The IvfFileWriter logs a warning in case frames have a different
resolution compared to the one of the first frame in the file.
While this is an issue, since the IVF header will have the resolution
of the first frame, in reality this is not a problem (e.g. tools like
VLC can open and play the IVF without issues).
For this reason, let's remove the log which gets printed for each
frame.
Bug: b/282678729
Change-Id: I540cd1b6ce4f5d888737725e7615918aa126647f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305280
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40069}
Also fix one instance where access was done wrongly.
This makes certain that the split between MediaChannel types is respected
for this variable (prior to splitting the actual C++ types).
Bug: webrtc:13931
Change-Id: I8cf48ff5eddef35fda75533bb9c5075083c4ab16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305220
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40065}
by not starting the receive stream whenever it is creating.
Instead, this is controlled by the direction of the media content.
BUG=webrtc:11013
Change-Id: Iaaa0ac0aa9f90a4be776a1348f53a0f9c2b84d99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304661
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40064}
Store just the fields that are used by RtpTransportController
Avoid redundand map lookup when updating that information
Bug: webrtc:13757
Change-Id: I1e5a000557bde1735979c1cf8fa762936a64ffd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305023
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40063}
The fcntl() call has variable arguments, therefore we need to pass 0 to
specify there are no other arguments for this call, otherwise we might
end up with an argument that is random garbage.
Bug: webrtc:15174
Change-Id: I34f16a942d80913b667d8ade7eed557b0233be01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305120
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#40060}
Before the channel split, the RTP modes were set by reading the
configuration of the send codec. After the split, this is done
via the SetReceiverFeedbackParams function.
This CL adds caching those parameters so that they are applied
to receive streams created after the SetReceiverFeedbackParams call.
Bug: webrtc:13931
Change-Id: I92eb651e5dd1ec68aca7f6a162e3521eb835a11d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305021
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40056}
RTX padding packets sent before media packets can legitimately have no
timestamps set (they are 0). Writing the TransmissionOffset extension
with capture time 0 will overflow once current time exceeds ~3 minutes.
Bug: webrtc:15172
Change-Id: I4dd1f341802d45016549b330f0e08cd3a00cfa19
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305020
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40055}
Step 1: Make reading RTCStatsMember look the same as reading
absl::optional (this CL).
Step 2: Migrate uses of "is_defined()" to "has_value()".
Step 3: Delete "is_defined()".
Step 4: Make RTCStatsMember+Interface an implementation detail of
RTCStats::Members(), only used for abstract iteration ("for
each metric"). Lazy instantiate it upon Members().
Step 5: Replace RTCStatsMember with absl::optional for use in RTCStats
dictionaries (rtcstats_objects.h/cc).
Bug: webrtc:15164
Change-Id: I5a2c9fe56707e3c7d89e8ea62fb37171ae806a7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304840
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40048}
The flexibility offered by the GN `rtc_jsoncpp_root` should be enough
to wire a different version of jsoncpp.
Bug: b/281848049
Change-Id: I8af39fec2406e27c3af7b7ec1949a4762dce762f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304861
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40045}
With intent to fully replace RtcpBandwidthObserver interface
and half of the TransportFeedbackObserver interface
RtcpBandwidthObserver interfaces passed bitrate and time variables as
raw ints, NetworkLinkRtcpObserver uses more expressive types.
Bug: webrtc:13757, webrtc:8239
Change-Id: I0a8c8de626fbe0c190a0a1a9f6733d863494401c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304700
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40043}
OpenSSL removed ability to generate C code:
a18cf8fc63
CL rewrites generation script to use pure Python asn1crypto library.
The changes in generated code leading to huge diff in generated file:
- Certificate array names are based on certificate fingerprints instead
of semi-human readable names, which were not referenced externally;
- Order of arrays in generated file matches the order of certificates
as they are appeared in source pem file. Previously re-ordering happen
due to writing temporary files on disk;
Bug: webrtc:11710
Change-Id: Ie7a97b3658f6ccb397f0fd0c21d341934a2cc12e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304642
Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40039}
If the caller calls RegisterObserver() on the network thread while the
state is not kOpen but there are queued received data, those received
data will be immediately delivered to the observer before the state is
transitioned to kOpen, which may break the observer's assertions and
cause problems.
The problem turns out to be that, when SctpDataChannel::RegisterObserver
calls DeliverQueuedReceivedData(), the data will be passed to the
observer without checking the |state_| first, meanwhile
SctpDataChannel::UpdateState does effectively check the state and
null-check |observer_| before delivering the received data. This CL
fixes this by simply making DeliverQueuedReceivedData() also check
`state_ == kOpen`. In case the state transitions to kOpen after
RegisterObserver() is called, the first DeliverQueuedReceivedData()
call will be no-op, while the second DeliverQueuedReceivedData() call
will do the work.
Bug: chromium:1442696
Change-Id: If25ce6a038d704939b1a8ae73d7ced110448b050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304687
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40036}
The VCMReceiveStatisticsCallback interface is both implemented (by ReceiveStatisticsProxy) and called (by VideoStreamBufferController) in `video/`, so there's no reason it should be declared in `modules/video_coding`. I also took the opportunity to update the name.
No functional changes are intended by this change, but following CLs will make some changes.
Bug: webrtc:15085
Change-Id: Ib8da30ca56675e4f638d0b9778c329b9c1138acf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304662
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40034}
This is especially needed for datachannels that get created in
response to an OPEN message and RegisterObserver() is called from
within the OnDataChannel callback. More details in the associated bug.
Bug: webrtc:15165
Change-Id: I833db6c3c503623d482808dc5a02f03b9821a5f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304721
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40032}
which avoids an infinitely growing SDP if the remote end rejects
the datachannel section. This will reactivate the m-line even if
all datachannels are closed.
BUG=chromium:1442604
Change-Id: If60f93b406271163df692d96102baab701923602
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304241
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40029}
RtpRtcpInterface::RTT follows discouraged style of using return values,
uses raw integers to represent time delta,
and returns values that no code uses (min, max, average RTT)
added LastRtt function addresses all these stylistic issues.
Bug: webrtc:13757
Change-Id: Iaf947dd1b7139026f2beb991e69634c606c6b608
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304520
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40028}
The field trial has been enabled-by-default for several years, I
suspect it was needed during its development but there doesn't seem to
be any reason to maintain it going forward.
Its very existence blocks our long term objective to have our APIs
behave according to the W3C standards and any apps still depending on
it, if there are any, should make sure to use the APIs correctly
instead. I assume they already do any any references to this is us
forgetting to clean things up.
Bug: webrtc:15161
Change-Id: I4a6a44a15219d2e045f3d8d857b5197a064f049c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304660
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40025}
The web compat requirement that was the reason for keeping
is now solved in Chromium and its stats bindings.
BUG=webrtc:9674
Change-Id: Ifb722769414b2bcc5f4d36d7dff87a875336e039
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303860
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40024}
These tests were failing when building WebRTC against OpenSSL instead of
BoringSSL. The reason is that OpenSSLStreamAdapter::SSLVerifyCallback in
the BoringSSL mode returns the full cert_chain by calling
SSL_get0_peer_certificates. This API does not exist in OpenSSL, instead
only a single certificate is fetched via X509_STORE_CTX_get0_cert.
ifdef out the parts of the test that assert on cert[1] and cert[2].
An alternative but more involved way to fix these tests could be to use
X509_STORE_CTX_get1_chain to fetch the full chain on the OpenSSL path.
Bug: webrtc:15153
Change-Id: I1ede6a3c5a63d4afd2de849f5e44fcd67592aa3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40022}
