Commit Graph

6455 Commits

Author SHA1 Message Date
ee2bf41a79 Update build files to use webrtc_overrides in Chromium instead of overrides.
This re-lands https://codereview.webrtc.org/1354933002/

This is a part of moving the overrides to Chromium. See bug comment #65 for all steps.

Depends on https://codereview.chromium.org/1345873004/

BUG=chromium:468375

Review URL: https://codereview.webrtc.org/1345313004

Cr-Commit-Position: refs/heads/master@{#10115}
2015-09-30 10:49:11 +00:00
6ba8e4a4f2 ACM: Remove a few local enums that were no longer used
BUG=webrtc:3520
R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1375863002 .

Cr-Commit-Position: refs/heads/master@{#10114}
2015-09-30 08:59:36 +00:00
d094c04baf Remove AgcManager.
It was not used anywhere.

R=andrew@webrtc.org

Review URL: https://codereview.webrtc.org/1299143003 .

Cr-Commit-Position: refs/heads/master@{#10113}
2015-09-29 22:45:23 +00:00
38778b046f Add unit test for nack bandwidth constraint.
BUG=

Review URL: https://codereview.webrtc.org/1341743002

Cr-Commit-Position: refs/heads/master@{#10111}
2015-09-29 16:48:30 +00:00
98db68fdaa If gather_continually is set to true, keep the last port allocator session running while stopping all existing process of getting ports (when p2ptransportchannel first becomes writable).
BUG=5034

Review URL: https://codereview.webrtc.org/1359363003

Cr-Commit-Position: refs/heads/master@{#10110}
2015-09-29 14:58:26 +00:00
86fd9ed6f9 Set RtcpSender transport at construction.
BUG=

Review URL: https://codereview.webrtc.org/1365043002

Cr-Commit-Position: refs/heads/master@{#10106}
2015-09-29 11:45:51 +00:00
092508a5c5 Fix bug in ramp-up tests stats where rtx was accounted for in the media ssrc.
BUG=chromium:536941

Review URL: https://codereview.webrtc.org/1380483002

Cr-Commit-Position: refs/heads/master@{#10104}
2015-09-29 09:26:50 +00:00
fb9e76369d Remove last use of ACMAMRPackingFormat
It was no-op used in FileRecorder.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1360173003

Cr-Commit-Position: refs/heads/master@{#10102}
2015-09-29 06:30:31 +00:00
d6024e3c34 Roll chromium_revision 310ea93..8cf53d6 (349094:351112)
Our perf test suite webrtc_perf_tests timed out, which caused most
of the delay landing this (https://crbug.comn/535973 and
https://codereview.chromium.org/1370133004).

Other problems with executing Android tests also needed to be
resolved in order to land this (http://crbug.com/534849).

Libvpx has moved from third_party/libvpx to third_party/libvpx_new
as of https://codereview.chromium.org/1323333002/

Android GN was blocking this roll due to a problem that ended up
being caused by a bug (http://crbug.com/534849).

Relevant changes:
* src/buildtools: f7310ee..8d89c1b
* src/third_party/boringssl/src: 1d128f3..4c60d35
* src/third_party/icu: 6b3ce81..423fc7e
* src/third_party/libjpeg_turbo: 631e2dd..e4e7503
* src/third_party/libvpx: ac1772e..70db223
* src/third_party/libyuv: fcacbfb..62c49dc
* src/tools/gyp: 5d01a8c..01528c7
* src/tools/swarming_client: 77f720b..6e5d2b2
Details: 310ea93..8cf53d6/DEPS

Clang version changed 245965:247874
Details: 310ea93..8cf53d6/tools/clang/scripts/update.sh

BUG=481034, 535973
TBR=marpan@webrtc.org

Review URL: https://codereview.webrtc.org/1355083002

Cr-Commit-Position: refs/heads/master@{#10101}
2015-09-29 04:16:53 +00:00
8108764552 Analyze support in gyp_webrtc
BUG=chromium:482463
TESTED=Manually tested using the JSON files attached to https://code.google.com/p/chromium/issues/detail?id=482463#c2 and:
webrtc/build/gyp_webrtc --analyzer nothing-files.json nothing-files-RESULT.json
webrtc/build/gyp_webrtc --analyzer everything-files.json everything-files-RESULT.json
webrtc/build/gyp_webrtc --analyzer test_support_unittests-files.json test_support_unittests-files-RESULT.json
Then I verified the result-json contained the expected output.

R=phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/1369683004 .

