130 Commits

Author SHA1 Message Date
Ivo Creusen
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
Benjamin Wright
3e94557b04 Adding fuzzing configuration files for Rtp Replay Fuzzing.
Configuring video decoding and rtp depacketization through json was introduced
in a prior change. This change introduces some basic configurations that will
be used in the initial round of fuzzers that are being added.

TBR=henrik.lundin@webrtc.org

Bug: webrtc:9599
Change-Id: I58aba6a6f24f8374126547deeef0ff4d1708327b
Reviewed-on: https://webrtc-review.googlesource.com/c/113834
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26005}
2018-12-13 19:58:39 +00:00
Benjamin Wright
f6b10fbe4a Basic fuzzing of rtc::s_url_decode.
rtc::s_url_decode internally calls transform on rtc::url_decode which operates
on raw char buffers. This is used in some core parts of ice server parsing so
it makes sense to add at least a basic fuzzer here. Corpus generation will be
tailored in a future CL.

Bug: webrtc:10117
Change-Id: If1685601c746c4a9f88c2a8d396eeb3f1b1688d4
Reviewed-on: https://webrtc-review.googlesource.com/c/113835
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25980}
2018-12-12 01:21:25 +00:00
Ilya Nikolaevskiy
4348ce240a Calculate min and max receive timestamps for packets in a video frame
Bug: webrtc:10106
Change-Id: I1d3469abb1e7bb7c91a5912d7b781505526abaca
Reviewed-on: https://webrtc-review.googlesource.com/c/113507
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25935}
2018-12-07 16:22:34 +00:00
Sam Zackrisson
6c95e2d56a Fuzz unfuzzed AEC3 killswitch field trials
Bug: webrtc:9413
Change-Id: Iccf861453c1c49c306ad18542074a792592491a9
Reviewed-on: https://webrtc-review.googlesource.com/c/113501
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25928}
2018-12-07 09:38:49 +00:00
Alessio Bazzica
dc107965bd Fix AGC2 fuzzer coverage.
Bug: webrtc:10084
Change-Id: Icc51994fe5ab16188c41452e887cbe7a6b8b9aff
Reviewed-on: https://webrtc-review.googlesource.com/c/112941
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25906}
2018-12-05 15:55:42 +00:00
Niels Möller
648a7cefe1 Delete method EncodedFrame::GetBitstream, part 1
Only caller was the RtpFrameObject constructor, so it's
not needed in the interface.

To be able to delete downstream overrides, add a temporary
default implementation. Method will be completely deleted in part 2.

Bug: webrtc:9378
Change-Id: I9083b6284313b6ebce854c6f2cec4617953331d9
Reviewed-on: https://webrtc-review.googlesource.com/c/112128
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25822}
2018-11-28 14:52:32 +00:00
Johannes Kron
09d6588d93 Change HdrMetadataExtension to ColorSpaceExtension
Bug: webrtc:8651
Change-Id: Ica6f8c6bd13bb07f89700b9c0a359b9a58feefbb
Reviewed-on: https://webrtc-review.googlesource.com/c/111758
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25800}
2018-11-27 14:05:31 +00:00
Mirko Bonadei
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
Jiawei Ou
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
Alessio Bazzica
1e2542f593 AGC2: adding level estimation option (RMS or peak-based).
This CL makes possible to choose the level estimation for the adaptive
digital GC of AGC2. The options are RMS (default and currently used
estimator) and peak-based (already computed, but not used).

Besides adding the new AGC2 config param for the level estimator, this CL
also refactors the config class by making it more structured.

Bug: webrtc:7494
Change-Id: I20eb558ca50f13536aa7bdea08d21de3b630f8bc
Reviewed-on: https://webrtc-review.googlesource.com/c/110144
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25620}
2018-11-13 14:32:13 +00:00
Alex Loiko
20f60f0dc6 Fuzzer crash in AGC2.
Gain specified by fuzzer in APM config was too high.

