The H264SpsPpsTracker class:
- Keeps track of all received SPS/PPS.
- Decides whether a packet should be inserted into the PacketBuffer or not.
- Don't insert if this packet only contains SPS and/or PPS.
- Don't insert if this is the first packet of and IDR and we have not
received the required SPS/PPS.
- Insert start codes, and in the case of the first packet of an IDR prepend
the bitstream with the given SPS/PPS for this IDR.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2466993003
Cr-Commit-Position: refs/heads/master@{#14906}
This change copies ScreenCapturerDifferWrapper to a new
DesktopCapturerDifferWrapper, and adds DesktopCapturer::CreateWindowCapturer and
DesktopCapturer::CreateScreenCapturer functions to replace
WindowCapturer::Create and ScreenCapturer::Create.
BUG=webrtc:6513
Committed: https://crrev.com/b763e39beba92b45baa09542f949daabbe6258a3
Review-Url: https://codereview.webrtc.org/2468753002
Cr-Original-Commit-Position: refs/heads/master@{#14880}
Cr-Commit-Position: refs/heads/master@{#14903}
Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
These methods should no longer be used anywhere and it's safe to remove
them.
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2405173006
Cr-Commit-Position: refs/heads/master@{#14902}
In the new APM statistics interface, the default values did not match those previously used in AudioSendStream::Stats.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2469783002
Cr-Commit-Position: refs/heads/master@{#14896}
This fix is made to remove the discrepancy between GYP and GN audio_decoder_factory_interface target.
BUG=webrtc:6412
NOTRY=True
Review-Url: https://codereview.webrtc.org/2472643003
Cr-Commit-Position: refs/heads/master@{#14894}
This CL replaces the use of webrtc::Video as an internal
variable in the H.264 encoder with the specific fields
that are used by this encoder.
In support of refactorings discussed around:
BUG=600254
Committed: https://crrev.com/2549437b5ccf5aae2e6f1a1491c5f505d1859f9c
Review-Url: https://codereview.webrtc.org/2468903003
Cr-Original-Commit-Position: refs/heads/master@{#14887}
Cr-Commit-Position: refs/heads/master@{#14892}
- Add histogram: "WebRTC.Video.RtpToNtpFreqOffsetInKhz"
The absolute value of the difference between the estimated frequency during RTP timestamp to NTP time conversion and the actual value (i.e. 90 kHz) is measured per received video frame. The max offset during 40 second intervals is stored. The average of these stored offsets per received video stream is recorded when a stream is removed.
Updated rtp_to_ntp.cc:
- Add validation for only inserting newer RTCP sender reports to the rtcp list.
- Move calculation of frequency/offset (from RTP/NTP timestamp pairs) to UpdateRtcpList. Calculated when a new RTCP SR in inserted (and not in RtpToNtpMs per packet).
BUG=webrtc:6579
Review-Url: https://codereview.webrtc.org/2385763002
Cr-Commit-Position: refs/heads/master@{#14891}
Reason for revert:
Landed the wrong patchset. Nothing broken.
Original issue's description:
> Remove webrtc::Video from H264 encoder internals
>
> This CL replaces the use of webrtc::Video as an internal
> variable in the H.264 encoder with the specific fields
> that are used by this encoder.
>
> In support of refactorings discussed around:
>
> BUG=600254
>
> Committed: https://crrev.com/2549437b5ccf5aae2e6f1a1491c5f505d1859f9c
> Cr-Commit-Position: refs/heads/master@{#14887}
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=600254
Review-Url: https://codereview.webrtc.org/2472673002
Cr-Commit-Position: refs/heads/master@{#14888}
This CL replaces the use of webrtc::Video as an internal
variable in the H.264 encoder with the specific fields
that are used by this encoder.
In support of refactorings discussed around:
BUG=600254
Review-Url: https://codereview.webrtc.org/2468903003
Cr-Commit-Position: refs/heads/master@{#14887}
Reason for revert:
Prevents WebRTC rolls into Chrome.
https://build.chromium.org/p/chromium.linux/builders/Blimp%20Linux%20%28dbg%29/builds/14848/steps/compile/logs/stdio
The reason for reverting is: Breaks
https://build.chromium.org/p/chromium.linux/builders/Blimp%20Linux%20%28dbg%2...
