I disabled the check on "video_tests" because it pulls
"//webrtc/media/rtc_unittest_main" as a dependency and it defines
the _main (that is already defined by "//webrtc/test:test_main").
I will file a bug to solve this in another CL.
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2832063003
Cr-Commit-Position: refs/heads/master@{#17859}
Introduce new small header-only targets in system_wrappers:
:cpu_features_api
:field_trial_api
:metrics_api
to untangle and optimize dependencies but still satisfy GN check.
In webrtc/p2p, previously uncovered header "base/fakecandidatepair.h"
is added to :p2p_test_utils target.
Refactor system_wrappers so 'rtc_p2p' can depend on only
system_wrappers:field_trial_api instead of all of system_wrappers
(which led to a breakage in Chromium that called for the revert of
https://codereview.webrtc.org/2735583002).
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2739863002
Cr-Commit-Position: refs/heads/master@{#17812}
When SSRCs aren't signaled, an SSRC of 0 is used internally to mean
"the default receive stream". But this wasn't working with the
implementation of GetRtpReceiveParameters in the audio/video
engines. This was breaking RtpReceiver.GetParameters in this situation,
as well as the new getStats implementation (which relies on
GetParameters).
The new implementation will fail if *no* default receive stream is
configured (meaning no default sink is set), and otherwise will return
a default RtpEncodingParameters (later it will be filled with relevant
SDP parameters as they're implemented).
BUG=webrtc:6971
Review-Url: https://codereview.webrtc.org/2806173002
Cr-Commit-Position: refs/heads/master@{#17803}
These tests are already built into rtc_unittests, so they end up being
run three times. Fixed by creating a "p2p_test_utils" target that
contains the test utils that ortc_unittests and rtc_media_unittests
depend on, but not the tests themselves.
BUG=None
TBR=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2820263004
Cr-Commit-Position: refs/heads/master@{#17752}
The example that brought up this issue was:
1. Do offer/answer exchange.
2. Later, remove the audio/video stream.
3. Add back a new stream, that contains only the audio track.
4. Do new offer/answer.
The new offer didn't have the new stream ID, but code elsewhere was
expecting one. As a result, the send stream is never hooked up to the
audio track, and audio packets aren't sent.
BUG=chromium:611708
Review-Url: https://codereview.webrtc.org/2810733003
Cr-Commit-Position: refs/heads/master@{#17709}
The compiler is allowed to leave a chopped-off horse's head on your
bed if you do.
Happily, we didn't even have to move this function to another namespace,
since it was unused.
BUG=webrtc:7484
Review-Url: https://codereview.webrtc.org/2815203002
Cr-Commit-Position: refs/heads/master@{#17696}
Reason for revert:
Breaks android buildbots.
Original issue's description:
> Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
>
> Reason for revert:
> Reland with appropriate changes to API to not break depending projects.
>
> Original issue's description:
> > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
> >
> > Reason for revert:
> > Breaks dependent projects.
> >
> > Original issue's description:
> > > Add content type information to Encoded Images and add corresponding RTP extension header.
> > > Use it to separate UMA e2e delay metric between screenshare from video.
> > > Content type extension is set based on encoder settings and processed and decoders.
> > >
> > > Also,
> > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> > >
> > > BUG=webrtc:7420
> > >
> > > Review-Url: https://codereview.webrtc.org/2772033002
> > > Cr-Commit-Position: refs/heads/master@{#17640}
> > > Committed: 64e739aeae
> >
> > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2816463002
> > Cr-Commit-Position: refs/heads/master@{#17644}
> > Committed: 5721866808
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2812913002
> Cr-Commit-Position: refs/heads/master@{#17651}
> Committed: 774f6b4b96TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420
Review-Url: https://codereview.webrtc.org/2809653004
Cr-Commit-Position: refs/heads/master@{#17653}
Reason for revert:
Reland with appropriate changes to API to not break depending projects.
Original issue's description:
> Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
>
> Reason for revert:
> Breaks dependent projects.