Bug: chromium:901661
Change-Id: Id3ea8d23a4284a35c827bb16125902d84e37ca1e
Reviewed-on: https://webrtc-review.googlesource.com/c/110604
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25594}
2018-11-12 12:16:47 +00:00
Johannes Kron
ad1d9f0d78 Add RTP header extension for HDR metadata
Bug: webrtc:8651
Change-Id: I1c956eaac1532ac0d3820084edb4054a4cc9c68d
Reviewed-on: https://webrtc-review.googlesource.com/c/109924
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25578}
2018-11-09 11:10:12 +00:00
Alessio Bazzica
b768e8800f Reland "Isolating APM API build target: making :api an actual target."
This reverts commit 61c6e5643e7ea058e653956980a90e033249c055.

Reason for revert: downstream projects prepared for this change

Original change's description:
> Revert "Isolating APM API build target: making :api an actual target."
> 
> This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff.
> 
> Reason for revert: breaking downstream
> 
> Original change's description:
> > Isolating APM API build target: making :api an actual target.
> > 
> > This CL is part of a refactoring work to unblock other CLs
> > that would generate a circular dependency when including
> > modules/audio_processing. It will also allow to easily move
> > the APM interface part under //api.
> > 
> > More in detail, this change moves the APM interface files from
> > the build target modules/audio_processing to
> > modules/audio_processing:api. It also adds :api as dependency
> > where needed.
> > 
> > Bug: webrtc:9535
> > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25539}
> 
> TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
> 
> Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9535
> Reviewed-on: https://webrtc-review.googlesource.com/c/109820
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25540}

TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org

Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109884
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:30:06 +00:00
Alessio Bazzica
61c6e5643e Revert "Isolating APM API build target: making :api an actual target."
This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff.

Reason for revert: breaking downstream

Original change's description:
> Isolating APM API build target: making :api an actual target.
> 
> This CL is part of a refactoring work to unblock other CLs
> that would generate a circular dependency when including
> modules/audio_processing. It will also allow to easily move
> the APM interface part under //api.
> 
> More in detail, this change moves the APM interface files from
> the build target modules/audio_processing to
> modules/audio_processing:api. It also adds :api as dependency
> where needed.
> 
> Bug: webrtc:9535
> Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25539}

TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org

Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109820
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25540}
2018-11-07 11:28:03 +00:00
Alessio Bazzica
a7f77a7c05 Isolating APM API build target: making :api an actual target.
This CL is part of a refactoring work to unblock other CLs
that would generate a circular dependency when including
modules/audio_processing. It will also allow to easily move
the APM interface part under //api.

More in detail, this change moves the APM interface files from
the build target modules/audio_processing to
modules/audio_processing:api. It also adds :api as dependency
where needed.

Bug: webrtc:9535
Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
Reviewed-on: https://webrtc-review.googlesource.com/c/109501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25539}
2018-11-07 10:34:51 +00:00
Karl Wiberg
2365936b87 Hide the AudioEncoderCng class behind a create function
And put codecs/cng/webrtc_cng.h in a non-public build target while
we're at it.

Bug: webrtc:8396
Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c
Reviewed-on: https://webrtc-review.googlesource.com/c/108782
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25486}
2018-11-02 13:00:05 +00:00
Sam Zackrisson
281276301c Remove deprecated AudioProcessing::GetStatistics function
Additionally, AudioProcessing::GetStatistics(bool) is made pure
virtual and the default implementation in AudioProcessing is removed.

Deprecation PSA:
https://groups.google.com/forum/#!msg/discuss-webrtc/NgqEPvkNuDE/7HtwnMmADgAJ

Bug: webrtc:9947, webrtc:8572
Change-Id: I123402bf7d6c49f3613154c469b818109d8fad43
Reviewed-on: https://webrtc-review.googlesource.com/c/108783
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25463}
2018-11-01 11:21:15 +00:00
Qingsi Wang
7852d291c9 Improve the documentation of MdnsResponderInterface and rename MDns.*
to Mdns.*.

MdnsResponderInterface now explicitly requires the reference counting
of created names to allow the coexistence of multiple users of the same
responder where one user would not remove identical names created by
others.

MDns.* is also renamed to Mdns.* per the style guide.

