WebRTC standalone tests show 24% speed up on foreman_cif, 16% speed up
on a 240p clip and 11% speed up on Bridge_r180.
Bug: None
Change-Id: I433b7a8841bd9df2402575f72dd1984cc5e011a9
Reviewed-on: https://webrtc-review.googlesource.com/29260
Commit-Queue: Jerome Jiang <jianj@google.com>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21096}
Alr state is now logged by the pacer. To avoid confusion,
loopback tools will now create two separate rtc event
logs for sender and receiver calls.
Bug: webrtc:8287, webrtc:8588
Change-Id: Ib3e47d109c3a65a7ed069b9a613e6a08fe6a2f30
Reviewed-on: https://webrtc-review.googlesource.com/26880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21084}
There's no downstream use of kFileFormatCompressedFile,
kFileFormatPreencodedFile or kFileFormatPcm48kHzFile.
Bug: None
Change-Id: I66cbe71151472d6348515a2432a280acbc3bbf85
Reviewed-on: https://webrtc-review.googlesource.com/28040
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21078}
A lot of WebRTC targets were depending on //third_party/libyuv using
public_deps instead of deps. This causes issues because a the
inclusion of libyuv headers is not declared to the build system and
this creates hidden dependencies that put the modularity of the project
at risk.
Bug: webrtc:8603
Change-Id: Ide0ceb84eb5640ae664dc782f3a722b55c3b601a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/28120
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21039}
In particular, PlayoutStereoData and StartPlayingAudioFile. This also
eliminates the dependency on system_wrappers FileWrapper.
Bug: None
Change-Id: I61df1eea1ad5f5035e36c8229febbf3668808f65
Reviewed-on: https://webrtc-review.googlesource.com/28121
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21038}
This will also cause us to use the new Android HardwareVideoEncoder,
instead of the deprecated MediaCodecVideoEncoderFactory. Unfortunately,
the new HW encoder does not seem to work as good as the old (or the new
encoder is more strict with return values or something). I don't think
it adds much value to continue testing the deprecated encoder, so I
filed a bug for fixing the new encoder, and in this CL I disabled the
tests on Android. I want to remove as many places as possible where we
use the old WebRtcVideoEncoderFactory interface, because it makes it
more difficult to migrate to the new interface.
Bug: webrtc:7925
Change-Id: If8e34752148a5e5139944d2dfbe7e231fe58aeb9
Reviewed-on: https://webrtc-review.googlesource.com/27540
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21037}
This reverts commit 0af8370cb38b0b0f35f4ed4ec4237d0e6c7d59da.
Reason for revert: Breaks downstream
Original change's description:
> Fix all circular deps in audio_processing (but one).
>
> Arguably we should add a few more targets, for instance a utility
> target, but I tried to create as few targets as possible here based on
> the current usage.
>
> Bug: webrtc:6828
> Change-Id: If2740de2e4374eeae64b3d7599a52bb051594c6a
> Reviewed-on: https://webrtc-review.googlesource.com/28020
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21025}
TBR=phoglund@webrtc.org,peah@webrtc.org
Change-Id: I423f027f6919cf4eb44b4e08c7cb38f0506ad0d7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6828
Reviewed-on: https://webrtc-review.googlesource.com/28940
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21027}
Arguably we should add a few more targets, for instance a utility
target, but I tried to create as few targets as possible here based on
the current usage.
Bug: webrtc:6828
Change-Id: If2740de2e4374eeae64b3d7599a52bb051594c6a
Reviewed-on: https://webrtc-review.googlesource.com/28020
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21025}
This CL centralizes the render buffering in AEC3 so that all render
buffers are updated and synchronized/aligned with the render alignment
buffer.
Bug: webrtc:8597, chromium:790905
Change-Id: I8a94e5c1f27316b6100b420eec9652ea31c1a91d
Reviewed-on: https://webrtc-review.googlesource.com/25680
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20989}
This reverts commit 8c319e951a6c59212e23af858a4c51d28b4eedc1.
Reason for revert: Increase in dropped frames and decreased send bandwidth in perf tests.
Original change's description:
> Now calculates RTT in SendSideCongestionController.
>
> Moved calculation of round trip time from transport feedback adapter to send side congestion
> controller. This reduces the role of the transport specific transport feedback adapter and
> gives more power to the congestion controller to decide how the feedback rtt should be
> calculated and used.
>
> Bug: webrtc:8415
> Change-Id: I7878d9fb32c3f4ed11993a6f39e6d9c69fab190a
> Reviewed-on: https://webrtc-review.googlesource.com/27980
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20973}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: I993d00de7171a163a41b486d68b9255fd5c0f5da
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/28300
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20984}
Moved calculation of round trip time from transport feedback adapter to send side congestion
controller. This reduces the role of the transport specific transport feedback adapter and
gives more power to the congestion controller to decide how the feedback rtt should be
calculated and used.