>
> Original issue's description:
> > Add content type information to Encoded Images and add corresponding RTP extension header.
> > Use it to separate UMA e2e delay metric between screenshare from video.
> > Content type extension is set based on encoder settings and processed and decoders.
> >
> > Also,
> > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> >
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2772033002
> > Cr-Commit-Position: refs/heads/master@{#17640}
> > Committed: 64e739aeae
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2816463002
> Cr-Commit-Position: refs/heads/master@{#17644}
> Committed: 5721866808TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420
Review-Url: https://codereview.webrtc.org/2812913002
Cr-Commit-Position: refs/heads/master@{#17651}
Reason for revert:
Relanded by mistake.
Original issue's description:
> Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
>
> Reason for revert:
> Reland with fixes which break API
>
> Original issue's description:
> > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
> >
> > Reason for revert:
> > Breaks dependent projects.
> >
> > Original issue's description:
> > > Add content type information to Encoded Images and add corresponding RTP extension header.
> > > Use it to separate UMA e2e delay metric between screenshare from video.
> > > Content type extension is set based on encoder settings and processed and decoders.
> > >
> > > Also,
> > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> > >
> > > BUG=webrtc:7420
> > >
> > > Review-Url: https://codereview.webrtc.org/2772033002
> > > Cr-Commit-Position: refs/heads/master@{#17640}
> > > Committed: 64e739aeae
> >
> > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2816463002
> > Cr-Commit-Position: refs/heads/master@{#17644}
> > Committed: 5721866808
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2811963002
> Cr-Commit-Position: refs/heads/master@{#17645}
> Committed: 4fa0c4f97fTBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420
Review-Url: https://codereview.webrtc.org/2810923004
Cr-Commit-Position: refs/heads/master@{#17648}
Reason for revert:
Reland with fixes which break API
Original issue's description:
> Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
>
> Reason for revert:
> Breaks dependent projects.
>
> Original issue's description:
> > Add content type information to Encoded Images and add corresponding RTP extension header.
> > Use it to separate UMA e2e delay metric between screenshare from video.
> > Content type extension is set based on encoder settings and processed and decoders.
> >
> > Also,
> > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> >
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2772033002
> > Cr-Commit-Position: refs/heads/master@{#17640}
> > Committed: 64e739aeae
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2816463002
> Cr-Commit-Position: refs/heads/master@{#17644}
> Committed: 5721866808TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420
Review-Url: https://codereview.webrtc.org/2811963002
Cr-Commit-Position: refs/heads/master@{#17645}
Reason for revert:
Breaks dependent projects.
Original issue's description:
> Add content type information to Encoded Images and add corresponding RTP extension header.
> Use it to separate UMA e2e delay metric between screenshare from video.
> Content type extension is set based on encoder settings and processed and decoders.
>
> Also,
> Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
>
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2772033002
> Cr-Commit-Position: refs/heads/master@{#17640}
> Committed: 64e739aeaeTBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420
Review-Url: https://codereview.webrtc.org/2816463002
Cr-Commit-Position: refs/heads/master@{#17644}
Use it to separate UMA e2e delay metric between screenshare from video.
Content type extension is set based on encoder settings and processed and decoders.
Also,
Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
BUG=webrtc:7420
Review-Url: https://codereview.webrtc.org/2772033002
Cr-Commit-Position: refs/heads/master@{#17640}
Reason for revert:
Re-land, reverting did not fix bug.
https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
Original issue's description:
> Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
>
> Reason for revert:
> Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
>
> https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
>
> Original issue's description:
> > Added the GetSources() to the RtpReceiverInterface and implemented
> > it for the AudioRtpReceiver.
> >
> > This method returns a vector of RtpSource(both CSRC source and SSRC
> > source) which contains the ID of a source, the timestamp, the source
> > type (SSRC or CSRC) and the audio level.
