46814941f2
Revert "Remove legacy delay manger field trial and update default config."
...
This reverts commit 93849d4b2a976b0a46059d6f74d9efd8f12eab92.
Reason for revert: AcmReceiverBitExactnessOldApi tests failing on MacARM64; first failing build https://ci.chromium.org/ui/p/webrtc/builders/ci/MacARM64%20M1%20Release/1038/overview
Example faliure
[ RUN ] AcmReceiverBitExactnessOldApi.8kHzOutput
...
(rtp_file_reader.cc:165): Failed to read
../../modules/audio_coding/acm2/audio_coding_module_unittest.cc:912: Failure
Expected equality of these values:
checksum_ref
Which is: "636efe6d0a148f22c5383f356da3deac"
checksum_string
Which is: "6a288942d67e82076b38b17777cdaee4"
Original change's description:
> Remove legacy delay manger field trial and update default config.
>
> Bug: webrtc:10333
> Change-Id: I20e55d8d111d93657d1afe556fe3a325337c074c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232820
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
> Cr-Commit-Position: refs/heads/main@{#35321}
TBR=ivoc@webrtc.org ,jakobi@webrtc.org ,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I0bd3832aacba8dcd8e836650786cea20b4c083be
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10333
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237441
Reviewed-by: Olga Sharonova <olka@webrtc.org >
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Commit-Queue: Olga Sharonova <olka@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35324}
2021-11-09 09:10:39 +00:00
93849d4b2a
Remove legacy delay manger field trial and update default config.
...
Bug: webrtc:10333
Change-Id: I20e55d8d111d93657d1afe556fe3a325337c074c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232820
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35321}
2021-11-08 11:14:09 +00:00
e5e78c4521
Fix -Wunused-but-set-variable.
...
Bug: None
Change-Id: I8943227108e46c4c942895e4bd8fb276947502e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236525
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org >
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35272}
2021-10-28 12:53:49 +00:00
d67e75a45d
Update m1 arm64 audio bitexactness tests hashes. Followup.
...
Update hashes that weren't updated at
https://webrtc-review.googlesource.com/c/src/+/232220
Bug: b/199885455
Change-Id: I4f9327fa48a9d5656f57c5dcbc6438f20f919acd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232320
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35023}
2021-09-17 10:19:01 +00:00
d14f98f635
Update m1 arm64 audio bitexactness tests hashes
...
Follow up for https://webrtc-review.googlesource.com/c/src/+/232061/5 . Updated mac M1 tests that was missed because they are not part of CQ
Bug: b/199885455
Change-Id: I77618ac2869ba601f322857f4391b63220d20252
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232220
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35021}
2021-09-17 08:59:05 +00:00
e742d8c163
Updae bitexactness tests to match new canonical results
...
This CL has to be submitted together with chromium roll
Roll chromium_revision 48501b3f18..a098e1265b (917742:921682)
Change log: 48501b3f18..a098e1265b48501b3f18..a098e1265bbdbd6f899c..ecf6ff89bff90eed6a5e..fabb3638a77de86cbf37..a5b6419452edf77b2d2d..44ea7aba6a1b17fd57e6..e8d5d843cf59835db543..7adcb6a6bfa299c990bc..21d7875bec2@0.10.cr0 ..version:2@1.1.2.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_annotation: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_annotations: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_check_api: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_core: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_type_annotations: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_guava_guava: version:2@30.1-jre.cr0..version:2@30.1.1-jre.cr0
* src/third_party/androidx: TnotTDnWGUJDh0mSOMrgnIwzbfWFiwo2NTtr2SlUJ0MC..QRXHawrfyCC8nr_u3VCPoilYsqHO01yv1EiduoQUPcgC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/dddb60eb97..27a3328a37
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d9a9ebbe89..6b072fac88
* src/third_party/depot_tools: 789dfc223b..9c67b230207482c98f15..5b626281f1955c7f837e..159c9ad23e280366ba6a..6602cbb706ece15d049f..3f443830bd15a75b4530..65a1751e5b49ebc996aa..0896c348738420673b4c..de98afcd4abdf3dd3f28..62d7d0c928e3721e5cf9..a27c09f45b48501b3f18..a098e1265b48501b3f18..a098e1265bTBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,marpan@webrtc.org , jianj@chromium.org ,
Bug: b/199885455
Change-Id: I64329c57949e9233f5cf65167f012db0637fe4fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232061
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35008}
2021-09-15 18:38:41 +00:00
1061686107
red: handle opus dtx 400ms timestamp gap
...
by not encoding redundancy. The timestamp gap of 400ms means a
rtp timestamp difference of 19200 which would overflow the 14 bit
RED timestamp difference field.