TBR=aleloi@webrtc.org

Bug: webrtc:9605
Change-Id: I047fc41f34de8d4e97c980409a7f373769c4c252
Reviewed-on: https://webrtc-review.googlesource.com/c/101921
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25458}
2018-11-01 02:39:59 +00:00
Per Åhgren
8b7d206d37 AEC3: Decrease latency until the delay has been detected
This CL utilizes the existing, but unused, ability to set
different histogram thresholds for early and late delay
estimation. It does so by tuning the parameters for these.

On top of that, some corrections are added to correctly
handle resets and the use of the hysteresis thresholds.

Bug: webrtc:19886,chromium:896334
Change-Id: I950ac107c124541af8f02b4403f477dda71cc1a1
Reviewed-on: https://webrtc-review.googlesource.com/c/106706
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25443}
2018-10-31 07:29:48 +00:00
Yves Gerey
9516c38538 [Fuzzer] Check FieldTrial bitmask size at compile time.
Rather fail at compile time than at run-time.

Bug: chromium:898373
Bug: webrtc:9855
Change-Id: Iaae81e04e4a8135814c1226f82d3a994de75e9ad
Reviewed-on: https://webrtc-review.googlesource.com/c/107886
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25364}
2018-10-25 13:21:31 +00:00
Sam Zackrisson
262047055d Update fuzzer max input length handling
The docs have been updated. max_len is libfuzzer specific, new way is
fuzzer agnostic.

Docs:
https://chromium.googlesource.com/chromium/src/+/master/testing/libfuzzer/getting_started.md#improving-your-fuzz-target

Bug: chromium:895082
Test: flexfec_sender_fuzzer input size still converges at <=200 after running locally for 5-10 minutes.
Change-Id: I7a5ce95cb4d8b8ca461f6e502b81b599daa855f9
Reviewed-on: https://webrtc-review.googlesource.com/c/107883
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25361}
2018-10-25 12:19:18 +00:00
Jonas Olsson
e068ad6262 Use a sufficiently large bitmask.
The fuzzer uses a bitmask to construct the field trials string.
Now that there's 33 relevant field trials it's no longer large enough, so switch to a 64-bit type.

Bug: chromium:898373
Change-Id: I1ea68d451ceadbd9b720079a577b573866293e4b
Reviewed-on: https://webrtc-review.googlesource.com/c/107650
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25330}
2018-10-24 09:27:18 +00:00
Alex Loiko
4842c78e93 Increasing APM fuzzer coverage.
Add AecDump to the list of fuzzed stuff. Attaches an AecDump to the
Audio Processing Module in the APM-fuzzer. The AecDump writes to
/dev/null.

Bug: webrtc:7820
Change-Id: I03916ce4d1c69906ca8bb7e6fbe29c11e4ea55e5
Reviewed-on: https://webrtc-review.googlesource.com/c/107622
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25321}
2018-10-23 15:11:02 +00:00
Gustaf Ullberg
6ed37ba1b5 AEC3: Enable fuzzer testing of old render buffering code.
The old render buffering code has been replaced, but can still be
activated by a killswitch. This change enables fuzzer testing of
the old code path.

Bug: webrtc:9726
Change-Id: I6e91cd4b4a95388cc63d1a65dade21b3c44be71b
Reviewed-on: https://webrtc-review.googlesource.com/c/107562
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25303}
2018-10-23 08:43:00 +00:00
Henrik Lundin
961dbeac82 NetEq fuzzer: Restrict fuzzer input to 90000 bytes
This is to avoid very long runs, resulting in time-outs.

NOTRY=True

Bug: chromium:895082
Change-Id: Iafdc3d10b3fb52f2d487547c954dca8ae7edb783
Reviewed-on: https://webrtc-review.googlesource.com/c/105960
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25175}
2018-10-15 15:36:55 +00:00
Gustaf Ullberg
53e22113fd AEC3: Kill kill-switches
"Perfection is achieved, not when there is nothing more to add,
but when there is nothing left to take away."

