7477 Commits

Author SHA1 Message Date
aleloi
d700bef583 Added new mixer to |check_targets| in .gn and fixed include/depend errors.
Also fixed one small chromium-style error in the new mixer.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2234293002
Cr-Commit-Position: refs/heads/master@{#13752}
2016-08-15 14:24:13 +00:00
kwiberg
963be23e62 RtpRtcp: Remove the SetSendREDPayloadType and SendREDPayloadType methods
The last in-tree call site recently disappeared, so they were unused.

BUG=webrtc:5922

Review-Url: https://codereview.webrtc.org/2066473002
Cr-Commit-Position: refs/heads/master@{#13751}
2016-08-15 14:08:39 +00:00
hbos
8f956dead6 FakeTiming added, an implementation of Timing that can be used for
tests.

Note: The webrtc/base/test/ folder is new.

Currently not used, I intend to use this in another CL.

BUG=chromium:627816
NOPRESUBMIT=TRUE
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2238073003
Cr-Commit-Position: refs/heads/master@{#13750}
2016-08-15 14:00:06 +00:00
peah
96dbc8f4b4 Adding comment regarding the disabling the flaky test VolumeTest.ManualRequiresMicrophoneCanSetMicrophoneVolumeWithAgcOff
NOTRY=True
TBR=henrika@webrtc.org
BUG=webrtc:6206

Review-Url: https://codereview.webrtc.org/2247733002
Cr-Commit-Position: refs/heads/master@{#13749}
2016-08-15 13:38:56 +00:00
stefan
3ab6614d10 Add video_loopback to gn.
BUG=webrtc:5949
NOTRY=True

Review-Url: https://codereview.webrtc.org/2236473002
Cr-Commit-Position: refs/heads/master@{#13748}
2016-08-15 13:29:19 +00:00
sakal
92c09509bd Make CameraCapturer.switchCamera try again if session is still opening.
R=magjed@webrtc.org

Review-Url: https://codereview.webrtc.org/2238263002
Cr-Commit-Position: refs/heads/master@{#13747}
2016-08-15 13:19:40 +00:00
peah
d7d05f8056 Disabling the test VolumeTest.ManualRequiresMicrophoneCanSetMicrophoneVolumeWithAgcOff
as it has been found to be flaky.

NOTRY=True
TBR=henrika@webrtc.org
BUG=webrtc:6206

Review-Url: https://codereview.webrtc.org/2248633003
Cr-Commit-Position: refs/heads/master@{#13746}
2016-08-15 13:13:21 +00:00
kthelgason
3e3ebe6937 remove unnecessary double allocation
BUG=

Review-Url: https://codereview.webrtc.org/2226933005
Cr-Commit-Position: refs/heads/master@{#13744}
2016-08-15 10:42:08 +00:00
kwiberg
0ccff57024 VoERTP_RTCP: Remove GetREDStatus and SetREDStatus
They always fail, because RED isn't supported.

BUG=webrtc:5922

Review-Url: https://codereview.webrtc.org/2055753002
Cr-Commit-Position: refs/heads/master@{#13743}
2016-08-15 10:34:52 +00:00
aleloi
5bcc00e538 Changed folder structure in new mixer and fixed simple lint errors.
The folder structure is now as was agreed on in the 'Slim and Modular
WebRTC' effort.  Also added some dependencies that were previously in
another part of the tree.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2238803002
Cr-Commit-Position: refs/heads/master@{#13742}
2016-08-15 10:01:37 +00:00
sakal
714dd4e532 GN: Update tests to have the correct shard timeout value on Android.
TBR=mflodman@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2231413002
Cr-Commit-Position: refs/heads/master@{#13741}
2016-08-15 09:29:19 +00:00
asapersson
5093b38497 Make variable for selecting if intervals without samples should be included in stats configurable (for rate counters).
BUG=

Review-Url: https://codereview.webrtc.org/2236923002
Cr-Commit-Position: refs/heads/master@{#13740}
2016-08-15 08:20:37 +00:00
Taylor Brandstetter
4cb5b64b16 Fix for data channels perpetually stuck in "closing" state.
If the data transport is destroyed while data is buffered (due to
the PC being closed, or a description set with data rejected), the
data channel was getting stuck in a "closing" state, waiting to
finish sending its buffered data. But since there's no more transport,
it will never get another chance to send buffered data.

