We use Optional in our public API, so its header should be in
webrtc/api/.
BUG=webrtc:8205
Review-Url: https://codereview.webrtc.org/3011943002
Cr-Commit-Position: refs/heads/master@{#19693}
Packets recovered by ULPFEC enter through
RtpVideoStreamReceiver::OnRecoveredPacket, which does RTP
header extension parsing. Packets recovered by FlexFEC, however,
enter through Call::OnRecoveredPacket, which prior to this
CL did not do RTP header extension parsing.
The lack of RTP header extension parsing for FlexFEC packets is a
regression since https://codereview.webrtc.org/2886993005/.
TESTED=Using Android app with FlexFEC field trial enabled.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/3002023002
Cr-Commit-Position: refs/heads/master@{#19460}
In preparation of running DirectTransport on a TaskQueue in unit-tests, change the thread-checkers to sequence-checkers. This is necessary for Mac and iOS, where the TaskQueue is guaranteed to run sequentially, but not guaranteed to do so on only one thread.
TODO: Add the relevant BUGs.
BUG=None
Review-Url: https://codereview.webrtc.org/2999973002
Cr-Commit-Position: refs/heads/master@{#19404}
Reason for revert:
Reland
Original issue's description:
> Revert of Add functionality which limits the number of bytes on the network. (patchset #26 id:500001 of https://codereview.webrtc.org/2918323002/ )
>
> Reason for revert:
> Speculative revert to see if this caused regressions in android perf tests.
>
> Original issue's description:
> > Add functionality which limits the number of bytes on the network.
> >
> > The limit is based on the bandwidth delay product, but also adds some additional slack to compensate for the sawtooth-like BWE pattern and the slowness of the encoder rate control. The delay is estimated based on the time from sending a packet until an ack is received. Since acks are received in bursts (feedback is only sent periodically), a min filter is used to estimate the rtt.
> >
> > Whenever the in flight bytes reaches the congestion window, the pacer is paused, which in turn will result in send-side queues growing. Eventually the encoders will be paused as the pacer queue grows large (currently 2 seconds).
> >
> > BUG=webrtc:7926
> >
> > Review-Url: https://codereview.webrtc.org/2918323002
> > Cr-Commit-Position: refs/heads/master@{#19289}
> > Committed: 8497fdde43
>
> TBR=terelius@webrtc.org,philipel@webrtc.org,tschumim@webrtc.org,gnish@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7926
>
> Review-Url: https://codereview.webrtc.org/3001653002
> Cr-Commit-Position: refs/heads/master@{#19339}
> Committed: 64136af364TBR=terelius@webrtc.org,philipel@webrtc.org,tschumim@webrtc.org,gnish@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7926
Review-Url: https://codereview.webrtc.org/2994343002
Cr-Commit-Position: refs/heads/master@{#19373}
Reason for revert:
Speculative revert to see if this caused regressions in android perf tests.
Original issue's description:
> Add functionality which limits the number of bytes on the network.
>
> The limit is based on the bandwidth delay product, but also adds some additional slack to compensate for the sawtooth-like BWE pattern and the slowness of the encoder rate control. The delay is estimated based on the time from sending a packet until an ack is received. Since acks are received in bursts (feedback is only sent periodically), a min filter is used to estimate the rtt.
>
> Whenever the in flight bytes reaches the congestion window, the pacer is paused, which in turn will result in send-side queues growing. Eventually the encoders will be paused as the pacer queue grows large (currently 2 seconds).
>
> BUG=webrtc:7926
>
> Review-Url: https://codereview.webrtc.org/2918323002
> Cr-Commit-Position: refs/heads/master@{#19289}
> Committed: 8497fdde43TBR=terelius@webrtc.org,philipel@webrtc.org,tschumim@webrtc.org,gnish@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7926
Review-Url: https://codereview.webrtc.org/3001653002
Cr-Commit-Position: refs/heads/master@{#19339}
The limit is based on the bandwidth delay product, but also adds some additional slack to compensate for the sawtooth-like BWE pattern and the slowness of the encoder rate control. The delay is estimated based on the time from sending a packet until an ack is received. Since acks are received in bursts (feedback is only sent periodically), a min filter is used to estimate the rtt.
