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}
