Commit Graph

13158 Commits

Author SHA1 Message Date
72333d2ca0 Add kjellander@webrtc.org to more BUILD.gn OWNERS files.
NOTRY=True

Review-Url: https://codereview.webrtc.org/2258983003
Cr-Commit-Position: refs/heads/master@{#13826}
2016-08-19 07:48:39 +00:00
96b6b8336a iOS: add type to peer connection local streams
BUG=

Review-Url: https://codereview.webrtc.org/2249173002
Cr-Commit-Position: refs/heads/master@{#13825}
2016-08-18 21:21:27 +00:00
c21560b3e1 Remove pbos@webrtc.org from autoroll TBRs.
BUG=
TBR=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/2259933002 .

Cr-Commit-Position: refs/heads/master@{#13824}
2016-08-18 19:10:49 +00:00
9b5306c4ef Adding test for unordered, fragmented SCTP message delivery.
This functionality broke after a recent usrsctp roll. This test would be
useful in catching issues that arise in the future.

BUG=633959
R=honghaiz@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2233033002 .

Cr-Commit-Position: refs/heads/master@{#13823}
2016-08-18 18:40:45 +00:00
b5b30908dc Corrected the testvectors for the level controller
bitexactness test. The activation of the test will
be done in another CL.

BUG=

Review-Url: https://codereview.webrtc.org/2257733002
Cr-Commit-Position: refs/heads/master@{#13822}
2016-08-18 16:47:52 +00:00
8df4d0e426 Add playout_delay_oracle_unittest as gn target
BUG=

Review-Url: https://codereview.webrtc.org/2256743002
Cr-Commit-Position: refs/heads/master@{#13821}
2016-08-18 14:53:44 +00:00
3a11933a63 Remove audio_device_test_func.
This code does not work and hasn't been used in a long time. It also
lacks a GN target. There's no reason to save it.

BUG=none

Review-Url: https://codereview.webrtc.org/2255173002
Cr-Commit-Position: refs/heads/master@{#13820}
2016-08-18 14:20:48 +00:00
644fa96886 Added recording of the configuration for the AudioFrame API call
BUG=webrtc:6227

Review-Url: https://codereview.webrtc.org/2252043003
Cr-Commit-Position: refs/heads/master@{#13819}
2016-08-18 13:48:38 +00:00
7320866091 Reland of Adding audio to video_quality_test.
The original commit was https://codereview.webrtc.org/2136573002/.

BUG=

Review-Url: https://codereview.webrtc.org/2259783002
Cr-Commit-Position: refs/heads/master@{#13818}
2016-08-18 13:28:59 +00:00
2b616397de Remove TMMBRSet class
by cleaning RTCPReceiveInfo class
and following cleaning of RTCPReceiver::BoundingSet function.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2254703003
Cr-Commit-Position: refs/heads/master@{#13817}
2016-08-18 13:17:48 +00:00
e1f5b4a7fe voice_engine: Removed old variants of Channel constructor and CreateChannel
These are no longer used internally and their interface is not to be
considered public. They were due to be changed in
https://codereview.webrtc.org/1993783002/ but remained due to a
misunderstanding.

Review-Url: https://codereview.webrtc.org/2082483003
Cr-Commit-Position: refs/heads/master@{#13816}
2016-08-18 11:23:04 +00:00
38d840c35a NetEq: Changing checked_cast to saturated_cast
The cast involves packet_len_samp, which is derived from the timestamps
and sequence numbers of incoming packets. Being values from the outside,
these should be treated as if any value is possible, making a
checked_cast unsuitable. Changing instead to saturated_cast to avoid
overflow with out-of-bounds values.