Whenever the in flight bytes reaches the congestion window, the pacer is paused, which in turn will result in send-side queues growing. Eventually the encoders will be paused as the pacer queue grows large (currently 2 seconds).
BUG=webrtc:7926
Review-Url: https://codereview.webrtc.org/2918323002
Cr-Commit-Position: refs/heads/master@{#19289}
[This CL is work in progress.]
Wire up the rtp keep-alive in webrtc::Call::Config using new
SetRtpTransportParameters() method on RtpTransportInterface.
BUG=webrtc:7907
Review-Url: https://codereview.webrtc.org/2981513002
Cr-Commit-Position: refs/heads/master@{#19287}
I don't think this line could never conceivably fail - if the ctor has reached that point, the object fit in memory, and its members have all been allocated legal memory addresses, none of which may be 0x00.
BUG=None
Review-Url: https://codereview.webrtc.org/2989813002
Cr-Commit-Position: refs/heads/master@{#19176}
through streams related to a call object.
The Call object does not know directly when packets pass through it, only which
AudioSendStreams are used. Each AudioSendStream has a pointer to the Transport
object through which its packets are send.
This CL:
By registering an internal wrapper class, TimedTransport, the AudioSendStream
can stay up-to-date on when packets have passed through its Transport. This
lifetime (as an interval) is then queried by the Call when the AudioSendStream
is destroyed. When Call is destroyed, all streams are guaranteed to have been
destroyed and hence Call is up-to-date on packet activity.
The class TimeInterval keeps the code in Call and AudioSendStream smaller, with
fewer get methods in their APIs and less code for updating values.
Also modifies the unit test for AudioSendStream: it previously enforced that
the stream registers (with its channel proxy) the same transport that it was
constructed with.
BUG=webrtc:7882
Review-Url: https://codereview.webrtc.org/2979833002
Cr-Commit-Position: refs/heads/master@{#19087}
Don't force requirement of media type for those packets.
BUG=webrtc:7964
Review-Url: https://codereview.webrtc.org/2973323002
Cr-Commit-Position: refs/heads/master@{#18966}
This makes more sense since logically it's a transport level feature,
not a media stream feature. Even if the implementation details forces it
to be an rtp stream detail, for the moment.
BUG=webrtc:7907
Review-Url: https://codereview.webrtc.org/2978503002
Cr-Commit-Position: refs/heads/master@{#18963}
Original issue:
https://codereview.webrtc.org/2957073002/
Reason for reland:
Failed Android unit tests and failed Windows compile.
The tests seemed related at the time, but not after more consideration.
Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.
BUG=webrtc:7882
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2970793003
Cr-Commit-Position: refs/heads/master@{#18886}
Reason for revert:
The following, seemingly related, unit tests crash on Android32 (M Nexus5X).
org.webrtc.PeerConnectionTest#testCompleteSession
org.webrtc.PeerConnectionTest#testDataChannelOnlySession
A Windows build fails with a mysterious compile error.
Original issue's description:
> Add received audio/video call duration metrics based on packets.
>
> Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.
>
> BUG=webrtc:7882
>
> Review-Url: https://codereview.webrtc.org/2957073002
> Cr-Commit-Position: refs/heads/master@{#18881}
> Committed: 746749237aTBR=stefan@webrtc.org,aleloi@webrtc.org,asapersson@webrtc.org,holmer@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7882
Review-Url: https://codereview.webrtc.org/2972613002
Cr-Commit-Position: refs/heads/master@{#18882}
Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.
BUG=webrtc:7882
Review-Url: https://codereview.webrtc.org/2957073002
Cr-Commit-Position: refs/heads/master@{#18881}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
And implementation class RtpStreamReceiverController.
