8673 Commits

Author SHA1 Message Date
jackychen
9bfa1063d1 Change the threshold for external VNR.
The change is based on visual evaluation results and improves the
denoising result on both desktop/laptop and Nexus.

Review-Url: https://codereview.webrtc.org/1935353002
Cr-Commit-Position: refs/heads/master@{#12612}
2016-05-03 18:21:34 +00:00
Peter Boström
c4deee49a3 Use RC_TIMESTAMP_MODE for OpenH264.
Performs rate control based on timestamp deltas instead of announced
frame rate.

BUG=webrtc:5855
R=hbos@webrtc.org

Review URL: https://codereview.webrtc.org/1945763002 .

Cr-Commit-Position: refs/heads/master@{#12611}
2016-05-03 18:00:05 +00:00
henrik.lundin
c8fe991a3d Removing SpatialAudio test code
The code has not been dead for almost four years (since
https://webrtc-codereview.appspot.com/636006).

NOTRY=True

Review-Url: https://codereview.webrtc.org/1947483002
Cr-Commit-Position: refs/heads/master@{#12610}
2016-05-03 15:40:13 +00:00
henrik.lundin
b1fb72bebb NetEq: Move counting of generated CNG samples from DecisionLogic
The counting is moved to NetEqImpl, and the new counter is realized as a
Stopwatch object. The DecisionLogic class still has to maintain record
of when the CNG period is shortened, in order to reduce the delay. This
is recorded in a new noise_fast_forward_ member in DecisionLogic.

BUG=webrtc:5608

Review-Url: https://codereview.webrtc.org/1914303004
Cr-Commit-Position: refs/heads/master@{#12608}
2016-05-03 15:18:54 +00:00
peah
b46083ed63 This CL introduces a new data logging functionality
to use for the APM. It allows simple and rapid
additions of exploratory data logpoints to use
during bug investigations and module performance
analysis.
The new data logging functionality is also in this CL
used to replace the existing data logging functionality
present in the AEC.

Additional information:
As there was an issue with that the build flag for
activating this feature was not present in all
compilation units that included the feature additional
changes were needed. A summary of the changes are
-The build files were modified to ensure that the
 logging build flag always is set to either 0 or 1
 for compilation units that include the feature.
-Build-time checks in the appropriate places were added
 to ensure that the above is fulfilled.
-The build object was added dynamically to the AEC state
 as a pointer to ensure that the size of that state is not
 dependent on whether the logging build flag is set or not.
-The constructor of the AEC class needed to be modified in
 order to construct the logging object. For this a destructor
 was also needed.
-An unused method without any declaration was removed in
 order to avoid any issues with the logging flag being set to
 0 or 1.

This CL will be immediately followed with an upcoming CL
that replaces the logging in echo_cancellation.cc with the
new functionality which will ensure that the  logging flag
is only used in one place within WebRTC, which in turn will
fully ensure that all compilation units that uses the feature
also have the flag properly set.

BUG=webrtc:5201, webrtc:5298

Review-Url: https://codereview.webrtc.org/1877713002
Cr-Commit-Position: refs/heads/master@{#12607}
2016-05-03 14:01:27 +00:00
philipel
696a802332 Re-enable Vp9FlexModeRefCount
Looks like this test was disable (https://codereview.webrtc.org/1556273002) but never re-enabled after the bug was fixed.

BUG=webrtc:5402

Review-Url: https://codereview.webrtc.org/1914893003
Cr-Commit-Position: refs/heads/master@{#12606}
2016-05-03 12:45:48 +00:00
pbos
35fdb2a914 Log WebRTC.Video.AVSyncOffsetInMs.
BUG=
R=asapersson@webrtc.org

Review-Url: https://codereview.webrtc.org/1941993002
Cr-Commit-Position: refs/heads/master@{#12605}
2016-05-03 10:32:16 +00:00
kwiberg
5178ee86ba NetEq: Use a BuiltinAudioDecoderFactory to create decoders
Later steps in the refactoring will have the factory injected from the
outside rather than owned by NetEq.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/1928293002
Cr-Commit-Position: refs/heads/master@{#12604}
2016-05-03 08:39:08 +00:00
magjed
ddf165393f Android EGL: Synchronize calls to eglCreateContext
Synchronize calls to EGL10/EGL14.eglCreateContext on EglBase.lock. The
reason is that a deadlock between the remote render thread in
eglSwapBuffers and MediaCodecVideoEncoder eglCreateContext was observed.

