Commit Graph

357 Commits

Author SHA1 Message Date
6a9bd74481 Fix a downstream test failure.
In rare case the packets number may loop around and in the same FEC-protected group the packet sequence number became out of order.

Bug: chromium:850493
Change-Id: Ice82aafd537e0edc1dbdb8b934e11e7c42a4cf60
Reviewed-on: https://webrtc-review.googlesource.com/82802
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23633}
2018-06-15 13:30:26 +00:00
d1f970dc43 Change echo detector to scoped_refptr
The echo detector is currently stored as a unique_ptr, but when injecting an echo detector, a scoped_refptr makes more sense since the ownership will be shared.

Bug: webrtc:8732
Change-Id: I2180014acb84f1cd5c361864a444b7b6574520f5
Reviewed-on: https://webrtc-review.googlesource.com/83325
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23610}
2018-06-14 09:51:41 +00:00
0c87e293c9 Update packet_buffer_fuzzer to fuzz full packets.
Bug: webrtc:7728
Change-Id: I9d33404470c2ecf8d6f91c57c9dc9fd4dd821a18
Reviewed-on: https://webrtc-review.googlesource.com/77424
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23485}
2018-06-01 10:00:36 +00:00
547e3169d9 Limit input length for SDP fuzzer.
This limits the SDP to 16KB, which sounds enough.

Bug: chromium:813328
Change-Id: I58c7b3e073108fd7b3495e8182b5c632e9619fe7
Reviewed-on: https://webrtc-review.googlesource.com/78280
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23360}
2018-05-23 10:33:40 +00:00
65ec0fc81e Delete unneeded includes of basictypes.h.
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.

Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.

Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
2018-05-21 19:35:08 +00:00
1e9cf7faf8 Fuzzing for video_coding::FrameBuffer2.
Bug: webrtc:7728
Change-Id: I712289a82d408dde1db73a1cc44f0c69a6b639ff
Reviewed-on: https://webrtc-review.googlesource.com/31841
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23282}
2018-05-17 12:41:41 +00:00
c1ee9d57d1 RtpFrameReferenceFinder fuzzer.
Bug: webrtc:7728
Change-Id: I641772837384a4d8070db2138b93f4157f997d03
Reviewed-on: https://webrtc-review.googlesource.com/74584
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23271}
2018-05-17 08:55:31 +00:00
a29b148557 Create a fuzzer for the Opus encoder
The fuzzer is very simple. It only considers the default encoder
configuration at this point.

Bug: chromium:826914
Change-Id: Ifa248a1dba80efb231807750e40082ec5580636a
Reviewed-on: https://webrtc-review.googlesource.com/75261
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23192}
2018-05-09 13:35:23 +00:00
bbf21a3fd6 Remove dependencies on modules:module_api from AudioProcessing.
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.

Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:05:27 +00:00
9df3cf3e8c Stats calls improve APM fuzzer coverage.
A part of the Audio Processing Module interface is GetStatistics. The
call collects stats from submodules. We make sure these calls are made
by the fuzzer to cover that code path.

Bug: webrtc:7820
Change-Id: Ia8f89d9838602dcb2599f676bd5c43e815bbf791
Reviewed-on: https://webrtc-review.googlesource.com/68980
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22817}
2018-04-11 08:47:04 +00:00
f0482ea9dd Add MID sending to FlexfecSender
Bug: webrtc:4050
Change-Id: I1eefd99cca1c02751d3f5a2d3b57625ccb45323f
Reviewed-on: https://webrtc-review.googlesource.com/64321
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22811}
2018-04-10 16:08:35 +00:00
8fabab1509 CNG fuzzer: avoid long fuzzer runs by limiting generator calls
The number of calls to ComfortNoiseDecoder::Generate() was determined
by the fuzzer input, and was chosen between 0 and 255. This would
sometimes lead to very long runs, with questionable merit. With this
change, the number of call to Generate() is limited to 17 (an
arbitrary small integer).

Bug: chromium:820078
Change-Id: I27b5c7f0b72d53370d002a6b157d4451079a0ba9
Reviewed-on: https://webrtc-review.googlesource.com/60941
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22360}
2018-03-09 13:16:44 +00:00
12edf4ce34 Separate build target for rtc_base/numerics/safe_minmax.h
So that we can avoid dependency cycles.

