This reverts commit 998e9bd5c55de253106b697af691169853a4e91f.
Reason for revert: Breaks downstream projects because some headers
have been renamed without providing a forward header for backwards
compatibility.
Original change's description:
> Linux capturers: organize X11 and Wayland implementations into separate folders
>
> Bug: webrtc:13429
> Change-Id: I2db727797c2ca2bd85937ff732ce3f68bb45469a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238173
> Reviewed-by: Mark Foltz <mfoltz@chromium.org>
> Commit-Queue: Mark Foltz <mfoltz@chromium.org>
> Cr-Commit-Position: refs/heads/main@{#35471}
TBR=tommi@webrtc.org,sprang@chromium.org,mfoltz@chromium.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com,grulja@gmail.com
Change-Id: I2aadfeb30151fcbe1a8c05e856be989d60bb10a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:13429
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239821
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#35472}
This patch adds reporting of relay protocol,
i.e how a client connect to the turn server.
This is added in the old stats api...cause there
are clients still using it.
Bug: none
Change-Id: Iac7fe3e3de0ba42d5897c304ebbae368edf498fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239640
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35469}
There are cases for each of these getters where other code later takes
a reference to the passed object, meaning that these getters should be
returning a refptr. To prevent additional overhead from places that
simply access the getter to call additional methods without needing to
worry about taking a ref, the return values are converted to const refs.
Bug: webrtc:13465
Change-Id: Ib27969c7f5ef9d6aadf3c95ac171ae6e778cdbfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239720
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35465}
This reverts commit 258ed1a38ad9d4f0da798c40b6976eff2dce864f.
Reason for revert: This cl breaks the dumping of the perf proto histogram.
Original change's description:
> Use gtest_parallel with 1 worker for webrtc_perf_tests.
>
> This will enable test results to be uploaded to ResultDB.
>
> Bug: b/197492097
> Change-Id: Iec28520c4cd8f35fcff2cbd105a4b851ef41b9fc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239641
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Christoffer Jansson <jansson@google.com>
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Cr-Commit-Position: refs/heads/main@{#35458}
TBR=mbonadei@webrtc.org,jansson@google.com,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com,jleconte@webrtc.org
Change-Id: Ic4ab03d0e7f8bc1ce799d30e74420200d2f8f224
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/197492097
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239644
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#35463}
If the number of samples does not fit in an AudioFrame, we should return
kSampleUnderrun to avoid crashes further downstream.
Bug: chromium:1265806
Change-Id: Ie94e1de53810167fd9b52ade72b3cb669a2a4f06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238666
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35459}
This will enable test results to be uploaded to ResultDB.
Bug: b/197492097
Change-Id: Iec28520c4cd8f35fcff2cbd105a4b851ef41b9fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239641
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#35458}
Changing to an index for-loop (instead of using std::transform) allows the compiler (clang for x86 at least) to use 3 different registers in the loop rather than just 1, resulting in less pipeline stall (I'd assume). Interestingly, the compiler unrolls the loop(s) completely in both cases.
Bug: None
Change-Id: I586773bc525e91bb6eb6638d5399928482306b9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239364
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Christian Schuldt <cschuldt@google.com>
Cr-Commit-Position: refs/heads/main@{#35453}
Current rate statistic tracker has assumption, the tracking window will
always be full after first filled up. This assumption looks not always
true. One example is the input_framerate_ tracker inside
video_stream_sender.cc which is used for setup frame droper and encoder.
Whenever there is a gap in video stream, like mute/unmute,
pacer pause/unpause etc. The fps detected from the rate_statistics
becomes samples_filled_partial_window / full_window_size, which could
be extremely low for a while. This creates a misalignment between the
fps we told encoder/frame dropper, and the real fps we fed into them,
which causes short-term serious overshot and very bad experience on
delay, avsync, congestion etc. This may also depends on how fast
encoder could react to the gap between set fps and real fps, but
libvpx and openh264 at least cannot handle this well.
So propose a fix to update first timestamp after tracker window
drained. This will give more accurate fps estimate similar based on
active window after sample gets drained
Bug: webrtc:13403
Change-Id: I96792c11091fe8bfa63e669f4360a3b3e95593e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237720
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35447}
This reverts commit 61a8d9caaa31ab4ef953415882f97be5a4248774.
Reason for revert: We have identified some downstream regressions caused by this change (https://crbug.com/webrtc/13437).
Original change's description:
> Call: Deduplicate SentPacket notifications
>
> When bundling is in effect, multiple senders may be sharing the same
> transport. It means every |sent_packet| will be multiply notified from
> different channels, WebRtcVoiceMediaChannel or WebRtcVideoChannel.
> Record |last_sent_packet_| to deduplicate redundant notifications to
> downstream objects.
>
> This CL reduces 50% PostTask/Wakeup of Dynamic Mode Pacer.
>
> [1] https://datatracker.ietf.org/doc/html/rfc8829#section-4.1.1
> [2] https://datatracker.ietf.org/doc/html/rfc8843
>
> Bug: webrtc:13417
> Change-Id: Ib121d5af07abe208bd7d36715a234f48cdabb032
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238720
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35417}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:13417
Change-Id: Ib1230fa07db56c33941a5b529a28f83d6d08d74d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239441
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Owners-Override: Jakob Ivarsson <jakobi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35442}
When a frame is assembled `packet_infos` is moved and must be
re-initialized before potentially being used in another iteration of the
loop. Clear `packet_infos` immediately instead of relying on it being
implicitly cleared in the next iteration of the loop.
Bug: None
Change-Id: I954aaa0c6df296cc2a27b3ab496e49fac200f135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238981
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35441}
the maximum used in practice is multiopus with
6 or 8 channels. 24 is the maximum number of channels
supported in the audio decoder.
BUG=chromium:1265806
Change-Id: Iba8e3185a1f235b846fed9c154e66fb3983664ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/main@{#35440}
This method is no longer useful after a previous refactoring, but it was
not removed from the interface.
Bug: webrtc:13444
Change-Id: I9c4761e8503acdec06c16cc37c2a804d4913eac8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239366
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35439}
This test being flaky impeded the landing of
https://webrtc-review.googlesource.com/c/src/+/239126. Fix by
ensuring the test's OnSendRtp guts don't execute past all streams
stopped.
Bug: None
Change-Id: Ie8aefb3bb03c09d2a9514acecd162e7c079c77c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239363
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35432}
The encoders wrapped in VideoStreamEncoder grossly over-estimates
available bitrate when capture FPS falls close to zero, and frames
re-commence highly frequent delivery. Avoid this by moving the input
RateStatistics inside VSE into the frame cadence adapter, and changing
the reported framerate under zero-hertz encoding mode to always return
the configured max FPS.
Bug: chromium:1255737
Change-Id: Iaa71ef51c0755b12e24e435d86d9562122ed494e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239126
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35431}
This avoids copying the payload at all. Future CL will change the
transport.
In performance tests, memcpy was visible in the performance profiles
prior to this change.
Bug: webrtc:12943
Change-Id: I507a1a316165db748e73cf0d58c1be62cc76a2d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236346
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35428}