The function calls that are now synchronized on EglBase.lock are:
eglCreateContext, eglMakeCurrent, eglSwapBuffers, and
SurfaceTexture.updateTexImage.

BUG=webrtc:5702

Review-Url: https://codereview.webrtc.org/1937933002
Cr-Commit-Position: refs/heads/master@{#12603}
2016-05-03 08:24:44 +00:00
nisse
30f118effd This cl deletes the class webrtc::VideoRendererCallback.
Replaced by VideoSinkInterface instead.

Also delete stream_id property of IncomingVideoStream.

BUG=webrtc:5426

Review-Url: https://codereview.webrtc.org/1813173002
Cr-Commit-Position: refs/heads/master@{#12602}
2016-05-03 08:09:17 +00:00
nisse
fc88ffe9d8 Fix allocation size in CricketToJavaI420Frame, taking stride into account.
BUG=

Review-Url: https://codereview.webrtc.org/1941773002
Cr-Commit-Position: refs/heads/master@{#12601}
2016-05-03 07:32:16 +00:00
asapersson
35151f35ec Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
- "WebRTC.Video.SendDelayInMs"

Change so that PacketOption packet id is always set in RtpSender (if having a TransportSequenceNumberAllocator).
Add SendDelayStats class for computing delays.
Add SendPacketObserver to RtpRtcp config and register SendDelayStats as observer.
Wire up OnSentPacket to SendDelayStats.

BUG=webrtc:5215

Review-Url: https://codereview.webrtc.org/1478253002
Cr-Commit-Position: refs/heads/master@{#12600}
2016-05-03 06:44:11 +00:00
Honghai Zhang
5a2463796e Do not stop a session unless the candidate of a writable connection belongs to the
latest generation.

BUG=webrtc:5644
R=deadbeef@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1857453002 .

Cr-Commit-Position: refs/heads/master@{#12599}
2016-05-03 00:28:42 +00:00
deadbeef
5dd42fd849 Fixing a segfault that can occur when changing the track of an RtpSender.
The reference to the old track needs to be kept alive until SetAudioSend/
SetSource is called, because otherwise it could be deleted while the audio/
video engine is still trying to use the track.

BUG=webrtc:5796

Review-Url: https://codereview.webrtc.org/1894283002
Cr-Commit-Position: refs/heads/master@{#12598}
2016-05-02 23:20:08 +00:00
minyue
acf143128f Removing unused resources from building files.
A number of resources files have been removed in
https://codereview.webrtc.org/1928923002/

This CL remove the them from the building files.

BUG=

Review-Url: https://codereview.webrtc.org/1940933002
Cr-Commit-Position: refs/heads/master@{#12597}
2016-05-02 19:10:12 +00:00
perkj
376b192ea3 Remove VideoCodingModule::VCMPacketizationCallback
And move encoder name cb to VCMSendStatisticsCallback.

BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/1900193004
Cr-Commit-Position: refs/heads/master@{#12596}
2016-05-02 18:35:33 +00:00
Peter Boström
16ac3280f5 Remove VCMRenderBufferSizeCallback.
Unused/dead code.

BUG=
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1923713002 .

Cr-Commit-Position: refs/heads/master@{#12595}
2016-05-02 16:28:15 +00:00
nisse
1bffc1d1a4 Rename rtc::Time64 --> rtc::TimeMillis.
In the discussion on https://codereview.webrtc.org/1888593004/, a more
decriptive name was suggested for Time64.

BUG=webrtc:5740

Review-Url: https://codereview.webrtc.org/1923213002
Cr-Commit-Position: refs/heads/master@{#12594}
2016-05-02 15:19:00 +00:00
perkj
bc75d97c32 Remove PayloadRouter dependency from ViEEncoder.
BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/1912653002
Cr-Commit-Position: refs/heads/master@{#12593}
2016-05-02 13:31:31 +00:00
minyue
5bd3397e53 Adding 120 ms frame length support in NetEq.
BUG=webrtc:1015

Review-Url: https://codereview.webrtc.org/1901633002
Cr-Commit-Position: refs/heads/master@{#12592}
2016-05-02 11:46:19 +00:00
henrika
7d4a6c3208 Adds timeout for audio record thread in Java layer
BUG=b/28448866
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1933123002 .

