This CL restructures the RtcEventLog protobuf format, by removing the DebugEvent message. This is done by moving the LOG_START and LOG_END events to the EventType enum and making a seperate message for audio playout events. In addition to these changes, some fields were added to the AudioReceiveConfig and AudioSendConfig messages, but these are for future use and are not currently logged yet.
This is a follow-up to CL 1340283002 which adds a SSRC to AudioPlayout events in the RtcEventLog.
BUG=webrtc:4741
R=henrik.lundin@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/1348113003 .
Cr-Commit-Position: refs/heads/master@{#10221}
What used to be the libpeerconnection library is now compiled
statically into the Chromium binary, so clean up references it.
BUG=chromium:482123
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1399513002 .
Cr-Commit-Position: refs/heads/master@{#10216}
The CL includes several changes:
- Adding flags describing the streams and spatial layers.
- Reorganizing the order of the flags, to make them easier to maintain.
- Adding a member .params_ to VideoQualityAnalyzer.
(instead of passing it to every member function manually)
- Updating VideoAnalyzer to support simulcast.
(select appropriate ssrc and fix timestamps which are sometimes increased by 1)
- VP9EncoderImpl already had code for automatic calculation of bitrate for each layer.
Changing to first read bitrates and resolution ratios from the flags, if specified.
If not specified, reverting to the old code are setting the values automatically.
- Changing the parameters in LayerFilteringTransport, replacing
xx_discard_thresholds with selected_xx, to make it easier to use for the end user.
Review URL: https://codereview.webrtc.org/1353263005
Cr-Commit-Position: refs/heads/master@{#10215}
Since the pacer is always enabled, removing enable/disable which makes
all packet queueing succeed. Also renaming one of the ::SendPackets
::InsertPacket to avoid confusion.
BUG=webrtc:1695, webrtc:2629
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1392513002 .
Cr-Commit-Position: refs/heads/master@{#10211}
In video_sender.cc, properly read the number of temporal layers for VP9 too.
Also, some cleanup in video_loopback.cc and video_quality_test.h.
Review URL: https://codereview.webrtc.org/1351693005
Cr-Commit-Position: refs/heads/master@{#10201}
Every now and then we get CLs to codereview.webrtc.org
that are created from a Chromium checkout by editing
the code in third_party/webrtc or third_party/libjingle.
By editing these lower-level codereview.settings files,
we instead cause a crash during 'git cl upload', but the
contents of the file will also be printed, which can work
as an error message. The alternative would be to entirely
remove the files.
BUG=
R=andrew@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1389963002 .
Cr-Commit-Position: refs/heads/master@{#10191}
The OwningThread member of CRITICAL_SECTION is declared as having type
HANDLE but it is actually the thread's Thread ID which is a DWORD. When
doing 64-bit builds of Chromium with VS 2015 this triggers a warning
because of the suspicious conversion from 32-bit integer to 64-bit
pointer.
This change adds a cast (and some comments) to make the conversion
explicit and avoid the warning.
R=henrikg@webrtc.org
BUG=440500
Review URL: https://codereview.webrtc.org/1386183002
Cr-Commit-Position: refs/heads/master@{#10190}
Every malformed URL should now produce an error message in JS, rather than
silently failing and possibly printing a warning message to the console (and
possibly crashing).
Also added some unit tests, and made "ParseIceServers" public.
BUG=445002
Review URL: https://codereview.webrtc.org/1344143002
Cr-Commit-Position: refs/heads/master@{#10186}
Also, in Sample struct, replacing double with the original type.
It makes more sense to save the original data as truthful as possible, and then
convert it to double later if necessary (in the plot script).
Review URL: https://codereview.webrtc.org/1374233002
Cr-Commit-Position: refs/heads/master@{#10184}
When fetching a packet from the rtp packet history, cuased by a
retransmission, the transport seq extension header is enabled but the
sequence number is set to 0. A new transport seq should be assigned in
this case.
BUG=
Review URL: https://codereview.webrtc.org/1385563005
Cr-Commit-Position: refs/heads/master@{#10183}
Since padding is no longer sent on Encoded() callbacks, dummy callbacks
aren't required to generate padding. This skip-frame behavior can then
be removed to get rid of dummy callbacks though nothing was encoded. As
frames don't have to be generated for frames that don't have to be sent
we skip encoding frames that aren't intended to be sent either, reducing
CPU load.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1369923005 .
Cr-Commit-Position: refs/heads/master@{#10181}
The AEC dump was not self-contented enough in the sense that APM configuration is missing, and therefore, given an AEC dump, it is sometimes not clear how to reproduce problems.
This CL tries to address the problem.
Note that this cannot guarantee a perfect reproduction in all cases. Dumping from the middle of a call makes the initial states unknown and thus may make the result non-reproducible.
BUG=
TEST= 1. new dump in Chromium and unpack
2. unpack old dump
R=andrew@webrtc.org, peah@webrtc.org
Review URL: https://codereview.webrtc.org/1348903004 .
Cr-Commit-Position: refs/heads/master@{#10155}
Microsoft introduced modern app from win8. Modern apps can be used cross Microsoft's platforms.
It was confirmed from Microsoft that there is no support for modern app's window capture.
BUG=526883
Review URL: https://codereview.webrtc.org/1371383003
Cr-Commit-Position: refs/heads/master@{#10154}
Adds a loopback button that will connect to itself by simulating another client connection to the web socket server.
Adds an audio only mode switch.
BUG=
Review URL: https://codereview.webrtc.org/1334003002
Cr-Commit-Position: refs/heads/master@{#10153}
Poller thread is currently started in the constructor, so the first call
to PollStats() may happen even before the streams have been configured.
This will blow up on RTC_DCHECK_GT(expected_bitrate_bps_, 0);
Thread should instead be started on PerformTest() call.
BUG=
Review URL: https://codereview.webrtc.org/1378303004
Cr-Commit-Position: refs/heads/master@{#10149}
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.
BUG=4173
Review URL: https://codereview.webrtc.org/1376673004
Cr-Commit-Position: refs/heads/master@{#10144}
