Commit Graph

3633 Commits

Author SHA1 Message Date
6b3d18164b Remove unused BWE field trial strings.
Bug: None
Change-Id: I38d2e5495ddfe0b9f1493efc38ef7df95e7fd207
Reviewed-on: https://webrtc-review.googlesource.com/c/111258
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25798}
2018-11-27 13:05:43 +00:00
ba2840ce4e Various VP9 high fps fixes
- Enable flexible mode in loopback tools and quality tests
- Ensure duplicate references are not set by the sender in video header
- Reset first active spatial layer on keyframe in encoder
- Make vp9 encoder to not generate spatial references for first active
  layer with external reference control in svc flexible mode

Bug: webrtc:10049
Change-Id: If9ff576ea8a1a2fef6116b17b5b5adff08c5f8c6
Reviewed-on: https://webrtc-review.googlesource.com/c/112080
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25795}
2018-11-27 12:20:56 +00:00
af52b68116 Populate VideoSendTime extension network2 field when configured
before this CL it was only configured when pacer is used.
This CL sets it also when pacer is not used.

Move block for setting TransmissionOffset/AbsoluteTime extensions after pacer_ check
to stress in pacer case there are set(overwritten) in another function.

Bug: None
Change-Id: I06a6dd6ec689a25439a75b3baa71340535cd1ff8
Reviewed-on: https://webrtc-review.googlesource.com/c/112126
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25794}
2018-11-27 10:54:40 +00:00
74cdf7874d add cstring include need for strncmp
Propose resolution of Issue 10011 : (GCC) build fails desktop_capturer.cc:66:66: error: ‘strncmp’ was not declared in this scope

Bug: webrtc:10011
Change-Id: I4afdfd96f8bbc8e39380a365138ab79e237568e3
Reviewed-on: https://webrtc-review.googlesource.com/c/111885
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25790}
2018-11-26 20:49:36 +00:00
622eedaf0f Bump variable sizes in response to fuzzer bug
The fuzzers detected a possible overflow in the multiplication of sum and gainQ10.
Since gainQ10 cannot be larger than 2048000 (see WebRtcIsac_kQGain2Levels) and sum cannot be larger than 2^16, a int64 is large enough to hold the result.

Bug: chromium:904909
Change-Id: Icb12821d4006aaaaf70a5735d2abd2b96f7a2f0e
Reviewed-on: https://webrtc-review.googlesource.com/c/111921
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25787}
2018-11-26 16:16:50 +00:00
b24c00f02d Add AudioProcessingCaptureStats and a level estimator replacement
This adds an interface for accessing stats on the capture stream, and
adds a level estimator to report one of the stats.

Bug: webrtc:9947
Change-Id: Id472534fa2e04d46c9ab700671f620584a246afb
Reviewed-on: https://webrtc-review.googlesource.com/c/109587
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25786}
2018-11-26 15:52:14 +00:00
856cf22996 In ReceiveStatistics use monotonic clock instead of ntp clock
for all time difference calculations.

Bug: None
Change-Id: I37f4a3c73ab275e661bedf991a471a1c2928180a
Reviewed-on: https://webrtc-review.googlesource.com/c/111884
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25782}
2018-11-26 10:31:44 +00:00
8ce0d2b956 In ReceiveStatistic require callbacks during construction
Remove RegisterRtcpStatisticsCallback callback functions
saving taking an extra lock when calling callbacks.

Bug: None
Change-Id: Ib4537deffa0ab0abf597228e7c0fab7067614f6a
Reviewed-on: https://webrtc-review.googlesource.com/c/111821
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25779}
2018-11-26 09:17:21 +00:00
4c0cc5bc5f 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}
2018-11-26 07:48:03 +00:00
57f3ad0f8d Adds stable bandwidth estimate to GoogCC.
The intention is to provide a bandwidth estimate that only updates if
the actual available bandwidth is known to have changed. This will be
used in media streams to avoid changing the configuration (such as
frame size, audio frame length etc), just because the control target
rate changed.

Bug: webrtc:9718
Change-Id: I17ba5a2f9e5bd408a71f89c690d45541655a68e2
Reviewed-on: https://webrtc-review.googlesource.com/c/107726
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25772}
2018-11-23 14:55:37 +00:00
88ce4ef46e Don't buffer encoded frames.
Pass encoded frames to packetizer immediately if encoder is configured
to drop whole superframe.

