464 Commits

Author SHA1 Message Date
perkj
8eb37a39e7 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
Reason for revert:
Failed on Win 10 Chrome FYI.

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3847/steps/content_browsertests/logs/stdio

#
# Fatal error in e:\b\c\b\win_builder\src\third_party\webrtc\base\task_queue_win.cc, line 138
# last system error: 87
# Check failed: ((DWORD)0xFFFFFFFF) != result (4294967295 vs. 4294967295)
#

WebRtcBrowserTest

#

Original issue's description:
> - Add task queue to Call with the intent of replacing the use of one of the process threads.
>
> - Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
>
> - BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
>
> - VideoEncoderConfig and VideoSendStream::Config support move semantics.
>
> - The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
>
> BUG=webrtc:5687
>
> Committed: https://crrev.com/cc168360f41322332860cb075edeb1cde21aa473
> Cr-Commit-Position: refs/heads/master@{#13767}

TBR=tommi@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org,sprang@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/2248713003
Cr-Commit-Position: refs/heads/master@{#13774}
2016-08-16 09:40:59 +00:00
perkj
cc168360f4 - Add task queue to Call with the intent of replacing the use of one of the process threads.
- Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.

- BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.

- VideoEncoderConfig and VideoSendStream::Config support move semantics.

- The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.

BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/2060403002
Cr-Commit-Position: refs/heads/master@{#13767}
2016-08-16 07:38:51 +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
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
mflodman
997684079e Remove a few meaningless TODOs
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/2203363002 .

Cr-Commit-Position: refs/heads/master@{#13624}
2016-08-03 13:38:20 +00:00
Sergey Ulanov
525df3ffd1 Add EncodedImageCallback::OnEncodedImage().
OnEncodedImage() is going to replace Encoded(), which is deprecated now.
The new OnEncodedImage() returns Result struct that contains frame_id,
which tells the encoder RTP timestamp for the frame.

BUG=chromium:621691
R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2089773002 .

Committed: https://crrev.com/4c7f4cd2ef76821edca6d773d733a924b0bedd25
Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
Cr-Original-Original-Commit-Position: refs/heads/master@{#13613}
Cr-Original-Commit-Position: refs/heads/master@{#13615}
Cr-Commit-Position: refs/heads/master@{#13617}
2016-08-03 00:46:47 +00:00
sergeyu
51db4dd1bd Revert of Add EncodedImageCallback::OnEncodedImage(). (patchset #14 id:300001 of https://codereview.chromium.org/2089773002/ )
Reason for revert:
broke browser_tests

Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/4c7f4cd2ef76821edca6d773d733a924b0bedd25
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Original-Commit-Position: refs/heads/master@{#13613}
> Cr-Commit-Position: refs/heads/master@{#13615}

TBR=pbos@webrtc.org,mflodman@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691

Review-Url: https://codereview.webrtc.org/2203233002
Cr-Commit-Position: refs/heads/master@{#13616}
2016-08-03 00:33:47 +00:00
Sergey Ulanov
4c7f4cd2ef Add EncodedImageCallback::OnEncodedImage().
OnEncodedImage() is going to replace Encoded(), which is deprecated now.
The new OnEncodedImage() returns Result struct that contains frame_id,
which tells the encoder RTP timestamp for the frame.

BUG=chromium:621691
R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2089773002 .

Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
Cr-Original-Commit-Position: refs/heads/master@{#13613}
Cr-Commit-Position: refs/heads/master@{#13615}
2016-08-02 22:14:51 +00:00
sergeyu
ac4dc2cefe Revert of Add EncodedImageCallback::OnEncodedImage(). (patchset #13 id:280001 of https://codereview.webrtc.org/2089773002/ )
Reason for revert:
broke internal tests

Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Commit-Position: refs/heads/master@{#13613}

TBR=pbos@webrtc.org,mflodman@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691

Review-Url: https://codereview.webrtc.org/2206743002
Cr-Commit-Position: refs/heads/master@{#13614}
2016-08-02 21:33:21 +00:00
Sergey Ulanov
ad34dbe934 Add EncodedImageCallback::OnEncodedImage().
OnEncodedImage() is going to replace Encoded(), which is deprecated now.
The new OnEncodedImage() returns Result struct that contains frame_id,
which tells the encoder RTP timestamp for the frame.

