These tests got flaky under the new jitter buffer.
Enhancements:
- Use send-side BWE.
- Let BWE ramp up before applying packet loss.
- Improve packet loss simulation for ULPFEC.
- Add delay to fake network pipe for FlexFEC.
(Not added for ULPFEC, since this makes those flaky...?)
- Add FlexFEC+NACK test, using RTX instead of "raw retransmits".
- Tighter checks of received packets' payload types and SSRCs.
TESTED=
$ ninja -C out/Debug video_engine_tests && third_party/gtest-parallel/gtest-parallel -r 1000 out/Debug/video_engine_tests --gtest_filter="*EndToEnd*Ulpfec*:*EndToEnd*Flexfec*"
ninja: Entering directory `out/Debug'
ninja: no work to do.
[12000/12000] TestWithNewVideoJitterBuffer/EndToEndTest.RecoversWithFlexfecAndNack/1 (14935 ms)
BUG=webrtc:7047
Review-Url: https://codereview.webrtc.org/2675573004
Cr-Commit-Position: refs/heads/master@{#16449}
Intervals when video is paused is no longer included in the stats:
"WebRTC.Video.BitrateSentInKbps"
"WebRTC.Video.MediaBitrateSentInKbps"
"WebRTC.Video.PaddingBitrateSentInKbps"
"WebRTC.Video.RetransmittedBitrateSentInKbps"
"WebRTC.Video.RtxBitrateSentInKbps"
"WebRTC.Video.FecBitrateSentInKbps"
BUG=webrtc:5283
Review-Url: https://codereview.webrtc.org/2536613002
Cr-Commit-Position: refs/heads/master@{#16447}
Improves possibility to debug the demo application using adb.
As an example, now allows 'adb run as' which is useful when profiling CPU. It enables us to do profiling on non-rooted devices (excluding details).
BUG=NONE
Review-Url: https://codereview.webrtc.org/2676983003
Cr-Commit-Position: refs/heads/master@{#16446}
AEC3 and AEC2 are separate submodules in APM. This CL ensures that AEC3
deactivates AEC2 if both are active at the same time.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2675863004
Cr-Commit-Position: refs/heads/master@{#16443}
These structs will be used for ORTC objects (and their WebRTC
equivalents).
This CL also introduces some minor changes to the existing implemented
structs:
- max_bitrate_bps uses rtc::Optional instead of "-1 means unset"
- "mime_type" turned into "name"/"kind" (which can be used to form the
MIME type string, if needed).
- clock_rate and channels changed to rtc::Optional, since they will
need to be for RtpSender.send().
- Renamed "channels" to "num_channels" (the ORTC name, which I prefer).
BUG=webrtc:7013, webrtc:7112
Review-Url: https://codereview.webrtc.org/2651883010
Cr-Commit-Position: refs/heads/master@{#16437}
Update libwertc AudioRtpSender::SetAudioSend with WEBRTC_WEBKIT_BUILD
This only introduces the WEBRTC_WEBKIT BUILD, inspired by WEBRTC_CHROMIUM_BUILD
macro. It is only defined by Webkit libwebrtc build system.
https://trac.webkit.org/changeset/210977
BUG=webrtc:7039
Review-Url: https://codereview.webrtc.org/2651273003
Cr-Commit-Position: refs/heads/master@{#16432}
Fix: Order of assignments is now correct, after being incorrect
due to an incorrect merge between
https://codereview.webrtc.org/2617373002/ and
https://codereview.webrtc.org/2589713003.
Improvement: Set parameters in more places, allowing for
correct reconfiguration. Add TODOs to point of minor issues
with current configuration.
TESTED=By locally patching an application using this code.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2660403004
Cr-Commit-Position: refs/heads/master@{#16431}
The timer is only used for logging purposes and we can save CPU by
checking the volume less often.
BUG=webrtc:7096
Review-Url: https://codereview.webrtc.org/2669323003
Cr-Commit-Position: refs/heads/master@{#16430}
We have seen that PostTask can consume some CPU and the way we used it
before (logging only) in the ADB is not worth the cost we see when
profiling.
This CL simply moves frequent (trivial) stat updates from the task queue
to the native threads to avoid calling PostTask in each callback.
The reason for doing so before was to avoid locks but we can live without
them since races are benign here.
BUG=webrtc:7096
Review-Url: https://codereview.webrtc.org/2663383004
Cr-Commit-Position: refs/heads/master@{#16429}
This CL replaces the previously-hardcoded DTLS timeout value of 50 ms with one that is picked dynamically as 2x ICE RTT (clamped between 50 and 3000 ms to keep it reasonable).
No tests yet - I'm working on these, but sending out this CL early to get feedback. I've tested it manually and on slow networks it results in successful DTLS setup in a single attempt - instead of clogging up the slow link with multiple retransmissions as done previosly.
