4b7024b572
Revert "dcsctp: Use rtc::CopyOnWriteBuffer"
...
This reverts commit 2db59a6584eca54245794a0e657ca9ded9e6707f.
Reason for revert: Causes msan-issue in crc32c, reading uninitialized
memory.
Bug: webrtc:12943, chromium:1275559
Change-Id: I05f1012d896aeaca86c4562e0df15fa7ea326d60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239560
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Victor Boivie <boivie@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35461}
2021-12-02 12:33:46 +00:00
4ad09762da
Delete legacy RtpPacketSendInfo::ssrc field
...
Bug: webrtc:12713
Change-Id: I8785eaabbb8725fb119067f5897ef245e6187996
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239642
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35460}
2021-12-02 11:59:37 +00:00
500d6e7f14
Return kSampleUnderrun if the number of samples does not fit in an AudioFrame.
...
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}
2021-12-02 11:48:17 +00:00
258ed1a38a
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}
2021-12-02 10:40:46 +00:00
0763ccc8e3
Don't configure SVC params without per layer bitrate configured.
...
Change-Id: Ieb200ce1a710078e380047ed8af73db0c5e0c751
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239442
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35457}
2021-12-02 09:52:32 +00:00
b5dcbbf9a6
Roll chromium_revision 28e5940d2a..ab1e4b1c09 (947206:947322)
...
Change log: 28e5940d2a..ab1e4b1c0928e5940d2a..ab1e4b1c094056842853..122962bd6310c5306b5f..83e1eefbf91da5f9f6aa..58323b68465c6ecb9673..ce98fdf6a94c819f4a1a..4c3d57f854e685543855..fc79a334664f3656c38b..e92d3d404328e5940d2a..ab1e4b1c09https://webrtc-review.googlesource.com/c/src/+/239620
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2021-12-02 02:25:17 +00:00
e889b5f801
Roll chromium_revision 27e299b2f4..28e5940d2a (947091:947206)
...
Change log: 27e299b2f4..28e5940d2a27e299b2f4..28e5940d2af510ba5363..4056842853f7800e3695..10c5306b5f665b74f7d1..89f2e82120c936d73ff7..a281fd3d4cf3f80e3620..1da5f9f6aa1018546e10..5c6ecb9673daa9aa1570..4c819f4a1a9296f1edfe..4a9b683ccb023af79284..4f3656c38b27e299b2f4..28e5940d2ahttps://webrtc-review.googlesource.com/c/src/+/239600
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Cr-Commit-Position: refs/heads/main@{#35455}
2021-12-01 22:27:12 +00:00
95e6b8d6f7
Roll chromium_revision abea99b512..27e299b2f4 (946947:947091)
...
Change log: abea99b512..27e299b2f4abea99b512..27e299b2f407fb44c465..f510ba5363acd04d0ee9..f7800e3695b521528b20..f3f80e3620f58d7ccc17..1018546e10f67165847e..daa9aa157070f73380a5..9296f1edfede32b0cbf7..023af79284abea99b512..27e299b2f4https://webrtc-review.googlesource.com/c/src/+/239525
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Cr-Commit-Position: refs/heads/main@{#35454}
2021-12-01 18:30:51 +00:00
794cc227dc
Optimize RenderBuffer.
...
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}
2021-12-01 14:52:21 +00:00
59b8a46d3f
Fix low_bandwidth_audio_test on Windows bots.
...
Bug: b/197492097
Change-Id: I29ae4d699c3c928f9d22da8cf9026c462f3afa7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239540
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@{#35452}
2021-12-01 13:04:52 +00:00
408a866cfc
Roll chromium_revision 39e73a0204..abea99b512 (946822:946947)
...
