4787 Commits

Author SHA1 Message Date
ivoc
112a3d81db Added functions on libjingle API to start and stop the recording of an RtcEventLog.
BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1374253002

Cr-Commit-Position: refs/heads/master@{#10297}
2015-10-16 09:22:23 +00:00
pbos
65e15bafaa Add native-handle support for single VP8 streams.
Implements SupportsNativeHandle() in SimulcastEncoderAdapter which works
when there's only a single encoder.

BUG=webrtc:5060
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1397653004

Cr-Commit-Position: refs/heads/master@{#10291}
2015-10-15 17:52:21 +00:00
sprang
4af0f1a098 Add screenshare perf tests with lossy links
BUG=

Review URL: https://codereview.webrtc.org/1409513005

Cr-Commit-Position: refs/heads/master@{#10290}
2015-10-15 15:34:06 +00:00
stefan
c1aeaf0dc3 Wire up packet_id / send time callbacks to webrtc via libjingle.
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1363573002

Cr-Commit-Position: refs/heads/master@{#10289}
2015-10-15 14:26:17 +00:00
Henrik Kjellander
27576e0b68 Landmines support to ease clobbering builds
Landmines is a feature used in Chromium that makes it possible to
clobber the build output directory when needed. Example scenarios
are when compiler/tool/infrastructure changes require a full rebuild.
This is mainly to ease clobbering on all bots, but will also ensure
developers don't have to waste time on figuring out what's wrong
(or rely on reading PSA e-mails announcing when such manual action
is required).

This CL depends on https://codereview.chromium.org/1407733002/
being landed and rolled into DEPS first.

BUG=5077
R=kjellander@chromium.org, machenbach@chromium.org

Review URL: https://codereview.webrtc.org/1402923003 .

Cr-Commit-Position: refs/heads/master@{#10287}
2015-10-15 12:24:29 +00:00
pbos
a2f30deea3 Log Call {audio, video} stream deletions.
BUG=
R=solenberg@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1400333002

Cr-Commit-Position: refs/heads/master@{#10286}
2015-10-15 12:22:21 +00:00
solenberg
5bdddf91d3 Move PRNG from BWE test framework to webrtc/test.
BUG=

Review URL: https://codereview.webrtc.org/1404953002

Cr-Commit-Position: refs/heads/master@{#10285}
2015-10-15 12:10:33 +00:00
Peter Boström
ab73d13c4b Remove internal encoders from VCMCodecDatabase.
Encoders need to be externally provided. To use software encoders they
need to be created and registered from the outside.

BUG=webrtc:1695
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1394823002 .

Cr-Commit-Position: refs/heads/master@{#10283}
2015-10-15 10:01:48 +00:00
deadbeef
d59daf8023 Merging BaseSession code into WebRtcSession.
After the TransportController CL, BaseSession does little more than
hold a state and an error, and act as an intermediary for the
TransportController. So it doesn't make sense for it to be its own
class.

Review URL: https://codereview.webrtc.org/1397973002

Cr-Commit-Position: refs/heads/master@{#10281}
2015-10-14 22:02:50 +00:00
perkj
a9046d0969 Add unit test to decode to a surface texture.
Also parameterise on PeerConnectionParameters to prepare for more test variations. (capture and encode to textures)

Review URL: https://codereview.webrtc.org/1404093002

Cr-Commit-Position: refs/heads/master@{#10279}
2015-10-14 19:55:25 +00:00
noahric
65220a70a3 Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
Also updated the RTPPayloadRegistry::RestoreOriginalPacket signature to not take the first arg as a **, since it isn't modified.

Review URL: https://codereview.webrtc.org/1394573004

Cr-Commit-Position: refs/heads/master@{#10276}
2015-10-14 18:29:56 +00:00
henrik.lundin
bd7de0c6ef Delete full-band mode from the iSAC codec
This mode is no longer used.

BUG=4210

Review URL: https://codereview.webrtc.org/1392173004

Cr-Commit-Position: refs/heads/master@{#10275}
2015-10-14 13:06:00 +00:00
henrik.lundin
06b869f11a Delete iSAC-fb from NetEq
This is no longer used. Related code in the iSAC codec itself will be
deleted a follow-up CL.

BUG=4210

Review URL: https://codereview.webrtc.org/1404463003

Cr-Commit-Position: refs/heads/master@{#10272}
2015-10-14 10:44:59 +00:00
stefan
457a61db61 Pause/resume pacer from Call instead of via SendStreams.
BUG=webrtc:5073

Review URL: https://codereview.webrtc.org/1398443007

Cr-Commit-Position: refs/heads/master@{#10271}
2015-10-14 10:13:04 +00:00
Henrik Kjellander
b79472a4fb Roll chromium_revision c089d37..159828f (353662:353696)
Due to https://codereview.chromium.org/1397493004 we're now adding
a build_overrides directory in WebRTC. Thanks to this, we no longer
need to pass --args="build_with_chromium=false" when running GN in
standalone WebRTC.

