This is a reland of e6cefdf9c572cdce55ff0497ad6e516c76132ee8.
Original change's description:
> Moved congestion controller to goog_cc folder.
>
> Bug: webrtc:8415
> Change-Id: I2070da0cacf1dbfc4b6a89285af3e68fd03497ab
> Reviewed-on: https://webrtc-review.googlesource.com/43841
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21928}
Bug: webrtc:8415
Change-Id: Ib5cf8641466655d64ac80f720561817f4cab49a9
Reviewed-on: https://webrtc-review.googlesource.com/53062
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22244}
The functionality in network_control is by design not rtp specific and
should live outside the rtp directory in congestion_controller.
Bug: webrtc:8415
Change-Id: I68778f1a630c8e93950040bc56ee864c201b5990
Reviewed-on: https://webrtc-review.googlesource.com/55383
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22240}
* Do not simulate freeze in decoded output file when frames have been dropped.
* Add more DCHECKs and consts.
* Remove unused members |num_encoded_frames_| and |num_decoded_frames_|.
* Move SdpVideoFormat conversion to TestConfig.
Bug: webrtc:8448
Change-Id: Ia879141f36dc23427cd1abcaa66716656fbaac2a
Reviewed-on: https://webrtc-review.googlesource.com/56802
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22239}
Makes the new task queue based congestion controller implement the
SendSideCongestionControllerInterface.
Bug: webrtc:8415
Change-Id: I3dfe11c2eb200bc8d85c83edf78d7fdd0129bbff
Reviewed-on: https://webrtc-review.googlesource.com/56781
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22235}
This reverts commit d2ed0a4c9e7f04060d8e3358eb0006c31579bb86.
Reason for revert: Breaks downstream projects.
Original change's description:
> Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers.
>
> temporal_layer_thresholds_bps served only one purpose: its size was used
> to infer number of temporal layers. I replaced it with num_temporal_layers,
> which does what is says.
>
> The practical reason for this change is the need to have possibility to
> distinguish between cases when VP9 SVC temporal layering was/not set
> through field trial. That was not possible with
> temporal_layer_thresholds_bps[] because empty vector means 1 temporal
> layer.
>
> Bug: webrtc:8518
> Change-Id: I275ec3a8c74e8ba409eb049878199f132a20ec51
> Reviewed-on: https://webrtc-review.googlesource.com/58084
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22230}
TBR=sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org
Change-Id: Ic2940f7f78a74312170940d51ad8967cde8ad42f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8518
Reviewed-on: https://webrtc-review.googlesource.com/58902
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22234}
temporal_layer_thresholds_bps served only one purpose: its size was used
to infer number of temporal layers. I replaced it with num_temporal_layers,
which does what is says.
The practical reason for this change is the need to have possibility to
distinguish between cases when VP9 SVC temporal layering was/not set
through field trial. That was not possible with
temporal_layer_thresholds_bps[] because empty vector means 1 temporal
layer.
Bug: webrtc:8518
Change-Id: I275ec3a8c74e8ba409eb049878199f132a20ec51
Reviewed-on: https://webrtc-review.googlesource.com/58084
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22230}
This CL activates unit tests for the task queue based send side
congestion controller that will replace the current one in the future.
To be able to have the tests run side by side with the tests of the old
congestion controller, the old tests have been prefixed with "Legacy".
This CL also contains some minor fixes to the new congestion controller
code.
Bug: webrtc:8415
Change-Id: I5e7474d42f17fcbfef402e26f638846fa3424695
Reviewed-on: https://webrtc-review.googlesource.com/55381
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22229}
SetTransportOverhead was used by send streams to signal the packet
overhead that they received from Call. However, call receives the value
from OnNetworkRouteChanged in WebRtcVideoChannel and
WebRtcVoiceMediaChannel which is already propagated to
RtpTransportControllerSend. By skipping the round trip, the interface on
the rtp transport controller can be reduced.
Bug: None
Change-Id: I759b1207aab214bbc2b993106f6ff7cc24e177f7
Reviewed-on: https://webrtc-review.googlesource.com/57182
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22226}
Uploading of the file to device causes timeouts. I removed it from
resources for iOS and Android builds and disabled the test that used
the file since we don't really need to run it on test bots.
