When an outgoing stream reset is sent, with sequence number A, and the
receiver can't perform it immediately, it will return IN_PROGRESS with
response sequence number A.
This is then retried with sequence number A+1, and the peer would then
possibly respond PERFORMED with response sequence number A+1.
Before this CL, whenever a request was sent that didn't immediately
succeed, it wouldn't increment its expected response sequence number.
So in the retry above, the socket would still expect the response
sequence number to stay at A, not at A+1.
Bug: webrtc:13994
Change-Id: I6f36d45229a7fb312e97ad15826e0377f4efb64f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261310
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36797}
This reverts commit 45361f78ed18c350b3edcaef19ae4c7cf167e95b.
Reason for revert: Perf alerts galore.
Original change's description:
> Calculate video stream max bitrate using expression.
>
> This replaces the ealier table-based caps.
> Apart from the VGA cap (now 1600kbps instead of 1700kbps), or if using
> "in between" resolutions, the caps are unchanged - but now cover high
> resolutions better.
>
> Bug: webrtc:14017
> Change-Id: I8649b528495d6c917e38ea8cb1a272df6c464c03
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260940
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36776}
Bug: webrtc:14017
Change-Id: I18ebc81c6054713c58d49bd227e37090686958c9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261309
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#36794}
To quote rfc6464:
The audio level for digital silence -- for a muted audio source, for
example -- MUST be represented as 127 (-127 dBov), regardless of the
dynamic range of the encoded audio format.
The behavior in webrtc is correct that digital silence is represented
with 127, but it is also possible to get a value of 127 for not quite
digitally silent audio buffer (as in, not strictly 0s).
Bug: webrtc:14029
Change-Id: I7ff8698a7e4d5c0960c667fd1cc961838e269456
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261244
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36793}
We already use RTC_NO_SANITIZE("cfi-icall") for most of the code and
it looks this one can be triggered recently with pw_loop_signal_event()
call.
Bug: webrtc:13659
Change-Id: I4dbb88f32de861e05be18254640db90b0f58c5e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261300
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36787}
* Structs with user-declared constructors are no longer considered
aggregates, so remove the declarations when possible
* Types of both arguments to "==" must match to avoid "ambiguous
function call" warning
* Various types of math involving enums are deprecated, so replace with
constexprs where necessary
* ABSL_CONST_INIT must be used on definition as well as declaration
* volatile memory may no longer be read from and written to by the same
operator, so replace e.g. "n++" with "n = n + 1"
* Replace an outdated check for no_unique_address support with
__has_cpp_attribute
* std::result_of(f(x)) has been removed, replace with
std::invoke_result(f, x)
Bug: chromium:1284275
Change-Id: I77b366ab1da7eb2c1e4c825b2714417c31ee5903
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261221
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36786}
This reverts commit 3180a5ad0663900a39adf4b9974052c356c835fe.
Reason for revert: Speculative revert due to failures in downstream tests.
Original change's description:
> dcsctp: Reset send queue when recreating TCB
>
> This is an issue found in fuzzer, and doesn't really happen in WebRTC
> as it never closes the connection and reconnects.
>
> The issue is that the send queue outlives any connection since you're
> allowed to send messages (well, enqueue them) before the association is
> fully connected. So the send queue is always present but the TCB
> (information about the connection) is torn down when the connection is
> closed for example. And the TCB holds the Stream Reset handler, which is
> responsible for e.g. keeping track of stream reset sequence numbers and
> such - which is tied to the connection.
>
> So to ensure that the Stream Reset Handler is in charge of deciding
> if a stream reset is taking place, make sure that the send queue is in
> a known good state when the Stream Reset handler is created.
>
> Bug: webrtc:13994, chromium:1320194
> Change-Id: I940e4690ac9237ac99dd69a9ffc060cdac61711d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261260
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Victor Boivie <boivie@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36779}
Bug: webrtc:13994, chromium:1320194
Change-Id: I89bb9cae60adc53902c1304e79047d18e72594a5
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261302
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Victor Boivie <boivie@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36783}
The webrtc::VideoStreamDecoderInterface was basically created as a public version of FrameBuffer2, but to hide the complexity of FrameBuffer2 it was also combined with decoding so that the public API could be reasonably simple to use. FrameBuffer3 has a simple API with a clear purpose, so its API can be exposed directly.
