Commit graph

373 commits

Author SHA1 Message Date
Niels Möller
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
Fredrik Solenberg
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
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
Ivo Creusen
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
Mirko Bonadei
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
Niels Möller
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
Sam Zackrisson
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
Niels Möller
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
Fredrik Solenberg
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
Niels Möller
b7180c09fc Replace RegisterExternalDecoder in NetEq test VerifyTimestampPropagation.
Bug: webrtc:10080
Change-Id: Ie93f130863115c2d288cfd9f3e273a9fbc982ed6
Reviewed-on: https://webrtc-review.googlesource.com/c/112904
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25927}
2018-12-07 09:28:47 +00:00
Fredrik Solenberg
657b296ff5 Reland "Remove CodecInst pt.1"
This is a reland of 056f9738bf

Original change's description:
> Remove CodecInst pt.1
> 
> Update audio_coding tests to not use CodecInst.
> 
> Bug: webrtc:7626
> Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/112594
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25879}

Bug: webrtc:7626
Change-Id: I5d6ca0baf6230bfe9bf95c2c25496d2a56812d90
Reviewed-on: https://webrtc-review.googlesource.com/c/112942
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25902}
2018-12-05 10:38:23 +00:00
Fredrik Solenberg
ec0f45be11 Revert "Remove CodecInst pt.1"
This reverts commit 056f9738bf.

Reason for revert: breaks downstream

Original change's description:
> Remove CodecInst pt.1
> 
> Update audio_coding tests to not use CodecInst.
> 
> Bug: webrtc:7626
> Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/112594
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25879}

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

Change-Id: I51d666969bcd63e2b7cb7d669ec2f59b5f8f9dde
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7626
Reviewed-on: https://webrtc-review.googlesource.com/c/112906
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25881}
2018-12-03 15:50:51 +00:00
Fredrik Solenberg
056f9738bf Remove CodecInst pt.1
Update audio_coding tests to not use CodecInst.

Bug: webrtc:7626
Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
Reviewed-on: https://webrtc-review.googlesource.com/c/112594
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25879}
2018-12-03 15:16:20 +00:00
Niels Möller
a0f4430b3a Replace RegisterExternalDecoder with decoder factory in NetEqImplTest120ms
Change-Id: I86b5f748f556be186f020a97fcc1211f953fd219

Bug: webrtc:10080
Change-Id: I86b5f748f556be186f020a97fcc1211f953fd219
Reviewed-on: https://webrtc-review.googlesource.com/c/112600
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@{#25869}
2018-12-03 08:34:50 +00:00
Yves Gerey
3e70781361 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.

Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 18:25:07 +00:00
Niels Möller
bb9f4c1252 Delete ssrc book-keeping in NetEq
The ssrc for a given NetEq instance shouldn't change.

Bug: webrtc:7135
Change-Id: Iee0d4cd8bd5d917e819fa2ecf45a40e203c6d9c0
Reviewed-on: https://webrtc-review.googlesource.com/c/111661
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25825}
2018-11-28 15:33:14 +00:00
Jakob Ivarsson
10403ae87c Add PeerConnection option to configure minimum audio jitter buffer delay.
Note that this value will override the minimum delay that is used for audio/video sync.

Bug: webrtc:10053
Change-Id: Ia129f6c9ee9da5d00a3d955afaaa6e8f0c2bee33
Reviewed-on: https://webrtc-review.googlesource.com/c/112121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25805}
2018-11-27 19:49:48 +00:00
Jakob Ivarsson
352ce5c419 Expose delayed packet outage as a cumulative metric of samples in the new getStats API.
The stat will be exposed through origin trial described in:
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI

Change-Id: Ib191a11c6bd9e617abbe9dd82239b0c5b4e6b4e0
Bug: webrtc:10043
Reviewed-on: https://webrtc-review.googlesource.com/c/111922
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25802}
2018-11-27 15:10:09 +00:00
Niels Möller
53382cb19f Move RtcpStatistics from common_types.h to a new header file
New location is modules/rtp_rtcp/include/rtcp_statistics.h.

Bug: webrtc:5876
Change-Id: I85f55b58658588228ed77175226b3479352fd5de
Reviewed-on: https://webrtc-review.googlesource.com/c/111961
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25799}
2018-11-27 13:46:42 +00:00
Jonas Olsson
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
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
Ruslan Burakov
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
Niels Möller
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
Minyue Li
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
Sebastian Jansson
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
Niels Möller
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
Niels Möller
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
Fredrik Solenberg
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
Yves Gerey
a038e71b48 Less strict audio codec tests to accomodate opus switch to SSE.
Expected checksums depend on whether libopus is built with SSE or not.
Since we have no robust way to know that and we cannot enforce all
clients to use SSE, we accept both results.

