I'll be doing some changes to code it tests (rtp_receiver_audio,
specifically) and want to make sure there are tests in place before I
touch anything.
Fixed test_api_audio not properly checking payload data. Required a
fix to LoopBackTransport in test_api to as to act like the regular
audio and video parts of WebRTC and separate payload from header data.
Also added a test for CNG and cleaned up constants.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2378403004
Cr-Commit-Position: refs/heads/master@{#14529}
The original CL (https://codereview.webrtc.org/2315633002) was
reverted since the fuzzer depended on gflags and files in the
resources folder; neither of this is allowed for a fuzzer test in
Chromium. This new version streamlines the dependencies, and changes
the test to generate a sinusoid input audio signal instead of reading
from a file.
Original commit message:
This CL introduces a new fuzzer target neteq_rtp_fuzzer that
manipulates the RTP header fields before inserting the packets into
NetEq. A few helper classes are also introduced.
BUG=webrtc:5447
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device
Review-Url: https://codereview.webrtc.org/2384423002
Cr-Commit-Position: refs/heads/master@{#14523}
Remove check on entropy_coding_mode_flag in PPS parser.
Parse entropy_coding_mode_flag from PPS and store it in the parser struct. Parse out extra data in NALU slices in case of entropy_coding_mode to avoid reporting incorrect QP.
BUG=webrtc:6338
Review-Url: https://codereview.webrtc.org/2373393002
Cr-Commit-Position: refs/heads/master@{#14522}
ClockTest.NtpTime was checking that the two methods for getting the
system time are returning a value that is within a fixed error margin
(100 ms) of each other. Unfortunately, even such a wide margin was
sometimes exceeded on heavily loaded machines
(https://build.chromium.org/p/client.webrtc/builders/Mac%20Asan/builds/9235/steps/system_wrappers_unittests/logs/stdio).
This CL changes the test to sandwich clock->CurrentNtp() between
clock->CurrentNtpTimeInMilliseconds(). This way the test will pass no
matter how much time elapses between the two method calls, as long as
the clock is monotonic.
Repeated test runs showed that there may be 1 ms worth of rounding error
between the two methods of getting time, so we have to allow that.
BUG=None.
Review-Url: https://codereview.webrtc.org/2393063002
Cr-Commit-Position: refs/heads/master@{#14520}
Reason for revert:
This CL breaks some downstream dependencies (contact me for more info).
Original issue's description:
> Delete unused code httprequest, httpclient, and socketpool.
>
> BUG=webrtc:6424
>
> Committed: https://crrev.com/4a255be3790a040cae2f6182ed70b7dd38c6839e
> Cr-Commit-Position: refs/heads/master@{#14514}
TBR=perkj@webrtc.org,pthatcher@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:6424
Review-Url: https://codereview.webrtc.org/2397673002
Cr-Commit-Position: refs/heads/master@{#14518}
Also rename some related minor methods. No functional changes
are intended/expected.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2391963002
Cr-Commit-Position: refs/heads/master@{#14513}
RTCStatsMemberInterface::Type's kBool and kSequenceBool.
This means that RTCStats-derived classes ("RTCStats-derived
dictionaries"[1]) can contain boolean and sequence of boolean members.
[1] https://w3c.github.io/webrtc-stats/
BUG=chromium:627816
NOTRY=True
Review-Url: https://codereview.webrtc.org/2387343002
Cr-Commit-Position: refs/heads/master@{#14509}
The test was heavily dependent on wall clock timing, so it ended up
being disabled in some build configurations. This CL switches it to use
a fake clock instead.
BUG=webrtc:5947
Review-Url: https://codereview.webrtc.org/2392613003
Cr-Commit-Position: refs/heads/master@{#14507}
NetEq already uses SdpAudioFormat internally; this CL adds an
AudioCodingModule::RegisterReceiveCodec overload that accepts
SdpAudioFormat, and propagates it through AcmReceiver into NetEq.
The intention is to get rid of the other ways to specify decoders and
always use SdpAudioFormat. (And eventually to do the same for encoders
too.)
NOTRY=true
BUG=5801
Review-Url: https://codereview.webrtc.org/2365653004
Cr-Commit-Position: refs/heads/master@{#14506}
This is a simple application limited region detector that is quite conservative
at the moment. We detect as being application-limited if we see sending rate
as less than 30% of the estimated bandwidth over 500 ms. The moment we detect
a single burst above 30% over a 100 ms period, we consider ourselves network
limited.
