Commit Graph

57 Commits

Author SHA1 Message Date
ee89e7870c Replace CHECK(x && y) with two separate CHECK() calls
That way, the debug printout will tell us which of x and y that was false.

BUG=none

Review-Url: https://codereview.webrtc.org/2988153003
Cr-Commit-Position: refs/heads/master@{#19297}
2017-08-10 00:22:01 +00:00
007d56229a Use default header extension map in rtc_event_log2text
Use defaults if the header extension map is missing from the config.

BUG=webrtc:6399

Review-Url: https://codereview.webrtc.org/2983283002
Cr-Commit-Position: refs/heads/master@{#19267}
2017-08-08 12:40:26 +00:00
333264f089 nit: Avoid pointer-to-unique_ptr in RtcEventLogImpl
BUG=None

Review-Url: https://codereview.webrtc.org/2983573002
Cr-Commit-Position: refs/heads/master@{#19052}
2017-07-16 23:44:08 +00:00
c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00
a80c16a67c Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
This reverts commit c3771cc4d37f5573fe53b7c7cff295a4f0f9560f.
(breaks downstream internal project)

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2972463002 .
Cr-Commit-Position: refs/heads/master@{#18873}
2017-07-01 14:48:18 +00:00
c3771cc4d3 Update includes for webrtc/{base => rtc_base} rename (2/3)
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}
2017-06-30 20:42:44 +00:00
c8e0552c07 Limit the number of simultaneous event logs.
BUG=webrtc:7887

Review-Url: https://codereview.webrtc.org/2956003003
Cr-Commit-Position: refs/heads/master@{#18808}
2017-06-28 13:40:51 +00:00
38ede13042 Support building WebRTC without audio and video.
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}
2017-06-15 19:52:32 +00:00
f3ba6484e3 Change rtp header extension AbsoluteSendTime::Write to take time in 24bit format
making it symmetric to AbsoluteSendTime::Parse function.

Bug: None
Change-Id: I9c71d840768064022ebebbbeb2962aeeecc68392
Reviewed-on: https://chromium-review.googlesource.com/531044
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18555}
2017-06-13 09:08:14 +00:00
84b4d2c1c2 Use rtp_header_extension_map.h instead of rtp_header_extension.h
Finish renaming started in the https://chromium-review.googlesource.com/c/520947/

Bug: webrtc:5565
Change-Id: If420e05165ef7c110b7d38f53dbe73c21a4059bc
Reviewed-on: https://chromium-review.googlesource.com/528095
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18538}
2017-06-12 14:01:20 +00:00
a8e781aedf Make rtc_event_log2text output header extensions
BUG=webrtc:none

Review-Url: https://codereview.webrtc.org/2918103002
Cr-Commit-Position: refs/heads/master@{#18530}
2017-06-12 08:02:46 +00:00
f53c4cd867 Delete rtc_event_log/ringbuffer.h
This code is unused since https://codereview.webrtc.org/2875823003/ and the current implementation is too specific to the event log to be useful elsewhere. I think we should get add a reusable cyclic buffer though and I made a rough draft of what it might look like: https://codereview.webrtc.org/2691073002/

BUG=webrtc:7732

Review-Url: https://codereview.webrtc.org/2923163006
Cr-Commit-Position: refs/heads/master@{#18465}
2017-06-07 08:53:30 +00:00
1c187dcd80 Replace RingBuffer by std::deque in RtcEventLog.
BUG=webrtc:7732

Review-Url: https://codereview.webrtc.org/2875823003
Cr-Commit-Position: refs/heads/master@{#18447}
2017-06-05 15:55:40 +00:00
5b542130d7 Print configured header extensions and codecs in rtc_event_log2text.
BUG=None

Review-Url: https://codereview.webrtc.org/2916053002
Cr-Commit-Position: refs/heads/master@{#18379}
2017-06-01 12:23:03 +00:00
8b2b9efab6 Use oneof in event log proto to reduce memory consumption.
BUG=webrtc:7732

Review-Url: https://codereview.webrtc.org/2919483002
Cr-Commit-Position: refs/heads/master@{#18350}
2017-05-31 12:45:56 +00:00
8fbc765421 Ensure the RtcEventLog parser is backwards compatible
The old video send stream configs could contain multiple SSRCs and RTX SSRCs (in case of simulcast). To ensure that the RtcEventLog parser is backwards compatible, we have to return one config for every SSRC.

Also update the parsing functions for the other config types to return the config instead of passing in an output parameter.