[881/894] SOLINK ./libcontent.so
FAILED: libcontent.so libcontent.so.TOC
../../third_party/webrtc/modules/desktop_capture/desktop_capturer.cc:45: error:
undefined reference to
'webrtc::DesktopCapturer::CreateRawWindowCapturer(webrtc::DesktopCaptureOptions
const&)'
../../third_party/webrtc/modules/desktop_capture/desktop_capturer.cc:56: error:
undefined reference to
'webrtc::DesktopCapturer::CreateRawScreenCapturer(webrtc::DesktopCaptureOptions
const&)'
clang: error: linker command failed with exit code 1 (use -v to see invocation)
ninja: build stopped: subcommand failed.
Original issue's description:
> Add CreateWindowCapturer() and CreateScreenCapturer() in DesktopCapturer
>
> This change copies ScreenCapturerDifferWrapper to a new
> DesktopCapturerDifferWrapper, and adds DesktopCapturer::CreateWindowCapturer and
> DesktopCapturer::CreateScreenCapturer functions to replace
> WindowCapturer::Create and ScreenCapturer::Create.
>
> BUG=webrtc:6513
>
> Committed: https://crrev.com/b763e39beba92b45baa09542f949daabbe6258a3
> Cr-Commit-Position: refs/heads/master@{#14880}
TBR=sergeyu@chromium.org,zijiehe@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2471773002
Cr-Commit-Position: refs/heads/master@{#14884}
This change copies ScreenCapturerDifferWrapper to a new
DesktopCapturerDifferWrapper, and adds DesktopCapturer::CreateWindowCapturer and
DesktopCapturer::CreateScreenCapturer functions to replace
WindowCapturer::Create and ScreenCapturer::Create.
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2468753002
Cr-Commit-Position: refs/heads/master@{#14880}
With this change, instead of
RTC_DCHECK_GE(unsigned_var, 17u);
we can simply write
RTC_DCHECK_GE(unsigned_var, 17);
or even
RTC_DCHECK_GE(unsigned_var, -17); // Always true.
and the mathematically sensible thing will happen.
Perhaps more importantly, we can replace checks like
// index is size_t, num_channels is int.
RTC_DCHECK(num_channels >= 0
&& index < static_cast<size_t>(num_channels));
or, even worse, just
// Surely num_channels isn't negative. That would be absurd!
RTC_DCHECK_LT(index, static_cast<size_t>(num_channels));
with simply
RTC_DCHECK_LT(index, num_channels);
In short, you no longer have to keep track of the signedness of the arguments, because the sensible thing will happen.
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2459793002
Cr-Commit-Position: refs/heads/master@{#14878}
In this CL:
- Don't insert a packet if we have explicitly cleared past it.
- Added some logging to ExpandBufferSize.
- Renamed IsContinuous to PotentialNewFrame.
- Unittests updated/added for this new behavior.
- Refactored TestPacketBuffer unittests.
BUG=webrtc:5514
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2399373002 .
Cr-Commit-Position: refs/heads/master@{#14871}
This is a proposal for a new RTCP message. Feel free to comment on the
message structure, selected type ids etc, as well as code for
serialization/deserialization. Once we agree on this, I'll continue
with wiring it up in the actual rtcp sender and receiver.
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2306873003
Cr-Commit-Position: refs/heads/master@{#14867}
The timestamps are 32 bit and can (conceivably) be spaced far enough
apart for the calculation, which is done in Q4, to overflow.
BUG=chromium:653268
Review-Url: https://codereview.webrtc.org/2460393002
Cr-Commit-Position: refs/heads/master@{#14856}
Design of individual block in ExtendedReports packet suggest there is
no point to have more than one block per type.
This CL reduce complexity of having several blocks of the same type in
same report.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2378113002
Cr-Commit-Position: refs/heads/master@{#14855}
Contains fixes for a non-perfect implementation in https://codereview.webrtc.org/2328433003/
Summary:
Adds WebRTC.Audio.RecordedOnlyZeros UMA stat when recording stops if:
- All level estimates during the audio session were zero, and
- If the audio session was longer than 10 seconds.
Adds four simple methods to the AudioDeviceBuffer (ADB) class to allow the ADM
to update the ADB about when media starts and stops in both directions.
Moves any "critical" parst out frome the timer (based on task queue) and ensures
that it only does trivial logging tasks.
