108 Commits

Author SHA1 Message Date
Artem Titov
ab30d72b72 Use backticks not vertical bars to denote variables in comments for /video
Bug: webrtc:12338
Change-Id: I47958800407482894ff6f17c1887dce907fdf35a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227030
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34585}
2021-07-28 13:22:27 +00:00
philipel
6ad542cf11 Remove temporary using webrtc::OnCompleteFrameCallback statement.
Bug: webrtc:12579
Change-Id: I9ba7735be20318b1f37bc7e75b07804ef694a7b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220362
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34162}
2021-05-31 11:45:33 +00:00
philipel
2182096e66 RtpFrameReferenceFinder return frames directly instead of via callback.
Bug: webrtc:12579
Change-Id: I41263f70a6f3dc60167e41f8b015a7d3b0dc3dd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219633
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@google.com>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34136}
2021-05-26 15:47:03 +00:00
Minyue Li
63b3095d2b Make local to capturer clock offset a separate entry in PacketInfo.
This also changes the meaning of |estimated_capture_clock_offset| in
|absolute_capture_time_| to become a remote to capturer clock offset.

Bug: chromium:1056230, webrtc:10739
Change-Id: Id658590e027bbe77ae0834ea224e1dc977a305f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219163
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#34067}
2021-05-20 13:42:57 +00:00
philipel
9599b3c582 Don't store RtpPacketInfo in the PacketBuffer.
Historically the PacketBuffer used a callback for assembled frames, and because of that RtpPacketInfos were piped through it even though they didn't have anything to do with the PacketBuffer.

With this CL RtpPacketInfos are stored in the RtpVideoStreamReceiver(2) instead.

Bug: webrtc:12579
Change-Id: Ia6285b59e135910eee7234b89b23425bb0fc0d2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215320
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33980}
2021-05-11 10:37:46 +00:00
philipel
b84931107c Update last received keyframe packet timestamp on all packets with the same RTP timestamp.
Bug: webrtc:12579,webrtc:12680
Change-Id: Id6e7b2c4199f52b3872ad407d8b602bed8b6cf3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215325
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33770}
2021-04-19 15:55:15 +00:00
philipel
dad500a728 Remove PacketBuffers internal mutex.
In RtpVideoStreamReceiver2 it can be protected by the `worker_task_checker_` instead.

Bug: webrtc:12579
Change-Id: I4f7d64f16172139eddc7a3e07d1dbbf338beaf2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215224
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33734}
2021-04-14 16:05:51 +00:00
philipel
ce423ce12d Track last packet receive times in RtpVideoStreamReceiver instead of the PacketBuffer.
Bug: webrtc:12579
Change-Id: I4adb8c6ada913127b9e65d97ddce0dc71ec6ccee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214784
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33713}
2021-04-13 18:24:45 +00:00
philipel
ca18809ee5 Move RtpFrameObject and EncodedFrame out of video_coding namespace.
Bug: webrtc:12579
Change-Id: Ib7ecd624eb5c54abb77fe08440a014aa1e963865
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33542}
2021-03-23 14:22:47 +00:00
philipel
6a6715042a Move RtpFrameReferenceFinder out of video_coding namespace.
Namespace used because of copy-pasting an old pattern, should never have been used in the first place. Removing it now to make followup refactoring prettier.

Bug: webrtc:12579
Change-Id: I00a80958401cfa368769dc0a1d8bbdd76aaa4ef5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212603
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33536}
2021-03-23 08:48:37 +00:00
Artem Titov
d15a575ec3 Use SequenceChecker from public API
Bug: webrtc:12419
Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 15:04:55 +00:00
Artem Titov
c8421c4c3e Replace rtc::ThreadChecker with webrtc::SequenceChecker
Bug: webrtc:12419
Change-Id: I825c014cc1c4b1dcba5ef300409d44859e971144
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205002
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33136}
2021-02-02 14:56:27 +00:00
Mirko Bonadei
20e4c80fbe Reland "Introduce RTC_NO_UNIQUE_ADDRESS."
This is a reland of f5e261aaf65cdf2eb903cdf40d651846be44f447

This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since
there have been some issues.

Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}

Bug: webrtc:11495, webrtc:12218
Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32668}
2020-11-23 11:29:36 +00:00
Mirko Bonadei
0abd518abd Revert "Introduce RTC_NO_UNIQUE_ADDRESS."
This reverts commit f5e261aaf65cdf2eb903cdf40d651846be44f447.

Reason for revert: Breaks downstream projects.

Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}

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

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

Bug: webrtc:11495
Change-Id: Ice318d1b11ca3dff09c190187a0b0a32ca945fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32335}
2020-10-07 07:37:01 +00:00
Mirko Bonadei
f5e261aaf6 Introduce RTC_NO_UNIQUE_ADDRESS.
This macro introduces the possibility to suggest the compiler that a
data member doesn't need an address different from other non static
data members.

The usage of a macro is to maintain portability since at the moment
the attribute [[no_unique_address]] is only supported by clang
with at least -std=c++11 but it should be supported by all the
compilers starting from C++20.

Bug: webrtc:11495
Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32246}
2020-09-30 09:52:49 +00:00
Niels Möller
3a749339be Delete obsolete TODO comment
Bug: webrtc:10198, webrtc:7408
Change-Id: I81e47dcc60abb7bdd2f0106a4370805994969980
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183364
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32046}
2020-09-07 11:10:55 +00:00
Niels Möller
f867ba8b5d Delete deprecated variant of RtpVideoStreamReceiver::AddReceiveCodec
Bug: None
Change-Id: I231967ca6777ec2fcbc6bdeef432267525182198
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181361
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31906}
2020-08-11 09:53:35 +00:00
Niels Möller
8e95ea92b2 Add method RtpVideoStreamReceiver::AddReceiveCodec with explicit payload type
Bug: None
Change-Id: If1008c9053a27b1e0d79299555675e17511069f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181240
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31894}
2020-08-10 14:26:41 +00:00
Markus Handell
a376518817 Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex.
Also migrates test/ partly.

Bug: webrtc:11567
Change-Id: If5b2eae65c5f297f364b6e3c67f94946a09b4a96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178862
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31672}
2020-07-08 12:21:08 +00:00
Markus Handell
a827a30bb7 Revert "Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex."
This reverts commit 0eba415fb40cc4e3958546a8ee53c698940df0a1.

Reason for revert: previously unknown lock recursion occurring downstream.

Original change's description:
> Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex.
> 
> Also migrates test/ partly.
> 
> Bug: webrtc:11567
> Change-Id: I4203919615c087e5faca3b2fa1d54cba9f171e07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178813
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31653}

TBR=sprang@webrtc.org,handellm@webrtc.org

Change-Id: I13f337e0de5b8f0eb19deb57cb5623444460ec4d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178842
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31656}
2020-07-07 20:46:48 +00:00
Markus Handell
0eba415fb4 Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex.
Also migrates test/ partly.

Bug: webrtc:11567
Change-Id: I4203919615c087e5faca3b2fa1d54cba9f171e07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178813
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31653}
2020-07-07 18:01:44 +00:00
Mirko Bonadei
9ca7365a8c Deprecate webrtc::NackModule.
This CL moves webrtc::NackModule to a deprecated folder and annotates
the type with RTC_DEPRECATED.

Since the header should not be used outside of WebRTC, this CL doesn't
created a forward header.

Bug: webrtc:11611
Change-Id: I4d5899d473d78b8c7f4a6a018e2805648244b5f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176360
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31394}
2020-05-30 16:34:44 +00:00
Tommi
d3807da009 Fork NackModule and RtpVideoStreamReceiver
Bug: webrtc:11595
Change-Id: I4d14c0bf9c32e09d1624099a256f2778afebd4df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175901
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31337}
2020-05-22 17:07:16 +00:00
Markus Handell
41acdc6074 Reland "RtpVideoStreamReceiver::RtcpFeedbackBuffer: remove lock recursions."
This reverts commit fde94a72d1e4f6bc38f8324bed390a9cebbc091c.

Reason for revert: there was a suspicion it broke the importer, but it was probably something else.

