Commit Graph

3633 Commits

Author SHA1 Message Date
c0a67baa36 AEC3: moving the dumping of the Erle to aec state
Bug: webrtc:10154
Change-Id: I3b4cbfe218f6ed1be273f4545b159dc4d90ba587
Reviewed-on: https://webrtc-review.googlesource.com/c/115402
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26082}
2018-12-21 10:29:42 +00:00
a1f78a4fa6 Revert "Refactor rate profile update."
This reverts commit b6cdfdc165d76d86a67d829e0ccec50c36106e73.

Reason for revert: breaks downstream projects

Original change's description:
> Refactor rate profile update.
> 
> RateProfile::frame_num specifies frame at which this rate profile
> should be applied.
> 
> Bug: none
> Change-Id: I003ee43f44299a49d83f547558284817bfaeacc0
> Reviewed-on: https://webrtc-review.googlesource.com/c/115242
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26080}

TBR=ilnik@webrtc.org,shampson@webrtc.org,ssilkin@webrtc.org

Change-Id: I5957a0169841008436d1db70403d3694bf25d5cf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/115400
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26081}
2018-12-21 09:05:01 +00:00
b6cdfdc165 Refactor rate profile update.
RateProfile::frame_num specifies frame at which this rate profile
should be applied.

Bug: none
Change-Id: I003ee43f44299a49d83f547558284817bfaeacc0
Reviewed-on: https://webrtc-review.googlesource.com/c/115242
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26080}
2018-12-21 08:32:35 +00:00
7f57788ab7 Removes trial to enable BBR congestion controller.
The BBR controller can still be injected, but the trials
will no longer work. This reduces the binary size.

Bug: webrtc:8415
Change-Id: I2c32c414d08ef0cc16bfd72651535a755cde9916
Reviewed-on: https://webrtc-review.googlesource.com/c/114120
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26077}
2018-12-20 16:42:07 +00:00
e2e000059d Make pacing buffer send interval configurable.
Bug: webrtc:10153
Change-Id: I2f71b5fc902e24b87f0b6cba474d988b8c27aefa
Reviewed-on: https://webrtc-review.googlesource.com/c/115301
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26076}
2018-12-20 15:29:27 +00:00
61832dd018 Propagate spatial index to EncodedImage.
Set spatial index of assembled VP9 picture equal to spatial index of
its top spatial layer frame.

Bug: webrtc:10151
Change-Id: Iae40505864b14b01cc6787f8da99a9e3fe283956
Reviewed-on: https://webrtc-review.googlesource.com/c/115280
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26075}
2018-12-20 15:21:22 +00:00
29a935a7fe Refactor NetEqDecoderPlc to use AudioDecoderProxyFactory
Bug: webrtc:10080
Change-Id: I651efc70fa020e345776c44d9510245c45f9b092
Reviewed-on: https://webrtc-review.googlesource.com/c/114547
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26067}
2018-12-20 10:17:15 +00:00
3f651d80a0 Reland "Add AudioDecoderFactory to NetEqTest constructor."
This is a reland of daa970f33e1923c5651a4a63c18e3d5361d0a795

Original change's description:
> Add AudioDecoderFactory to NetEqTest constructor.
>
> Update EventLogAnalyzer to not depend on builtin audio decoders.
>
> Bug: webrtc:8396, webrtc:10080
> Change-Id: Ie02ed9cda6d4f11bfdf2e65eb6482283b7520738
> Reviewed-on: https://webrtc-review.googlesource.com/c/114301
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26026}

Tbr: kwiberg@webrtc.org
Bug: webrtc:8396, webrtc:10080
Change-Id: I598ce1cd41676b1992b0973b09476eeeb0e602d2
Reviewed-on: https://webrtc-review.googlesource.com/c/114940
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26058}
2018-12-19 15:08:47 +00:00
31d8b52075 Delete unneeded includes of rtc_base/stringutils.h.
Also delete corresponding dependencies on rtc_base:stringutils.

Bug: webrtc:6424
Change-Id: I2be5e021292eea2d788c76a63cc0e4f7cefd927d
Reviewed-on: https://webrtc-review.googlesource.com/c/114544
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26057}
2018-12-19 11:04:27 +00:00
d1208c26b1 Desktop capturer: Add OnDisplayChanged callback
This adds support for a new DesktopCapturer::Callback method
OnDisplayChanged that is sent at the start of a desktop capture
session and whenever the display geometry changes.

