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}
Now it check if rtp timestamp can be calculating instead of checking number of rtp packets. This way it works for reconfigured streams too.
It also moved deeper into rtcp_sender class to prevent SR no matter the reason it need to be genereated. This way it prevents creating compound rtcp packets that have to start with Sender Report and Sender Reports as response to (mostly theoretical) sr-request rtcp packet.
BUG=webrtc:1600
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1639253007 .
Cr-Commit-Position: refs/heads/master@{#13503}
Retransmissions are supposed to be sent before normal packets by the pacer, but the current implementation will only use it if the second packet is a retransmission and the first packet is not. It misses the case where the first packet is retransmission and the second packet is not.
This CL fixes the comparator and adds a unit test.
Also changed the SendAndExpectPacket function to propagate the retransmission flag to the expectations. Previously, all packets were expected to be normal packets.
BUG=webrtc:6124
Review-Url: https://codereview.webrtc.org/2156063004
Cr-Commit-Position: refs/heads/master@{#13502}
The added unittest triggers this CHECK:
433ed06800/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc (146)
It happens because the unwrap of the sequence number fails if the unwrappers last sequence number is small, but the newly added sequence number is large (greater than last seq num + 2^15), and therefore should have been interpreted as a reordering and a backwards wrap. Since that would mean the sequence number returned from the unwrapper would be negative, it simply returns the original sequence number instead. This causes problems later where the wrap is correctly handled, and everything breaks.
The real solution should be to correctly handle wraps, but to prevent the crash this is a reasonable workaround for now.
BUG=
Review-Url: https://codereview.webrtc.org/2157843002
Cr-Commit-Position: refs/heads/master@{#13496}
After https://codereview.webrtc.org/1827263002, audio devices are no
longer (ever) initialized if they return true from
RecordingIsInitialized. Since this was left as "return true;" for
file_audio_device, the recording buffer was never set up correctly, and
the audio buffer would assert when called (in debug) and FileAudioDevice
would cause memory corruption (in release).
BUG=
Review-Url: https://codereview.webrtc.org/2116003003
Cr-Commit-Position: refs/heads/master@{#13489}
In order to correctly determine the references of a frame when using Vp9
with GOF one has to wait for all frames on the lower temporal layers
to make sure no up-switch point is missed.
This patch fix a bug where upon receiving a frame the RtpFrameReferenceFinder
would try to add missing frame for a group with a not yet knows scalability
structure.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2127073002
Cr-Commit-Position: refs/heads/master@{#13487}
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}
The decode thread should be stopped before triggering shutdown of the
video receiver, so that the decoder doesn't try to insert a new frame
while the jitter buffer is being shut down.
BUG=webrtc:6102
Review-Url: https://codereview.webrtc.org/2146883002
Cr-Commit-Position: refs/heads/master@{#13467}
Reason for revert:
For some reason, payload_type_mapper.cc is not being picked up in Chrome builds, leading to undefined references. Reverting while investigating.
Original issue's description:
> WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs.
>
> Changed WebRtcVoiceEngine to present receive codecs from the formats
> provided by its decoder factory. Added supported formats to
> BuiltinAudioDecoderFactory. Added helper functions for creating some
> simple decoder factories for mocking.
>
> Created a PayloadTypeMapper for assigning payload types to formats. I
> think we'll eventually want to use this further up, or possibly in
> both the audio and video sides. It would be best if the engines didn't
> have to talk payload types at all, but it might be more difficult to
> get around when payload types depend on each-other, like the RTX
> codecs for video.
>
> This CL also includes some changes to rtc::Optional. I've put them in
> a separate CL that should (or should not) land first, making these
> changes void.
> See: https://codereview.webrtc.org/2072713002/
>
> BUG=webrtc:5805
>
> Committed: https://crrev.com/95eb1ba0db79d8fd134ae61b0a24648598684e8a
> Cr-Commit-Position: refs/heads/master@{#13459}
TBR=ivoc@webrtc.org,tina.legrand@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2151453002
Cr-Commit-Position: refs/heads/master@{#13460}
Changed WebRtcVoiceEngine to present receive codecs from the formats
provided by its decoder factory. Added supported formats to
BuiltinAudioDecoderFactory. Added helper functions for creating some
simple decoder factories for mocking.
Created a PayloadTypeMapper for assigning payload types to formats. I
think we'll eventually want to use this further up, or possibly in
both the audio and video sides. It would be best if the engines didn't
have to talk payload types at all, but it might be more difficult to
get around when payload types depend on each-other, like the RTX
codecs for video.
This CL also includes some changes to rtc::Optional. I've put them in
a separate CL that should (or should not) land first, making these
changes void.
See: https://codereview.webrtc.org/2072713002/
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2072753002
Cr-Commit-Position: refs/heads/master@{#13459}
Reason for revert:
My original patch broke things that are now fixed by https://codereview.webrtc.org/2141193002/.
Hence I am relanding my original change.
Original issue's description:
> Revert of Adds data logging in native AudioDeviceBuffer class (patchset #10 id:180001 of https://codereview.webrtc.org/2132613002/ )
>
> Reason for revert:
> Seems to break things upstream.
>
> Original issue's description:
> > Adds data logging in native AudioDeviceBuffer class.
> >
> > Goal is to provide periodic logging of most essential audio parameters
> > for playout and recording sides. It will allow us to track if the native audio layer is working as intended.
> >
> > BUG=NONE
> >
> > Committed: https://crrev.com/348e411dd27e6dbe9b84b27ce46e9b7c657c1eae
> > Cr-Commit-Position: refs/heads/master@{#13440}
>
> TBR=stefan@webrtc.org,henrika@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=NONE
>
> Committed: https://crrev.com/025aa94ccb85e4c6fe20a3fecdac5d27ec9ba3da
> Cr-Commit-Position: refs/heads/master@{#13441}
TBR=stefan@webrtc.org,sprang@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/2138403003
Cr-Commit-Position: refs/heads/master@{#13455}
Reason for revert:
Seems to break things upstream.
Original issue's description:
> Adds data logging in native AudioDeviceBuffer class.
>
> Goal is to provide periodic logging of most essential audio parameters
> for playout and recording sides. It will allow us to track if the native audio layer is working as intended.
>
> BUG=NONE
>
> Committed: https://crrev.com/348e411dd27e6dbe9b84b27ce46e9b7c657c1eae
> Cr-Commit-Position: refs/heads/master@{#13440}
TBR=stefan@webrtc.org,henrika@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/2139233002
Cr-Commit-Position: refs/heads/master@{#13441}
