Previously ProbeController would overflow int when calculating
min_bitrate_to_probe_further_bps and when probing bitrate is
above 17 Mbps. The problem was introduced in
https://codereview.webrtc.org/2504023002. Fixed ProbeController to use
int64_t internally for bitrate calculations.
BUG=6332
Review-Url: https://codereview.webrtc.org/2574533002
Cr-Commit-Position: refs/heads/master@{#15642}
In order for the VCMTiming object to be correctly updated with decoding timings
when running the WebRTC-NewVideoJitterBuffer experiment the VCMTiming object
has to be available in both the VideoReceiver and the video_coding::FrameBuffer
class. Therefore the VCMTiming object is created in VideoRecieveStream and
then passed to VideoReceiver/video_coding::FrameBuffer as they are constructed.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2575473004
Cr-Commit-Position: refs/heads/master@{#15638}
Reason for revert:
Multiple definitions of TestEstimator
Original issue's description:
> Avoid precision loss in TrendlineEstimator by passing the arrival time as an int64_t instead of a double.
>
> BUG=webrtc:6884
>
> Committed: https://crrev.com/c12cbaf9dd0729dd45f3fc45a1938d1b3455e40a
> Cr-Commit-Position: refs/heads/master@{#15631}
TBR=stefan@webrtc.org,brandtr@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6884
Review-Url: https://codereview.webrtc.org/2582513002
Cr-Commit-Position: refs/heads/master@{#15636}
Reason for revert:
Multiple definitions of TestEstimator
Original issue's description:
> Pass arrival time as an int64_t rather than a double to the MedianSlopeEstimator to avoid precision loss.
>
> Also clean up the unit test.
>
> BUG=webrtc:6892
>
> Committed: https://crrev.com/ebcbcc3b2451f5c4fb07f7b37815bd54f364d057
> Cr-Commit-Position: refs/heads/master@{#15634}
TBR=brandtr@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:6892
Review-Url: https://codereview.webrtc.org/2572353003
Cr-Commit-Position: refs/heads/master@{#15635}
Reason for revert:
This change broke Chrome too. It's stats processing wants to make a copy of webrtc's stats mapping, which is no longer possible with std::unique_ptr.
Original issue's description:
> Reland of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2576673002/ )
>
> Reason for revert:
> Downstream project fixed to not make copies while iterating over the stats mapping.
>
> Original issue's description:
> > Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2567143003/ )
> >
> > Reason for revert:
> > The change from rtc::linked_ptr to std::unique_ptr broke a downstream project.
> >
> > Original issue's description:
> > > Delete rtc::linked_ptr. Only use, in statstypes.h, replaced with std::unique_ptr.
> > >
> > > BUG=webrtc:6424
> > >
> > > Committed: https://crrev.com/36f74e55792cae19db8b222c29aa38d6e0eb1225
> > > Cr-Commit-Position: refs/heads/master@{#15588}
> >
> > TBR=solenberg@webrtc.org,pthatcher@webrtc.org,hta@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:6424
> >
> > Committed: https://crrev.com/8afbc8cba65d99bb7a0feece8fb3055b144106b1
> > Cr-Commit-Position: refs/heads/master@{#15589}
>
> TBR=solenberg@webrtc.org,pthatcher@webrtc.org,hta@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:6424
>
> Committed: https://crrev.com/06035cf53abad80b0525f286a3b81e388cc7ee00
> Cr-Commit-Position: refs/heads/master@{#15627}
TBR=solenberg@webrtc.org,pthatcher@webrtc.org,hta@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2579753002
Cr-Commit-Position: refs/heads/master@{#15629}
This makes sure that the referenced stats dictionaries exist.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2577033002
Cr-Commit-Position: refs/heads/master@{#15628}
Underlying stats gatherers may otherwise default it to -1.
BUG=chromium:669877, chromium:627816
Review-Url: https://codereview.webrtc.org/2562703007
Cr-Commit-Position: refs/heads/master@{#15625}
This CL moves all codec specific definitions into their own
header files in the respective codec directory, and replaces
it with an include in the top level directory.
This is to facilitate getting the code out of the header files.
No behavioral changes are expected.