Cr-Commit-Position: refs/heads/master@{#12590}
2016-05-02 09:01:02 +00:00
minyue
53ff70f582 Reland "Avoiding overflow in cross correlation in NetEq."
The original CL is https://codereview.webrtc.org/1908623002/

An error was caused by that and this CL fix that problem and reland the CL.

BUG=

Review-Url: https://codereview.webrtc.org/1931933004
Cr-Commit-Position: refs/heads/master@{#12589}
2016-05-02 08:50:34 +00:00
asapersson
a017b8ed2e Remove asapersson from webrtc/modules/utility/OWNERS.
BUG=

Review-Url: https://codereview.webrtc.org/1842893003
Cr-Commit-Position: refs/heads/master@{#12588}
2016-05-02 08:22:35 +00:00
Magnus Jedvert
d1d96b2508 VideoCapturerAndroid: Remove deprecated create function with egl context argument
R=glaznev@webrtc.org, perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1900413002 .

Cr-Commit-Position: refs/heads/master@{#12587}
2016-05-02 07:43:32 +00:00
pbos
1ba8d39a9c Remove webrtc/stream.h and unutilized inheritance.
Removes inheritance and a virtual call. Also removes a root header that
would have needed to be moved into a subdirectory otherwise to prevent
circular dependencies.

BUG=webrtc:4243
R=kjellander@webrtc.org, solenberg@webrtc.org
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/1924793002
Cr-Commit-Position: refs/heads/master@{#12586}
2016-05-02 03:18:36 +00:00
pbos
c04305200e Reland of move VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1935753002/ )
Reason for revert:
Not root cause for perf regression (regression still ongoing).

Original issue's description:
> Revert of Remove VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1917083003/ )
>
> Reason for revert:
> Speculative revert for perf regression.
>
> Original issue's description:
> > Remove VCMQmRobustness.
> >
> > Class contained a lot of not-really-wired-up functionality that ended up
> > being complicated ways of saying return 1; or return false;. This
> > removes this dependency that complicates code readability significantly.
> >
> > BUG=webrtc:5066
> > R=marpan@google.com, marpan@webrtc.org
> > TBR=stefan@webrtc.org
> >
> > Committed: https://crrev.com/73894369791cb5eedc8788baf918ec07d11d351d
> > Cr-Commit-Position: refs/heads/master@{#12516}
>
> TBR=marpan@webrtc.org,stefan@webrtc.org,marpan@google.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5066, chromium:607838
>
> Committed: https://crrev.com/602316c3cd8556cc78d44f3ea4cd5fc8e70d9417
> Cr-Commit-Position: refs/heads/master@{#12572}

TBR=marpan@webrtc.org,stefan@webrtc.org,marpan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5066, chromium:607838

Review-Url: https://codereview.webrtc.org/1941643002
Cr-Commit-Position: refs/heads/master@{#12583}
2016-05-02 00:19:13 +00:00
kwiberg
bfefb03ec1 Replace scoped_ptr with unique_ptr everywhere
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.

BUG=webrtc:5520

Review-Url: https://codereview.webrtc.org/1937693002
Cr-Commit-Position: refs/heads/master@{#12581}
2016-05-01 21:53:55 +00:00
kwiberg
322c4a0b3a Replace scoped_ptr with unique_ptr in webrtc/libjingle/
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.

BUG=webrtc:5520

Review-Url: https://codereview.webrtc.org/1935893002
Cr-Commit-Position: refs/heads/master@{#12577}
2016-04-30 09:40:26 +00:00
mikescarlett
a97611a43f Stop QuicDataChannel and QuicDataTransport unit tests from segfaulting
Minor; needed because QuicTransportChannel now owns the ICE transport
channel as a result of the QuicTransport CL.

TBR=pthatcher@webrtc.org

BUG=

Review-Url: https://codereview.webrtc.org/1934723003
Cr-Commit-Position: refs/heads/master@{#12576}
2016-04-30 05:09:32 +00:00
mikescarlett
e7748674ee Allow TransportController to create a QuicTransportChannel
A QuicTransport is implemented that subclasses Transport
and takes ownership of the QuicTransportChannel/P2PTransportChannel.

Split from CL https://codereview.webrtc.org/1844803002/.