BUG=webrtc:7099
Review-Url: https://codereview.webrtc.org/2670113002
Cr-Commit-Position: refs/heads/master@{#16426}
As it is, the test fails to compile on some downstream compilers with the following error:
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc:316:25: error: lambda capture 'kPayloadLength' is not required to be captured for this use [-Werror,-Wunused-lambda-capture]
.WillOnce(Invoke([kPayloadLength, kFirstSequenceNumber, kFirstTimestamp,
^
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc:316:41: error: lambda capture 'kFirstSequenceNumber' is not required to be captured for this use [-Werror,-Wunused-lambda-capture]
.WillOnce(Invoke([kPayloadLength, kFirstSequenceNumber, kFirstTimestamp,
^
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc:316:63: error: lambda capture 'kFirstTimestamp' is not required to be captured for this use [-Werror,-Wunused-lambda-capture]
.WillOnce(Invoke([kPayloadLength, kFirstSequenceNumber, kFirstTimestamp,
^
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc:317:25: error: lambda capture 'kFirstReceiveTime' is not required to be captured for this use [-Werror,-Wunused-lambda-capture]
kFirstReceiveTime](const SdpAudioFormat& format,
BUG=webrtc:7107
Review-Url: https://codereview.webrtc.org/2672823002
Cr-Commit-Position: refs/heads/master@{#16422}
If an application sets a non-null value in RTCConfiguration.iceCheckMinInterval, we do not sent STUN pings more often than that. This is useful for bandwidth constrained scenarios.
This CL also increases the maximum STUN ping timeout to 60 seconds up from its previous value of 5 (which meant that a ping response received 5 seconds later would not be counted), and allows the RTT estimate to go up to 60 seconds from its previous limit of 3. RTTs above 3 seconds are possible on mobile links. (webrtc:7109)
This CL was originally written by pthatcher@, I am just submitting it after a minor cleanup.
BUG=webrtc:7082, webrtc:7109
Review-Url: https://codereview.webrtc.org/2670053002
Cr-Commit-Position: refs/heads/master@{#16421}
Since we can't know when a H264 frame really starts we instead group all packets together by timestamp when a frame seems to be complete (only in the case of H264).
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2675693002
Cr-Commit-Position: refs/heads/master@{#16419}
Proper error handling was missing, using VERIFY to crash in debug
builds, while release builds would ignore the error and leak the
attribute memory. The check of attribute type consistency was changed
to a RTC_DCHECK.
Also removes a large number of uses of the deprecated VERIFY macro.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2665343002
Cr-Commit-Position: refs/heads/master@{#16413}
Adds ignore for all lint errors in Chromium code. Changes minimum SDK for
instrumentation tests to 16 from 14. Adds TargetApi annotations.
BUG=webrtc:6597
Review-Url: https://codereview.webrtc.org/2670473004
Cr-Commit-Position: refs/heads/master@{#16412}
It's currently not possible on Mac and iOS due to libtool. See webrtc:6418
for more info.
BUG=webrtc:6418
NOTRY=True
Review-Url: https://codereview.webrtc.org/2367313002
Cr-Commit-Position: refs/heads/master@{#16411}
Previously in the spec, there was a createDtmfSender method on
PeerConnection, but that's been replaced by a "dtmf" attribute
on RtpSender, which allows getting a DTMF sender without having
an audio track.
This also simplifies the code slightly, since tracks are now not
necessary for identification.
BUG=webrtc:4180
Review-Url: https://codereview.webrtc.org/2666853002
Cr-Commit-Position: refs/heads/master@{#16409}
BUG=webrtc:7097
TEST=Set "ptime=120", try WebRTC calls over custom build Chromium with and without Opus 120ms. Try both Chromium w <-> Chromium w and Chromium w <-> Chromium w/o
Review-Url: https://codereview.webrtc.org/2668633004
Cr-Commit-Position: refs/heads/master@{#16408}
Useful if bitrate probing is to be used with audio streams.
BUG=webrtc:7043
Review-Url: https://codereview.webrtc.org/2652893004
Cr-Commit-Position: refs/heads/master@{#16404}
Use AdaptDown/AdaptUp instead of ScaleDown/ScaleUp, since we may want to
adapt using other means than resolution.
Also, extend vie_encoder with unit test that actually uses frames scaled
to resolution as determined by VideoAdapter, since that seems to be the
default implementation.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2652893015
Cr-Commit-Position: refs/heads/master@{#16402}
Note:
* PLR is calculated over all of the known packets.
* RPLR is calculated over all of the known packet *pairs*. That is, only over sets of subsequent packets where the reception status is known for both.
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2629883003
Cr-Commit-Position: refs/heads/master@{#16401}