Change log: 39e73a0204..abea99b51239e73a0204..abea99b512241200489d..07fb44c4656bc4ab94a3..acd04d0ee9f3bfdf6913..b521528b20dddf6836b0..f58d7ccc1717ad36f7fb..f67165847ead7c145263..e685543855bf66d373ae..edf883ad2d3702b35757..de32b0cbf739e73a0204..abea99b512https://webrtc-review.googlesource.com/c/src/+/239522
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Cr-Commit-Position: refs/heads/main@{#35451}
2021-12-01 12:58:02 +00:00
ff05f1fd89
TaskQueuePacedSender: Process packets after outstanding data update
...
Bug: webrtc:10809
Change-Id: I31785c529d331891830777c93d7067f96580ea39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238967
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35450}
2021-12-01 11:59:01 +00:00
8d89f461c9
Remove --isolated from mb.py
...
This option is not supported now.
Bug: chromium:1271841
Change-Id: I27dd1531589ac497ef345fff41897a087ace4362
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239461
Reviewed-by: Takuto Ikuta <tikuta@google.com >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Auto-Submit: Junji Watanabe <jwata@google.com >
Cr-Commit-Position: refs/heads/main@{#35449}
2021-12-01 06:15:41 +00:00
d4516d86d3
Roll chromium_revision 1932b16547..39e73a0204 (946682:946822)
...
Change log: 1932b16547..39e73a02041932b16547..39e73a0204dbc5f6096e..241200489dc82934398a..6bc4ab94a36d5a3ef834..f3bfdf6913f5addf91c9..dddf6836b02ea1121894..17ad36f7fbee2ad61263..0ae29c99d1https://chromium.googlesource.com/catapult.git/+log/bd8b0c2c47..5695a8426c
* src/third_party/depot_tools: 320a1caf81..ad7c145263abe1a8bc88..70f73380a50fdcc84d75..3702b357571932b16547..39e73a0204https://webrtc-review.googlesource.com/c/src/+/239480
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Cr-Commit-Position: refs/heads/main@{#35448}
2021-12-01 03:00:30 +00:00
4774a9fcb8
Fix rate statistic when time window running out of samples
...
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}
2021-11-30 23:57:40 +00:00
f0290eb6fd
Roll chromium_revision b956d14f88..1932b16547 (946538:946682)
...
Change log: b956d14f88..1932b16547b956d14f88..1932b165472ade335c68..dbc5f6096ecc2a3e818b..c82934398ab4943c7220..6d5a3ef8342f3258e975..f5addf91c9368033fffc..2ea1121894https://chromium.googlesource.com/catapult.git/+log/8cf0922154..bd8b0c2c47
* src/third_party/depot_tools: 804165b445..320a1caf813b45f564e9..03380482ce9e18a83edc..abe1a8bc88fad05cbf2a..0fdcc84d75b956d14f88..1932b16547https://webrtc-review.googlesource.com/c/src/+/239434
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2021-11-30 22:51:32 +00:00
063bb384f8
sdp: temporarily raise mid limit to 32
...
to avoid breaking existing deployments. Also measure usage.
BUG=webrtc:12517
Change-Id: Ic38f1b45e79e46da9ff6fe927b0c5351443ccd96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239188
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
Cr-Commit-Position: refs/heads/main@{#35445}
2021-11-30 21:05:51 +00:00
0ce7f165fd
Roll chromium_revision add2d2ddee..b956d14f88 (946436:946538)
...
Change log: add2d2ddee..b956d14f88add2d2ddee..b956d14f8845c59f126d..2ade335c68726a5fd8ba..cc2a3e818b302ce56025..b4943c722032c8bef651..2f3258e975755e7d848a..368033fffc3dd5b80bc4..b52786888db9dad73621..9e18a83edc2128c47e06..fad05cbf2aadd2d2ddee..b956d14f88https://webrtc-review.googlesource.com/c/src/+/239432
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2021-11-30 18:47:14 +00:00
f42e00097d
Roll chromium_revision a3e750c9b0..add2d2ddee (946315:946436)
...