Cr-Commit-Position: refs/heads/master@{#10097}
2015-09-28 19:56:50 +00:00
2d566686a2 Unify Transport and newapi::Transport interfaces.
BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1369263002

Cr-Commit-Position: refs/heads/master@{#10096}
2015-09-28 16:59:36 +00:00
8387c5f449 Remove AMR format parameter from AudioCoder in utility
The parameter was never used.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1365283002

Cr-Commit-Position: refs/heads/master@{#10095}
2015-09-28 16:24:56 +00:00
1968d3f357 Simplify VCMTimestampMap.
Fixes code formatting and uses size_t properly. Also makes use of
IsNewerTimestamp instead of a simple > check, which should fix an
edge-case bug.

BUG=
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1358863002

Cr-Commit-Position: refs/heads/master@{#10094}
2015-09-28 15:55:52 +00:00
8c404fab8d When doing DisableEquivalentPhases, exclude those AllocationSequences
whose network has ever been removed. It is unlikely the sockets/ports/candidates created from
those AllocationSequences will still be valid.

BUG=

Review URL: https://codereview.webrtc.org/1361183004

Cr-Commit-Position: refs/heads/master@{#10093}
2015-09-28 14:59:50 +00:00
1f429e3418 Passing the new policy from PeerConnection RTCConfiguration to
p2ptransportchannel.  This CL does not use the new policy yet.
BUG=

Review URL: https://codereview.webrtc.org/1369773003

Cr-Commit-Position: refs/heads/master@{#10092}
2015-09-28 14:57:39 +00:00
4b808eee85 ACM: Remove unused and deprecated types
None of these were used.

BUG=webrtc:3520
R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1364703007 .

Cr-Commit-Position: refs/heads/master@{#10090}
2015-09-28 13:52:56 +00:00
1bd0e03ce5 ACM: Removing runtime APIs related to playout mode
The playout mode in NetEq can still be set through the constructor
configuration.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1362943004

Cr-Commit-Position: refs/heads/master@{#10089}
2015-09-28 13:12:21 +00:00
d417523194 Minor fix for debug logging on Android
BUG=NONE

Review URL: https://codereview.webrtc.org/1372873002

Cr-Commit-Position: refs/heads/master@{#10088}
2015-09-28 11:50:18 +00:00
4fbd145dce Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-side.
In addition to this the ramp-up tests are refactored to use a receive call instead of only a remote bitrate estimator, and to make use of BaseTest.

BUG=webrtc:4836

Review URL: https://codereview.webrtc.org/1368943002

Cr-Commit-Position: refs/heads/master@{#10087}
2015-09-28 10:57:23 +00:00
82d6f2a3f7 ACM: Remove ACMVQMonCallback object
It was never used, and the underlying functionality was removed long
ago.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1365193003 .

Cr-Commit-Position: refs/heads/master@{#10083}
2015-09-28 08:25:33 +00:00
69984f0533 Fixes logging levels in WebRtcAudioXXX.java classes
BUG=NONE
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1363673005 .

Cr-Commit-Position: refs/heads/master@{#10082}
2015-09-28 07:24:16 +00:00
d6d27e7340 Update isolate.gypi to support Swarming + move .isolate files
This updates the isolate.gypi copies we have to maintain in our
code repo to Chromium's revision 310ea93.
The changes about generating .isolated.gen.json files are needed
to support running with Swarming (https://www.chromium.org/developers/testing/isolated-testing)

Since isolated testing is now using a new launch script
in tools: isolate_driver.py, that's added to our links
script.

In order to use isolate_driver.py, the .isolate files must be in the
same directory as the test_name_run target is defined, which meant
I had to move around some of the isolate files and targets below
webrtc/modules.

BUG=497757
R=maruel@chromium.org
TBR=henrik.lundin@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org
TESTED=Clobbered trybots:
git cl try -c --bot=linux_compile_rel --bot=mac_compile_rel --bot=win_compile_rel --bot=android_compile_rel --bot=ios_rel -m tryserver.webrtc

Review URL: https://codereview.webrtc.org/1373513002 .

Cr-Commit-Position: refs/heads/master@{#10081}
2015-09-25 20:19:21 +00:00
c97be6a741 Disable TestUdpReadyToSendIPv4 under MSan.
It has become extra flaky lately, and is preventing people from
using the CQ.

BUG=webrtc:4958

Review URL: https://codereview.webrtc.org/1368763002

Cr-Commit-Position: refs/heads/master@{#10080}
2015-09-25 18:00:54 +00:00
4d47aa335c Fallback to system log when webrtc tracing not enabled.
BUG=
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1368053002 .

Cr-Commit-Position: refs/heads/master@{#10079}
2015-09-25 17:04:36 +00:00
1741770742 Implement a high-QP threshold for Android H.264.
Android hardware H.264 seems to keep a steady high-QP flow instead of
dropping frames, so framedrops aren't sufficient to detect a bad state
where downscaling would be beneficial.

