Reason for revert:
Reverting since this seems to break multiple WebRTC Perf buildbots
Original issue's description:
> Don't hardcode MediaType::ANY in FakeNetworkPipe.
>
> Instead let each test set the appropriate media type. This simplifies
> demuxing in Call and later in RtpTransportController.
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2774463003
> Cr-Commit-Position: refs/heads/master@{#17418}
> Committed: 9c47b00e24TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2784543002
Cr-Commit-Position: refs/heads/master@{#17427}
Instead let each test set the appropriate media type. This simplifies
demuxing in Call and later in RtpTransportController.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2774463003
Cr-Commit-Position: refs/heads/master@{#17418}
In ViEEncoder, try to reduce framerate instead of resolution if the
current degradation preference is maintain-resolution rather than
balanced.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2716643002
Cr-Commit-Position: refs/heads/master@{#17327}
This avoids the situation where an encoder, not supporting certain
screen content settings, is created for a config where screencast is
off, and later ReconfigureEncoder() is called updating the configuration
but not the encoder instance, causing an inconsistency in the encoder's
InitEncode() call.
TBR=pthatcher@webrtc.org
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2710493008
Cr-Commit-Position: refs/heads/master@{#16921}
Also fixes a bug where RTCP transport feedback was sent even though RTCP was disabled.
May affect perf numbers since the behavior of the send-side BWE differs a lot from the recv-side BWE.
BUG=webrtc:7111
Review-Url: https://codereview.webrtc.org/2669413003
Cr-Commit-Position: refs/heads/master@{#16451}
Also mark the render_time_ms getter function and the ntp timestamp
as deprecated.
BUG=webrtc:6977
Review-Url: https://codereview.webrtc.org/2633493002
Cr-Commit-Position: refs/heads/master@{#16354}
Test disabled on Windows due to failures on Win Msan, Win64 Debug, Win
SyzyAsan, Win32 Debug and others.
TBR=sprang@webrtc.org
BUG=webrtc:6886
NOTRY=True
Review-Url: https://codereview.webrtc.org/2657233002
Cr-Commit-Position: refs/heads/master@{#16320}
The MultithreadedFakeH264Encoder is a derived class from FakeEncoder
and FakeH264Encoder, and these should thus also be thread safe.
TESTED=Ran "out/Tsan/video_engine_tests --gtest_filter="*Multithreaded*" --gtest_repeat=100" with is_debug=false, dcheck_always_on=true, is_tsan=true.
BUG=webrtc:6943
Review-Url: https://codereview.webrtc.org/2604403003
Cr-Commit-Position: refs/heads/master@{#16093}
Add RTC_DEPRACATed anonymous unions to not break downstream projects.
Orignal issue's description:
> commit 0ad21111fcc57a7e978edba3c4263f0062d7f9ff
> Author: danilchap <danilchap@webrtc.org>
> Date: Mon Dec 19 09:36:33 2016 -0800
>
> Revert of Rename RTPVideoHeader.isFirstPacket to
> .is_first_packet_in_frame. (patchset #1 id:1 of
> https://codereview.webrtc.org/2574943003/ )
>
> Reason for revert:
> breaks downstream project.
>
> Can you make this change in a compatible way using anonymous
> union:
> union {
> bool is_first_packet_in_frame;
> RTC_DEPRECATED bool isFirstPacket;
> };
> (unfortunetly this this treak breaks braced initialization in
> rtp_rtcp_impl_unittest.cc,
> so that should be rewritting in a more classic way)
>
> Original issue's description:
> > Rename RTPVideoHeader.isFirstPacket to
> > .is_first_packet_in_frame.
> >
> > Name should represent the actual meaning.
> >
> > BUG=None
> >
> > Review-Url: https://codereview.webrtc.org/2574943003
> > Cr-Commit-Position: refs/heads/master@{#15684}
> > Committed:
> > efde908380
>
> TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days
> ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2589783003
> Cr-Commit-Position: refs/heads/master@{#15686}
>
BUG=None
Review-Url: https://codereview.webrtc.org/2614503002
Cr-Commit-Position: refs/heads/master@{#15987}
This test was inadvertently made useless in https://codereview.webrtc.org/2558303002/.
Prior to that CL, at least 100 media packets -- but no ULPFEC packets -- had to
be sent before the test claimed that ULPFEC was not utilized. This CL reinstates
that behaviour, but instead of counting packets we expect the test to time out
when ULPFEC should not be utilized.
This solution is simple in code, but increases the test duration from 70 ms
to 10000 ms. Note, however, that the 70 ms duration was for a test that didn't
test anything interesting...