To test in Chrome, go to
https://webrtc.github.io/samples/src/content/peerconnection/audio/
set `useDtx = true` in the console and be very quiet.
BUG=webrtc:13182,webrtc:11640
Change-Id: I1cedc7d846ac7ae821bb7cb5c0f37a17511ac727
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231940
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35005}
2021-09-15 15:09:28 +00:00
bc503c95b5
Remove support for nack threshold.
...
The nack threshold feature is unlikely to provide any value, since
reordered packets are rare. This CL also removes the factory method
from the NackTracker class, since it did not add much value.
Bug: webrtc:10178
Change-Id: Ib6bece4a2d9f95bd4298799aaa15627f5c014b61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231953
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#34993}
2021-09-14 16:51:17 +00:00
018cd3d6fc
Avoid NACKing after DTX.
...
This is done by not adding missing packets to the NACK list if the number of samples per packet is too large.
Bug: webrtc:10178
Change-Id: If46398d6d05ea35f30d7028040d3b808559e950b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231841
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#34984}
2021-09-13 16:31:42 +00:00
7a4ab1a063
red: handle 0-redundancy encoding case
...
which is a "degenerated" case that just adds a one-byte header.
BUG=webrtc:11640
Change-Id: I173f4cb56270e0090219e4450b036bbe1b51756a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231696
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#34976}
2021-09-13 07:23:11 +00:00
3a9640a5aa
Improve nacking logic
...
Requesting nacks in more cases allows the delay adaptation to better
predict if it's worth it to increase the jitter buffer delay to wait for
the nacked packets to arrive.
Bug: webrtc:10178
Change-Id: I46adb76c013dbb8df0b99eb3f7a398f8f21c2bf6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231648
Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#34970}
2021-09-10 14:21:22 +00:00
552c83171b
Update nack tracker when receiving empty packets.
...
Bug: webrtc:7611
Change-Id: Ied50829f245ea99f8d737cffdf9a3d646de3fbaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231641
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#34958}
2021-09-09 10:40:35 +00:00
58ed02eae3
Improve adaptation to reordered packets in delay manager.
...
This is done by adding a reorder optimizer that estimates the probability of receiving reordered packets.
The optimal delay is decided by balancing the cost of increasing the delay against the probability of missing a reordered packet, resulting in a loss. This balance is decided using the `ms_per_loss_percent` parameter.
The usage and parameters can be controlled via field trial.
Bug: webrtc:10178
Change-Id: Ic484df0412af35610e74b3a6070f2bac7a926a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231541
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#34954}
2021-09-08 16:41:27 +00:00
74158ff761
Refactor delay manager.
...
Split out `RelativeArrivalDelayTracker` and `DelayOptimizer` logic.
This is in preparation for adding another `DelayOptimizer` specialized in handling reordered packets.
Bug: webrtc:10178
Change-Id: Id3c1746d91980b171fa524f9b2b71cf11fc75f64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231224
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#34938}
2021-09-07 13:45:47 +00:00
7cca016721
Hard code rtx handling option in NetEq.
...
This allows NetEq to adapt to late reordered packets which are common when using retransmissions.
Remaining cleanup of the plumbing from WebRTC API will be done in a follow-up cl.
Bug: webrtc:10178
Change-Id: Ia9911eaafdabd3b69441dc089116d79e24f1b2b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231002
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#34898}
2021-09-01 18:18:59 +00:00
25edb62a94
Calculate relative arrival delay for reordered packets.
...
This changes behavior slightly but results in a better delay estimate and cleaner code.
Bug: webrtc:10178
Change-Id: If150258bc1ea58149940f17c5660733ff61159c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230740
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#34883}
2021-08-31 12:26:41 +00:00
1039392209
Add a unittest to NetEq StatisticsCalculator for discarded packets counter.