Bug: webrtc:14026
Change-Id: I81dc84b869e4d16c5e02feb5c876fbcede3d4a25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261181
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36781}
Repeatedly open and close data channels on a peer connection
to check that the channels are properly negotiated and SCTP
stream IDs properly recycled.
Bug: webrtc:13994, chromium:1320194
Change-Id: I244911abb5abaf0a290de07a0d790cd1edffe8cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260984
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36780}
This is an issue found in fuzzer, and doesn't really happen in WebRTC
as it never closes the connection and reconnects.
The issue is that the send queue outlives any connection since you're
allowed to send messages (well, enqueue them) before the association is
fully connected. So the send queue is always present but the TCB
(information about the connection) is torn down when the connection is
closed for example. And the TCB holds the Stream Reset handler, which is
responsible for e.g. keeping track of stream reset sequence numbers and
such - which is tied to the connection.
So to ensure that the Stream Reset Handler is in charge of deciding
if a stream reset is taking place, make sure that the send queue is in
a known good state when the Stream Reset handler is created.
Bug: webrtc:13994, chromium:1320194
Change-Id: I940e4690ac9237ac99dd69a9ffc060cdac61711d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261260
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36779}
If media_engine is not passed in init parameters, the PC can't handle
media, but can be used for datachannels. This CL adds testing that
datachannels work without media engine, and adds failure returns
to PeerConnection APIs that manipulate media when media engine is
not present.
Bug: webrtc:13931
Change-Id: Iecdf17a0a0bb89e0ad39eb74d6ed077303b875c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261246
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36778}
This replaces the ealier table-based caps.
Apart from the VGA cap (now 1600kbps instead of 1700kbps), or if using
"in between" resolutions, the caps are unchanged - but now cover high
resolutions better.
Bug: webrtc:14017
Change-Id: I8649b528495d6c917e38ea8cb1a272df6c464c03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260940
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36776}
When an SCTP stream is closing, a stream reset needs
to be sent from both ends.
The remote was not sending a stream reset and quickly
opening another stream with the same StreamID could
cause SCTP errors.
Bug: webrtc:13994
Change-Id: I3abc74ddc88b3fcf7e6495d76e7d77f52280b5d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260922
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36773}
When streams were to be reset, but there was already an ongoing
stream reset command in-flight, those streams wouldn't be properly
reset. When multiple streams were reset close to each other (within
an RTT), some streams would not have their SSNs reset, which resulted
in the stream resuming the SSN sequence. This could result in ordered
streams not delivering all messages as the receiver wouldn't deliver any
messages with SSN different from the expected SSN=0.
In WebRTC data channels, this would be triggered if multiple channels
were closed at roughly the same time, then re-opened, and continued
to be used in ordered mode. Unordered messages would still be delivered,
but the stream state could be wrong as the DATA_CHANNEL_ACK message is
sent ordered, and possibly not delivered.
There were unit tests for this, but not on the socket level using
real components, but just on the stream reset handler using mocks,
where this issue wasn't found. Also, those mocks didn't validate that
the correct parameters were provided, so that's fixed now.
The root cause was the PrepareResetStreams was only called if there
wasn't an ongoing stream reset operation in progress. One may try to
solve it by always calling PrepareResetStreams also when there is an
ongoing request, or to call it when the request has finished. One would
then realize that when the response of the outgoing stream request is
received, and CommitResetStreams is called, it would reset all paused
and (prepared) to-be-reset streams - not just the ones in the outgoing
stream request.
One cause of this was the lack of a single source of truth of the stream
states. The SendQueue kept track of which streams that were paused, but
the stream reset handler kept track of which streams that were
resetting. As that's error prone, this CL moves the source of truth
completely to the SendQueue and defining explicit stream pause states. A
stream can be in one of these possible states:
* Not paused. This is the default for an active stream.
