Rationale:
1. sinks_ is not properly differentiated from rsid_sinks_.
2. Consistency with RtcpDemuxer.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2958283002
Cr-Commit-Position: refs/heads/master@{#18813}
Patch set 1 is a reland + trivial rebase.
Patch set >= 2 contains bug fixes.
> Original issue's description:
> > Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
> >
> > That however exposes a bunch of failed test, so this CL also fixed a few other things:
> > * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> > * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> > * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> > * Fix test
> >
> > BUG=7664
> >
> > Review-Url: https://codereview.webrtc.org/2883963002
> > Cr-Commit-Position: refs/heads/master@{#18473}
> > Committed: 6431e21da6
BUG=webrtc:7664
Review-Url: https://codereview.webrtc.org/2953053002
Cr-Commit-Position: refs/heads/master@{#18782}
Reason for revert:
About to fix problem and reland.
Original issue's description:
> Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ )
>
> Reason for revert:
> Breaks Chromium FYI bots.
>
> The problem is in the BUILD.gn file.
>
> Sample failure:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829
>
> Sample logs:
> use_goma = true
> """ to /b/c/b/Linux_Builder/src/out/Release/args.gn.
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
> -> returned 1
> ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file.
> "//webrtc/base:rtc_base_approved",
> ^--------------------------------
>
> Original issue's description:
> > Create RtcpDemuxer. Capabilities:
> > 1. Demux RTCP messages according to the sender-SSRC.
> > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP).
> > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks").
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/2943693003
> > Cr-Commit-Position: refs/heads/master@{#18763}
> > Committed: cb83bdf01f
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2957763002
> Cr-Commit-Position: refs/heads/master@{#18764}
> Committed: 0e7e7869e7
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2960623002
Cr-Commit-Position: refs/heads/master@{#18768}
Reason for revert:
Breaks Chromium FYI bots.
The problem is in the BUILD.gn file.
Sample failure:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829
Sample logs:
use_goma = true
""" to /b/c/b/Linux_Builder/src/out/Release/args.gn.
/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
-> returned 1
ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file.
"//webrtc/base:rtc_base_approved",
^--------------------------------
Original issue's description:
> Create RtcpDemuxer. Capabilities:
> 1. Demux RTCP messages according to the sender-SSRC.
> 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP).
> 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks").
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2943693003
> Cr-Commit-Position: refs/heads/master@{#18763}
> Committed: cb83bdf01fTBR=stefan@webrtc.org,danilchap@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org,holmer@google.com,eladalon@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/2957763002
Cr-Commit-Position: refs/heads/master@{#18764}
1. Demux RTCP messages according to the sender-SSRC.
2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP).
3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks").
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2943693003
Cr-Commit-Position: refs/heads/master@{#18763}
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}
WebRtcVideoChannel and and WebRtcVideoEngine seem to have been removed, and only WebRtcVideoChannel2 and WebRtcVideoEngine2 remain, which removes the need for the "2" postfix.
BUG=None
Review-Url: https://codereview.webrtc.org/2932073002
Cr-Commit-Position: refs/heads/master@{#18531}
Make RtpDemuxer able to demux RTP packets according to RSID (RTP Stream ID), as well as the (pre-existing) ability to demux according to SSRC.
BUG=None
Review-Url: https://codereview.webrtc.org/2920993002
Cr-Commit-Position: refs/heads/master@{#18495}
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}
1. Create unit tests for RtpDemuxer.
2. Add an RTC_DCHECK in RtpDemuxer that makes sure that the sink<->ssrc multimap does not allow multiple instances of the same association.
BUG=None
Review-Url: https://codereview.webrtc.org/2902823004
Cr-Commit-Position: refs/heads/master@{#18357}
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}
probing_interval as a name is used for the period that BWE attempt to increase its estimate. The name is confusing since it is not related to "probing" which is a special mechanism for estimating BWE.
BUG=None
Review-Url: https://codereview.webrtc.org/2888893002
Cr-Commit-Position: refs/heads/master@{#18203}
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}
Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2799033006
Cr-Commit-Position: refs/heads/master@{#17977}
These are the changes made to WebRtcVoiceEngine and surrounding
code. It still contains some things that are inelegant, like how
AudioCodecSpec and AudioFormatInfo is ferried around in
SendCodecSpec. This should probably be resolved before landing.
There are also a few test still that are disabled. They should be
removed or fixed, as the case may be.
I've put this CL up to get a better overview of the changes made and
how reviewable they are.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2705093002
Cr-Commit-Position: refs/heads/master@{#17904}
This target keeps track of .h the files under webrtc/ that are not part
of any target.
If a .h file is not part of a target the 'gn check' utility is not
able to spot if a target is missing a dependency because even if
it parses '#include' directives it is not able to find a target that
contains these headers.
