Commit Graph

211 Commits

Author SHA1 Message Date
f88aa97c52 Introduce RtpSequenceNumberMap
The video encoder should be informed of incoming LossNotification
RTCP messages. Since it is unaware of RTP sequence numbers,
or anything else RTP-related, the sequence numbers mentioned
in the RTCP message must first be mapped to timestamps,
since those are meaningful to the encoder.

This CL introduces RtpSequenceNumberMap, which maps RTP sequence
numbers to timestamps, while providing:
1. Capping the number of entries.
2. Wrap-around handling.

RtpSequenceNumberMap also remembers which packets were first
and/or last in the frame.

Later CLs will wire this up.

Bug: webrtc:10501
Change-Id: Ie0662cdb5706a3bcf63aa2934816a9df88439357
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130497
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27448}
2019-04-04 10:50:12 +00:00
9d8eaac4ee Delete unneeded direct includes of common_types.h
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:

api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/

There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.

Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
2019-04-01 07:18:13 +00:00
91c2606ca1 Use Abseil container algorithms in modules/rtp_rtcp/
Bug: None
Change-Id: Ica2e9795ec6195e044403f5ee25e476f6c47cf93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129600
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27361}
2019-03-29 16:47:33 +00:00
a7de698675 Add functions IsLegalMidName and IsLegalRsidName
This is a preparation for deleting the class StringRtpHeaderExtension.

Bug: webrtc:10440
Change-Id: I3480e58d96e67d10c4d78597c8ab7f01b63e37ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128761
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27228}
2019-03-21 16:10:31 +00:00
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
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
ef1052a134 Reland "Move api/rtp_headers.h to its own build target."
This is a reland of a67050debcb5a3461a452a7928d7aaea1562747e

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
2baef3509f Revert "Move api/rtp_headers.h to its own build target."
This reverts commit a67050debcb5a3461a452a7928d7aaea1562747e.

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
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
dd025d85b0 Avoid using global task queue factory in RtcpTransciever tests
by not using convenient rtc::TaskQueue constructor
and using TaskQueueForTest instead.

Bug: webrtc:10284
Change-Id: I882ce4d1404a1cb9ab8bf97fca6beac3f89bb41d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128604
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27212}
2019-03-20 15:20:20 +00:00
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
1aa7581701 Replace all usage of rtc::NewClosure with webrtc::ToQueuedTask
Bug: webrtc:10191
Change-Id: I795c8a6f281ccdf60031500a4fb5a411f2afdb70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125400
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26975}
2019-03-05 18:40:44 +00:00
e11b7d2e80 Replace field trials with WebRtcKeyValueConfig in RtpRtcpModule
Replaces use of field trials in RtpSender and RtpVideoSender with injectable WebRtcKeyValueConfig.
Implementation still defaults to field trials.

BUG: webrtc:10335
Change-Id: I5fc6d327ee5d011ccc41385734b38344df172627
Reviewed-on: https://webrtc-review.googlesource.com/c/123447
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26795}
2019-02-21 14:25:34 +00:00
e7b9e6b17d Move RtpSenderVideo tests to separate file.
Also refactor most of the RtpSender tests to not use the frame-level
method RTPSender::SendOutgoingData.

Bug: webrtc:7135
Change-Id: I9b0af6aa45e9b908d8197e48b143fede7e2804c7
Reviewed-on: https://webrtc-review.googlesource.com/c/121461
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26577}
2019-02-06 18:00:39 +00:00
a34d7766c5 Move RtpSenderAudioTest to its own file
Update RtpSenderAudioTest to call methods on RTPSenderAudio rather
than RTPSender, when possible. In particular, avoid
RTPSender::SendOutgoingData. Drop parameterization on the
WebRTC-SendSideBwe-WithOverhead field trial, since that appears
unrelated to these tests.

Also delete some unused parts of the RtpSender test.