Bug: webrtc:9530
Bug: webrtc:9995
Change-Id: I9f0464ffec15df91eafe15d89c61e2140f341cb1
Reviewed-on: https://webrtc-review.googlesource.com/c/110789
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25633}
2018-11-14 10:16:04 +00:00
Piotr (Peter) Slatala
42b715adb7 Add visibility to ana config proto
Downstream projects need to be able to configure ANA without hacking or redefining protos.

Bug: webrtc:9719
Change-Id: Idd80471066ff41a9265adbdb738cc98cc97b2e6e
Reviewed-on: https://webrtc-review.googlesource.com/c/110765
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25629}
2018-11-13 20:49:29 +00:00
Karl Wiberg
105edcaeaf Remove some unused RentACodec static methods
We want to get rid of the whole thing, really, but these two were
easy.

Bug: webrtc:8396
Change-Id: I9292bf077caca171c9f5fe63695b6333cf22547d
Reviewed-on: https://webrtc-review.googlesource.com/c/104763
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25618}
2018-11-13 12:03:37 +00:00
Karl Wiberg
49c33ce528 AudioCodingModule: Remove support for creating encoders
After this CL, all audio encoders have to be injected by the caller.
This means that there is no special "built-in" set of codecs, and
users won't have to pay the binary size and security costs of codecs
they aren't using.

Bug: webrtc:8396
Change-Id: Idb0959ce395940c8bb3bbb49256cdcd84fc87bb6
Reviewed-on: https://webrtc-review.googlesource.com/c/103821
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25600}
2018-11-12 14:02:11 +00:00
Yves Gerey
09102a02cf Revert "Roll "Enable SSE, SSE2, and run-time detected SSE4.1 for libopus.""
This reverts commit 466620b326.

Reason for revert: Break downstream clients which are still expecting the previous references for NetEqDecodingTest.TestOpusBitExactness.

Original change's description:
> Roll "Enable SSE, SSE2, and run-time detected SSE4.1 for libopus."
> 
> We manually roll third_party since we need to update impacted tests,
> namely bit-exact comparison of expected neteq_opus results.
> They have changed due to SSE optimization enabled here:
> 85d355e530
> 
> For consistency sake roll_deps has been invoked:
> 
> Roll chromium_revision db720b4ab9..ae94013397 (606025:606579)
> 
> Change log: db720b4ab9..ae94013397
> Full diff: db720b4ab9..ae94013397
> 
> Changed dependencies
> * src/base: fee916f36b..f428263956
> * src/build: 02b0a894b0..3f61809684
> * src/ios: 95aadfb43f..fb48cd850c
> * src/testing: 03b25bebb5..f6a2832441
> * src/third_party: 360db5b8aa..8209b47661
> * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/dd412c428a..384d0eaf19
> * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2e722b007d..f04a3a61ad
> * src/third_party/depot_tools: 4d2d5b4bbe..edcefdcf7d
> * src/third_party/freetype/src: f56830ed40..fb0d66d04c
> * src/tools: a8e76f0ca5..f8ace9b670
> DEPS diff: db720b4ab9..ae94013397/DEPS
> 
> Clang version changed 344066:346388
> Details: db720b4ab9..ae94013397/tools/clang/scripts/update.py
> 
> Bug: webrtc:9530
> Change-Id: I8a016c5834c4f05fc17e3a84a5e139eeaeaee510
> Reviewed-on: https://webrtc-review.googlesource.com/c/110040
> Commit-Queue: Yves Gerey <yvesg@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25588}

TBR=phoglund@webrtc.org,ivoc@webrtc.org,yvesg@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:9530
Change-Id: I01549abdcfbcad70881ff9aeee913df68d0f0052
Reviewed-on: https://webrtc-review.googlesource.com/c/110602
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#25591}
2018-11-12 09:55:10 +00:00
Elad Alon
0b1b5c1b2a Hide RtcEvent members behind accessors
Bug: webrtc:8111
Change-Id: I3d350a6e159330aed7362162006860ac86ed7c32
Reviewed-on: https://webrtc-review.googlesource.com/c/109881
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25590}
2018-11-10 23:34:07 +00:00
Yves Gerey
466620b326 Roll "Enable SSE, SSE2, and run-time detected SSE4.1 for libopus."
We manually roll third_party since we need to update impacted tests,
namely bit-exact comparison of expected neteq_opus results.
They have changed due to SSE optimization enabled here:
85d355e530

For consistency sake roll_deps has been invoked:

Roll chromium_revision db720b4ab9..ae94013397 (606025:606579)