Bug: webrtc:8936
Change-Id: Ia5e04c4630544eca8e56826c9e89c9c9f4dcb600
Reviewed-on: https://webrtc-review.googlesource.com/58090
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22222}
This makes it easier to have the new and the old send side congestion
controller side by side. This namespace is only temporary. As soon the
new task queue based congestion controller is fully functional, the old
will be deprecated and removed together with the temporary namespace.
Bug: webrtc:8415
Change-Id: Ie817511345c91cab2ebca68f038075875c7e6529
Reviewed-on: https://webrtc-review.googlesource.com/56720
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22221}
Added sanity checks to interval budget to protect against integer
overflow. To avoid tests failing due to initializing paced sender
in another time domain, the elapsed time that is input into the
interval budget is limited to max 2 seconds.
Bug: webrtc:8942
Change-Id: I9ed32f059e65df7898c37bb34a008189ce79dc60
Reviewed-on: https://webrtc-review.googlesource.com/58087
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22220}
To be able to safely move over to the new code, the revised
code is added alongside the old code. Most of the files added
in this CL are more or less direct copies of the previous code.
This new version of send side congestion controller will be
activated under a field trial in a followup CL.
Bug: webrtc:8415
Change-Id: I034e583cf891a8f6357119739a1517cc0a4abe88
Reviewed-on: https://webrtc-review.googlesource.com/53322
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22212}
The flag is passed as --isolated-script-test-perf-output=/b/whatever
on the bots, but this code expected a blank space instead of =.
Bug: webrtc:8932
Change-Id: I9ca48c9b285e365ac23a04ea2e89d9a8e75f5540
Reviewed-on: https://webrtc-review.googlesource.com/58088
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22211}
Now VCMTiming::MaxWaitingTime will not clip negative values. Thus frame
buffer will be able to distinguish between late frames and when waiting
cycle was simply interrupted by a new inserted frame right before the
waiting timer would expire.
Bug: webrtc:8917
Change-Id: I6b253f459fcb3a346064a103cc92ee332b074e1b
Reviewed-on: https://webrtc-review.googlesource.com/57741
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22210}
Allows mixing up to 4 input streams. Useful for profiling and manual
tests. Allows testing different combinations of input/output rates and
number of channels. Reads and writes WAV files. Can also configure
whether to use the Limiter component of the AudioMixer.
Bug: webrtc:8925
Change-Id: Iaf4fee5284980f6ed01f4bb721e49bb1af8dd392
Reviewed-on: https://webrtc-review.googlesource.com/56842
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22209}
The webrtc::AudioMixer uses a limiter component. This CL allows
changes the APM-AGC limiter to the APM-AGC2 limiter though a Chrome
field trial.
The AGC2 limiter has a float interface. We plan to eventually switch
to the AGC2 limiter. Therefore, we will now mix in de-interleaved
floats. Float mixing will happen both when using the old limiter and
when using the new one.
After this CL the mixer will support two limiters. The limiters have
different interfaces and need different processing steps. Because of
that, we make (rather big) changes to the control flow in
FrameCombiner. For a short while, we will mix in deinterleaved floats
when using any limiter.
Originally landed in https://webrtc-review.googlesource.com/c/src/+/56141/
Reverted in https://webrtc-review.googlesource.com/c/src/+/57940
because of both breaking compilation and having a severe error. The
error is fixed and a test is added. The compilation issue is fixed.
Bug: webrtc:8925
Change-Id: Ieba138dee9652c826459fe637ae2dccbbc06bcf0
Reviewed-on: https://webrtc-review.googlesource.com/58085
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22207}
This reverts commit bd7b461f16b53363e2c510893d8d3aade5737f3a.
Reason for revert: Broke the internal project. The issue maybe related to the apm_debug_dump configuration.
Original change's description:
> Choose between APM-AGC-Limiter and Apm-AGC2-fixed-gain_controller.