It just needs to terminate non-gracefully and discard the buffered data
in this situation.

R=skvlad@webrtc.org, zhihuang@webrtc.org

Review URL: https://codereview.webrtc.org/2235843003 .

Cr-Commit-Position: refs/heads/master@{#13737}
2016-08-12 17:10:42 +00:00
flim
64a7eab891 Update tests and DTX check for Opus 1.1.3.
DTX is now indicated by packets that may have a size of up to 2 bytes.
Ref: https://git.xiph.org/?p=opus.git;a=commit;h=1c311423c86b89eba27a494e17c79fefd7d75ab0

BUG=

Review-Url: https://codereview.webrtc.org/2158293003
Cr-Commit-Position: refs/heads/master@{#13736}
2016-08-12 11:36:14 +00:00
sakal
9591e3e82d Convert PeerConnectionTest to use the new capture APIs.
Review-Url: https://codereview.webrtc.org/2236323002
Cr-Commit-Position: refs/heads/master@{#13735}
2016-08-12 07:06:22 +00:00
Taylor Brandstetter
62351c9923 Fixing problems with ICE candidate pair prioritization.
The main issue was that upon receiving a binding response with a srflx
mapped address attribute, the local candidate was not updated from local
to srflx. This means the two ICE agents view the same pair differently;
one sees it as "X<->srflx" while the other sees it as "local<->X". This
causes sub-optimal prioritization and could result in the wrong pair
being selected if using aggressive nomination.

The other issue was that TCP prflx candidates were not differentiated from
UDP prflx candidates. This lead to TCP prflx candidates prioritized above TCP
host candidates.

After fixing these issues, I was able to re-enable many disabled tests, as well
as restore the check for the candidate types of the controlled agent.

BUG=webrtc:1953,webrtc:2383
R=honghaiz@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2125823004 .

Cr-Commit-Position: refs/heads/master@{#13734}
2016-08-11 23:05:15 +00:00
Taylor Brandstetter
6f82535f45 Enabling IPv6 socket recv timestamp test, and making less flaky.
The test worked by sleeping a certain time, then checking that the
difference between recv timestamps before and after the sleep was
within some margin of the requested sleep time.

However, this means that imprecision of SleepMs makes the test flaky.
This source of flakiness can be removed by comparing to the actual
time slept instead of the requested time.

Also making the margin larger, to further reduce the likelihood of
flakiness.

R=pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2111043004 .

Cr-Commit-Position: refs/heads/master@{#13733}
2016-08-11 22:38:40 +00:00
skvlad
588783adcd Return nil from RTCPeerConnectionFactory when creation fails
RTCPeerConnectionFactory.createPeerConnection did not check the return
value of the native createPeerConnection() - so when the native PC
fails to be created, it could end up attempting to use a null pointer.

The change makes it return nil when the creation fails. The application
can then detect and respond to the failure.

Review-Url: https://codereview.webrtc.org/2240633004
Cr-Commit-Position: refs/heads/master@{#13732}
2016-08-11 21:29:32 +00:00
johan
fe1ffb141b Remove unused SessionId from TransportChannel and PortAllocatorSession.
BUG=

Review-Url: https://codereview.webrtc.org/2237853002
Cr-Commit-Position: refs/heads/master@{#13731}
2016-08-11 19:37:51 +00:00
Taylor Brandstetter
c8762a838f Remove StartSSLWithServer from SSLStreamAdapter.
It's not used by anything any more. We only use SSLStreamAdapter in
the mode where it verifies the peer's certificate using a signaled
digest.

