First patch set uses a script attached in an issue comment:
https://bugs.chromium.org/p/webrtc/issues/detail?id=5118#c24
This discards the boilerplate prefix of WEBRTC_TRACE log strings, but it appears to be discarded anyway by all users.
Second patch set removes the header and makes small fixes to four of the log messages.
BUG=webrtc:5118
Review-Url: https://codereview.webrtc.org/2958273002
Cr-Commit-Position: refs/heads/master@{#18941}
Ensure that the ring buffer does not return a pointer into the buffer if
no data is available to read.
The ring buffer fix is not directly applicable to issue webrtc:7845, but may cause related memory errors.
BUG=webrtc:7845
Review-Url: https://codereview.webrtc.org/2971313002
Cr-Commit-Position: refs/heads/master@{#18940}
Prior to https://codereview.webrtc.org/2750783004/ Reset() intentionally
did not zero out the buffer. After that change, callers calling Reset()
and then mutable_data() were performing a wasteful zeroing.
This change adds ResetWithoutMuting() to match the old behavior and
switches the sole non-test caller of Reset() to use ResetWithoutMuting()
instead.
Prior to this change (optimized, Linux):
$ out/Default/webrtc_perf_tests --gtest_filter=NetEqPerformanceTest.Run* \
--gtest_repeat=10 | grep neteq_performance
*RESULT neteq_performance: 10_pl_10_drift= 4051 ms
*RESULT neteq_performance: 0_pl_0_drift= 1768 ms
*RESULT neteq_performance: 10_pl_10_drift= 3666 ms
*RESULT neteq_performance: 0_pl_0_drift= 1690 ms
*RESULT neteq_performance: 10_pl_10_drift= 3685 ms
*RESULT neteq_performance: 0_pl_0_drift= 1693 ms
*RESULT neteq_performance: 10_pl_10_drift= 3720 ms
*RESULT neteq_performance: 0_pl_0_drift= 1690 ms
*RESULT neteq_performance: 10_pl_10_drift= 3780 ms
*RESULT neteq_performance: 0_pl_0_drift= 1728 ms
*RESULT neteq_performance: 10_pl_10_drift= 3733 ms
*RESULT neteq_performance: 0_pl_0_drift= 1737 ms
*RESULT neteq_performance: 10_pl_10_drift= 3781 ms
*RESULT neteq_performance: 0_pl_0_drift= 1744 ms
*RESULT neteq_performance: 10_pl_10_drift= 3712 ms
*RESULT neteq_performance: 0_pl_0_drift= 1731 ms
*RESULT neteq_performance: 10_pl_10_drift= 3681 ms
*RESULT neteq_performance: 0_pl_0_drift= 1691 ms
*RESULT neteq_performance: 10_pl_10_drift= 3681 ms
*RESULT neteq_performance: 0_pl_0_drift= 1690 ms
With this change:
$ out/Default/webrtc_perf_tests --gtest_filter=NetEqPerformanceTest.Run* \
--gtest_repeat=10 | grep neteq_performance
*RESULT neteq_performance: 10_pl_10_drift= 3824 ms
*RESULT neteq_performance: 0_pl_0_drift= 1632 ms
*RESULT neteq_performance: 10_pl_10_drift= 3502 ms
*RESULT neteq_performance: 0_pl_0_drift= 1521 ms
*RESULT neteq_performance: 10_pl_10_drift= 3520 ms
*RESULT neteq_performance: 0_pl_0_drift= 1534 ms
*RESULT neteq_performance: 10_pl_10_drift= 3517 ms
*RESULT neteq_performance: 0_pl_0_drift= 1530 ms
*RESULT neteq_performance: 10_pl_10_drift= 3521 ms
*RESULT neteq_performance: 0_pl_0_drift= 1527 ms
*RESULT neteq_performance: 10_pl_10_drift= 3511 ms
*RESULT neteq_performance: 0_pl_0_drift= 1533 ms
*RESULT neteq_performance: 10_pl_10_drift= 3518 ms
*RESULT neteq_performance: 0_pl_0_drift= 1523 ms
*RESULT neteq_performance: 10_pl_10_drift= 3503 ms
*RESULT neteq_performance: 0_pl_0_drift= 1524 ms
*RESULT neteq_performance: 10_pl_10_drift= 3514 ms
*RESULT neteq_performance: 0_pl_0_drift= 1534 ms
*RESULT neteq_performance: 10_pl_10_drift= 3501 ms
*RESULT neteq_performance: 0_pl_0_drift= 1530 ms
BUG=webrtc:7343,chromium:738852,chromium:738839
Change-Id: Idcbb276ca0ed27fff95164a73f1c1fa310175ee5
Reviewed-on: https://chromium-review.googlesource.com/563021
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18939}
Added documentation of thread expectations for video tracks and sources to the API.
BUG=None
Review-Url: https://codereview.webrtc.org/2964863002
Cr-Commit-Position: refs/heads/master@{#18938}
This CL removes code that supported the now removed
downstream dependencies in the support for using an
external audio processing module.
BUG=webrtc:7939
Review-Url: https://codereview.webrtc.org/2969213002
Cr-Commit-Position: refs/heads/master@{#18929}
Initial CL for this effort, with a working RTCVideoEncoder/Decoder for H264
(wrapping the VideoToolbox codec).
Some notes / things left to do:
- There are some hard-coded references to codec types that are supported by
webrtc::VideoCodec, cricket::VideoCodec, webrtc::CodecSpecificInfo etc
since we need to convert to/from these types in ObjCVideoEncoder/Decoder.
These types would need to be more codec agnostic to avoid this.