BUG=webrtc:7731

Review-Url: https://codereview.webrtc.org/2912113002
Cr-Commit-Position: refs/heads/master@{#18343}
2017-05-31 09:03:16 +00:00
77cd58e140 This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.
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}
2017-05-30 10:52:10 +00:00
33bb86d5a1 Remove final from RtcEventLogNullImpl
The reason is that there might be implementations that do not want to implement all methods.
To allow easier modification of the RtcEventLog interface, allow these implementation to inherit the RtcEventLogNullImpl implementation.

BUG=none

Review-Url: https://codereview.webrtc.org/2903003002
Cr-Commit-Position: refs/heads/master@{#18298}
2017-05-29 09:46:05 +00:00
f472699bbd Replace AudioSendStream::Config with rtclog::StreamConfig.
BUG=webrtc:7538

Review-Url: https://codereview.webrtc.org/2856063003
Cr-Commit-Position: refs/heads/master@{#18224}
2017-05-22 17:12:26 +00:00
ac8f52de70 Replace AudioReceiveStream::Config with rtclog::StreamConfig.
BUG=webrtc:7538

Review-Url: https://codereview.webrtc.org/2851303007
Cr-Commit-Position: refs/heads/master@{#18223}
2017-05-22 16:36:28 +00:00
c0876aab46 Replace VideoSendStream::Config with new rtclog::StreamConfig in RtcEventLog.
BUG=webrtc:7538

Review-Url: https://codereview.webrtc.org/2857933002
Cr-Commit-Position: refs/heads/master@{#18221}
2017-05-22 11:08:28 +00:00
09e71daec5 Replace VideoReceiveStream::Config with new rtclog::StreamConfig in RtcEventLog.
BUG=webrtc:7538

Review-Url: https://codereview.webrtc.org/2850793002
Cr-Commit-Position: refs/heads/master@{#18220}
2017-05-22 10:26:49 +00:00
c52bd61f65 Change rtp_event_log2text to ignore webrtc::MediaType from proto.
BUG=webrtc:7538

Review-Url: https://codereview.webrtc.org/2894833003
Cr-Commit-Position: refs/heads/master@{#18210}
2017-05-19 12:32:56 +00:00
8a329652de DCHECK that we don't insert nullptr into event log.
BUG=None

Review-Url: https://codereview.webrtc.org/2874753003
Cr-Commit-Position: refs/heads/master@{#18100}
2017-05-11 13:05:27 +00:00
ee37e86de8 Create tool to print statistics about the file size usage of an RTC event log.
BUG=webrtc:7502

Review-Url: https://codereview.webrtc.org/2717553004
Cr-Commit-Position: refs/heads/master@{#17932}
2017-04-28 14:48:17 +00:00
10fc0e6385 Delay based logging.
BUG=none

Review-Url: https://codereview.webrtc.org/2808833002
Cr-Commit-Position: refs/heads/master@{#17641}
2017-04-11 08:50:23 +00:00
9765370416 Resolve dependency between rtc_event_log_api and remote_bitrate_estimator
BUG=webrtc:7257

Review-Url: https://codereview.webrtc.org/2800633004
Cr-Commit-Position: refs/heads/master@{#17638}
2017-04-11 07:49:44 +00:00
7c2c8438f1 Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )
Reason for revert:
Trying to re-land after solving some related issues.

There are no changes compared to the original CL.

Original issue's description:
> Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
>
> Reason for revert:
> I will try to reland next week because it is causing some problems.
>
> Original issue's description:
> > To accommodate some downstream WebRTC users we need to loosen
> > the coupling between our code and the //third_party/protobuf.
> >
> > This includes using typedefs to define strings instead of
> > assuming std::string.
> >
> > After this refactoring it will be possible to link with other
> > protobuf implementations than the current one.
> >
> > We moved the PRESUBMIT check to another CL [1]. The goal of this
> > presubmit is to avoid the direct usage of google::protobuf outside
> > of the webrtc/base/protobuf_utils.h header file.
> >
> > [1] - https://codereview.webrtc.org/2753823003/
> >
> > BUG=webrtc:7340
> > NOTRY=True
> >
> > Review-Url: https://codereview.webrtc.org/2747863003
> > Cr-Commit-Position: refs/heads/master@{#17466}
> > Committed: 16ab93b952
>
> TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7340
>
> Review-Url: https://codereview.webrtc.org/2786363002
> Cr-Commit-Position: refs/heads/master@{#17483}
> Committed: d00aad5eb2

TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7340
NOTRY=True

Review-Url: https://codereview.webrtc.org/2791963003
Cr-Commit-Position: refs/heads/master@{#17584}
2017-04-07 07:59:12 +00:00
cde46b7278 Resolve cyclic dependency between audio network adaptor and event log api
BUG=webrtc:7257

Review-Url: https://codereview.webrtc.org/2745473003
Cr-Commit-Position: refs/heads/master@{#17565}
2017-04-06 12:59:10 +00:00
368f5cf27e Replace use of system_wrappers/include/logging.h by base/logging.h.
BUG=webrtc:5118

Review-Url: https://codereview.webrtc.org/2781343002
Cr-Commit-Position: refs/heads/master@{#17539}
2017-04-05 12:00:33 +00:00
d00aad5eb2 Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
Reason for revert:
I will try to reland next week because it is causing some problems.

