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
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.orgTBR=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
- Add max_bit_rate and max_payload_size_bytes to config structs.
- Fix support for 48 kHz sample rate.
- Fix iSAC-RED.
- Add method UpdateDecoderSampleRate().
- Update locking structure with a separate lock for local member
variables used by the encoder methods.
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41659004
Cr-Commit-Position: refs/heads/master@{#8204}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8204 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
This fixes a two year old TODO of deleting dead code :)
In cases where the _id or id_ member variable is being used for tracing,
I changed the member to at least be const.
It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them.
BUG=
R=henrika@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37849004
Cr-Commit-Position: refs/heads/master@{#8201}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
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
The latter file was more up-to-date. The files are now identical
with the following exceptions:
* The namespace used (rtc vs. webrtc).
* The name of the include guard.
* base/scoped_ptr.h still has two extra methods, accept() and use().
* base/scoped_ptr.h still includes webrtc/base/common.h even though
it doesn't need it itself, since several .cc files expect to get
it for free by incuding base/scoped_ptr.h. This is of course bad
manners, and the "unused" include will be removed in a future CL.
A later CL will remove system_wrappers/interface/scoped_ptr.h.
R=andrew@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8147
And reverted again, because out-of-tree code using this file was defining nullptr to 0: https://code.google.com/p/webrtc/source/detail?r=8149
Review URL: https://webrtc-codereview.appspot.com/36919004
Cr-Commit-Position: refs/heads/master@{#8196}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8196 4adac7df-926f-26a2-2b94-8c16560cd09d
When using the paced sender, packets will be put into the rtp packet
history and then retreived from there again when it is time to send.
In some cases (low send bitrate and very large frames created) this
may overflow, causing packets to be overwritten in the packet history
before they have been sent.
Check this condition and expand history size if needed.
This is primarily triggered during screenshare, when
switching to a large picture with lots of high frequency
details in it.
BUG=4171
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34879004
Cr-Commit-Position: refs/heads/master@{#8195}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8195 4adac7df-926f-26a2-2b94-8c16560cd09d
Currently, in ipc_network_manager.cc, the UMA WebRTC.PeerConnection.IPv4Interfaces and its IPv6
counter part counts the addresses, instead of the interfaces as when
chromium delivers available networks to WebRTC, each address is wrapped
inside an individual network object.
The plan is to replace the current MergeNetworkList with the new one once it's rolled into chromium.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36779004
Cr-Commit-Position: refs/heads/master@{#8188}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8188 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
The following three methods are added:
rtp_timestamp_rate_hz()
SetTargetBitrate()
SetProjectedPacketLossRate()
Default implementations are provided, and a few overrides are
implemented. AudioEncoderCopyRed and AudioEncoderCng propagate the new
methods to the underlying speech codec.
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34049004
Cr-Commit-Position: refs/heads/master@{#8171}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8171 4adac7df-926f-26a2-2b94-8c16560cd09d
Previously, the delay estimate history was reset every time the metrics were pulled. This required all clients to be on the same thread and make use of one call.
Now we use a fixed aggregation window of one second and when a client pulls the metrics you get the latest value.
Under certain circumstances like tests you would like to have the aggregation window set to the recording length. We therefore turn on the fixed aggregation window after the first call.
BUG=2994
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38759004
Cr-Commit-Position: refs/heads/master@{#8170}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8170 4adac7df-926f-26a2-2b94-8c16560cd09d
