This CL performs the following renames of targets to
make GYP and GN more unified and make the targets that
have the same name as the module and include the external
render/capture implementation (the internal one is only
used by WebRTC tests).
This makes it natural to declare dependencies in GN
without having to specify the target.
Summary of the renames:
GYP:
video_render_module_impl -> video_render (new target)
video_capture_module_impl -> video_capture (new target)
GN:
video_capture -> video_capture_module (now identical to the GYP target)
video_capture_impl -> video_capture
video_render -> video_render_module (now identical to the GYP target)
video_render_impl -> video_render
BUG=456815
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35099004
Cr-Commit-Position: refs/heads/master@{#8323}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8323 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL moves the pacer out from the regular module process thread to
instead use one thread per pacer. This is to get better accuracy for the
paced packets and to avoid overusing the module process thread.
BUG=
TEST=existing tests
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41839004
Cr-Commit-Position: refs/heads/master@{#8308}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8308 4adac7df-926f-26a2-2b94-8c16560cd09d
Removes AddRenderCallback from ViERenderer and implements
VideoReceiveStream on top of DeliverI420Frame like WebRtcVideoEngine
currently does today.
Also adds ::IsTextureSupported() to the VideoRenderer interface to
permit querying whether an external renderer supports texture rendering.
R=stefan@webrtc.orgTBR=mflodman@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/34169004
Cr-Commit-Position: refs/heads/master@{#8299}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8299 4adac7df-926f-26a2-2b94-8c16560cd09d
The issue that was causing the thread checker to report error, turned out to be unrelated.
> Revert 8203 "Reducing locking in OveruseFrameDetector and increa..."
>
> Broke tests in Chrome for some reason:
>
> [ RUN ] WebRtcAecDumpBrowserTest.CallWithAecDump
> [80131:1287:0129/074432:30561723987517:ERROR:vt_video_decode_accelerator.cc(132)] Failed to create VTDecompressionSession: codecOpenErr (-8973)
> [80129:1287:0129/074432:30562276677373:INFO:CONSOLE(64)] "Looking at video in element remote-view-1", source: http://127.0.0.1:61401/media/webrtc_test_utilities.js (64)
> [80129:1287:0129/074432:30562281435788:INFO:CONSOLE(64)] "Looking at video in element remote-view-2", source: http://127.0.0.1:61401/media/webrtc_test_utilities.js (64)
> [80129:1287:0129/074432:30562315329399:INFO:CONSOLE(800)] "Negotiating call...", source: http://127.0.0.1:61401/media/peerconnection-call.html (800)
> [80133:29187:0129/074432:30562402039578:FATAL:overuse_frame_detector.cc(388)] Check failed: processing_thread_.CalledOnValidThread().
> 0 libbase.dylib 0x000000010dfd688f base::debug::StackTrace::StackTrace() + 47
> 1 libbase.dylib 0x000000010dfd68e3 base::debug::StackTrace::StackTrace() + 35
> 2 libbase.dylib 0x000000010e030076 logging::LogMessage::~LogMessage() + 70
> 3 libbase.dylib 0x000000010e02f0c3 logging::LogMessage::~LogMessage() + 35
> 4 libcontent.dylib 0x000000011d8c0cd5 webrtc::OveruseFrameDetector::TimeUntilNextProcess() + 245
> 5 libcontent.dylib 0x000000011d31ddfd webrtc::ProcessThreadImpl::Process() + 525
> 6 libcontent.dylib 0x000000011d31d836 webrtc::ProcessThreadImpl::Run(void*) + 38
> 7 libcontent.dylib 0x000000011d10c390 webrtc::ThreadPosix::Run() + 288
> 8 libcontent.dylib 0x000000011d10c076 webrtc::StartThread(void*) + 38
> 9 libsystem_pthread.dylib 0x00007fff8e667899 _pthread_body + 138
> 10 libsystem_pthread.dylib 0x00007fff8e66772a _pthread_struct_init + 0
> 11 libsystem_pthread.dylib 0x00007fff8e66bfc9 thread_start + 13
>
>
> > Reducing locking in OveruseFrameDetector and increasing constness.
> >
> > I also added a few TODOs there to see what we can do to reduce the chance of contention.
> > To catch regressions, I've started using the ThreadChecker class on the processing thread but it might also be a good idea to add similar checks for other known threads such as the thread we receive frames on. I'm sure we can reduce locking even further.
> >
> > BUG=2822
> > R=asapersson@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/33129004
>
> TBR=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/34079004TBR=tommi@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/35029004
Cr-Commit-Position: refs/heads/master@{#8287}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8287 4adac7df-926f-26a2-2b94-8c16560cd09d
sending RTP module for the specified simulcast layer a frame belongs to.
This CL also removes the corresponding functionality from the RTP RTCP
module and fixes lint warnings in the files touched.
BUG=769
TEST=New unittest and manual tests
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39629004
Cr-Commit-Position: refs/heads/master@{#8267}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8267 4adac7df-926f-26a2-2b94-8c16560cd09d
* Add a new WakeUp method that gives a module a chance to be called back right away on the worker thread.
* Wrote unit tests for the class.
* Significantly reduce the amount of locking.
- ProcessThreadImpl itself does a lot less locking.