BUG=chromium:621691
R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2089773002 .

Cr-Commit-Position: refs/heads/master@{#13613}
2016-08-02 20:44:25 +00:00
Stefan Holmer
13181035bc Reland: Add BWE plot to event log analyzer.
The plot is constructed by actually running the congestion controller with
the logged rtp headers and rtcp feedback messages to reproduce the same behavior
as in the real call.

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

Review URL: https://codereview.webrtc.org/2193763002 .

Cr-Commit-Position: refs/heads/master@{#13574}
2016-07-29 12:49:04 +00:00
terelius
59c1ad58e6 Revert of Add BWE plot to event log analyzer. (patchset #6 id:100001 of https://codereview.webrtc.org/2188033004/ )
Reason for revert:
Breaks downstream targets.

Original issue's description:
> Add BWE plot to event log analyzer.
>
> The plot is constructed by actually running the congestion controller with
> the logged rtp headers and rtcp feedback messages to reproduce the same behavior
> as in the real call.
>
> R=phoglund@webrtc.org, terelius@webrtc.org
>
> Committed: https://crrev.com/2beea2a8c920000ef19eea20cce397507fc3d5e7
> Cr-Commit-Position: refs/heads/master@{#13558}

TBR=phoglund@webrtc.org,stefan@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/2190013002
Cr-Commit-Position: refs/heads/master@{#13559}
2016-07-28 16:21:32 +00:00
Stefan Holmer
2beea2a8c9 Add BWE plot to event log analyzer.
The plot is constructed by actually running the congestion controller with
the logged rtp headers and rtcp feedback messages to reproduce the same behavior
as in the real call.

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

Review URL: https://codereview.webrtc.org/2188033004 .

Cr-Commit-Position: refs/heads/master@{#13558}
2016-07-28 15:42:31 +00:00
mflodman
86cc6ffc7c Variable audio bitrate.
This is a first CL wiring up AudioSendStream to BitrateAllocator. This
is still experimental and there is a test added for the audio only case,
combined audio video variable bitrate test cases will be added as a
follow up.

BUG=5079

Review-Url: https://codereview.webrtc.org/2165743003
Cr-Commit-Position: refs/heads/master@{#13527}
2016-07-26 11:44:12 +00:00
stefan
bded44b79b Add a CongestionController fuzzer.
BUG=

Review-Url: https://codereview.webrtc.org/2157783002
Cr-Commit-Position: refs/heads/master@{#13497}
2016-07-18 16:26:15 +00:00
katrielc
44905cf49f Add a fuzzer for the pseudoTCP implementation used by chromoting. This
found chromium:620694.

Unfortunately it depends on unsafe GN targets, so do not build in
Chromium.

NOTRY=true

Review-Url: https://codereview.webrtc.org/2129603003
Cr-Commit-Position: refs/heads/master@{#13407}
2016-07-07 12:52:29 +00:00
katrielc
1d4fefbbaf Reland of https://codereview.webrtc.org/2044523002.
Landing these in WebRTC under a guard so they don't build in
Chromium. The guard can be removed once Chromium has migrated to use the
new GN targets.

BUG=webrtc:6081
NOTRY=true

Review-Url: https://codereview.webrtc.org/2117183005
Cr-Commit-Position: refs/heads/master@{#13397}
2016-07-06 15:02:09 +00:00
katrielc
262ba77d78 Add fuzzer for TURN unwrap.
This is called on received network packets if dump_rtp_packets_ is on.

NOTRY=true

Review-Url: https://codereview.webrtc.org/2126463002
Cr-Commit-Position: refs/heads/master@{#13394}
2016-07-06 12:59:34 +00:00
katrielc
36a321d2e3 Update the current RTP parser fuzzer to handle header extensions.
This changes the corpus semantics, but libfuzzer should be smart enough to figure it out, and if not then we can add a seed_corpus to help.

