While the target has a restricted visibility, since it was in rtc_base_approved
public deps, a lot of targets were able to bypass the visibility check.
So we remove the visibility restrictions and use the dependency explicitely
everywhere instead.
Bug: webrtc:8603
Change-Id: I94a03fdf7f94c54ab72081a58dd648e2cca73d17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36566}
This is a slight change in behavior that fixes a bug where all expansions are not counted due to more than 10ms available in the sync buffer, which can happen after repeated expansions.
The counter should also be updated when in muted mode.
Bug: webrtc:13322
Change-Id: I067689ee251d3d1ae990a27cdd271f718b0d6f2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257360
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36483}
This is used until the first RTT measurement becomes available.
100ms is a reasonable default and used in other places.
Bug: webrtc:10178
Change-Id: I14f530504a4866fbe75f025dfe184fd6e296b75e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256861
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36384}
Recent Clang versions have enhanced -Wunused-but-set-variable which now
warns about this.
Bug: chromium:1309955
Change-Id: Ie70df85f5a6d2cbabb4e10960bfd926ff7bf32cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257162
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Hans Wennborg <hans@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36381}
Add field trials to audio api.
It is added as a pointer with nullptr as default.
It is not (yet) used anywhere.
Usage of field trials comes in subsequent patches.
Bug: webrtc:10335
Change-Id: Icbe22d95c356a6fefde34590f11ea63f005ab09e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255521
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36213}
The problem was fixed by implementing the methid PacketDuration() in
AudioDecoderG722StereoImpl, which catches the issue in
AudioDecoder::Decode().
Bug: chromium:1280851
Change-Id: I31f974b9999f3c1c62b0e5dc39bb3e56a9a9388d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251842
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36034}
This CL removes even more top-level const from parameters in function
declarations. This change is safe because top-level const in function
declarations (not function definitions) are ignored by the compiler
and so change is just a no-op cleanup.
Bug: webrtc:13610
Change-Id: Icf6868c27b1fdb9d9915b3a7020eb34bdcf07a09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249989
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35866}
This reverts commit a37384899b.
Reason for revert: It breaks some downstream tests, let's reland on Monday adding a fix for them as well (Mac M1 is still broken).
Original change's description:
> Update NetEq bitexactness tests to only run on Linux.
>
> Running bitexactness tests only on Linux makes it significantly easier to
> update them, while still giving many of the same benefits.
>
> Bug: webrtc:12518, b/216736217
> Change-Id: I7f3c9a27c0fc14b7ee0e83aede2e7702cfa79141
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249787
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35829}
TBR=mbonadei@webrtc.org,ivoc@webrtc.org,titovartem@webrtc.org,jakobi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I53e3d18d53949eb9dded9ce29de99e091a480705
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12518, b/216736217
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249980
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35836}
Running bitexactness tests only on Linux makes it significantly easier to
update them, while still giving many of the same benefits.
Bug: webrtc:12518, b/216736217
Change-Id: I7f3c9a27c0fc14b7ee0e83aede2e7702cfa79141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249787
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35829}
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.
Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
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}
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}
This reverts commit 4cbfe4192c.
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}
The number of audio channels can be configured in SDP, and can thus be
set to arbitrary values. However, the audio code has limitations that
prevent a high number of channels from working well in practice.
Bug: chromium:1265806
Change-Id: I6f6c3f68a3791bb189a614eece6bd0ed7874f252
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237807
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35359}
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}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
This is a reland of 320c57b7c6
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}
This reverts commit 320c57b7c6.
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
This reverts commit 793bac569f.
Reason for revert: rare compilation error fixed
Original change's description:
> Revert "Refactor the PlatformThread API."
>
> This reverts commit c89fdd716c.
>
> Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
> See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
>
> Original change's description:
> > Refactor the PlatformThread API.
> >
> > PlatformThread's API is using old style function pointers, causes
> > casting, is unintuitive and forces artificial call sequences, and
> > is additionally possible to misuse in release mode.
> >
> > Fix this by an API face lift:
> > 1. The class is turned into a handle, which can be empty.
> > 2. The only way of getting a non-empty PlatformThread is by calling
> > SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> > code reader.
> > 3. Handles can be Finalized, which works differently for joinable and
> > detached threads:
> > a) Handles for detached threads are simply closed where applicable.