TESTED=Using gtest-parallel locally: 2000 test runs without flakes.
BUG=webrtc:6851
Review-Url: https://codereview.webrtc.org/2592693002
Cr-Commit-Position: refs/heads/master@{#15859}
Reason for revert:
breaks downstream project.
Can you make this change in a compatible way using anonymous union:
union {
bool is_first_packet_in_frame;
RTC_DEPRECATED bool isFirstPacket;
};
(unfortunetly this this treak breaks braced initialization in rtp_rtcp_impl_unittest.cc,
so that should be rewritting in a more classic way)
Original issue's description:
> Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
>
> Name should represent the actual meaning.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2574943003
> Cr-Commit-Position: refs/heads/master@{#15684}
> Committed: efde908380TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2589783003
Cr-Commit-Position: refs/heads/master@{#15686}
The changes here are the same as in https://codereview.webrtc.org/2523993002/:
- reduce packet loss from 50% to 5%, to allow the BWE to ramp up better.
- Do not wait for 100 packets to be sent before letting the test pass.
BUG=webrtc:6851
Review-Url: https://codereview.webrtc.org/2558303002
Cr-Commit-Position: refs/heads/master@{#15542}
The intention is to make the tests less flaky.
BUG=webrtc:6744
Review-Url: https://codereview.webrtc.org/2552713002
Cr-Commit-Position: refs/heads/master@{#15421}
There's no longer any need to make the two arguments have the same
signedness, so we can drop the "u" suffix on literal integer
arguments.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2535593002
Cr-Commit-Position: refs/heads/master@{#15280}
Previously ProbeController was starting probing as soon as SetBitrates()
is called. As result these probes would often timeout while connection
is being established. Now ProbeController receives notifications about
network route changes. This allows to start probing only when transport
is connected. This also makes it possible to restart probing whenever
transport route changes (will be done in a separate change).
BUG=webrtc:6332
R=honghaiz@webrtc.org, philipel@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2458863002 .
Committed: https://crrev.com/5c99c76255ee7bface3c742c25fb5617748ac86e
Cr-Original-Commit-Position: refs/heads/master@{#15094}
Cr-Commit-Position: refs/heads/master@{#15204}
Reason for revert:
The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change:
function removeVideoCodec(offerSdp) {
- offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n',
- 'a=rtpmap:100 XVP8/90000\r\n');
+ offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/,
+ 'a=rtpmap:$1 XVP8/90000\r\n');
return offerSdp;
}
Original issue's description:
> Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ )
>
> Reason for revert:
> Breaks chromium.fyi test:
> WebRtcBrowserTest.NegotiateUnsupportedVideoCodec
>
> Original issue's description:
> > Stop using hardcoded payload types for video codecs
> >
> > This CL stops using hardcoded payload types for different video codecs
> > and will dynamically assign them payload types incrementally from 96 to
> > 127 instead.
> >
> > This CL:
> > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in
> > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for
> > internally supported software codecs instead. The purpose is to
> > streamline the payload type assignment in webrtcvideoengine2.cc which
> > will now have two encoder factories of the same
> > WebRtcVideoEncoderFactory type; one internal and one external.
> > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec
> > instead.
> > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and
> > moves the create function to the internal encoder factory instead.
> > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an
> > interface without any static functions.
> > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies
> > the internal and external codecs and assigns them payload types
> > incrementally from 96 to 127.
> > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about
> > what payload types will be used.
> >
> > BUG=webrtc:6677,webrtc:6705
> > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org
> >
> > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b
> > Cr-Commit-Position: refs/heads/master@{#15135}
>
> TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6677,webrtc:6705
>
> Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591
> Cr-Commit-Position: refs/heads/master@{#15140}
TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6677,webrtc:6705
Review-Url: https://codereview.webrtc.org/2511933002
Cr-Commit-Position: refs/heads/master@{#15148}
Reason for revert:
Breaks chromium.fyi test:
WebRtcBrowserTest.NegotiateUnsupportedVideoCodec
Original issue's description:
> Stop using hardcoded payload types for video codecs
>
> This CL stops using hardcoded payload types for different video codecs
> and will dynamically assign them payload types incrementally from 96 to
> 127 instead.
>
> This CL:
> * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in
> webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for
> internally supported software codecs instead. The purpose is to
> streamline the payload type assignment in webrtcvideoengine2.cc which
> will now have two encoder factories of the same
> WebRtcVideoEncoderFactory type; one internal and one external.
> * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec
> instead.
> * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and
> moves the create function to the internal encoder factory instead.