Bug: webrtc:7135
Change-Id: I535bf48bb1720e2727f4a62fa3e49b2bb84394a0
Reviewed-on: https://webrtc-review.googlesource.com/c/120920
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26516}
2019-02-01 15:15:56 +00:00
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
f8e7ccb967 Create new RTCP feedback message - LossIndication
Create a new RTCP feedback message for reporting the loss and/or non-decodability of video frames, to be used by the upcoming injectable VideoFrameBufferController. The new feedback message should report:
1. The sequence number of the last decoded non-discardable video frame. (TBD: If a multi-packet frame, should it be the sequence number of the first, last, or any of the packets?)
2. The sequence number of the last received RTP packet in the stream.
3. A decodability flag, whose specific meaning depends on the last-received
   RTP sequence number. The decodability flag is true if and only if all of
   the frame's dependencies are known to be decodable, and the frame itself
   is not yet known to be unassemblable.
   * Clarification #1: In a multi-packet frame, the first packet's
     dependencies are known, but it is not yet known whether all parts
     of the current frame will be received.
   * Clarification #2: In a multi-packet frame, the dependencies would be
     unknown if the first packet was not received. Then, the packet will
     be known-unassemblable.

Bug: webrtc:10226
Change-Id: I1563c944477e3ed40235e82ab99a439414632aff
Reviewed-on: https://webrtc-review.googlesource.com/c/118931
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26387}
2019-01-24 12:21:00 +00:00
ecb6897ade Adds repeating task class.
This CL adds a single class to manage the use case of having a task
that repeats itself by a fixed or variable interval. It replaces the
repeating task previously locally defined for rtp transport controller
send as well as the cancelable periodic task. Furthermore, it is
introduced where one off repeating tasks were created before.

It provides the currently used functionality of the cancelable periodic
task, but not some of the unused features, such as allowing cancellation
of tasks before they are started and cancellation of a task after the
owning task queue has been destroyed.

Bug: webrtc:9883
Change-Id: Ifa7edee836c2a64fce16a7d0f682eb09c879eaca
Reviewed-on: https://webrtc-review.googlesource.com/c/116182
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26313}
2019-01-18 10:55:41 +00:00
f93eda1705 Move some video codec constants to separate file.
kMaxSimulcastStreams, kMaxSpatialLayers and kMaxTemporalStreams don't
really beling on VideoBitrateAllocation.
common_types.h is going away and it feels dubious to requrie include
of the full VideoEncoder api to use them. Therefore moving them into a
seprate file/target.

Also includes some remaining cleanup of includes.

Bug: webrtc:9271
Change-Id: I7ded3d97a9a835ac756159700774445a2b93a697
Reviewed-on: https://webrtc-review.googlesource.com/c/117305
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26299}
2019-01-17 15:29:53 +00:00
31d8b52075 Delete unneeded includes of rtc_base/stringutils.h.
Also delete corresponding dependencies on rtc_base:stringutils.

Bug: webrtc:6424
Change-Id: I2be5e021292eea2d788c76a63cc0e4f7cefd927d
Reviewed-on: https://webrtc-review.googlesource.com/c/114544
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26057}
2018-12-19 11:04:27 +00:00
f1ab9b9b3b Refactor creation of ColorSpace test data
Bug: webrtc:8651
Change-Id: I2ebb5fcdc260af19d04513ab5f3d76f81a3b4ca9
Reviewed-on: https://webrtc-review.googlesource.com/c/114282
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26012}
2018-12-14 10:15:10 +00:00
18f0c3c038 Add RegisterAudioSendPayload() method
In preparation of removing CodecInst.

Bug: webrtc:7626
Change-Id: I8955d17dbb3ec15177e505ae420376b542d48410
Reviewed-on: https://webrtc-review.googlesource.com/c/113306
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25919}
2018-12-06 12:44:53 +00:00
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
aa3c1cc927 Delete _strnicmp. Uses replaced with abseil functions.
The replacements are absl::EqualsIgnoreCase and
absl::StartsWithIgnoreCase. Also delete the alias
RtpUtility::StringCompare.