...
Bug: webrtc:8199
Change-Id: I32127af1ae6692717f28dbf2d820cd67c0b6a66a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229300
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Commit-Queue: Minyue Li <minyue@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34808}
2021-08-19 17:17:37 +00:00
57d13105e5
A test to verify counter on discarded packet upon reordering.
...
Bug: webrtc:8199
Change-Id: Ia8c545e4ae0ff98c22031b633d1bca4ba58e9e7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229380
Commit-Queue: Minyue Li <minyue@webrtc.org >
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34804}
2021-08-19 14:13:30 +00:00
9367cff59b
red: change default redundancy level to 1
...
BUG=webrtc:11640
Change-Id: Ide66ae4803349ebb908d372e811efd9ef93d51e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228424
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34720}
2021-08-11 12:59:13 +00:00
e9655736e5
Reland "red: remove special-casing of no-redundancy"
...
This is a reland of 320c57b7c6bf68d8612a4f135f44b3d29e802113
Original change's description:
> red: remove special-casing of no-redundancy
>
> removes the special-casing of not sending a RED header when there is no redundant payload.
> This avoids switching back and forth between the primary and the red payload format (primarily at the start of the connection).
>
> BUG=webrtc:11640
>
> Change-Id: I8e0044bef1ed7c4168d9527645522392db2ed068
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220932
> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org >
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#34703}
Bug: webrtc:11640
Change-Id: I5e5687be575183ee16d74df4a8170e4fedad739f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228422
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34713}
2021-08-11 08:26:59 +00:00
1118ebac0a
Revert "red: remove special-casing of no-redundancy"
...
This reverts commit 320c57b7c6bf68d8612a4f135f44b3d29e802113.
Reason for revert:
Breaks CI tests: https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux32%20Release/27236/overview
All CI tests: https://ci.chromium.org/p/webrtc/g/ci/console
Error is of the following type:
```
../../modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc:195: Failure
Expected equality of these values:
1u
Which is: 1
encoded_info_.encoded_bytes
Which is: 2
Stack trace:
0x56b298d9: webrtc::AudioEncoderCopyRedTest_CheckPayloadSizesSingle_Test::TestBody()
0x572fe317: testing::internal::HandleExceptionsInMethodIfSupported<>()
0x572fe1d4: testing::Test::Run()
0x572ff2ee: testing::TestInfo::Run()
```
Original change's description:
> red: remove special-casing of no-redundancy
>
> removes the special-casing of not sending a RED header when there is no redundant payload.
> This avoids switching back and forth between the primary and the red payload format (primarily at the start of the connection).
>
> BUG=webrtc:11640
>
> Change-Id: I8e0044bef1ed7c4168d9527645522392db2ed068
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220932
> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org >
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#34703}
TBR=henrik.lundin@webrtc.org ,devicentepena@webrtc.org ,minyue@webrtc.org ,philipp.hancke@googlemail.com ,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: Ide409232720df32b24022f99228f3b6ae81f06fe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228421
Reviewed-by: Andrey Logvin <landrey@webrtc.org >
Commit-Queue: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34707}
2021-08-10 18:09:16 +00:00
320c57b7c6
red: remove special-casing of no-redundancy
...
removes the special-casing of not sending a RED header when there is no redundant payload.
This avoids switching back and forth between the primary and the red payload format (primarily at the start of the connection).
BUG=webrtc:11640
Change-Id: I8e0044bef1ed7c4168d9527645522392db2ed068
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220932
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34703}
2021-08-10 16:56:05 +00:00
43f25e36f7
red: fix redundancy shift and add tests
...
fixes an incorrect redundancy shift and add tests that would have caught this bug.
BUG=webrtc:11640
Change-Id: I6fe2fb21587fffc5fee4d403ac898e12d525a1cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224120
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34702}
2021-08-10 16:51:06 +00:00
3abd10889f
Fix define if chain in audio decoder unittest
...