BUG=webrtc:6842
Review-Url: https://codereview.webrtc.org/2555993003
Cr-Commit-Position: refs/heads/master@{#15623}
In order to get rid of the Chromium checkout for WebRTC, the plan is to
instead of cloning all of Chromium, only clone the build, third_party and
tools sub-directories. In order to do so, we must first move all things
checked into the WebRTC tools/ directory somewhere else.
Due to many hardcoded assumptions of tools/ existing in Chrome, this
is only manageble solution to the problem.
This first step only moves stuff not used by the build system or bots,
and deletes a few unused directories.
BUG=webrtc:5006
R=henrika@webrtc.org
Review-Url: https://codereview.webrtc.org/2584433002 .
Cr-Commit-Position: refs/heads/master@{#15622}
third_party/nss is no longer used.
third_party/llvm-build and third_party/syzygy links are not be needed;
these dirs are created during 'gclient runhooks' now that we moved
to executing hooks in our own DEPS file instead of the ones in
chromium/src/DEPS as per https://codereview.webrtc.org/2524673002
BUG=webrtc:5006
NOTRY=True
TESTED=Ran all compile trybots with the --clobber flag.
Review-Url: https://codereview.webrtc.org/2570993003
Cr-Commit-Position: refs/heads/master@{#15621}
WriteIsolatedOutput() functions write large content into swarming isolated
output folder, which are useful to log large test data for debugging purpose.
BUG=webrtc:6732
TBR=holmer@webrtc.org
Review-Url: https://codereview.webrtc.org/2558693002
Cr-Commit-Position: refs/heads/master@{#15616}
This change inserts a RTC_CHECK for illegal packetization modes
when RTP packetizers are constructed.
This should help find places where this field is not initialized.
BUG=webrtc:6858
Review-Url: https://codereview.webrtc.org/2575073002
Cr-Commit-Position: refs/heads/master@{#15614}
BUG=webrtc:6649
- Supports Bluetooth Headset profile.
- Detects new BT headset:
+ enabled at start, and
+ powered on during active call.
- Enables/disables BT SCO channel when BT device is selected.
- Removes proximity sensor usage to avoid conflicts (will be added again later).
- Adds new (unused) APIs to explicitly select audio device.
- Starts routing audio to BT headset when enabled, i.e, BT is default.
Review-Url: https://codereview.webrtc.org/2501983002
Cr-Commit-Position: refs/heads/master@{#15610}
# Legal requires us to keep the original license header.
NOPRESUBMIT=true
BUG=None
Review-Url: https://codereview.webrtc.org/2574143002
Cr-Commit-Position: refs/heads/master@{#15609}
This is a prerequisite to decode fmtp sprop-parameter-sets into
the right encoding for H264SpsPpsTracker.
# Legal requires us to keep the original license header.
NOPRESUBMIT=true
BUG=webrtc:5948
Review-Url: https://codereview.webrtc.org/2539153002
Cr-Commit-Position: refs/heads/master@{#15604}
Add a flag that makes the script more useful for local development, when
you normally develop on a local branch.
BUG=webrtc:5006
NOTRY=True
Review-Url: https://codereview.webrtc.org/2566223007
Cr-Commit-Position: refs/heads/master@{#15602}
The previous CL that added the ability to add
and artificial nearend signal had an issue with
null pointer access.
This is addressed in this CL.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2573033003
Cr-Commit-Position: refs/heads/master@{#15600}
Reason for revert:
Crashes perf tests, e.g.,
./out/Debug/webrtc_perf_tests --gtest_filter='FullStackTest.ScreenshareSlidesVP8_2TL_VeryLossyNet'
dies with an assert related to rtc::Optional.
Original issue's description:
> Delete VideoFrame default constructor, and IsZeroSize method.
>
> This ensures that the video_frame_buffer method never can return a
> null pointer.
>
> BUG=webrtc:6591
>
> Committed: https://crrev.com/bfcf561923a42005e4c7d66d8e72e5932155f997
> Cr-Commit-Position: refs/heads/master@{#15574}
TBR=magjed@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:6591
Review-Url: https://codereview.webrtc.org/2574123002
Cr-Commit-Position: refs/heads/master@{#15597}
This test is flaky on all platforms, not just Android. Disabling it entirely until webrtc:6057 is fixed.
BUG=webrtc:6057
Review-Url: https://codereview.webrtc.org/2568743007
Cr-Commit-Position: refs/heads/master@{#15594}