Bug: none
Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8
Reviewed-on: https://webrtc-review.googlesource.com/60500
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-07 14:12:00 +00:00
38c15d3995 Template argument and corpora for Audio Processing Fuzzer.
We found out that

  int16_t x = test::FuzzDataHelper::ReadOrDefaultValue(0)

reads 4 bytes from the fuzzer input instead of 2. That means that
almost half the bits in the input data to audio_processing_fuzzer are
ignored. This change adds template arguments to force reading 2 bytes
when we only need 2.

We also add a small manually generated corpus. During local testing we
let the fuzzer run for a few hours on an empty corpus. Adding the
manually-generated files resulted in an immediate coverage increase by
~3%, and then by another 3% over the next few hours.

The manually generated corpus contains a short segment of speech with
real echo. We suspect that triggering Voice Activity Detection or echo
estimation filter convergence can be difficult for an automatic
fuzzer.

We remove the Level Controller config. We read 20 bytes extra after the
config to guard against future configuration changes.

Bug: webrtc:7820
Change-Id: If60c04f53b27c519c349a40bd13664eef7999368
Reviewed-on: https://webrtc-review.googlesource.com/58744
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22269}
2018-03-02 14:00:39 +00:00
0efa941d2f Move EchoCanceller3Factory to api/auido
The AEC3 factory is now part of the WebRTC API.

Bug: webrtc:8844
Change-Id: If6f419b4ca0354e2d346c0e6474086e456ba747e
Reviewed-on: https://webrtc-review.googlesource.com/57141
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22204}
2018-02-27 14:09:59 +00:00
151be2dffc comfort_noise_decoder_fuzzer: limit the fuzzer input size to avoid timeout
The length of the fuzzer input can sometimes be really long (more than
1000000 bytes), and this take a very long time to execute. Typically,
the fuzzer times out instead. This change limits the used length of
the fuzzer to 200000 bytes.

NOTRY=TRUE

Bug: chromium:802149
Change-Id: Ia9d2f080602bba8ff70c5f0575bb9ecfa99c537c
Reviewed-on: https://webrtc-review.googlesource.com/57581
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22183}
2018-02-26 10:52:30 +00:00
06fa1539f8 neteq_rtp_fuzzer: limit the fuzzer input size to avoid timeout
The length of the fuzzer input can sometimes be really long (more than
600000 bytes), and this take a very long time to execute. Typically,
the fuzzer times out instead. This change limits the used length of
the fuzzer to 100000 bytes.

NOTRY=TRUE

Bug: chromium:802193
Change-Id: Id32174611fadb480f4e2c6b4f553a2ba0fa5b493
Reviewed-on: https://webrtc-review.googlesource.com/57580
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22182}
2018-02-26 10:49:20 +00:00
2a6d864264 neteq_signal_fuzzer: limit the fuzzer input size to avoid timeout
The length of the fuzzer input can sometimes be really long (more than
600000 bytes), and this take a very long time to execute. Typically,
the fuzzer times out instead. This change limits the used length of
the fuzzer to 100000 bytes.

NOTRY=TRUE

Bug: chromium:802245
Change-Id: Ibe02b6de932d900408f870d9ba440b7b8e08dc0e
Reviewed-on: https://webrtc-review.googlesource.com/57180
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22181}
2018-02-26 10:42:30 +00:00
f35c6667d6 Separate build targets for aec3 and aec3_unittests
Bug: webrtc:8844
Change-Id: Id6a98eae19aaedc87c3f402a004f58f0290d5c28
Reviewed-on: https://webrtc-review.googlesource.com/56580
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22173}
2018-02-23 13:16:16 +00:00
a05ee82c4c Fixed Digital mode of AGC2 implementation finished.
This CL adds the GainCurveApplier (GCA). It owns a
FixedDigitalLevelEstimator (LE) and an InterpolatedGainCurve
(IGC). The GCA uses the LE to compute the input signal level, looks up
a gain from IGC and applies it on the signal.

The other IGC and LE submodules were added in previous CLs [1] and
[2].

This CL also turns on AGC2 in the APM fuzzer.