The task queue is now owned by a unique pointer to improve control of when it
starts and stops.
Adds time measurements (for logging) of both total time playing out and total
recording time. Units are in milliseconds.
BUG=webrtc:6592
Review-Url: https://codereview.webrtc.org/2445363003
Cr-Commit-Position: refs/heads/master@{#14854}
The new code is only exercised in tests so far. The H264 profile-level-id
parsing is not complete, but it should be enough for our purposes for
now.
BUG=webrtc:6400,webrtc:6337
Review-Url: https://codereview.webrtc.org/2459633002
Cr-Commit-Position: refs/heads/master@{#14850}
This change will allow for a audio source to report its sampling rate
to the audio mixer. It is needed in order to mix at a lower sampling
rate. Mixing at a lower sampling rate can in many cases lead to big
efficiency improvements, as reported by experiments.
The code affected is all implementations of the Source interface:
AudioReceiveStream and a mock class. The AudioReceiveStream now
queries its underlying voe::Channel object for the needed frequency.
Note that the changes to the mixing algorithm are done in a later CL.
BUG=webrtc:6346
NOTRY=True
TBR=solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2448113009
Cr-Commit-Position: refs/heads/master@{#14839}
I have chosen part of 2435603010 changes to compose this change.
According to the discussion we have made in previous change, this CL contains,
1. Source structure to represent a source of a DesktopCapturer.
2. GetSourceList / SelectSource / FocusOnSelectedSource functions in
DesktopCapturer.
3. ScreenCapturer and WindowCapturer forward corresponding functions to the new
DesktopCapturer APIs.
After this change, We can remove WindowCapturer & ScreenCapturer references from
Chromium, and use the new APIs.
BUG=webrtc:6513
Committed: https://crrev.com/9cb0b3b4ac916cdf52d97a63d923dfbe73f0541e
Review-Url: https://codereview.webrtc.org/2452263003
Cr-Original-Commit-Position: refs/heads/master@{#14830}
Cr-Commit-Position: refs/heads/master@{#14832}
Reason for revert:
Build break in Chromium
Original issue's description:
> Add DesktopCapturer GetSourceList SelectSource FocusOnSelectedSource functions
>
> I have chosen part of 2435603010 changes to compose this change.
> According to the discussion we have made in previous change, this CL contains,
> 1. Source structure to represent a source of a DesktopCapturer.
> 2. GetSourceList / SelectSource / FocusOnSelectedSource functions in
> DesktopCapturer.
> 3. ScreenCapturer and WindowCapturer forward corresponding functions to the new
> DesktopCapturer APIs.
>
> After this change, We can remove WindowCapturer & ScreenCapturer references from
> Chromium, and use the new APIs.
>
> BUG=webrtc:6513
>
> Committed: https://crrev.com/9cb0b3b4ac916cdf52d97a63d923dfbe73f0541e
> Cr-Commit-Position: refs/heads/master@{#14830}
TBR=sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2464553002
Cr-Commit-Position: refs/heads/master@{#14831}
I have chosen part of 2435603010 changes to compose this change.
According to the discussion we have made in previous change, this CL contains,
1. Source structure to represent a source of a DesktopCapturer.
2. GetSourceList / SelectSource / FocusOnSelectedSource functions in
DesktopCapturer.
3. ScreenCapturer and WindowCapturer forward corresponding functions to the new
DesktopCapturer APIs.
After this change, We can remove WindowCapturer & ScreenCapturer references from
Chromium, and use the new APIs.
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2452263003
Cr-Commit-Position: refs/heads/master@{#14830}
renamed kName to kUri and make it more const.
remove IsSupportedBy to reduce header dependency.
BUG=webrtc:1994
Review-Url: https://codereview.webrtc.org/2457783005
Cr-Commit-Position: refs/heads/master@{#14825}
This algorithm calculates an estimate of the Pearson product-moment correlation coefficient between the power of 10ms audio buffers taken from the render and capture sides, for various different delay values.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2419563003
Cr-Commit-Position: refs/heads/master@{#14824}
Introduce a convention on categorizing GN targets:
1. Production code
2. Tests
3. Examples
4. Tools
The first two have targets spread out all over the tree,
while the latter are isolated to examples/ and tools/ directories.