> >
> > The RtpSource objects are buffered and maintained by the
> > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> > the info of the contributing source will be pulled along the object
> > chain:
> > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> > AudioReceiveStream -> voe::Channel -> RtpRtcp module
> >
> > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
> >
> > BUG=chromium:703122
> > TBR=stefan@webrtc.org, danilchap@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2770233003
> > Cr-Commit-Position: refs/heads/master@{#17591}
> > Committed: 292084c376
>
> TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:703122
>
> Review-Url: https://codereview.webrtc.org/2809613002
> Cr-Commit-Position: refs/heads/master@{#17616}
> Committed: fbcc5cb386TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:703122
Review-Url: https://codereview.webrtc.org/2810623003
Cr-Commit-Position: refs/heads/master@{#17621}
AudioDecoder and AudioDecoderFactory are in webrtc/api/ now, so move
their mocks to someplace central where tests from all over WebRTC are
allowed to #include them.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2798063004
Cr-Commit-Position: refs/heads/master@{#17619}
Reason for revert:
Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
Original issue's description:
> Added the GetSources() to the RtpReceiverInterface and implemented
> it for the AudioRtpReceiver.
>
> This method returns a vector of RtpSource(both CSRC source and SSRC
> source) which contains the ID of a source, the timestamp, the source
> type (SSRC or CSRC) and the audio level.
>
> The RtpSource objects are buffered and maintained by the
> RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> the info of the contributing source will be pulled along the object
> chain:
> AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> AudioReceiveStream -> voe::Channel -> RtpRtcp module
>
> Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
>
> BUG=chromium:703122
> TBR=stefan@webrtc.org, danilchap@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2770233003
> Cr-Commit-Position: refs/heads/master@{#17591}
> Committed: 292084c376TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:703122
Review-Url: https://codereview.webrtc.org/2809613002
Cr-Commit-Position: refs/heads/master@{#17616}
This CL contains all the changes made to audio_coding while making
audio encoders injectable. Apart from some small changes to
webrtcvoiceengine, nothing here is hooked up to the outside
world. Those changes will be added to a follow-up CL.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2695243005
Cr-Commit-Position: refs/heads/master@{#17569}
Reason for revert:
Reland with temporary deprecated API to not break chromium and google3.
Original issue's description:
> Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ )
>
> Reason for revert:
> Suspect of breaking Chrome FYI bots.
>
> See
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder
>
> Example logs:
> ../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory
> #include "third_party/webrtc/video_encoder.h"
> ^
>
> Original issue's description:
> > Move video_encoder.h and video_decoder.h to /api and create GN targets for them
> >
> > BUG=webrtc:5881
> > # Because PRESUBMIT ignores LINT blacklist for moved files and these
> > # headers have some not easy to resolve issues.
> > NOPRESUBMIT=True
> >
> > Review-Url: https://codereview.webrtc.org/2780943003
> > Cr-Commit-Position: refs/heads/master@{#17511}
> > Committed: c42f540570
>
> TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5881
>
> Review-Url: https://codereview.webrtc.org/2794033002
> Cr-Commit-Position: refs/heads/master@{#17514}
> Committed: 716d7ac5c1TBR=solenberg@webrtc.org,sprang@webrtc.org,guidou@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5881
Review-Url: https://codereview.webrtc.org/2795163002
Cr-Commit-Position: refs/heads/master@{#17537}
Reason for revert:
Breaks Chrome FYI Android bots.
See:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus9%29/builds/20418https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus6%29/builds/14724https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/builds/20133https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/15111
Original issue's description:
> Deliver video frames on Android, on the decode thread.
>
> VideoCoding
> * Adding a method for polling for frames on Android only until the capture implementation takes care of this (longer term plan).
>
> CodecDatabase
> * Add an accessor for the current decoder
> * Use std::unique_ptr<> for ownership.
> * Remove "Release()" and "ReleaseDecoder()". Instead just delete.
> * Remove |friend| relationship between CodecDatabase and VCMGenericDecoder.