> > b) Joinable threads are joined before handles are closed.
> > 4. The destructor finalizes handles. No explicit call is needed.
> >
> > Fixed: webrtc:12727
> > Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> > Commit-Queue: Markus Handell <handellm@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33923}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=handellm@webrtc.org
>
> Bug: webrtc:12727
> Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33936}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:12727
Change-Id: Ifd6f44eac72fed84474277a1be03eb84d2f4376e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217881
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33950}
This reverts commit c89fdd716c.
Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
Original change's description:
> Refactor the PlatformThread API.
>
> PlatformThread's API is using old style function pointers, causes
> casting, is unintuitive and forces artificial call sequences, and
> is additionally possible to misuse in release mode.
>
> Fix this by an API face lift:
> 1. The class is turned into a handle, which can be empty.
> 2. The only way of getting a non-empty PlatformThread is by calling
> SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> code reader.
> 3. Handles can be Finalized, which works differently for joinable and
> detached threads:
> a) Handles for detached threads are simply closed where applicable.
> b) Joinable threads are joined before handles are closed.
> 4. The destructor finalizes handles. No explicit call is needed.
>
> Fixed: webrtc:12727
> Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33923}
# Not skipping CQ checks because original CL landed > 1 day ago.
TBR=handellm@webrtc.org
Bug: webrtc:12727
Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33936}
PlatformThread's API is using old style function pointers, causes
casting, is unintuitive and forces artificial call sequences, and
is additionally possible to misuse in release mode.
Fix this by an API face lift:
1. The class is turned into a handle, which can be empty.
2. The only way of getting a non-empty PlatformThread is by calling
SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
code reader.
3. Handles can be Finalized, which works differently for joinable and
detached threads:
a) Handles for detached threads are simply closed where applicable.
b) Joinable threads are joined before handles are closed.
4. The destructor finalizes handles. No explicit call is needed.
Fixed: webrtc:12727
Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33923}
Before this CL, timestamps of received packets were rounded
to the nearest millisecond and stored as int64_t. Due to the
rounding it sometimes happened that timestamps later in the
pipeline that are not rounded seem to occur even before the
video frame was received.
Change-Id: I92d8f3540b23baae2d4a1dc6a7cb3f58bcdaad18
Bug: webrtc:12722
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216398
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33916}
potentially allowing distances of more than 2.
BUG=webrtc:11640
Change-Id: I0d8c831218285d57cf07f0a8e5829810afd4ab3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188383
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33913}
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.
Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
The change introduces support for detachable PlatformThreads, for which
the Stop() call doesn't wait until the thread has finished executing.
The change also introduces rtc::ThreadAttributes that carries priority
and detachability thread attributes. It additionally refactors all
known use to use the new semantics.
Bug: b:181572711, webrtc:12659
Change-Id: Id96e87c2a0dafabc8047767d241fd5da4505d14c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214704
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33796}
Many things are omitted in this doc and it can definitely be improved,
but I hope it captures the most important parts.
Bug: webrtc:12568
Change-Id: I13097d633ca19cecc9dd43bdb777b0ca48f151dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215142
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33724}
This is required to be able to update the Opus version and will be
rolled back after.
Bug: webrtc:12518
Change-Id: Icc649039787db44bd55a0dc8e5ba4089df3a9566
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209363
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33375}
This remove webrtc-specific macro that has no reason to be webrtc specific
ABSL_DEPRECATED takes a message parameter encouraging to write text how class or function is deprecated.
Bug: webrtc:12484
Change-Id: I89f1398f91dacadc37f7db469dcd985e3724e444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208282
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33314}
Audio interruption metric is not implemented for codecs doing their own PLC.
R=ivoc@webrtc.org, jakobi@webrtc.org
Bug: b/177523033 webrtc:12456
Change-Id: I0aca6fa5c0ff617e76ee1e4ed8d95703c7097223
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206561
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Pablo Barrera González <barrerap@google.com>
Cr-Commit-Position: refs/heads/master@{#33229}
Isac is not able to effectively compress all types of signals. This
should be extremely rare with real audio signals, but mostly happens
with artificially created test signals. When this happens, we should
avoid crashing and just carry on.
Bug: chromium:1170167
Change-Id: I97b551fbbdcccb0186f3e6497991ac52d2301f68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205626
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33193}
We need to be able build chromium with rtc_include_tests = true. It
reveals a lot of targets that are not compatible with chromium but
aren't marked so.
`rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated.
Bug: webrtc:12404
Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33124}
To be able to build these targets in chromium we need to replace all abseil dependencies with "//third_party/abseil-cpp:absl".
Bug: webrtc:12404
Change-Id: Ie0f6af73f2abc73e5744520cfd9a6414e2f948e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202762
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33108}
Finding the array element with the largest argmax is a fairly common
operation, so it makes sense to have a Neon optimized version. The
implementation is done by first finding both the min and max value, and
then returning whichever has the largest argmax.
Bug: chromium:12355
Change-Id: I088bd4f7d469b2424a7265de10fffb42764567a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201622
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33052}
To determine the appropriate amount of shifting to prevent overflow in a
cross correlation, it is necessary to have the max value of both
sequences. However, only one was calculated in the ilbc code. This CL
calculates the max of the other sequence and correctly takes both into
account.
Bug: chromium:1161837
Change-Id: I3ba8eee0814bb5eda3769c0ce6caf2681c7525e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202253
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33043}
The abs max of a 16 bit integer cannot be represented as a 16 bit integer, because abs(-2^16) is too large. To work around this, we can instead use the index of the max element, convert it to a 32-bit int and then take the absolute value.
Bug: chromium:1158070, chromium:1146835, chromium:1161837
Change-Id: If56177c55ec62b4bd578986a5deae38a91bbc821
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198123
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32950}
Prior to this commit, most .c files in modules/audio_coding/codecs/ilbc
don't include their corresponding headers, nor do they order #includes
as per the Google Style Guide [1]. The former is especially harmful,
since in C it can silently allow the function signature to diverge from
its prototype, thus causing disaster at runtime.
This CL fixes both issues. In effect, this allows the common_audio and
modules/audio_coding:ilbc targets to be compiled with Clang's
-Wmissing-prototypes, though this CL does not add that change.
[1]: https://google.github.io/styleguide/cppguide.html#Names_and_Order_of_Includes
Bug: webrtc:12314
Change-Id: I8299968ed3cc86ff35d9de045072b846298043af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198362
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Timothy Gu <timothygu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32896}
The calculation of the necessary number of shifts is not correct, leading to an overflow.
Bug: chromium:1158070
Change-Id: I6545e9da46debf33ce169c33d762915fe755d606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197981
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32855}
I was not able to reproduce chromium:1146676 locally, so the change in merge.cc is a speculative fix.
Bug: chromium:1146835, chromium:1146676, chromium:1137226
Change-Id: I14472ba5b41e58b2d5f27d9833249c14505af18f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194264
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32759}
Instead of flushing all packets, it makes sense to flush down to the target level instead. This CL also initiates a flush when the packet buffer is a multiple of the target level, instead of waiting until it is completely full.
Bug: webrtc:12201
Change-Id: I8775147624536824eb88752f6e8ffe57ec6199cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193941
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32701}
Add a new field trial with more flexible parsing and new options:
- Resample packet delays to only update histogram with maximum observed
delay every X ms.
- Setting the maximum history size (in ms) used for calculating the
relative arrival delay.
Legacy field trial used for configuration is maintained.
Bug: webrtc:10333
Change-Id: I35b004f5d8209c85b33cb49def3816db51650946
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192789
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32591}
And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.
Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}
The variables that are used to track the amount of preemptive expansion
and acceleration are not initialized before being passed to their
respective functions. However, these function can fail in certain cases,
and when they do the uninitialized memory will pollute the NetEq statistics.
Bug: chromium:1140376
Change-Id: I004fbaaf8d24de01dd1997fb73bdf93ca88ceaaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191480
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32544}
After recently changing .pylintrc (see [1]) we discovered that
the presubmit check always checks all the python files when just
one python file gets updated.
This CL moves all these files one step closer to what the linter
wants.
Autogenerated with:
# Added all the files under pylint control to ~/Desktop/to-reformat
cat ~/Desktop/to-reformat | xargs sed -i '1i\\'
git cl format --python --full
This is part 1 out of 2. The second part will fix function names and
will not be automated.