BUG=

Review-Url: https://codereview.webrtc.org/1856943002
Cr-Commit-Position: refs/heads/master@{#12575}
2016-04-30 03:21:04 +00:00
mikescarlett
9bc517f123 Add QuicDataChannel and QuicDataTransport classes
QuicDataChannel implements DataChannelInterface. It
replaces SCTP data channels by using a QuicTransportChannel
to create a ReliableQuicStream for each message.
QuicDataChannel only implements unordered, reliable delivery
for the initial implementation and does not send a hello message.

QuicDataTransport is a helper class that dispatches each incoming
ReliableQuicStream to a QuicDataChannel when the remote
peer receives a message by parsing the data channel id and message id
from the message header. It is also responsible for encoding the header
before QuicDataChannel sends the message.

Split from CL https://codereview.chromium.org/1844803002/.

BUG=

Review-Url: https://codereview.webrtc.org/1886623002
Cr-Commit-Position: refs/heads/master@{#12574}
2016-04-30 01:31:03 +00:00
mikescarlett
70035cae4d Fix QuicSession to unbuffer data when the QuicTransportChannel reconnects
The QuicWriteBlockedList needs to register outgoing QUIC
streams so that when the QuicTransportChannel becomes
unwritable and QUIC streams have buffered data, they can
send data once the QuicTransportChannel becomes writable.

Otherwise the QUIC streams will remain write blocked
after the QuicTransportChannel is writable.

BUG=

Review-Url: https://codereview.webrtc.org/1888903002
Cr-Commit-Position: refs/heads/master@{#12573}
2016-04-30 01:14:47 +00:00
pbos
602316c3cd Revert of Remove VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1917083003/ )
Reason for revert:
Speculative revert for perf regression.

Original issue's description:
> Remove VCMQmRobustness.
>
> Class contained a lot of not-really-wired-up functionality that ended up
> being complicated ways of saying return 1; or return false;. This
> removes this dependency that complicates code readability significantly.
>
> BUG=webrtc:5066
> R=marpan@google.com, marpan@webrtc.org
> TBR=stefan@webrtc.org
>
> Committed: https://crrev.com/73894369791cb5eedc8788baf918ec07d11d351d
> Cr-Commit-Position: refs/heads/master@{#12516}

TBR=marpan@webrtc.org,stefan@webrtc.org,marpan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5066, chromium:607838

Review-Url: https://codereview.webrtc.org/1935753002
Cr-Commit-Position: refs/heads/master@{#12572}
2016-04-29 23:10:36 +00:00
mikescarlett
8d37d2941e Update QuicTransportChannel to latest version of libquic (Chromium: f03d2c62)
These changes are necessary to incorporate the latest
changes to QUIC sessions and the QUIC crypto handshake.

BUG=

Review-Url: https://codereview.webrtc.org/1910633003
Cr-Commit-Position: refs/heads/master@{#12571}
2016-04-29 22:35:09 +00:00
skvlad
f3569c8a8f Added the API to create an RTCRtpSender to the Objective C wrapper.
Objective C applications can now create new RTCRtpSenders and change their tracks, which gives them more fine grained control than MediaStreams.

BUG=

Review-Url: https://codereview.webrtc.org/1888633002
Cr-Commit-Position: refs/heads/master@{#12570}
2016-04-29 22:30:24 +00:00
Karl Wiberg
0bdebd4b21 Re-add a (dummy) webrtc/base/buffer.cc to hopefully unbreak the Chromium build
Remove this file once Chromium doesn't need it anymore.

TBR=tommi@webrtc.org

BUG=webrtc:5845

Review URL: https://codereview.webrtc.org/1928633006 .

Cr-Commit-Position: refs/heads/master@{#12568}
2016-04-29 18:18:55 +00:00
minyue
4f90677527 Making NetEq bitexactness test independent on reference files.
NetEq bitexactness test depended on reference files which differs from platform to platform. This makes it very hard to update Neteq.

New method maintains the ability to save output into files. But it verifies the checksum only. With this, when bitexactness test fails, we can still check closely to the output file if need, but the test becomes much easier to modify.