Bug: webrtc:9950
Change-Id: Iedee9618bb146307efd5a86cb35bf14b5e64b341
Reviewed-on: https://webrtc-review.googlesource.com/c/109002
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25771}
2018-11-23 13:48:00 +00:00
885cf60106 Moves ProbeBitrateEstimator from DelayBasedBwe.
This prepares for providing an additional implementation of delay based
rate control. By moving the probe controller, less code will have to be
added in the upcoming CL.

Bug: webrtc:9718
Change-Id: I64eb2c8f5f7950b6e9d209f110dc0a757c710b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/111860
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25770}
2018-11-23 13:43:51 +00:00
e3abb8134f Decouple //rtc_base:rtc_base_tests_utils from gunit.
This CL decouples //rtc_base:rtc_base_tests_utils from gunit by
moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h
(which depends on gunit helpers) to their own build target.

It also removes some unused dependencies in the WebRTC build graph.

Bug: None
Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623
Reviewed-on: https://webrtc-review.googlesource.com/c/111861
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 12:52:46 +00:00
8af8896596 Expose jitter buffer flushes metric in new getStats api.
Origin trial experiment proposal (new statistic part):
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI/edit?ts=5bf5535c#

Bug: chromium:907113
Change-Id: I1d005291f9b47665f70c26148dbdcbb55564bef8
Reviewed-on: https://webrtc-review.googlesource.com/c/111505
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#25768}
2018-11-23 11:41:43 +00:00
b357e54dd5 Add field trial config to disable pacer emergency stops.
Bug: none
Change-Id: Ie92c4ef82e5ce3e222ec85df21acfb233b16b85d
Reviewed-on: https://webrtc-review.googlesource.com/c/111883
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25767}
2018-11-23 11:06:25 +00:00
6d254bcd5e Delete unused method NetEq::PacketBufferStatistics
Bug: None
Change-Id: I9f87e445e2b5b54f78474489172f694abff38363
Reviewed-on: https://webrtc-review.googlesource.com/c/111660
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25766}
2018-11-23 09:39:32 +00:00
f40150d874 Removing ANA enabling field trials.
This is to let ANA config proto to fully control it.

Bug: b/119788974
Change-Id: Ib7842f784bdf879cb7d753c7077ce845f435a379
Reviewed-on: https://webrtc-review.googlesource.com/c/111741
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25764}
2018-11-22 22:26:28 +00:00
777cf26328 AEC3: Clockdrift detection
This change introduces a clockdrift detector operating on the estimated
delay of the echo path delay estimator. Each time the delay estimate
changes it is compared to previous estimates. If the estimates are
slowly increasing or decreasing, clockdrift is detected.

Four different patterns are considered clockdrift:
- k, k+1, k+2, k+3
- k, k+2, k+1, k+3
- k, k-1, k-2, k-3
- k, k-2, k-1, k-3

A delay estimate history matching the three last elements in one of the
patterns is considered probable clockdrift. Matching all four elements
is considered verified clockdrift.

If the delay is constant for some time after clockdrift is detected the
clockdrift detector will revert to no detected clockdrift.

The level of clockdrift is reported via an UMA histogram.

Bug: webrtc:10014
Change-Id: I1cce4d593e101a8b3fa99df6935e59b4243cb97a
Reviewed-on: https://webrtc-review.googlesource.com/c/111381
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25758}
2018-11-22 16:02:44 +00:00
ebb50c217d Fix setting max reordering threshold in ReceiveStatistics
By ensuring new max reordering threshold applies to future statisticians too.

Bug: b/38179459
Change-Id: I0df32fb893a930b93faaf2161cd03626f9544a74
Reviewed-on: https://webrtc-review.googlesource.com/c/111752
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25756}
2018-11-22 15:31:20 +00:00
eccfc47ffa Cleanup AimdRateController and remove RateControlRegion enum.
- Rename avg_max_bitrate_kbps to link_capacity_estimate_kbps and change
  the type to optional.
- Remove the RateControlRegion enum. The old code seems to have the invariant
  that the region is kRcMaxUnknown iff avg_max_bitrate_kbps is uninitialized.
- Change floats to double.

