Commit Graph

984 Commits

Author SHA1 Message Date
ef8d773d26 Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
BUG=webrtc:7433

Review-Url: https://codereview.webrtc.org/2805023002
Cr-Commit-Position: refs/heads/master@{#17759}
2017-04-19 09:59:48 +00:00
ba6aa90c04 Ensure SetREMBStatus(false) disables sending REMB
support CL for upcoming https://codereview.webrtc.org/2789843002/

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2813693003
Cr-Commit-Position: refs/heads/master@{#17741}
2017-04-18 13:57:02 +00:00
0426222f4c Modified the rtp_receiver_unittests.
Implemented operator == in RtpSource and use the gmock EXPECT_THAT to make the test cleaner.

Related CL: https://codereview.webrtc.org/2770233003/

BUG=chromium:703122

Review-Url: https://codereview.webrtc.org/2813753002
Cr-Commit-Position: refs/heads/master@{#17659}
2017-04-11 18:28:10 +00:00
00d802b6ee Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
Reason for revert:
Fix failing bots.

BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2816493002
Cr-Commit-Position: refs/heads/master@{#17658}
2017-04-11 17:34:31 +00:00
27c46e2872 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ )
Reason for revert:
Breaks android buildbots.

Original issue's description:
> Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
>
> Reason for revert:
> Reland with appropriate changes to API to not break depending projects.
>
> Original issue's description:
> > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
> >
> > Reason for revert:
> > Breaks dependent projects.
> >
> > Original issue's description:
> > > Add content type information to Encoded Images and add corresponding RTP extension header.
> > > Use it to separate UMA e2e delay metric between screenshare from video.
> > > Content type extension is set based on encoder settings and processed and decoders.
> > >
> > > Also,
> > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> > >
> > > BUG=webrtc:7420
> > >
> > > Review-Url: https://codereview.webrtc.org/2772033002
> > > Cr-Commit-Position: refs/heads/master@{#17640}
> > > Committed: 64e739aeae
> >
> > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2816463002
> > Cr-Commit-Position: refs/heads/master@{#17644}
> > Committed: 5721866808
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2812913002
> Cr-Commit-Position: refs/heads/master@{#17651}
> Committed: 774f6b4b96

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2809653004
Cr-Commit-Position: refs/heads/master@{#17653}
2017-04-11 13:20:05 +00:00
774f6b4b96 Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
Reason for revert:
Reland with appropriate changes to API to not break depending projects.

Original issue's description:
> Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
>
> Reason for revert:
> Breaks dependent projects.
>
> Original issue's description:
> > Add content type information to Encoded Images and add corresponding RTP extension header.
> > Use it to separate UMA e2e delay metric between screenshare from video.
> > Content type extension is set based on encoder settings and processed and decoders.
> >
> > Also,
> > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> >
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2772033002
> > Cr-Commit-Position: refs/heads/master@{#17640}
> > Committed: 64e739aeae
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2816463002
> Cr-Commit-Position: refs/heads/master@{#17644}
> Committed: 5721866808

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2812913002
Cr-Commit-Position: refs/heads/master@{#17651}
2017-04-11 13:12:37 +00:00
29dbb1992a Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ )
Reason for revert:
Relanded by mistake.

Original issue's description:
> Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
>
> Reason for revert:
> Reland with fixes which break API
>
> Original issue's description:
> > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
> >
> > Reason for revert:
> > Breaks dependent projects.
> >
> > Original issue's description:
> > > Add content type information to Encoded Images and add corresponding RTP extension header.
> > > Use it to separate UMA e2e delay metric between screenshare from video.
> > > Content type extension is set based on encoder settings and processed and decoders.
> > >
> > > Also,
> > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> > >
> > > BUG=webrtc:7420
> > >
> > > Review-Url: https://codereview.webrtc.org/2772033002
> > > Cr-Commit-Position: refs/heads/master@{#17640}
> > > Committed: 64e739aeae
> >
> > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2816463002
> > Cr-Commit-Position: refs/heads/master@{#17644}
> > Committed: 5721866808
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2811963002
> Cr-Commit-Position: refs/heads/master@{#17645}
> Committed: 4fa0c4f97f