Change log: db720b4ab9..ae94013397
Full diff: db720b4ab9..ae94013397

Changed dependencies
* src/base: fee916f36b..f428263956
* src/build: 02b0a894b0..3f61809684
* src/ios: 95aadfb43f..fb48cd850c
* src/testing: 03b25bebb5..f6a2832441
* src/third_party: 360db5b8aa..8209b47661
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/dd412c428a..384d0eaf19
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2e722b007d..f04a3a61ad
* src/third_party/depot_tools: 4d2d5b4bbe..edcefdcf7d
* src/third_party/freetype/src: f56830ed40..fb0d66d04c
* src/tools: a8e76f0ca5..f8ace9b670
DEPS diff: db720b4ab9..ae94013397/DEPS

Clang version changed 344066:346388
Details: db720b4ab9..ae94013397/tools/clang/scripts/update.py

Bug: webrtc:9530
Change-Id: I8a016c5834c4f05fc17e3a84a5e139eeaeaee510
Reviewed-on: https://webrtc-review.googlesource.com/c/110040
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25588}
2018-11-09 22:30:47 +00:00
Sam Zackrisson
c496d58882 Add flag for fast jitter buffer playout in neteq simulation
It is currently not possible to run e.g. neteq_rtpplay in the fast
accelerate mode.

Bug: None
Change-Id: I5e0ce3fae2ad5585fe9fb545109bb0c9a87fd201
Reviewed-on: https://webrtc-review.googlesource.com/c/110162
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25561}
2018-11-08 14:32:48 +00:00
Sam Zackrisson
f0e7440a35 Add missing conditional defines to neteq test and tools targets
The .cc source files listed below #ifdef for WEBRTC_CODEC_OPUS and
WEBRTC_CODEC_ILBC but the build files don't include the defines.

modules/audio_coding/neteq/tools/neteq_test.cc
modules/audio_coding/neteq/tools/neteq_test_factory.cc

Bug: None
Change-Id: I6065021f68e58d0e5663acd006a9865bf265adc0
Reviewed-on: https://webrtc-review.googlesource.com/c/109925
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25555}
2018-11-08 11:25:10 +00:00
Niels Möller
c572ff3c71 Add default constructor for rtc::Event
Bug: webrtc:9962
Change-Id: Icaa91e657e6881fcb1553f354c07866109a0ea68
Reviewed-on: https://webrtc-review.googlesource.com/c/109500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25535}
2018-11-07 08:57:50 +00:00
tzik
254d3db59a Add missing #include to absl/memory/memory.h from audio_encoder_cng.cc
absl::make_unique is used in this file without absl/memory/memory.h
#include, that causes a build error on C++17 build of Chromium.

Bug: chromium:752720
Change-Id: I78fe9f76a6ea670a4250b4cf25c3c02cf4c4beb6
Reviewed-on: https://webrtc-review.googlesource.com/c/109540
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25514}
2018-11-06 10:57:47 +00:00
Henrik Lundin
007065522a Removing ancient and unused test scripts and data files
None of these scripts or files have been used in a very long time. They
are removed for the same reason, and since the data files add to the
weight of the resources folder.

Bug: webrtc:5289
Change-Id: Ia14a46aed180f286fa881fe5f60da6973a5fe027
Reviewed-on: https://webrtc-review.googlesource.com/c/109022
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25502}
2018-11-05 16:08:46 +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
Yves Gerey
69807e8871 Depend directly on destination targets.
Makes 'gn check' happy.
Followup to https://webrtc-review.googlesource.com/c/src/+/106820

Bug: webrtc:5876, webrtc:9855
Change-Id: I33fa2c31ba26dc10c9a9c17da0ffed255c1f4d5b
Reviewed-on: https://webrtc-review.googlesource.com/c/108760
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25447}
2018-10-31 10:21:40 +00:00
Yves Gerey
21cddffd99 Harmonize paths to dependent targets.
This CL consistently use:
 * relative paths for WebRTC dependent targets (test_support)
 * absolute paths for shared dependent targets (abseil)
This is a necessary (but insufficient) step to build WebRTC tests
from Chromium tree (rtc_include_tests=true), since test/ doesn't
sit anymore in the top level directory.

We also make sure that target declarations and uses are
consistent in regard to build_with_chromium flag.