Bug: webrtc:8415
Change-Id: I7878d9fb32c3f4ed11993a6f39e6d9c69fab190a
Reviewed-on: https://webrtc-review.googlesource.com/27980
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20973}
Header files base/videosinkinterface.h and base/videosourceinterface.h
were not part of any target (because they cause 2 dependency cycles).
This CL uncomment them so GN can keep dependencies under control, the
2 dependency cycles will be removed as part of webrtc:6828.
Bug: webrtc:6828
Change-Id: I5c5580facc010ba619e105a9b8a572ac70169a01
Reviewed-on: https://webrtc-review.googlesource.com/27621
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20970}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=sprang@webrtc.org
Bug: None
Change-Id: Ic429f28a8610ca798e29c45ec1f64604d6f9687f
Reviewed-on: https://webrtc-review.googlesource.com/23603
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20955}
This is a reland of 20f2133d5dbd1591b89425b24db3b1e09fbcf0b1
Original change's description:
> Add stereo codec header and pass it through RTP
>
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
>
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
>
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}
TBR=danilchap@webrtc.org, sprang@webrtc.org, niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: If8f0c7e6e3a2a704f19161f0e8bf1880906e7fe0
Reviewed-on: https://webrtc-review.googlesource.com/27160
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20946}
This reverts commit 94f39301067b9fbf820100cbd4018aad3a32cc52.
Reason for revert: Broke internal builds
Original change's description:
> Removes deprecated ADM APIs (reland).
>
> Usage should now be removed and this change can be relanded.
> It was reverted here: https://webrtc-review.googlesource.com/c/src/+/25320
>
> TBR=solenberg
>
> Bug: webrtc:7306
> Change-Id: I1afea773eff51bf5ec80711f0d7753ac0b7be77b
> Reviewed-on: https://webrtc-review.googlesource.com/27000
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20936}
TBR=henrika@webrtc.org
Change-Id: If91ff815fa69f7c36b0531e295f553a8c4a95590
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7306
Reviewed-on: https://webrtc-review.googlesource.com/27221
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20944}
This is a minor change to generated Python code used for testing the echo likelihood metric.
Bug: webrtc:8573
Change-Id: Ifb2438fdd36c3ade8cd830df0d05917af0f77dec
Reviewed-on: https://webrtc-review.googlesource.com/26281
Commit-Queue: Daniel Johansson <dajo@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20939}
This reverts commit 20f2133d5dbd1591b89425b24db3b1e09fbcf0b1.
Reason for revert: Breaks downstream project.
Original change's description:
> Add stereo codec header and pass it through RTP
>
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
>
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
>
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}
TBR=danilchap@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org,emircan@webrtc.org
Change-Id: I57f3172ca3c60a84537d577a574dc8018e12d634
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/26940
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20931}
The UMA metric will log the same information that goes into the
googPreferredJitterBufferMs stat.
Bug: webrtc:8488
Change-Id: I4e4e1e362dd42377105d52d2c4cd49c1ecb1a90d
Reviewed-on: https://webrtc-review.googlesource.com/26740
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20923}
- Defines CodecSpecificInfoStereo that carries stereo specific header info from
encoded image.
- Defines RTPVideoHeaderStereo that carries the above info to packetizer,
see module_common_types.h.
- Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
header.
- Uses new data containers in StereoAdapter classes.
This CL is the step 3 for adding alpha channel support over the wire in webrtc.
See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
Reviewed-on: https://webrtc-review.googlesource.com/22900
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20920}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=tommi@webrtc.org
Bug: None
Change-Id: I0ca1b624859a6561e227480b7dac8c254d26ad57
Reviewed-on: https://webrtc-review.googlesource.com/23562
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20916}
class to pass packet to RtcpPacketParser
This helpers make tests setup cleaner and
makes explicit expectation on number of packets passed to the transport.
Bug: webrtc:8239
Change-Id: I2d5975be59327cee440e87dbd0701b93514c9726
Reviewed-on: https://webrtc-review.googlesource.com/22460
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20911}
This tests NetEq with a stream encoded with Opus using it's internal
DTX/CNG.
Also adding a new resource file which is encoded using Opus with DTX.
Bug: webrtc:8488
Change-Id: Icfba5bc5dc7f9c9d0e637a90f4df674e8ba40358
Reviewed-on: https://webrtc-review.googlesource.com/26028
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20905}
The root cause of the flakiness is unknown, the possible issue is that the
console window running the test case is hidden or minimized. So this change
adds a SetWindow(SW_MAXIMIZE) to ensure the console window is showing.
I have run the tests against win_asan for hundreds times during the
thanksgiving. So far, no flakiness were caught.
Bug: webrtc:8568
Change-Id: Ib2c93e9bd511257213254bdaa0079c14ea50f3e4
Reviewed-on: https://webrtc-review.googlesource.com/25286
Reviewed-by: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#20902}