Change log: c089d37..159828f
Full diff: c089d37..159828f

No dependencies changed.
No update to Clang.

BUG=webrtc:5070,chromium:541791
TBR=tommi@webrtc.org
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1403453003 .

Cr-Commit-Position: refs/heads/master@{#10270}
2015-10-14 06:14:10 +00:00
sprang
7dc39f331a Avoid data race in RtcpReceiver.
See eg https://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/3930/steps/video_engine_tests/logs/stdio

Also some cleanup, lock annotations.

BUG=

Review URL: https://codereview.webrtc.org/1401463003

Cr-Commit-Position: refs/heads/master@{#10266}
2015-10-13 16:17:56 +00:00
henrik.lundin
9ea2147e5c Delete iSAC-fb from AudioCodingModule
This is no longer used. Related code in NetEq and the iSAC codec itself
will be deleted in follow-up CLs.

BUG=4210

Review URL: https://codereview.webrtc.org/1404623002

Cr-Commit-Position: refs/heads/master@{#10264}
2015-10-13 13:28:04 +00:00
dcheng
3402bcda56 Make the WARN_UNUSED_RESULT macro match the Chromium one.
BUG=none

Review URL: https://codereview.webrtc.org/1399313002

Cr-Commit-Position: refs/heads/master@{#10259}
2015-10-12 23:28:20 +00:00
Magnus Jedvert
fc950848e3 Fix: RefCountInterface: Make AddRef() and Release() const
The landed CL contained some unwanted changes.

TBR=tommi

Review URL: https://codereview.webrtc.org/1401743002 .

Cr-Commit-Position: refs/heads/master@{#10255}
2015-10-12 14:10:50 +00:00
Magnus Jedvert
1b40a9a8af RefCountInterface: Make AddRef() and Release() const
This CL makes AddRef() and Release() const member methods and the refcount integer mutable. This is reasonable, because they only manage the lifetime of the object, and this is also how it's done in Chromium.

The purpose is to be able to capture a const pointer in a scoped_refptr, which is currenty impossible. The practial problem this CL solves is this:

void Foo::Bar() const {}

rtc::Callback0<void> Foo::MakeClosure() const {
  return rtc::Bind(&Foo::Bar, this);
}

We currently capture |this| as const Foo*. With this CL, |this| will be captured as scoped_refptr<const Foo>.

A test is also added in bind_unittest to check this behaviour.

BUG=webrtc:5065
R=perkj@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1403683004 .

Cr-Commit-Position: refs/heads/master@{#10253}
2015-10-12 13:50:50 +00:00
sprang
7a975f75e7 Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ )
Reason for revert:
Temporarily reverting as this causes some issues with perf tests. Especially tests with packet loss no longer works.

Original issue's description:
> Adding support for simulcast and spatial layers into VideoQualityTest
>
> 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.
>
> Committed: https://crrev.com/87f83a9a27d657731ccb54025bc04ccad0da136e
> Cr-Commit-Position: refs/heads/master@{#10215}

TBR=pbos@webrtc.org,mflodman@webrtc.org,ivica@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1397363002

Cr-Commit-Position: refs/heads/master@{#10252}
2015-10-12 13:33:24 +00:00
Marco
e7f6b565e4 VP9: Enable multi-threading for SVC.
This was disabled due to issues with multi-threading
and spatial layers, but have since been fixed.

R=stefan@webrtc.org
TBR=mflodman@webrtc.org, stefan@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1390353003 .

Cr-Commit-Position: refs/heads/master@{#10229}
2015-10-08 20:57:15 +00:00
torbjorng
4e572470a3 Provide RSA2048 as per RFC
Original CL here:
https://codereview.webrtc.org/1329493005

That CL is in patch set #1 of this CL.
This CL resolves a method collision in Chrome.

BUG=webrtc:4972

Review URL: https://codereview.webrtc.org/1394223002

Cr-Commit-Position: refs/heads/master@{#10222}
2015-10-08 16:43:03 +00:00
Ivo Creusen
301aaed813 Update to the RtcEventLog protobuf to remove the DebugEvent message.
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}
2015-10-08 16:07:53 +00:00
tfarina
8ac544e811 Get rid of deprecated SocketAddress::IsAny() method.
This patch converts the usage of IsAny() to IsAnyIP() and removes the
deprecated method.