BUG=webrtc:4968
R=magjed@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1364253002 .

Cr-Commit-Position: refs/heads/master@{#10078}
2015-09-25 15:03:37 +00:00
a323fd66de Removes Nexus 6 from OpenSL ES blacklist.
BUG=b/1370703002
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1370703002 .

Cr-Commit-Position: refs/heads/master@{#10077}
2015-09-25 14:25:40 +00:00
702f39726b GN: Do not use forward_dependent_configs_from variable.
It is deprecated and public_deps should be used instead, which will have
the
same effect.

BUG=None
R=brettw@chromium.org,kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1365063004

Cr-Commit-Position: refs/heads/master@{#10074}
2015-09-25 12:57:47 +00:00
5c389d3e09 Split webrtc/video into webrtc/{audio,call,video}.
Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.

BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1227923005 .

Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 11:58:39 +00:00
82e20554cb Modifies invalid DCHECK in AudioRecordJni::OnCacheDirectBufferAddress()
Ensures that we can restart audio recording on Android without hitting
a DCHECK. Also adds a symmetric design for the playout side.

BUG=webrtc:5000
TEST=modules_unittests --gtest_filter=AudioDevice*

Review URL: https://codereview.webrtc.org/1373443003

Cr-Commit-Position: refs/heads/master@{#10072}
2015-09-25 11:26:19 +00:00
3fd7be4cb1 Revert of Don't link with audio codecs that we don't use (patchset #4 id:60001 of https://codereview.webrtc.org/1349393003/ )
Reason for revert:
Breaking Chromium FYI bots.

Original issue's description:
> Don't link with audio codecs that we don't use
>
> We used to link with all audio codecs unconditionally (except Opus);
> this patch makes gyp and gn only link to the ones that are used.
>
> (This unfortunately fails to have a measurable impact on Chromium
> binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
> fix were already being excluded from Chromium by some other means
> (likely just the linker omitting compilation units with no incoming
> references).)
>
> BUG=webrtc:4557
>
> Committed: https://crrev.com/f66a9251424351ea6d631c54dd1feb64cc13d809
> Cr-Commit-Position: refs/heads/master@{#10046}

TBR=henrik.lundin@webrtc.org,tina.legrand@webrtc.org,kjellander@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1368933002

Cr-Commit-Position: refs/heads/master@{#10069}
2015-09-25 08:36:11 +00:00
a53e383d7d Revert of CodecOwner: Don't look at definitions for classes we don't link with (patchset #1 id:1 of https://codereview.webrtc.org/1364233002/ )
Reason for revert:
Breaking Chromium FYI bots.

Original issue's description:
> CodecOwner: Don't look at definitions for classes we don't link with
>
> It's good hygiene and just generally the right thing to do. And
> apparently at least sometimes required by Microsoft's compiler.
>
> Committed: https://crrev.com/f4d38ea57aa739b525066b095468cb4af1d2799b
> Cr-Commit-Position: refs/heads/master@{#10060}

TBR=henrik.lundin@webrtc.org,tommi@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1368083002

Cr-Commit-Position: refs/heads/master@{#10068}
2015-09-25 08:31:01 +00:00
67e0cf15d3 Android AppRTCDemo: Add slider for changing camera capture quality during call
This CL adds a slider that can change capture resolution and fps during a call. The camera will no be reconfigured, but the frames will be downscaled/dropped in software by cricket::VideoAdapter in the cricket::VideoCapturer. This is controlled with VideoCapturerAndroid.onOutputFormatRequest(). The slider is turned off by default and can be enabled with a checkbox under 'WebRTC Video Settings'.

R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1361083002 .

Cr-Commit-Position: refs/heads/master@{#10067}
2015-09-25 06:23:49 +00:00
574d5daa6d CodecOwner::SetEncoders: Return error code when given bad arguments
Instead of FATAL on a bad codec specification, log and return an error
code. This is a band-aid until callers are taught to only give it good
specifications.

BUG=webrtc:5033, chromium:526478

Review URL: https://codereview.webrtc.org/1364193002

Cr-Commit-Position: refs/heads/master@{#10066}
2015-09-25 05:54:00 +00:00
4ba059d218 Remove custom handler since the logger already logs to console by default.
BUG=
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1367523009 .

Cr-Commit-Position: refs/heads/master@{#10064}
2015-09-24 21:36:17 +00:00
8937437872 Do not prune if the current best connection is weak.
Otherwise, we may delete a useful connection because the current best connection may be failing.

