Commit graph

155 commits

Author SHA1 Message Date
Alex Loiko
e5b94160b5 Decoder for multistream Opus.
See https://webrtc-review.googlesource.com/c/src/+/121764 for the
overall vision.

This CL adds a multistream Opus decoder. It's a new code-path to not
interfere with the standard Opus decoder. We introduce new SDP syntax,
which uses terminology of RFC 7845. We also set up the decoder side to
parse it. The encoder part will come in a later CL.

E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"

Bug: webrtc:8649
Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27493}
2019-04-08 16:15:37 +00:00
Artem Titov
741daaf039 Move rtc::FunctionView to the public API
Bug: webrtc:10138
Change-Id: Icc25a2a277a9608701aaddd546882366739991ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127898
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27227}
2019-03-21 15:23:05 +00:00
Artem Titov
94b57c044e Cleanup BUILD.gn files from imports like foo:foo
Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn
files.

Done with Sublime regex replace.
Find: \b([-a-zA-Z0-9_]+):+\1\b
In: *.gn
Replace with: \1

Bug: None
Change-Id: I40aba1b14face687a595b852ffe443cb20197611
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:05:28 +00:00
Niels Möller
ef1052a134 Reland "Move api/rtp_headers.h to its own build target."
This is a reland of a67050debc

Original change's description:
> Move api/rtp_headers.h to its own build target.
>
> Reduces dependencies on the libjingle_peerconnection_api target from
> lower-level code.
>
> Bug: None
> Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27213}

Bug: None
Tbr: kwiberg@webrtc.org
Change-Id: If15b05957e50bb8f18a33c2ed1321e672311b626
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127895
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27216}
2019-03-21 09:17:07 +00:00
Steve Anton
2baef3509f Revert "Move api/rtp_headers.h to its own build target."
This reverts commit a67050debc.

Reason for revert: breaks downstream projects

Original change's description:
> Move api/rtp_headers.h to its own build target.
> 
> Reduces dependencies on the libjingle_peerconnection_api target from
> lower-level code.
> 
> Bug: None
> Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27213}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org

Change-Id: I8cccaa8be1700ca8db141db7252eb6ce588ba2e0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128645
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27215}
2019-03-20 16:47:30 +00:00
Niels Möller
a67050debc Move api/rtp_headers.h to its own build target.
Reduces dependencies on the libjingle_peerconnection_api target from
lower-level code.

Bug: None
Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27213}
2019-03-20 16:00:49 +00:00
Niels Möller
c936cb6a86 Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h
Bug: webrtc:5876
Change-Id: I0c92f9410fcf0832bfa321229b3437134255dba6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128085
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27190}
2019-03-19 16:59:27 +00:00
Mirko Bonadei
7edc49cb31 Mark neteq_rtpplay as publicly visible.
Bug: None
Change-Id: I051c7c23851ab15345c8e0f0322458d4f9a7e187
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128123
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27162}
2019-03-18 14:51:50 +00:00
Alessio Bazzica
5ad789ceff Reland "NetEQ RTP Play: Optionally write output audio file"
This reverts commit c4b391a257.

Reason for revert: issue fixed

Original change's description:
> Revert "NetEQ RTP Play: Optionally write output audio file"
>
> This reverts commit 6330818ec8.
>
> Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots.
>
> Original change's description:
> > NetEQ RTP Play: Optionally write output audio file
> >
> > This CL makes the output audio file optional to more
> > quickly run neteq_rtpplay when no audio output is needed.
> > The CL also includes necessary adaptations because of pre-existing
> > dependencies (e.g., the output audio file name is used to create
> > the plotting script file names).
> >
> > The command line arguments are retro-compatible - i.e., same behavior
> > when specifying the output audio file and the new flag
> > --output_files_base_name is not used.
> >
> > This CL also includes a test script with which the retro-compatibility
> > has been verified.
> >
> > Bug: webrtc:10337
> > Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27067}
>
> TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org
>
> Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10337
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27078}

TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org

Change-Id: Ia7061f7c2d69db61638ad612e82cd429eb49d539
Bug: webrtc:10337
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127540
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27106}
2019-03-13 15:33:29 +00:00
Ivo Creusen
c4b391a257 Revert "NetEQ RTP Play: Optionally write output audio file"
This reverts commit 6330818ec8.

Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots.

Original change's description:
> NetEQ RTP Play: Optionally write output audio file
> 
> This CL makes the output audio file optional to more
> quickly run neteq_rtpplay when no audio output is needed.
> The CL also includes necessary adaptations because of pre-existing
> dependencies (e.g., the output audio file name is used to create
> the plotting script file names).
> 
> The command line arguments are retro-compatible - i.e., same behavior
> when specifying the output audio file and the new flag
> --output_files_base_name is not used.
> 
> This CL also includes a test script with which the retro-compatibility
> has been verified.
> 
> Bug: webrtc:10337
> Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27067}

TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org

Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10337
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27078}
2019-03-12 14:28:23 +00:00
Alessio Bazzica
6330818ec8 NetEQ RTP Play: Optionally write output audio file
This CL makes the output audio file optional to more
quickly run neteq_rtpplay when no audio output is needed.
The CL also includes necessary adaptations because of pre-existing
dependencies (e.g., the output audio file name is used to create
the plotting script file names).

The command line arguments are retro-compatible - i.e., same behavior
when specifying the output audio file and the new flag
--output_files_base_name is not used.

This CL also includes a test script with which the retro-compatibility
has been verified.

Bug: webrtc:10337
Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27067}
2019-03-12 08:37:57 +00:00
Takuto Ikuta
745cfb9997 use link_deps in ana_debug_dump_proto
I will deprecate deps in proto_library for improved build throughput.
We can use link_deps here instead.

Bug: chromium:938011
Change-Id: Iafa83000c3f7f9ffdc0c376a2297b4a9380b7594
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125820
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Takuto Ikuta <tikuta@google.com>
Cr-Commit-Position: refs/heads/master@{#26989}
2019-03-06 12:12:10 +00:00
Mirko Bonadei
fc52b912a3 Implicitly suppress //build/config/clang:find_bad_constructs.
Since there is no way to enable/disable these diagnostics at runtime,
this CL moves the suppression into the rtc_* templates in order to
remove the need to explicitly add the snippet of code needed to
suppress it (currently copy/pasted in 144 locations).

The diagnostic that causes the most problems is the one about "complex
class/struct explicit ctor/dtor" [1] because WebRTC doesn't find
it useful enough.

Other diagnostics are good (for example the one that warns about
using "virtual" instead of "override", but that will be covered by
this clang-tidy check [2]) while others are Chromium related so
they have never triggered.

[1] - https://cs.chromium.org/chromium/src/tools/clang/plugins/FindBadConstructsConsumer.cpp?l=147-167&rcl=b4bebe1aa15dba7ca5fcc6456a81a55665327c3a
[2] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html

Bug: webrtc:163
Change-Id: Icbf27efa5b369100a31e6a32df1a0913729b3b34
Reviewed-on: https://webrtc-review.googlesource.com/c/125088
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26918}
2019-03-01 10:18:17 +00:00
Ivo Creusen
ba7886b076 Move command line flags out of NetEqTestFactory
This replaces the use of command-line flags with the use of a config
struct. This makes it easier for non command-line applications to use
the NetEqTestFactory to run simulations.

Bug: webrtc:10337
Change-Id: I24533bf206e70e12db9af8d9675769c1ff7c7d48
Reviewed-on: https://webrtc-review.googlesource.com/c/123600
Reviewed-by: Pablo Barrera González <barrerap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26887}
2019-02-28 10:01:25 +00:00
Kimmo Kinnunen
08f6a6c672 Import proto_library.gni when rtc_enable_protobuf is true
Import proto_library.gni when rtc_enable_protobuf is true instead of when
build_with_mozilla is false.

Makes it maybe easier to reason about the intention (e.g. intention is to not
compile any protobuf in, hence flag rtc_enable_protobuf)

The build file could not work if build_with_mozilla = true but
rtc_enable_protobuf = true.