This CL removes the following kill-switches from AEC3
- WebRTC-Aec3DownSamplingFactor8KillSwitch
- WebRTC-Aec3NewSuppressionKillSwitch
- WebRTC-Aec3ShadowFilterJumpstartKillSwitch
- WebRTC-Aec3SlowFilterAdaptationKillSwitch
- WebRTC-Aec3SuppressorNearendAveragingKillSwitch

It also removes code paths and configuration parameters that are no
longer in use. The list of kill-switches in the audio processing
fuzzer test is updated.

The change has been tested for bit-exactness.

Bug: webrtc:8671
Change-Id: Ie0af86a14baf853548bf9c00b2b9b3bbc32c1aaa
Reviewed-on: https://webrtc-review.googlesource.com/c/105324
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25120}
2018-10-11 16:11:07 +00:00
Jonathan Metzman
9f80b97309 Fix fuzzer build failures on Windows
Fix the following issues with fuzz targets when built on Windows:
1. Fix audio_processing_fuzzer by making types match in
invocations of CheckedDivExact by explicitly casting to size_t.
2. Fix packet_buffer_fuzzer by including "frame_object.h" for
declaration of RtpFrameObject.
3. Fix rtcp_receiver_fuzzer by including "tmmb_item.h" for declaration
of TmmbItem.

Bug: chromium:891867
Change-Id: Iddc338360ca37d5fc31488ec908eb4cdb5cc7b94
Reviewed-on: https://webrtc-review.googlesource.com/c/103844
Commit-Queue: Jonathan Metzman <metzman@chromium.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25028}
2018-10-05 18:13:32 +00:00
philipel
c71cd6c31d Don't ovewrite complex member VCMPacket::generic_descriptor when fuzzing.
In https://webrtc-review.googlesource.com/c/src/+/102720 a new complex member
was added to VCMPacket. This member was overwritten with random data in the
fuzzer, which put it in an invalid state. To avoid that we save/restore it.

Bug: chromium:891597
Change-Id: I7b489afa727a028a542fbec55a4ee27ac54fa698
Reviewed-on: https://webrtc-review.googlesource.com/c/103462
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24972}
2018-10-04 09:22:13 +00:00
Mirko Bonadei
e0d455b409 Remove runtime_enabled_feature.
This features is not needed anymore, with this CL it is also possible
to address two issues:
- The need to pick a default implementation.
- The need to use -Wno-global-constructors.

Bug: webrtc:9631, webrtc:9693
Change-Id: Id3daf34179fbc8db26969fc701ccbfa7182c6a9b
Reviewed-on: https://webrtc-review.googlesource.com/102543
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24904}
2018-10-01 07:03:25 +00:00
Mirko Bonadei
17f4878419 Remove deprecated field_trial_default and metrics_default.
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default

It also refreshes all the dependencies on field_trial.h and metrics.h.

A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm

Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 07:21:07 +00:00
Sam Zackrisson
1c718f91de Update configuration of builtin AECs in APM fuzzer
The old pointer-to-submodule interface is replaced with
AudioProcessing::Config settings.

Bug: webrtc:9535
Change-Id: I5580d690fdd7664f48fa274b39f12cc41f69da37
Reviewed-on: https://webrtc-review.googlesource.com/102020
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24847}
2018-09-26 11:19:46 +00:00
Christian Fremerey
ee002e6185 Fix WebRTC fuzzers tests in Chromium missing field trial implementation
Follow-up to https://webrtc-review.googlesource.com/c/src/+/100940.

When WebRTC fuzzers tests are built on Chromium bots they need to link
with Chromium's implementation of field_trial.

This is for fixing the roll out WebRTC into Chromium. Example failure:
https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux-libfuzzer-asan-rel/4551

TBR=phoglund@webrtc.org

Bug: webrtc:9631
Change-Id: I353a2d293beafe016ce0c03d88e09fc5af23598f
Reviewed-on: https://webrtc-review.googlesource.com/101102
Commit-Queue: Christian Fremerey <chfremer@webrtc.org>
Reviewed-by: Christian Fremerey <chfremer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24778}
2018-09-19 20:09:14 +00:00
Mirko Bonadei
15212f3d4e Fix WebRTC fuzzers tests in Chromium.
When WebRTC fuzzers tests are built on Chromium bots they need to link
with Chromium's implementation of metrics.