Bug: webrtc:6424
Change-Id: I4bed71540264450f85137ad0c2564125c5c6213f
Reviewed-on: https://webrtc-review.googlesource.com/c/109006
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25481}
2018-11-02 11:03:38 +00:00
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
6dd7f9120e Remove deprecated deregistration functions in RtcpTransceiver
These functions were deprecated in
https://webrtc-review.googlesource.com/c/src/+/107305

Bug: None
Change-Id: I3246f1e2f16be7591daec520e90c6b75071eb92d
Reviewed-on: https://webrtc-review.googlesource.com/c/107730
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25388}
2018-10-26 12:31:37 +00:00
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
98f5f6cdea In RtcpTransceiver functions with callback avoid relying on PostTaskAndReply
deprecated version guarantees (using PostTaskAndReply) callback task will run on the task queue,
and thus doesn't guarantee to run it if task queue is destroyed,

new callback versions instead guarantee callback will always run,
but may run off the task queue if task queue is destroyed.

Both keep guarantee observer callbacks will not run after on_destroyed/on_removed is called.

Bug: None
Change-Id: I61bf52127f3084c0186aa8bc89037bf9296801d8
Reviewed-on: https://webrtc-review.googlesource.com/c/107305
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25301}
2018-10-23 08:29:34 +00:00
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
23b2a25675 Remove unlimited retransmission for screenshare experiment code
Bug: webrtc:9659
Change-Id: I29d8f0d20b0faee5ec2e8e196581338770b1a74d
Reviewed-on: https://webrtc-review.googlesource.com/c/105001
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25103}
2018-10-11 07:53:47 +00:00
44b384d013 Delete support for VoIP metrics (RFC 3611 4.7)
Bug: None
Change-Id: I2f3cd622d3863fa88a9e1971894eced8eeb777e6
Reviewed-on: https://webrtc-review.googlesource.com/c/103805
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25007}
2018-10-05 10:07:57 +00:00
4a72ba99a7 Delete RtpReceiver and related code.
The RtpReceiver class is no longer used. Together with it, delete
RTPPayloadRegistry, RtpReceiverStrategy, and the tests under
modules/rtp_rtcp/test/testAPI/.

Bug: webrtc:8995
Change-Id: Ia9924d2f0f4315914a0dce6b7375ebb3601a6f96
Reviewed-on: https://webrtc-review.googlesource.com/c/103503
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24968}
2018-10-04 08:46:16 +00:00
7b1899224b Move RtpHeaderExtensionMap::GetTotalLengthInBytes into own file
Rename to better match what it does,
Adjust to support two-byte header extension

Bug: webrtc:7990
Change-Id: I2786d70e7cf9cd3d722f54fb1d07c9cfaafab947
Reviewed-on: https://webrtc-review.googlesource.com/103201
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24958}
2018-10-03 17:25:31 +00:00
5c25010c86 Set public visibility for rtp_rtcp and video_coding targets
Though discouraged, those folders are listed in native-api

NOTRY=True

Bug: webrtc:9808
Change-Id: I9407c8d69a0d75196cfa9435f5e459264c64e046
Reviewed-on: https://webrtc-review.googlesource.com/c/103364
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24953}
2018-10-03 13:30:52 +00:00
1f3206cca4 Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1
Add new method OnRtpPacket, but leave
ReceiveStatisticsImpl::IncomingPacket and most of the implementation
unchanged. Deleting the old method and converting implementation from
RTPHeader to RtpPacketreceived is planned for a followup, after
downstream code is updated.

Bug: webrtc:7135, webrtc:8016
Change-Id: I697ec12804618859f8d69415622d1b957e1d0847
Reviewed-on: https://webrtc-review.googlesource.com/100104
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24889}
2018-09-28 12:00:28 +00:00
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
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
f0076d31f8 Add RtcpTransceiver::Stop to allow non-blocking destruction
As downside it disallows to destroy RtcpTransceiver on the TaskQueue
without prio call to the Stop function

BUG: webrtc:8239
Change-Id: I236b9aff7a0746044dd764c619174cc5ac03d27f
Reviewed-on: https://webrtc-review.googlesource.com/98120
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24587}
2018-09-05 15:42:24 +00:00
1a6a4f3d62 Replace WeakPtr with CancelablePeriodicTask in RtcpTransceiverImpl
This allow to destroy the RtcpTransceiverImpl off the task queue
with assumption it is destroyed after task queue.
i.e. it allows to post destruction of RtcpTransceiverImpl to the TaskQueue without waiting.