[1] - https://webrtc-review.googlesource.com/c/src/+/186664
No-Presubmit: True
Bug: webrtc:12114
Change-Id: Idfec4d759f209a2090440d0af2413a1ddc01b841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32530}
Large gaps can cause issues in NetEq simulations, so the simulation is
ended whenever we encounter one. However, the time span of the gap is
still included in the simulation time, leading to incorrect results.
Bug: webrtc:10337
Change-Id: I94a1a0b46259e3718b1b73522a3886a17bedbb7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190287
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32514}
This code used to have a reader-writer lock, and call
std::queue::pop() with only a reader lock, which appears unsafe. Code
changed to use a plain webrtc::Mutex.
Bug: webrtc:12102
Change-Id: Icbea17a824c91975dfebd4d05bbd0c21e1abeadc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190700
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32511}
- Add buffer level filter and delay manager mocks and make them
injectable for easier testing.
- Add a basic set of tests for simple cases and recently added features.
Bug: webrtc:10333
Change-Id: I8b6f73b8ad99ad6859ed1279086c0bd68b7687be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188623
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32433}
This CL also puts the arguments in a struct to allow for easier future additions.
Bug: webrtc:11005
Change-Id: I47bf664e7106b724eb1fc42299c42bbf022393ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188385
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32409}
This is a reland of 2a7c57c34f
Original change's description:
> Reland "Refactor NetEq delay manager logic."
>
> This is a reland of f8e62fcb14
>
> Original change's description:
> > Refactor NetEq delay manager logic.
> >
> > - Removes dependence on sequence number for calculating target delay.
> > - Changes target delay unit to milliseconds instead of number of
> > packets.
> > - Moves acceleration/preemptive expand thresholds to decision logic.
> > Tests for this will be added in a follow up cl.
> >
> > Bug: webrtc:10333
> > Change-Id: If690aae4abf41ef1d9353f0ff01fb7d121cf8a26
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186265
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32326}
>
> Bug: webrtc:10333
> Change-Id: Iad5e7063f63b84762959ee5b412f5f14a7b2cd06
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186943
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32332}
Bug: webrtc:10333
Change-Id: If2244ee9a3d56a0cfa9b602e7bdf448dc6340147
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187356
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32367}
This is a reland of f8e62fcb14
Original change's description:
> Refactor NetEq delay manager logic.
>
> - Removes dependence on sequence number for calculating target delay.
> - Changes target delay unit to milliseconds instead of number of
> packets.
> - Moves acceleration/preemptive expand thresholds to decision logic.
> Tests for this will be added in a follow up cl.
>
> Bug: webrtc:10333
> Change-Id: If690aae4abf41ef1d9353f0ff01fb7d121cf8a26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186265
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32326}
Bug: webrtc:10333
Change-Id: Iad5e7063f63b84762959ee5b412f5f14a7b2cd06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186943
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32332}
During muted state NetEq shortcircuits a large part of the internals to
quickly return a buffer filled with zeros. It can be beneficial for the
controller to be aware that it is in muted state.
Bug: webrtc:11005
Change-Id: I5fe24b4a3704d953cbd68b5a24bbb7ef58b30be0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186760
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32330}
This reverts commit f8e62fcb14.
Reason for revert: breaks downstream test.
Original change's description:
> Refactor NetEq delay manager logic.
>
> - Removes dependence on sequence number for calculating target delay.
> - Changes target delay unit to milliseconds instead of number of
> packets.
> - Moves acceleration/preemptive expand thresholds to decision logic.
> Tests for this will be added in a follow up cl.
>
> Bug: webrtc:10333
> Change-Id: If690aae4abf41ef1d9353f0ff01fb7d121cf8a26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186265
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32326}
TBR=ivoc@webrtc.org,jakobi@webrtc.org
Change-Id: I1bdeacce61b902a0003a40c740f6acccf1443e3e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10333
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186942
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32329}
- Removes dependence on sequence number for calculating target delay.
- Changes target delay unit to milliseconds instead of number of
packets.
- Moves acceleration/preemptive expand thresholds to decision logic.
Tests for this will be added in a follow up cl.
Bug: webrtc:10333
Change-Id: If690aae4abf41ef1d9353f0ff01fb7d121cf8a26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186265
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32326}
This is a reland of ad148272b8
Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}
Bug: webrtc:11663, chromium:1134234
Change-Id: I0cb34cf08d4d14bc3aee055254493c9c9ee8faa0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186401
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32303}
This is a reland of ad148272b8
Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}
Bug: webrtc:11663
Change-Id: Ib41dc1d1c5865f2828699c462939d15d5562df47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186262
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32270}
This reverts commit ad148272b8.
