This CL will start to distinguish H264 profiles during SDP negotiation.
We currently don't look at the H264 profile at all and assume they are
all Constrained Baseline Level 3.1. This CL will start to check profiles
for equality when matching, and will generate the correct answer H264
level.
Each local supported H264 profile needs to be listed explicitly in the
list of local supported codecs, even if they are redundant. For example,
Baseline profile should be listed explicitly even though another profile
that is a superset of Baseline is also listed. The reason for this is to
simplify the code and avoid profile intersection during matching. So
VideoCodec::Matches will check for profile equality, and not check if
one codec is a subset of the other. This also leads to the nice property
that VideoCodec::Matches is symmetric, i.e. iif a.Matches(b) then
b.Matches(a).
BUG=webrtc:6337
TBR=tkchin@webrtc.org
Review-Url: https://codereview.webrtc.org/2483173002
Cr-Commit-Position: refs/heads/master@{#15051}
This is a trivial change to remove MockScreenCapturerCallback, and use
MockDesktopCapturerCallback to replace it.
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2494013003
Cr-Commit-Position: refs/heads/master@{#15047}
A tiny but critical change to avoid a crash failure in DirectX capturer.
A good news is this failure is caught by ScreenCapturer integration tests.
BUG=314516
Review-Url: https://codereview.webrtc.org/2494893002
Cr-Commit-Position: refs/heads/master@{#15046}
The old CroppingWindowCapturer::Create() function returns a raw pointer, which
cannot explain the ownership.
So this change adds a CreateCapturer() function to return a unique_ptr.
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2496993003
Cr-Commit-Position: refs/heads/master@{#15045}
We want to keep these warnings on. They were switched off temporarily
to unblock chromium roll.
BUG=webrtc:6520
Review-Url: https://codereview.webrtc.org/2493873005
Cr-Commit-Position: refs/heads/master@{#15043}
These functions currently copy cricket::Codec classes by value which is
expensive since they contain e.g. std::map<std::string, std::string>
containers with parameters. This CL avoids copying them altogether.
BUG=webrtc:6337
Review-Url: https://codereview.webrtc.org/2493733003
Cr-Commit-Position: refs/heads/master@{#15040}
In older Chrome versions, the associated payload type in the RTX header
of retransmitted packets was always set to be the original media payload type,
regardless of the actual payload type of the packet. This meant that packets
encapsulated with RED headers had incorrect payload type information in the
RTX header. Due to an assumption in the receiver, this incorrect payload type
information would effectively be undone, leading to a working system.
Albeit working, this behaviour was undesired, and thus removed. In the interim,
several workarounds were introduced to not destroy interop between old and
new Chrome versions:
(1) https://codereview.webrtc.org/1649493004
- If no payload type mapping existed for RED over RTX, the payload type
of the underlying media would be used.
- If RED had been negotiated, received RTX packets would always be
assumed to contain RED.
(2) https://codereview.webrtc.org/1964473002
- If RED was removed from the remote description answer, it would be
disabled in the local receiver as well.
(3) https://codereview.webrtc.org/2033763002
- If RED was negotiated in the SDP, it would always be used, regardless
if ULPFEC was negotiated and used, or not.
Since the Chrome versions that exhibited the original bug now are very old,
this CL removes the workarounds from (1) and (2). In particular, after this
change, we will have the following behaviour:
- We assume that a payload type mapping for RED over RTX always is set.
If this is not the case, the RTX packet is not sent.
- The associated payload type of received RTX packets will always be obeyed.
- The (non)-existence of RED in the remote description does not affect the
local receiver.
The workaround in (3) still needs to exist, in order to interop with receivers
that did not have the workarounds in (1) and (2) removed. The change in (3)
can be removed in a couple of Chrome versions.
TESTED=Using AppRTC between patched Chrome (connected to ethernet) and standard Chrome M54 (connected to lossy internal Google WiFi), with and without FEC turned off using AppRTC flag. Also using "Munge SDP" sample on patched Chrome over loopback interface, with 100ms delay and 5% packet loss simulated using tc.
