixes that. R=juberti@google.com TBR=pthatcher@webrtc.org BUG=3951 TEST=Verify that the G722 is advertised as a=rtpmap:9 G722/8000, not /16000. Review URL: https://webrtc-codereview.appspot.com/27879004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7645 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05This fixes a small memory leak (found using Xcode/Instruments on iOS) intkchin@webrtc.org
the ObjC bindings of PeerConnection. The generated session description has to be released by the recipient BUG=3985 R=tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28959004 Patch from Matthias Liebig <matthias.gcode@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7636 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05Wire up bandwidth stats to the new API and webrtcvideoengine2.stefan@webrtc.org
Adds stats to verify bandwidth and pacer stats. BUG=1788 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24969004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05(Auto)update libjingle 79205306-> 79244016buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7633 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05(Auto)update libjingle 79200114-> 79205306buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7627 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Merge third_party/libjingle/source/talk from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 19c8d5c35a3e7a7341124f3865a3a117985e7c08 This commit was generated by merge_from_chromium.py. Change-Id: I75181570213a02c85bd866a22896805d4b1f966f
2014-11-04Cleanup RTCVideoRenderer interface.tkchin@webrtc.org
RTCVideoRenderer should be a protocol not a class. This change includes an adapter for use with the C++ apis. The video views have been refactored to implement that protocol. BUG=3795 R=glaznev@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7626 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04(Auto)update libjingle 79169148-> 79192489buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7624 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04AppRTCDemoActivity: use differnet Themes for different API levelsmcasas@webrtc.org
The current AndroidManifest.xml hardcodes a Theme that is only available in Android L or later (Material). To maintain backwards compatibility, and for better App style, create a single Theme/Style and define it for different APIs. I tested this in two Nexus %, one with prerelease L and another with a KK 4.4.2 and the Theme is indeed automagically updated :) Note that this is compatible with https://webrtc-codereview.appspot.com/26979004/ R=glaznev@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26019004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7619 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Remove protected files from talk/PRESUBMIT.py.pbos@webrtc.org
All files may now be committed to. R=pthatcher@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/23359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7616 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Falling back on single-stream on multiple SSRC.pbos@webrtc.org
Instead of failing, use one stream. Also clamp video min bitrate. R=stefan@webrtc.org BUG=1788 Review URL: https://webrtc-codereview.appspot.com/31949004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7615 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Merge third_party/libjingle/source/talk from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 4fb0d5af90819ec9666b847f6295b933f58c8301 This commit was generated by merge_from_chromium.py. Change-Id: I9300049afc40d0a2f93ff9ffcb144ca3c68863d7
2014-11-04ReAdd PeerConnectionInterface::AddStream to fix Chrome build.perkj@webrtc.org
AddStream(MediaStreamInterface* stream, const MediaConstraintsInterface* constraints); This will be removed once Chrome has been updated. R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7608 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Change the PeerConnection proxy templates to use blocking method calls ↵tommi@webrtc.org
instead of using Thread::Send. The problem with Thread::Send is that it processes incoming pending messages and for the proxies, this can mean that multiple incoming calls can concurrently run on the same thread, resulting in unexpected behavior. See e.g. crbug.com/429740 (and more) R=perkj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28939004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7607 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04Prepare for removal of PeerConnectionObserver::OnError.perkj@webrtc.org
Prepare for removal of constraints to PeerConnection::AddStream. OnError has never been implemented and has been removed from the spec. Also, constraints to PeerConnection::AddStream has also been removed from the spec and have never been implemented. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23319004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7605 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03(Auto)update libjingle 79104430-> 79104922buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7602 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Android AppRTCDemo improvements:glaznev@webrtc.org
- Add a room list to ConnectActivity with buttons to add/remove rooms. - Add loopback call button. - Add option to toggle full screen / letterbox video. - Add camera fps settings. - Fix device to landscape orientation for HD video until issue 3936 will be fixed. - Fix a few crashes by avoiding calling peer connection and GAE signaling function while connection is closing. - Better handling GAE channel error - catch channel exceptions and display dialog with error messages. BUG=3939, 3935 R=kjellander@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26979004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7601 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Implement external decoder support in WebRtcVideoEngine2.pbos@webrtc.org
R=stefan@webrtc.org BUG=1788 Review URL: https://webrtc-codereview.appspot.com/30839004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7594 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03Disable PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate under MSanhenrik.lundin@webrtc.org