Reason for revert: Speculative revert to investigate test failures
Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}
TBR=mbonadei@webrtc.org,saza@webrtc.org,peah@webrtc.org,kwiberg@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11663
Change-Id: Ibb019e8c702dce45ebf47f1c1e8db19069b4964d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186081
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32259}
This change lets the fuzzer modify the first few bytes of the RTP
payload. One of the benefits is that it can cover the RED header
splitter functionality.
The CL also fixes an issue found while running the fuzzer locally.
Bug: webrtc:11640
Change-Id: I7ca73676440897a14a0aaca796f70d381e016575
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185819
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32242}
The former was unused, the latter is replaced with the explicit C++11
deletions. The related RTC_DISALLOW_COPY_AND_ASSIGN is left for now,
it is used in a lot more places.
Bug: None
Change-Id: I49503e7f2b9ff43c6285f8695833479bbc18c380
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185500
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32224}
This is a reland of ad148272b8
Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}
Bug: webrtc:11663
Change-Id: I669435c2f4e451ee0766d809443484f2dde09d8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185482
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32200}
This reverts commit ad148272b8.
Reason for revert: Causing test failures downstream.
Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}
TBR=mbonadei@webrtc.org,saza@webrtc.org,peah@webrtc.org,kwiberg@webrtc.org
Change-Id: If2287a0a4b37931ce5f85baae093a66b19d0a78b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11663
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185481
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32196}
This CL activates the newly added AVX2 support by default.
The activation is done beneath a kill-switch.
Beyond the above, the CL also changes an incorrect DCHECK_GT
to a DCHECK_GE.
Bug: webrtc:11663
Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32193}
Add a get_and_clear_legacy_stats flag to AudioReceiveStream::GetStats,
to distinguish calls from standard GetStats and legacy GetStats.
Add const method NetEq::CurrentNetworkStatistics to get current
values of stateless NetEq stats. Standard GetStats will then call this
method instead of NetEq::NetworkStatistics.
Bug: webrtc:11622
Change-Id: I3833a246a9e39b18c99657a738da22c6e2bd5f5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183600
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32092}
It is meant for Pinpoint to run only the relevant tests when running a bisection.
The Pinpoint side of this change can be found here:
https://crrev.com/c/2404161
Bug: webrtc:11084
Change-Id: I466f39816b83e2f83a3a49845c99605f4d5a857b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183763
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32082}
Currently is_linux is set to true on Chrome OS build,
but it is planned to be set false. This CL is the preparation
to keep the compatibility.
Bug: chromium:1110266
Test: Build locally.
Change-Id: Ic79a202b0b3baeff157955cd03a07556bfb958a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Hidehiko Abe <hidehiko@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32073}
This is a reland of 2b242d8fba
Original change's description:
> Merge cpu_features build targets into //system_wrappers.
>
> Before this CL, functions declared in cpu_features_wrapper.h where
> not defined in the same build target, causing brittle builds that
> might fail at link time if the binary was not depending on
> //system_wrappers (the target with the definitions), violating [1].
>
> This CL moves everything into //system_wrappers and also moves
> cpu_features_wrapper.h definitions from C to C++ (in order to be able
> to add the definitions to a C++ build target like //system_wrappers).
>
> [1] - https://webrtc.googlesource.com/src/+/refs/heads/master/style-guide.md#h-cc-pairs
>
> Bug: None
> Change-Id: I5a0009cddb17206b19f2a71eeba722faacc4bcae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183380
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32039}
TBR=kwiberg@webrtc.org
Bug: None
Change-Id: I1695b9a34d3ec20c50c1202a745f64fac58edef0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183444
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32042}
This reverts commit 2b242d8fba.
Reason for revert: Breaks downstream project.
Original change's description:
> Merge cpu_features build targets into //system_wrappers.
>
> Before this CL, functions declared in cpu_features_wrapper.h where
> not defined in the same build target, causing brittle builds that
> might fail at link time if the binary was not depending on
> //system_wrappers (the target with the definitions), violating [1].
>
> This CL moves everything into //system_wrappers and also moves
> cpu_features_wrapper.h definitions from C to C++ (in order to be able
> to add the definitions to a C++ build target like //system_wrappers).