BUG=webrtc:6650
Review-Url: https://codereview.webrtc.org/2469093003
Cr-Commit-Position: refs/heads/master@{#15038}
This change removes ScreenCapturer and WindowCapturer from WebRTC.
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2490063002
Cr-Commit-Position: refs/heads/master@{#15033}
This approach extends the H.264 specific information with
a packetization mode enum.
Status: Parameter is in code. No way to set it yet.
Rebase of CL 2009213002
BUG=600254
Review-Url: https://codereview.webrtc.org/2337453002
Cr-Commit-Position: refs/heads/master@{#15032}
Chrome coding standard now discourages use of references to smart
pointers. This cl updates some recent methods to the new conventions.
BUG=webrtc:6672
Review-Url: https://codereview.webrtc.org/2477233004
Cr-Commit-Position: refs/heads/master@{#15028}
Reason for revert:
Broke a google3 build
Original issue's description:
> Declare VideoCodec.codec_specific_info private
>
> This completes the privatization of the codec specific
> information in VideoCodec.
>
> BUG=webrtc:6603
>
> Committed: https://crrev.com/792738640234d81c916ac4458ac72286cb2548a4
> Cr-Commit-Position: refs/heads/master@{#15013}
TBR=tommi@chromium.org,magjed@chromium.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6603
Review-Url: https://codereview.webrtc.org/2491933002
Cr-Commit-Position: refs/heads/master@{#15027}
Reason for revert:
Seems to be causing flakiness in perf test:
FullStackTest.ScreenshareSlidesVP8_2TL_LossyNet
Original issue's description:
> Reland of Issue 2434073003: Extract bitrate allocation ...
>
> This is a reland of https://codereview.webrtc.org/2434073003/ including
> some fixes for failing test cases.
>
> Original description:
>
> Extract bitrate allocation of spatial/temporal layers out of codec impl.
>
> This CL makes a number of intervowen changes:
>
> * Add BitrateAllocation struct, that contains a codec independent view
> of how the target bitrate is distributed over spatial and temporal
> layers.
>
> * Adds the BitrateAllocator interface, which takes a bitrate and frame
> rate and produces a BitrateAllocation.
>
> * A default (non layered) implementation is added, and
> SimulcastRateAllocator is extended to fully handle VP8 allocation.
> This includes capturing TemporalLayer instances created by the
> encoder.
>
> * ViEEncoder now owns both the bitrate allocator and the temporal layer
> factories for VP8. This allows allocation to happen fully outside of
> the encoder implementation.
>
> This refactoring will make it possible for ViEEncoder to signal the
> full picture of target bitrates to the RTCP module.
>
> BUG=webrtc:6301
>
> Committed: https://crrev.com/647bf43dcb2fd16fccf276bd94dc4400728bb405
> Cr-Commit-Position: refs/heads/master@{#15023}
TBR=mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2491393002
Cr-Commit-Position: refs/heads/master@{#15026}
The likelihood is logged as a percentage, with 100 bins in the histogram.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2487243002
Cr-Commit-Position: refs/heads/master@{#15025}
Due to bugs.webrtc.org/216 code that includes
//webrtc/modules/audio_device:mock_audio_device fails to compile on
Windows unless a warning flag is switched off. It seems that
googlemock changes types of overridden virtual method parameters from
'const int' to 'int', which causes the VS compiler to report an error.
The problem was not solved by suppressing the flag wd4373 in
//webrtc/modules/audio_device:mock_audio_device, because targets that
include the headers are compiled separately.
This CL adds the flag suppression to the GN variable
|all_dependent_configs|. Then GN will apply the configuration to all
reachable dependencies. This is needed to reduce clutter and extra
conditions in dependency build targets.
The reason for |all_dependent_configs| before |public_configs| is
for a situation in which a targets headers include the headers from
this target. Then dependencies of dependencies will have a copy of
this targets' code after preprocessing, and compilation will fail.
This will happen if we e.g. change mock_voice_engine to return a
MockAudioTransport or MockAudioDevice.
This change has been tested by compiling a dependent CL
(https://codereview.webrtc.org/2454373002/) which uses these mocks
on Windows without suppressing the flag.
There is no GYP change, because test code has been removed from GYP.