Bug: webrtc:9943
Bug: webrtc:9855
Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b
Reviewed-on: https://webrtc-review.googlesource.com/c/108720
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25445}
2018-10-31 10:04:59 +00:00
Niels Möller
2c16cc61c2 Replace some usage of EventWrapper with rtc::Event.
Bug: webrtc:3380
Change-Id: Id33b19bf107273e6f838aa633784db73d02ae2c2
Reviewed-on: https://webrtc-review.googlesource.com/c/107888
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25407}
2018-10-29 09:37:24 +00:00
Yves Gerey
988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00
Niels Möller
2edab4c026 Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase.
Bug: webrtc:5876
Change-Id: Ica2d47ca45b8ef01a548d8dbe31dbed740a0ebda
Reviewed-on: https://webrtc-review.googlesource.com/c/106820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25306}
2018-10-23 09:24:15 +00:00
Mirko Bonadei
2dfa998be2 Reland "Prefix flag macros with WEBRTC_."
This is a reland of 5ccdc1331f

Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
>   for y in bool int float string FLAG; do
>     git grep -l "\b$x\_$y\b" | \
>     xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
>   done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}

TBR=kwiberg@webrtc.org

Bug: webrtc:9884
Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25277}
2018-10-19 15:06:43 +00:00
Mirko Bonadei
c538fc77b0 Revert "Prefix flag macros with WEBRTC_."
This reverts commit 5ccdc1331f.

Reason for revert: Breaks downstream project.

Original change's description:
> Prefix flag macros with WEBRTC_.
> 
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
> 
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
> 
> This CL adds the 'WEBRTC_' prefix to them.
> 
> Generated with:
> 
> for x in DECLARE DEFINE; do
>   for y in bool int float string FLAG; do
>     git grep -l "\b$x\_$y\b" | \
>     xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
>   done
> done
> git cl format
> 
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}

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

Change-Id: Ia79cd6066ecfd1511c34f1b30fd423e560ed6854
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9884
Reviewed-on: https://webrtc-review.googlesource.com/c/107160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25276}
2018-10-19 15:04:13 +00:00
Mirko Bonadei
5ccdc1331f Prefix flag macros with WEBRTC_.
Macros defined in rtc_base/flags.h are intended to be used to define
flags in WebRTC's binaries (e.g. tests).

They are currently not prefixed and this could cause problems with
downstream clients since these names are quite common.

This CL adds the 'WEBRTC_' prefix to them.

Generated with:

for x in DECLARE DEFINE; do
  for y in bool int float string FLAG; do
    git grep -l "\b$x\_$y\b" | \
    xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
  done
done
git cl format

Bug: webrtc:9884
Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
Reviewed-on: https://webrtc-review.googlesource.com/c/106682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25270}
2018-10-19 10:55:20 +00:00
Sebastian Jansson
b9972fa37b Adds AudioNetworkAdaptation support to Scenario tests.
Bug: webrtc:9718
Change-Id: I6cb976df5767797fec670134d29e030ec0f9d3a2
Reviewed-on: https://webrtc-review.googlesource.com/c/106340
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25236}
2018-10-17 15:42:58 +00:00
Minyue Li
34d990fef9 Adding NetEq buffer full metric to UMA.
BUG: webrtc:9882
Change-Id: Idbcbbbd99855b2251fbb66629efeab4f2d1f6498
Reviewed-on: https://webrtc-review.googlesource.com/c/106400
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25230}
2018-10-17 12:54:19 +00:00
Ivo Creusen
ed04912ccd Stop simulations when a LOG_END event is reached.
When a LOG_END event is reached, it makes no sense to continue simulating NetEq.

Bug: webrtc:9667
Change-Id: Ie4f6811cdec0d0632f6e7906059e0e74e9f10438
Reviewed-on: https://webrtc-review.googlesource.com/c/105643
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25176}
2018-10-15 16:06:40 +00:00
Ivo Creusen
d8a52b3ff4 Make ivoc owner of audio_coding.
Bug: None
Change-Id: I9e20031cd292b3459d5bead1a5763af9af18a325
Reviewed-on: https://webrtc-review.googlesource.com/c/106021
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25174}
2018-10-15 15:08:28 +00:00
Artem Titov
40a7a35eaa Extract functionality of test_main into separate library.
Extract functionality of test_main into separate library to be able to
reuse it if another main will be required.

Bug: webrtc:5996
Change-Id: I2925b4240bd0e4fb884b43bb16667ca2d6216bbd
Reviewed-on: https://webrtc-review.googlesource.com/c/105921
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25172}
2018-10-15 14:13:06 +00:00
Ivo Creusen
d2d2ecb4a8 Add command-line flag for setting the max number of packets in the buffer.
There is currently no way to set this for simulations in neteq_rtpplay.

Bug: webrtc:9667
Change-Id: I34f34565538bd3c378cdb9d355f5173c3517d59a
Reviewed-on: https://webrtc-review.googlesource.com/c/105982
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25171}
2018-10-15 14:10:24 +00:00
Jakob Ivarsson
83bd37cda4 Add field trials for configuring Opus encoder packet loss rate.
Add options to:
1. Bypass optimization (use reported packet loss).
2. Set a maximum value.
3. Set a coefficient.

