We use Optional in our public API, so its header should be in
webrtc/api/.
BUG=webrtc:8205
Review-Url: https://codereview.webrtc.org/3011943002
Cr-Commit-Position: refs/heads/master@{#19693}
We use ArrayView in our public API, so its header should be in
webrtc/api/.
BUG=none
Review-Url: https://codereview.webrtc.org/3007763002
Cr-Commit-Position: refs/heads/master@{#19658}
directives in our DEPS files are not needed anymore.
Includes from webrtc/rtc_base are also whitelisted in webrtc/DEPS
so we don't have to whitelist it in all the others DEPS files.
BUG=webrtc:7634
NOTRY=True
Review-Url: https://codereview.webrtc.org/3006583002
Cr-Commit-Position: refs/heads/master@{#19601}
Like YuvFileGenerator, this also updates the display with a new slide on every Nth frame, but it generates the slides itself instead of reading them from files.
BUG=webrtc:8138
Review-Url: https://codereview.webrtc.org/3003193002
Cr-Commit-Position: refs/heads/master@{#19585}
DirectTransport::task_queue_ previously could not be const, because of a hack for the deprecated version of DirectTransport. That is no longer the case.
BUG=webrtc:8125
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/3012483002
Cr-Commit-Position: refs/heads/master@{#19583}
Headers webrtc/video_receive_stream.h and webrtc/video_send_stream.h
have been moved to webrtc/call in https://codereview.webrtc.org/3000253002,
this CL is just switching WebRTC internal dependencies to actual headers
instead of depending on the backward compatibility ones.
BUG=webrtc:8107
Review-Url: https://codereview.webrtc.org/3007553002
Cr-Commit-Position: refs/heads/master@{#19561}
Original implementation of jpeg writer didn't compile on iOS at all.
This required clients to exclude some code using defines, which leads to
more complicated code.
Now, instead, jpeg writer will compile but will do nothing on iOS. Clients'
code don't need any additional checks now.
BUG=none
Review-Url: https://codereview.webrtc.org/3004603002
Cr-Commit-Position: refs/heads/master@{#19558}
Add GUARDED_BY annotation to DirectTransport to demonstrate that next_scheduled_task_ is properly protected.
SendRtp(), SendRtcp(), etc., remain as unprotected as they were before the migration from using a thread to using a task-queue.
BUG=webrtc:8113
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/3002163003
Cr-Commit-Position: refs/heads/master@{#19487}
also C++ code (see https://bugs.chromium.org/p/webrtc/issues/detail?id=7743
for more information).
We have moved .mm files out of test_support and fileutils (into test_support_objc
and fileutils_objc).
To achieve the goal for run_test and test_renderer (in the next part of the phrase
X is run_test or test_renderer) we have created 2 targets (X_objc and X_generic)
and X will act as a proxy between these targets (this way we can avoid a circular
dependency between X_generic and X_objc).
BUG=webrtc:7743
Review-Url: https://codereview.webrtc.org/2991323003
Cr-Commit-Position: refs/heads/master@{#19479}
Also, use it to save worst psnr frame in video quality tests. It is indented that these saved frames from perfbots will be uploaded to the cloud and will be available in chrome perf dashboard. Because of that size of the saved frame is somewhat an issue. Also, y4m is not convenient to view.
BUG=webrtc:8030
Review-Url: https://codereview.webrtc.org/2998143002
Cr-Commit-Position: refs/heads/master@{#19450}
Moved the headers video_receive_stream.h and video_send_stream.h from
webrtc/ into webrtc/call/ as part of the Slim and Modular work.
The GN target webrtc:video_stream_api has moved to
webrtc/call:video_stream_api.
There are headers left in webrtc/ with the same name including the
moved headers in webrtc/call/ for not breaking external projects
depending on WebRTC.
At the same time, some minor cleanup is done: Non-pure-virtual functions declared in the two affected headers now have definitions in the same target. After making this change, our 'chromium-style' plugin detected some style violations that have now been fixed: non-inlined constructors and destructors have been added to a number of classes, both inside the GN target of the two affected headers, and in other targets.
BUG=webrtc:8107
Review-Url: https://codereview.webrtc.org/3000253002
Cr-Commit-Position: refs/heads/master@{#19448}
DirectTransport has so far used its own thread, which led to a different threading-model for in the unit-tests than is used in actual WebRTC. Because of that, some critical-sections that weren't truly necessary in WebRTC could not be replaced with thread-checks, because those checks failed in unit-tests.
This CL introduces SingleThreadedTaskQueue - a TaskQueue which guarantees to run all of its tasks on the same thread (rtc::TaskQueue doesn't guarantee that on Mac) - and uses that for DirectTransport. CLs based on top of this will uncomment thread-checks which had to be commented out before, and remove unnecessary critical-sections.
Future work would probably replace the thread-checkers by more sophisticated serialized-access checks, allowing us to move from the SingleThreadedTaskQueue to a normal TaskQueue.
Related implementation notes:
* This CL has made DirectTransport::StopSending() superfluous, and so it was deleted.