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2810923004
Cr-Commit-Position: refs/heads/master@{#17648}
2017-04-11 11:49:07 +00:00
4fa0c4f97f Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
Reason for revert:
Reland with fixes which break API

Original issue's description:
> Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
>
> Reason for revert:
> Breaks dependent projects.
>
> Original issue's description:
> > Add content type information to Encoded Images and add corresponding RTP extension header.
> > Use it to separate UMA e2e delay metric between screenshare from video.
> > Content type extension is set based on encoder settings and processed and decoders.
> >
> > Also,
> > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> >
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2772033002
> > Cr-Commit-Position: refs/heads/master@{#17640}
> > Committed: 64e739aeae
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2816463002
> Cr-Commit-Position: refs/heads/master@{#17644}
> Committed: 5721866808

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2811963002
Cr-Commit-Position: refs/heads/master@{#17645}
2017-04-11 11:01:43 +00:00
5721866808 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
Reason for revert:
Breaks dependent projects.

Original issue's description:
> Add content type information to Encoded Images and add corresponding RTP extension header.
> Use it to separate UMA e2e delay metric between screenshare from video.
> Content type extension is set based on encoder settings and processed and decoders.
>
> Also,
> Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
>
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2772033002
> Cr-Commit-Position: refs/heads/master@{#17640}
> Committed: 64e739aeae

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2816463002
Cr-Commit-Position: refs/heads/master@{#17644}
2017-04-11 10:59:43 +00:00
64e739aeae Add content type information to Encoded Images and add corresponding RTP extension header.
Use it to separate UMA e2e delay metric between screenshare from video.
Content type extension is set based on encoder settings and processed and decoders.

Also,
Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.

BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2772033002
Cr-Commit-Position: refs/heads/master@{#17640}
2017-04-11 08:46:04 +00:00
ca31f175e1 Remove deprecated RTPPayloadStrategy
Remove deprecated set_use_rtx_payload_mapping_on_restore()
Remove unused headers

BUG=None

Review-Url: https://codereview.webrtc.org/2808743002
Cr-Commit-Position: refs/heads/master@{#17623}
2017-04-10 15:45:29 +00:00
8d609f6b6d Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
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: fbcc5cb386

TBR=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}
2017-04-10 14:39:05 +00:00
fbcc5cb386 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}
2017-04-10 11:38:13 +00:00
c547e84ec5 Allow rtp::Packet::*RawExtension to take 0 as an extension id
BUG=webrtc:7433

Review-Url: https://codereview.webrtc.org/2803623004
Cr-Commit-Position: refs/heads/master@{#17610}
2017-04-10 08:31:49 +00:00
292084c376 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}
2017-04-07 17:57:22 +00:00
978504e25c Move rtp header extension length check from Packet::FindExtension to ExtensionT::Parse
to allow to read variable-length extensions.

BUG=webrtc:7433

Review-Url: https://codereview.webrtc.org/2801733002
Cr-Commit-Position: refs/heads/master@{#17554}
2017-04-06 08:03:53 +00:00
fab482b326 Simplify RTPSender::RegisterRtpHeaderExtension
DCHECK instead of run-time check extension type is correct.

BUG=webrtc:1994

Review-Url: https://codereview.webrtc.org/2795723002
Cr-Commit-Position: refs/heads/master@{#17524}
2017-04-04 09:33:48 +00:00
653063f6fd Add functions to get/set rtp header extension by id.
This helps with dynamic size extensions.