Bug: webrtc:9866
Change-Id: I3fef43e5186a4f0f50fda3506e445860518cfbd7
Reviewed-on: https://webrtc-review.googlesource.com/c/105304
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25161}
2018-10-15 08:59:43 +00:00
Bjorn Terelius
5350d1cafd RtcEventLogSource no longer uses deprecated parsing functions.
Also remove header extension map from NetEqEventLogInput and RtcEventLogSource.

Bug: webrtc:8111
Change-Id: Ic9be7b03e32ab8aa12284596e21e53b6763f483a
Reviewed-on: https://webrtc-review.googlesource.com/c/102622
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25122}
2018-10-11 16:13:17 +00:00
Jakob Ivarsson
88b68ace17 Create field trial for setting a minimum value for Opus encoder packet loss rate
Bug: webrtc:9848
Change-Id: I0663ee3af7729a220de7aff08cd74545e1a7409a
Reviewed-on: https://webrtc-review.googlesource.com/c/104800
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25081}
2018-10-10 08:52:56 +00:00
Patrik Höglund
7730193a49 Remove SetExecutablePath, simplify ResourcePath
SetExecutablePath isn't used anymore.

Nobody was using the fancy select-per-platform functionality, and the
documentation was wrong anyway. In the cases somebody needed an
override per platform, they were using defines in their own test
instead. I think that is more verbose but more predictable and easy
to understand (see how it's done in audio_processing_unittest.cc
when loading output_data_mac, for instance).

Bug: webrtc:9792
Change-Id: I7289bf5883fe43852638922d7c7583eae0c08601
Reviewed-on: https://webrtc-review.googlesource.com/c/104482
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25062}
2018-10-09 14:01:16 +00:00
Niels Möller
433eafe1f5 Delete unused includes of assert.h
Bug: None
Change-Id: Iadc531710dca0ba34a00ac363bfe0784355bb6f9
Reviewed-on: https://webrtc-review.googlesource.com/c/103501
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24995}
2018-10-04 14:01:44 +00:00
Niels Möller
eddd3665a2 Delete unused method AudioCodingModuleImpl::SetOpusApplication.
Bug: None
Change-Id: I8fc1b4b9a4521444867c8b34ee54187c86dd6027
Reviewed-on: https://webrtc-review.googlesource.com/c/102040
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24993}
2018-10-04 13:46:31 +00:00
Ivo Creusen
dc6d5533e1 Add more NetEq information to NetEqState.
Some important NetEq information was not available in NetEqState, which
meant it was not available on the API. This CL adds additional
information.

Bug: webrtc:9667
Change-Id: I702707c7d60472f488047d48fb286f839c5608dc
Reviewed-on: https://webrtc-review.googlesource.com/c/102300
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24985}
2018-10-04 11:50:29 +00:00
Mirko Bonadei
8ca5c5216d Temporarily increase visibility of publicly used build targets.
Bug: webrtc:9808
Change-Id: I4ad2402dc288766732a2d81a289f717deec56629
Reviewed-on: https://webrtc-review.googlesource.com/c/103460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24981}
2018-10-04 11:29:21 +00:00
Karl Wiberg
c2c4d042ae AudioCodingModuleTest.TestRedFec: Don't let the ACM create audio encoders
It will soon lose the ability to do so.

Bug: webrtc:8396
Change-Id: If120afa37325c00ae2c3e9a9bd75bf89c8897f8c
Reviewed-on: https://webrtc-review.googlesource.com/c/103441
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24979}
2018-10-04 11:20:57 +00:00
Pablo Barrera González
bc2959072d NetEq: Fix an UBSan error
UBSan will trigger when time_stretched_samples overflows using a
big number. This change avoids this problem by storing the
intermediate result into a int64_t.

Bug: chromium:886904
Change-Id: Id09dc4b468f841f03b523d5f21763f610b163a42
Reviewed-on: https://webrtc-review.googlesource.com/c/103123
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Pablo Barrera González <barrerap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24977}
2018-10-04 10:51:08 +00:00
Mirko Bonadei
311c13b3c2 Remove noop system_wrappers_default build target.
After the removal of field_trial_default, metrics_default and
runtime_enabled_features_default, this build target doesn't build
anything and can be safely removed.

Bug: webrtc:9631
Change-Id: Iee1111e065ffefe0b4b9a695ee67a594e6d82caa
Reviewed-on: https://webrtc-review.googlesource.com/c/103702
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24976}
2018-10-04 10:25:37 +00:00
Minyue Li
002fbb8c7d Adding field trial to force target level percentile in NetEQ.
Bug: webrtc:9822
Change-Id: I636f75de10851729825311ee5783e836f3b583cd
Reviewed-on: https://webrtc-review.googlesource.com/c/101220
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24975}
2018-10-04 10:00:54 +00:00
Minyue Li
7f6417f480 Restricting NetEq postpone decoding after expand.
Bug: webrtc:9289
Change-Id: I923f304e6c12423fe5323c62484a27346033b19a
Reviewed-on: https://webrtc-review.googlesource.com/c/98320
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24966}
2018-10-04 08:01:09 +00:00
Karl Wiberg
5cc8e14586 audio_coding_module_unittest: Don't rely on the ACM to create encoders
It will soon lose the ability to do so.