This cl adds the basic structure to call this api at the start
of the capture session. Currently Windows only.

A follow-up cl will add support to call this whenever the display
geometry changes.

Bug: webrtc:10122, chromium:915411
Change-Id: Ie7283be5992454180daab1a60f58a3b2efdfed56
Reviewed-on: https://webrtc-review.googlesource.com/c/114020
Commit-Queue: Gary Kacmarcik <garykac@chromium.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26053}
2018-12-18 21:56:48 +00:00
503da94350 Adds first unacknowledged packet send time.
This will be used to calculate a lower bound for the round trip time in
a later CL.

Bug: webrtc:9718
Change-Id: I0a1d22045961fe6bd343d1d6ce9b36490b036bb1
Reviewed-on: https://webrtc-review.googlesource.com/c/114680
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26050}
2018-12-18 16:55:33 +00:00
11b8703201 Base ApmTest.Process on AudioProcessingStats.output_rms_dbfs
This replaces the current usage of AudioProcessing::level_estimator()
in that test.

The unit tests that specifically test the level_estimator API are left
in place, until the level_estimator API itself is removed.

Bug: webrtc:9947
Change-Id: I73301c1478d2c9763bb49598a692142229102876
Reviewed-on: https://webrtc-review.googlesource.com/c/114550
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26049}
2018-12-18 16:45:03 +00:00
358fba1f9d Removes NetworkControllerTester
Replacing NetworkControllerTester usages with SimulatedTimeClient since
they have corresponding functionality.

Bug: webrtc:9510
Change-Id: I4a6a78142a9922e53b862eb8cb71ba9091236346
Reviewed-on: https://webrtc-review.googlesource.com/c/114660
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26048}
2018-12-18 16:11:22 +00:00
b3564c1cb2 Back off relative to current estimate rather than ack rate when in ALR.
If we're in ALR, the acked rate is going to be significantly lower than
the current estimate for the link capacity. If we need to back off in
this situation (usually caused by latency spikes), this CL makes us back
off relative to current estimate if. We then immediately send a new
probe just in case the network did actually change.

All of this is behind experiment flags for now.

Bug: webrtc:10144
Change-Id: I062a259c36417eea2211d44592ef7fc979aa22b7
Reviewed-on: https://webrtc-review.googlesource.com/c/113880
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26045}
2018-12-18 12:33:08 +00:00
a1bec23f6c Pass on explicit color space for VP8 and H264
Bug: webtc:8651
Change-Id: I9d478e7123e915bff858d725d6008fcfeeb0779d
Reviewed-on: https://webrtc-review.googlesource.com/c/114424
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26044}
2018-12-18 12:32:03 +00:00
0fcf4b1dbd Delete unused I420 "codec"
Previous attempt: https://codereview.webrtc.org/1882733006/. There
might be some benefit of having dummy encoder/decoder available in
video_loopback.

Bug: webrtc:5791
Change-Id: Iec316296754178c92b18dd3cf92f67ce6aed9439
Reviewed-on: https://webrtc-review.googlesource.com/c/112596
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26043}
2018-12-18 12:30:58 +00:00
7d92de69fe Deprecating legacy SendSideCongestionController.
For somewhat similar funtionality, GoogCcNetworkController can
be used via GoogCcNetworkControllerFactory.

Bug: webrtc:9586
Change-Id: I298050184513f50c1b9ef5c21b8c9b7a6ca46fd5
Reviewed-on: https://webrtc-review.googlesource.com/c/114543
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26040}
2018-12-18 10:22:30 +00:00
41f3a43c74 Remove CodecInst pt.3
Finally remove CodecInst from common_types.h, including remaining code referencing it.

TBR=kwiberg

Bug: webrtc:7626
Change-Id: I5e6b949ae9093641e33972af8438d1126fc48556
Reviewed-on: https://webrtc-review.googlesource.com/c/114546
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26036}
2018-12-18 07:42:21 +00:00
f7f753b320 Revert "Add AudioDecoderFactory to NetEqTest constructor."
This reverts commit daa970f33e1923c5651a4a63c18e3d5361d0a795.