BUG=webrtc:4771
NOTRY=true

Review-Url: https://codereview.webrtc.org/2072473002
Cr-Commit-Position: refs/heads/master@{#13384}
2016-07-05 14:20:30 +00:00
ivoc
14d5dbe5b3 Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
The breaking tests in Chromium have been temporarily disabled, they will be fixed and reenabled soon.

Original CLs: https://codereview.webrtc.org/1748403002/, https://codereview.webrtc.org/2107253002/ and https://codereview.webrtc.org/2106103003/.

TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org,tkchin@webrtc.org
BUG=webrtc:4741, webrtc:5603, chromium:609749

Review-Url: https://codereview.webrtc.org/2110113003
Cr-Commit-Position: refs/heads/master@{#13379}
2016-07-04 14:07:03 +00:00
ivoc
9e03c3b372 Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.

Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}

TBR=solenberg@webrtc.org,tommi@webrtc.org,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=webrtc:4741,webrtc:5603,chromium:609749

Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
2016-06-30 07:59:49 +00:00
Ivo Creusen
1895526c61 Move RtcEventLog object from inside VoiceEngine to Call.
In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.

BUG=webrtc:4741,webrtc:5603,chromium:609749
R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1748403002 .

Cr-Commit-Position: refs/heads/master@{#13321}
2016-06-29 11:57:01 +00:00
katrielc
d4bcdad263 Add a libfuzzer for RtpHeaderParser.
NOTRY=true

Review-Url: https://codereview.webrtc.org/2062103002
Cr-Commit-Position: refs/heads/master@{#13271}
2016-06-23 10:50:43 +00:00
katrielc
bddc94bca2 Add fuzzer corpora.
- RTP and RTCP corpora for existing fuzzers
  - STUN/SDP/pseudotcp for upcoming ones
  - STUN/SDP tokens as well

NOTRY=true

Review-Url: https://codereview.webrtc.org/2082943002
Cr-Commit-Position: refs/heads/master@{#13253}
2016-06-22 13:43:30 +00:00
katrielc
839315beca Use the Chromium libfuzzer template instead of rolling our own.
This lets us use their fancy features, including seed_corpus which is
super handy.

NOTRY=true

Review-Url: https://codereview.webrtc.org/2081683002
Cr-Commit-Position: refs/heads/master@{#13216}
2016-06-20 13:04:01 +00:00
nisse
ac62bd4a3b Rewrite CreateBlackFrame in webrtcvideoengine.
Don't use VideoFrameBuffer::MutableDataY and friends, instead, use
I420Buffer::SetToBlack.

Also introduce static method I420Buffer::Create, to create an object and
return a scoped_refptr.

TBR=marpan@webrtc.org # Trivial change to video_denoiser.cc
BUG=webrtc:5921

Review-Url: https://codereview.webrtc.org/2078943002
Cr-Commit-Position: refs/heads/master@{#13212}
2016-06-20 10:39:00 +00:00
solenberg
217fb66e16 Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
Removes the need to use VoEVolume::SetChannelOutputVolumeScaling().

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2062193002
Cr-Commit-Position: refs/heads/master@{#13194}
2016-06-17 15:30:58 +00:00
kjellander
68208897be Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420)
A fix was needed to make Android tests pass after
https://codereview.chromium.org/2043803003

Another fix was needed for the webrtc_perf_tests.isolate sinc eit was missing
one file.

Change log: 7fa6701bc5..1a73d11e65
Full diff: 7fa6701bc5..1a73d11e65

Changed dependencies:
* src/buildtools: 8dd3c8e39a..099f1da55b
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/0fc7df55c0..171b5403ee
* src/third_party/ffmpeg: 7f03319b9d..bcb8b67b8b
DEPS diff: 7fa6701bc5..1a73d11e65/DEPS

No update to Clang.

TBR=
BUG=webrtc:5990
NOTRY=True

Review-Url: https://codereview.webrtc.org/2074043002
Cr-Commit-Position: refs/heads/master@{#13178}
2016-06-17 06:29:38 +00:00
solenberg
9421853e17 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
Removes the need to use VoEVolume::SetInputMute()/GetInputMute().

