BUG=webrtc:4690
Defined classes Stream, SendStream and ReceiveStream. Inherited existing stream classes from either SendStream or ReceiveStream.
This is a step towards having a Transport associated with streams instead of a Call. It will allow a lot of code in the Call to be media type agnostic.
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1226123005 .
Cr-Commit-Position: refs/heads/master@{#9591}
Tested that this doesn't break compatibility with Firefox or older
versions of Chrome, no matter which side generates the initial offer.
BUG=webrtc:2796
Review URL: https://codereview.webrtc.org/1219333002
Cr-Commit-Position: refs/heads/master@{#9589}
Using a right-sided (absolute value), truncated gaussian distribution originally with zero mean.
Currently truncated at x = 3 * std_dev.
Added expected value computation.
Modified jitter unittests accordingly.
BUG=webrtc:4848
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1237303002 .
Cr-Commit-Position: refs/heads/master@{#9587}
---- Dynamic receiving rate.
---- Dynamic packet-loss.
---- Dynamic objective function.
---- Dynamic available capacity.
---- Dynamic available capacity per flow.
---- Average delay Histogram with standard deviation or 5th/95th percentiles.
---- Average bitrate Histogram with error bars.
---- Optimal average bitrate dashed line.
---- Average packet-loss Histogram.
---- Total objective function Histogram.
Added media Pause/Resume methods to Video and TcpSender.
Modified LinkedSet: computing GlobalPacketLossRatio even if packet's sequence_number overflows.
Added small randomization to frame send times, modified bwe_test_framework_unittest accordingly.
Taking offset time into account for plotting.
Added nada_unittests.
Added bwe_unittests.
Added a RateCounter to BweReceiver (replaced ReceivingRate)
Added LossAccount.
Fixed NadaBweReceiver issue: using sender_timestamp instead of creation_time.
Fixed memory leaks.
Fixed int division rounding issues.
Supporting plots on bandwidth Estimators:
Logging received packet information on on SubClassesBweReceiver::ReceivePacket
Updating RateCounter, updating packet loss account and relieving LinkedSet when necessary.
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1202253003 .
Cr-Commit-Position: refs/heads/master@{#9585}
This patch tests separate iSAC encoder and decoder in more cases (32
kHz in addition to 16 kHz, and 30 ms adaptive and 60 ms nonadaptive).
In order to handle 32 kHz adaptive, the decoder needs to be told of
the encoder's sample rate (16 kHz worked already because that's the
default). And since we can't set the encoder's frame size without also
setting its bit rate, we need a way to set the decoder's bit rate as
well.
It turned out to be way too messy to continue verifying that the
bandwidth estimator does something reasonable in all these cases,
because it seems it doesn't. So the GetSetBandwidthInfo is now just
responsible for ensuring that split encoder/decoder behaves the same
as conjoined encoder/decoder; the job of verifying that the bandwidth
estimator does its job properly falls on some other test (that doesn't
exist yet).
Review URL: https://codereview.webrtc.org/1225093005
Cr-Commit-Position: refs/heads/master@{#9583}
Also fixed an arithmetic issue where a 0 0 3 at the end of the rbsp would include the 3 (that's not a legal bitstream anyway, so it probably wasn't a real bug, but it was incorrect).
This maintains the underflow fix from an earlier CL (https://codereview.webrtc.org/1219493004/). The overflow fix is virtually impossible to hit (hence no unit tests), but is there for strict correctness.
BUG=
Review URL: https://codereview.webrtc.org/1226203002
Cr-Commit-Position: refs/heads/master@{#9581}
2. provide an implementation for SetIceConnectionReceivingTimeout so that Chrome does not complain.
BUG=
Review URL: https://codereview.webrtc.org/1227843006
Cr-Commit-Position: refs/heads/master@{#9574}
Adds a class used to classify whether packet loss events are a single packet or multiple packets as well as how many packets have been lost. Also exposes a new function in the RtpRtcp interface to retrieve these statistics.
BUG=
Review URL: https://codereview.webrtc.org/1198853004
Cr-Commit-Position: refs/heads/master@{#9568}
- New option for computing variance that is more adaptive with lower complexity.
- Fixed related off-by-one errors.
- Added intelligibility unittests.
- Do not enhance if experiencing variance underflow.
R=andrew@webrtc.org, henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1207353002 .