Change log: a3e750c9b0..add2d2ddeea3e750c9b0..add2d2ddeee6e0ee5bae..45c59f126de0d5dbffaa..726a5fd8baaf6e3bb63e..302ce5602569383604c0..32c8bef65192767b0feb..755e7d848ae2f3978937..bb9216085f0e7a87dbf5..2128c47e06a3e750c9b0..add2d2ddeehttps://webrtc-review.googlesource.com/c/src/+/239430
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Cr-Commit-Position: refs/heads/main@{#35443}
2021-11-30 14:31:14 +00:00
90cc4fe0e9
Revert "Call: Deduplicate SentPacket notifications"
...
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}
2021-11-30 11:49:51 +00:00
af5ca5af75
Fix potential use after move in RtpVideoStreamReceiver
...
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}
2021-11-30 11:20:21 +00:00
d58ac5adf8
sdp: reject large number of channels
...
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}
2021-11-30 11:01:14 +00:00
6c167d8278
Remove NetEq::Create.
...
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}
2021-11-30 10:31:16 +00:00
0680c4871c
Roll chromium_revision 22cbef27a6..a3e750c9b0 (946167:946315)
...
Change log: 22cbef27a6..a3e750c9b022cbef27a6..a3e750c9b0f8009dc41e..e6e0ee5bae2f9e5b5867..e0d5dbffaad11bb31289..af6e3bb63e42caab826d..69383604c017c7be145e..92767b0febhttps://chromium.googlesource.com/catapult.git/+log/75423c310e..8cf0922154
* src/third_party/depot_tools: 51d99b2f3a..804165b44525c7e11129..0e7a87dbf522cbef27a6..a3e750c9b0https://webrtc-review.googlesource.com/c/src/+/239425
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2021-11-30 04:30:11 +00:00
c56645c0cd
Roll chromium_revision 0e21474081..22cbef27a6 (946029:946167)
...
Change log: 0e21474081..22cbef27a60e21474081..22cbef27a61d1c400730..f8009dc41e245517235e..2f9e5b58675bea6e0b34..d11bb31289bebfd47d77..42caab826d51f3bc37e1..17c7be145e02f4003fdb..51d99b2f3a64e26ad3a2..3b45f564e9889d9a924c..b9dad7362169300210d4..25c7e111290e21474081..22cbef27a6https://webrtc-review.googlesource.com/c/src/+/239421
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2021-11-29 22:39:01 +00:00
6dd4997950
Fix TestTurnServer to not depend on rtc::Thread::socketserver() accessor
...
Updated all calls to constructor to pass appropriate
rtc::SocketFactory explicitly.
Bug: webrtc:13145
Change-Id: Ibaa764c7eaffbdb8dd6825fd518c54b37316ecb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238807
Auto-Submit: Niels Moller <nisse@webrtc.org >
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org >
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35436}
2021-11-29 21:10:31 +00:00
bf7532116b
Create socket server explicitly in PeerConnectionEndToEndBaseTest
...
Bug: webrtc:13145
Change-Id: I674382b3a784d371681426848461502b711dfe6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238430
Auto-Submit: Niels Moller <nisse@webrtc.org >
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org >
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35435}
2021-11-29 20:13:01 +00:00
7d6e34ae0b
Roll chromium_revision fe555f2e69..0e21474081 (945923:946029)
...
Change log: fe555f2e69..0e21474081fe555f2e69..0e21474081c8a98d6969..1d1c400730bc9340854d..5bea6e0b346410d12020..bebfd47d7751b895633d..51f3bc37e12ffa1bde79..02f4003fdbcff026d415..64e26ad3a27c7f2ccfb1..69300210d4fe555f2e69..0e21474081https://webrtc-review.googlesource.com/c/src/+/239400
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2021-11-29 18:46:00 +00:00
9937c32371
Roll chromium_revision 1ce7d592c3..fe555f2e69 (944520:945923)
...
This CL also adds COM_DECLSPEC_NOTHROW where needed since Windows
is compiled with -std=c++17.