Reason for revert: Speculative revert due to downstream breakage

Original change's description:
> Add AudioDecoderFactory to NetEqTest constructor.
>
> Update EventLogAnalyzer to not depend on builtin audio decoders.
>
> Bug: webrtc:8396, webrtc:10080
> Change-Id: Ie02ed9cda6d4f11bfdf2e65eb6482283b7520738
> Reviewed-on: https://webrtc-review.googlesource.com/c/114301
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26026}

TBR=mbonadei@webrtc.org,aleloi@webrtc.org,kwiberg@webrtc.org,terelius@webrtc.org,nisse@webrtc.org,ivoc@webrtc.org

No-Try: True
Bug: webrtc:8396, webrtc:10080
Change-Id: Ided750d8ed800d8a38f7cce8f72095d8ed1bc6cb
Reviewed-on: https://webrtc-review.googlesource.com/c/114552
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26030}
2018-12-17 15:16:30 +00:00
daa970f33e Add AudioDecoderFactory to NetEqTest constructor.
Update EventLogAnalyzer to not depend on builtin audio decoders.

Bug: webrtc:8396, webrtc:10080
Change-Id: Ie02ed9cda6d4f11bfdf2e65eb6482283b7520738
Reviewed-on: https://webrtc-review.googlesource.com/c/114301
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26026}
2018-12-17 11:15:50 +00:00
f693bfae5f Remove CodecInst pt.2
The following APIs on AudioCodingModule are deprecated with this CL:
  static int NumberOfCodecs();
  static int Codec(int, CodecInst*);
  static int Codec(const char*, CodecInst*, int, size_t);
  static int Codec(const char*, int, size_t);
  absl::optional<CodecInst> SendCodec() const;
  bool RegisterReceiveCodec(int, const SdpAudioFormat&);
  int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&);
  int UnregisterReceiveCodec(uint8_t);
  int32_t ReceiveCodec(CodecInst*);
  absl::optional<SdpAudioFormat> ReceiveFormat();

As well as this method on RtpRtcp module:
  int32_t RegisterSendPayload(const CodecInst&);

Bug: webrtc:7626
Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/113816
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26025}
2018-12-17 10:33:55 +00:00
194d4d20fb Delete unused send-side methods of VideoCodingModule
Bug: webrtc:8064
Change-Id: Icb7a452dfefce01ff59f6568b4766d609c2725bf
Reviewed-on: https://webrtc-review.googlesource.com/c/14900
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26023}
2018-12-17 08:26:12 +00:00
2db46b0fb7 Added new feature to print a text log to neteq_rtpplay
This will print out the major events during a NetEq simulation.

Bug: b/116685514
Change-Id: Iab172e9a9115695b42c67628d5523c727359bb89
Reviewed-on: https://webrtc-review.googlesource.com/c/114320
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26019}
2018-12-14 16:38:45 +00:00
94f107454e Only use GetAudio events that correspond to an ssrc matching at least one incoming packet.
Using GetAudio events from SSRCs without incoming packets doesn't make sense, and should be prevented.

Bug: b/116685514
Change-Id: I48e38bb780549c71cb5f68d370a6819634ad487d
Reviewed-on: https://webrtc-review.googlesource.com/c/114321
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26017}
2018-12-14 15:05:15 +00:00
24779d8229 Missing packet send time should not cause BWE backoff.
The removed coded causes problems if the same RTCP packet is forwarded
to the congestion controller multiple times.

Bug: webrtc:10125
Change-Id: I659d8f8f3ce3c643710156fa81176ceeaedd714a
Reviewed-on: https://webrtc-review.googlesource.com/c/114165
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26016}
2018-12-14 14:48:48 +00:00
3073f3d148 Revert "Reland "Default to dlopening the PipeWire.""
This reverts commit 0cc42d47389c039c57e47d7ec0c76b97e2da2b0b.

Reason for revert: Sorry, broke WebRTC roll to Chromium again: https://chromium-review.googlesource.com/c/chromium/src/+/1377299. This time the define now set enabled code around the feature flag already landed and there were failures related to that. I suggest to revert that Chromium CL and re-land it after this CL has landed and been rolled into Chromium (if possible to do so).