Follow up https://webrtc-review.googlesource.com/c/src/+/228247 . Turned out "#elif defined(WEBRTC_MAC) && defined(WEBRTC_ARCH_ARM64) // M1 Mac" branch was unreachable
Bug: webrtc:13053
Change-Id: Icf1aa5147347a1fad0dce8cca893bb3c598f658e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228381
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Commit-Queue: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34699}
2021-08-10 13:00:33 +00:00
8c654aa059
Update bit exactness tests to match changes
...
Follow up for https://webrtc-review.googlesource.com/c/src/+/227773 , updating M1 checksums that were not updated in the previous CL.
Example M1 failed run: https://ci.chromium.org/ui/p/webrtc/builders/ci/MacARM64%20M1%20Release/401/overview
Bug: webrtc:13053
Change-Id: I111d1d3c4bf5828ee499f20799b527ca916d77e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228247
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Commit-Queue: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34697}
2021-08-10 12:19:13 +00:00
cfea2182f8
Use backticks not vertical bars to denote variables in comments
...
Bug: webrtc:12338
Change-Id: I89c8b3a328d04203177522cbdfd9e606fd4bce4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228246
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34696}
2021-08-10 10:40:03 +00:00
5ce7d14f81
Delete legacy rtp header parser as no longer used
...
Bug: None
Change-Id: I3c532eee7f2d9e5295874dd538730625c8d423ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227086
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Minyue Li <minyue@google.com >
Reviewed-by: Minyue Li <minyue@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34676}
2021-08-09 12:14:52 +00:00
ee96675eda
Reland "Roll chromium_revision de0d050e64..42d795c24f (908789:908899)"
...
This is a reland of e369928e040623340e04390ce933e5dc491af22f
Original change's description:
> Roll chromium_revision de0d050e64..42d795c24f (908789:908899)
>
> Change log: de0d050e64..42d795c24fde0d050e64..42d795c24ff14f1b7600..6551b66fbfe360729c13..496f4dc82b2965e1969a..fda9d9017836299f559a..cb835b4820e99cff4446..aec4ec11c20a4dd4181a..cc487710bb00e6f338d0..7d0822e5b15219d6859a..667c51bbcade0d050e64..42d795c24fde0d050e64..42d795c24fhttps://webrtc-review.googlesource.com/c/src/+/227770
> Commit-Queue: Artem Titov <titovartem@webrtc.org >
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#34658}
Bug: None
Change-Id: Ibc843ef6e4e50d9d62b6b3550d5cde6eaebc02e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227773
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34662}
2021-08-06 13:51:07 +00:00
d937f50e20
Revert "Roll chromium_revision de0d050e64..42d795c24f (908789:908899)"
...
This reverts commit e369928e040623340e04390ce933e5dc491af22f.
Reason for revert: Breaks downstream project
Original change's description:
> Roll chromium_revision de0d050e64..42d795c24f (908789:908899)
>
> Change log: de0d050e64..42d795c24fde0d050e64..42d795c24ff14f1b7600..6551b66fbfe360729c13..496f4dc82b2965e1969a..fda9d9017836299f559a..cb835b4820e99cff4446..aec4ec11c20a4dd4181a..cc487710bb00e6f338d0..7d0822e5b15219d6859a..667c51bbcade0d050e64..42d795c24fde0d050e64..42d795c24fhttps://webrtc-review.googlesource.com/c/src/+/227770
> Commit-Queue: Artem Titov <titovartem@webrtc.org >
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#34658}
TBR=mbonadei@webrtc.org ,ivoc@webrtc.org ,titovartem@webrtc.org ,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: Id01cdb6a6344d7d08ee38fb152cb209a4705aa39
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227772
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34659}
2021-08-06 11:07:58 +00:00
e369928e04
Roll chromium_revision de0d050e64..42d795c24f (908789:908899)
...
Change log: de0d050e64..42d795c24fde0d050e64..42d795c24ff14f1b7600..6551b66fbfe360729c13..496f4dc82b2965e1969a..fda9d9017836299f559a..cb835b4820e99cff4446..aec4ec11c20a4dd4181a..cc487710bb00e6f338d0..7d0822e5b15219d6859a..667c51bbcade0d050e64..42d795c24fde0d050e64..42d795c24fhttps://webrtc-review.googlesource.com/c/src/+/227770
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34658}
2021-08-06 10:51:58 +00:00
deac4dea4f
red: copy audio level from main packet for recovery packet
...
fill the audio level of the recovery packets from the main packet.
