We detect an unreasonable state (caused by a bad encoded stream)
before it can lead to problems, and handle it by resetting the
decoder.
NOPRESUBMIT=true
BUG=chromium:617124
Review-Url: https://codereview.webrtc.org/2255203002
Cr-Commit-Position: refs/heads/master@{#13888}
Move the webrtc/test/test_support/metrics sources into
test_support[_unittests] targets.
This is essentially reverting https://webrtc-codereview.appspot.com/5789004
and moving these sources back to the right target.
Add missing foreman_cif.yuv resource needed for these tests.
For MIPS, a compile error was surfacing for logcat_trace_context.h when
flipping bot to GN, which was fixed.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2267113002
Cr-Commit-Position: refs/heads/master@{#13860}
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}
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}
Reason for revert:
Breaks downstream code, so revert again. Yay.
Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> (This is a re-land of https://codereview.webrtc.org/2037623002, which
> had to be reverted.)
>
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/dc65ea29b3270ad418050658ad962ddd33ee70c1
> Cr-Commit-Position: refs/heads/master@{#13757}
TBR=perkj@webrtc.org,kjellander@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/2245153002
Cr-Commit-Position: refs/heads/master@{#13758}
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}
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}
The ProbingCalculator class calculates and validates the results from
probing attempts.
BUG=webrtc:5859
Review-Url: https://codereview.webrtc.org/2121183002
Cr-Commit-Position: refs/heads/master@{#13589}
OutputMixer and AudioConferenceMixer communicated via a callback. OutputMixer implemented an AudioMixerOutputReceiver interface, which defines the callback function NewMixedAudio. This has been removed and replaced by a simple function in the new mixer. The audio frame with mixed audio is now copied one time less. I have also removed one forward declaration.
Review-Url: https://codereview.webrtc.org/2111293003
Cr-Commit-Position: refs/heads/master@{#13550}
Also adds a copy of the BWE test suite to the new DelayBasedBwe class.
BUG=webrtc:6079
Review-Url: https://codereview.webrtc.org/2126793002
Cr-Commit-Position: refs/heads/master@{#13428}
I have added build files and renamed the mixer so that it doesn't conflict with the old one. The header includes now point to this copy of the mixer. I have also fixed some of the more obvious cases of style guide non-conformance and run 'PRESUBMIT' on the old mixer.
This is a first step in the creation of a new mixing module that will replace AudioConferencMixer and OutputMixer.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2104363003
Cr-Commit-Position: refs/heads/master@{#13378}
I'm also removing media_optimization_unittest.cc, since it only tested the
suspension logic and nothing else.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/2119503002 .
Cr-Commit-Position: refs/heads/master@{#13355}
Before this change the ChannelBuffer had a fixed number of channels. This meant for example that when the Beamformer would reduce the number of channels to one, the merging filter bank was still merging all the channels, which was unnecessary since they were not processed and just discarded later. This change doesn't change the signal at all. It just reflects the number of channels in the ChannelBuffer, reducing the complexity.
R=henrik.lundin@webrtc.org, peah@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/2053773002 .
Cr-Commit-Position: refs/heads/master@{#13352}
Reason for revert:
voice_engine_unittests: FilePlayerTest.PlayWavPcm16File and FilePlayerTest.PlayWavPcmuFile fail on 32-bit android (android_rel and android-dbg try bots, Android32 Tests (L Nexus5) and Android32 Tests (L Nexus7.2) build bots).
Not sure why this would happen, since I just moved the test without modifying it. Some test filtering that no longer manages to disable them? Anyway, reverting until I know how to fix.
This was actually caught by the try bots, but I missed it because I was manually ignoring them because of an error with the bots. :-(
Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> R=perkj@webrtc.org, solenberg@webrtc.org
>
> Committed: 65874b163eTBR=perkj@webrtc.org,solenberg@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/2092633002
Cr-Commit-Position: refs/heads/master@{#13267}
Introduced new class DelayBasedProbingEstimator which is a copy of
RemoteBitrateEstimatorAbsSendTime with only minor changes. This makes probing
more reliable but is still not usable for mid-call probing.
BUG=
Review-Url: https://codereview.webrtc.org/2038023002
Cr-Commit-Position: refs/heads/master@{#13195}
Changes:
* Enabled protobuf for iOS globally.
* Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global
scope similar to GYP since tests depend on it.
* Added missing rtc_libvpx_build_vp9 variable.
* Moved out audio_coding defines into .gni file to avoid code duplication
* Renamed files to avoid object naming conflicts that GN disallows:
* webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc}
* webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc}
BUG=webrtc:5949
TESTED=Built and ran the tests on Mac. Also ran:
gn gen out/Default --args="rtc_enable_bwe_test_logging=true"
and verified that more objects are being built (1885 vs 1883)
when compiling modules_unittests.
