Reason for revert:
Perf test broke as it made assumptions that quality scaling was turned off. This turns out not to be the case. Fixed by turning quality scaling off for the tests.
Original issue's description:
> Revert of Drop frames until specified bitrate is achieved. (patchset #12 id:240001 of https://codereview.webrtc.org/2630333002/ )
>
> Reason for revert:
> due to failures on perf tests (not on perf stats, but fails running due to dcheck failures), see e.g., https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20(K%20Nexus5)
>
> Original issue's description:
> > Drop frames until specified bitrate is achieved.
> >
> > This CL fixes a regression introduced with the new quality scaler
> > where the video would no longer start in a scaled mode. This CL adds
> > code that compares incoming captured frames to the target bitrate,
> > and if they are found to be too large, they are dropped and sinkWants
> > set to a lower resolution. The number of dropped frames should be low
> > (0-4 in most cases) and should not introduce a noticeable delay, or
> > at least should be preferrable to having the first 2-4 seconds of video
> > have very low quality.
> >
> > BUG=webrtc:6953
> >
> > Review-Url: https://codereview.webrtc.org/2630333002
> > Cr-Commit-Position: refs/heads/master@{#16391}
> > Committed: 83399caec5
>
> TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6953
>
> Review-Url: https://codereview.webrtc.org/2666303002
> Cr-Commit-Position: refs/heads/master@{#16395}
> Committed: 35fc2aa82fTBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,minyue@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6953
Review-Url: https://codereview.webrtc.org/2675223002
Cr-Commit-Position: refs/heads/master@{#16473}
These structs will be used for ORTC objects (and their WebRTC
equivalents).
This CL also introduces some minor changes to the existing implemented
structs:
- max_bitrate_bps uses rtc::Optional instead of "-1 means unset"
- "mime_type" turned into "name"/"kind" (which can be used to form the
MIME type string, if needed).
- clock_rate and channels changed to rtc::Optional, since they will
need to be for RtpSender.send().
- Renamed "channels" to "num_channels" (the ORTC name, which I prefer).
BUG=webrtc:7013, webrtc:7112
Review-Url: https://codereview.webrtc.org/2651883010
Cr-Commit-Position: refs/heads/master@{#16437}
Fix: Order of assignments is now correct, after being incorrect
due to an incorrect merge between
https://codereview.webrtc.org/2617373002/ and
https://codereview.webrtc.org/2589713003.
Improvement: Set parameters in more places, allowing for
correct reconfiguration. Add TODOs to point of minor issues
with current configuration.
TESTED=By locally patching an application using this code.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2660403004
Cr-Commit-Position: refs/heads/master@{#16431}
Reason for revert:
due to failures on perf tests (not on perf stats, but fails running due to dcheck failures), see e.g., https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20(K%20Nexus5)
Original issue's description:
> Drop frames until specified bitrate is achieved.
>
> This CL fixes a regression introduced with the new quality scaler
> where the video would no longer start in a scaled mode. This CL adds
> code that compares incoming captured frames to the target bitrate,
> and if they are found to be too large, they are dropped and sinkWants
> set to a lower resolution. The number of dropped frames should be low
> (0-4 in most cases) and should not introduce a noticeable delay, or
> at least should be preferrable to having the first 2-4 seconds of video
> have very low quality.
>
> BUG=webrtc:6953
>
> Review-Url: https://codereview.webrtc.org/2630333002
> Cr-Commit-Position: refs/heads/master@{#16391}
> Committed: 83399caec5TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6953
Review-Url: https://codereview.webrtc.org/2666303002
Cr-Commit-Position: refs/heads/master@{#16395}
This CL fixes a regression introduced with the new quality scaler
where the video would no longer start in a scaled mode. This CL adds
code that compares incoming captured frames to the target bitrate,
and if they are found to be too large, they are dropped and sinkWants
set to a lower resolution. The number of dropped frames should be low
(0-4 in most cases) and should not introduce a noticeable delay, or
at least should be preferrable to having the first 2-4 seconds of video
have very low quality.
BUG=webrtc:6953
Review-Url: https://codereview.webrtc.org/2630333002
Cr-Commit-Position: refs/heads/master@{#16391}
Reason for revert:
Downstream project relied on changed struct.
Transition made possible by https://codereview.webrtc.org/2655243006/.
Original issue's description:
> Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ )
>
> Reason for revert:
> Breaks internal downstream project.
>
> Original issue's description:
> > Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters.
> >
> > Prior to this CL, received RTX (associated) payload types were only configured
> > when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX
> > SSRC was set up.
> >
> > After this CL, the RTX (associated) payload types are set in
> > WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set
> > them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since
> > that is the code path that sets other SSRCs.
> >
> > As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed.