Cr-Commit-Position: refs/heads/master@{#9567}
Fixes
..\..\third_party\webrtc/base/stringutils.h(295,49) : warning(clang): extra qualification on member "empty_str" [-Wmicrosoft]
No behavior change, but makes the code more standards-conformant.
BUG=chromium:505296
Review URL: https://codereview.webrtc.org/1228193002
Cr-Commit-Position: refs/heads/master@{#9562}
Comparing with 1 is less clear than using the input flags as
booleans.
BUG=5008276
Review URL: https://codereview.webrtc.org/1231663002
Cr-Commit-Position: refs/heads/master@{#9561}
This reduces the risk of getting a small initial estimate when doing combined a/v BWE, and the audio stream is received earlier than the video stream.
In addition a check is added to make sure a probe can't reduce the BWE.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1219303002 .
Cr-Commit-Position: refs/heads/master@{#9560}
Allows removing MediaRecorder which isn't in use apart from channel
unittests, along with it unittests for MediaRecorder that are flaky when
run in parallel can also go.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1219663008
Cr-Commit-Position: refs/heads/master@{#9558}
This fixes a bug in the WebRtcSessionDescriptionFactory where messages would be dropped or worse yet processed after the factory was deleted.
BUG=chromium:507307
Review URL: https://codereview.webrtc.org/1231823002
Cr-Commit-Position: refs/heads/master@{#9557}
To make it possible to exclude the examples when running
GYP on all.gyp.
The webrtc_examples.gyp already has an OS=="android" condition
inside it, so there's no need to check that before including it.
BUG=webrtc:4242
Review URL: https://codereview.webrtc.org/1196623006
Cr-Commit-Position: refs/heads/master@{#9556}
Note: Regarding the ICMP6_CLOSE_FUNC variable in winping.cc,
Icmp6CloseHandle does not exist, and IcmpCloseHandle is the correct way
to close an IPv6 handle. Therefore the existing code is correct to use
close_ on both types of connections and this variable is unnecessary.
BUG=505319
Review URL: https://codereview.webrtc.org/1231653003
Cr-Commit-Position: refs/heads/master@{#9555}
Relevant changes:
* src/third_party/icu: 7fe225d..c81a1a3
Details: 3ead4bc..f8d6ba9/DEPS
Clang version was not updated in this roll.
TBR=phoglund
Review URL: https://codereview.webrtc.org/1223013002
Cr-Commit-Position: refs/heads/master@{#9552}
To be used in tests that depend on specific field-trial settings without
overwriting the command-line flag for overriding field trials.
BUG=webrtc:4820
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1227653002
Cr-Commit-Position: refs/heads/master@{#9547}
These payload types aren't directly connected to any payload type, and
the payload type still has to be negotiated externally. As such these
constants are just a source of confusion.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1215603003
Cr-Commit-Position: refs/heads/master@{#9546}
This improves self-fairness and competing for resources with TCP flows.
BUG=4711
Review URL: https://codereview.webrtc.org/1151603008
Cr-Commit-Position: refs/heads/master@{#9545}
We have two histograms today that trigger on large jumps in either platform reported stream delays (WebRTC.Audio.PlatformReportedStreamDelayJump) or the system delay in the AEC (WebRTC.Audio.AecSystemDelayJump). The latter is the internal buffer size in the AEC.
The sizes of such jumps are of relevance since it can harm the AEC and even put it in a complete failure state. It is hard, not to say impossible, to tell how frequent it is.
Therefore, two complementary histograms are added; number of jumps in each metric.
This way we get a quick way to determine how often a jump occurs in general and also how frequent it is within a call.
This is solved by adding a counter for each metric.
The counter is activated either upon an event trigger or if we know for sure when the AEC is running.
Unfortunately, we can't rely on the destructor at the end of a call so we add a public API for the user to take on the action of calling it at the end of a call.
Tested locally by building ToT chromium including changes and three triggered jumps (200, 50 and 60 ms).
The stats picked up the 60 and 200 ms jumps as expected.
BUG=488124
R=asapersson@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1229443003.
Cr-Commit-Position: refs/heads/master@{#9544}
Bugs found by manual inspection of code, not by fuzzing or packet
replays. At least one of them confirmed by local fuzzing.
BUG=chromium:496094, webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1182793002
Cr-Commit-Position: refs/heads/master@{#9542}