BUG=webrtc:4690
NOTRY=true

Review-Url: https://codereview.webrtc.org/2066973002
Cr-Commit-Position: refs/heads/master@{#13172}
2016-06-16 17:53:28 +00:00
tommi
a6219cc3ef FileWrapper[Impl] modifications and actually remove the "Impl" class.
This is a somewhat involved refactoring of this class. Here's an overview of the changes:

* FileWrapper can now be used as a regular class and instances allocated on the stack.
* The type now has support for move semantics and copy isn't allowed.
* New public ctor with FILE* that can be used instead of OpenFromFileHandle.
* New static Open() method.  The intent of this is to allow opening a file and getting back a FileWrapper instance.  Using this method instead of Create(), will allow us in the future to make the FILE* member pointer, to be const and simplify threading (get rid of the lock).
* Rename the Open() method to is_open() and make it inline.
* The FileWrapper interface is no longer a pure virtual interface.  There's only one implementation so there's no need to go through a vtable for everything.
* Functionality offered by the class, is now reduced.  No support for looping (not clear if that was actually useful to users of that flag), no need to implement the 'read_only_' functionality in the class, since file APIs implement that already, no support for *not* managing the file handle (this wasn't used).  OpenFromFileHandle always "manages" the file.
* Delete the unused WriteText() method and don't support opening files in text mode.  Text mode is only different on Windows and on Windows it translates \n to \r\n, which means that files such as log files, could have a slightly different format on Windows than other platforms.  Besides, tools on Windows can handle UNIX line endings.
* Remove FileName(), change Trace code to manage its own path.
* Rename id_ member variable to file_.
* Removed the open_ member variable since the same functionality can be gotten from just checking the file pointer.
* Don't call CloseFile inside of Write.  Write shouldn't be changing the state of the class beyond just attempting to write.
* Remove concept of looping from FileWrapper and never close inside of Read()
* Changed stream base classes to inherit from a common base class instead of both defining the Rewind method. Ultimately, Id' like to remove these interfaces and just have FileWrapper.
* Remove read_only param from OpenFromFileHandle
* Renamed size_in_bytes_ to position_, since it gets set to 0 when Rewind() is called (and the size actually does not change).
* Switch out rw lock for CriticalSection. The r/w lock was only used for reading when checking the open_ flag.

BUG=

Review-Url: https://codereview.webrtc.org/2054373002
Cr-Commit-Position: refs/heads/master@{#13155}
2016-06-15 17:30:18 +00:00
solenberg
971cab0d93 Configure VoE NACK through AudioSendStream::Config, for send streams.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/1955363003
Cr-Commit-Position: refs/heads/master@{#13136}
2016-06-14 17:02:46 +00:00
Peter Boström
0208322ee3 GN: Add video_engine_tests
Adds separate source_sets for the video_engine_tests subtargets inside
audio, call and video and merges them together into video_engine_tests.

BUG=webrtc:5949
R=kjellander@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/2064523002 .

Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 10:53:09 +00:00
kjellander
dfe6937666 Revert of Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420) (patchset #2 id:20001 of https://codereview.webrtc.org/2061723002/ )
Reason for revert:
webrtc_perf_tests fails on all Android Release testers:
http://build.chromium.org/p/client.webrtc/builders/Android64%20Tests%20%28L%20Nexus9%29/builds/3673

Original issue's description:
> Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420)
>
> A fix was needed to make Android tests pass after
> https://codereview.chromium.org/2043803003
>
> Change log: 7fa6701bc5..1a73d11e65
> Full diff: 7fa6701bc5..1a73d11e65
>
> Changed dependencies:
> * src/buildtools: 8dd3c8e39a..099f1da55b
> * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/0fc7df55c0..171b5403ee
> * src/third_party/ffmpeg: 7f03319b9d..bcb8b67b8b
> DEPS diff: 7fa6701bc5..1a73d11e65/DEPS
>
> No update to Clang.
>
> TBR=
> BUG=webrtc:5990
> NOTRY=True
>
> Committed: https://crrev.com/c49cf1308f879ce029f5f0e804b4eef3b08b67c5
> Cr-Commit-Position: refs/heads/master@{#13117}