BUG=

Review-Url: https://codereview.webrtc.org/1928923002
Cr-Commit-Position: refs/heads/master@{#12567}
2016-04-29 18:05:18 +00:00
solenberg
05e61edd8f Remove usage of VoENetwork from VoEWrapper and FakeWebRtcVoiceEngine.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/1934513002
Cr-Commit-Position: refs/heads/master@{#12566}
2016-04-29 16:05:35 +00:00
pbos
79e2842381 Add tracing to MessageQueue::Dispatch.
Accounts for additional blocking yet unaccounted for that's not visible
through invoke.

BUG=
R=tommi@webrtc.org

Review-Url: https://codereview.webrtc.org/1932753002
Cr-Commit-Position: refs/heads/master@{#12565}
2016-04-29 15:48:12 +00:00
kwiberg
a4ac4786a8 Define rtc::BufferT, like rtc::Buffer but for any trivial type
And redefine rtc::Buffer as

  using Buffer = BufferT<uint8_t>;

(In the long run, I'd like to remove the type alias and rename the
template to just rtc::Buffer, but that requires all current users of
Buffer to start saying Buffer<uint8_t> instead, and since Buffer is
used in the API, we can't do that in one step.)

The immediate reason for the new template is that we'd like to use
BufferT<int16_t> in the AudioDecoder interface.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/1929903002
Cr-Commit-Position: refs/heads/master@{#12564}
2016-04-29 15:00:28 +00:00
nisse
ef8b61e110 Enable -Winconsistent-missing-override flag.
The problem with gmock is worked around by commenting out any other override declarations in classes using gmock.

NOPRESUBMIT=True
BUG=webrtc:3970

Review-Url: https://codereview.webrtc.org/1921653002
Cr-Commit-Position: refs/heads/master@{#12563}
2016-04-29 13:09:23 +00:00
tommi
b296d0591c Revert of New task queueing primitive for async tasks: TaskQueue. (patchset #5 id:80001 of https://codereview.webrtc.org/1919733002/ )
Reason for revert:
Reverting this temporarily while I figure out the issues with the Chrome on android GN debug build.

Original issue's description:
> New task queueing primitive for async tasks: TaskQueue.
> TaskQueue is a new way to asynchronously execute tasks sequentially
> in a thread safe manner with minimal locking.  The implementation
> uses OS supported APIs to do this that are compatible with async IO
> notifications from things like sockets and files.
>
> This class is a part of rtc_base_approved, so can be used by both
> the webrtc and libjingle parts of the WebRTC library.  Moving forward,
> we can replace rtc::Thread and webrtc::ProcessThread with this implementation.
>
> NOTE: It should not be assumed that all tasks that execute on a TaskQueue,
> run on the same thread.  E.g. on Mac and iOS, we use GCD dispatch queues
> which means that tasks might execute on different threads depending on
> what's the most efficient thing to do.

TBR=perkj@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/1935483002
Cr-Commit-Position: refs/heads/master@{#12562}
2016-04-29 13:03:38 +00:00
tommi
0c9df5e568 New task queueing primitive for async tasks: TaskQueue.
TaskQueue is a new way to asynchronously execute tasks sequentially
in a thread safe manner with minimal locking.  The implementation
uses OS supported APIs to do this that are compatible with async IO
notifications from things like sockets and files.

This class is a part of rtc_base_approved, so can be used by both
the webrtc and libjingle parts of the WebRTC library.  Moving forward,
we can replace rtc::Thread and webrtc::ProcessThread with this implementation.

NOTE: It should not be assumed that all tasks that execute on a TaskQueue,
run on the same thread.  E.g. on Mac and iOS, we use GCD dispatch queues
which means that tasks might execute on different threads depending on
what's the most efficient thing to do.

Review-Url: https://codereview.webrtc.org/1919733002
Cr-Commit-Position: refs/heads/master@{#12561}
2016-04-29 11:49:14 +00:00
danilchap
4edf93bcc6 Remove deprecated functions in rtp_rtcp module
Review-Url: https://codereview.webrtc.org/1859273003
Cr-Commit-Position: refs/heads/master@{#12560}
2016-04-29 10:01:33 +00:00
nisse
0565451820 Reland of Delete cricket::VideoFrame methods GetYPlane and GetYPitch. (patchset #1 id:1 of https://codereview.webrtc.org/1921493004/ )
Reason for revert:
Chrome has been updated, cl https://codereview.chromium.org/1919283005/