Original change's description:
> Reland "Default to dlopening the PipeWire."
> 
> This is a reland of a099877d8946eb942046ca1295cc142e4fa7ea6f
> 
> Original change's description:
> > Reland "Default to dlopening the PipeWire."
> >
> > This is a reland of a13be019017449c57f48203d0fb778f34f7553a7
> >
> > Original change's description:
> > > Default to dlopening the PipeWire.
> > >
> > > Reuse the existing infra from Chromium to do that. Additionally the
> > > target_gen_dir needs to the added to the include directories, otherwise
> > > the Chromium build will fail as it won't find the generated stubs. Also the
> > > pw_properties_new() was replaced with pw_properties_new_string() as it doesn't
> > > require a variadic parameter because the //tools/generate_stubs/generate_stubs.py
> > > doesn't work with them correctly. With all these changes in place the PipeWire
> > > support is enabled when compiling on Linux.
> > >
> > > Bug: chromium:682122
> > > Change-Id: I3bbc5efaecd9a08e20cbcf998b2cb534224eae7d
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/111081
> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > > Reviewed-by: Brave Yao <braveyao@webrtc.org>
> > > Commit-Queue: Tomáš Popela <tomas.popela@gmail.com>
> > > Cr-Commit-Position: refs/heads/master@{#25720}
> >
> > Bug: chromium:682122
> > Change-Id: I3cca3d4d961dc7a088346c8fd3c970d3dfde3b79
> > Reviewed-on: https://webrtc-review.googlesource.com/c/113040
> > Reviewed-by: Weiyong Yao <braveyao@chromium.org>
> > Reviewed-by: Brave Yao <braveyao@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25981}
> 
> Bug: chromium:682122
> Change-Id: Ief26c93069f946f981340664a267fcb412229285
> Reviewed-on: https://webrtc-review.googlesource.com/c/114163
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26004}

TBR=phoglund@webrtc.org,mbonadei@webrtc.org,oprypin@webrtc.org,braveyao@webrtc.org,braveyao@chromium.org,tomas.popela@gmail.com

Change-Id: I9ca52c61210e94182dd6b6a26a136c7f79a2dd0f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:682122
Reviewed-on: https://webrtc-review.googlesource.com/c/114427
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Henrik Grunell <henrikg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26014}
2018-12-14 14:23:58 +00:00
3be607f2bc Use output_dir instead of output_name
This is to make second build no-op in mac_asan builder.
e.g. https://ci.chromium.org/p/webrtc/builders/luci.webrtc.try/mac_asan/15219

We can use output_dir to override default_output_dir of executable.
https://gn.googlesource.com/gn/+/master/docs/reference.md#tool-variables


confirm no-op step for this CL does not complain.
https://ci.chromium.org/p/webrtc/builders/luci.webrtc.try/mac_asan/15305

Bug: chromium:914264
Change-Id: Ia1196280064703dcb08e208e91c704cce25a925c
Reviewed-on: https://webrtc-review.googlesource.com/c/114180
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Takuto Ikuta <tikuta@google.com>
Cr-Commit-Position: refs/heads/master@{#26013}
2018-12-14 14:22:52 +00:00
f1ab9b9b3b Refactor creation of ColorSpace test data
Bug: webrtc:8651
Change-Id: I2ebb5fcdc260af19d04513ab5f3d76f81a3b4ca9
Reviewed-on: https://webrtc-review.googlesource.com/c/114282
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26012}
2018-12-14 10:15:10 +00:00
0cc42d4738 Reland "Default to dlopening the PipeWire."
This is a reland of a099877d8946eb942046ca1295cc142e4fa7ea6f

Original change's description:
> Reland "Default to dlopening the PipeWire."
>
> This is a reland of a13be019017449c57f48203d0fb778f34f7553a7
>
> Original change's description:
> > Default to dlopening the PipeWire.
> >
> > Reuse the existing infra from Chromium to do that. Additionally the
> > target_gen_dir needs to the added to the include directories, otherwise
> > the Chromium build will fail as it won't find the generated stubs. Also the
> > pw_properties_new() was replaced with pw_properties_new_string() as it doesn't
> > require a variadic parameter because the //tools/generate_stubs/generate_stubs.py
> > doesn't work with them correctly. With all these changes in place the PipeWire
> > support is enabled when compiling on Linux.
> >
> > Bug: chromium:682122
> > Change-Id: I3bbc5efaecd9a08e20cbcf998b2cb534224eae7d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/111081
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Brave Yao <braveyao@webrtc.org>
> > Commit-Queue: Tomáš Popela <tomas.popela@gmail.com>
> > Cr-Commit-Position: refs/heads/master@{#25720}
>
> Bug: chromium:682122
> Change-Id: I3cca3d4d961dc7a088346c8fd3c970d3dfde3b79
> Reviewed-on: https://webrtc-review.googlesource.com/c/113040
> Reviewed-by: Weiyong Yao <braveyao@chromium.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25981}