Bug: webrtc:10338
Change-Id: I26e5983bd1519aa46c308b11796d518de5ef7597
Reviewed-on: https://webrtc-review.googlesource.com/c/123763
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26868}
2019-02-27 09:56:42 +00:00
Jakob Ivarsson
1eb3d7ea0f Refactor DelayManager into separate Histogram class and make it injectable for testing purposes.
Change-Id: I98aa3f992169e598fc1a3dd850400183395fe1fe
Bug: webrtc:10333
Reviewed-on: https://webrtc-review.googlesource.com/c/123445
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26797}
2019-02-21 15:45:31 +00:00
Mirko Bonadei
e45c688e67 Remove webrtc::ProtoString.
Bug: None
Change-Id: If99a977532eda41eada25f57ff0ff6fe17085986
Reviewed-on: https://webrtc-review.googlesource.com/c/122581
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26726}
2019-02-16 11:11:45 +00:00
Mirko Bonadei
d970807e0c Remove rtc_base/scoped_ref_ptr.h.
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.

More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.

Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:29:58 +00:00
Bjorn Terelius
5c2f1f053f Add some missing includes and dependencies.
In particular, time_utils.h is currently pulled in via rtc_event.h
This CL is in preparation of moving parts of the RTC event log to api/.

Bug: webrtc:10206
Change-Id: Idd35aa9404afded4d29b1296344996c45b8c2e91
Reviewed-on: https://webrtc-review.googlesource.com/c/117921
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26326}
2019-01-18 15:30:26 +00:00
Niels Möller
7289906437 Delete enum NetEqDecoder.
A trimmed down version is moved to legacy_encoded_audio_frame_unittest.cc
where it's used for test parameterization.

Bug: webrtc:10185
Change-Id: I9abda22f9806b831b6ca4b27d6bcc888285f50f2
Reviewed-on: https://webrtc-review.googlesource.com/c/116961
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26218}
2019-01-11 13:52:25 +00:00
Niels Möller
9c277dd1dd Delete NetEq::RegisterExternalDecoder.
Bug: webrtc:10080
Change-Id: Ie36b10af6ab22f498636e38f36bef11f28fc7f7e
Reviewed-on: https://webrtc-review.googlesource.com/c/112081
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26174}
2019-01-09 10:38:08 +00:00
Steve Anton
40d55331d7 Include absl/memory/memory.h if absl::make_unique is used
Tbr: kwiberg@webrtc.org
Bug: None
Change-Id: Iaf4533d2ce0e80b351a8a664ef8cf7ba0e5ec583
Reviewed-on: https://webrtc-review.googlesource.com/c/115746
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26168}
2019-01-08 20:08:32 +00:00
Niels Möller
d375f1c8d1 Refactor NetEqTestFactory to not use "external" decoders
Bug: webrtc:10080
Change-Id: Icfca98d6d91fc5139e678c1aa3de1e2c35abff5c
Reviewed-on: https://webrtc-review.googlesource.com/c/115240
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26090}
2018-12-21 15:05:03 +00:00
Niels Möller
3f651d80a0 Reland "Add AudioDecoderFactory to NetEqTest constructor."
This is a reland of daa970f33e

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

Tbr: kwiberg@webrtc.org
Bug: webrtc:8396, webrtc:10080
Change-Id: I598ce1cd41676b1992b0973b09476eeeb0e602d2
Reviewed-on: https://webrtc-review.googlesource.com/c/114940
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26058}
2018-12-19 15:08:47 +00:00
Fredrik Solenberg
41f3a43c74 Remove CodecInst pt.3
Finally remove CodecInst from common_types.h, including remaining code referencing it.

TBR=kwiberg

Bug: webrtc:7626
Change-Id: I5e6b949ae9093641e33972af8438d1126fc48556
Reviewed-on: https://webrtc-review.googlesource.com/c/114546
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26036}
2018-12-18 07:42:21 +00:00
Oleh Prypin
f7f753b320 Revert "Add AudioDecoderFactory to NetEqTest constructor."
This reverts commit daa970f33e.

Reason for revert: Speculative revert due to downstream breakage

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

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

No-Try: True
Bug: webrtc:8396, webrtc:10080
Change-Id: Ided750d8ed800d8a38f7cce8f72095d8ed1bc6cb
Reviewed-on: https://webrtc-review.googlesource.com/c/114552
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26030}
2018-12-17 15:16:30 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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