Original issue's description:
> To accommodate some downstream WebRTC users we need to loosen
> the coupling between our code and the //third_party/protobuf.
>
> This includes using typedefs to define strings instead of
> assuming std::string.
>
> After this refactoring it will be possible to link with other
> protobuf implementations than the current one.
>
> We moved the PRESUBMIT check to another CL [1]. The goal of this
> presubmit is to avoid the direct usage of google::protobuf outside
> of the webrtc/base/protobuf_utils.h header file.
>
> [1] - https://codereview.webrtc.org/2753823003/
>
> BUG=webrtc:7340
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2747863003
> Cr-Commit-Position: refs/heads/master@{#17466}
> Committed: 16ab93b952

TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7340

Review-Url: https://codereview.webrtc.org/2786363002
Cr-Commit-Position: refs/heads/master@{#17483}
2017-03-31 10:08:07 +00:00
16ab93b952 To accommodate some downstream WebRTC users we need to loosen
the coupling between our code and the //third_party/protobuf.

This includes using typedefs to define strings instead of
assuming std::string.

After this refactoring it will be possible to link with other
protobuf implementations than the current one.

We moved the PRESUBMIT check to another CL [1]. The goal of this
presubmit is to avoid the direct usage of google::protobuf outside
of the webrtc/base/protobuf_utils.h header file.

[1] - https://codereview.webrtc.org/2753823003/

BUG=webrtc:7340
NOTRY=True

Review-Url: https://codereview.webrtc.org/2747863003
Cr-Commit-Position: refs/heads/master@{#17466}
2017-03-30 08:24:20 +00:00
e127e7a0ed Visualize events related to probing in the total bitrate graph.
BUG=webrtc:6984
R=terelius@webrtc.org

Review-Url: https://codereview.webrtc.org/2782553005 .
Cr-Commit-Position: refs/heads/master@{#17449}
2017-03-29 14:28:54 +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
d9cbd5138d Remove unused include from rtc_event_log_parser.cc
BUG=None

Review-Url: https://codereview.webrtc.org/2723533002
Cr-Commit-Position: refs/heads/master@{#16893}
2017-02-28 09:46:19 +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
0f8b403eb5 Introduce a new constructor to PlatformThread.
The new constructor introduces two new changes:

* Support specifying thread priority at construction time.
  - Moving forward, the SetPriority() method will be removed.
* New thread function type.
  - The new type has 'void' as a return type and a polling loop
    inside PlatformThread, is not used.

The old function type is still supported until all places have been moved over.

In this CL, the first steps towards deprecating the old mechanism are taken
by moving parts of the code that were simple to move, over to the new callback
type.

BUG=webrtc:7187

Review-Url: https://codereview.webrtc.org/2708723003
Cr-Commit-Position: refs/heads/master@{#16779}
2017-02-22 19:22:05 +00:00
7d59f6b1c4 Reland of Delete class SSRCDatabase, and its global ssrc registry. (patchset #1 id:1 of https://codereview.webrtc.org/2700413002/ )
Reason for revert:
Intend to fix perf problem and reland.

Original issue's description:
> Revert of Delete class SSRCDatabase, and its global ssrc registry. (patchset #20 id:370001 of https://codereview.webrtc.org/2644303002/ )
>
> Reason for revert:
> Breaks webrtc_perf_tests reliably:
> https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/1780
> https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus4%29/builds/178
>
> We're actively working on getting a quick version of webrtc_perf_tests up on the trybots again to prevent breakages like this: https://bugs.chromium.org/p/webrtc/issues/detail?id=7101
>
> Original issue's description:
> > Delete class SSRCDatabase, and its global ssrc registry,
> > and the method RTPSender::GenerateNewSSRC.
> >
> > It's now mandatory for higher layers to call SetSSRC, RTPSender
> > no longer allocates any ssrc by default.
> >
> > BUG=webrtc:4306,webrtc:6887
> >
> > Review-Url: https://codereview.webrtc.org/2644303002
> > Cr-Commit-Position: refs/heads/master@{#16670}
> > Committed: b78d4d1383
>
> TBR=solenberg@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,ivoc@webrtc.org,nisse@webrtc.org
> NOTRY=True
> BUG=webrtc:4306,webrtc:6887
>
> Review-Url: https://codereview.webrtc.org/2700413002
> Cr-Commit-Position: refs/heads/master@{#16693}
> Committed: b5848ecbf5