Review-Url: https://codereview.webrtc.org/2243403007
Cr-Commit-Position: refs/heads/master@{#13815}
2016-08-18 10:49:41 +00:00
96bbdd585e WebRtcSpl_SynthesisQMF: Fix UBSan fuzzer bug (left shift of negative value)
BUG=chromium:614033

Review-Url: https://codereview.webrtc.org/2253943002
Cr-Commit-Position: refs/heads/master@{#13814}
2016-08-18 10:17:10 +00:00
e9a6acfbf5 Added missing unittest to the modules/BUILD.gn build file
NOTRY=True

BUG=

Review-Url: https://codereview.webrtc.org/2255093002
Cr-Commit-Position: refs/heads/master@{#13813}
2016-08-18 09:41:51 +00:00
cb2d701946 Add kjellander as BUILD.gn OWNER in webrtc/modules
NOTRY=True

Review-Url: https://codereview.webrtc.org/2258593003
Cr-Commit-Position: refs/heads/master@{#13812}
2016-08-18 09:39:14 +00:00
71fead2146 Reland of StartTimestamp generated randomly in RtpSender constructor (patchset #1 id:1 of https://codereview.webrtc.org/2248413002/ )
Reason for revert:
Reland: downstream code expectation about rtp_sender timestamp adjusted.

Original issue's description:
> Revert of StartTimestamp generated randomly in RtpSender constructor (patchset #4 id:60001 of https://codereview.webrtc.org/2241193002/ )
>
> Reason for revert:
> Breaks downstream code.
>
> Original issue's description:
> > StartTimestamp generated randomly in RtpSender constructor
> > instead of not-randomly at SetSendingState(true)
> > Renamed to timestamp_offset_ to better match meaning of the variable.
> >
> > R=asapersson@webrtc.org, terelius@webrtc.org
> >
> > Committed: https://crrev.com/4466782ae43e1b1125a55ee7e18abd10dd37cede
> > Cr-Commit-Position: refs/heads/master@{#13796}
>
> TBR=asapersson@webrtc.org,terelius@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/86c96948e340cf8b879bddb0c7293f3b5ad4dad4
> Cr-Commit-Position: refs/heads/master@{#13798}

TBR=asapersson@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2257083002
Cr-Commit-Position: refs/heads/master@{#13811}
2016-08-18 09:02:16 +00:00
d4e9f62ea7 Updated AudioDecoderFactory to list AudioCodecSpecs instead of SdpAudioFormats.
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2123923004
Cr-Commit-Position: refs/heads/master@{#13810}
2016-08-18 09:02:15 +00:00
235020dba6 Roll chromium_revision 915e47250f..e3860bd297 (412201:412289)
Change log: 915e47250f..e3860bd297
Full diff: 915e47250f..e3860bd297

No dependencies changed.
No update to Clang.

NOTRY=TRUE
TBR=
BUG=webrtc:6219

Review-Url: https://codereview.webrtc.org/2253973002
Cr-Commit-Position: refs/heads/master@{#13809}
2016-08-18 08:45:53 +00:00
010f092919 GN: Add Android support to video_engine_tests.
R=kjellander@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2246423002
Cr-Commit-Position: refs/heads/master@{#13808}
2016-08-18 07:42:05 +00:00
fd16da290c Do not switch to a high-cost connection that is not receiving.
This prevents connection switching due to remote-side network down.

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

Review URL: https://codereview.webrtc.org/2232563002 .

Cr-Commit-Position: refs/heads/master@{#13807}
2016-08-17 23:12:58 +00:00
41a3287472 Nil out EAGLContext explicitly on RTCEAGLVideoView dealloc.
Theoretical fix to address some EAGLContext issues from other UIImageViews that could be active.

NOTRY=True
BUG=

Review-Url: https://codereview.webrtc.org/2259513002
Cr-Commit-Position: refs/heads/master@{#13806}
2016-08-17 23:03:09 +00:00
869dab775c Disable Intel VP8 HW encoder.
Need to investigate dequeueOutputBuffer failure on Asus
Zenfones before re-enabling back.

BUG=b/30890961
R=jiayl@chromium.org

Review URL: https://codereview.webrtc.org/2249743007 .

Cr-Commit-Position: refs/heads/master@{#13805}
2016-08-17 22:41:22 +00:00
6a35590d14 Add code for dummy file audio to fallback to dummy audio.
BUG=

Review-Url: https://codereview.webrtc.org/2250853002
Cr-Commit-Position: refs/heads/master@{#13804}
2016-08-17 22:19:55 +00:00
7c0f8ee67a Avoid null pointer exception if Android getCameraInfo fails.
BUG=b/30890971
R=magjed@webrtc.org, sakal@webrtc.org

Review URL: https://codereview.webrtc.org/2250283002 .