Original change's description:
> Revert "RtpVideoStreamReceiver::RtcpFeedbackBuffer: remove lock recursions."
> 
> This reverts commit ef93a26180660eaed00571996bb8e530be89320c.
> 
> Reason for revert: Checking if this broke things downstream.
> 
> Original change's description:
> > RtpVideoStreamReceiver::RtcpFeedbackBuffer: remove lock recursions.
> > 
> > This change removes lock recursions and adds thread annotations.
> > 
> > Bug: webrtc:11567
> > Change-Id: I68f62d0d62c8ad8dd8276e48f5876b75932bac61
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175113
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Markus Handell <handellm@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31314}
> 
> TBR=stefan@webrtc.org,handellm@webrtc.org
> 
> Change-Id: I99b622a0f88f3a264f1943f2457b9c9b89962b86
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11567
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175644
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31315}

TBR=tommi@webrtc.org,stefan@webrtc.org,handellm@webrtc.org

# Not skipping CQ checks because this is a reland.

Bug: webrtc:11567
Change-Id: I1171127bc4a0520561caf92ddb787ec7e649e7af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175651
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31322}
2020-05-19 13:04:20 +00:00
Tommi
fde94a72d1 Revert "RtpVideoStreamReceiver::RtcpFeedbackBuffer: remove lock recursions."
This reverts commit ef93a26180660eaed00571996bb8e530be89320c.

Reason for revert: Checking if this broke things downstream.

Original change's description:
> RtpVideoStreamReceiver::RtcpFeedbackBuffer: remove lock recursions.
> 
> This change removes lock recursions and adds thread annotations.
> 
> Bug: webrtc:11567
> Change-Id: I68f62d0d62c8ad8dd8276e48f5876b75932bac61
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175113
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31314}

TBR=stefan@webrtc.org,handellm@webrtc.org

Change-Id: I99b622a0f88f3a264f1943f2457b9c9b89962b86
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175644
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31315}
2020-05-19 05:58:44 +00:00
Markus Handell
ef93a26180 RtpVideoStreamReceiver::RtcpFeedbackBuffer: remove lock recursions.
This change removes lock recursions and adds thread annotations.

Bug: webrtc:11567
Change-Id: I68f62d0d62c8ad8dd8276e48f5876b75932bac61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175113
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31314}
2020-05-18 22:27:09 +00:00
Johannes Kron
14a23a32c4 Add field trial to force playout delay
This CL adds the field trial WebRTC-ForcePlayoutDelay with parameters
min_ms and max_ms. If both of these values are set, the playout delay
of any received packet will be overridden by the specified values.

Bug: None
Change-Id: I353282097e3ffa437dfc5affdfdf7780b09474e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174180
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31149}
2020-05-04 09:03:34 +00:00
Tommi
74fc574cbc Fork a few VideoReceiveStream related classes.
We'll need to deprecate the previous classes due to being used externally
as an API.

Bug: webrtc:11489
Change-Id: I64de29c8adae304d0b7628e24dd0abc5be6387ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31136}
2020-04-27 09:25:47 +00:00
Guido Urdaneta
e1aa22f892 [InsertableStreams] Set video frame transformer if RTP stream already started.
Test in https://chromium-review.googlesource.com/c/chromium/src/+/2127927

Bug: chromium:1065836
Change-Id: Idf3f41285e23ac829f69f1bc95b1def3a73af8d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172400
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Marina Ciocea <marinaciocea@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30948}
2020-03-31 14:07:29 +00:00
Marina Ciocea
6c08e4b57d Remove deprecated RtpVideoStreamReceiver constructor.
The dependencies have been updated to use the new constructor.

Bug: webrtc:11380
Change-Id: I1ded1816b94fd069d729df50ff83842eca054acc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170223
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30766}
2020-03-11 17:38:34 +00:00
Marina Ciocea
78964c1e0a Transform encoded frames in RtpVideoStreamReceiver.
This change is part of the implementation of the Insertable Streams Web
API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: If4ffcfe5761492a2ae5513ec46deb9f837e8aee8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169130
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30755}
2020-03-11 09:46:57 +00:00
Marina Ciocea
412a31bbf8 Insert frame transformer between Depacketizer and Decoder.
Add a new API in RTReceiverInterface, to be called from the browser side
to insert a frame transformer between the Depacketizer and the Decoder.