Change log: 1ce7d592c3..fe555f2e691ce7d592c3..fe555f2e69a0e30222a6..c8a98d6969749ecdaeea..245517235ee504863f9e..665b74f7d1038090f742..c936d73ff7d9a982f504..bc9340854d4e888a310b..6410d120207d86ceacba..51b895633dhttps://chromium.googlesource.com/catapult.git/+log/4b9301e9c5..75423c310e
* src/third_party/depot_tools: a29f589a15..2ffa1bde793cabd142ce..cff026d4153e0e32ba30..e2f397893749836d72bd..02959c3ee1ec80f88c5d..13f984c216d5cb19a57d..889d9a924c0e9c2f5abf..7c7f2ccfb11ce7d592c3..fe555f2e69https://webrtc-review.googlesource.com/c/src/+/239355
Reviewed-by: Tommi <tommi@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35433}
2021-11-29 16:23:30 +00:00
6844e51f47
VideoSendStreamTest: Fix flaky UAF.
...
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}
2021-11-29 15:58:40 +00:00
ee22543829
Zero-hertz encoding mode: avoid encoder bitrate overshooting.
...
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}
2021-11-29 13:48:00 +00:00
b55c299f4d
Avoid TBR in DEPS autoroller (use CC instead).
...
No-Presubmit: True
No-Try: True
Bug: None
Change-Id: I71c71de4ff9cfb1b9e59620b56a13ff4e94639eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239361
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35430}
2021-11-29 12:54:01 +00:00
e2044664bd
Add unit test of AudioRtpReceiver
...
Bug: chromium:1272566
Change-Id: Ic10c586062cc8fdded2dd6249b49fab527c99823
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238987
Commit-Queue: Tony Herre <toprice@chromium.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35429}
2021-11-29 12:18:29 +00:00
2db59a6584
dcsctp: Use rtc::CopyOnWriteBuffer
...
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}
2021-11-29 11:53:19 +00:00
8695282243
Remove unnecessary copy of suspended_ssrcs.
...
Also removing pass-by-value in ctor.
Bug: none
Change-Id: I09e36fd955c8f306c4a347d8befc6eea38384cb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239183
Auto-Submit: Tommi <tommi@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Tommi <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35427}
2021-11-29 10:54:30 +00:00
9345bee860
DEPS: use python3
...
Bug: chromium:1208028
Change-Id: I1117855556e4a9e7cbaf54764c8a23c5b8581ded
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239160
Auto-Submit: Takuto Ikuta <tikuta@google.com >
Reviewed-by: Jeremy Leconte <jleconte@google.com >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35426}
2021-11-29 07:27:52 +00:00
ef5b21e637
Deprecate and remove usage for WARNING log level
...
Bug: webrtc:13362
Change-Id: Ida112158e4ac5f667e533a0ebfedb400c84df4d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239124
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35425}
2021-11-27 22:21:54 +00:00
150503566c
Delete unused classes Win32SocketServer, Win32Socket, Win32Thread.
...
Bug: webrtc:6424. webrtc:13065
Change-Id: I7f180831d3b01ad6d127b7eec51ecc9b2d8046a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239125
Reviewed-by: Tommi <tommi@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35424}
2021-11-26 16:51:56 +00:00
b2860b0033
Update peerconnection example to not use Win32Socket
...
Followup to https://webrtc-review.googlesource.com/c/src/+/223540
Bug: webrtc:6424. webrtc:13065
Change-Id: I85ccfa9e6997a24464d240b0e9b3f1ac90112e8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224541
Reviewed-by: Henrik Andreassson <henrika@webrtc.org >
Reviewed-by: Tommi <tommi@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35423}
2021-11-26 12:24:25 +00:00
deb1b1bc70
Always call IsOk() to ensure audio codec configuration is valid when negotiating.
...
We should avoid creating codecs with invalid parameters, since this can
expose security issues. For many codecs the IsOk() method to check the
codec config is only called in DCHECKs. This CL ensures IsOk() is always
called, also in non-debug builds.