[1] https://webrtc-review.googlesource.com/c/src/+/51920
[2] https://webrtc-review.googlesource.com/c/src/+/52381

Bug: webrtc:7949
Change-Id: Idb10cc3ca9d6d2e4ac5824cc3391ed8aa680f6cd
Reviewed-on: https://webrtc-review.googlesource.com/54361
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22103}
2018-02-20 15:59:25 +00:00
61405bcb19 Fix infinite loop in rtp packet parsing
when rtp header extension is larger than 2^16 bytes

Bug: chromium:811613
Change-Id: I05b725d734dd628056d603b596d3523e827ddb54
Reviewed-on: https://webrtc-review.googlesource.com/52345
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22003}
2018-02-13 14:42:45 +00:00
c2dd59c25d Skip oversized rtp header extension when parsing Rtp Packet.
Rtp Packets in webrtc expected to be less that 1500,
i.e. way less that 2^16 bytes for extensions block.
This CL explicitly discards longer extension.

Bug: chromium:809046
Change-Id: Ibed33b51bafc3fd4804ec135f66110c6d2796734
Reviewed-on: https://webrtc-review.googlesource.com/48061
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21910}
2018-02-06 11:30:08 +00:00
970b088878 Reland "Break up rtc_event_log_api to solve circular dependencies."
This is a reland of 001546da953275c7a39eb220592b440c9b47d756
Original change's description:
> Break up rtc_event_log_api to solve circular dependencies.
>
> The original rtc_event_log_api is refactored to a pure API target plus
> multiple targets coupled with WebRTC implementations.
>
> Bug: None
> Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
> Reviewed-on: https://webrtc-review.googlesource.com/43247
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#21811}

TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org

Bug: None
Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647
Reviewed-on: https://webrtc-review.googlesource.com/46900
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21862}
2018-02-01 22:47:52 +00:00
75df7282eb Revert "Break up rtc_event_log_api to solve circular dependencies."
This reverts commit 001546da953275c7a39eb220592b440c9b47d756.

Reason for revert: breaks downstream projects.

Original change's description:
> Break up rtc_event_log_api to solve circular dependencies.
> 
> The original rtc_event_log_api is refactored to a pure API target plus
> multiple targets coupled with WebRTC implementations.
> 
> Bug: None
> Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
> Reviewed-on: https://webrtc-review.googlesource.com/43247
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#21811}

TBR=phoglund@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,pthatcher@google.com,pthatcher@webrtc.org,qingsi@google.com

Change-Id: I82540eac176c4abfb7e50dc51671585b32a1bace
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/46581
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21823}
2018-01-31 09:39:44 +00:00
001546da95 Break up rtc_event_log_api to solve circular dependencies.
The original rtc_event_log_api is refactored to a pure API target plus
multiple targets coupled with WebRTC implementations.

Bug: None
Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
Reviewed-on: https://webrtc-review.googlesource.com/43247
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#21811}
2018-01-30 17:54:06 +00:00
393e266470 Use correct RTP header length in RED generation for ULPFEC packets.
Prior to this change, in certain circumstances the RTP header length
used when creating a RedPacket was incorrect. This was due to an
assumption that a new media packet would _always_ be added to the
UlpfecGenerator's internal media packet buffer. This is not correct,
and the fix is to keep track of whatever RTP header length that is
currently correct.

Bug: webrtc:8767
Change-Id: I6d61429a19d4693dde9330f0469d13c5dfbeac52
Reviewed-on: https://webrtc-review.googlesource.com/40600
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21720}
2018-01-22 15:12:08 +00:00
2ffe3e80db Reland Use runtime enabled features API to enable dual stream mode
This is an unchanged patch after dependency fixes in downstream projects are implemented.

Original patch was reviewed here:
https://webrtc-review.googlesource.com/c/src/+/39008

TBR=phoglund@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org,philipel@webrtc.org,lliuu@webrtc.org

Change-Id: I648bbf63d34282a48cabc854615005ec65b28cb3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8287
Reviewed-on: https://webrtc-review.googlesource.com/40420
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21674}
2018-01-18 12:22:49 +00:00
c1094eb81d Revert "Use runtime enabled features API to enable dual stream mode"
This reverts commit 6f011dfdd413a903dcdf5a23a49109e64432326d.