The frame transformer is passed from RTReceiverInterface through the
library to be eventually set in RtpVideoStreamReceiver, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169130.

This change is part of the implementation of the Insertable Streams Web
API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: I6b73cd16e3907e8b7709b852d6a2540ee11b4fed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169129
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30654}
2020-03-02 08:33:44 +00:00
Danil Chapovalov
e8f4e09be9 Parse DependencyDescriptor rtp header extension
Bug: webrtc:10342
Change-Id: I1b5914232f73803774523fae215cf719c92da305
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168481
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30563}
2020-02-20 09:09:27 +00:00
Danil Chapovalov
b42c54f949 Refactor parsing generic descriptor extension into own function
Before making it even more complicated that it is right now.

Bug: webrtc:10342
Change-Id: I54f67309b8832cd85b6c5213f9b090908814ebd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168766
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30557}
2020-02-19 13:50:36 +00:00
Danil Chapovalov
159c414ff8 Detach LossNotificationController from RtpGenericFrameDescriptor
To allow to use the LossNotificationController with
an updated version of the frame descriptor extension

Bug: webrtc:10342
Change-Id: I5ac44dc5549dfcfc73bf81ad1e8eab8bd5dd136e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166166
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30369}
2020-01-24 11:53:28 +00:00
Danil Chapovalov
cebdbf650d switch RtpVideoStreamReceiver to use VideoRtpDepacketizer interface
instead of creating each time an object with RtpDepacketizer interface

this moves packet payload memcpy from RtpVideoStreamReceiver into
the depacketizers with possibility to remove it from there in follow ups.

Bug: webrtc:11152
Change-Id: If474207eb84d7e9d0207075bd395e60895f0d842
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162185
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30095}
2019-12-16 12:08:11 +00:00
Ruslan Burakov
d08bb1e12f Propagate absolute capture time through video receiving side.
Prototype link:
https://webrtc-review.googlesource.com/c/src/+/158520


Bug: webrtc:10739
Change-Id: I8d30b729ac5bca484608af7f0378998987df7d53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160341
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29939}
2019-11-27 17:07:59 +00:00
Ilya Nikolaevskiy
815e00c102 Revert "Reset RtpFrameReferenceFinder on long pause"
This reverts commit 7a4db6eb0ef5a998019f03428072f0cc6afae866.

Reason for revert: Caused regression on perf tests.

Original change's description:
> Reset RtpFrameReferenceFinder on long pause
> 
> Bug: webrtc:11074
> Change-Id: I4c9a8761e9039d32885ccf9ac0eebdffdf67f48d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159240
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29747}

TBR=ilnik@webrtc.org,sprang@webrtc.org

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

Bug: webrtc:11074
Change-Id: Ic40779087bf8e6bd94f02d38161f6abb9ca395f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159690
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29775}
2019-11-12 16:26:38 +00:00
Ilya Nikolaevskiy
7a4db6eb0e Reset RtpFrameReferenceFinder on long pause
Bug: webrtc:11074
Change-Id: I4c9a8761e9039d32885ccf9ac0eebdffdf67f48d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159240
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29747}
2019-11-08 16:52:14 +00:00
Danil Chapovalov
09860e0bc3 Split out counting unique rtp timestamps from packet_buffer
Bug: None
Change-Id: Ia6fd05f284e8304cf56ab9ddf944fb222a4c9573
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158676
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29656}
2019-10-30 15:27:48 +00:00
Danil Chapovalov
ce1ffcdc06 change PacketBuffer to return it's result rather that use callback
Bug: None
Change-Id: I8cc05dd46e811d6db37af520d2106af21c671def
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157893
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29589}
2019-10-23 16:50:57 +00:00
Danil Chapovalov
c71d85bc4e Pass full RtpPacket to RtpVideoStreamReceiver::OnReceivedPayload
that brings RtpPacketReceived closer to the packet buffer
to allow strore original packets rather than VCMPacket in it.