BUG=webrtc:7433

Review-Url: https://codereview.webrtc.org/2789773004
Cr-Commit-Position: refs/heads/master@{#17505}
2017-04-03 13:16:30 +00:00
3a407eee1c Making RtpSender tests cover BWE with overhead.
BUG=webrtc:7418, webrtc:6315

Review-Url: https://codereview.webrtc.org/2783743003
Cr-Commit-Position: refs/heads/master@{#17499}
2017-04-03 08:10:33 +00:00
0ffdcc51bc Delete unneeded includes of deprecated system_wrappers include files.
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}
2017-03-30 07:31:15 +00:00
07a01b3357 Allow RtpPacket::SetPayloadSize to increase payload size
Make SetPayloadSize return buffer to write to so that it can replace
AllocatePayload function.

BUG=None

Review-Url: https://codereview.webrtc.org/2785713002
Cr-Commit-Position: refs/heads/master@{#17450}
2017-03-29 14:33:13 +00:00
0f0a849cba H264 Supplemental Enhancement Information no longer considered a keyframe.
BUG=none

Review-Url: https://codereview.webrtc.org/2769863007
Cr-Commit-Position: refs/heads/master@{#17435}
2017-03-29 09:19:54 +00:00
1c07c70d88 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType"
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2774833003
Cr-Commit-Position: refs/heads/master@{#17391}
2017-03-27 14:15:49 +00:00
670a7f3611 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ )
Reason for revert:
Makes perf and Chromium FYI bots unhappy.

Original issue's description:
> WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
>
> This removes one more place where we were unable to handle codecs not
> in the built-in set.
>
> BUG=webrtc:5805
>
> Review-Url: https://codereview.webrtc.org/2686043006
> Cr-Commit-Position: refs/heads/master@{#17370}
> Committed: 1724cfbdba

TBR=ossu@webrtc.org,solenberg@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/2772043002
Cr-Commit-Position: refs/heads/master@{#17374}
2017-03-24 12:56:21 +00:00
60c5668f1e Fix cpplint errors in locations that are already being checked
Otherwise cpplint will trigger during presubmit for unrelated changes
in these files.

BUG=webrtc:5149
NOTRY=True

Review-Url: https://codereview.webrtc.org/2767393003
Cr-Commit-Position: refs/heads/master@{#17371}
2017-03-24 10:22:49 +00:00
1724cfbdba WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
This removes one more place where we were unable to handle codecs not
in the built-in set.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2686043006
Cr-Commit-Position: refs/heads/master@{#17370}
2017-03-24 10:16:04 +00:00
d12a8e1c8e Attach TransportFeedbackPacketLossTracker to ANA (PLR only)
This CL is one in a series. To finish the work, the following CLs will be added:
1. CL for connecting RPLR as well
2. CL for RPLR-based FecController
3. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)

BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2638083002
Cr-Commit-Position: refs/heads/master@{#17365}
2017-03-23 18:04:48 +00:00
b0bf93a4b7 DCHECK that no RTPPayloadRegistry instance is used for both audio and video
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2763323002
Cr-Commit-Position: refs/heads/master@{#17352}
2017-03-23 07:10:09 +00:00
25d0bdc1bc Delete support for receiving RTCP RPSI and SLI message.
This code has been unused for years, and at least the RTCP RSPI sending
logic appears broken.

This cl is part 3, following

  https://codereview.webrtc.org/2746413003 (delete sending)
  https://codereview.webrtc.org/2753783002 (delete vp8 feedback mode)

BUG=webrtc:7338

Review-Url: https://codereview.webrtc.org/2742383004
Cr-Commit-Position: refs/heads/master@{#17342}
2017-03-22 14:15:09 +00:00
14adba77ec Don't allocate any RTPSender object for a receive only RtpRtcp module.
This is one step towards separation of send-side and receive-side
processing.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2740163002
Cr-Commit-Position: refs/heads/master@{#17306}
2017-03-20 10:52:39 +00:00
40ba3addc7 Delete unused member RTCPSender::FeedbackState::send_payload_type.
It became unused with cl https://codereview.webrtc.org/2746413003/

BUG=webrtc:7338

Review-Url: https://codereview.webrtc.org/2759433005
Cr-Commit-Position: refs/heads/master@{#17293}
2017-03-17 14:04:00 +00:00
3b941bef7a Minor FlexFEC fixes.
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2720693007
Cr-Commit-Position: refs/heads/master@{#17292}
2017-03-17 14:02:46 +00:00
994ebb4702 Provide default impl of TimeUntilNextProcess in MockRtpRtcp instead of mocking.
This method isn't currently mocked or required by any test, so the safe thing
is to return a reasonably large value from the implementation to avoid busy loops.