BUG: webrtc:8239
Change-Id: I4bea7a6d2edc97061ebd00f2f275c1ab827bc3c5
Reviewed-on: https://webrtc-review.googlesource.com/97160
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24574}
2018-09-05 11:06:16 +00:00
376e1147e6 in RtpPacketizerVp8 factor out payload splitter function
so that it can be shared between different packetizers
and thus easier to extend

Bug: webrtc:9680
Change-Id: Ie5e904ad27afb8dd2ed35ef9e009f7f408017b2f
Reviewed-on: https://webrtc-review.googlesource.com/97661
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24555}
2018-09-04 15:28:05 +00:00
8a3c166fff Cleanup RtpPacketizerVP8 tests
Remove partition support for test helper and from tests.
Merge Init function into constructor
Replace extra macroses in favor of Bit helper function
Replace extra members in favor of local variables
Remove fixture

Bug: None
Change-Id: Ibf1600dda9f59abe5afd2bbe40c3e232a2d269ea
Reviewed-on: https://webrtc-review.googlesource.com/96940
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24508}
2018-08-31 08:13:45 +00:00
523b4c4330 Add unlimited retransmission experiment for screenshare
Bug: webrtc:9659
Change-Id: Idcdc647c112ed2c7c027a7a0056b145ce8f45788
Reviewed-on: https://webrtc-review.googlesource.com/95724
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24422}
2018-08-24 08:48:30 +00:00
af17595879 Refactor RtpReceiverImpl, extracting CSRC book-keeping to its own class
Bug: webrtc:7135
Change-Id: I7ce9afe575241542e4e3f7e2e8459ee3257eec76
Reviewed-on: https://webrtc-review.googlesource.com/93466
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24271}
2018-08-13 11:59:08 +00:00
f8d81d33ed Add members for the codec agnostic descriptor to RTPVideoHeader.
TBR=danilchap@webrtc.org

Bug: webrtc:9361, webrtc:9582
Change-Id: I0303fc89bafab59e68ec81979e0e4372e79a4f51
Reviewed-on: https://webrtc-review.googlesource.com/91866
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24170}
2018-08-02 09:12:31 +00:00
a12c42a6b2 Delete root header file typedef.h.
Usage replaced with stdint.h, rtc_base/system/arch.h and
rtc_base/system/unused.h, as appropriate.

Bug: webrtc:6854
Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18
Reviewed-on: https://webrtc-review.googlesource.com/90249
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 14:59:26 +00:00
4206a0a849 Exposing video bitrate allocator into API
In order to have public video bitrate allocator factory, the video bitrate allocator has be part of
the api.

Bug: webrtc:9513
Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2
Reviewed-on: https://webrtc-review.googlesource.com/88083
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#24073}
2018-07-23 21:23:21 +00:00
1a4746a563 Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file.
Bug: none
Change-Id: If28f57c5ae250afbb47c5d20c9854e9a11182642
Reviewed-on: https://webrtc-review.googlesource.com/87561
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23904}
2018-07-10 11:57:46 +00:00
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
fb8e7ef842 Implement PayloadUnion as variant instead of pair of optionals
Bug: None
Change-Id: I2e54f5a0561804bc59c4d4c8e35ccdaa9536b8e4
Reviewed-on: https://webrtc-review.googlesource.com/85366
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23745}
2018-06-26 15:58:06 +00:00
beb2d9813c Removing usage of //build/config/compiler:no_size_t_to_int_warning.
Bug: webrtc:9251, webrtc:1348
Change-Id: I76e52abbfab5666cad73044b49172a9799539108
Reviewed-on: https://webrtc-review.googlesource.com/84144
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23686}
2018-06-20 13:44:26 +00:00