Bug: chromium:682122
Change-Id: Ief26c93069f946f981340664a267fcb412229285
Reviewed-on: https://webrtc-review.googlesource.com/c/114163
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26004}
2018-12-13 19:45:59 +00:00
d96b275cd6 Refactor EncodeParameters usage, remove unused rtt/loss
Bug: webrtc:10126
Change-Id: Ib93f5e65b25540576c026197f72a5902cf43fc16
Reviewed-on: https://webrtc-review.googlesource.com/c/114281
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26001}
2018-12-13 12:15:09 +00:00
aa7bc7e0bb Create field trial for vp8 number of thread on iOS.
Without the added preprocessor check, iOS device will be using the desktop logic to determine the number of thread. This put iPhone 8 and iPhone X to use 3 threads and all other iPhones after iPhone 5 to use a single thread.
This CL added a preprocessor for WEBRTC_IOS to have it own thread number calculation logic. In which, the maximum number of thread is fetched from a field_trial and capped by the number of CPU available on the device.

Bug: webrtc:10005
Change-Id: I8c6257fcbf85b07bc986b5f733dbabb3feee37f7
Reviewed-on: https://webrtc-review.googlesource.com/c/110941
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25997}
2018-12-13 07:35:59 +00:00
1618095100 Cleanup of RtpTransportControllerSend.
This CL simplifies a lot of code that can be cleaned up after the merge
of RtpTransportControllerSend and SendSideCongestionController.

In particular, the role of CongestionControlHandler is reduced to only
handle the pacer pushback and stream pausing mechanism.

Bug: webrtc:9586
Change-Id: Idbc1e968efd35e6df6129bc307f6bc1db18d20f2
Reviewed-on: https://webrtc-review.googlesource.com/c/113947
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25994}
2018-12-12 16:36:45 +00:00
c13f4be5f4 Add chroma siting to color space RTP extension
- Add chroma siting to color space RTP extension.
- Use 16 bits for max/min luminance.
- Change denominator of chromaticity and luminance.
- Add RTC_DCHECKs to protect against overflows.

Bug: webrtc:8651
Change-Id: If8b95bad6241381224eaba9c5bccce06a65a9195
Reviewed-on: https://webrtc-review.googlesource.com/c/113804
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25990}
2018-12-12 13:13:15 +00:00
0697ce2a76 Revert "Reland "Default to dlopening the PipeWire.""
This reverts commit a099877d8946eb942046ca1295cc142e4fa7ea6f.

Reason for revert: Breaks WebRTC roll into Chromium. See https://chromium-review.googlesource.com/c/chromium/src/+/1373891:

In file included from ../../third_party/webrtc/modules/desktop_capture/linux/window_capturer_pipewire.cc:11:
In file included from ../../third_party/webrtc/modules/desktop_capture/linux/window_capturer_pipewire.h:16:
../../third_party/webrtc/modules/desktop_capture/linux/base_capturer_pipewire.h:16:10: fatal error: 'pipewire/pipewire.h' file not found
#include <pipewire/pipewire.h>
         ^~~~~~~~~~~~~~~~~~~~~

Original change's description:
> Reland "Default to dlopening the PipeWire."
> 
> This is a reland of a13be019017449c57f48203d0fb778f34f7553a7
> 
> Original change's description:
> > Default to dlopening the PipeWire.
> >
> > Reuse the existing infra from Chromium to do that. Additionally the
> > target_gen_dir needs to the added to the include directories, otherwise
> > the Chromium build will fail as it won't find the generated stubs. Also the
> > pw_properties_new() was replaced with pw_properties_new_string() as it doesn't
> > require a variadic parameter because the //tools/generate_stubs/generate_stubs.py
> > doesn't work with them correctly. With all these changes in place the PipeWire
> > support is enabled when compiling on Linux.
> >
> > Bug: chromium:682122
> > Change-Id: I3bbc5efaecd9a08e20cbcf998b2cb534224eae7d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/111081
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Brave Yao <braveyao@webrtc.org>
> > Commit-Queue: Tomáš Popela <tomas.popela@gmail.com>
> > Cr-Commit-Position: refs/heads/master@{#25720}
> 
> Bug: chromium:682122
> Change-Id: I3cca3d4d961dc7a088346c8fd3c970d3dfde3b79
> Reviewed-on: https://webrtc-review.googlesource.com/c/113040
> Reviewed-by: Weiyong Yao <braveyao@chromium.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25981}