> > We remove the possibility for each video payload type to have an associated
> > specific RTX SSRC. Although the config previously allowed for this, all payload
> > types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry
> > did not support multiple SSRCs. This change to the config struct should thus not
> > have any functional impact. The change does however affect the RtcEventLog, since
> > that is used for storing the VideoReceiveStream::Configs. For simplicity,
> > this CL does not change the event log proto definitions, instead duplicating
> > the serialized RTX SSRCs such that they fit in the existing proto definition.
> >
> > BUG=webrtc:7011
> >
> > Review-Url: https://codereview.webrtc.org/2646073004
> > Cr-Commit-Position: refs/heads/master@{#16302}
> > Committed: fe2bef39cd
>
> TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7011
>
> Review-Url: https://codereview.webrtc.org/2649323010
> Cr-Commit-Position: refs/heads/master@{#16307}
> Committed: e4974953ceTBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
# NOTREECHECKS=true
# NOTRY=true
BUG=webrtc:7011
Review-Url: https://codereview.webrtc.org/2654163006
Cr-Commit-Position: refs/heads/master@{#16322}
Reason for revert:
Breaks internal downstream project.
Original issue's description:
> Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters.
>
> Prior to this CL, received RTX (associated) payload types were only configured
> when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX
> SSRC was set up.
>
> After this CL, the RTX (associated) payload types are set in
> WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set
> them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since
> that is the code path that sets other SSRCs.
>
> As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed.
> We remove the possibility for each video payload type to have an associated
> specific RTX SSRC. Although the config previously allowed for this, all payload
> types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry
> did not support multiple SSRCs. This change to the config struct should thus not
> have any functional impact. The change does however affect the RtcEventLog, since
> that is used for storing the VideoReceiveStream::Configs. For simplicity,
> this CL does not change the event log proto definitions, instead duplicating
> the serialized RTX SSRCs such that they fit in the existing proto definition.
>
> BUG=webrtc:7011
>
> Review-Url: https://codereview.webrtc.org/2646073004
> Cr-Commit-Position: refs/heads/master@{#16302}
> Committed: fe2bef39cdTBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7011
Review-Url: https://codereview.webrtc.org/2649323010
Cr-Commit-Position: refs/heads/master@{#16307}
Prior to this CL, received RTX (associated) payload types were only configured
when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX
SSRC was set up.
After this CL, the RTX (associated) payload types are set in
WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set
them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since
that is the code path that sets other SSRCs.
As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed.
We remove the possibility for each video payload type to have an associated
specific RTX SSRC. Although the config previously allowed for this, all payload
types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry
did not support multiple SSRCs. This change to the config struct should thus not
have any functional impact. The change does however affect the RtcEventLog, since
that is used for storing the VideoReceiveStream::Configs. For simplicity,
this CL does not change the event log proto definitions, instead duplicating
the serialized RTX SSRCs such that they fit in the existing proto definition.
BUG=webrtc:7011
Review-Url: https://codereview.webrtc.org/2646073004
Cr-Commit-Position: refs/heads/master@{#16302}
The original CL was reverted because of a bug discovered by the
chromium bots. Description of that CL:
> Review-Url: https://codereview.webrtc.org/2636443002
> Cr-Commit-Position: refs/heads/master@{#16135}
> Committed: a28e971e3b
The first patch set of this CL is the same as r16135.
Subsequence patch sets are the fixes applied.
Some new test cases have been added, which reveal a few more bugs that
have also been fixed.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2641133002
Cr-Commit-Position: refs/heads/master@{#16299}
The existence of FlexfecConfig is due to a naive design. Now when it
is not used on the receiving side (see https://codereview.webrtc.org/2542413002),
it is time to remove it from the sending side as well.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2621573002
Cr-Commit-Position: refs/heads/master@{#16097}
Earlier, the FlexFEC codec would receive the same default RTCP feedback
params as the media codecs. Since most of these are not used, there is
no point negotiating them.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2623513002
Cr-Commit-Position: refs/heads/master@{#16057}
Moves webrtc/common_video/rotation.h and parts of
webrtc/common_video/include/video_frame_buffer.h and
webrtc/video_frame.h, and adds to a new GN target api:video_frame_api.
BUG=webrtc:5880
Review-Url: https://codereview.webrtc.org/2517173004
Cr-Commit-Position: refs/heads/master@{#15993}
With this, RtpSender and RtpReceiver will always return an SSRC if one
is available. Also, attempts to change the SSRC with SetParameters will
fail, rather than silently doing nothing.
BUG=webrtc:6888
Review-Url: https://codereview.webrtc.org/2567333004
Cr-Commit-Position: refs/heads/master@{#15939}
A WebRtcVideoEngine2 object seems to be reused between PeerConnections,
which means that the field trial added in
https://codereview.webrtc.org/2511703002/ may not activate/deactivate
as intended between calls. This CL removes the caching of video codecs,
which gets rid of this problem.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2521393004
Cr-Commit-Position: refs/heads/master@{#15265}
It will be necessary to keep the H264 profile information in
VideoReceiveStream::Decoder. I think it will be easier now and for the
future to just store all of the codec parameters unmodified in
VideoReceiveStream::Decoder instead of extracting a subset of them to an
ad hoc class.