TBR=solenberg@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,ivoc@webrtc.org,kjellander@webrtc.org,kjellander@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:4306,webrtc:6887

Review-Url: https://codereview.webrtc.org/2702203002
Cr-Commit-Position: refs/heads/master@{#16737}
2017-02-21 11:40:24 +00:00
424e6cfd58 Rename some variables and methods in RTC event log.
Rename loss based and delay based bwe updates in proto (and correspondingly in the C++ code).

BUG=webrtc:6423

Review-Url: https://codereview.webrtc.org/2705613002
Cr-Commit-Position: refs/heads/master@{#16719}
2017-02-20 13:14:41 +00:00
b5848ecbf5 Revert of Delete class SSRCDatabase, and its global ssrc registry. (patchset #20 id:370001 of https://codereview.webrtc.org/2644303002/ )
Reason for revert:
Breaks webrtc_perf_tests reliably:
https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/1780
https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus4%29/builds/178

We're actively working on getting a quick version of webrtc_perf_tests up on the trybots again to prevent breakages like this: https://bugs.chromium.org/p/webrtc/issues/detail?id=7101

Original issue's description:
> Delete class SSRCDatabase, and its global ssrc registry,
> and the method RTPSender::GenerateNewSSRC.
>
> It's now mandatory for higher layers to call SetSSRC, RTPSender
> no longer allocates any ssrc by default.
>
> BUG=webrtc:4306,webrtc:6887
>
> Review-Url: https://codereview.webrtc.org/2644303002
> Cr-Commit-Position: refs/heads/master@{#16670}
> Committed: b78d4d1383

TBR=solenberg@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,ivoc@webrtc.org,nisse@webrtc.org
NOTRY=True
BUG=webrtc:4306,webrtc:6887

Review-Url: https://codereview.webrtc.org/2700413002
Cr-Commit-Position: refs/heads/master@{#16693}
2017-02-18 20:00:50 +00:00
b78d4d1383 Delete class SSRCDatabase, and its global ssrc registry,
and the method RTPSender::GenerateNewSSRC.

It's now mandatory for higher layers to call SetSSRC, RTPSender
no longer allocates any ssrc by default.

BUG=webrtc:4306,webrtc:6887

Review-Url: https://codereview.webrtc.org/2644303002
Cr-Commit-Position: refs/heads/master@{#16670}
2017-02-17 16:34:35 +00:00
0baf55d23b Add logging of delay-based bandwidth estimate.
BUG=webrtc:6423

Review-Url: https://codereview.webrtc.org/2695923004
Cr-Commit-Position: refs/heads/master@{#16663}
2017-02-17 11:38:28 +00:00
bb46b95dbe Add option to print information about configured SSRCs from RTC event logs.
BUG=webrtc:7118

Review-Url: https://codereview.webrtc.org/2686823002
Cr-Commit-Position: refs/heads/master@{#16500}
2017-02-08 17:37:30 +00:00
d4ed7f59e4 New tool for printing basic packet information from an RTC event log to stdout.
BUG=webrtc:7118

Review-Url: https://codereview.webrtc.org/2673403002
Cr-Commit-Position: refs/heads/master@{#16488}
2017-02-08 12:22:53 +00:00
1474212895 Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ )
Reason for revert:
Downstream project relied on changed struct.

Transition made possible by https://codereview.webrtc.org/2655243006/.

Original issue's description:
> Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ )
>
> Reason for revert:
> Breaks internal downstream project.
>
> Original issue's description:
> > Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters.
> >
> > Prior to this CL, received RTX (associated) payload types were only configured
> > when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX
> > SSRC was set up.
> >
> > After this CL, the RTX (associated) payload types are set in
> > WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set
> > them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since
> > that is the code path that sets other SSRCs.
> >
> > As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed.
> > We remove the possibility for each video payload type to have an associated
> > specific RTX SSRC. Although the config previously allowed for this, all payload
> > types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry
> > did not support multiple SSRCs. This change to the config struct should thus not
> > have any functional impact. The change does however affect the RtcEventLog, since
> > that is used for storing the VideoReceiveStream::Configs. For simplicity,
> > this CL does not change the event log proto definitions, instead duplicating
> > the serialized RTX SSRCs such that they fit in the existing proto definition.
> >
> > BUG=webrtc:7011
> >
> > Review-Url: https://codereview.webrtc.org/2646073004
> > Cr-Commit-Position: refs/heads/master@{#16302}
> > Committed: fe2bef39cd
>
> TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7011
>
> Review-Url: https://codereview.webrtc.org/2649323010
> Cr-Commit-Position: refs/heads/master@{#16307}
> Committed: e4974953ce

TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
# NOTREECHECKS=true
# NOTRY=true
BUG=webrtc:7011

Review-Url: https://codereview.webrtc.org/2654163006
Cr-Commit-Position: refs/heads/master@{#16322}
2017-01-27 12:53:07 +00:00
e4974953ce Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ )
Reason for revert:
Breaks internal downstream project.

Original issue's description:
> Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters.
>
> Prior to this CL, received RTX (associated) payload types were only configured
> when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX
> SSRC was set up.
>
> After this CL, the RTX (associated) payload types are set in
> WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set
> them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since
> that is the code path that sets other SSRCs.
>
> As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed.
> We remove the possibility for each video payload type to have an associated
> specific RTX SSRC. Although the config previously allowed for this, all payload
> types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry
> did not support multiple SSRCs. This change to the config struct should thus not
> have any functional impact. The change does however affect the RtcEventLog, since
> that is used for storing the VideoReceiveStream::Configs. For simplicity,
> this CL does not change the event log proto definitions, instead duplicating
> the serialized RTX SSRCs such that they fit in the existing proto definition.
>
> BUG=webrtc:7011
>
> Review-Url: https://codereview.webrtc.org/2646073004
> Cr-Commit-Position: refs/heads/master@{#16302}
> Committed: fe2bef39cd

TBR=stefan@webrtc.org,magjed@webrtc.org,terelius@webrtc.org,brandtr@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7011

Review-Url: https://codereview.webrtc.org/2649323010
Cr-Commit-Position: refs/heads/master@{#16307}
2017-01-26 21:22:37 +00:00
fe2bef39cd Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters.
Prior to this CL, received RTX (associated) payload types were only configured
when WebRtcVideoChannel2::AddRecvStream was called. In the same method, the RTX
SSRC was set up.

After this CL, the RTX (associated) payload types are set in
WebRtcVideoChannel2::SetRecvParameters, which is the appropriate place to set
them. The RTX SSRC is still set in WebRtcVideoChannel2::AddRecvStream, since
that is the code path that sets other SSRCs.

As part of this fix, the VideoReceiveStream::Config::Rtp struct is changed.
We remove the possibility for each video payload type to have an associated
specific RTX SSRC. Although the config previously allowed for this, all payload
types always had the same RTX SSRC set, and the underlying RtpPayloadRegistry
did not support multiple SSRCs. This change to the config struct should thus not
have any functional impact. The change does however affect the RtcEventLog, since
that is used for storing the VideoReceiveStream::Configs. For simplicity,
this CL does not change the event log proto definitions, instead duplicating
the serialized RTX SSRCs such that they fit in the existing proto definition.

BUG=webrtc:7011

Review-Url: https://codereview.webrtc.org/2646073004
Cr-Commit-Position: refs/heads/master@{#16302}
2017-01-26 16:03:58 +00:00
4b7c952376 Reland of "Log audio network adapter decisions in event log."
This was originally reviewed https://codereview.webrtc.org/2559953002/

It was reverted due to a bug in the original CL, see https://codereview.webrtc.org/2631703002/

This CL is to fix and reland.

BUG=webrtc:6845

Review-Url: https://codereview.webrtc.org/2644863002
Cr-Commit-Position: refs/heads/master@{#16242}
2017-01-24 12:54:59 +00:00
363a29157a Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ )
Reason for revert:
Breaks chromium.webrtc.fyi.

Original issue's description:
> Log audio network adapter decisions in event log.
>
> BUG=webrtc:6845
>
> Review-Url: https://codereview.webrtc.org/2559953002
> Cr-Commit-Position: refs/heads/master@{#16053}
> Committed: 3663681b5d

TBR=minyue@webrtc.org,henrik.lundin@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6845

Review-Url: https://codereview.webrtc.org/2631703002
Cr-Commit-Position: refs/heads/master@{#16054}
2017-01-13 14:52:12 +00:00
3663681b5d Log audio network adapter decisions in event log.
BUG=webrtc:6845

Review-Url: https://codereview.webrtc.org/2559953002
Cr-Commit-Position: refs/heads/master@{#16053}
2017-01-13 14:10:16 +00:00