* Pending to be paused. This is when it's about to be reset, but
there is a message that has been partly sent, with fragments
remaining to be sent before it can be paused.
* Paused, with no partly sent message. In this state, it's ready to
be reset.
* Resetting. A stream transitions into this state when it has been
paused and has been included in an outgoing stream reset request.
When this request has been responded to, the stream can really be
reset (SSN=0, MID=0).
This CL also improves logging, and adds socket tests to catch this
issue.
Bug: webrtc:13994, chromium:1320194
Change-Id: I883570d1f277bc01e52b1afad62d6be2aca930a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36771}
This ensures that only the compilation units that actually need
ChannelManager details can see it.
Bug: webrtc:13931
Change-Id: Iddd37580c0ceceba5b7095e84b981e6a525b2800
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261200
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36762}
The field trial will remain as a kill-switch for a few weeks while
decisions about sync decoding are being made.
Change-Id: I6034d25a129404e94ab8830f51e83667e285c785
Bug: webrtc:14003
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260327
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36760}
Metrics for calculating sent/recv key frame rate:
num_encoded_frames -- number of encoded frames
num_decoded_frames -- number of decoded frames
num_send_key_frames -- number of sent keyframes
num_recv_key_frames -- number of received keyframes
Metrics for frame size (in bytes) distribution:
recv_key_frame_size_bytes -- key frame size
recv_delta_frame_size_bytes -- delta frame size
The stats are only reported if the user so requests. This is
because the stats may not be useful for regression tracking,
but mainly for adhoc analysis.
Tested:
```
~/s/w/src (frame_stats↑1|✚1) $ autoninja -C out/Debug && out/Debug/test_support_unittests --gtest_filter="PeerConnectionE2EQualityTestSmokeTest.Smoke" 2>/dev/null
Note: Google Test filter = PeerConnectionE2EQualityTestSmokeTest.Smoke
[==========] Running 1 test from 1 test suite.
[----------] Global test environment set-up.
[----------] 1 test from PeerConnectionE2EQualityTestSmokeTest
[ RUN ] PeerConnectionE2EQualityTestSmokeTest.Smoke
...
RESULT num_encoded_frames: Smoke/alice-video= 31 count_biggerIsBetter
RESULT num_decoded_frames: Smoke/alice-video= 31 count_biggerIsBetter
RESULT num_send_key_frames: Smoke/alice-video= 2 count_biggerIsBetter
RESULT num_recv_key_frames: Smoke/alice-video= 2 count_biggerIsBetter
RESULT recv_key_frame_size_bytes: Smoke/alice-video= {1847,126} count_biggerIsBetter
RESULT recv_delta_frame_size_bytes: Smoke/alice-video= {1150.1034,393.66329} count_biggerIsBetter
...
RESULT num_encoded_frames: Smoke/charlie-video= 31 count_biggerIsBetter
RESULT num_decoded_frames: Smoke/charlie-video= 31 count_biggerIsBetter
RESULT num_send_key_frames: Smoke/charlie-video= 2 count_biggerIsBetter
RESULT num_recv_key_frames: Smoke/charlie-video= 2 count_biggerIsBetter
RESULT recv_key_frame_size_bytes: Smoke/charlie-video= {1847,126} count_biggerIsBetter
RESULT recv_delta_frame_size_bytes: Smoke/charlie-video= {1150.1034,393.66329} count_biggerIsBetter
...
[----------] 1 test from PeerConnectionE2EQualityTestSmokeTest (2196 ms total)
[----------] Global test environment tear-down
[==========] 1 test from 1 test suite ran. (2196 ms total)
[ PASSED ] 1 test.
```
Bug: webrtc:14019
Change-Id: I3d5ea0cf659a6299c6d94f5004cbf0c763b267e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260924
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36759}
This breaks the link from sdp_offer_answer.cc to channel.h.
Bug: webrtc:13931
Change-Id: I75608f75713bf4e69013ac5f5b17c19e53d07519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261060
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36757}