>
> The webrtc::AudioMixer uses a limiter component. This CL changes the
> APM-AGC limiter to the APM-AGC2 limiter though a Chrome field trial.
>
> The new limiter has a float interface. Since we're moving to it, we
> now mix in floats as well. After this CL the mixer will support two
> limiters. The limiters have different interfaces and need different
> processing steps. Because of that, we make (rather big) changes to the
> control flow in FrameCombiner. For a short while, we will mix in
> deinterleaved floats when using any limiter.
>
> NOTRY=true
>
> Bug: webrtc:8925
> Change-Id: Ie296c2b0d94f3f0078811a2a58f6fbf0f3e6e4a8
> Reviewed-on: https://webrtc-review.googlesource.com/56141
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22185}
TBR=gustaf@webrtc.org,aleloi@webrtc.org
Change-Id: I3dd1a2b1fca32c4dd046e6fc325744079e3ac5ca
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8925
Reviewed-on: https://webrtc-review.googlesource.com/57940
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22189}
The webrtc::AudioMixer uses a limiter component. This CL changes the
APM-AGC limiter to the APM-AGC2 limiter though a Chrome field trial.
The new limiter has a float interface. Since we're moving to it, we
now mix in floats as well. After this CL the mixer will support two
limiters. The limiters have different interfaces and need different
processing steps. Because of that, we make (rather big) changes to the
control flow in FrameCombiner. For a short while, we will mix in
deinterleaved floats when using any limiter.
NOTRY=true
Bug: webrtc:8925
Change-Id: Ie296c2b0d94f3f0078811a2a58f6fbf0f3e6e4a8
Reviewed-on: https://webrtc-review.googlesource.com/56141
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22185}
In rare and pathological circumstances, it could happen that the input
length to the merge function is very short. This CL will avoid one of
the problems with out-of-bounds read that could result from this.
Bug: chromium:799499
Change-Id: I6bde105ae88f9d130764b6dfb3d25443d07e214b
Reviewed-on: https://webrtc-review.googlesource.com/57582
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22180}
This CL fixes the bug that we must have some padding for H264 codec
when used in Multiplex encoder/decoder, otherwise H264 decoder will
crash.
And this CL fixes a bug that potential infinite loop exists in
MultiplexEncoderAdapter
Bug: webrtc:8921
Change-Id: I4124579c31ee69f72e29d118378aa1f8b3f05eb4
Reviewed-on: https://webrtc-review.googlesource.com/56960
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22175}
Pass pointer to application_data from RtpPacketToSend arriving via RtpSender::SendToNetwork through to Transport::SendRtp, in PacketOptions.
Bug: webrtc:8906
Change-Id: Ie75013ed472710f4efcfbcc160e46a6119a1f41d
Reviewed-on: https://webrtc-review.googlesource.com/55600
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dino Radaković <dinor@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22174}
This is a reland of c75f1e45093a8d5cc62937c7708b87aa5c5bf0b0.
Original change's description:
> Reduce locking in VideoReceiver and check the threading model.
>
> Note: This is a subset of code that was previously reviewed here:
> - https://codereview.webrtc.org/2764573002/
>
> * Added two notification methods, DecoderThreadStarting() and DecoderThreadStopped()
> * Allows us to establish a period when the decoder thread is not running and it is
> safe to modify variables such as callbacks, that are only read when the decoder
> thread is running.
> * Allows us to DCHECK thread guarantees/correctness.
> * Allows synchronizing callbacks from the module process thread and have them only
> active while the decoder thread is running.
> * The above, allows us to establish two modes for the thread,
> single-threaded-mutable and multi-threaded-const.
> * Using that knowledge, we can remove |receive_crit_| as well as locking for a
> number of member variables.
> * Removed |VCMFrameBuffer _frameFromFile| (unused).
> * Clean up several of my TODOs
>
> Bug: webrtc:7361, chromium:695438
> Change-Id: Id0048ee9624f76103c088d02825eb5c0d6c8913c
> Reviewed-on: https://webrtc-review.googlesource.com/55000
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22133}
Bug: webrtc:7361, chromium:695438
Change-Id: I32e1dc6c62cb30ad96e6366106f39fe415de49f1
Tbr: philipel@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/56803
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22163}
Using a mock of rtp transport controller send in audio send stream unit
tests. This reduces the dependencies and makes the tests more focused
on testing the functionality of audio send stream itself.