TBR=phoglund@webrtc.org,mbonadei@webrtc.org,oprypin@webrtc.org,braveyao@webrtc.org,braveyao@chromium.org,tomas.popela@gmail.com

Change-Id: Icdb6a94c8825f13d75ddc12219e99cee8fef51a0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:682122
Reviewed-on: https://webrtc-review.googlesource.com/c/114162
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Henrik Grunell <henrikg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25989}
2018-12-12 13:05:56 +00:00
1d8307d706 Delete VideoCodec::targetBitrate
This member is unused by encoders.

Bug: None
Change-Id: I867013bfdb89f48782e84842de05bb57648e0b64
Reviewed-on: https://webrtc-review.googlesource.com/c/113882
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25988}
2018-12-12 12:48:15 +00:00
e10b163dd4 Stop using 'using namespace'.
This CL removes all the instances of 'using namespace' from C++ code
(more info https://abseil.io/tips/153).

Bug: webrtc:9855
Change-Id: Ic940fe87c5047742cfa6d60857d2f97be380ed18
Reviewed-on: https://webrtc-review.googlesource.com/c/113948
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25985}
2018-12-12 11:08:40 +00:00
50b66d55f8 Convert NetEq Cng-related test to not use RegisterExternalDecoder
Bug: webrtc:10080
Change-Id: Ie91e967cd68efede71108458b912bf1e062ffea6
Reviewed-on: https://webrtc-review.googlesource.com/c/113943
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25982}
2018-12-12 09:19:22 +00:00
a099877d89 Reland "Default to dlopening the PipeWire."
This is a reland of a13be019017449c57f48203d0fb778f34f7553a7

Original change's description:
> Default to dlopening the PipeWire.
>
> Reuse the existing infra from Chromium to do that. Additionally the
> target_gen_dir needs to the added to the include directories, otherwise
> the Chromium build will fail as it won't find the generated stubs. Also the
> pw_properties_new() was replaced with pw_properties_new_string() as it doesn't
> require a variadic parameter because the //tools/generate_stubs/generate_stubs.py
> doesn't work with them correctly. With all these changes in place the PipeWire
> support is enabled when compiling on Linux.
>
> Bug: chromium:682122
> Change-Id: I3bbc5efaecd9a08e20cbcf998b2cb534224eae7d
> Reviewed-on: https://webrtc-review.googlesource.com/c/111081
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Commit-Queue: Tomáš Popela <tomas.popela@gmail.com>
> Cr-Commit-Position: refs/heads/master@{#25720}

Bug: chromium:682122
Change-Id: I3cca3d4d961dc7a088346c8fd3c970d3dfde3b79
Reviewed-on: https://webrtc-review.googlesource.com/c/113040
Reviewed-by: Weiyong Yao <braveyao@chromium.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25981}
2018-12-12 08:22:57 +00:00
7b3a568f6a Reland 2: Add VP9 Profile 2 to default profiles
This is a reland of 4c0cc5bc5fa027b9392ff2886e731bea3aac7602
I added more Chrome checks for munging profiles in the below patch
that will allow us to land this without regressions.
https://chromium-review.googlesource.com/c/chromium/src/+/1366898

Original change's description:
> Reland Profile 2 to default profiles
>
> This is a reland after chrome browser tests are updated.
>
> Bug: webrtc:9376
> Change-Id: I818bf5d447da7901ffe49f2c452decb89196e829
> TBR: niklas.enbom@webrtc.org
> Reviewed-on: https://webrtc-review.googlesource.com/c/112060
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25778}

Bug: webrtc:9376
Change-Id: I8998537816a773961e519535c6afdde3801b5918
TBR: niklas.enbom@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/113980
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25977}
2018-12-11 23:38:26 +00:00
5d4740170a Reduce pacing buffer padding rate during pushback.
Bug: webrtc:10112
Change-Id: I2cd2d07bd5bcbff5b3808ee63eea251a52e45b79
Reviewed-on: https://webrtc-review.googlesource.com/c/113808
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25968}
2018-12-11 15:22:27 +00:00
698d6c4f30 Change the type of indW32 back to int32_t
It was changed to size_t in https://codereview.webrtc.org/1227163003,
which makes sense if the pitch lags in the code are also guaranteed
to be non-negative. Otherwise, integer wraparounds may happen, which
causes the code to circumvent the check for too low values here:
https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c?q=webrtcisacfix_pitchfilter&sq=package:chromium&g=0&l=112