BUG=

Review URL: https://codereview.webrtc.org/1364683002

Cr-Commit-Position: refs/heads/master@{#10063}
2015-09-24 20:14:51 +00:00
59e72ab49b Enable logging for Mac by default on debug builds.
Was previously using the wrong preprocessor define (DEBUG vs _DEBUG).

Review URL: https://codereview.webrtc.org/1361173002

Cr-Commit-Position: refs/heads/master@{#10061}
2015-09-24 17:42:47 +00:00
f4d38ea57a CodecOwner: Don't look at definitions for classes we don't link with
It's good hygiene and just generally the right thing to do. And
apparently at least sometimes required by Microsoft's compiler.

Review URL: https://codereview.webrtc.org/1364233002

Cr-Commit-Position: refs/heads/master@{#10060}
2015-09-24 17:21:02 +00:00
a58ea7806a 1. Add receiving state as part of the connection sorting criteria. So if a connection's receiving state changes, it will re-select a better connection if there is any.
This will paves the way for continuous nomination lite and multi-networking.
2. Combined checking and pinging to remove some redundant checking and to make it switch to more frequent ping mode earlier.

Review URL: https://codereview.webrtc.org/1311433009

Cr-Commit-Position: refs/heads/master@{#10057}
2015-09-24 15:13:45 +00:00
8a88dd271e Stability improvement for audio recording on Android
BUG=NONE
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1363323002 .

Cr-Commit-Position: refs/heads/master@{#10056}
2015-09-24 14:45:14 +00:00
7076729c57 Enable SurfaceViewRenderer for AppRTCDemo
BUG=webrtc:4742,webrtc:4910,webrtc:4909
R=glaznev@webrtc.org, perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1356603004 .

Cr-Commit-Position: refs/heads/master@{#10054}
2015-09-24 14:02:15 +00:00
9236bb1e08 Minor fix for improving logging of supported platform effects
BUG=NONE
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1370443003 .

Cr-Commit-Position: refs/heads/master@{#10053}
2015-09-24 13:58:46 +00:00
6b8d355168 Reland "Wire up send-side bandwidth estimation."
Revert was patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/

The culprit was RTC_DCHECK(poller_thread_->Start()); in rampup_test.cc

BUG=webrtc:4173
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1362303002 .

Cr-Commit-Position: refs/heads/master@{#10052}
2015-09-24 13:07:17 +00:00
8c266e6baf H264 bitstream parser.
Parsing the encoded bitstream is required for doing downscaling
decisions based on average encoded QP to improve perceived quality.

BUG=webrtc:4968
R=noahric@chromium.org, stefan@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1314473008 .

Cr-Commit-Position: refs/heads/master@{#10051}
2015-09-24 13:07:04 +00:00
ec249d4eae ACMCodecDB: Remove unused stuff, and move private stuff to anonymous namespace
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1360123002

Cr-Commit-Position: refs/heads/master@{#10048}
2015-09-24 11:32:11 +00:00
f66a925142 Don't link with audio codecs that we don't use
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.

(This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means
(likely just the linker omitting compilation units with no incoming
references).)

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1349393003

Cr-Commit-Position: refs/heads/master@{#10046}
2015-09-24 10:18:48 +00:00
c675ddd830 video_capture: Better support for UYVY
A couple of places were missing handling of UYVY video formats.

BUG=webrtc:4816

Review URL: https://codereview.webrtc.org/1317613003

Cr-Commit-Position: refs/heads/master@{#10044}
2015-09-24 08:11:45 +00:00
74d85e19ae Reduce locking in overuse frame detector now that (as of r9508) the observer_ and options_ can only be set at construction time. E.g. no lock is any longer held while doing the callback.
BUG=

Review URL: https://codereview.webrtc.org/1228853004

Cr-Commit-Position: refs/heads/master@{#10043}
2015-09-24 07:53:38 +00:00
53eee43e78 Address the comment from 1367553002.
Remove duplication introduced by
https://codereview.webrtc.org/1367553002

BUG=webrtc:5030
TBR=juberti@webrtc.org

Review URL: https://codereview.webrtc.org/1360203003 .

Cr-Commit-Position: refs/heads/master@{#10039}
2015-09-23 21:09:18 +00:00
2e4b620471 TcpPort doesn't connect when calling gmail with non-proxied UDP disabled.
The same check has been made into turnport.cc but missed this place.

BUG=webrtc:5030
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1367553002 .

Cr-Commit-Position: refs/heads/master@{#10038}
2015-09-23 20:57:17 +00:00
cdfe20bfc1 Fix the maximum native sample rate in AudioProcessing
BUG=webrtc:4983
R=andrew@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1338833002 .

Cr-Commit-Position: refs/heads/master@{#10037}
2015-09-23 19:49:21 +00:00