Reason for revert:
broke browser_tests
Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/4c7f4cd2ef76821edca6d773d733a924b0bedd25
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Original-Commit-Position: refs/heads/master@{#13613}
> Cr-Commit-Position: refs/heads/master@{#13615}
TBR=pbos@webrtc.org,mflodman@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.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/2203233002
Cr-Commit-Position: refs/heads/master@{#13616}
Reason for revert:
broke internal tests
Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Commit-Position: refs/heads/master@{#13613}
TBR=pbos@webrtc.org,mflodman@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.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/2206743002
Cr-Commit-Position: refs/heads/master@{#13614}
- Removes unnecessary casts to compute timedelta.
- Renames ProbingState for clarity. This should help when we probe mid-call.
- Enables probing by default to avoid checking on each incoming packet.
- Removes duplicate probing state tracking in paced sender. These duplicate
states were conflicting at times.
- Removes passing through packets for bug 5307 which seems long fixed.
- Cleanup handling of time_last_send_ms and avoid side effects of changing
probing state at TimeUntilNextProbe().
- Clear cluster data when probing is restarted to avoid having old data after a reset.
BUG=5859
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2182603002 .
Cr-Commit-Position: refs/heads/master@{#13612}
A reset of Bwe and rates on network route change triggers
switching of the RemoteBitrateEstimator from DelayBasedBwe
to RemoteBitrateEstimatorAbsSendTime. This is unexpected,
fix this.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2193043002 .
Cr-Commit-Position: refs/heads/master@{#13594}
The intended behavior of the pacer is that it should send audio (i.e high priority) packets immediatelly even if the pacer is paused and even if the available media budget has been used. The existing code will not send audio if the pacing budget has been used. It will normally send audio when the pacer is paused, but if the media budget was already used up when the pacer was paused, it would neither send audio nor update the budget.
Updated the implementatation and the unit test. Added logging for when the pacer is paused, and when it receives a zero bitrate estimate.
BUG=webrtc:6155
Review-Url: https://codereview.webrtc.org/2193183003
Cr-Commit-Position: refs/heads/master@{#13592}
The ProbingCalculator class calculates and validates the results from
probing attempts.
BUG=webrtc:5859
Review-Url: https://codereview.webrtc.org/2121183002
Cr-Commit-Position: refs/heads/master@{#13589}
double check rtp_sender in sending mode when altering sequence_number
adjust test to skip validating timestamp on rtx streams
fix test by waiting for all 3 media streams instead of 3 out 6 media and rtx streams.
BUG=webrtc:4332
Review-Url: https://codereview.webrtc.org/2177523002
Cr-Commit-Position: refs/heads/master@{#13587}
The plot is constructed by actually running the congestion controller with
the logged rtp headers and rtcp feedback messages to reproduce the same behavior
as in the real call.
R=phoglund@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2193763002 .
Cr-Commit-Position: refs/heads/master@{#13574}
The new version is much shorter than the old one, and hopefully easier
to read. This is part of the effort to rewrite the old mixer.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2132563002
Cr-Commit-Position: refs/heads/master@{#13570}
Reason for revert:
Multiple definitions of webrtc::MockMixerParticipant::MockMixerParticipant() during linking of modules_unittests. Please investigate and resubmit.
Original issue's description:
> Rewrote UpdateToMix in the audio mixer.
>
> The new version is much shorter than the old one, and hopefully easier
> to read. This is part of the effort to rewrite the old mixer.
>
> Committed: https://crrev.com/2942e240f4a985752714dac18c141064c97696d4
> Cr-Commit-Position: refs/heads/master@{#13568}
TBR=ossu@webrtc.org,ivoc@webrtc.org,aleloi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2195633002
Cr-Commit-Position: refs/heads/master@{#13569}
The new version is much shorter than the old one, and hopefully easier
to read. This is part of the effort to rewrite the old mixer.
Review-Url: https://codereview.webrtc.org/2132563002
Cr-Commit-Position: refs/heads/master@{#13568}
- Renamed variables and some function to comply with style guide.
- Removed default argument values.