BUG=webrtc:7512
NOTRY=True
Review-Url: https://codereview.webrtc.org/2841873002
Cr-Commit-Position: refs/heads/master@{#17874}
Make all rtc_source_test target that contains tests that
are included in a test executable only be visible to the
rtc_test target. Doing this exposed a couple of errors and
dependency problems that were resolved. Having this could
have prevented duplicated execution of tests like the case that
was recently fixed by deadbeef@ in
https://codereview.webrtc.org/2820263004
New targets:
* //webrtc/modules/rtp_rtcp:fec_test_helper
* //webrtc/modules/rtp_rtcp:mock_rtp_rtcp
* //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer
The mock files and targets should probably be moved into webrtc/test in
the future, but that's out of the scope of this CL.
BUG=webrtc:5716
NOTRY=True
Review-Url: https://codereview.webrtc.org/2828793003
Cr-Commit-Position: refs/heads/master@{#17863}
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}
Reason for revert:
Re-land, reverting did not fix bug.
https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
Original issue's description:
> Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
>
> Reason for revert:
> Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
>
> https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
>
> Original issue's description:
> > Added the GetSources() to the RtpReceiverInterface and implemented
> > it for the AudioRtpReceiver.
> >
> > This method returns a vector of RtpSource(both CSRC source and SSRC
> > source) which contains the ID of a source, the timestamp, the source
> > type (SSRC or CSRC) and the audio level.
> >
> > The RtpSource objects are buffered and maintained by the
> > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> > the info of the contributing source will be pulled along the object
> > chain:
> > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> > AudioReceiveStream -> voe::Channel -> RtpRtcp module
> >
> > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
> >
> > BUG=chromium:703122
> > TBR=stefan@webrtc.org, danilchap@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2770233003
> > Cr-Commit-Position: refs/heads/master@{#17591}
> > Committed: 292084c376
>
> TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:703122
>
> Review-Url: https://codereview.webrtc.org/2809613002
> Cr-Commit-Position: refs/heads/master@{#17616}
> Committed: fbcc5cb386TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:703122
Review-Url: https://codereview.webrtc.org/2810623003
Cr-Commit-Position: refs/heads/master@{#17621}
AudioDecoder and AudioDecoderFactory are in webrtc/api/ now, so move
their mocks to someplace central where tests from all over WebRTC are
allowed to #include them.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2798063004
Cr-Commit-Position: refs/heads/master@{#17619}
Reason for revert:
Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
Original issue's description:
> Added the GetSources() to the RtpReceiverInterface and implemented
> it for the AudioRtpReceiver.
>
> This method returns a vector of RtpSource(both CSRC source and SSRC
> source) which contains the ID of a source, the timestamp, the source
> type (SSRC or CSRC) and the audio level.
>
> The RtpSource objects are buffered and maintained by the
> RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> the info of the contributing source will be pulled along the object
> chain:
> AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> AudioReceiveStream -> voe::Channel -> RtpRtcp module
>
> Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
>
> BUG=chromium:703122
> TBR=stefan@webrtc.org, danilchap@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2770233003
> Cr-Commit-Position: refs/heads/master@{#17591}
> Committed: 292084c376TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:703122
Review-Url: https://codereview.webrtc.org/2809613002
Cr-Commit-Position: refs/heads/master@{#17616}
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}
This provides a better default for audio-only tests.
BUG=None
Review-Url: https://codereview.webrtc.org/2794193003
Cr-Commit-Position: refs/heads/master@{#17536}
Reason for revert:
Seem to be a flaky test rather than an issue with this cl. Creating reland, will add code to reduce flakiness to that test.
Original issue's description:
> Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
>
> Reason for revert:
> This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780
>
> Original issue's description:
> > Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ )
> >
> > Reason for revert:
> > Found issue with test case, will add fix to reland cl.
> >
> > Original issue's description:
> > > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
> > >
> > > Reason for revert:
> > > Breaks perf tests:
> > > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679
> > > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325
> > >
> > > Original issue's description:
> > > > Add framerate to VideoSinkWants and ability to signal on overuse
> > > >
> > > > In ViEEncoder, try to reduce framerate instead of resolution if the
> > > > current degradation preference is maintain-resolution rather than
> > > > balanced.
> > > >
> > > > BUG=webrtc:4172
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2716643002
> > > > Cr-Commit-Position: refs/heads/master@{#17327}
> > > > Committed: 72acf25261
> > >
> > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,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=webrtc:4172
> > >
> > > Review-Url: https://codereview.webrtc.org/2764133002
> > > Cr-Commit-Position: refs/heads/master@{#17331}
> > > Committed: 8b45b11144
> >
> > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:4172
> >
> > Review-Url: https://codereview.webrtc.org/2781433002
> > Cr-Commit-Position: refs/heads/master@{#17474}
> > Committed: 3ea3c77e93
>
> TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4172
>
> Review-Url: https://codereview.webrtc.org/2783183003
> Cr-Commit-Position: refs/heads/master@{#17477}
> Committed: f9ed235c9bR=ilnik@webrtc.org,stefan@webrtc.org
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2789823002
Cr-Commit-Position: refs/heads/master@{#17498}