Cr-Commit-Position: refs/heads/master@{#13803}
2016-08-17 22:18:27 +00:00
d8a72f0ab2 Close input file in FileAudioDevice::StopRecording.
Also added some more logging, to help track down start/stop, start
failure, and the name of the file used.

BUG=

Review-Url: https://codereview.webrtc.org/2253763002
Cr-Commit-Position: refs/heads/master@{#13802}
2016-08-17 22:14:57 +00:00
78810b633c Expose media constraint string constants as ObjC NSStrings
Review-Url: https://codereview.webrtc.org/2252783003
Cr-Commit-Position: refs/heads/master@{#13801}
2016-08-17 18:07:44 +00:00
d22854bf7d FilePlayer: Remove unused default values for arguments
The functions in question were virtual, so we would've wanted to get
rid of the default values even if callers had relied on them.

Review-Url: https://codereview.webrtc.org/2045943004
Cr-Commit-Position: refs/heads/master@{#13800}
2016-08-17 16:27:08 +00:00
4a42900540 Removes redundant log warning in WebRtcAudioManager.
Trivial patch which avoids logs that are of no value.

BUG=NONE

Review-Url: https://codereview.webrtc.org/2250403002
Cr-Commit-Position: refs/heads/master@{#13799}
2016-08-17 15:43:59 +00:00
86c96948e3 Revert of StartTimestamp generated randomly in RtpSender constructor (patchset #4 id:60001 of https://codereview.webrtc.org/2241193002/ )
Reason for revert:
Breaks downstream code.

Original issue's description:
> StartTimestamp generated randomly in RtpSender constructor
> instead of not-randomly at SetSendingState(true)
> Renamed to timestamp_offset_ to better match meaning of the variable.
>
> R=asapersson@webrtc.org, terelius@webrtc.org
>
> Committed: https://crrev.com/4466782ae43e1b1125a55ee7e18abd10dd37cede
> Cr-Commit-Position: refs/heads/master@{#13796}

TBR=asapersson@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2248413002
Cr-Commit-Position: refs/heads/master@{#13798}
2016-08-17 15:12:27 +00:00
5a25d9504a FileRecorder + FilePlayer: Let Create functions return unique_ptr
Because passing ownership in raw pointers makes kittens cry.

This also means we can ditch the Destroy functions and the protected
destructors. (Well, almost. We need to keep the old CreateFilePlayer
and DestroyFilePlayer around for a little while longer because of an
external caller.)

Review-Url: https://codereview.webrtc.org/2049683003
Cr-Commit-Position: refs/heads/master@{#13797}
2016-08-17 14:31:18 +00:00
4466782ae4 StartTimestamp generated randomly in RtpSender constructor
instead of not-randomly at SetSendingState(true)
Renamed to timestamp_offset_ to better match meaning of the variable.

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

Review URL: https://codereview.webrtc.org/2241193002 .

Cr-Commit-Position: refs/heads/master@{#13796}
2016-08-17 13:07:49 +00:00
2ae1fb62f6 Fix get_landmines.py script.
BUG=webrtc:6216
NOTRY=True

Review-Url: https://codereview.webrtc.org/2250343002
Cr-Commit-Position: refs/heads/master@{#13795}
2016-08-17 11:00:47 +00:00
144dd27056 FileRecorderImpl and FilePlayerImpl don't need their own .h and .cc files
They are implementations of interfaces that are only ever exposed
via "create" functions, so the entire class definitions can be put in
anonymous namespaces in the .cc files that defines the "create"
functions.

NOPRESUBMIT=true

Review-Url: https://codereview.webrtc.org/2038513002
Cr-Commit-Position: refs/heads/master@{#13794}
2016-08-17 09:46:57 +00:00
c54071d8ab WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs.
Reland of https://codereview.webrtc.org/2072753002/ which broke
chromium due to how their build was setup. This issue should now be
resolved.