Another new convention: Each directory's BUILD.gn file shall contain
a target named similar to the directory name. This target shall
contain the 'most common' production code, i.e. so that most
consumers of the directory can depend on only the directory
(which implicitly means that target in GN).
//webrtc:webrtc_tests is changed to depend on all WebRTC tests.
From now on, it's necessary to add new test targets to this dependency
tree in order to get them compiled.
Two new group targets are created:
//webrtc/modules/audio_coding:audio_coding_tests
//webrtc/modules/audio_processing:audio_processing_tests
to reduce the long list of tests in //webrtc:webrtc_tests.
Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted
to the root target, to avoid circular dependencies due to the monolithic
property of these targets (a problem we've had in the past).
The 'root' target at the top level is renamed to 'default', which means GN will
build this target instead of _all_ generated targets
(see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default).
This target now depends on everything we want to build, meaning all targets now
explicitly needs to be wired up from the root target in order to get build.
Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build.
BUG=webrtc:6440
TESTED=git cl try --clobber
NOTRY=True
Review-Url: https://codereview.webrtc.org/2441383002
Cr-Commit-Position: refs/heads/master@{#14821}
This CL does not contain the actual algorithm, but only creates an empty processing component and connects the right signals to it. The algorithm will be added in a follow-up CL.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2405403003
Cr-Commit-Position: refs/heads/master@{#14820}
The test program audioproc_f does not call the stream_analog_level
method. This should be done do
1) Ensure that proper log output is produced when reproducing a call.
2) Ensure that this method is properly tested.
3) Ensure that the correct side-effects are triggered (this method
is not const).
BUG=webrtc:6564
Review-Url: https://codereview.webrtc.org/2449043008
Cr-Commit-Position: refs/heads/master@{#14817}
Reason for revert:
This CL breaks internal dependencies.
Original issue's description:
> New statistics interface for APM
>
> This adds functions to enable and retrieve statistics from APM. These functions are not yet called, which will happen in a follow-up CL.
>
> BUG=webrtc:6525
>
> Committed: https://crrev.com/8b8d3e4c30e8ea3846b58dfd36d1fd35a7799df4
> Cr-Commit-Position: refs/heads/master@{#14810}
TBR=peah@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2456333002
Cr-Commit-Position: refs/heads/master@{#14814}
aecdump recordings in an efficient manner, it is
important to be able to use a standardized analysis
script. For this to be feasible, data log points should
be present.
This CL adds those logpoints as well as the framework
needed to for those to work.
BUG=webrtc:6564
Review-Url: https://codereview.webrtc.org/2457783003
Cr-Commit-Position: refs/heads/master@{#14812}
This adds functions to enable and retrieve statistics from APM. These functions are not yet called, which will happen in a follow-up CL.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2433153003
Cr-Commit-Position: refs/heads/master@{#14810}
Replaced with a size() method, returning the corresponding attribute
(_length) of the underlying EncodedImage.
BUG=None
Review-Url: https://codereview.webrtc.org/2444193010
Cr-Commit-Position: refs/heads/master@{#14809}
Fixing compile error for non Intel platforms
when using C implemented fft functions
(for example, optimizations are not implemented for mips64el
and C functions must be used)
Adding bypass of presubmit to avoid code style and header
errors caused by the fact that files with legacy code are
being renamed.
NOPRESUBMIT=true
BUG=webrtc:6595
TEST=gn gen out-gn/mips64-android-webrtc --args="is_debug=false target_os=\"android\" target_cpu=\"mips64el\""
ninja -C out-gn/mips64-android-webrtc audio_processing
Review-Url: https://codereview.webrtc.org/2442773002
Cr-Commit-Position: refs/heads/master@{#14808}
This is a trivial change, I just cutted and pasted part of the code in
screen_capturer_unittest.cc to screen_capturer_integration_test.cc, removed
DISABLED_ prefixes, and updated build file.
BUG=webrtc:6366
Review-Url: https://codereview.webrtc.org/2444583002
Cr-Commit-Position: refs/heads/master@{#14806}
This helps a lot to avoid reducing the bitrate too quickly when there's a short period of very few packets delivered, followed by the rate resuming at the regular rate. It specifically avoids the BWE going down to super low values as a response delay spikes.
BUG=webrtc:6566
R=terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2422063002 .
Cr-Commit-Position: refs/heads/master@{#14802}