Bug: webrtc:9942
Change-Id: Ic071a11ec4950053ec92beaa06f28f43192521d7
Reviewed-on: https://webrtc-review.googlesource.com/c/111247
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25752}
2018-11-22 13:51:28 +00:00
44727b48d6 Cleanup rtcp StreamStatistician::OnRtpPacket
inline InOrder check
remove it from IsRetransmit check as redundant
avoid call to IsRetransmitOfOldPacket when packet arrived in order
take current time once
Remove packet overhead counting as unused

Bug: None
Change-Id: Icd8bf69b5076e4469c349529c9ac79a1b15d9515
Reviewed-on: https://webrtc-review.googlesource.com/c/111746
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25749}
2018-11-22 11:42:13 +00:00
ecd62056e3 Disable GoogCcNetworkControllerTest.DetectsHighRateInSafeResetTrial
Test is flaky. See linked bug.

TBR=srte@webrtc.org

Bug: webrtc:10036
Change-Id: I21dd0daceaca6071364cb3aec50da79480f4dfcb
Reviewed-on: https://webrtc-review.googlesource.com/c/111747
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25746}
2018-11-22 10:51:11 +00:00
8ac05ccaa7 Adds trial to use link capacity estimate in Opus encoder.
Since the link capacity is designed to be a more stable value, we don't
need the smoothing. This allows us to react faster to changes in link
capacity while still avoiding to react to changes in target bitrate due
to normal control behavior.

Bug: webrtc:9718
Change-Id: I2fbf6bb882f312a7b28ea43d27057886d035ac45
Reviewed-on: https://webrtc-review.googlesource.com/c/111511
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25745}
2018-11-22 09:21:12 +00:00
d51b3553db Delete unused NetEq Rtcp stats.
Bug: webrtc:7135
Change-Id: Ib3ca9e02b051b8b41c2eac4e43a4f1f37999bf75
Reviewed-on: https://webrtc-review.googlesource.com/c/111640
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25743}
2018-11-22 08:00:54 +00:00
8b5d9d8650 Remove the audio/video split for the RTCP report intervals.
This is a follow up of a comment in
https://webrtc-review.googlesource.com/c/src/+/110105

It was not very useful to split the audio and video report interval since the RTCP module can only either be audio or video.

The recent it was written that way in https://webrtc-review.googlesource.com/c/src/+/43201/ was because that was a straightforward transition from two global constants to two variable.

Bug: webrtc:8789
Change-Id: I2293de14ba5f363351f379a02022ed5dc7b8d458
Reviewed-on: https://webrtc-review.googlesource.com/c/110824
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#25741}
2018-11-22 01:39:41 +00:00
4749e4e221 Move HdrMetadata to ColorSpace
Move HdrMetadata to ColorSpace as part of preparing for joint transmission
of these two objects.

Bug: webrtc:8651
Change-Id: Ie948011a2c0106d5967cb5ef3b9565217e798272
Reviewed-on: https://webrtc-review.googlesource.com/c/111481
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25730}
2018-11-21 15:09:24 +00:00
ecf6315a7f AGC2 adaptive digital: remove unnecessary flag.
Bug: webrtc:7494
Change-Id: I03d854ab082cb8fcf3f01a431c06496f93d3063b
Reviewed-on: https://webrtc-review.googlesource.com/c/111601
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25729}
2018-11-21 15:01:28 +00:00
8da7b350cf AGC2 adaptive digital false by default
Avoid that the client code relies on the adaptive digital mode being
enabled by default (error prone).

Bug: webrtc:7494
Change-Id: I765fecf535cf31a2163e10595a42520473c233b6
Reviewed-on: https://webrtc-review.googlesource.com/c/111586
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25728}
2018-11-21 14:20:15 +00:00
2b5b0e9521 Disabling ScreenDrawerTest.TwoScreenDrawerLocks
Test disabled on TSAN due to repeated failures. There are data races
in a low-level syncronization primitive (semaphore). Since
syncronization primitives should handle that, I think TSAN may be
configured incorrectly.

The locking scheme is written entirely in the unit test. This means we
are losing some test coverage of *unit tests*.

