The reason is that the desktop apprtcdemo only handle one MediaStream and this doesn't play audio if it receive two streams.
TEST=Test that a call with audio and video can be setup between an Android device and a desktop client using apprtc.appspot.com.
BUG=4051,3786
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7781 4adac7df-926f-26a2-2b94-8c16560cd09d
This reverts commit 308e7ff613.
Original cl description:
This adds an Android apk for running tests on the Java layer of PeerConnection.
The only testcase is currently the same test we run on Java standalone.
To run the test adb shell am instrument -w org.webrtc.test/android.test.InstrumentationTestRunner
BUG=4031
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7748 4adac7df-926f-26a2-2b94-8c16560cd09d
These parameters are set by the x-google-*-bitrate SDP parameters. This
is implemented on a Call level instead of per-stream like the currently
underlying VideoEngine implementation to allow this refactoring to not
reconfigure the VideoCodec at all but rather adjust bandwidth-estimator
parameters.
Also implements SetMaxSendBandwidth in WebRtcVideoEngine2 as it's a SDP
parameter and allowing it to be dynamically readjusted in Call.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/26199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7746 4adac7df-926f-26a2-2b94-8c16560cd09d
for Android AppRTCDemo when using WebSocket signaling.
- Add loopback support for new signaling. In loopback mode
only room connection is established, WebSocket connection is
not opened and all candidate/sdp messages are automatically
routed back.
- Fix command line support both for channek and new signaling.
Exit from application when room connection is closed and add
an option to run application for certain time period and exit.
- Plus some fixes for WebSocket signaling - support
POST (not used for now) and DELETE request to WebSocket server
and making sure that all available TURN server are used by
peer connection client.
BUG=3995,3937
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7725 4adac7df-926f-26a2-2b94-8c16560cd09d
The purpose of this CL is to add a new file in libjingle without breaking Chromium in the process. The plan is to do the following:
1. Land a no-op videoframefactory.cc in webrtc (this file).
2. Wait for it to roll into Chromium.
3. Modify libjingle.gyp in Chromium to include this file.
4. Make the real change in webrtc with the real implementation of this file.
5. Wait for the change to roll into Chromium.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7712 4adac7df-926f-26a2-2b94-8c16560cd09d
This didn't compile on the FYI bots. Example error:
FAILED: E:\b\depot_tools\python276_bin\python.exe gyp-win-tool link-with-manifests environment.x86 True chrome_child.dll "E:\b\depot_tools\python276_bin\python.exe gyp-win-tool link-wrapper environment.x86 False link.exe /nologo /IMPLIB:chrome_child.dll.lib /DLL /OUT:chrome_child.dll @chrome_child.dll.rsp" 2 mt.exe rc.exe "obj\chrome\chrome_child_dll.chrome_child.dll.intermediate.manifest" obj\chrome\chrome_child_dll.chrome_child.dll.generated.manifest
content_renderer.lib(content_renderer.webrtc_video_capturer_adapter.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)
libjingle_webrtc_common.lib(libjingle_webrtc_common.peerconnectionfactory.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)
libjingle_webrtc_common.lib(libjingle_webrtc_common.videocapturer.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)
libjingle_webrtc_common.lib(libjingle_webrtc_common.dummydevicemanager.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)
chrome_child.dll : fatal error LNK1120: 1 unresolved externals
> cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
>
> In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
>
> This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
>
> R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
>
> Committed: https://code.google.com/p/webrtc/source/detail?r=7702
>
> Review URL: https://webrtc-codereview.appspot.com/29949004TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7708 4adac7df-926f-26a2-2b94-8c16560cd09d
In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7702
Review URL: https://webrtc-codereview.appspot.com/29949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7707 4adac7df-926f-26a2-2b94-8c16560cd09d
Rease for revert: failed internal test cases
> cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
>
> In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
>
> This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
>
> R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/29949004TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7703 4adac7df-926f-26a2-2b94-8c16560cd09d
In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7702 4adac7df-926f-26a2-2b94-8c16560cd09d
Previously DataChannel::SendQueuedDataMessages continues the loop of sending queued messages if the channel is blocked, which will cause message reordering if the channel becomes unblocked during the loop, i.e. messages attempted after the unblocking will be sent earlier than the older messages attempted before the unblocking.
BUG=3979
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7690 4adac7df-926f-26a2-2b94-8c16560cd09d
Addresses issue where SetDefaultEncoderConfig modifies the codec list
rather than just the targeted codec. This was previously done just to
pass more unit tests rather than be done properly. This incidentally
addresses a TODO causing this to work with external codecs as well.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/32009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7667 4adac7df-926f-26a2-2b94-8c16560cd09d
- 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@7601 4adac7df-926f-26a2-2b94-8c16560cd09d
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@7587 4adac7df-926f-26a2-2b94-8c16560cd09d
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@7584 4adac7df-926f-26a2-2b94-8c16560cd09d
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@7582 4adac7df-926f-26a2-2b94-8c16560cd09d
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@7573 4adac7df-926f-26a2-2b94-8c16560cd09d
Removes direct VideoCodec use from the new API, exposes VideoDecoders
through webrtc/video_decoder.h similar to VideoEncoders.
Also includes some preparation for wiring up external decoders in
WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they
were allocated internally or externally.
Additionally addresses a data race in VideoReceiver that was exposed with this change.
R=mflodman@webrtc.org, stefan@webrtc.orgTBR=pthatcher@webrtc.org
BUG=2854,1667
Review URL: https://webrtc-codereview.appspot.com/27829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