BUG=webrtc:6743,webrtc:5948
Review-Url: https://codereview.webrtc.org/2523773003
Cr-Commit-Position: refs/heads/master@{#15239}
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}
This CL interfaces the SDP information (payload types and
SSRCs) about FlexFEC with the corresponding configs at the
Call layer. It also adds a field trial, which when active
will expose FlexFEC in the default codec list, thus showing
up in the default SDP.
BUG=webrtc:5654
R=magjed@webrtc.org, stefan@webrtc.orgCC=perkj@webrtc.org
Review-Url: https://codereview.webrtc.org/2511703002
Cr-Commit-Position: refs/heads/master@{#15184}
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}
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 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}
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 removes the VideoSendStream::LoadObserver interface and the implementation in WebrtcVideoSendStream and replace it with VideoSinkWants through the VideoSourceInterface.
To do that that, some stats for CPU adaptation is moved into VideoSendStream. Also handling of the CVO rtp header extension is moved to VideoSendStreamImpl.
BUG=webrtc:5687
TBR=mflodman@webrtc.org
Review-Url: https://codereview.webrtc.org/2304363002
Cr-Commit-Position: refs/heads/master@{#14877}
This CL introduces two new functions to the WebRtcVideoEncoderFactory
interface based on cricket::VideoFormat instead of
WebRtcVideoEncoderFactory::VideoCodec. The functions are:
WebRtcVideoEncoderFactory::CreateVideoEncoder() and
WebRtcVideoEncoderFactory::supported_codecs(). In order to make a smooth
transition to the new interface, the old functions are kept, and default
implementations are provided for both the old and new functions so that
external clients can switch from the old to the new functions in peace.
The default implementations will just convert between
cricket::VideoFormat and WebRtcVideoEncoderFactory::VideoCodec. Once all
external clients have updated their code, the plan is to remove the old
functions and all default implementations to make
WebRtcVideoEncoderFactory a pure interface again.
BUG=webrtc:6402,webrtc:6337
Review-Url: https://codereview.webrtc.org/2449993003
Cr-Commit-Position: refs/heads/master@{#14826}
Reason for revert:
Seems to break WebRTC perf tests.
Original issue's description:
> Change ViEEncoder to not reconfigure the encoder until the video resolution is known.
>
> BUG=b/32285861
>
> Committed: https://crrev.com/461c29e436b5bd7ed019e83024e24dc8e86ec9b9
> Cr-Commit-Position: refs/heads/master@{#14813}
TBR=skvlad@webrtc.org,pbos@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=b/32285861
Review-Url: https://codereview.webrtc.org/2457083002
Cr-Commit-Position: refs/heads/master@{#14815}
Since WebRtcVideoSendStream have reconfigures the send codec to match the incoming captured frames widht and height they have not been used.
Framerate has just been set when parsing sdp to 60fps and not changed elsewhere.
This cl require some upstream projects to change first.
BUG=webrtc:5332
Review-Url: https://codereview.webrtc.org/2408153002
Cr-Commit-Position: refs/heads/master@{#14733}
This CL is a pure refactoring which should not result in any functinal
changes. It moves ownership of the RtcEventLog from webrtc::Call to the
webrtc::PeerConnection object.
This is done so that we can add RtcEventLog support for ICE events -
which will require the TransportController to have a pointer to the
RtcEventLog. PeerConnection is the closest common owner of both Call and
TransportController (through WebRtcSession).
BUG=webrtc:6393
Review-Url: https://codereview.webrtc.org/2353033005
Cr-Commit-Position: refs/heads/master@{#14578}
Also rename some related minor methods. No functional changes
are intended/expected.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2391963002
Cr-Commit-Position: refs/heads/master@{#14513}
The original landed cl is in patchset 1.
The following patchset fix VideoQualityTest as well as fix the case where max_bitrate is set in the SendParams. A unit test is added for that as well.
Original cl description:
Let ViEEncoder handle resolution changes.
This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.
With this change, many variables in WebRtcVideoSendStream no longer need to be locked.
BUG=webrtc:5687, webrtc:6371, webrtc:5332
Review-Url: https://codereview.webrtc.org/2386573002
Cr-Commit-Position: refs/heads/master@{#14467}
This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.
With this change, many variables in WebRtcVideoSendStream no longer need to be locked.
BUG=webrtc:5687, webrtc:6371, webrtc:5332
Review-Url: https://codereview.webrtc.org/2351633002
Cr-Commit-Position: refs/heads/master@{#14445}