While not exact, this should be close enough. Without this,
the audio level in getStats() will be filled but the audio level
in getSynchronizationSources() will be empty.
In chrome this is easy to test, the audio level graph on
https://webrtc.github.io/samples/src/content/peerconnection/audio/
will be empty all the time prior to this fix.
BUG=webrtc:11640
Change-Id: Ia1e61fd1852445239021a76d08032120a92d83aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226840
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34635}
2021-08-03 14:26:02 +00:00
d00ce747c7
Use backticks not vertical bars to denote variables in comments for /modules/audio_coding
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Bug: webrtc:12338
Change-Id: I02613d9fca45d00e2477f334b7a0416e7912e26b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227037
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34621}
2021-08-02 10:45:40 +00:00
55ec1a43bb
Fix some instances of -Wunused-but-set-variable.
...
Bug: chromium:1203071
Change-Id: I1ef3c8fd1f8e2bbf980d5d5217257e919f4564c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226961
Commit-Queue: Peter Kasting <pkasting@chromium.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34579}
2021-07-28 02:08:30 +00:00
b0ed12099f
Update links to point at main branch
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As part of go/coil update code search links to not point to the
"master" branch.
Bug: chromium:1226942
Change-Id: I0ae9e84ecc660f789a69fe0b226f93bbc39a8a66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226081
Commit-Queue: Tony Herre <toprice@chromium.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34531}
2021-07-22 16:41:26 +00:00
190244bb59
Remove all #include <assert.h>/<cassert> and usage in Obj-C code.
...
This CL completes the removal of assert() and relative headers from
the codebase (excluded
//examples/objc/AppRTCMobile/third_party/SocketRocket which is in a
third_party sub-directory).
Bug: webrtc:6779
Change-Id: I93ed57168d2c0e011626873d66529488c5f484f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225546
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34528}
2021-07-22 14:00:26 +00:00
28a2c63526
Adding packetsDiscarded to RTCReceivedRtpStreamStats.
...
Bug: webrtc:12532, webrtc:7065, webrtc:8199
Change-Id: I3ba62ec65e5660e98787f629aec3ee7a0889207a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225261
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Commit-Queue: Minyue Li <minyue@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34468}
2021-07-13 20:34:45 +00:00
25ab3228f3
Replace assert() with RTC_DCHECK().
...
CL partially auto-generated with:
git grep -l "\bassert(" | grep "\.[c|h]" | \
xargs sed -i 's/\bassert(/RTC_DCHECK(/g'
And with:
git grep -l "RTC_DCHECK(false)" | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'
With some manual changes to include "rtc_base/checks.h" where
needed.
A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.
The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.
This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).
Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34442}
2021-07-09 07:49:43 +00:00
6d92fcd364
Roll chromium_revision ba5ff58b6c..94a136c73d (898571:898790)
...
This CL also includes updates to bit-exactness tests that started
to fail on linux_x86 after the update of clang that is part of
the Chromium Roll CL.
Change log: ba5ff58b6c..94a136c73dba5ff58b6c..94a136c73decfc5939e4..da70c03d5c6f773f2fd2..b11e004f56837dc401ee..2d44844c9e537028df55..7ec8dcae8bddfda49030..326e9a8fc7f4ffdc1c0d..1f54e94bc3b3f11721ed..0587b769f6ba5ff58b6c..94a136c73dba5ff58b6c..94a136c73dTBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,
BUG=webrtc:12941
Change-Id: Ibbbb25952bc6f33f418fec37b189c0068d3a6928
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225141
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34423}
2021-07-06 17:04:38 +00:00
6832ee25c0
Delete unneeded references to string_encode.h
...
Bug: webrtc:6424
Change-Id: Ia521bcdfa8b887447ca9ed6f9d89f3ddb0e1dd15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223665
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34400}
2021-07-01 09:35:23 +00:00
2ae4ed223a
Fix the last checksum
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This should be the last checksum CL for audio tests.