TBR=jamiewalch@chromium.org

Bug: webrtc:10019
Change-Id: Ieafa00a5a789acf8d0bacf6ad669c6daca7efa17
Reviewed-on: https://webrtc-review.googlesource.com/c/111585
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25723}
2018-11-21 10:27:52 +00:00
c4d5642e13 Revert "Default to dlopening the PipeWire."
This reverts commit a13be019017449c57f48203d0fb778f34f7553a7.

Reason for revert: The GN definitions cause problems for downstream tooling. They're also generally complicated and reach deep into Chromium's build which is undesirable. Setting `rtc_use_pipewire = true` by default should also be re-evaluated.

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}

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

Change-Id: Iec20b07cb1cff7d57f8114ac6ec2d0d250e61214
No-Try: true
Bug: chromium:682122
Reviewed-on: https://webrtc-review.googlesource.com/c/111584
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25722}
2018-11-21 09:20:41 +00:00
a13be01901 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}
2018-11-21 08:33:04 +00:00
bd04f4ae7f Increase buffer level threshold in VP8/9 tests.
This increases expected value of maximum buffer level in VP8/9 tests
up to 1 second and thus alignes it with the value that WebRTC uses by
default for these codecs.

Bug: webrtc:10017
Change-Id: I8fd41e8006f11c230d844a053c04656408c2ec97
Reviewed-on: https://webrtc-review.googlesource.com/c/111503
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25716}
2018-11-20 16:33:18 +00:00
2222a80e79 Delete unneeded includes of common_types.h and gn deps on webrtc_common.
Bug: webrtc:5876
Change-Id: Iae14e5f1679067a5a5e0584ca830aee0870c8807
Reviewed-on: https://webrtc-review.googlesource.com/c/111463
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25715}
2018-11-20 16:28:39 +00:00
4bc60452f7 Add output directory option for audioproc_f data dump files.
Bug: webrtc:10000
Change-Id: Iac21f826e78d6cb339c68fdeeedf9fe39920ac31
Reviewed-on: https://webrtc-review.googlesource.com/c/110904
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25713}
2018-11-20 13:30:24 +00:00
22b70ff1d4 Move VideoCodecType from common_types.h to api/video/video_codec_type.h
Bug: webrtc:7660
Change-Id: I9381364a64113dbb622b26acbf2b71228c3c4b96
Reviewed-on: https://webrtc-review.googlesource.com/c/111480
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25710}
2018-11-20 13:12:57 +00:00
22ff1a437a Fix threshold in VideoCodecTestLibvpx.ChangeFramerateVP9.
Libvpx has been recently updated and this test was failing because
of a slightly different value.

TBR=sprang@webrtc.org

Bug: webrtc:10017
Change-Id: I5fe9161eef5c3e1ff8e0dceb36a663648d8f4617
Reviewed-on: https://webrtc-review.googlesource.com/c/111461
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25709}
2018-11-20 13:09:33 +00:00
68170388f4 APM audioproc_f: flag for AGC2 adaptive level estimator.
Bug: webrtc:7494
Change-Id: I603211570a0a46d8884749dab887cd572827cca6
Reviewed-on: https://webrtc-review.googlesource.com/c/110250
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25708}
2018-11-20 12:50:23 +00:00
44974e143c AEC3: Adding a correction factor for the Erle estimation that depends on the portion of the filter that is currently in use.
In this CL a more precise estimation of the Erle is introduced. This is done by creating different estimators that are specialized in different regions of the linear filter. An estimation of which regions were used for generating the current echo estimate is performed and used for selecting the right Erle estimator.