- Removed some dead code.
- Cleaned up comments formatting in rtp_rtcp.h
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2067673004 .
Cr-Commit-Position: refs/heads/master@{#13565}
Due to a recent interface change for svc_params in vp9 svc, which
allows speed setting per layer, svc_params should be inited to 0
for safety.
Review-Url: https://codereview.webrtc.org/2179753003
Cr-Commit-Position: refs/heads/master@{#13561}
It was generating a random ID using the test case's "this" pointer
and the current time. However, the current time may be imprecise. And
the "this" pointer may have repeatable values.
BUG=webrtc:5898
Review-Url: https://codereview.webrtc.org/2190533004
Cr-Commit-Position: refs/heads/master@{#13560}
Reason for revert:
Breaks downstream targets.
Original issue's description:
> Add BWE plot to event log analyzer.
>
> The plot is constructed by actually running the congestion controller with
> the logged rtp headers and rtcp feedback messages to reproduce the same behavior
> as in the real call.
>
> R=phoglund@webrtc.org, terelius@webrtc.org
>
> Committed: https://crrev.com/2beea2a8c920000ef19eea20cce397507fc3d5e7
> Cr-Commit-Position: refs/heads/master@{#13558}
TBR=phoglund@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2190013002
Cr-Commit-Position: refs/heads/master@{#13559}
The plot is constructed by actually running the congestion controller with
the logged rtp headers and rtcp feedback messages to reproduce the same behavior
as in the real call.
R=phoglund@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2188033004 .
Cr-Commit-Position: refs/heads/master@{#13558}
This is an issue if the sequence numbers are to be used to compute packet loss statistics since it introduces gaps which are not related to loss.
Also making sure that the header extensions are properly guarded by the send crit sect.
Review-Url: https://codereview.webrtc.org/2190913002
Cr-Commit-Position: refs/heads/master@{#13557}
The goal of this change is to log the volume level for the
current audio stream so we can keep track of what volume the
user selects during a call.
BUG=b/30376577
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/2182043005 .
Cr-Commit-Position: refs/heads/master@{#13555}
Memory frames are now expected to be owned by the mixing participants.
Review-Url: https://codereview.webrtc.org/2127763002
Cr-Commit-Position: refs/heads/master@{#13554}
OutputMixer and AudioConferenceMixer communicated via a callback. OutputMixer implemented an AudioMixerOutputReceiver interface, which defines the callback function NewMixedAudio. This has been removed and replaced by a simple function in the new mixer. The audio frame with mixed audio is now copied one time less. I have also removed one forward declaration.
Review-Url: https://codereview.webrtc.org/2111293003
Cr-Commit-Position: refs/heads/master@{#13550}
Updated the sources in audio_processing:audioproc_test_utils to match the configuration on
"webrtc/modules/audio_processing/audio_processing_tests.gypi"
Removed audio_buffer_tools from modules_unittests to match the gyp file.
BUG=webrtc:6041
Review-Url: https://codereview.webrtc.org/2178963002
Cr-Commit-Position: refs/heads/master@{#13541}
These tests will be reenabled and fixed after Opus 1.1.3 has landed in
Chromium and is rolled into WebRTC.
BUG=
Review-Url: https://codereview.webrtc.org/2185673002
Cr-Commit-Position: refs/heads/master@{#13534}
maximum allowed sized raised from limited by physical udp packet size to
limited by theoritical maximum rtcp packet size.
BUG=webrtc:5260
R=åsapersson
Review-Url: https://codereview.webrtc.org/1998633002
Cr-Commit-Position: refs/heads/master@{#13532}
The iOS H264 video toolbox encoder is currently undershooting the
intended bitrate. This seems to be caused by the data rate limit
property. This CL increases the data rate limit to a set
percentage above the intended bitrate to avoid undershooting. The
AverageBitRate property is still set to the intended bitrate, which
keeps it from overshooting the intended bitrate.
BUG=b/28713684
Review-Url: https://codereview.webrtc.org/2177873003
Cr-Commit-Position: refs/heads/master@{#13526}