Bug: chromium:1265806
Change-Id: Ibd3c6c65d3bb547cd2603e11808ac40ac693a8b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238801
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35422}
2021-11-26 10:11:21 +00:00
789a0f361f
Delete deprecated RtpExtension::FindHeaderExtensionByUri variant
...
this variant was deprecated 6 month ago in
https://webrtc-review.googlesource.com/c/src/+/219081
with a trivial replacement.
Bug: None
Change-Id: Ib9cd686280edf36da5f39e8e22b6073530837147
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238983
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35421}
2021-11-26 07:57:26 +00:00
a40e6de242
Allow extremely low resolution for simulcast path
...
Some screen capturers may occasionally send an extremely small frame,
e.g. 2x2. If a scale_resolution_down_by is specified, WebrtcVideoEngine
would enforce configured resolution to be at least 16x16, which would
then break VideoStreamEncoder and cause a crash.
This changes disables scaling and alignment for extremely low resolutions.
Bug: chromium:1265303, webrtc:13371
Change-Id: Icdb736043e1fdf91fdde5a8e4b3c6a89f6b90577
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236850
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35420}
2021-11-25 19:07:57 +00:00
e1bbef1e6b
Add options to only NACK if there is a valid RTT and if loss rate is below a configured value.
...
Bug: webrtc:10178
Change-Id: I16a74ed6fb380cecaf82a303bb14bf215c944a73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238988
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35419}
2021-11-25 16:30:17 +00:00
144e5bf87d
Use NONE if scalability mode is not specified
...
Bug: none
Change-Id: I8ffdb7fc41dec3c5b37483a6dcbb8fe7f03b59da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238984
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35418}
2021-11-25 13:50:14 +00:00
61a8d9caaa
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}
2021-11-25 10:55:04 +00:00
5f34130f26
Declare LERROR deprecated and remove all usage in webrtc
...
Bug: webrtc:13362
Change-Id: I1c6c6eccd950d73be616b34f96db7832ff94377e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238804
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35416}
2021-11-24 14:34:24 +00:00
4e8e36c745
Try to fix the version updated bot.
...
No-Presubmit: True
Bug: webrtc:13373
Change-Id: I30418f760cda6be31c75ff7e59311cb59c1874c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238805
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35415}
2021-11-24 13:04:06 +00:00
a018e677f2
Optimize block_delay_buffer.
...
Reducing pointer following. This will allow the compiler to optimize more efficiently with the "-fno-strict-aliasing" flag.
Bug: None
Change-Id: Ic126bd2d53969a7e9d15e1c1081d5278e27a816c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238664
Reviewed-by: Per Åhgren <peah@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Christian Schuldt <cschuldt@google.com >
Cr-Commit-Position: refs/heads/main@{#35414}
2021-11-24 12:20:28 +00:00
624fb67bbc
Revert "Fix out-of-bounds memory access due to large number of audio channels."
...
This reverts commit 4cbfe4192cd5b8289f7896ce14e0bd8c4ae41a97.
Reason for revert: The fix in this CL is ineffective. A better one has been created here: https://webrtc-review.googlesource.com/c/src/+/238666
Original change's description:
> Fix out-of-bounds memory access due to large number of audio channels.
>
> The number of audio channels can be configured in SDP, and can thus be
> set to arbitrary values by an attacker. This CL fixes an out-of-bounds
> memory access that could occur when the number of channels is set to a
> large number.
>
> Bug: chromium:1265806
> Change-Id: Ic88ff6d85b978b8eb99bf03cc52457a4552e8c24
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237808
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
> Cr-Commit-Position: refs/heads/main@{#35354}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1265806
Change-Id: If695ed92f831c2a9631efdf47f1568f5a15c1841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238803
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35413}
2021-11-24 11:45:55 +00:00
c3a1ea1b3f
dcsctp: assert that CreateTimeout returns a usable pointer
...
BUG=None
Change-Id: I97852d5222a4a722e61b450fbe446dd949b6840e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238802
Reviewed-by: Victor Boivie <boivie@webrtc.org >
Commit-Queue: Victor Boivie <boivie@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35412}
2021-11-24 11:44:15 +00:00