BUG=None
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1392153002

Cr-Commit-Position: refs/heads/master@{#10220}
2015-10-08 14:15:49 +00:00
phoglund
c671139ef2 Removing M API call for now to green up downstream.
BUG=None

Review URL: https://codereview.webrtc.org/1392903005

Cr-Commit-Position: refs/heads/master@{#10219}
2015-10-08 13:33:56 +00:00
Henrik Kjellander
6ffc3309de Remove references to libpeerconnection.
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}
2015-10-08 12:41:05 +00:00
ivica
87f83a9a27 Adding support for simulcast and spatial layers into VideoQualityTest
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}
2015-10-08 12:13:37 +00:00
ivica
c1cc854d54 Fixing perf regression caused by refactoring full stack tests
Calling CreateCapturer after CreateStreams. The wrong order of calling those methods seems to have caused perf regressions.

Testing has been done here: https://codereview.webrtc.org/1371113004/

BUG=chromium:534220

Review URL: https://codereview.webrtc.org/1394463002

Cr-Commit-Position: refs/heads/master@{#10212}
2015-10-08 10:44:11 +00:00
Peter Boström
e23e737177 Disable pacer disabling.
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}
2015-10-08 09:44:29 +00:00
torbjorng
335204c550 Revert of Provide RSA2048 as per RFC (patchset #9 id:200001 of https://codereview.webrtc.org/1329493005/ )
Reason for revert:
Breaks chrome.

Original issue's description:
> provide RSA2048 as per RFC
>
> BUG=webrtc:4972
>
> Committed: https://crrev.com/0df3eb03c9a6a8299d7e18c8c314ca58c2f0681e
> Cr-Commit-Position: refs/heads/master@{#10209}

TBR=hbos@webrtc.org,juberti@google.com,jbauch@webrtc.org,henrikg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4972

Review URL: https://codereview.webrtc.org/1397703002

Cr-Commit-Position: refs/heads/master@{#10210}
2015-10-08 09:30:21 +00:00
torbjorng
0df3eb03c9 provide RSA2048 as per RFC
BUG=webrtc:4972

Review URL: https://codereview.webrtc.org/1329493005

Cr-Commit-Position: refs/heads/master@{#10209}
2015-10-08 09:06:20 +00:00
asapersson
f839dcc870 Add stats for rendered pixels (sqrt(w*h)) per second:
- "WebRTC.Video.RenderSqrtPixelsPerSecond"

BUG=chromium:512752

Review URL: https://codereview.webrtc.org/1366583002

Cr-Commit-Position: refs/heads/master@{#10208}
2015-10-08 07:42:07 +00:00
ivica
e78e2c714b Using different sequence numbers for different SSRCs
This seems to solve the unexpected behavior when selecting lower layers.
Also, this replaces https://codereview.webrtc.org/1327153002/

Review URL: https://codereview.webrtc.org/1350383004

Cr-Commit-Position: refs/heads/master@{#10206}
2015-10-08 06:44:35 +00:00
Alex Glaznev
fddf6e526c Use WebRTC logging in MediaCodec JNI code.
Also enable HW encoder scaling in AppRTCDemo.

R=wzh@webrtc.org

Review URL: https://codereview.webrtc.org/1396653002 .

Cr-Commit-Position: refs/heads/master@{#10205}
2015-10-07 23:51:20 +00:00
ivica
c7199c2d0b Read the number of TLs for VP9 too + cleanup
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}
2015-10-07 13:43:43 +00:00
Henrik Kjellander
78543284d0 Fix minor GYP error in webrtc/tools/internal_tools.gyp
It seems 'deps' is similar to 'dependencies' for the ninja and make
generators in GYP, but some generators does not support it.
Better use the correct key.
This was introduced in https://codereview.webrtc.org/1387243002/

TBR=phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/1386353003 .

Cr-Commit-Position: refs/heads/master@{#10200}
2015-10-07 12:51:40 +00:00
torbjorng
172f009be2 Get rid of SCHANNEL code.
BUG=webrtc:5045

Review URL: https://codereview.webrtc.org/1383253002

Cr-Commit-Position: refs/heads/master@{#10199}
2015-10-07 11:58:00 +00:00
Peter Boström
70a5e0ead6 Remove (u)int typedefs from basictypes.h.
BUG=webrtc:5024
R=henrikg@webrtc.org

Review URL: https://codereview.webrtc.org/1387413002 .

Cr-Commit-Position: refs/heads/master@{#10197}
2015-10-07 11:19:47 +00:00
Peter Boström
0c4e06b4c6 Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.

BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1362503003 .

Cr-Commit-Position: refs/heads/master@{#10196}
2015-10-07 10:23:32 +00:00
ivica
8d15bd6dab Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
Reason for revert:
The reverted commit didn't affect the tests, but the one before: https://codereview.webrtc.org/1385563005/

I've run the test that was failing (EndToEndTest.AssignsTransportSequenceNumbers) locally multiple times, and it works fine (finishes successfully in 150-170ms).