BUG=webrtc:8113, webrtc:7405, webrtc:8056, webrtc:8116
Review-Url: https://codereview.webrtc.org/2998923002
Cr-Commit-Position: refs/heads/master@{#19445}
Reason for revert:
Breaks webrtc.linux
Original issue's description:
> Add Jpeg frame writer for test support.
>
> Also, use it to save worst psnr frame in video quality tests. It is indented that these saved frames from perfbots will be uploaded to the cloud and will be available in chrome perf dashboard. Because of that size of the saved frame is somewhat an issue. Also, y4m is not convenient to view.
>
> BUG=webrtc:8030
>
> Review-Url: https://codereview.webrtc.org/2990563002
> Cr-Commit-Position: refs/heads/master@{#19414}
> Committed: 26e5cbd6bbTBR=stefan@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8030
Review-Url: https://codereview.webrtc.org/2998133002
Cr-Commit-Position: refs/heads/master@{#19419}
Also, use it to save worst psnr frame in video quality tests. It is indented that these saved frames from perfbots will be uploaded to the cloud and will be available in chrome perf dashboard. Because of that size of the saved frame is somewhat an issue. Also, y4m is not convenient to view.
BUG=webrtc:8030
Review-Url: https://codereview.webrtc.org/2990563002
Cr-Commit-Position: refs/heads/master@{#19414}
Instead explicitly ignore only the flags we know should be ignored.
BUG=webrtc:7568
Review-Url: https://codereview.webrtc.org/2968003003
Cr-Commit-Position: refs/heads/master@{#19412}
I want to publish an API for iSAC in webrtc/api/, and I want to use
the class names Audio{De,En}coderIsac{Fix,Float}.
BUG=webrtc:7835, webrtc:7841
Review-Url: https://codereview.webrtc.org/2996593002
Cr-Commit-Position: refs/heads/master@{#19381}
This CL brings us one step closer to removing CodecDatabase and
GenericEncoder, by removing the static VCM::Codec(). Codec specific
methods are moved to video_encoder.cc (they already belonged to this
class) and getting default generic codec settings has been moved to a
test specific file.
This CL also makes video_encoder.h pass style guide and lint checks,
since these checks are triggered with the new video_encoder.cc file.
BUG=webrtc:8064
Review-Url: https://codereview.webrtc.org/2993923002
Cr-Commit-Position: refs/heads/master@{#19303}
[This CL is work in progress.]
Wire up the rtp keep-alive in webrtc::Call::Config using new
SetRtpTransportParameters() method on RtpTransportInterface.
BUG=webrtc:7907
Review-Url: https://codereview.webrtc.org/2981513002
Cr-Commit-Position: refs/heads/master@{#19287}
This CL:
- Renames the ViEEncoder class to VideoStreamEncoder, according to discussions.
- Renames variables 'vie_encode' to 'video_stream_encoder'.
- Formatting to match style guide.
- No other changes.
BUG=webrtc:8064
Review-Url: https://codereview.webrtc.org/2995433002
Cr-Commit-Position: refs/heads/master@{#19237}
The destructor was not explicitly marked as "virtual" nor as "override".
BUG=None
Review-Url: https://codereview.webrtc.org/2986363002
Cr-Commit-Position: refs/heads/master@{#19236}
In preparation of making RTP packet demuxing many-to-one (one SSRC goes to one sink, but one sink may have multiple SSRCs), we need to remove FlexFEC's dependence on being able to register itself with the demuxer. Instead, we register FlexFEC streams with the streams they protect; when those streams get packets, they'll forward them to their associated FlexFEC streams, too.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2974453002
Cr-Commit-Position: refs/heads/master@{#19219}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2976293002
Cr-Commit-Position: refs/heads/master@{#19094}
Defaults are consistent with these used in CallTest.
BUG=none
Review-Url: https://codereview.webrtc.org/2972423002
Cr-Commit-Position: refs/heads/master@{#18961}
Since the keep-alive payload type is not registered in the payload type
map of FakeNetworkPipe, it will cause a DCHECK to trigger unless we're
able to destroy the call before that.
Just register it in the fake network as media type "any", it will be
discarded early on the receive side anyway.
BUG=webrt:7964
Review-Url: https://codereview.webrtc.org/2979543002
Cr-Commit-Position: refs/heads/master@{#18953}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2973183002
Cr-Commit-Position: refs/heads/master@{#18948}
Some frames are already marked as 'timing frames' via video-timing RTP header extension. Timestamps along full WebRTC pipeline are gathered for these frames. This CL implements reporting of these timestamps for a single
timing frame since the last GetStats(). The frame with the longest end-to-end delay between two consecutive GetStats calls is reported.
The purpose of this timing information is not to provide a realtime statistics but to provide debugging information as it will help identify problematic places in video pipeline for outliers (frames which took longest to process).
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2946413002
Cr-Commit-Position: refs/heads/master@{#18909}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
(This got reverted because of a problem with the Opus encoder parts.
Re-landing without changes.)
BUG=webrtc:7837
Review-Url: https://codereview.webrtc.org/2950453002
Cr-Commit-Position: refs/heads/master@{#18855}