TBR=
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5990

Review-Url: https://codereview.webrtc.org/2059423002
Cr-Commit-Position: refs/heads/master@{#13118}
2016-06-13 21:11:09 +00:00
kjellander
c49cf1308f Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420)
A fix was needed to make Android tests pass after
https://codereview.chromium.org/2043803003

Change log: 7fa6701bc5..1a73d11e65
Full diff: 7fa6701bc5..1a73d11e65

Changed dependencies:
* src/buildtools: 8dd3c8e39a..099f1da55b
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/0fc7df55c0..171b5403ee
* src/third_party/ffmpeg: 7f03319b9d..bcb8b67b8b
DEPS diff: 7fa6701bc5..1a73d11e65/DEPS

No update to Clang.

TBR=
BUG=webrtc:5990
NOTRY=True

Review-Url: https://codereview.webrtc.org/2061723002
Cr-Commit-Position: refs/heads/master@{#13117}
2016-06-13 19:12:47 +00:00
ossu
29b1a8d7ec Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
CreatePeerConnectionFactory does not yet expose the ability to set the
factory from the outside.

Added notry due to android_dbg being broken.

NOTRY=True
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/1991233004
Cr-Commit-Position: refs/heads/master@{#13112}
2016-06-13 14:35:01 +00:00
Tommi
733b5478dd Movable support for VideoReceiveStream::Config and avoid copies.
Instead of the default copy constructor, the Copy() method has to be used.  In this CL, the number of copies has been reduced significantly in production code. One case in the video engine remains, where we need to restart a video stream.  Even in that case, I'm sure we could avoid it, but for this particular CL, I decided against it to keep things simple (and it's also an edge case).  Most importantly, creating copies is made harder and the interface encourages ownership transfers.

R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/2042603002 .

Cr-Commit-Position: refs/heads/master@{#13102}
2016-06-10 15:58:12 +00:00
katrielc
81ca73586e Remove new fuzzers until their GN targets work properly in Chromium.
Chromium uses gn gen --check, which doesn't like some of the includes
used in the new gn targets the fuzzers use. This breaks Chromium
libfuzzer compiles, for which there isn't yet a webrtc FYI bot.

I'm working on fixing the includes, at which point these can come back.

BUG=chromium:618901
NOTRY=true

Review-Url: https://codereview.webrtc.org/2053293002
Cr-Commit-Position: refs/heads/master@{#13098}
2016-06-10 09:59:46 +00:00
kjellander
94cee3111c GN: Enable api,media,pc and p2p for the 'webrtc' target.
These parts were commented out to avoid breaking the Chromium
WebRTC FYI bots. Include them in the WebRTC build to make our bots
build as many as possible of our GN targets.

BUG=webrtc:5949
NOTRY=True
TBR=phoglund@webrtc.org

Review-Url: https://codereview.webrtc.org/2054903002
Cr-Commit-Position: refs/heads/master@{#13097}
2016-06-10 08:57:05 +00:00
Peter Boström
555cfe9e6e Use relative paths for api/p2p fuzzers.
BUG=
R=aizatsky@chromium.org
TBR=katrielc@webrtc.org

Review URL: https://codereview.webrtc.org/2053093002 .

Cr-Commit-Position: refs/heads/master@{#13095}
2016-06-09 21:25:25 +00:00
katrielc
7b496e026b Add fuzzers for SDP and STUN parsing.
The STUN fuzzer is split into two parts: validation and parsing. The
latter should be able to handle invalid packets instead of assuming
the validation deals with them, since an adversary could set a valid
HMAC on an invalid packet.

