The bug hasn't caused us any problems, since we don't run CNG together with Opus (our only real 48 kHz codec), but would cause problems if used with PCB16b @ 48 kHz.
BUG=webrtc:5303
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1496243002 .
Cr-Commit-Position: refs/heads/master@{#10929}
Reason for revert:
Broke downstream compile step, possibly relandable when using a MSVC version that has constexpr, other than that I'm out of ideas.
.../webrtc/base/atomicops.h:71:8: note: no known conversion for argument 1 from '<brace-enclosed initializer list>' to 'const rtc::AtomicInt&'
Original issue's description:
> Reland of "Create rtc::AtomicInt POD struct."
>
> Relands https://codereview.webrtc.org/1420043008/ with brace initializers
> instead of constructors hoping that they won't introduce static
> initializers.
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/84f0970d100e67a1dc4fe9a1b16b7d293302044e
> Cr-Commit-Position: refs/heads/master@{#10920}
TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1505053002
Cr-Commit-Position: refs/heads/master@{#10922}
This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.
BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1479023002 .
Cr-Commit-Position: refs/heads/master@{#10909}
This CL is the first in a series of CLs to refactor
VideoProcessing(Module) to follow Google C++ style guide and make the
code more readable.
This CL removed inheritance from Module, renames variables and makes
VideoProcessingImpl::PreprocessFrame return a frame pointer if there
is a frame to send, nullptr otherwise. The affected CLs also passes git
cl lint.
BUG=webrtc:5259
Review URL: https://codereview.webrtc.org/1482913003
Cr-Commit-Position: refs/heads/master@{#10907}
Specify kf_min_dist to get correct key frame interval in svc mode.
Also set QP-max/min per temporal and spatial layer (was previously only allowed to be set per spatial layer).
BUG=chromium:500602
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1492633005 .
Cr-Commit-Position: refs/heads/master@{#10890}
To try to resolve the problem I replaced the custom synchronization with rtc::Event which made the code cleaner, faster, and less error prone.
However, in the end the source of the test locks was that during TearDown one of the threads was stuck in a waiting loop.
I added a fix for the TearDown issue but still decided to keep the rtc:Event - based code change metioned above as that gave a more clean code.
BUG=
Review URL: https://codereview.webrtc.org/1490113004
Cr-Commit-Position: refs/heads/master@{#10880}
The two added macros simplifies the logging code when a value which is not stored in a variable should be logged.
BUG=
Review URL: https://codereview.webrtc.org/1488613002
Cr-Commit-Position: refs/heads/master@{#10870}
Also doing some simplifications inside video_coding. No CHECKs added,
since they appear to have introduced breakages in downstream tests.
Overall reducing the number of potential ways a decoder could possibly
be set null. Removing deregistration of external decoders should also
give a quicker shutdown time since that may attempt to register
internal decoders.
BUG=chromium:563299
TBR=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1483423002 .
Cr-Commit-Position: refs/heads/master@{#10858}
Reason for revert:
Speculative revert since a downstream test started failing with this.
Original issue's description:
> Add _decoder CHECK to VCMGenericDecoder constructor.
>
> This should never be using a null decoder, but it looks like it's
> crashing out in the field. Adding a CHECK to see if it catches any
> interesting stack traces.
>
> Also making the _decoder pointer const to show that it should never be
> changing.
>
> BUG=chromium:563299
> R=stefan@webrtc.org
>
> Committed: a443ec1a75TBR=stefan@webrtc.org,pbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:563299
Review URL: https://codereview.webrtc.org/1490703002
Cr-Commit-Position: refs/heads/master@{#10851}
The callback keeps a reference to an object until the callback goes out of scope.
Review URL: https://codereview.webrtc.org/1487493002
Cr-Commit-Position: refs/heads/master@{#10847}
This should never be using a null decoder, but it looks like it's
crashing out in the field. Adding a CHECK to see if it catches any
interesting stack traces.
Also making the _decoder pointer const to show that it should never be
changing.
BUG=chromium:563299
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1485713002 .
Cr-Commit-Position: refs/heads/master@{#10843}
-Renamed the TimeToFrequency and FrequencyToTime functions.
-Moved the windowing from the TimeToFrequency function.
-Simplified the EchoSubtraction function.