R=pthatcher@webrtc.org, zhihuang@webrtc.org

Review URL: https://codereview.webrtc.org/2204883004 .

Cr-Commit-Position: refs/heads/master@{#13730}
2016-08-11 19:01:58 +00:00
asapersson
2e5cfcd6c2 Add periodic logging of video stats.
Add ToString method to: Call::Stats, VideoSendStream::Stats, VideoReceiveStream::Stats and log stats periodically (every 10 seconds).

BUG=

Review-Url: https://codereview.webrtc.org/2133073002
Cr-Commit-Position: refs/heads/master@{#13727}
2016-08-11 15:41:26 +00:00
stefan
b17976763d Add an HD resolution perf test.
Also update existing perf tests to use send side bwe.

BUG=webrtc:4604, chromium:522001

Review-Url: https://codereview.webrtc.org/2227733004
Cr-Commit-Position: refs/heads/master@{#13726}
2016-08-11 14:01:03 +00:00
philipel
17deeb47ed PacketBuffer is now ref counted.
Since all FrameObjects have a reference to its PacketBuffer and since
the PacketBuffer can be thrown away at any moment the PacketBuffer
has to be ref counted in order to avoid FrameObjects dereferencing a potentially
destroyed object.

BUG=webrtc:5514
R=danilchap@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2199133004 .

Cr-Commit-Position: refs/heads/master@{#13725}
2016-08-11 13:09:40 +00:00
ehmaldonado
a3a1fde020 Add Mac bots to MB.
BUG=589510
NOTRY=True

Review-Url: https://codereview.webrtc.org/2232643003
Cr-Commit-Position: refs/heads/master@{#13724}
2016-08-11 08:04:39 +00:00
sakal
d30e0ad3e7 Session based capturing for Camera2Capturer.
BUG=webrtc:6148
TBR=tommi@webrtc.org

Review-Url: https://codereview.webrtc.org/2186253002
Cr-Commit-Position: refs/heads/master@{#13723}
2016-08-11 08:00:43 +00:00
sakal
bd59c71ff8 GN: Add dependency libjingle_peerconnection_java to modules_unittests.
This change only affects the Android build.

BUG=webrtc:6189
TBR=tommi@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2233903002
Cr-Commit-Position: refs/heads/master@{#13722}
2016-08-11 07:59:23 +00:00
kjellander
0ae7878bbc MB: Add Windows configurations
BUG=589510, 636414
NOTRY=True

Review-Url: https://codereview.webrtc.org/2235833002
Cr-Commit-Position: refs/heads/master@{#13721}
2016-08-11 07:49:55 +00:00
Honghai Zhang
3d31bd65cf Do not create incompatible TurnPort if the server address family is known.
In the existing code, if the server address and the local IP family does not
match, we still create a TurnPort and destroy it later.
Instead, we could avoid creating it at the beginning.
This does not affect the client behavior except for the port creation.

BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org, zhihuang@webrtc.org

Review URL: https://codereview.webrtc.org/2206713004 .

Cr-Commit-Position: refs/heads/master@{#13720}
2016-08-10 17:33:17 +00:00
philipel
bf8a2c94ce Probe bitrate estimator correction.
Since the interval between the timestamps does not include the send/receive
time of the last/first packet we correct the interval by adding the average
of the interval between probing packets.

BUG=webrtc:5859
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2224173003 .

Cr-Commit-Position: refs/heads/master@{#13719}
2016-08-10 17:00:52 +00:00
kjellander
68815bf977 MB: Make all Android debug builds static
In https://codereview.webrtc.org/2228923002 all Android debug
bots were configured to build with component=shared_library.
This is not the behavior we had before:
https://cs.chromium.org/chromium/build/scripts/slave/recipe_modules/chromium/config.py?rcl=0&l=713
This CL changes them back to static library.