Original issue's description:
> Revert of Delete cricket::VideoFrame methods GetYPlane and GetYPitch. (patchset #5 id:80001 of https://codereview.webrtc.org/1901973002/ )
>
> Reason for revert:
> GetYPlane, GetYPitch etc is used by Chromium.
>
> Original issue's description:
> > Delete cricket::VideoFrame methods GetYPlane and GetYPitch.
> >
> > (And similarly for U and V). Also change video_frame_buffer method to
> > return a const ref to a scoped_ref_ptr.
> >
> > This cl is analogous to https://codereview.webrtc.org/1900673002/,
> > which delete corresponding methods in webrtc::VideoFrame.
> >
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/1c27c6bf4cf0476dd2f09425509afaae4cdfe599
> > Cr-Commit-Position: refs/heads/master@{#12492}
>
> TBR=magjed@webrtc.org,perkj@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/b05f994bb6f3055c852891c8acb531aee916a668
> Cr-Commit-Position: refs/heads/master@{#12494}

TBR=magjed@webrtc.org,perkj@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,terelius@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1923903002
Cr-Commit-Position: refs/heads/master@{#12559}
2016-04-29 09:56:06 +00:00
nisse
5b3c443d30 Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
Reason for revert:
Breaks chrome FYI bots.

Original issue's description:
> Delete webrtc::VideoFrame methods buffer and stride.
>
> To make the HasOneRef/IsMutable hack work, also had to change the
> video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> to not imply an AddRef.
>
> BUG=webrtc:5682

TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1935443002
Cr-Commit-Position: refs/heads/master@{#12558}
2016-04-29 09:39:33 +00:00
nisse
a0591b5473 Delete webrtc::VideoFrame methods buffer and stride.
To make the HasOneRef/IsMutable hack work, also had to change the
video_frame_buffer method to return a const ref to a scoped_ref_ptr,
to not imply an AddRef.

BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1900673002
Cr-Commit-Position: refs/heads/master@{#12557}
2016-04-29 09:09:33 +00:00
nisse
b99395a544 Reland of Delete video_render module. (patchset #1 id:1 of https://codereview.webrtc.org/1923613003/ )
Reason for revert:
Chrome's build files have now been updated, see cl https://codereview.chromium.org/1929933002/

Original issue's description:
> Revert of Delete video_render module. (patchset #12 id:220001 of https://codereview.webrtc.org/1912143002/ )
>
> Reason for revert:
> This breaks every buildbot in chromium.webrtc.fyi and I don't see any roll in progress to address this (and I don't see how that would be possible either).
> Usage in Chrome: https://code.google.com/p/chromium/codesearch#search/&q=modules.gyp%3Avideo_render&sq=package:chromium&type=cs
>
> Example failures:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/5420
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/4526
>
> I think it's fine to delete our video_render_module_internal_impl target and those files, but video_render target needs to remain.
>
> Original issue's description:
> > Delete video_render module.
> >
> > BUG=webrtc:5817
> >
> > Committed: https://crrev.com/97cfd1ec05d07ef233356e57f7aa4b028b74ffba
> > Cr-Commit-Position: refs/heads/master@{#12526}
>
> TBR=mflodman@webrtc.org,pbos@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5817

TBR=mflodman@webrtc.org,pbos@webrtc.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5817

Review-Url: https://codereview.webrtc.org/1929223003
Cr-Commit-Position: refs/heads/master@{#12556}
2016-04-29 07:58:48 +00:00
mflodman
3d7db263b9 Switch voice transport to use Call and Stream instead of VoENetwork.
VoENetwork is kept for now, but is not really used anylonger.

webrtcvoiceengine is changed to have the same behavior for unsignaled
ssrc as video has, which is reflected by disabling one test case and
this will be discussed and followed up.

BUG=webrtc:5079

TBR=tommi

Review-Url: https://codereview.webrtc.org/1909333002
Cr-Commit-Position: refs/heads/master@{#12555}
2016-04-29 07:57:21 +00:00
henrik.lundin
8f8c96d192 NetEq: Use TickTimer in DelayManager
This change replaces packet_iat_count_ms_ and max_timer_ms_, two
time-counting member variables in DelayManager, with Stopwatch objects
obtained from a TickTimer.

BUG=webrtc:5608

Review-Url: https://codereview.webrtc.org/1929863002
Cr-Commit-Position: refs/heads/master@{#12554}
2016-04-29 06:19:27 +00:00