This test is flaky on MSan bots. BUG=3980 TBR=jiayl@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7591 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Update Android projects to API level 21.kjellander@webrtc.org
The update in https://webrtc-codereview.appspot.com/23309004 was not enough, so this updates to 21 instead. This is required in order to roll chromium_revision to keep up with Chrome, as third_party/android_tools have now dropped support for API level 20. Commands used: third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-21 --path webrtc/examples/android/opensl_loopback third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-21 --path webrtc/examples/android/media_demo/ third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-21 --path talk/examples/android/ Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when build/android/envsetup.sh is sourced. BUG= R=glaznev@webrtc.org, henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7587 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Change default JVM location to /usr/lib/jvm/java-7-openjdk-amd64kjellander@webrtc.org
Given that OpenJDK 1.7 is the recommended Java SDK for Chromium these days, we should get rid of linking to the old non-standardized link referring to a Sun Java 1.6 SDK. Instead of requiring all users to set JAVA_HOME, I prefer have the most common path as default and and close webrtc:2113 as won't fix after this is submitted. BUG=2113 R=henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29839004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7584 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Update all .isolate files for the new format.kjellander@webrtc.org
R=kjellander@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/27809004 Patch from Marc-Antoine Ruel <maruel@chromium.org>. git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Update Android projects to API level 20.kjellander@webrtc.org
This is required in order to roll chromium_revision to keep up with Chrome, as third_party/android_tools have now dropped support for API level 19. Commands used: third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-20 --path webrtc/examples/android/opensl_loopback third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-20 --path webrtc/examples/android/media_demo/ third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-20 --path talk/examples/android/ Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when build/android/envsetup.sh is sourced. BUG= R=glaznev@webrtc.org, henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23309004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7582 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Merge third_party/libjingle/source/talk from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 5910fdfb2ca4b12096bdc5c64aee0afc7f93d426 This commit was generated by merge_from_chromium.py. Change-Id: Ib80d377fe0896551345c165d3b3809b4f66ded80
2014-10-31Implement conference-mode temporal-layer screencast.pbos@webrtc.org
Renames VideoStream::temporal_layers to temporal_layer_thresholds_bps to convey that it contains thresholds needed to ramp up between them (1 threshold -> 2 temporal layers, etc.). R=mflodman@webrtc.org, stefan@webrtc.org BUG=1788,1667 Review URL: https://webrtc-codereview.appspot.com/23269004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7578 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Configure A/V sync in WebRtcVideoEngine2.pbos@webrtc.org
Sets up A/V sync for the first video receive channel with the default voice channel. This is only done when conference mode is disabled to preserve existing behavior. Ideally we'd know which voice channel to sync with here. R=mflodman@webrtc.org, stefan@webrtc.org BUG=1788 Review URL: https://webrtc-codereview.appspot.com/23249004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7577 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31Adapting bitrate according to maxplaybackrate for Opus.minyue@webrtc.org
BUG= R=mflodman@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29929004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7575 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31arm64 iOS build.tkchin@webrtc.org
Allows successful build of arm64 libraries using GYP_DEFINES="OS=ios target_arch=arm64 target_subarch=arm64". Note that not all libraries will be NEON optimized (eg common_audio), however most importantly libvpx will be. WEBRTC_ARCH_ARM needs to be defined so that libvpx doesn't post-process, which is significantly detrimental to performance. BUG=3898 R=kjellander@webrtc.org, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7573 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30Improve the logging when a TCP connection is deleted.jiayl@webrtc.org
BUG= R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7572 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30Cleaning up r7562-7567.minyue@webrtc.org
Wrongly used git svn dcommit for committing a CL. Then two reverts were applied. Still something needs to be cleaned. BUG= TBR=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23299004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7568 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30(Auto)update libjingle 78822708-> 78823675buildbot@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7567 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30Revert 7563 "before rebase" due to wrong submissionminyue@webrtc.org
> before rebase TBR=minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31859004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7566 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30Revert 7564 "to submit" due to wrong submissionminyue@webrtc.org
> to submit TBR=minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23289004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7565 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30to submitminyue@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7564 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-06 01:36:31 +0000