>
> [1] - https://webrtc.googlesource.com/src/+/refs/heads/master/style-guide.md#h-cc-pairs
>
> Bug: None
> Change-Id: I5a0009cddb17206b19f2a71eeba722faacc4bcae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183380
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32039}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: I4daa7582e55a0343eef72f08ed023c73e0b6456b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183443
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32040}
Before this CL, functions declared in cpu_features_wrapper.h where
not defined in the same build target, causing brittle builds that
might fail at link time if the binary was not depending on
//system_wrappers (the target with the definitions), violating [1].
This CL moves everything into //system_wrappers and also moves
cpu_features_wrapper.h definitions from C to C++ (in order to be able
to add the definitions to a C++ build target like //system_wrappers).
[1] - https://webrtc.googlesource.com/src/+/refs/heads/master/style-guide.md#h-cc-pairs
Bug: None
Change-Id: I5a0009cddb17206b19f2a71eeba722faacc4bcae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183380
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32039}
This CL fixes 2 issues that affect NetEq simulations.
- When using event logs with multiple SSRCs, it does not make sense to
use more than a single SSRC. If the user does not provide an SSRC
filter, we should use the first SSRC we find and no others.
- It is possible for event logs to have a gap in the middle, and
sometimes we don't store/mark the gap properly. If is possible to
detect gaps by looking at the wallclock time delta between getAudio
events. These should be 10 ms nominally, so values greater than 1000
should never happen and indicate an error.
Bug: webrtc:11855
Change-Id: Idc3b8a7902be4159da48b063ef5c5c82fd484071
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181940
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31960}
and do not generate redundancy for packets that are larger
than 1024 bytes which is the maximum size red can encode.
Bug: webrtc:11640
Change-Id: I211cb196eee2a0659f22a601a6dee4b7dd4e5116
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178781
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31846}
This avoids a difference in behaviour between mobile and
desktop platforms since the bitrate is now too low for
CELT mode.
BUG=webrtc:11643
Change-Id: I9ac1439bea0ccbbfee7388516932e30d6cb06bf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179522
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#31757}
In production code, the maximum number of packets is by default set to
200, so we should adopt the same behavior in tests.
Bug: None
Change-Id: I415790b7cd9fb170ea7ac94685cc6bbe14efac4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178744
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31646}
Extends the RED implementation to support a distance of two, i.e. two
packets redundancy.
BUG=webrtc:11640
Change-Id: I5113a97a4e3d45d836d7952a0c19c5381069c158
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178565
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31625}
A slight simplification of the NetEq code is also included.
The subtrees below common_audio, modules/audio_coding and
modules/audio_processing were scanned while making this CL.
Bug: webrtc:11680
Change-Id: I33bb1c75b2e3d1c6793fd1c5741ca59f4b6e8455
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178361
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31578}
modifies the RED encoder to send the actual RFC 2198 format
described in
https://tools.ietf.org/html/rfc2198
Decoding is handled in neteq, see red_payload_splitter.h
BUG=webrtc:11640
Change-Id: Ib3005882a3ceee49d2b05c43357f552432a984ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176371
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31560}
This will result in slightly higher encode bitrates and longer frame
lengths compared to using the smoothing filter.
Bug: webrtc:10981
Change-Id: I64704196c56b0ad910895c908baad38c994a971b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177425
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31556}
Setting gtest_enable_absl_printers to false in .gn uncovers some missing
dependencies that were pulled in by gtest.
Bug: None
Change-Id: Ibd7772f6e2af9c798c97161c24f70b1658e3723c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177843
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31551}
Not just at construction time.
Bug: webrtc:11704
Change-Id: I952c7dbe20774cc976065c7d2f992a80074ebf63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177663
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31550}
This has been proven to not be useful.
Bug: chromium:1086942
Change-Id: Ib71b194f59301851791a1a056f5f10b98c5a1d57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177520
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31548}
This will be used when adaptivePtime is enabled.
Bug: chromium:1086942
Change-Id: I63c947c53a8c5b8e0825b78b847c3f7900197d6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177421
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31544}
Adding field trial WebRTC-Audio-NetEqExtraDelay with a parameter value
to set the extra delay in NetEq. This overrides the
extra_output_delay_ms parameter in NetEq::Config.