Bug: webrtc:8415
Change-Id: Ia8d9cf47d93decc74b10ca75a6771f39df658dc2
Reviewed-on: https://webrtc-review.googlesource.com/56600
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22161}
Flakiness of the test reveals this assumption doesn't hold and shouldn't be rely on.
Currently there is no code that use it. Plans to rely on it silently adjusted.
Bug: webrtc:8610
Change-Id: Id24f2a36c8fb188b518f5301c4b278836885d140
Reviewed-on: https://webrtc-review.googlesource.com/56860
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22160}
The plan is to:
1. Move FrameObject to api/video.
2. Rename FrameObject to EncodedFrame.
3. Move EncodedFrame out of the video_coding namespace.
This is the 2nd CL.
Bug: webrtc:8909
Change-Id: I5e76a0a3b306156b8bc1de67834b4adf14bebef9
Reviewed-on: https://webrtc-review.googlesource.com/56182
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22158}
The unit tests for AudioSendStream was generating a lot of warnings
about "Uninteresting mock function call" on mocked objects. This is due
to the default gmock implementation being NaggyMock and there was no
NiceMock override.
With this change the mocks are replaced with NiceMock implementations
which do not output warnings for unexpected calls. This makes the error
output from the test runner much easier to visually parse to find the
actual errors in failing tests.
Bug: None
Change-Id: Ic40db78159536ddeaa72a468fc2cb3ec17386d44
Reviewed-on: https://webrtc-review.googlesource.com/56220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22152}
This prepares for a later CL providing two implementations of
SendSideCongestionController.
Bug: webrtc:8415
Change-Id: I890dbe4b88bf609921558e03aac66b42629857c8
Reviewed-on: https://webrtc-review.googlesource.com/56700
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22151}
MockAudioEncoder was calling a mocked Die function on itself in its
destructor. This outputs "Uninteresting mock function call" warning if
the Die call was not expected. This is true even if a NiceMock is used
to suppress the warnings.
The purpose of testing that the destructor is called might be to protect
against memory leaks when audio encoder ownership is transferred using a
raw pointer. However, this case is already covered by msan checks.
Bug: None
Change-Id: I0603c417b4b239027859228e05ebcf83ff5aaf18
Reviewed-on: https://webrtc-review.googlesource.com/56183
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22146}
This CL moves the AEC3 API call skew estimator into a separate file.
This has the advantage that it can more easily be tested.
The CL also simplifies the code and adds unittests.
Bug: webrtc:8671
Change-Id: I19bc31ca5666cdc87a1ed14770ef20ead1b5b80d
Reviewed-on: https://webrtc-review.googlesource.com/55860
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22144}
This reverts commit a4e71b9e7e59be21b98d63cf8cb676096d9c74b0.
Reason for revert: Breaking internal project
Original change's description:
> VCMGenericDecoder threading updates for all but Android.
>
> * All methods now have thread checks.
> * Variable access associated with thread checkers.
> * Remove need for |rtc::CriticalSection lock_|
>
> Since the android decoder is inherently asynchronous, and
> FrameBuffer2's decoder doesn't support posting tasks to it
> yet (for async decode completion), we need to tackle android
> separately. Once FrameBuffer2 gets changed to use a TaskQueue
> or ProcessThread, we can move Android over to delivering decoded
> frames on the right thread/queue and delete generic_decoder_android.*.
>
> Note: This is a subset of code that was previously reviewed here:
> - https://codereview.webrtc.org/2764573002/
>
> Bug: webrtc:7361, webrtc:8907, chromium:695438
> Change-Id: I118609dfa5c0f0180287d8c2b6d62987b7473c5c
> Reviewed-on: https://webrtc-review.googlesource.com/55060
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22119}
TBR=sakal@webrtc.org,tommi@webrtc.org
Change-Id: I3afe4671f9d06bb4a2b17e4f14c21d79f773e067
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7361, webrtc:8907, chromium:695438
Reviewed-on: https://webrtc-review.googlesource.com/56282
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22143}
This reverts commit c75f1e45093a8d5cc62937c7708b87aa5c5bf0b0.