Bug: chromium:906379
Change-Id: Id88c6c38bf30059181ed593968cea29ca87adf76
Reviewed-on: https://webrtc-review.googlesource.com/c/113810
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25964}
2018-12-11 13:10:12 +00:00
aa4f100225 Adds trial to fall back to probe rate if ack rate is missing.
Bug: webrtc:9718
Change-Id: I7b6e1d3c051e67b97f6de1ec95e84631af9c5b0d
Reviewed-on: https://webrtc-review.googlesource.com/c/113600
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25953}
2018-12-10 16:12:18 +00:00
f3ef6cd863 Using more accurate receive time calculation in scenario tests.
Some tests had to be updated due to this change.

Bug: webrtc:9510
Change-Id: I79c4c0166d8ba5e8190a607d5d35b67dc30a3c14
Reviewed-on: https://webrtc-review.googlesource.com/c/113522
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25952}
2018-12-10 15:54:33 +00:00
69540f4419 Use android Nullable instead of javax Nullable
This is a propagation of upstream chromium change needed to
resume DEPS autorolls into WebRTC.

Original comment from upstream change:

> This change is made in preparation for an ErrorProne
> check to catch this at compile time. See bug for details.

Bug: chromium:771683
Change-Id: I56aed15f73a633dcadae7ece6c645cd3596f9257
Reviewed-on: https://webrtc-review.googlesource.com/c/113505
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25951}
2018-12-10 15:03:58 +00:00
57011626bd Re-tuning of VAD in AGC2.
Changing VAD (voice activity detector) confidence threshold from 40%
to 90%. The proportion of samples classified as speech drops to ca 80%
of what it was when the threshold was 40%. Therefore,
kFullBufferSizeMs has to be increased by 1.0/0.8. We increase it from
1600ms to 2000ms.

TESTED = Did run the new and old configs on AEC dumps. With one minute
of kitchen noise, the new tuning boosted the noise by 3-4 db less.

Bug: chromium:913430
Change-Id: I4a2ebb6d1d309c6c20dd23c3685818b1b5ad4a66
Reviewed-on: https://webrtc-review.googlesource.com/c/113806
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25950}
2018-12-10 14:47:29 +00:00
f04feee41e Remove redundant return-statement in VCMGenericEncoder::RequestFrame
Bug: None
Change-Id: I0da8747729ec309a37146397d6bc1f32bf22c329
Reviewed-on: https://webrtc-review.googlesource.com/c/113660
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25947}
2018-12-10 13:54:39 +00:00
a1eb9c7e9b Convert NetEq tests to not use RegisterExternalDecoder.
This change converts all tests but CodecInternalCng and
DecodingErrorDuringInternalCng, which depend on the obsolete Decode
method.

Bug: webrtc:10080
Change-Id: I34b068b3aa7139ed24bd63b417a5adcfc1de7922
Reviewed-on: https://webrtc-review.googlesource.com/c/113506
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25946}
2018-12-10 13:01:21 +00:00
8b9b5f98db Activate/deactivate VP9 spatial layers.
* Stop encoding spatial layers S(n >= N) if application deactivates
spatial layer N by setting RTCRtpEncodingParameters.active = false.

* Move calculation of padding bitrate to SvcRateAllocator class.

* Pad up to minimum required bitrate of base layer if ALR probing is
enabled.

Bug: webrtc:9350
Change-Id: I398284c943d43348def535c83263fc234c9767fa
Reviewed-on: https://webrtc-review.googlesource.com/c/113240
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25945}
2018-12-10 12:55:51 +00:00
b47ccc38e7 Add chroma siting to ColorSpace
Bug: webrtc:8651
Change-Id: I82263e8b6cdcc3ebf699f5e3ebbde04e46982efb
Reviewed-on: https://webrtc-review.googlesource.com/c/113424
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25944}
2018-12-10 11:19:35 +00:00
a59db7481c Remove unnecessary includes of common_types.h
Bug: webrtc:7626
Change-Id: I2d9275e5dc8eea6419d3c80cd68c4a01deafa9b7
Reviewed-on: https://webrtc-review.googlesource.com/c/113524
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25940}
2018-12-07 21:21:13 +00:00