>
> VCMDecodedFrameCallback
> * DCHECKs for thread correctness.
> * Remove |lock_| now that a threading model has been established and verified.
>
> VCMGenericDecoder
> * All methods now have thread checks.
> * Variable access associated with thread checkers.
>
> VideoReceiver
> * Added two notification methods, DecoderThreadStarting() and DecoderThreadStopped()
> * Allows us to establish a period when the decoder thread is not running and it is safe to modify variables such as callbacks, that are only read when the decoder thread is running.
> * Allows us to DCHECK thread guarantees.
> * Allows synchronizing callbacks from the module process thread and have them only active while the decoder thread is running.
> * The above, allows us to establish two modes for the thread, single-threaded-mutable and multi-threaded-const.
> * Using that knowledge, we can remove |receive_crit_| as well as locking for a number of member variables.
>
> MediaCodecVideoDecoder
> * Removed frame polling code from this class, since this is now done from the root thread function in VideoReceiveStream.
>
> VideoReceiveStream
> * On Android: Polls for decoded frames every 10ms (same interval as previously in MediaCodecVideoDecoder)
> * [Un]Registers the |video_receiver_| with the module thread only around the time the decoder thread is started/stopped.
> * Notifies the receiver of start/stop events of the decoder thread.
> * Changed the decoder thread to use the new PlatformThread callback type.
>
> BUG=webrtc:7361, 695438
>
> Review-Url: https://codereview.webrtc.org/2764573002
> Cr-Commit-Position: refs/heads/master@{#17527}
> Committed: e3aa88bbd5TBR=sakal@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7361, 695438
Review-Url: https://codereview.webrtc.org/2792033003
Cr-Commit-Position: refs/heads/master@{#17530}
VideoCoding
* Adding a method for polling for frames on Android only until the capture implementation takes care of this (longer term plan).
CodecDatabase
* Add an accessor for the current decoder
* Use std::unique_ptr<> for ownership.
* Remove "Release()" and "ReleaseDecoder()". Instead just delete.
* Remove |friend| relationship between CodecDatabase and VCMGenericDecoder.
VCMDecodedFrameCallback
* DCHECKs for thread correctness.
* Remove |lock_| now that a threading model has been established and verified.
VCMGenericDecoder
* All methods now have thread checks.
* Variable access associated with thread checkers.
VideoReceiver
* Added two notification methods, DecoderThreadStarting() and DecoderThreadStopped()
* Allows us to establish a period when the decoder thread is not running and it is safe to modify variables such as callbacks, that are only read when the decoder thread is running.
* Allows us to DCHECK thread guarantees.
* Allows synchronizing callbacks from the module process thread and have them only active while the decoder thread is running.
* The above, allows us to establish two modes for the thread, single-threaded-mutable and multi-threaded-const.
* Using that knowledge, we can remove |receive_crit_| as well as locking for a number of member variables.
MediaCodecVideoDecoder
* Removed frame polling code from this class, since this is now done from the root thread function in VideoReceiveStream.
VideoReceiveStream
* On Android: Polls for decoded frames every 10ms (same interval as previously in MediaCodecVideoDecoder)
* [Un]Registers the |video_receiver_| with the module thread only around the time the decoder thread is started/stopped.
* Notifies the receiver of start/stop events of the decoder thread.
* Changed the decoder thread to use the new PlatformThread callback type.
BUG=webrtc:7361, 695438
Review-Url: https://codereview.webrtc.org/2764573002
Cr-Commit-Position: refs/heads/master@{#17527}
Reason for revert:
Suspect of breaking Chrome FYI bots.
See
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder
Example logs:
../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory
#include "third_party/webrtc/video_encoder.h"
^
Original issue's description:
> Move video_encoder.h and video_decoder.h to /api and create GN targets for them
>
> BUG=webrtc:5881
> # Because PRESUBMIT ignores LINT blacklist for moved files and these
> # headers have some not easy to resolve issues.