- Most interfaces are borrowed from the design document for injectable
codecs in Android. Some data in the corresponding C++ classes is discarded
when converting to the Obj-C version, since it has fewer fields. I have not
verified whether all data that we do keep is needed, or whether we might be
losing anything useful in these conversions.
- Implement the VideoToolbox codec code directly in the RTCVideoEncoderH264
classes, instead of wrapping webrtc::H264VideoToolboxEncoder / decoder.
Eliminates converting between ObjC/C++ types outside the ObjCVideoEncoder/
Decoder wrapper classes.
- List the injected codec factory's supported codecs in the list of codecs in
AppRTCMobile.
BUG=webrtc:7924
R=magjed@webrtc.org
Review-Url: https://codereview.webrtc.org/2966023002 .
Cr-Commit-Position: refs/heads/master@{#18928}
On Windows8/10, we prefer cropping desired window out from a whole screen
capture due to some reasons. The problem is Win10 has an invisible border
around the window. If we leave the border, it will expose background a bit.
This cl is about to always remove the border of desired window on Win8/10.
This will help a lot to capturing still windows during window sharing.
This cl still can't handle the background exposure issue when you move the
target window around during capturing. More investigation is needed.
BUG=chromium:737278
Review-Url: https://codereview.webrtc.org/2973853002
Cr-Commit-Position: refs/heads/master@{#18921}
In practice, this change will make AudioFrame::muted_ replicate the
explicit muted variable, passed as a pointer to NetEq::GetAudio.
BUG=webrtc:7944
Review-Url: https://codereview.webrtc.org/2965203002
Cr-Commit-Position: refs/heads/master@{#18914}
Lower then bitrate so that the delta between the highest layer of the
lower stream and lowest layer of the higher stream is not too large.
BUG=webrtc:4172
This is a reland of the following CL:
Review-Url: https://codereview.webrtc.org/2791273002
Cr-Commit-Position: refs/heads/master@{#18232}
Committed: dceb42da3e
https: //codereview.webrtc.org/2883963002
Review-Url: https://codereview.webrtc.org/2966833002
Cr-Commit-Position: refs/heads/master@{#18913}
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}
* Remove the ReleaseDecoder and Release methods that were used in combination with deleting the decoder object. Now simply deleting the object does the right thing.
* Remove 'friend' relationship between the two classes since they don't need to touch each other's state directly anymore.
* Use std::unique_ptr for holding pointers and transferring ownership.
These changes were previously reviewed here:
https://codereview.webrtc.org/2764573002/
BUG=webrtc:7361, 695438
Review-Url: https://codereview.webrtc.org/2966823002
Cr-Commit-Position: refs/heads/master@{#18908}
The implementation of this method did not follow the description in
the method comment. It was supposed to delete all packets in a range
[A, B], but if at least one packet in the buffer had a timestamp lower
than A, then no packets at all were discarded. This is now fixed.
BUG=webrtc:7937
Review-Url: https://codereview.webrtc.org/2969123003
Cr-Commit-Position: refs/heads/master@{#18903}
1) Function responsible for receiving feedback, digesting data and deciding switch scenarios.
2) Function which enters Startup mode.
3) Function which exits Startup mode.
4) Function which calculates, whether or not full bandwidth is reached.
BUG=webrtc:7713
Review-Url: https://codereview.webrtc.org/2924603002
Cr-Commit-Position: refs/heads/master@{#18901}
This change extends the definition of wired headset to also include USB
devices. The effect is that audio will now be routed to USB audio devices
when used in combination with AppRTCMobile.
BUG=webrtc:7931
Review-Url: https://codereview.webrtc.org/2971613003
Cr-Commit-Position: refs/heads/master@{#18889}
Original issue:
https://codereview.webrtc.org/2957073002/
Reason for reland:
Failed Android unit tests and failed Windows compile.
The tests seemed related at the time, but not after more consideration.
Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.
BUG=webrtc:7882
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2970793003
Cr-Commit-Position: refs/heads/master@{#18886}
This CL addresses the issue of echo leakage of low level
echoes by making the echo canceller more restrictive for
that scenario.
BUG=webrtc:7930
Review-Url: https://codereview.webrtc.org/2969943002
Cr-Commit-Position: refs/heads/master@{#18884}
Reason for revert:
The following, seemingly related, unit tests crash on Android32 (M Nexus5X).
org.webrtc.PeerConnectionTest#testCompleteSession
org.webrtc.PeerConnectionTest#testDataChannelOnlySession
A Windows build fails with a mysterious compile error.
Original issue's description:
> Add received audio/video call duration metrics based on packets.
>
> Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.
>
> BUG=webrtc:7882
>
> Review-Url: https://codereview.webrtc.org/2957073002
> Cr-Commit-Position: refs/heads/master@{#18881}
> Committed: 746749237aTBR=stefan@webrtc.org,aleloi@webrtc.org,asapersson@webrtc.org,holmer@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7882
Review-Url: https://codereview.webrtc.org/2972613002
Cr-Commit-Position: refs/heads/master@{#18882}
Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.
BUG=webrtc:7882
Review-Url: https://codereview.webrtc.org/2957073002
Cr-Commit-Position: refs/heads/master@{#18881}
The complexity test for the audio processing module have long proven
to give false alarms of complexity regressions for which no related
changes can be identified. Attempts to address that has improved the
that, but the tests do still give false alarms.
This CL deactivates the complexity tests until a better way of
testing this is available.
BUG=chromium:713507, webrtc:5846,webrtc:6685,webrtc:7712
Review-Url: https://codereview.webrtc.org/2897403006
Cr-Commit-Position: refs/heads/master@{#18879}