Bug: webrtc:8396
Change-Id: I1d7c62fbc2585233cf1656fdcc4bb5380c2f41a5
Reviewed-on: https://webrtc-review.googlesource.com/c/100980
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24947}
2018-10-03 09:47:10 +00:00
Karl Wiberg
895ce82cab VAD/DTX tests: Don't let the ACM create audio encoders
It will soon lose the ability to do so.

Bug: webrtc:8396
Change-Id: I06dce417bba855b57130bd1a052988b2f235dcbd
Reviewed-on: https://webrtc-review.googlesource.com/102882
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24921}
2018-10-02 08:19:32 +00:00
Karl Wiberg
3ff52ffa22 Remove the useless ACMTest base class
Bug: webrtc:8396
Change-Id: I021a2429910b21ffe4829e0ed51b9290bc715c0c
Reviewed-on: https://webrtc-review.googlesource.com/102884
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24907}
2018-10-01 12:01:44 +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
Karl Wiberg
91957c1540 AudioCodingModuleTest.TwoWayCommunication: Don't let the ACM create encoders
It will soon lose the ability to do so.

Bug: webrtc:8396
Change-Id: I7d8e1549c44628fc9bdf2480468a0f1d3ae812f2
Reviewed-on: https://webrtc-review.googlesource.com/102062
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24853}
2018-09-27 00:27:26 +00:00
Karl Wiberg
3a6b6bda17 AudioCodingModuleTest.TwoWayCommunication: Remove non-automatic mode
The tests only use the automatic mode, and I'd rather not maintain
(and test!) the rest.

Bug: webrtc:8396
Change-Id: I4cd1096e088d2ea8807a605b8448bd44ff9e88ed
Reviewed-on: https://webrtc-review.googlesource.com/102060
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24852}
2018-09-27 00:18:46 +00:00
Karl Wiberg
377a231c0d acm_receiver_unittest: Don't rely on the ACM to create encoders
It will soon lose the ability to do so.

Bug: webrtc:8396
Change-Id: Ifca101fce0c349dba8c402ab2b6ad614061a88f6
Reviewed-on: https://webrtc-review.googlesource.com/101281
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24836}
2018-09-25 17:16:56 +00:00
Karl Wiberg
bf7a0463da AudioCodingModuleTest.TestIsac: Don't rely on the ACM to create encoders
It will soon lose the ability to do so.

Bug: webrtc:8396
Change-Id: Id413204e53afec28495dff0873f027a56caed80f
Reviewed-on: https://webrtc-review.googlesource.com/101861
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24834}
2018-09-25 16:54:36 +00:00
Karl Wiberg
9a60e9a5b0 Remove the delay_test binary
It hasn't been changed in any meaningful way since 2013, the same year
it was created.

Bug: webrtc:8396
Change-Id: I5633188134f71f24311fbd3098d046632fc4ee3a
Reviewed-on: https://webrtc-review.googlesource.com/101563
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24816}
2018-09-25 08:25:56 +00:00
Karl Wiberg
d363db1907 TestStereo: Don't rely on the ACM to create encoders
It will soon lose the ability to do so.

Also, the ACM no longer creates comfort noise encoders for us, so
don't bother testing that.

Bug: webrtc:8396
Change-Id: I24a12e26bef142f9f8e7532b764f28572e0c6ace
Reviewed-on: https://webrtc-review.googlesource.com/101640
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24803}
2018-09-24 15:11:05 +00:00
Karl Wiberg
36b37dce8f AudioCodingModuleTest.TestStereo: Delete write-only variables
Bug: webrtc:8396
Change-Id: I96c744c39ed15a2e20a45b120db9304dff486b76
Reviewed-on: https://webrtc-review.googlesource.com/101542
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24792}
2018-09-24 10:46:36 +00:00
Jonas Olsson
84df1c724e Make fewer copies when using StringBuilder.
Replace calls to .str() which copies with .Release which moves in cases where that's safe.

This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"

Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
2018-09-24 09:39:19 +00:00
Danil Chapovalov
db1285676b Cleanup modules_common_types
Uninline RTPFragmentaion functions
fix RTPFragmentation move constructor and assign operators (was recursive for win)
replace assert with rtc::dchecked_cast
Remove unused includes and dependencies.
Fix other targets that used those includes transitively instead of directly

Bug: None
Change-Id: I647cb1eda107dc7d87d25234095545bc2842fa40
Reviewed-on: https://webrtc-review.googlesource.com/100500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24759}
2018-09-18 08:08:33 +00:00
Henrik Lundin
00eb12a20c Let NetEq use the PLC output from a decoder
This change enables NetEq to use the packet concealment audio (aka
PLC) produced by a decoder. The change also includes a new API to the
AudioDecoder interface, which lets the decoder implementation generate
and deliver concealment audio.