TBR=phoglund@webrtc.org

Bug: webrtc:9631
Change-Id: I1c955e646366b6b37d3ca595888e8cc94fe1b00e
Reviewed-on: https://webrtc-review.googlesource.com/100940
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christian Fremerey <chfremer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24771}
2018-09-18 22:09:06 +00:00
Jonas Olsson
941a07cca3 Remove all remaining non-test uses of std::stringstream.
Bug: webrtc:8982
Change-Id: I635a8545c46dc8c89663d64af351e22e65cbcb33
Reviewed-on: https://webrtc-review.googlesource.com/98880
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24715}
2018-09-13 08:52:05 +00:00
Johnny Lee
e0c8b230e7 Frame marking RTP header extension (PART 1: implement extension)
Bug: webrtc:7765
Change-Id: I23896d121afd6be4bce5ff4deaf736149efebcdb
Reviewed-on: https://webrtc-review.googlesource.com/85200
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24695}
2018-09-11 22:35:30 +00:00
Qingsi Wang
558b93b3e9 Add the multicast DNS message format.
This CL adds the utilities to generate and parse mDNS messages (RFC 1035
and RFC 6762).

TBR=phoglund@webrtc.org

Bug: webrtc:9605
Change-Id: Id6121c17926887cd3a41a2dfc829462fd15f3a4c
Reviewed-on: https://webrtc-review.googlesource.com/93241
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24505}
2018-08-31 00:02:44 +00:00
Alessio Bazzica
cc22f51988 Removing the intelligibility enhancer.
The intelligibility enhancer is always disabled and it is the only non-test
target using the lapped transform in common_audio (which we planned to remove).

Bug: webrtc:9689, webrtc:5298
Change-Id: Ida65d3aa11ac366471e7e5cbc053108b376c67d8
Reviewed-on: https://webrtc-review.googlesource.com/96460
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24504}
2018-08-30 21:29:57 +00:00
Niels Möller
2377588c82 Add accessor methods for RTP timestamp of EncodedImage.
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.

Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
2018-08-21 09:15:51 +00:00
Per Åhgren
ef5d5af3a0 AEC3: Increasing the accuracy of the detection for early reverb
This CL introduces an adaptive estimation of the early reverb
in the estimation for the room reverberation. The benefits of
this is that for room with long early reflections there is
a lower risk of underestimating the reverberation.

This CL is for a landing the code in
https://webrtc-review.googlesource.com/c/src/+/87420,
and the review of the code was done in that CL. The author of
code is devicentepena@webrtc.org

Bug: webrtc:9479, chromium:865397
Change-Id: Id6f57e2a684664aef96e8c502e66775f37da59da
Reviewed-on: https://webrtc-review.googlesource.com/91162
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24146}
2018-07-30 22:34:19 +00:00
Sam Zackrisson
151ba0f077 Fuzz unfuzzed AEC3 killswitch field trials
Bug: webrtc:9413
Change-Id: I09d8c673d6d8e2efd77bc9f311001a5843a556a2
Reviewed-on: https://webrtc-review.googlesource.com/90870
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24131}
2018-07-27 14:17:37 +00:00
Niels Möller
a12c42a6b2 Delete root header file typedef.h.
Usage replaced with stdint.h, rtc_base/system/arch.h and
rtc_base/system/unused.h, as appropriate.

Bug: webrtc:6854
Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18
Reviewed-on: https://webrtc-review.googlesource.com/90249
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 14:59:26 +00:00
Mirko Bonadei
e9b1854b69 Adding missing dependencies on absl.
Bug: None
Change-Id: I8b429bd3e590bebfb58c622eb750490e7c062b3f
Reviewed-on: https://webrtc-review.googlesource.com/90241
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24077}
2018-07-24 08:50:26 +00:00
Sam Zackrisson
2a959d96c9 Revert "Add one-stop-shop for built-in AEC toggling in APM"
This reverts commit 771b50ca0b92477ce8aabba025e959790dd1a949.