Original issue's description:
> Revert of Collecting encode_time_ms for each frame (patchset #13 id:220001 of https://codereview.webrtc.org/1374233002/ )
>
> Reason for revert:
> Breaks EndToEndTest.AssignsTransportSequenceNumbers in video_engine_tests
> on several bots:
> http://build.chromium.org/p/client.webrtc/builders/Linux64%20Debug/builds/5507
> http://build.chromium.org/p/client.webrtc/builders/Mac64%20Debug/builds/4815
> http://build.chromium.org/p/client.webrtc/builders/Win%20SyzyASan/builds/3272
> http://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/4414
>
> It seems very unfortunate that it breaks on _exactly_ the bot configs that aren't covered by the CQ trybots.
>
> Original issue's description:
> > Collecting encode_time_ms for each frame.
> >
> > 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).
> >
> > Committed: https://crrev.com/092b13384e57b33e2003d9736dfa1f491e76f938
> > Cr-Commit-Position: refs/heads/master@{#10184}
>
> TBR=sprang@webrtc.org,pbos@webrtc.org,mflodman@webrtc.org,asapersson@webrtc.org,ivica@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/810447972425e890bc7911af27f894b86e9b7e6f
> Cr-Commit-Position: refs/heads/master@{#10185}

TBR=sprang@webrtc.org,pbos@webrtc.org,mflodman@webrtc.org,asapersson@webrtc.org,kjellander@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1390163002

Cr-Commit-Position: refs/heads/master@{#10195}
2015-10-07 09:43:25 +00:00
Henrik Kjellander
67bcb609a3 GN: Port frame_analyzer and rgba_to_i420_converter targets
Original patch by tfarina@chromium.org at
https://webrtc-codereview.appspot.com/42999004/

BUG=chromium:461019, webrtc:4504
TESTED=Tested on Linux with the following command lines:
$ gn gen/out/Debug --args='is_debug=true build_with_chromium=false'
$ ninja -C out/Debug frame_analyzer rgba_to_i420_converter
Also successfully compiled from a Chromium checkout using the steps in webrtc:4504.

R=tfarina@chromium.org

Review URL: https://codereview.webrtc.org/1387243002 .

Cr-Commit-Position: refs/heads/master@{#10193}
2015-10-07 06:43:07 +00:00
Henrik Kjellander
a38e31a054 Update lower-level codereview.settings files.
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}
2015-10-06 21:19:34 +00:00
brucedawson
a10492ff33 Fix VS 2015 warning by adding an additional cast
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}
2015-10-06 20:34:34 +00:00
Alejandro Luebs
10950692d6 Revert "Transport sequence number should be set also for retransmissions."
After this CL, video_engine_test started failing flakily in different bots for different CLs.

TBR=sprang@webrtc.org

Review URL: https://codereview.webrtc.org/1393553003 .

Cr-Commit-Position: refs/heads/master@{#10188}
2015-10-06 19:27:12 +00:00
deadbeef
0a6c4ca942 Catching more errors when parsing ICE server URLs.
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}
2015-10-06 18:38:33 +00:00
kjellander
8104479724 Revert of Collecting encode_time_ms for each frame (patchset #13 id:220001 of https://codereview.webrtc.org/1374233002/ )
Reason for revert:
Breaks EndToEndTest.AssignsTransportSequenceNumbers in video_engine_tests
on several bots:
http://build.chromium.org/p/client.webrtc/builders/Linux64%20Debug/builds/5507
http://build.chromium.org/p/client.webrtc/builders/Mac64%20Debug/builds/4815
http://build.chromium.org/p/client.webrtc/builders/Win%20SyzyASan/builds/3272
http://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/4414

It seems very unfortunate that it breaks on _exactly_ the bot configs that aren't covered by the CQ trybots.

Original issue's description:
> Collecting encode_time_ms for each frame.
>
> 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).
>
> Committed: https://crrev.com/092b13384e57b33e2003d9736dfa1f491e76f938
> Cr-Commit-Position: refs/heads/master@{#10184}

TBR=sprang@webrtc.org,pbos@webrtc.org,mflodman@webrtc.org,asapersson@webrtc.org,ivica@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1383283005

Cr-Commit-Position: refs/heads/master@{#10185}
2015-10-06 18:34:14 +00:00
ivica
092b13384e Collecting encode_time_ms for each frame.
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}
2015-10-06 14:13:50 +00:00
sprang
af4ced986b Transport sequence number should be set also for retransmissions.
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}
2015-10-06 13:02:57 +00:00
Peter Boström
5d0379da2c Remove kSkipFrame usage.
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}
2015-10-06 12:05:03 +00:00