Reason for revert: Breaking internal project
Original change's description:
> Reduce locking in VideoReceiver and check the threading model.
>
> Note: This is a subset of code that was previously reviewed here:
> - https://codereview.webrtc.org/2764573002/
>
> * Added two notification methods, DecoderThreadStarting() and DecoderThreadStopped()
> * Allows us to establish a period when the decoder thread is not running and it is
> safe to modify variables such as callbacks, that are only read when the decoder
> thread is running.
> * Allows us to DCHECK thread guarantees/correctness.
> * Allows synchronizing callbacks from the module process thread and have them only
> active while the decoder thread is running.
> * The above, allows us to establish two modes for the thread,
> single-threaded-mutable and multi-threaded-const.
> * Using that knowledge, we can remove |receive_crit_| as well as locking for a
> number of member variables.
> * Removed |VCMFrameBuffer _frameFromFile| (unused).
> * Clean up several of my TODOs
>
> Bug: webrtc:7361, chromium:695438
> Change-Id: Id0048ee9624f76103c088d02825eb5c0d6c8913c
> Reviewed-on: https://webrtc-review.googlesource.com/55000
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22133}
TBR=tommi@webrtc.org,philipel@webrtc.org
Change-Id: I4d78e8b2c05b36e1a3f64cb38d652579b3a23f22
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7361, chromium:695438
Reviewed-on: https://webrtc-review.googlesource.com/56280
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22141}
This is a reland of 18cf4b67ddc66041d6b114ea15d78eea74d0592b.
Original change's description:
> Base pacer padding in pause state on time since last send.
>
> This clarifies the logic behind the pacer packet interval
> in paused state and prepares for future congestion window
> functionality.
>
> Bug: None
> Change-Id: Ibf6e23f73523b43742830353915b2b94d09a6fc9
> Reviewed-on: https://webrtc-review.googlesource.com/52060
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22004}
Bug: None
Change-Id: I19fc02bc226ad59cb4cbd2a6ef8ac6f47212f834
Reviewed-on: https://webrtc-review.googlesource.com/53080
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22138}
This CL adds functionality for passing the information about the
estimated delay to the echo remover in AEC3.
The CL also adds information about how long ago the delay changed,
and how long ago the delay estimate was updated.
Bug: webrtc:8671
Change-Id: If274ffe0465eb550f3e186d0599c6dc6fef7f5e8
Reviewed-on: https://webrtc-review.googlesource.com/55261
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22137}
The plan is to:
1. Move FrameObject to api/video.
2. Rename FrameObject to EncodedFrame.
3. Move EncodedFrame out of the video_coding namespace.
This is the 1st CL.
Bug: webrtc:8909
Change-Id: I2e5100eda6c51bcefb32295e03b73cf1f5c213a4
Reviewed-on: https://webrtc-review.googlesource.com/55560
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22135}
Note: This is a subset of code that was previously reviewed here:
- https://codereview.webrtc.org/2764573002/
* Added two notification methods, DecoderThreadStarting() and DecoderThreadStopped()
* Allows us to establish a period when the decoder thread is not running and it is
safe to modify variables such as callbacks, that are only read when the decoder
thread is running.
* Allows us to DCHECK thread guarantees/correctness.
* Allows synchronizing callbacks from the module process thread and have them only
active while the decoder thread is running.
* The above, allows us to establish two modes for the thread,
single-threaded-mutable and multi-threaded-const.
* Using that knowledge, we can remove |receive_crit_| as well as locking for a
number of member variables.
* Removed |VCMFrameBuffer _frameFromFile| (unused).
* Clean up several of my TODOs
Bug: webrtc:7361, chromium:695438
Change-Id: Id0048ee9624f76103c088d02825eb5c0d6c8913c
Reviewed-on: https://webrtc-review.googlesource.com/55000
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22133}