- Reimplemented the way we keep track of when to make calls to Process.
This reduces the amount of calls to TimeUntilNextProcess and since most implementations of that function grab a lock, this means less locking.
* Renamed ProcessThread::CreateProcessThread to ProcessThread::Create.
* Added thread checks for Start/Stop. Threading model of other functions is now documented.
* We now log an error if an implementation of TimeUntilNextProcess returns a negative value (some implementations do, but the method should only return a positive nr of ms).
* Removed the DestroyProcessThread method and instead force callers to use scoped_ptr<> to maintain object lifetime.
BUG=2822
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35999004
Cr-Commit-Position: refs/heads/master@{#8261}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8261 4adac7df-926f-26a2-2b94-8c16560cd09d
The surrounding similar methods all used unsigned char, using unsigned int in
this case looks like an accident, especially since the function passes on the
value in question to a function expecting a uint8.
BUG=none
TEST=none
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40529004
Cr-Commit-Position: refs/heads/master@{#8228}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8228 4adac7df-926f-26a2-2b94-8c16560cd09d
Broke tests in Chrome for some reason:
[ RUN ] WebRtcAecDumpBrowserTest.CallWithAecDump
[80131:1287:0129/074432:30561723987517:ERROR:vt_video_decode_accelerator.cc(132)] Failed to create VTDecompressionSession: codecOpenErr (-8973)
[80129:1287:0129/074432:30562276677373:INFO:CONSOLE(64)] "Looking at video in element remote-view-1", source: http://127.0.0.1:61401/media/webrtc_test_utilities.js (64)
[80129:1287:0129/074432:30562281435788:INFO:CONSOLE(64)] "Looking at video in element remote-view-2", source: http://127.0.0.1:61401/media/webrtc_test_utilities.js (64)
[80129:1287:0129/074432:30562315329399:INFO:CONSOLE(800)] "Negotiating call...", source: http://127.0.0.1:61401/media/peerconnection-call.html (800)
[80133:29187:0129/074432:30562402039578:FATAL:overuse_frame_detector.cc(388)] Check failed: processing_thread_.CalledOnValidThread().
0 libbase.dylib 0x000000010dfd688f base::debug::StackTrace::StackTrace() + 47
1 libbase.dylib 0x000000010dfd68e3 base::debug::StackTrace::StackTrace() + 35
2 libbase.dylib 0x000000010e030076 logging::LogMessage::~LogMessage() + 70
3 libbase.dylib 0x000000010e02f0c3 logging::LogMessage::~LogMessage() + 35
4 libcontent.dylib 0x000000011d8c0cd5 webrtc::OveruseFrameDetector::TimeUntilNextProcess() + 245
5 libcontent.dylib 0x000000011d31ddfd webrtc::ProcessThreadImpl::Process() + 525
6 libcontent.dylib 0x000000011d31d836 webrtc::ProcessThreadImpl::Run(void*) + 38
7 libcontent.dylib 0x000000011d10c390 webrtc::ThreadPosix::Run() + 288
8 libcontent.dylib 0x000000011d10c076 webrtc::StartThread(void*) + 38
9 libsystem_pthread.dylib 0x00007fff8e667899 _pthread_body + 138
10 libsystem_pthread.dylib 0x00007fff8e66772a _pthread_struct_init + 0
11 libsystem_pthread.dylib 0x00007fff8e66bfc9 thread_start + 13
> Reducing locking in OveruseFrameDetector and increasing constness.
>
> I also added a few TODOs there to see what we can do to reduce the chance of contention.
> To catch regressions, I've started using the ThreadChecker class on the processing thread but it might also be a good idea to add similar checks for other known threads such as the thread we receive frames on. I'm sure we can reduce locking even further.
>
> BUG=2822
> R=asapersson@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/33129004TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34079004
Cr-Commit-Position: refs/heads/master@{#8206}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8206 4adac7df-926f-26a2-2b94-8c16560cd09d
I also added a few TODOs there to see what we can do to reduce the chance of contention.
To catch regressions, I've started using the ThreadChecker class on the processing thread but it might also be a good idea to add similar checks for other known threads such as the thread we receive frames on. I'm sure we can reduce locking even further.
BUG=2822
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33129004
Cr-Commit-Position: refs/heads/master@{#8203}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8203 4adac7df-926f-26a2-2b94-8c16560cd09d
Make a few member variables in the Transport class officially const so that it's clear that locking isn't needed for access. There are getters for some of these (e.g. content_name()) that don't have locking or checking, so making the variables const is at least a way to guard against regressions. Also making the clock_ member in overuse_frame_detector.h const for clarity that it doesn't require a lock for access.
No code change.
Review URL: https://webrtc-codereview.appspot.com/35949004
Cr-Commit-Position: refs/heads/master@{#8186}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8186 4adac7df-926f-26a2-2b94-8c16560cd09d
Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.
> Support associated payload type when registering Rtx payload type.
>
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
>
> BUG=4024
> R=pbos@webrtc.org, stefan@webrtc.org
> TBR=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/26259004
>
> Patch from Changbin Shao <changbin.shao@intel.com>.
TBR=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.
This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".
BUG=chromium:81439
TEST=none
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