Reason for revert: Broke internal builds

Original change's description:
> Use runtime enabled features API to enable dual stream mode
> 
> Bug: webrtc:8287
> Change-Id: I1a366d959a8b7f2a704baa7ea8ace64c1c398d52
> Reviewed-on: https://webrtc-review.googlesource.com/39008
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21661}

TBR=phoglund@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org,philipel@webrtc.org

Change-Id: I0af406066231b67dd0b8eb6808bdc3e3f77560b6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8287
Reviewed-on: https://webrtc-review.googlesource.com/40321
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21663}
2018-01-17 19:45:48 +00:00
6f011dfdd4 Use runtime enabled features API to enable dual stream mode
Bug: webrtc:8287
Change-Id: I1a366d959a8b7f2a704baa7ea8ace64c1c398d52
Reviewed-on: https://webrtc-review.googlesource.com/39008
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21661}
2018-01-17 16:29:37 +00:00
a8b7c7f4c6 Move remaining traces of VoiceEngine
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
  utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.

NOPRESUBMIT=true

Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 13:27:47 +00:00
ab20a6016c AEC-m and AEC-2 fuzzing.
Going through the coverage of audio_processing_fuzzer, it was noticed
that it didn't cover AEC-m and AEC-2 code. Therefore this CL adds 2
fuzzer targets that only fuzz the previous generation echo cancellers.

To avoid code duplication, the APM running code was broken out in a
new GN target. We have also changed all fuzzing code to use the
FuzzDataHelper class to avoid manual pointer arithmetic.

Bug: webrtc:7820
Change-Id: Ifea3266e396b487952a736945577fccea15d0e01
Reviewed-on: https://webrtc-review.googlesource.com/36500
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21638}
2018-01-16 13:15:04 +00:00
3ac67a736b Make aleloi@webrtc.org owner of test/fuzzers
aleloi has read the Chromium fuzzer guides and covered lots of APM
code with fuzzing tests. He'd like to share responsibility for making
fuzzers faster and have high coverage.

Bug: None
NOTRY: True
Change-Id: I45db63349ca9d4432ebc69ed3c84ec2fc0f3f227
Reviewed-on: https://webrtc-review.googlesource.com/39923
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21637}
2018-01-16 12:57:44 +00:00
75baa498fa Stop using public_deps in media/.
Bug: webrtc:8603
Change-Id: I7a6dad323ac298dc784feb5aa1fdc2ae5876cb5c
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/33180
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21591}
2018-01-12 08:15:01 +00:00
62337e59dd Use AudioProcessingBuilder everywhere AudioProcessing is created.
The AudioProcessingBuilder was recently introduced in https://webrtc-review.googlesource.com/c/src/+/34651 to make it easier to create APM instances. This CL replaces all calls to the old Create methods with the new AudioProcessingBuilder.

Bug: webrtc:8668
Change-Id: Ibb5f0fc0dbcc85fcf3355b01bec916f20fe0eb67
Reviewed-on: https://webrtc-review.googlesource.com/36082
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21534}
2018-01-09 13:45:20 +00:00
97cb448d25 Update Webrtc to new AudioProcessing API.
webrtc::PostProcessor changed to webrtc::CustomProcessor and one APM
factory method has been deprecated.

The APM API changed in this cl: https://webrtc-review.googlesource.com/c/src/+/29201

TBR=henrik.lundin@webrtc.org, sakal@webrtc.org

Bug: webrtc:8665
Change-Id: I76dfc7831575d4dfce7e60cbe22007bd2a50e946
Reviewed-on: https://webrtc-review.googlesource.com/34381
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21451}
2017-12-27 09:03:59 +00:00
3e113438b1 Fix circular dependencies in webrtc_common.
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.

I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.

Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:33:26 +00:00
a8005cfd8b Fix circular dependencies between optional, array_view, and rtc_base.
This splits things out of rtc_base and makes dependencies explicit.

Bug: webrtc:6828
Change-Id: Id521896c3c43595349021c857bec216e429a0c8d
Reviewed-on: https://webrtc-review.googlesource.com/32780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-14 06:49:11 +00:00
d37709b659 Revert "Fix circular dependencies between optional, array_view, and rtc_base."
This reverts commit a9e0924fa7688c4e4558e179c6608ce1093e15f8.