BUG=webrtc:7187
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/2744233002
Cr-Commit-Position: refs/heads/master@{#17284}
2017-03-17 11:01:30 +00:00
74e8df8feb Fix rtcp_sender to support sdes with 31 chunk
Previosly it supported up to only 15 chunks which is a limit for csrcs in an rtp packet.

BUG=None

Review-Url: https://codereview.webrtc.org/2758533002
Cr-Commit-Position: refs/heads/master@{#17274}
2017-03-16 15:04:08 +00:00
150708e061 Make ModuleRtpRtcpImpl member variables private.
This is a follow-up to https://codereview.webrtc.org/2747743002.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2750283003
Cr-Commit-Position: refs/heads/master@{#17271}
2017-03-16 12:02:53 +00:00
9ea46b5286 Ignore packets sent on old network route when receiving feedback.
BUG=webrtc:7347
R=philipel@webrtc.org

Review-Url: https://codereview.webrtc.org/2755553003 .
Cr-Commit-Position: refs/heads/master@{#17243}
2017-03-15 11:40:25 +00:00
cd386eb13f Delete support for sending RTCP RPSI and SLI messages.
BUG=webrtc:7338

Review-Url: https://codereview.webrtc.org/2746413003
Cr-Commit-Position: refs/heads/master@{#17229}
2017-03-14 15:54:43 +00:00
a33c62ee65 Add accessor functions for protected member variables of ModuleRtpRtcpImpl.
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2747743002
Cr-Commit-Position: refs/heads/master@{#17220}
2017-03-14 07:49:45 +00:00
7be1dcb98e Delete method ModuleRtpRtcpImpl::SendPayloadType.
This was a trivial delegation wrapper, with only a single use.

BUG=None

Review-Url: https://codereview.webrtc.org/2741413003
Cr-Commit-Position: refs/heads/master@{#17205}
2017-03-13 12:09:27 +00:00
ec304f96b3 GetTransportFeedbackVector return vector with lost packets too, sorted by seq-num
1. GetTransportFeedbackVector will now return a vector which also explicitly states lost packets.
2. The returned vector is unsorted (uses default order - by sequence number). It's up to the users to sort otherwise, if they need a different order.

BUG=None

Review-Url: https://codereview.webrtc.org/2707383006
Cr-Commit-Position: refs/heads/master@{#17114}
2017-03-08 13:03:53 +00:00
21dc1890f4 Replace Clock::CurrentNtp with Clock::CurrentNtpTime
CurrentNtp return time by taking two output parameters by reference
(also breaks style guide)
CurrentNtpTime treat ntp time as single entity and returns it using NtpTime structure.
(making interface clearer)

BUG=None

Review-Url: https://codereview.webrtc.org/2733823002
Cr-Commit-Position: refs/heads/master@{#17088}
2017-03-07 10:51:09 +00:00
f949000834 Rename webrtc::PacketInfo to webrtc::PacketFeedback. This resolves ambiguity with a similarly named RTCPReceiver::PacketInformation and RtpPacketizerVp9::PacketInfo.
BUG=None

Review-Url: https://codereview.webrtc.org/2710093004
Cr-Commit-Position: refs/heads/master@{#17066}
2017-03-06 13:32:21 +00:00
609ab2b3dc Make ExtendedReports::target_bitrate() accessor const
as it supposed to be

BUG=None

Review-Url: https://codereview.webrtc.org/2726843007
Cr-Commit-Position: refs/heads/master@{#17059}
2017-03-06 11:12:12 +00:00
db23ea69b6 Add performance tracing for PlatformThread and parts of the video code.
BUG=webrtc:7219

Review-Url: https://codereview.webrtc.org/2729783004
Cr-Commit-Position: refs/heads/master@{#17009}
2017-03-03 15:21:18 +00:00
16ccfdf457 Reland Move fake_audio_device to its own target.
Patchset 1 is patchset #5 id:80001 of https://codereview.webrtc.org/2717983003/
Patchset 2 fix call_perf_test dep on fake_audio_device.