NOTRY=true

Review-Url: https://codereview.webrtc.org/2044523002
Cr-Commit-Position: refs/heads/master@{#13050}
2016-06-06 16:45:32 +00:00
katrielc
e1e951f965 Allow fuzzers to depend on anything, since they want access to as many targets as possible.
NOTRY=true

Review-Url: https://codereview.webrtc.org/2046493002
Cr-Commit-Position: refs/heads/master@{#13048}
2016-06-06 15:05:03 +00:00
danilchap
46b89b9a3f Collapse most stdout spammy output of webrtc_perf_tests with PrintResultList
BUG=webrtc:5032
NOTRY=true

Review-Url: https://codereview.webrtc.org/2030073002
Cr-Commit-Position: refs/heads/master@{#13037}
2016-06-03 16:27:45 +00:00
kwiberg
abe95ba323 AudioDecoderIsacT: Require caller to always specify sample rate
This gets rid of the complex & icky state where the sample rate is not
yet determined.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2020353003
Cr-Commit-Position: refs/heads/master@{#13011}
2016-06-02 09:59:02 +00:00
kjellander
8f4419b074 GN: Replace Windows suppressions of warning 4267 with config.
This makes the GN configurations easier to read.

BUG=webrtc:5949
NOTRY=True

Review-Url: https://codereview.webrtc.org/2020343003
Cr-Commit-Position: refs/heads/master@{#13006}
2016-06-02 09:09:56 +00:00
danilchap
a6a70073fb Fix FakeNetworkPipe to not deliver packet faster than requested.
BUG=webrtc:5938

Review-Url: https://codereview.webrtc.org/2024293003
Cr-Commit-Position: refs/heads/master@{#12997}
2016-06-01 18:20:48 +00:00
kjellander
0f380d8ffa GN: Add test_support_unittests target
BUG=webrtc:5949
TESTED=Passing run on Linux, Windows and Mac:
gn gen out/Default
ninja -C out/Default
out/Default/test_support_unittests

And for Android:
gn gen out/Default --args='target_os="android" target_cpu="arm"'
ninja -C out/Default
out/Default/bin/run_test_support_unittests

NOTRY=True

Review-Url: https://codereview.webrtc.org/2021243002
Cr-Commit-Position: refs/heads/master@{#12989}
2016-06-01 11:48:32 +00:00
kjellander
208d19845d Rename APK tests workaround to make it more generic.
We plan to add junit tests running with Robolectric
so naming these files "apk" is slightly confusing.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2020213002
Cr-Commit-Position: refs/heads/master@{#12971}
2016-05-31 11:01:47 +00:00
philipel
536378bf37 Allow FakeNetworkPipe to drop packets in bursts.
The fake network pipe will still only drop packets at an average rate of
|loss_percent| but in bursts at an average length specified by
|avg_burst_loss_length|.

Also added the flag -avg_burst_loss_length to video loopback.

BUG=

Review-Url: https://codereview.webrtc.org/1995683003
Cr-Commit-Position: refs/heads/master@{#12969}
2016-05-31 10:20:28 +00:00
ossu
5f7cfa50e5 Moved CreateBuiltinDecoderFactory out to VoEBaseImpl.
VoEBase is plumbed to optionally take an AudioDecoderFactory, or create
a builtin factory if none is provided.

Retained the CreateChannel interfaces in Channel and ChannelManager
and added variants for injecting an AudioDecoderFactory. The
"old-style" variants call CreateBuiltinAudioDecoderFactory to get a
factory to use.

(Just realized this means each channel uses a separate factory with the
old-style calls. Probably ok.)

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/1993783002
Cr-Commit-Position: refs/heads/master@{#12961}
2016-05-30 15:11:36 +00:00
isheriff
6f8d686d35 Remove use of RtpHeaderExtension and clean up
Currently there are two structs that are identical and track extension details:
webrtc::RtpExtension
cricket::RtpHeaderExtension

The use of the structs is mixed in the code to track the extensions being
supported. This results in duplicate definition of
the URI constants and there is code to convert between the two structs.

Clean up to use a single RtpHeader throughout the codebase. The actual location
of RtpHeader may change in future (perhaps to be located in api/). Additionally,
this CL renames some of the constants to clarify Uri and Id use.

BUG= webrtc:5895

Review-Url: https://codereview.webrtc.org/1984983002
Cr-Commit-Position: refs/heads/master@{#12924}
2016-05-26 18:25:04 +00:00