Reason for revert: Breaks because of RTC_LAST_SYSTEM_ERROR

Original change's description:
> Fix circular dependencies between optional, array_view, and rtc_base.
> 
> This splits things out of rtc_base and makes dependencies explicit.
> 
> Bug: webrtc:6828
> Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
> Reviewed-on: https://webrtc-review.googlesource.com/31940
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21245}

TBR=phoglund@webrtc.org,kwiberg@webrtc.org

Change-Id: I1a5dcf2223f00ae7c46f9f2a12b990ab3a84397d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6828
Reviewed-on: https://webrtc-review.googlesource.com/32760
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21251}
2017-12-13 14:56:33 +00:00
49ccbdb9d6 Add fuzzer for ForwardErrorCorrection::DecodeFec.
Bug: webrtc:8481
Change-Id: I23aa59ffee542c1c0b31c82186876ccc21e28592
Reviewed-on: https://webrtc-review.googlesource.com/32305
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21248}
2017-12-13 14:29:41 +00:00
a9e0924fa7 Fix circular dependencies between optional, array_view, and rtc_base.
This splits things out of rtc_base and makes dependencies explicit.

Bug: webrtc:6828
Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
Reviewed-on: https://webrtc-review.googlesource.com/31940
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21245}
2017-12-13 13:44:21 +00:00
e51f785043 Stop using public_deps in pc/.
TBR=phoglund@webrtc.org

Bug: webrtc:8603
Change-Id: If18e5a4d212392bbd9b4e1f9c2f00ee79a2ab348
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/29864
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21139}
2017-12-07 13:57:57 +00:00
5dcbbfd153 Create a fuzzer for ComfortNoiseDecoder
The fuzzer will hammer on the UpdateSid and Generate methods of
ComfortNoiseDecoder.

The change also includes a fix to an issue in WebRtcSpl_FilterAR, which
was immediately found by running the fuzzer locally.

Bug: none
Change-Id: I5283427cb27844fb953e2caa35423ea873aca2ff
Reviewed-on: https://webrtc-review.googlesource.com/28100
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21134}
2017-12-07 08:53:37 +00:00
b5728d9b0f Stop using public_deps in modules/rtp_rtcp.
TBR=phoglund@webrtc.org

Bug: webrtc:8603
Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/29780
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:37:52 +00:00
3fb614bc93 Remove unused UlpfecGenerator::BuildRedPacket.
BUG=none

Change-Id: I998e23beee9c46dc696631195790e8821d1cc967
Reviewed-on: https://webrtc-review.googlesource.com/24821
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20917}
2017-11-28 16:18:28 +00:00
d7e251378b Fix potential overflow in congestion controller fuzzer.
Bug: chromium:787753
Change-Id: I43d765379216db35f3df748b16599b34bffd388f
Reviewed-on: https://webrtc-review.googlesource.com/25480
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20851}
2017-11-23 14:18:38 +00:00
e40468ba3d Move some numeric utility code from rtc_base/ to rtc_base/numerics/
Specifically, I'm moving

  safe_compare.h
  safe_conversions.h
  safe_minmax.h

They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.

BUG=webrtc:8445

Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
2017-11-22 11:21:47 +00:00
df0822b102 Optional: Use nullopt and implicit construction in /test/fuzzers
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=henrik.lundin@webrtc.org

Bug: None
Change-Id: I446549a385e020c68bafc83d5dd1aabd11d7ae18
Reviewed-on: https://webrtc-review.googlesource.com/23563
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20718}
2017-11-16 18:01:03 +00:00
de6914508e Remove pbos@webrtc.org from all OWNERS.
Bug: None
Change-Id: I49c4df3873f359c20f46a64592a05c3d001b708d
Reviewed-on: https://webrtc-review.googlesource.com/17720
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20517}
2017-11-01 08:03:46 +00:00
ddfd9c5fd2 Fix AudioProcessing fuzzer crash.
When audio_processing_fuzzer runs with 'DCHECK_ALWAYS_ON', it crashes
when both AEC and AECM is enabled at the same time. This change
detects that case and fixes
https://clusterfuzz.com/v2/testcase-detail/6389429496446976.

It also removes an unnecessary safeguard that didn't allow fuzzing
with 8kHz input signals.

Bug: chromium:776358
Change-Id: I33c18a2a235e50ae410f7be24637872823e432eb
Reviewed-on: https://webrtc-review.googlesource.com/15320
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20428}
2017-10-25 12:00:36 +00:00
0c720505af Adding libFuzzer target for UlpFEC receiver.
Bug: none
Change-Id: I20e622455aee2f5aebad835e915d65f3475fbd17
Reviewed-on: https://webrtc-review.googlesource.com/14300
Commit-Queue: Mark Brand <markbrand@google.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20384}
2017-10-23 11:37:07 +00:00