NOTRY=True
BUG=webrtc:216
Review-Url: https://codereview.webrtc.org/2492713003
Cr-Commit-Position: refs/heads/master@{#15024}
This is a reland of https://codereview.webrtc.org/2434073003/ including
some fixes for failing test cases.
Original description:
Extract bitrate allocation of spatial/temporal layers out of codec impl.
This CL makes a number of intervowen changes:
* Add BitrateAllocation struct, that contains a codec independent view
of how the target bitrate is distributed over spatial and temporal
layers.
* Adds the BitrateAllocator interface, which takes a bitrate and frame
rate and produces a BitrateAllocation.
* A default (non layered) implementation is added, and
SimulcastRateAllocator is extended to fully handle VP8 allocation.
This includes capturing TemporalLayer instances created by the
encoder.
* ViEEncoder now owns both the bitrate allocator and the temporal layer
factories for VP8. This allows allocation to happen fully outside of
the encoder implementation.
This refactoring will make it possible for ViEEncoder to signal the
full picture of target bitrates to the RTCP module.
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2488833004
Cr-Commit-Position: refs/heads/master@{#15023}
This is not the right place for a SequencedTaskChecker, as we can
not make any guarantees about the thread this method runs on.
We were hitting this check on Android and iOS whenever the encoder
would be reconfigured. Access to these ivars should be guarded
by a lock.
As a bonus, an unused method declaration was removed.
BUG=webrtc:6686
Review-Url: https://codereview.webrtc.org/2495483002
Cr-Commit-Position: refs/heads/master@{#15019}
This CL adds the ability for RTPSenderVideo to generate and send
FlexFEC packets.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2490523002
Cr-Commit-Position: refs/heads/master@{#15016}
Reason for revert:
This CL probably broke Chromium FYI.
Original issue's description:
> Stop caching supported codecs in WebRtcVideoEngine2
>
> We currently cache the result of GetSupportedCodecs in a member variable
> |video_codecs_| in WebRtcVideoEngine2. This means we need to keep
> |video_codecs_| and the result of GetSupportedCodecs in sync, which is
> error prone. It's simpler to just call GetSupportedCodecs when we need
> it, and we actually end up making fewer calls, so it's faster as well.
> This CL also returns all std::vectors by-value instead of by-ref. Move
> semantic together with in-place filtering of codecs actually end up with
> fewer copies, and it's also simpler to not return references.
>
> BUG=webrtc:6337
>
> Committed: https://crrev.com/9f71ec5a3e3175751f4475b126cfda89767363f2
> Cr-Commit-Position: refs/heads/master@{#15007}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6337
Review-Url: https://codereview.webrtc.org/2489173004
Cr-Commit-Position: refs/heads/master@{#15014}
This completes the privatization of the codec specific
information in VideoCodec.
BUG=webrtc:6603
Review-Url: https://codereview.webrtc.org/2452963002
Cr-Commit-Position: refs/heads/master@{#15013}
There are currently two nearly identical classes called
MockAudioTransport defined in two unit tests:
android/audio_transport_unittest.cc and
/ios/audio_transport_unittest_ios.cc
This change defines a common mocked AudioTransport. The two current
mocks are rewritten to use the common one. A GN target is created for
this mock and MockAudioDevice.
This change will allow to provide a mocked AudioTransport to
AudioState in a dependent CL https://codereview.webrtc.org/2454373002/
BUG=webrtc:6346
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2493483002
Cr-Commit-Position: refs/heads/master@{#15010}
We currently cache the result of GetSupportedCodecs in a member variable
|video_codecs_| in WebRtcVideoEngine2. This means we need to keep
|video_codecs_| and the result of GetSupportedCodecs in sync, which is
error prone. It's simpler to just call GetSupportedCodecs when we need
it, and we actually end up making fewer calls, so it's faster as well.
This CL also returns all std::vectors by-value instead of by-ref. Move
semantic together with in-place filtering of codecs actually end up with
fewer copies, and it's also simpler to not return references.
BUG=webrtc:6337
Review-Url: https://codereview.webrtc.org/2492473002
Cr-Commit-Position: refs/heads/master@{#15007}