This class is currently not used. A follow up CL will leverage this to enable probing.
BUG=webrtc:6332
Review-Url: https://codereview.webrtc.org/2340763004
Cr-Commit-Position: refs/heads/master@{#14505}
Currently, BitrateProber does not scale higher than 2 Mbps to 6 Mbps. The actual
number is dependent on the size of the last packet. If a packet of around 250
bytes is used for probing, it fails above 2 Mbps.
BitrateProber now provides a recommendation on probe size instead of a
packet size. PacedSender utilizes this to decide on the number of packets
per probe. This enables BitrateProber to scale up-to higher bitrates.
Tests with chromoting show it stalls at about 10 Mbps (perhaps due to the
limitation on the simulation pipeline to deliver packets).
BUG=webrtc:6332
Review-Url: https://codereview.webrtc.org/2347023002
Cr-Commit-Position: refs/heads/master@{#14503}
Also update gyp dependency from rtc_base to rtc_base_approved.
BUG=None.
Review-Url: https://codereview.webrtc.org/2368203002
Cr-Commit-Position: refs/heads/master@{#14497}
This CL introduces changes that clearly demarcate
where we disable Unequal Protection in the FEC.
No functional changes are expected.
BUG=webrtc:5654
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2314743002 .
Cr-Commit-Position: refs/heads/master@{#14496}
Addresses a regression in the NetEq performance test.
# Added NOTRY due to android_arm64_rel being swamped.
NOTRY=True
BUG=chromium:651426
Review-Url: https://codereview.webrtc.org/2383723002
Cr-Commit-Position: refs/heads/master@{#14495}
RunPlayoutAndRecordingInFullDuplex fails sometimes on Android swarming
bots, presumably because the timing is hardware dependent.
This test ensures that audio starts pumping. The exact performance is
not that important.
R=kjellander@webrtc.org, henrika@webrtc.org
BUG=webrtc:6464
NOTRY=True
Review-Url: https://codereview.webrtc.org/2391563002
Cr-Commit-Position: refs/heads/master@{#14492}
- Change some member functions to be private. These were only
called by other private member functions.
- Replace DeleteMediaPackets() with direct calls to
media_packets_.clear()
- Rename GetFecPacketsAsRed to GetUlpfecPacketsAsRed.
No functional changes are intended by this CL.
BUG=webrtc:5654
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2305793003 .
Cr-Commit-Position: refs/heads/master@{#14491}
After https://codereview.webrtc.org/2386573002 changed where resolution
changes are handled, a few VideoSendStreamTests became flaky. They were
waiting for an InitEncode call they triggered, but sometimes were
getting one triggered by the resolution change when the first frame was
generated.
The fix was to make the tests wait for two InitEncode calls first -
one when the stream is created, and the second when the first frame was
encoded.
BUG=webrtc:6467, webrtc:6371
R=perkj@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2387293002 .
Cr-Commit-Position: refs/heads/master@{#14490}
- Place all member function definitions between test fixture
declaration and unit tests.
- Rename GenerateFrame -> PacketizeFrame.
No functional changes are intended by this CL.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2275303003
Cr-Commit-Position: refs/heads/master@{#14488}
Helper class for FlexFEC unit tests to come.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2282473002
Cr-Commit-Position: refs/heads/master@{#14487}
The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.
After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).
The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).
This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.
BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2380683005 .
Cr-Commit-Position: refs/heads/master@{#14485}
hbos and hta are already OWNERS of webrtc/stats/ and of rtcstats* files
(per-file rtcstats*=) in webrtc/api/. When the webrtc/api/stats/ folder
was created we forgot to add this OWNERS file (per-file OWNERS does not
apply to subfolders apparently).
BUG=chromium:627816
NOTRY=True
Review-Url: https://codereview.webrtc.org/2392633002
Cr-Commit-Position: refs/heads/master@{#14482}
This DCHECK is triggered by org.webrtc.PeerConnectionTest#testTrackRemoval.
BUG=webrtc:6465
Review-Url: https://codereview.webrtc.org/2389703002
Cr-Commit-Position: refs/heads/master@{#14481}