> NOPRESUBMIT=True
>
> Review-Url: https://codereview.webrtc.org/2780943003
> Cr-Commit-Position: refs/heads/master@{#17511}
> Committed: c42f540570TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5881
Review-Url: https://codereview.webrtc.org/2794033002
Cr-Commit-Position: refs/heads/master@{#17514}
BUG=webrtc:5881
# Because PRESUBMIT ignores LINT blacklist for moved files and these
# headers have some not easy to resolve issues.
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2780943003
Cr-Commit-Position: refs/heads/master@{#17511}
Reason for revert:
Seem to be a flaky test rather than an issue with this cl. Creating reland, will add code to reduce flakiness to that test.
Original issue's description:
> Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
>
> Reason for revert:
> This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780
>
> Original issue's description:
> > Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ )
> >
> > Reason for revert:
> > Found issue with test case, will add fix to reland cl.
> >
> > Original issue's description:
> > > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
> > >
> > > Reason for revert:
> > > Breaks perf tests:
> > > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679
> > > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325
> > >
> > > Original issue's description:
> > > > Add framerate to VideoSinkWants and ability to signal on overuse
> > > >
> > > > In ViEEncoder, try to reduce framerate instead of resolution if the
> > > > current degradation preference is maintain-resolution rather than
> > > > balanced.
> > > >
> > > > BUG=webrtc:4172
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2716643002
> > > > Cr-Commit-Position: refs/heads/master@{#17327}
> > > > Committed: 72acf25261
> > >
> > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:4172
> > >
> > > Review-Url: https://codereview.webrtc.org/2764133002
> > > Cr-Commit-Position: refs/heads/master@{#17331}
> > > Committed: 8b45b11144
> >
> > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:4172
> >
> > Review-Url: https://codereview.webrtc.org/2781433002
> > Cr-Commit-Position: refs/heads/master@{#17474}
> > Committed: 3ea3c77e93
>
> TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4172
>
> Review-Url: https://codereview.webrtc.org/2783183003
> Cr-Commit-Position: refs/heads/master@{#17477}
> Committed: f9ed235c9bR=ilnik@webrtc.org,stefan@webrtc.org
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2789823002
Cr-Commit-Position: refs/heads/master@{#17498}
Reason for revert:
This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780
Original issue's description:
> Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ )
>
> Reason for revert:
> Found issue with test case, will add fix to reland cl.
>
> Original issue's description:
> > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
> >
> > Reason for revert:
> > Breaks perf tests:
> > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679
> > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325
> >
> > Original issue's description:
> > > Add framerate to VideoSinkWants and ability to signal on overuse
> > >
> > > In ViEEncoder, try to reduce framerate instead of resolution if the
> > > current degradation preference is maintain-resolution rather than
> > > balanced.
> > >
> > > BUG=webrtc:4172
> > >
> > > Review-Url: https://codereview.webrtc.org/2716643002
> > > Cr-Commit-Position: refs/heads/master@{#17327}
> > > Committed: 72acf25261
> >
> > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:4172
> >
> > Review-Url: https://codereview.webrtc.org/2764133002
> > Cr-Commit-Position: refs/heads/master@{#17331}
> > Committed: 8b45b11144
>
> TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:4172
>
> Review-Url: https://codereview.webrtc.org/2781433002
> Cr-Commit-Position: refs/heads/master@{#17474}
> Committed: 3ea3c77e93TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2783183003
Cr-Commit-Position: refs/heads/master@{#17477}
Reason for revert:
Intend to fix perf failures and reland.
Original issue's description:
> Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
>
> Reason for revert:
> Reverting since this seems to break multiple WebRTC Perf buildbots
>
> Original issue's description:
> > Don't hardcode MediaType::ANY in FakeNetworkPipe.
> >
> > Instead let each test set the appropriate media type. This simplifies
> > demuxing in Call and later in RtpTransportController.