Reason for revert: Introduces error-prone config.

Original change's description:
> Add one-stop-shop for built-in AEC toggling in APM
> 
> This does not change what AEC functionality is available.
> However, a client that only uses this interface - and not the submodule
> pointer accessors - gets simpler code, and is guaranteed not to run any
> two AECs in tandem.
> 
> The submodule interface EchoControlMobile is being deprecated in
> https://webrtc-review.googlesource.com/c/src/+/89392
> 
> Bug: webrtc:9535
> Change-Id: Id9326074e566be6d8768010fc421c457beff402c
> Reviewed-on: https://webrtc-review.googlesource.com/89386
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24066}

TBR=saza@webrtc.org,peah@webrtc.org

Change-Id: I43283a1b22538a4caa77313499989146b2ce67f1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/90060
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24067}
2018-07-23 14:48:17 +00:00
Sam Zackrisson
771b50ca0b Add one-stop-shop for built-in AEC toggling in APM
This does not change what AEC functionality is available.
However, a client that only uses this interface - and not the submodule
pointer accessors - gets simpler code, and is guaranteed not to run any
two AECs in tandem.

The submodule interface EchoControlMobile is being deprecated in
https://webrtc-review.googlesource.com/c/src/+/89392

Bug: webrtc:9535
Change-Id: Id9326074e566be6d8768010fc421c457beff402c
Reviewed-on: https://webrtc-review.googlesource.com/89386
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24066}
2018-07-23 14:12:26 +00:00
Alex Loiko
f5c3ba15f0 Fuzz more kinds of floats in the APM fuzzer.
Previously, the fuzzer read a int16_t and converted to float. That is
how float audio samples were generated. This CL changes the fuzzer to
read floats directly, and then sanitize them.

Bug: webrtc:7820
Change-Id: Icc526611466c10dd4222b19a4d4b4fd26643812a
Reviewed-on: https://webrtc-review.googlesource.com/85343
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24001}
2018-07-17 09:36:35 +00:00
Sam Zackrisson
35c773dad6 Cap the number of fuzzed decoder packets to 200
The fuzzer figured out that 3 bytes is enough to fuzz a package.
2 bytes for packet length, and 1 byte of actual packet. A 20K test case
can generate > 6000 packets. It does not seem like efficient fuzzing.

This CL simply stops execution when 200 packets have been generated.
That corresponds to 4 seconds of 20 ms packets.

Bug: chromium:840115
Change-Id: Id2742a6f8021134bacd8a6e8c71b32f20c7f1086
Reviewed-on: https://webrtc-review.googlesource.com/88566
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24000}
2018-07-17 09:14:45 +00:00
Per Åhgren
88cf0501f3 AEC3: Adding explicit handling of microphone gain changes
This CL re-activates the explicit handling of microphone
gain changes in the AEC3 code. The implementation is done
beneath a kill-switch so that when that switch is active
the changes in this CL are bitexact.


Bug: webrtc:9526,chromium:863826
Change-Id: I58e93d8bc0bce7bec91e102de9891ad48ebc55d8
Reviewed-on: https://webrtc-review.googlesource.com/88620
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23986}
2018-07-16 16:02:07 +00:00
Sam Zackrisson
b5d3802b7a Fuzz more aec3 field trials
There are many new killswitches for AEC3 code.
By fuzzing them, we ensure the code is more thoroughly tested in case
we need to trigger any switches.

Bug: webrtc:9413
Change-Id: I0bc7609e4d71fc820abfbba7d481b2374c3587cb
Reviewed-on: https://webrtc-review.googlesource.com/88225
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23960}
2018-07-13 08:42:11 +00:00
Sam Zackrisson
58c79f66dd Add saza to fuzzer owners
Bug: None
Change-Id: Ib9803f3f1519822eb1cdae9fec6291f807f6c0f3
Reviewed-on: https://webrtc-review.googlesource.com/88367
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23950}
2018-07-12 11:12:00 +00:00