This reverts commit 985371bda999c6db51286586c5850d2ff58f3511.

Original cl description:

Move fake_audio_device to its own target.
The purpose is to make it usefull for test targets that does not need or can use test_common.

For some reason this also triggered override issues in rtp module tests that are fixed in the same cl.

BUG=none
TBR=kjellander@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2718363002
Cr-Commit-Position: refs/heads/master@{#16922}
2017-02-28 22:41:05 +00:00
c1b57a15bf Test field trial group with startswith rather than equals.
BUG=webrtc:7266

Review-Url: https://codereview.webrtc.org/2717973005
Cr-Commit-Position: refs/heads/master@{#16915}
2017-02-28 16:50:47 +00:00
985371bda9 Revert of Move fake_audio_device to its own target. (patchset #5 id:80001 of https://codereview.webrtc.org/2717983003/ )
Reason for revert:
Breaks build DEPS.

Original issue's description:
> Move fake_audio_device to its own target.
> The purpose is to make it usefull for test targets that does not need or can use test_common.
>
> For some reason this also triggered override issues in rtp module tests that are fixed in the same cl.
>
> BUG=none
>
> Review-Url: https://codereview.webrtc.org/2717983003
> Cr-Commit-Position: refs/heads/master@{#16889}
> Committed: 03d850ddf9

TBR=ehmaldonado@webrtc.org,danilchap@webrtc.org,kjellander@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/2718083003
Cr-Commit-Position: refs/heads/master@{#16890}
2017-02-28 08:56:28 +00:00
03d850ddf9 Move fake_audio_device to its own target.
The purpose is to make it usefull for test targets that does not need or can use test_common.

For some reason this also triggered override issues in rtp module tests that are fixed in the same cl.

BUG=none

Review-Url: https://codereview.webrtc.org/2717983003
Cr-Commit-Position: refs/heads/master@{#16889}
2017-02-28 08:49:48 +00:00
32d0010d86 Add probe logging to RtcEventLog.
In this CL:
 - Add message BweProbeCluster and BweProbeResult to rtc_event_log.proto.
 - Add corresponding log functions to RtcEventLog.
 - Add optional field |probe_cluster_id| to RtpPacket message and added
   an overload function to log with this information.
 - Propagate the probe_cluster_id to where RTP packets are logged.

BUG=webrtc:6984

Review-Url: https://codereview.webrtc.org/2666533002
Cr-Commit-Position: refs/heads/master@{#16857}
2017-02-27 10:18:46 +00:00
8aadd50b96 Propagate packet pacing information to SendTimeHistory.
In order to not make this CL too large I have broken it down into at least two
steps. Previous CL: https://codereview.chromium.org/2628563003/

webrtc::PacedSender::Process                        <--- previous CL start here
webrtc::PacedSender::SendPacket
webrtc::PacketRouter::TimeToSendPacket
webrtc::ModuleRtpRtcpImpl::TimeToSendPacket         <--- previous CL end here, this Cl start here
webrtc::RTPSender::TimeToSendPacket
webrtc::RTPSender::PrepareAndSendPacket
webrtc::RTPSender::AddPacketToTransportFeedback
webrtc::TransportFeedbackAdapter::AddPacket
webrtc::SendTimeHistory::AddAndRemoveOld            <--- this CL end here

BUG=webrtc:6822

Review-Url: https://codereview.webrtc.org/2708873003
Cr-Commit-Position: refs/heads/master@{#16796}
2017-02-23 10:56:13 +00:00