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/2774463003
> > Cr-Commit-Position: refs/heads/master@{#17418}
> > Committed: 9c47b00e24
>
> TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2784543002
> Cr-Commit-Position: refs/heads/master@{#17427}
> Committed: 3a3bd50610TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2783853002
Cr-Commit-Position: refs/heads/master@{#17459}
Also renames "peerconnection_unittests" to "peerconnection_integrationtests",
and moves the ICE URL parsing code to separate files.
The main problem previously was that the test assertions
occurred in various places in the main test class, and this shared test
code was overly complex and stateful. As a result, it was difficult to
tell what a test even does, let alone what assertions it's meant to be
making. And writing a new test that does what you want can be a
frustrating ordeal.
The new code still uses helper methods, but they have intuitive names
and a smaller role; all of the important parts of the test's logic are
in the test case itself.
We're planning on merging PeerConnection and WebRtcSession at some point
soon, so it seemed valuable to do this, so that the WebRtcSession tests
can be rewritten as PeerConnection tests using better patterns.
BUG=None
Review-Url: https://codereview.webrtc.org/2738353003
Cr-Commit-Position: refs/heads/master@{#17458}
Reason for revert:
Reverting since this seems to break multiple WebRTC Perf buildbots
Original issue's description:
> Don't hardcode MediaType::ANY in FakeNetworkPipe.
>
> Instead let each test set the appropriate media type. This simplifies
> demuxing in Call and later in RtpTransportController.
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2774463003
> Cr-Commit-Position: refs/heads/master@{#17418}
> Committed: 9c47b00e24TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2784543002
Cr-Commit-Position: refs/heads/master@{#17427}
Instead let each test set the appropriate media type. This simplifies
demuxing in Call and later in RtpTransportController.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2774463003
Cr-Commit-Position: refs/heads/master@{#17418}
This reverts to previous behavior where b=AS only affects the codec bitrate for audio streams, and not the max bandwidth estimate.
BUG=chromium:703903
Review-Url: https://codereview.webrtc.org/2774123002
Cr-Commit-Position: refs/heads/master@{#17386}
Reason for revert:
Makes perf and Chromium FYI bots unhappy.
Original issue's description:
> WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
>
> This removes one more place where we were unable to handle codecs not
> in the built-in set.
>
> BUG=webrtc:5805
>
> Review-Url: https://codereview.webrtc.org/2686043006
> Cr-Commit-Position: refs/heads/master@{#17370}
> Committed: 1724cfbdbaTBR=ossu@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2772043002
Cr-Commit-Position: refs/heads/master@{#17374}
This removes one more place where we were unable to handle codecs not
in the built-in set.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2686043006
Cr-Commit-Position: refs/heads/master@{#17370}
In ViEEncoder, try to reduce framerate instead of resolution if the
current degradation preference is maintain-resolution rather than
balanced.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2716643002
Cr-Commit-Position: refs/heads/master@{#17327}
Reason to go back is that we may end up with a bunch of streams that are never cleaned up and consume resources.
BUG=webrtc:7175, b/35863246
Review-Url: https://codereview.webrtc.org/2746763002
Cr-Commit-Position: refs/heads/master@{#17210}
Packets on source ports 32768-49151 got identified as RTP packets by
"IsRtpPacket" and were ignored by the SCTP transport.
This CL changes this to check the packet flags for "PF_SRTP_BYPASS".
BUG=webrtc:6959
Review-Url: https://codereview.webrtc.org/2743653005
Cr-Commit-Position: refs/heads/master@{#17179}
Use of FlexFEC is known when streams are created in
WebRtcVideoChannel2, so this replaces the code in Call to infer
FlexFEC config of video streams from the configuration of the FlexFEC
stream(s). This also allows us to switch to a more logical creation
order, where media streams are created before the FlexFEC stream.
This is done in preparation for a larger refactoring of the RTP
demuxing done in Call.
BUG=None
Review-Url: https://codereview.webrtc.org/2712683002
Cr-Commit-Position: refs/heads/master@{#17143}