Bug: b/156734419
Change-Id: Iae7d439fafa3059494249959ac13a02de63d6b7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176858
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31493}
gone for a while
BUG=webrtc:5922
Change-Id: Ie5d2f6dbffbc349686dbaf05a378375dbff0dce0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175914
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31352}
This change adds an optional delay to NetEq's output. Note, this is not
equivalent to increasing the jitter buffer with the same extra length.
Bug: b/156734419
Change-Id: I8b70b6b3bffcfd3da296ccf29853864baa03d6bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175110
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31343}
This CL introduced 2 new macros that affect the WebRTC OBJC API symbols:
- RTC_OBJC_TYPE_PREFIX:
Macro used to prepend a prefix to the API types that are exported with
RTC_OBJC_EXPORT.
Clients can patch the definition of this macro locally and build
WebRTC.framework with their own prefix in case symbol clashing is a
problem.
This macro must only be defined by changing the value in
sdk/objc/base/RTCMacros.h and not on via compiler flag to ensure
it has a unique value.
- RCT_OBJC_TYPE:
Macro used internally to reference API types. Declaring an API type
without using this macro will not include the declared type in the
set of types that will be affected by the configurable
RTC_OBJC_TYPE_PREFIX.
Manual changes:
https://webrtc-review.googlesource.com/c/src/+/173781/5..10
The auto-generated changes in PS#5 have been done with:
https://webrtc-review.googlesource.com/c/src/+/174061.
Bug: None
Change-Id: I0d54ca94db764fb3b6cb4365873f79e14cd879b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31153}
This reverts commit 7b201012bc.
Reason for revert: Seems to work, but need to get low bw tests working first
Original change's description:
> Flip histograms to true by default, fix unit in isac_fix_test.
>
> Requires downstream changes for all WebRTC perf tests, and
> a corresponding recipe change so isac_fix_test starts using the new
> flow.
>
> Bug: chromium:1029452
> Change-Id: I8918fca9bef003d365037c1c6bf7c55747dfed99
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170633
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30906}
TBR=phoglund@webrtc.org,mbonadei@webrtc.org
Change-Id: I96c2309cd71be14c5a27b515736a32f1b256453c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029452
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171865
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30913}
Requires downstream changes for all WebRTC perf tests, and
a corresponding recipe change so isac_fix_test starts using the new
flow.
Bug: chromium:1029452
Change-Id: I8918fca9bef003d365037c1c6bf7c55747dfed99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170633
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30906}
The WebRTC-SendSideBwe-WithOverhead field trial requires audio
encoders to properly implement the
AudioEncoder::GetFrameLengthRange() function. Thic CL implements
the function for all audio encoders in WebRTC in preparation for
making that function pure virtual in the interface.
Bug: webrtc:11427
Change-Id: Ieab6b6c72c62af6ac9525a20fcb39bd477079551
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171503
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30890}
This CL corrects the temporary buffers size in the
pre-processing of the capture audio before encoding.
As part of this it removes the ACM-specific hardcoding
of the size and instead ensures that the size of the
temporary buffer matches that of the AudioFrame.
(cherry picked from commit d82a02c837)
No-Try: True
TBR: kwiberg@webrtc.org
Bug: webrtc:11242, chromium:1060647
Change-Id: I56dd6cadfd4e140e8e159966c33d1027383ea9fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170340
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#30775}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170780
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/branch-heads/4044@{#10}
Cr-Branched-From: be99ee8f17f93e06c81e3deb4897dfa8253d3211-refs/heads/master@{#30432}
This CL corrects the temporary buffers size in the
pre-processing of the capture audio before encoding.
As part of this it removes the ACM-specific hardcoding
of the size and instead ensures that the size of the
temporary buffer matches that of the AudioFrame.
Bug: webrtc:11242
Change-Id: I56dd6cadfd4e140e8e159966c33d1027383ea9fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170340
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30775}
We need to write protos as "wb" and not "w", otherwise we get CRLF
on Windows which corrupts the proto.
Bug: chromium:1029452
Change-Id: Iabf841405134d7bc2523ac48219ca7cb9d8214c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170320
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30772}
This callback is enabled via the method
AudioCodingModule::RegisterVADCallback, which is unused, and deleted
in this cl.