Bug: webrtc:12882
Change-Id: Ie7033434e920a2f923c521cca00d1c270c406370
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224086
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Commit-Queue: Christoffer Jansson <jansson@google.com >
Cr-Commit-Position: refs/heads/master@{#34391}
2021-06-30 07:32:00 +00:00
46d002cb36
Add M1 Mac expected results for AudioDecoderIsacFixTest
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Bug: webrtc:12882
Change-Id: I56c1fcdd85fab88924b9a9f53a1a20485633f840
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223660
Commit-Queue: Christoffer Jansson <jansson@google.com >
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34389}
2021-06-30 07:03:52 +00:00
7208457e80
Same length for all ARM64 platforms
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Update more audio checksums for M1
Bug: webrtc:12882
Change-Id: I527a43a01afe2b2e4af137852174159bf3111652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224081
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Commit-Queue: Christoffer Jansson <jansson@google.com >
Cr-Commit-Position: refs/heads/master@{#34372}
2021-06-28 11:18:40 +00:00
2b3a10e62d
Add MAC arm64 platform and update checksums for acm unittest
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Bug: webrtc:12882
Change-Id: Ie820746dd66d28a2a57c2e2a3b9f12b4c43f56a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223668
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Commit-Queue: Christoffer Jansson <jansson@google.com >
Cr-Commit-Position: refs/heads/master@{#34370}
2021-06-28 08:18:07 +00:00
f2ed401679
Fix unscaled timestamps passed to nack_tracker
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If timestamp_scaler_ is used, then rtp_header.timestamp, passed to UpdateLastDecodedPacket, will advance at a different rate than the scaled timestamp packet->timestamp, passed to UpdateLastDecodedPacket.
NackTracker::EstimateTimestamp uses timestamp_last_received_rtp_, and NackTracker::TimeToPlay uses timestamp_last_decoded_rtp_.
This difference causes TimeToPlay to continuously increase to huge values, so that every missing packet will be returned from GetNackList, even if RTT > real time to play.
Change-Id: Ie6ca347972edea98a202c9cdd26c6ab3f45a73c4
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222841
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34361}
2021-06-23 08:41:50 +00:00
c6d76489e3
Add jakobi to modules/audio_coding OWNERS
...
Bug: None
Change-Id: I299f38126dc1bb419448dcf6f61d3d0323e33885
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223040
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34331}
2021-06-18 11:52:58 +00:00
b4100ad06a
Avoid using legacy rtp parser in neteq test::Packet
...
Bug: None
Change-Id: I9184954d9c99f0a34ae335d03843171864071e5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222648
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34316}
2021-06-17 08:38:14 +00:00
3cc68ec32e
Report stats from ChannelReceive::GetAudioFrameWithInfo at 1Hz.
...
This is a change from the previous 100Hz frequency.
Also changing the locks slightly in AcmReceiver so that grabbing the
neteq lock right after we've let it go, isn't necessary inside of
AcmReceiver::GetAudio and also to avoid grabbing the neteq lock while
holding the AcmReceiver lock.
Bug: webrtc:12868
Change-Id: If6ee35f3dca20eb5bdbc615123aa099ccecf57c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221371
Commit-Queue: Tommi <tommi@webrtc.org >
Reviewed-by: Minyue Li <minyue@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Reviewed-by: Markus Handell <handellm@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34258}
2021-06-09 18:41:47 +00:00
36b7d10a1f
Delete unused test method in neteq that uses RtcpStatistics
...
Bug: webrtc:10678
Change-Id: I759b635037ab7d2d113fbf8359cdbc46e7712ea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218843
Reviewed-by: Minyue Li <minyue@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34018}
2021-05-17 12:43:44 +00:00
d674ec77af
Not dropping the refresh DTX packets but substituting them by 1 byte packets.
...
Bug: webrtc:12380
Change-Id: I27029c591ac2555d6ae61b706adcf97c9498a9fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217880
Reviewed-by: Minyue Li <minyue@webrtc.org >
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33983}
2021-05-11 19:47:34 +00:00
b1606a3352
red: make amount of redundancy controllable via field trial parameter
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BUG=webrtc:11640
Change-Id: If46e3e6c84fa8781affaa400028b73571dae9108
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217384
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org >
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33975}
2021-05-11 08:36:14 +00:00