Bug: webrtc:9180
Change-Id: Icaacebccf645d4694b0d2d6310f6f2c7132881c4
Reviewed-on: https://webrtc-review.googlesource.com/96340
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24738}
2018-09-14 07:05:20 +00:00
Ivo Creusen
d1c2f78bfe Implement new stats interface on NetEq to monitor the operations and internal state.
Currently we use the NetworkStatistics to monitor these metrics, but because these get reset on every call, this makes it impossible to use them for other purposes.

Bug: webrtc:9667
Change-Id: If648085f04d2d58aae263cff5b9491bcad373a96
Reviewed-on: https://webrtc-review.googlesource.com/99740
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24727}
2018-09-13 14:03:47 +00:00
Minyue Li
1a80018a3c Avoid wrong parsing of padding length and its use in NetEq simulation.
Bug: b/113648474, webrtc:9730
Change-Id: Ieff7ab8697f5c8742548897a9b452a20b0bd2e7c
Reviewed-on: https://webrtc-review.googlesource.com/98461
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24703}
2018-09-12 11:23:03 +00:00
Ivo Creusen
f81b0f11a6 Move code for setting field trials from NetEqTestFactory to the main function in neteq_rtpplay.
It is problematic to set field trials more than once, so to avoid running into problems, this functionality has been placed in the main function of neteq_rtpplay.

Bug: webrtc:9667
Change-Id: Ib9b9990f30a1715b50889dbfc4d74787bcbe5dae
Reviewed-on: https://webrtc-review.googlesource.com/98541
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24673}
2018-09-11 09:27:11 +00:00
Mirko Bonadei
8b0aed1dd6 Fix no_global_constructors/no_exit_time_destructors in Neteq.
Bug: webrtc:9693
Change-Id: I0135e934c638ec391546928ba1e623d137b27b75
Reviewed-on: https://webrtc-review.googlesource.com/98600
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24668}
2018-09-11 06:39:14 +00:00
Henrik Lundin
9be7745509 NetEq tools: Fixing an issue with measuring the simulation time
The NetEqTest class was recently refactored. In the process, the
functionality for measuring the simulation time suffered a bug. This
CL fixes it.

Bug: webrtc:9667
Change-Id: I139e697ede21584ef77ae23cfa8e77f6dac65b51
Reviewed-on: https://webrtc-review.googlesource.com/98982
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24658}
2018-09-10 16:16:22 +00:00
Ivo Creusen
4384f53285 Add more useful information to NetEqState and implement action_times_ms
This CL adds more useful information to NetEqState, and implements setting action_times_ms, which can be used to get a better idea of what actually happened during a timestep.

Bug: webrtc:9667
Change-Id: I789a3e1ad852066fdf4e9b4c96b8fb6033dacb27
Reviewed-on: https://webrtc-review.googlesource.com/98163
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24643}
2018-09-10 09:10:53 +00:00
Niels Möller
fe3240aeae Reland "Delete class EventTimerWrapper."
This is a reland of a421775a6d

Original change's description:
> Delete class EventTimerWrapper.
>
> Only user, iSACTest, refactored to use a sleep instead.
>
> Bug: webrtc:3380
> Change-Id: I683a5a05349f75a17e5d2a02d4a20a9cf059a28f
> Reviewed-on: https://webrtc-review.googlesource.com/96802
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24541}

Tbr: henrik.lundin@webrtc.org
Bug: webrtc:3380
Change-Id: I541473b9c3ce2020f76d420598a7b10766f1d2a9
Reviewed-on: https://webrtc-review.googlesource.com/98481
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24620}
2018-09-07 09:54:55 +00:00
Jonas Olsson
366a50c4ef Remove simple stringstream usages.
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.

The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.

Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
2018-09-06 12:53:19 +00:00
Mirko Bonadei
96ede16a4e Enable -Wexit-time-destructors and -Wglobal-constructors.
This CL enables -Wexit-time-destructors and -Wglobal-constructors on
rtc_static_library and rtc_source_set build targets.

It also adds the possibility to suppress these warnings because
they trigger in a few places.

The long term goal is to avoid regressions on this and remove all the
suppressions.