Bug: webrtc:10979
Change-Id: Ia0fc0abf3551a843b19b0ee66ca0f20cae014479
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157164
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29516}
2019-10-17 14:48:32 +00:00
philipel
7acc4a4a3a Reset |reference_finder_| on codec switch.
In this CL:
 - Moved critical section out of RtpFrameReferenceFinder.
 - RtpFrameReferenceFinder can now assign picture ids with an offset.
 - RtpVideoStreamReceiver will now reset the |reference_finder_| in case
   of a codec switch.

Bug: webrtc:10795, webrtc:10828
Change-Id: I22631c121a465c434de24af5ce8be2a647fe3556
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154353
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29317}
2019-09-26 11:05:59 +00:00
Danil Chapovalov
f7457e55fe Store PacketBuffer by value instead of as reference counted object
Bug: None
Change-Id: I5a594972e8a8dad731c927a1a374301e549f5d71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153887
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29267}
2019-09-23 13:28:09 +00:00
Chen Xing
b56cca3282 Remove the old ContributingSources class.
This change removes the old `ContributingSources` class. It has been replaced by the new `SourceTracker`.

Bug: webrtc:10793
Change-Id: Ibd481cf6584837c46b229b9fc2a071362f07d361
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147878
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28756}
2019-08-05 12:30:22 +00:00
Elad Alon
a47ba4119f Get rid of deprecated version of NackSender::SendNack [1/2]
[1/2] - Make new version pure-virtual, and deprecated version non-pure.
        This will allow deleting the deprecated version from downstream
	projects.
[2/2] - Remove deprecated version.

TBR=stefan@webrtc.org

Bug: webrtc:10336
Change-Id: Ia132ef071b1f379fc74834178e75e981ca908125
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144042
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28413}
2019-06-28 13:04:48 +00:00
Chen Xing
90f3b89550 Replace the implementation of GetContributingSources() on the video side.
This change replaces the `ContributingSources`-implementation of `GetContributingSources()` and `GetSynchronizationSources()` on the video side with the spec-compliant `SourceTracker`-implementation.

The most noticeable impact is that the per-frame dictionaries are now updated when frames are delivered to the RTCRtpReceiver's MediaStreamTrack rather than when RTP packets are received on the network.

Bug: webrtc:10545
Change-Id: I895b5790280ac94c1501801d226c643633c67349
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143177
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28386}
2019-06-26 11:57:33 +00:00
Elad Alon
e86af2c75f Allowing buffering a LNTF (loss notification) feedback message in RTCPSender
Loss notifications may either be sent immediately, or wait until another
RTCP feedback message is sent.

Bug: webrtc:10336
Change-Id: I40601d9fa1dec6c17b2ce905cb0c8cd2dcff7893
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139242
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28142}
2019-06-03 16:28:34 +00:00
Elad Alon
ef09c5b734 Buffer RTCP feedback messages in RtpVideoStreamReceiver
Currently, if LNTF and NACK messages are both created, they will
be sent out in separate RTCP messages. This is wasteful.
This CL is the first of in a series of CLs that will ensure that
these feedback messages can be buffered together, without introducing
more of a delay than the CPU time required to process both messages.

Bug: webrtc:10336
Change-Id: I950324112ee346695a12a17d025483ea5e99c732
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139112
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28136}
2019-06-03 12:19:36 +00:00
Niels Möller
2f5554dae5 Make KeyFrameRequestSender injectable in RtpVideoStreamReceiver
This is a partial revert of
https://webrtc-review.googlesource.com/c/src/+/130101.

The KeyFrameRequestSender argument is added back to the constructor of
RtpVideoStreamReceiver. It is optional; if a null pointer is passed,
key frame requests are sent via the internal RtpRtcp module, and this is
how the class is used by VideoReceiveStream. An injectable
KeyFrameRequestSender is useful for tests, for downstream applications
that want to route key frame requests elsewhere, and should also aid
later migration to RtcpTransciever.

Bug: None
Change-Id: Idf9baeed21570625ad74e9afbe38f7ea5bf79feb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139107
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28102}
2019-05-29 12:24:02 +00:00