Note that the aec state is still an input to the EchoSubtraction function, and it currently needs to be that in order to support the output of the debug file. The longer-term goal is, however, to order the state into substates. This will simplify the parameter lists to the EchoCancellation function as well as replace the aec state as a parameter
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1456123003
Cr-Commit-Position: refs/heads/master@{#10830}
In https://codereview.webrtc.org/1481493004/ some duplicated headers
were left to make it possible to update downstream without breakage.
Now that's done and we can remove these to avoid confusion.
BUG=webrtc:5095
TBR=henrik.lundin@webrtc.org, kwiberg@webrtc.org
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
Review URL: https://codereview.webrtc.org/1477423002
Cr-Commit-Position: refs/heads/master@{#10829}
Created rtcp::Psfb abstract class between rtcp::Pli and rtcp::RtcpPacket to hold common data for Feedback Message.
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1446513002
Cr-Commit-Position: refs/heads/master@{#10823}
Prevents double-initialization of decoders due to resolution changes
between initial database settings and first incoming frame.
BUG=webrtc:5251
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1474193002 .
Cr-Commit-Position: refs/heads/master@{#10822}
Also adds a RTC_CHECK in VideoReceiveStream that verifies that decoders
aren't null, since this will attempt to deregister a codec which would
previously fail with an obscure stack trace not indicating what actually
was wrong.
BUG=webrtc:5249
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1479793002 .
Cr-Commit-Position: refs/heads/master@{#10821}
* Move PlatformThread to rtc::.
* Remove ::CreateThread factory method.
* Make non-scoped_ptr from a lot of invocations.
* Make Start/Stop void.
* Remove rtc::Thread priorities, which were unused and would collide.
* Add ::IsRunning() to PlatformThread.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1476453002 .
Cr-Commit-Position: refs/heads/master@{#10812}
This is the last piece of the old directory layout of the modules.
Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1481493004
Cr-Commit-Position: refs/heads/master@{#10803}
This is part of the project that makes RTC rendering more
smooth. We've already finished the developement of the
frame selection algorithm in WebMediaPlayerMS, where we
managed a frame pool, and based on the vsync interval, we
actively select the best frame to render in order to
maximize the rendering smoothness.
Thus the frame timeline control in IncomingVideoStream is
no longer needed, because with sophisticated frame
selection algorithm in WebMediaPlayerMS, the time control
in IncomingVideoStream will do nothing but add some extra
delay.
BUG=514873
Review URL: https://codereview.webrtc.org/1419673014
Cr-Commit-Position: refs/heads/master@{#10781}
Remove the headers that were kept to provide non-breaking updates
of downstream code for https://codereview.webrtc.org/1418913006/
and https://codereview.webrtc.org/1417283007/.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
Review URL: https://codereview.webrtc.org/1467173003
Cr-Commit-Position: refs/heads/master@{#10773}
This works aims to:
-More clearly separate the functionalities in the AEC.
-Make the inputs and outputs to functions more clear (currently the state struct is often passed as a parameter to the functions and the functions use members of the state both as inputs and outputs, which reduces the readability of the code and makes it difficult to change/refactor.
What is done in this CL:
-Most of what belongs to the echo subtraction functionality has been moved to a separate function.
-The NonLinearProcessing function has been renamed to EchoSuppressor which I think is more appropriate.
-Part of the code was replaced by a call to the TimeToFrequency function (which was also suggested by an existing todo).
-For consistency, a function FrequencyToTime doing the opposite of TimeToFrequency was added and part of the code was moved to that.
-The ScaleErrorSignal function was changed to no longer have the state as an input parameter. This entailed also changing the corresponding assembly optimized files accordingly.
Testing:
-The changes have been tested for bitexactness on Linux using a fairly extensive test.
-All the unittests pass on linux.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1455163006
Cr-Commit-Position: refs/heads/master@{#10764}
With this in, the only compilation errors left seems
related to yasm and libjpeg_turbo.
Notice the below example builds x86 builds (not ARM) since if
specifying target_cpu="arm", the gn step fails (separate issue).
BUG=webrtc:5213, webrtc:5195, chromium:459705
TESTED=Passing compilation with:
gn gen --args="target_os=\"ios\"" out/Default
ninja -C out/Default rtc_base audio_device
Review URL: https://codereview.webrtc.org/1471663002
Cr-Commit-Position: refs/heads/master@{#10763}