Changed WebRtcVoiceEngine to present receive codecs from the formats
provided by its decoder factory. Added supported formats to
BuiltinAudioDecoderFactory. Added helper functions for creating some
simple decoder factories for mocking.

Created a PayloadTypeMapper for assigning payload types to formats. I
think we'll eventually want to use this further up, or possibly in
both the audio and video sides. It would be best if the engines didn't
have to talk payload types at all, but it might be more difficult to
get around when payload types depend on each-other, like the RTX
codecs for video.

BUG=webrtc:5805
TBR=ivoc@webrtc.org

Review-Url: https://codereview.webrtc.org/2250683002
Cr-Commit-Position: refs/heads/master@{#13793}
2016-08-17 09:45:47 +00:00
a93d5ac019 Don't simulate probing based on rtc event logs since we don't have that info logged.
BUG=webrtc:6217

Review-Url: https://codereview.webrtc.org/2250963002
Cr-Commit-Position: refs/heads/master@{#13792}
2016-08-17 09:14:38 +00:00
eb680eac5d CongestionController::SetBweBitrates may now trigger probing.
BUG=webrtc:5859
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2246403002 .

Cr-Commit-Position: refs/heads/master@{#13791}
2016-08-17 09:12:14 +00:00
c594aa61bc Add a gyp/gn option to use dummy audio file devices.
Conceptually, dummy audio file devices are a "platform", like
win/mac/linux, and so the conditional slots under
include_internal_audio_device. When enabled, use_dummy_audio_file_devices
disables whatever platform-specific audio layer would have been used and
turns on dummy file device support.

BUG=

Review-Url: https://codereview.webrtc.org/2250483002
Cr-Commit-Position: refs/heads/master@{#13790}
2016-08-17 01:21:23 +00:00
e05bcc22b3 Do not switch a connection if the new connection is not ready to send packets.
There is no benefit of making such switches.

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

Review URL: https://codereview.webrtc.org/2212683002 .

Cr-Commit-Position: refs/heads/master@{#13789}
2016-08-17 01:19:21 +00:00
49c01d7f34 Currently there is not way to programmically test whether a ScreenCapturer
implementation can accurately capture updated regions. Especially in
ScreenCapturerWinDirectx, which has a specific updated region spreading logic
and cannot be tested through regular code path. So we need a controllable
ScreenDrawer to draw some basic shapes on the screen. And a platform independent
test case can use the ScreenDrawer to test a ScreenCapturer.

So this change addes a ScreenDrawer virtual class, and its Windows
implementation ScreenDrawerWin. A disabled gtest ScreenDrawerTest.DrawRectangles
is also added to manually test whether ScreenDrawer can work on a certain
platform.

BUG=314516

TBR=kjellander@webrtc.org

Review-Url: https://codereview.webrtc.org/2210443002
Cr-Commit-Position: refs/heads/master@{#13788}
2016-08-17 00:34:00 +00:00
895e1a9dc3 Change the default backup connection ping interval to 25 seconds.
This avoids the issue that the backup connection may be pinged faster
than the stable rate (once every 2.5 second) we have chosen for
non-backup connections.

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

Review URL: https://codereview.webrtc.org/2239423002 .

Cr-Commit-Position: refs/heads/master@{#13787}
2016-08-16 23:48:14 +00:00
287e54820b Cleanup RtcpReceiver::TMMBRReceived function
BUG=webrtc:951

Review-Url: https://codereview.webrtc.org/2250633002
Cr-Commit-Position: refs/heads/master@{#13786}
2016-08-16 22:15:46 +00:00
f095012dc2 Revert of Adding audio to video_quality_test. (patchset #10 id:230001 of https://codereview.webrtc.org/2136573002/ )
Reason for revert:
This CL breaks https://build.chromium.org/p/client.webrtc/waterfall?builder=Win64%20Debug%20(Clang)

Need to align values to struct Params {} in a proper way. Relanding will follow.