Bug: None
Change-Id: I04c8690fbb673305e69fe5b1c32d88efd6c72d1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148420
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30735}
This updates various bitexactness tests and other tests that no longer
pass.
Bug: webrtc:11325
Change-Id: Ifa3e4b42e303f5573e028dfdf8a108a76f6318ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168952
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30688}
Use speech content instead of white noise and enable target vs measured
bitrate tests.
Bug: webrtc:11360
Change-Id: If8c8e73f943eda14efeb22ba406c7a1bed7d32b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168660
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30630}
OPUS_GET_IN_DTX was added 2019-04-15, but we still need to support
building on systems with older versions of the Opus headers (eg. Debian
Jessie, released 2015-04-25). This is needed to fix the "Build From
Tarball" bot [1].
[1] https://ci.chromium.org/p/infra/builders/cron/Build%20From%20Tarball
BUG=chromium:1047860,webrtc:11085
R=minyue@webrtc.org,henrick.lundin@webrtc.org
Change-Id: I5418c3caf4d2c7da9b9ba43ce85879b1e0eec6e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168560
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Thomas Anderson <thomasanderson@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30612}
Move definition of AlignedArray to the only code using it, the
test-only LappedTransform class, and delete unused methods.
Bug: webrtc:6424, webrtc:9577
Change-Id: I1bb5f57400f7217345b7ec7376235ad4c4bae858
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168701
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30576}
It is now possible to set the target bitrate for iSAC for the fixed
point implementation. Unit tests added.
Bug: webrtc:11360
Change-Id: I60225d4ca1363cdacf18931e7cf412c5aec8d8fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168529
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30526}
Unit test to test the iSAC webrtc API wrapper, plus a minor
change in the c iSAC wrapper.
Bug: webrtc:10584
Change-Id: Iecbf6f3e7db5b3bdba41f8428254ae6a6a73e24a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168492
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30514}
AudioDecoder::Decode() is obsolete. This CL replaces it with
ParsePayload() in the audio decoder NetEQ unit tests.
Bug: webrtc:10098
Change-Id: I602b0330adbe1d0921b0c4524aa7305b500f2ebf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168486
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30511}
I've not worked in these parts for years!
Bug: webrtc:10381
Change-Id: Ie78947b3d5ed9106bc05749ab21b4dbca1da88d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168346
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30488}
The upgrade to opus 1.3 is easier to carry out while the opus
bitexactness tests are temporarily disabled.
Bug: webrtc:11325
Change-Id: I96eecdbc93a01da88b92ae7f6473034c9795f3a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167726
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30425}
This is a reland of 086055d0fd
ANA was accitendly disabled even when transport sequence numbers were
negotiated due to a bug in how the audio send stream is configured. To
solve this we simply continue to always allow enabling ANA and leave it
up to the application to ensure that it's not used together with receive
side estimation.
Original change's description:
> Reland "Only include overhead if using send side bandwidth estimation."
>
> This is a reland of 8c79c6e1af
>
> Original change's description:
> > Only include overhead if using send side bandwidth estimation.
> >
> > Bug: webrtc:11298
> > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30382}
>
> Bug: webrtc:11298
> Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30390}
Bug: webrtc:11298
Change-Id: If2ad91e17ebfc85dc51edcd9607996e18c5d1f13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167883
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30413}
This is a reland of 8c79c6e1af
Original change's description:
> Only include overhead if using send side bandwidth estimation.
>
> Bug: webrtc:11298
> Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30382}
Bug: webrtc:11298
Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30390}
This reverts commit 8c79c6e1af.
Reason for revert: Introduced a Bug that can happen if the include overhead state changes between pushing and poping a packet from the pacer packet queue.
Original change's description:
> Only include overhead if using send side bandwidth estimation.
>
> Bug: webrtc:11298
> Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30382}
TBR=saza@webrtc.org,ossu@webrtc.org,sprang@webrtc.org,srte@webrtc.org,alito@webrtc.org
Change-Id: I0cacbc26408b7bec5bc3855a628e62407c081117
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11298
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167523
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30383}
When creating a NetEqController it can be useful to have access to a
webrtc::Clock*. Also, NetEqControllers should have access to the
contents of the sync buffer when making decisions.
Bug: webrtc:11005
Change-Id: I7fdba75ce661b2ace52458620a8c1f3c990e5ac2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167208
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30368}