Bug: webrtc:9961
Change-Id: Iba6eb24596c067c3c66d40df590be379d3e1bb7b
Reviewed-on: https://webrtc-review.googlesource.com/c/109400
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25707}
2018-11-20 12:28:05 +00:00
5f00995964 Using unit classes in AimdRateControl.
Bug: webrtc:9718
Change-Id: I1efed4e55c9d1ccec3c32ed012cb3cd82d7f4ee8
Reviewed-on: https://webrtc-review.googlesource.com/c/110788
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25705}
2018-11-20 08:04:11 +00:00
b6787bcd79 Using data unit classes in DelayBasedBwe.
Bug: webrtc:9718
Change-Id: I1b6ed37afd7680dfad6267addfe46155c378525d
Reviewed-on: https://webrtc-review.googlesource.com/c/110903
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25702}
2018-11-19 20:18:36 +00:00
3a83748422 New loss-based bandwidth control mechanism.
Bug: none
Change-Id: Ie60e9225e2a2260624342ffbadb08cb887b2b6f5
Reviewed-on: https://webrtc-review.googlesource.com/c/109923
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25696}
2018-11-19 15:09:04 +00:00
78e88fe602 Move NetworkStatistics and AudioDecodingCallStats from common_types.h
Bug: webrtc:7626
Change-Id: I1b933b8be7acbca1f1043a374a7cafb95aa9ffde
Reviewed-on: https://webrtc-review.googlesource.com/c/111249
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25688}
2018-11-19 11:55:34 +00:00
95adedb9c2 Always compile VP9 source files.
Instead of optionally compile VP9 source files based on the value of
the GN argument 'rtc_libvpx_build_vp9', this CL uses the preprocessor
macro RTC_ENABLE_VP9 to decide if VP9 related code needs to be compiled
or not.

Bug: None
Change-Id: I5c1b69d7ec35e8446181d98c912277d0ae8fdba2
Reviewed-on: https://webrtc-review.googlesource.com/c/111063
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25685}
2018-11-19 10:18:18 +00:00
601504c5cc in RtcpTransceiver remove workaround for old bug in RtcpReceiver
the bug in RtcpReceiver was fixed Jan 30, i.e. 10.5 month ago

Bug: webrtc:8805
Change-Id: I5f5f00fba5e984ede906c5dbbe841ee5f4992e09
Reviewed-on: https://webrtc-review.googlesource.com/c/99822
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25683}
2018-11-19 10:13:02 +00:00
8ef57932b1 Switch from RTC_DISABLE_VP9 to RTC_ENABLE_VP9.
RTC_ENABLE_VP9 is more natural to deal with then RTC_DISABLE_VP9.
In all the places this macro is used, WebRTC needs to do more things
so it is easier to "do more if RTC_ENABLE_VP9 is defined" than
"do more if RTC_DISABLE_VP9 is not defined".

Bug: None
Change-Id: If992e5c554173e6af3f030f6e0fd21bd82acf9eb
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/111242
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25679}
2018-11-19 08:30:55 +00:00
eee39206a2 Don't poll EncoderInfo from encoder twice per frame
Bug: webrtc:9890
Change-Id: Id4c2062a1c0c6be699f2096b4c0b334c98f3c4ba
Reviewed-on: https://webrtc-review.googlesource.com/c/111083
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25672}
2018-11-16 14:11:38 +00:00
a32d7e2a2f Add default values for PlayoutDelay in RTPVideoHeader.
There have been several bugs where the members of PlayoutDelay were
zero initialized when handling RTP packets without the corresponding
extensions. Initializing to {-1, -1} (meaning not provided) is less
brittle.

Bug: None
Change-Id: I196850377128d5e67a19bdaf9298403b2e9f5a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/111181
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25670}
2018-11-16 12:10:23 +00:00
7dbb7c311f Adding missing build target for audio_device_default.
The header modules/audio_device/include/audio_device_default.h was not
owned by any build target.

Bug: webrtc:8946
Change-Id: I3266a613c10963688c3bea701384e1d1bb68daac
Reviewed-on: https://webrtc-review.googlesource.com/c/111201
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25669}
2018-11-16 11:15:08 +00:00
dd886082c5 AGC2 flags: remove deprecated fields.
Downstream projects adapted, clean up.

Bug: webrtc:7494
Change-Id: I019b8dd79c6bc55c900fb5595d5e2ee8aa0a2400
Reviewed-on: https://webrtc-review.googlesource.com/c/110865
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25656}
2018-11-15 13:47:24 +00:00
dbb988b016 Change ReceiveStatistics to implement RtpPacketSinkInterface, part 2.
Delete the deprecated IncomingPacket method, and convert implementation
to use RtpPacketReceived rather than RTPHeader.

Part 1 was https://webrtc-review.googlesource.com/c/src/+/100104

Bug: webrtc:7135, webrtc:8016
Change-Id: Ib4840d947870403deea2f9067f847e4b0f182479
Reviewed-on: https://webrtc-review.googlesource.com/c/6762
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25648}
2018-11-15 07:38:26 +00:00