Original issue's description:
> Adding audio to video_quality_test.
>
> This CL adds an audio loopback to video_quality_test (only RunWithVideoRenderer)
>
> BUG=
>
> Committed: https://crrev.com/65a6578e339f52eb5bc400c5715e60498e4af2c1
> Cr-Commit-Position: refs/heads/master@{#13784}

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

Review-Url: https://codereview.webrtc.org/2249163002
Cr-Commit-Position: refs/heads/master@{#13785}
2016-08-16 15:25:48 +00:00
65a6578e33 Adding audio to video_quality_test.
This CL adds an audio loopback to video_quality_test (only RunWithVideoRenderer)

BUG=

Review-Url: https://codereview.webrtc.org/2136573002
Cr-Commit-Position: refs/heads/master@{#13784}
2016-08-16 14:43:54 +00:00
75c287e383 Fix incorrect example in mod_ops.h
TBR=mflodman@webrtc.org
NOTRY=True
BUG=

Review-Url: https://codereview.webrtc.org/2247253003
Cr-Commit-Position: refs/heads/master@{#13783}
2016-08-16 13:38:23 +00:00
a06ce499d6 Run "git cl format" on some files before I start to modify them
This CL does literally nothing else but run "git cl format --full"
on the touched files.

NOPRESUBMIT=true

Review-Url: https://codereview.webrtc.org/2035663002
Cr-Commit-Position: refs/heads/master@{#13782}
2016-08-16 12:35:29 +00:00
b789439b23 Roll chromium_revision 2b53ee0889..915e47250f (411979:412201)
Change log: 2b53ee0889..915e47250f
Full diff: 2b53ee0889..915e47250f

Changed dependencies:
* src/third_party/libFuzzer/src: 764f3890a0..df7f2835bc
* src/third_party/libyuv: 68786ccd53..74491ba0c5
DEPS diff: 2b53ee0889..915e47250f/DEPS

No update to Clang.

TBR=

Review-Url: https://codereview.webrtc.org/2247213003
Cr-Commit-Position: refs/heads/master@{#13781}
2016-08-16 11:56:35 +00:00
90920d5bec GN: Enable msse2 flag in Mac.
It was disabled for some reason, even though in GYP it's enabled.

BUG=626067
NOTRY=True

Review-Url: https://codereview.webrtc.org/2247293002
Cr-Commit-Position: refs/heads/master@{#13780}
2016-08-16 11:13:07 +00:00
9d7eb13c40 Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #3 id:40001 of https://codereview.webrtc.org/2247033003/ )
Reason for revert:
Reverting, because it turns out that third-party code was using webrtc::FilePlayer. I'm not at all sure that this is something WebRTC ought to be exporting, but since we did export it, we have to live with it for now.

Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> (This has been landed twice before, as
> https://codereview.webrtc.org/2037623002 and
> https://codereview.webrtc.org/2240163002. Third time's a charm!)
>
> NOPRESUBMIT=True
> TBR=kjellander@webrtc.org
>
> Committed: https://crrev.com/427ce3d86f6328dc994f84a15c28bb7bfbaa46ef
> Cr-Commit-Position: refs/heads/master@{#13777}

TBR=
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2245413002
Cr-Commit-Position: refs/heads/master@{#13779}
2016-08-16 11:08:39 +00:00
e252d3ce3a MB: Fix incorrect iOS builder names.
In https://codereview.webrtc.org/2239643002/ MB was turned on
for the iOS GYP bots. This exposed an incorrect config for the
iOS simulator bots in client.webrtc.

BUG=589510
NOTRY=True

Review-Url: https://codereview.webrtc.org/2245393003
Cr-Commit-Position: refs/heads/master@{#13778}
2016-08-16 10:46:29 +00:00
427ce3d86f Move FilePlayer and FileRecorder to Voice Engine
Because Voice Engine was the only user.

(This has been landed twice before, as
https://codereview.webrtc.org/2037623002 and
https://codereview.webrtc.org/2240163002. Third time's a charm!)

NOPRESUBMIT=True
TBR=kjellander@webrtc.org

Review-Url: https://codereview.webrtc.org/2247033003
Cr-Commit-Position: refs/heads/master@{#13777}
2016-08-16 10:34:50 +00:00