It's responsible for demuxing, and acts as factory for
RtpStreamReceiverInterface.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2886993005
Cr-Commit-Position: refs/heads/master@{#18696}
This CL makes the WebRTC more modular and allows the users to build
WebRTC without audio and video(DataChannel only).
The BUILD files in call/, logging/, media/ and pc/ are modified to
support modular WebRTC.
The dependencies on Call and RtcEventLog are removed from the
PeerConnection. Instead of being created internally, they would be
passed in by the PeerConnectionFactory.
Add the CreateModularPeerConnectionFactory function which allow the
users to create a PeerConnectionFactory with the modules they need.
If the users want to build WebRTC without audio and video, they can
pass in null pointers for modules they don't need. (MediaEngine,
VideoEncoderFactory etc.)
BUG=webrtc:7613
Review-Url: https://codereview.webrtc.org/2854123003
Cr-Commit-Position: refs/heads/master@{#18617}
Reason for revert:
Broken downstream project.
Original issue's description:
> Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
>
> BUG=webrtc:7395
>
> Review-Url: https://codereview.webrtc.org/2888303005
> Cr-Commit-Position: refs/heads/master@{#18417}
> Committed: 9641c13327TBR=deadbeef@webrtc.org,stefan@webrtc.org,kwiberg@webrtc.org,solenberg@webrtc.org,holmer@google.com,zstein@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7395
Review-Url: https://codereview.webrtc.org/2914413002
Cr-Commit-Position: refs/heads/master@{#18420}
Biggest change is to Remove MediaType as argument to RtcEventLog::LogRtpHeader and RtcEventLog::LogRtcpHeader.
Since the type is used by tools, these tools are rewritten to figure out the media type from the configurations instead.
BUG=webrtc:7538
TBR=solenberg@webrtc.org // For call.cc and voiceengine.cc
Review-Url: https://codereview.webrtc.org/2855143002
Cr-Commit-Position: refs/heads/master@{#18324}
This functionality already exists for video streams, so not having it
for audio is unexpected and has lead to problems.
BUG=webrtc:7631
Review-Url: https://codereview.webrtc.org/2887733002
Cr-Commit-Position: refs/heads/master@{#18231}
To make the distinction for stats, add a |recovered| flag to
RtpPacketReceived.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2693123002
Cr-Commit-Position: refs/heads/master@{#18103}
FakeRtpTransportController moves to a common header and its constructor is changed to take a SendSideCongestionController to enable injecting the mock.
BUG=webrtc:7395
Review-Url: https://codereview.webrtc.org/2834663003
Cr-Commit-Position: refs/heads/master@{#18055}
Also move RtpTransportControllerSendInterface to its own header file.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2808043002
Cr-Commit-Position: refs/heads/master@{#17840}
Instead of using the time on the first callback to Call::OnSentPacket, use the time when the first packet is sent from the pacer (to make sure this packet corresponds to an audio/video RTP packet).
BUG=webrtc:6244
Review-Url: https://codereview.webrtc.org/2825333002
Cr-Commit-Position: refs/heads/master@{#17777}
These are invalid since:
- An allocated bitrate of 0 means that the stream should be disabled. Changing the behavior to send audio at min bitrate even though the allocator asks for the stream to be disabled.
- It should be OK to set a min bitrate equal to the start bitrate.
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2806163003
Cr-Commit-Position: refs/heads/master@{#17613}
Deletes left-over includes of trace.h and critical_section_wrapper.h.
BUG=webrtc:7035
Review-Url: https://codereview.webrtc.org/2784873002
Cr-Commit-Position: refs/heads/master@{#17460}
Reason for revert:
Intend to fix perf failures and reland.
Original issue's description:
> Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
>
> 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: 9c47b00e24
>
> TBR=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}
> Committed: 3a3bd50610TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@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/2783853002
Cr-Commit-Position: refs/heads/master@{#17459}
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}