Bug: webrtc:9693
Change-Id: I4c1ecc137ef9e87ec5e66981ce95d96fb082727c
Reviewed-on: https://webrtc-review.googlesource.com/98380
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24604}
2018-09-06 12:43:20 +00:00
Niels Moller
85e6e826a2 Revert "Delete class EventTimerWrapper."
This reverts commit a421775a6d.

Reason for revert: Depends on https://webrtc-review.googlesource.com/c/src/+/97320, which will be reverted due to breakage in video_engine_tests.

Original change's description:
> Delete class EventTimerWrapper.
> 
> Only user, iSACTest, refactored to use a sleep instead.
> 
> Bug: webrtc:3380
> Change-Id: I683a5a05349f75a17e5d2a02d4a20a9cf059a28f
> Reviewed-on: https://webrtc-review.googlesource.com/96802
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24541}

TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,nisse@webrtc.org

Change-Id: Iea92618c87cb4eb4595f22674528920171a9defb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:3380
Reviewed-on: https://webrtc-review.googlesource.com/97681
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24551}
2018-09-04 11:50:03 +00:00
Mirko Bonadei
01cd853d2e Add nogncheck for headers of codecs not used in Chromium.
iLBC [1] and Red [2] are not used in Chromium, this means that WebRTC
doeesn't add the GN dependency on them but the include checker
complains because when it parses the code it sees iLBC and Red headers
included (the GN checker doesn't run the c preprocessor).

[1] - https://cs.chromium.org/chromium/src/.gn?l=62-65&rcl=3f6c31f0fdba128d810c4e3e391fae3b1aca7e7c
[2] - https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_coding/audio_coding.gni?l=28-30&rcl=eb254b40b33380fcec43028dd89f3f6bab3d08a7

Bug: chromium:824831
Change-Id: I5059c1773fbe35f568c2fe3d8db3807f29973e7e
Reviewed-on: https://webrtc-review.googlesource.com/97620
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24546}
2018-09-04 09:45:56 +00:00
Niels Möller
a421775a6d Delete class EventTimerWrapper.
Only user, iSACTest, refactored to use a sleep instead.

Bug: webrtc:3380
Change-Id: I683a5a05349f75a17e5d2a02d4a20a9cf059a28f
Reviewed-on: https://webrtc-review.googlesource.com/96802
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24541}
2018-09-04 06:53:28 +00:00
Ivo Creusen
55de08e7ef Restructure neteq_rtpplay into a library with small executable wrapper.
Most of the code in neteq_rtpplay is moved into a factory class for
NetEqTest. The factory method takes the same argc and argv arguments as
neteq_rtpplay.
This CL also adds a small public API for neteq_test to allow easy
integration into external software.

Bug: webrtc:9667
Change-Id: I5241c1f51736cb6fbe47b0ad25f4bc83dabd727d
Reviewed-on: https://webrtc-review.googlesource.com/96100
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24531}
2018-09-03 10:42:40 +00:00
Alessio Bazzica
d4161a3c9d Moving LappedTransform, Blocker and AudioRingBuffer.
LappedTransform is only used in BandwidthAdaptationTest and therefore it
should not be anymore a visible target under common_audio.
This CL moves LappedTransform and other two classes it depends on (and which
are not used elsewhere) to modules/audio_coding/codecs/opus/test.

Bug: webrtc:9577, webrtc:5298
Change-Id: I1aa8052c2df2b2b150c279c0c9b1001474aed47a
Reviewed-on: https://webrtc-review.googlesource.com/96440
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24509}
2018-08-31 15:27:50 +00:00
Niels Möller
d941c09bc0 Delete unimplemented methods from the NetEq interface.
Bug: None
Change-Id: I51949a096c445813acc6649676e32c575732ef40
Reviewed-on: https://webrtc-review.googlesource.com/95643
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24469}
2018-08-28 15:48:26 +00:00
Niels Möller
764c14c87d Delete unused AudioCodingModule methods.
Methods deleted: IsCodecValid (static), QueryEncoder, SendFrequency.

Bug: None
Change-Id: Id63ea7cdc364583e896d3301d04fa9caae1e4d94
Reviewed-on: https://webrtc-review.googlesource.com/95486
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@{#24440}
2018-08-27 10:00:59 +00:00
Niels Möller
18f1adc0da Delete AudioCodingModule::LeastRequiredDelayMs and related NetEq code.
Bug: None
Change-Id: I2f68502d19415899b3694f7bf5da523da831b223
Reviewed-on: https://webrtc-review.googlesource.com/95640
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@{#24439}
2018-08-27 09:58:19 +00:00
Niels Möller
ec93075c00 Delete deprecated methods on AudioCodingModule
Bug: None
Change-Id: I05f1ab6cdd6ac52972835af7ea94aacf5f64d673
Reviewed-on: https://webrtc-review.googlesource.com/95482
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24388}
2018-08-22 13:26:17 +00:00