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2041233006
Cr-Commit-Position: refs/heads/master@{#13108}
Sync the GYP and GN targets and update the name of the GN one
to 'remote_bitrate_estimator'.
Move the GYP variable 'enable_bwe_test_logging' into the local scope.
Remove redundant entries in modules.gyp.
These are preparations related to the GN migration.
BUG=webrtc:5949
TESTED=Ran GYP with the default variables and with
-Denable_bwe_test_logging=1. Compiled remote_bitrate_estimator
and verified that bwe_test_logging.cc is compiled only when
set.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2040313004
Cr-Commit-Position: refs/heads/master@{#13087}
The changes are done in several patches in order to make
the review easier.
NOTRY=True
BUG=webrtc:5949
Review-Url: https://codereview.webrtc.org/2051443002
Cr-Commit-Position: refs/heads/master@{#13068}
This CL adds support for an extension on RTP frames to allow the sender
to specify the minimum and maximum playout delay limits.
The receiver makes a best-effort attempt to keep the capture-to-render delay
within this range. This allows different types of application to specify
different end-to-end delay goals. For example gaming can support rendering
of frames as soon as received on receiver to minimize delay. A movie playback
application can specify a minimum playout delay to allow fixed buffering
in presence of network jitter.
There are no tests at this time and most of testing is done with chromium
webrtc prototype.
On chromoting performance tests, this extension helps bring down end-to-end
delay by about 150 ms on small frames.
BUG=webrtc:5895
Review-Url: https://codereview.webrtc.org/2007743003
Cr-Commit-Position: refs/heads/master@{#13059}
This cl split the class MediaOptimization into two parts. One that deals with frame dropping and stats and one new class called ProtectionBitrateCalculator that deals with calculating the needed FEC parameters and how much of the estimated network bitrate that can be used by an encoder
Note that the logic of how FEC and the needed bitrates is not changed.
BUG=webrtc:5687
R=asapersson@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1972083002 .
Cr-Commit-Position: refs/heads/master@{#13018}
The VUI part an SPS may specify max_num_reorder_frames and
max_dec_frame_buffering. These may cause a decoder to buffer a number
of frame prior allowing decode, leading to delays, even if no frames
using such references (ie B-frames) are sent.
Because of this we update any SPS block emitted by the encoder.
Also, a bunch of refactoring of H264-related code to reduce code
duplication.
BUG=
Review-Url: https://codereview.webrtc.org/1979443004
Cr-Commit-Position: refs/heads/master@{#13010}
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}
Wires up existing libvpx_build_vp9==0 GYP flag into WebRTC and makes VP9
optional. Change is GYP only for now since libvpx's GN files build VP9
unconditionally.
BUG=webrtc:5884
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1970343002 .
Cr-Commit-Position: refs/heads/master@{#12741}
This reverts commit e30c27205148b34ba421184efe65f6a0780b436d (https://codereview.webrtc.org/1958053002/)
Original reverted cl is in patch set #1.
Changes in following patch sets.
The cl now also make sure SendPacer starts with the configured bitrate provided in a call to CongestionController::SetBweBitrates)()
It turns out that the failing tests in 609816 is due to a bug in the current code that runs the proper at 300kbit regardless of configured start bitrate.
Original cl description:
Remove SendPacer from ViEEncoder
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator)
BUG=chromium:609816, webrtc:5687
TBR=mflodman@webrtc.org
NOTRY=True // Due to bug in android_x86 cq builder....
Review-Url: https://codereview.webrtc.org/1958113003
Cr-Commit-Position: refs/heads/master@{#12688}
This reverts commit 825eb58d59940a4c3c9837595c4b3b07059c93ca.
This Relands the cl reviewed in https://codereview.webrtc.org/1917793002/
patchset #1 is a pure reland.
patchset #2 fix an overflow in BitrateProber that caused WebRtcVideoChannel2BaseTest.TwoStreamsSendAndReceive to fail.
Original cl description:
Remove SendPacer from ViEEncoder
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator)
R=stefan@webrtc.orgTBR=mflodman@webrtc.org
BUG=webrtc:5687
Review URL: https://codereview.webrtc.org/1947873002 .
Cr-Commit-Position: refs/heads/master@{#12630}
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator)
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1917793002
Cr-Commit-Position: refs/heads/master@{#12620}
This is a first draft of what we're hoping to use to create all
AudioDecoder instances. Follow-up CLs will start using this internally
in NetEq instead of calling constructors manually.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/1917163002
Cr-Commit-Position: refs/heads/master@{#12548}
The new class is intended to be used as a central time-keeping object
inside NetEq. The actual use of the class will come in subsequent
changes.
BUG=webrtc:5608
Review URL: https://codereview.webrtc.org/1910523003
Cr-Commit-Position: refs/heads/master@{#12477}
