This cl is in preparation for https://codereview.webrtc.org/2060403002/ Add task queue to Call.
In the coming cl the video_sender, and i420_buffer_pool will be used on a task queue and therefore SequencedTaskChecker is needed instead of a ThreadChecker.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2149553002
Cr-Commit-Position: refs/heads/master@{#13474}
When the target bitrate is zero, currently VideoSendStream.Stats.target_media_bitrate_bps show the last set rate before the target was set to zero.
BUG=webrtc::5687 b/29574845
Review-Url: https://codereview.webrtc.org/2122743003
Cr-Commit-Position: refs/heads/master@{#13386}
I'm also removing media_optimization_unittest.cc, since it only tested the
suspension logic and nothing else.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/2119503002 .
Cr-Commit-Position: refs/heads/master@{#13355}
This cl change so that VideoSendStream::Start adds the stream as a BitrateObserver and VideoSendStream::Stop removes the stream as observer.
That also means that start will trigger a VideoEncoder::SetRate call with the most recent bitrate estimate.
VideoSendStream::Stop will trigger a VideoEncoder::SetRate with bitrate = 0.
BUG=webrtc:5687 b/28636240
Review-Url: https://codereview.webrtc.org/2070343002
Cr-Commit-Position: refs/heads/master@{#13192}
This cl split the class MediaOptimization into two parts. One that deals with frame dropping and stats and one new class called ProtectionBitrateCalculator that deals with calculating the needed FEC parameters and how much of the estimated network bitrate that can be used by an encoder
Note that the logic of how FEC and the needed bitrates is not changed.
BUG=webrtc:5687
R=asapersson@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1972083002 .
Cr-Commit-Position: refs/heads/master@{#13018}
Remove ViEEncoder::SetNetworkStatus.
Original cl description:
This cl removed ViEEncoder::SetNetworkStatus. Instead the PacedSender will report that frames can not be sent when the network is down and the BitrateController will report an estimated available bandwidth of 0 bps.
Patchset #1 is a pure reland.
Patchset #2 change the bitrate allocator to always return an initial bitrate >0 regardless of the estimates. The observer will be notified though if the network is down.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1972183004
Cr-Commit-Position: refs/heads/master@{#12743}
SSRC knowledge is contained withing VideoSendStream. That also means that debug recording is moved to VideoSendStream.
I think that make sence since that allows debug recording with external encoder implementations one day.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1936503002
Cr-Commit-Position: refs/heads/master@{#12632}
And move encoder name cb to VCMSendStatisticsCallback.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1900193004
Cr-Commit-Position: refs/heads/master@{#12596}
Framerate-reduction code is disabled on all platforms, and this code
adds complexity. It's necessary to react fast, especially on mobile
platforms or other bad network conditions and framerate reduction adds
another step between HD and QVGA.
BUG=webrtc:5678, webrtc:5830
R=jackychen@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1885893002 .
Cr-Commit-Position: refs/heads/master@{#12503}
Instead, use the corresponding method on VideoFrameBuffer. In the process,
reduce code duplication in frame comparison functions used in
the test code.
Make FramesEqual use FrameBufsEqual. Make the latter support texture frames.
The cl also refactors VideoFrame::CopyFrame to use I420Buffer::Copy. This
has possibly undesired side effects of never reusing the frame buffer of
the destination frame, and producing a frame buffer which may use different
stride than the source frame.
BUG=webrtc:5682
Review URL: https://codereview.webrtc.org/1881953002
Cr-Commit-Position: refs/heads/master@{#12373}
Also refactor GenericEncoder to use these file writers, and remove use
of preprocessor to enable file writing.
BUG=
Review URL: https://codereview.webrtc.org/1853813002
Cr-Commit-Position: refs/heads/master@{#12372}
An earlier change moved SetRates to happen on every input frame, but
encoders with internal sources don't receive input frames, so they
weren't getting updated bitrate (and framerate) signals.
BUG=
Review URL: https://codereview.webrtc.org/1682253005
Cr-Commit-Position: refs/heads/master@{#12354}
1. Fix the case of key frame accumulation being incorrect due to the chunk
size being computed at the time of leak based on input frame rate. The issue
is that the count is computed based on key frame ratio and the actual chunk
size computed from current input frame rate. These can be wildly different
especially at the beginning of the stream (key frame ratio defaults based
on 30 fps) resulting in incorrect key frame accumulation causing large frame
drops when the input frame rate is low.
2. Add large delta frame compensation. The current code accounts for key frames
but not large delta frames. This is a common occurence in some application
(remote desktop as an example)
3. Fixes an issue identified by the unit tests. The accumulation of
key frames had an issue in the scenario of a high key frame ratio where
the full key frame was not being accounted for.
3. Removes fast mode and other methods that are mostly dead code.
4. Cleans up variable names as per chromium style.
Review URL: https://codereview.webrtc.org/1750493002
Cr-Commit-Position: refs/heads/master@{#11884}
Reason for revert:
Revert breaks other uses, a fix will be rolled into Chromium instead.
Original issue's description:
> Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ )
>
> Reason for revert:
> Breaks Chromium.
>
> Original issue's description:
> > Remove ignored return code from modules.
> >
> > ModuleProcessImpl doesn't act on return codes and having them around is
> > confusing (it's unclear what an error return code here would do even).
> >
> > BUG=
> > R=tommi@webrtc.org
> >
> > Committed: f14c47a58c
>
> TBR=tommi@webrtc.org,pbos@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/da33a8a2a22f6d19ba2a8cce963beafbdbaa8fd8
> Cr-Commit-Position: refs/heads/master@{#11761}
TBR=tommi@webrtc.org,torbjorng@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1737013002
Cr-Commit-Position: refs/heads/master@{#11762}
Reason for revert:
Breaks Chromium.
Original issue's description:
> Remove ignored return code from modules.
>
> ModuleProcessImpl doesn't act on return codes and having them around is
> confusing (it's unclear what an error return code here would do even).
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: f14c47a58cTBR=tommi@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1736663004
Cr-Commit-Position: refs/heads/master@{#11761}
ModuleProcessImpl doesn't act on return codes and having them around is
confusing (it's unclear what an error return code here would do even).
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1703833002 .
Cr-Commit-Position: refs/heads/master@{#11747}
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}
This makes it clearer this code not meant to be used as an API.
I could not find any use of this in downstream code.
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_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
R=stefan@webrtc.orgTBR=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1440873005 .
Cr-Commit-Position: refs/heads/master@{#10699}
The main purpose was the interface-> include rename, but other files
were also moved, eliminating the "main" dir.
To avoid breaking downstream, the "interface" directories were copied
into a new "video_coding/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
Other files also moved:
video_coding/main/source -> video_coding
video_coding/main/test -> video_coding/test
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
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417283007 .
Cr-Commit-Position: refs/heads/master@{#10694}