> * Removes video_encoder.cc. webrtc::VideoEncoder is now just an
> interface without any static functions.
> * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies
> the internal and external codecs and assigns them payload types
> incrementally from 96 to 127.
> * Updates webrtcvideoengine2_unittest.cc and removes assumptions about
> what payload types will be used.
>
> BUG=webrtc:6677,webrtc:6705
> R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b
> Cr-Commit-Position: refs/heads/master@{#15135}
TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6677,webrtc:6705
Review-Url: https://codereview.webrtc.org/2513633002
Cr-Commit-Position: refs/heads/master@{#15140}
This CL stops using hardcoded payload types for different video codecs
and will dynamically assign them payload types incrementally from 96 to
127 instead.
This CL:
* Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in
webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for
internally supported software codecs instead. The purpose is to
streamline the payload type assignment in webrtcvideoengine2.cc which
will now have two encoder factories of the same
WebRtcVideoEncoderFactory type; one internal and one external.
* Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec
instead.
* Removes 'static VideoEncoder* Create(EncoderType codec_type)' and
moves the create function to the internal encoder factory instead.
* Removes video_encoder.cc. webrtc::VideoEncoder is now just an
interface without any static functions.
* The function GetSupportedCodecs in webrtcvideoengine2.cc unifies
the internal and external codecs and assigns them payload types
incrementally from 96 to 127.
* Updates webrtcvideoengine2_unittest.cc and removes assumptions about
what payload types will be used.
BUG=webrtc:6677,webrtc:6705
R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2493133002 .
Cr-Commit-Position: refs/heads/master@{#15135}
This is yet another reland of https://codereview.webrtc.org/2434073003/
including two fixes:
1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that.
2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams.
Please review only the changes after patch set 1.
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
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2510583002 .
Cr-Commit-Position: refs/heads/master@{#15105}
Reason for revert:
It broke downstream test.
Original issue's description:
> Start probes only after network is connected.
>
> Previously ProbeController was starting probing as soon as SetBitrates()
> is called. As result these probes would often timeout while connection
> is being established. Now ProbeController receives notifications about
> network route changes. This allows to start probing only when transport
> is connected. This also makes it possible to restart probing whenever
> transport route changes (will be done in a separate change).
>
> BUG=webrtc:6332
>
> Committed: https://crrev.com/5c99c76255ee7bface3c742c25fb5617748ac86e
> Cr-Commit-Position: refs/heads/master@{#15094}
TBR=philipel@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6332
Review-Url: https://codereview.webrtc.org/2504783002
Cr-Commit-Position: refs/heads/master@{#15098}
Previously ProbeController was starting probing as soon as SetBitrates()
is called. As result these probes would often timeout while connection
is being established. Now ProbeController receives notifications about
network route changes. This allows to start probing only when transport
is connected. This also makes it possible to restart probing whenever
transport route changes (will be done in a separate change).
BUG=webrtc:6332
Review-Url: https://codereview.webrtc.org/2458863002
Cr-Commit-Position: refs/heads/master@{#15094}
This is needed for the following coming tests: VideoSendStream, end-to-end,
full stack, and video_loopback.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2500943002
Cr-Commit-Position: refs/heads/master@{#15087}
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 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}
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}
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}
Reason for revert:
Breaks perf tests.
Original issue's 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/8f46c679d24a05b3f08e02c6d91ec9637f34e24f
> Cr-Commit-Position: refs/heads/master@{#14998}
TBR=stefan@webrtc.org,perkj@webrtc.org,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/2489843002
Cr-Commit-Position: refs/heads/master@{#15001}
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/2434073003
Cr-Commit-Position: refs/heads/master@{#14998}
Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
These methods should no longer be used anywhere and it's safe to remove
them.
BUG=chromium:621691
Committed: https://crrev.com/c681250aaa2025836db7669694e323898e5c2ca7
Review-Url: https://codereview.webrtc.org/2405173006
Cr-Original-Commit-Position: refs/heads/master@{#14923}
Cr-Commit-Position: refs/heads/master@{#14935}
Reason for revert:
Still breaks internal downstream project.
Sergey: Please update internal project before relanding this.
Original issue's description:
> Remove deprected functions from EncodedImageCallback and RtpRtcp
>
> Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
> These methods should no longer be used anywhere and it's safe to remove
> them.
>
> BUG=chromium:621691
>
> Committed: https://crrev.com/c681250aaa2025836db7669694e323898e5c2ca7
> Cr-Commit-Position: refs/heads/master@{#14923}
TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2479643002
Cr-Commit-Position: refs/heads/master@{#14925}