BUG=589510
NOTRY=True
TBR=ehmaldonado@webrtc.org

Review-Url: https://codereview.webrtc.org/2236573002
Cr-Commit-Position: refs/heads/master@{#13718}
2016-08-10 16:49:45 +00:00
stefan
c99d5a6132 Add stefan@ to webrtc/OWNERS.
NOTRY=true

Review-Url: https://codereview.webrtc.org/2224353002
Cr-Commit-Position: refs/heads/master@{#13717}
2016-08-10 15:49:59 +00:00
kjellander
63cb17294c MB: Fix typo for android_arm64_rel trybot.
This was missed in review of https://codereview.webrtc.org/2228923002

BUG=589510
NOTRY=True
TBR=ehmaldonado@webrtc.org

Review-Url: https://codereview.webrtc.org/2233913002
Cr-Commit-Position: refs/heads/master@{#13716}
2016-08-10 15:28:32 +00:00
magjed
fb372f0074 iOS render: Handle frame rotation in OpenGL
This CL handles frame rotation by updating the OpenGL vertex data in
RTCOpenGLVideoRenderer, instead of calling the expensive
libyuv::I420Rotate that will rotate the actual memory. Also, we can
handle rotated native frames instead of falling back to
NativeToI420Buffer.

Review-Url: https://codereview.webrtc.org/2176623002
Cr-Commit-Position: refs/heads/master@{#13715}
2016-08-10 14:58:35 +00:00
ehmaldonado
4556b45076 Fix tools_unittests in GN.
tools_unittests were failing when generated by GN and running on a device.

BUG=webrtc:5949
NOTRY=True

Review-Url: https://codereview.webrtc.org/2233843002
Cr-Commit-Position: refs/heads/master@{#13714}
2016-08-10 14:47:20 +00:00
terelius
ccbbf8da38 Visualize delay changes based on both abs-send-time and capture time.
Adds reusable template function objects to extract interesting
statistics from data sets. A few more of these will be added later
to reduce the code size.

Review-Url: https://codereview.webrtc.org/2220383004
Cr-Commit-Position: refs/heads/master@{#13713}
2016-08-10 14:34:37 +00:00
terelius
d49a37b1d9 Rename main file for visualization tool.
Review-Url: https://codereview.webrtc.org/2226273002
Cr-Commit-Position: refs/heads/master@{#13712}
2016-08-10 14:12:55 +00:00
aleloi
c4ac700781 Migrated GN target :neteq_pcmu_quality_test
Migrated GN target :neteq_pcmu_quality_test from
webrtc/modules/audio_coding/neteq/neteq_tests.gypi

NOTRY=True

BUG=webrtc:6190

Review-Url: https://codereview.webrtc.org/2224313002
Cr-Commit-Position: refs/heads/master@{#13711}
2016-08-10 12:06:31 +00:00
aleloi
6df36dc054 Migrated GN target :neteq_isac_quality_test
Migrated GN target :neteq_isac_quality_test from
webrtc/modules/audio_coding/neteq/neteq_tests.gypi

NOTRY=True

BUG=webrtc:6190

Review-Url: https://codereview.webrtc.org/2226253002
Cr-Commit-Position: refs/heads/master@{#13710}
2016-08-10 12:04:51 +00:00
aleloi
e6ca9ec7b9 Broke out 'level_indicator' of the voice_engine GN target. This is
needed for the new audio_mixer to depend on just
'voice_engine:level_indicator' instead of the whole voice_engine and
avoid a circular dependency.

TBR=henrika@webrtc.org

NOTRY=True

Review-Url: https://codereview.webrtc.org/2221103004
Cr-Commit-Position: refs/heads/master@{#13709}
2016-08-10 12:01:59 +00:00
aleloi
0e0be0a2f5 Migrated GN target :neteq_ilbc_quality_test
Migrated GN target :neteq_ilbc_quality_test from
webrtc/modules/audio_coding/neteq/neteq_tests.gypi

NOTRY=True

BUG=webrtc:6190

Review-Url: https://codereview.webrtc.org/2221193004
Cr-Commit-Position: refs/heads/master@{#13708}
2016-08-10 11:55:29 +00:00
Magnus Jedvert
6e6e70f55b Add magjed@webrtc.org as owner of webrtc/base/java/
magjed@ is the author of two out of three files in webrtc/base/java
(ThreadUtils.java and Size.java). They were originally created in
webrtc/api/android/java, but were moved to base/ in order to be
accessible from e.g. webrtc/modules/audio_device/android/java.

TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2233793002 .

Cr-Commit-Position: refs/heads/master@{#13707}
2016-08-10 11:49:35 +00:00
aleloi
6391012514 Migrated GN target :audio_classifier_test
Migrated GN target :audio_classifier_test from
webrtc/modules/audio_coding/neteq/neteq_tests.gypi

NOTRY=True

BUG=webrtc:6190

Review-Url: https://codereview.webrtc.org/2225333002
Cr-Commit-Position: refs/heads/master@{#13706}
2016-08-10 11:41:23 +00:00
Danil Chapovalov
bcdad0f3a8 Generate random rtp packets with RtpPacketToSend instead of RtpSender
in rtc event log unittest

R=stefan@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/2228493002 .

Cr-Commit-Position: refs/heads/master@{#13705}
2016-08-10 11:23:38 +00:00
aleloi
116fd61599 Migrated GN target :neteq_speed_test
Migrated GN target :neteq_speed_test from
webrtc/modules/audio_coding/neteq/neteq.gypi

NOTRY=True

BUG=webrtc:6190

Review-Url: https://codereview.webrtc.org/2223343004
Cr-Commit-Position: refs/heads/master@{#13704}
2016-08-10 11:16:45 +00:00
ehmaldonado
3651d8f97f Add libjingle_peerconnection_unittest to GN targets.
BUG=webrtc:6037
NOTRY=True

Review-Url: https://codereview.webrtc.org/2226093003
Cr-Commit-Position: refs/heads/master@{#13703}
2016-08-10 10:10:53 +00:00
Danil Chapovalov
daa90a7e35 Cleaned out boundingSet member from TMMBRHelp class
BUG=webrtc:5565
R=philipel@webrtc.org

Review URL: https://codereview.webrtc.org/2224933002 .

Cr-Commit-Position: refs/heads/master@{#13702}
2016-08-10 09:30:08 +00:00
aleloi
630c6d58d3 Migrated GN target :neteq_opus_quality_test
Migrated GN target :neteq_opus_quality_test from
webrtc/modules/audio_coding/neteq/neteq_tests.gypi

NOTRY=True

BUG=webrtc:6190, webrtc:2692

Review-Url: https://codereview.webrtc.org/2223933004
Cr-Commit-Position: refs/heads/master@{#13701}
2016-08-10 09:11:49 +00:00
brandtr
74811e5fa3 Style updates to ProducerFec/FecReceiver.
- Make more use of std::unique_ptr.
- Auto type deduction for iterator type names.
- More extensive comments.
- Variable renaming.
- Make ProducerFec::BuildRedPacket() static.
- Avoid dynamic allocation of ProducerFec::fec_.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2110763002
Cr-Commit-Position: refs/heads/master@{#13700}
2016-08-10 07:51:59 +00:00
kjellander
5d6b84c484 MB: Set crosscompilation config for Linux ARM bot.
This apparently was missed as well for our initial Linux configs,
thus failing compilation.

BUG=589510
NOTRY=True
TBR=ehmaldonado@webrtc.org

Review-Url: https://codereview.webrtc.org/2228123003
Cr-Commit-Position: refs/heads/master@{#13699}
2016-08-09 20:17:48 +00:00
tkchin
4c45234b81 Fix regression for h264 VideoToolbox entering background.
We need to check the pool to make sure that the encoder session is valid. Otherwise, it appears as if the encoder just does not output frames.

NOTRY=True
BUG=

Review-Url: https://codereview.webrtc.org/2226383002
Cr-Commit-Position: refs/heads/master@{#13698}
2016-08-09 20:02:03 +00:00