383 Commits

Author SHA1 Message Date
Henrik Grunell
e1301a8b3a Revert "Implement read-only codecPayloadType in RtpParameters"
This reverts commit 806e06d1366b58878ced05cdd8d1d56394982fe6.

Reason for revert: Breaks WebRTC roll to Chromium. https://chromium-review.googlesource.com/c/chromium/src/+/1375538

02:52:35.346 7748   [6936:11248:1213/025234.206:ERROR:mediaengine.cc(80)] Attempted to set RtpParameters with modified codecPayloadType (INVALID_MODIFICATION)

Original change's description:
> Implement read-only codecPayloadType in RtpParameters
> 
> Bug: webrtc:7580
> Change-Id: I6d901afa97262b6c6d9fe6c7366df465ec77bfb3
> Reviewed-on: https://webrtc-review.googlesource.com/c/113944
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25993}

TBR=steveanton@webrtc.org,sakal@webrtc.org,andersc@webrtc.org,shampson@webrtc.org,orphis@webrtc.org

Change-Id: I157f9a79ae7133395431891e15e2c053559d359b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7580
Reviewed-on: https://webrtc-review.googlesource.com/c/114300
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Henrik Grunell <henrikg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26000}
2018-12-13 12:13:30 +00:00
Florent Castelli
806e06d136 Implement read-only codecPayloadType in RtpParameters
Bug: webrtc:7580
Change-Id: I6d901afa97262b6c6d9fe6c7366df465ec77bfb3
Reviewed-on: https://webrtc-review.googlesource.com/c/113944
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25993}
2018-12-12 16:24:29 +00:00
Niels Möller
1d8307d706 Delete VideoCodec::targetBitrate
This member is unused by encoders.

Bug: None
Change-Id: I867013bfdb89f48782e84842de05bb57648e0b64
Reviewed-on: https://webrtc-review.googlesource.com/c/113882
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25988}
2018-12-12 12:48:15 +00:00
Anders Carlsson
01092957f5 Mark functions using old factory classes as deprecated.
The flag rtc_use_builtin_sw_codecs will be removed in a later CL and
this marks usage of the various entry points using the old video factory
API as deprecated.

Bug: webrtc:7925, webrtc:10044
Change-Id: I5c75516a41b0666e77539c028808cc2b173ed4bd
Reviewed-on: https://webrtc-review.googlesource.com/c/113061
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25970}
2018-12-11 16:17:33 +00:00
Benjamin Wright
514f084c26 New statistic added to VideoReceiveStream to determine latency to first decode.
This change introduces a new measurement into the VideoReceiveStream::Stats
structure to measure the latency between the first frame being received and
the first frame being decoded in WebRTC. The goal here is to measure the latency
difference when a FrameEncryptor is attached and not attached.

Change-Id: I0f0178aff73b66f25dbc6617098033e226da2958
Bug: webrtc:10105
Reviewed-on: https://webrtc-review.googlesource.com/c/113328
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25956}
2018-12-10 18:49:34 +00:00
Sergey Silkin
8b9b5f98db Activate/deactivate VP9 spatial layers.
* Stop encoding spatial layers S(n >= N) if application deactivates
spatial layer N by setting RTCRtpEncodingParameters.active = false.

* Move calculation of padding bitrate to SvcRateAllocator class.

* Pad up to minimum required bitrate of base layer if ALR probing is
enabled.

Bug: webrtc:9350
Change-Id: I398284c943d43348def535c83263fc234c9767fa
Reviewed-on: https://webrtc-review.googlesource.com/c/113240
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25945}
2018-12-10 12:55:51 +00:00
Mirta Dvornicic
1ec2a16121 Revert "Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo"
This reverts commit cdc5eb0de179dcc866ef770ea303879c64466879.

Reason for revert: Causes wrong CPU adaptation to be used for some HW codecs since GetEncoderInfo() is polled before InitEncode().

Original change's description:
> Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo
> 
> Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
> until it is removed downstream and remove all implementations of it.
> 
> Bug: webrtc:10065
> Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
> Reviewed-on: https://webrtc-review.googlesource.com/c/113065
> Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25924}

TBR=brandtr@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,mirtad@webrtc.org

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

Bug: webrtc:10065
Change-Id: Idaa452e1d8c1c58cdb4ec69b88fce9042589cc3c
Reviewed-on: https://webrtc-review.googlesource.com/c/113800
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25943}
2018-12-10 10:36:00 +00:00
Mirta Dvornicic
cdc5eb0de1 Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo
Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
until it is removed downstream and remove all implementations of it.

Bug: webrtc:10065
Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
Reviewed-on: https://webrtc-review.googlesource.com/c/113065
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25924}
2018-12-06 15:24:45 +00:00
Florent Castelli
e7862cc6b5 Copy VP8EncoderSimulcastProxy to EncoderSimulcastProxy
Use the new class internally where appropriate too.

The objective is to rename it, but due to some external dependency,
it is better to copy, update dependencies and remove.

Bug: webrtc:10069
Change-Id: I8477ce5a2982933db27513cc9509f51558dafaf3
Reviewed-on: https://webrtc-review.googlesource.com/c/113265
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25920}
2018-12-06 13:24:07 +00:00
Ilya Nikolaevskiy
5546aef682 Vp9 flexible mode fixes
- Enable vp9 flexible mode in VideoEngine if 3 spatial layers are set.
- Enable flexible mode in loopback tools and quality tests.
- Reset first active spatial layer on keyframe in encoder.
- Ensure duplicate references are not set by the sender in video header.
- Set references manually for flexible mode in vp9 encoder.
- Delay new activated layers until next base layer frame.
- On receive side put each spatial layer as a separate frame to FrameBuffer
  and return several frames combined from FrameBuffer.

Bug: webrtc:10049,webrtc:9794,webrtc:9784
Change-Id: I01e69f134cc145deba666ccc92deb1d37a324ede
Reviewed-on: https://webrtc-review.googlesource.com/c/112289
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25895}
2018-12-04 15:36:28 +00:00
Johannes Kron
d0b69a8c50 Send and receive color space information if available
Bug: webrtc:8651
Change-Id: I244647cb1ccbda66fce83ae925cf4273c5a6568b
Reviewed-on: https://webrtc-review.googlesource.com/c/112383
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25884}
2018-12-03 21:07:45 +00:00
Mirta Dvornicic
897a991618 Add metadata from VideoEncoderFactory::CodecInfo to VideoEncoder::EncoderInfo
This is the first step in moving the metadata and eventually replacing
VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo.

Bug: webrtc:10065
Change-Id: If925b895718e1b1225d2cf49bede1adb3ff281b8
Reviewed-on: https://webrtc-review.googlesource.com/c/112285
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25856}
2018-11-30 12:58:53 +00:00
Erik Språng
72e52ee619 Make simulcast screenshare default-on
Bug: chromium:690537
Change-Id: I7380f5e7b3faa20ba60bebee8b8b4d74db885faf
Reviewed-on: https://webrtc-review.googlesource.com/c/112381
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25840}
2018-11-29 14:17:41 +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
Jakob Ivarsson
10403ae87c Add PeerConnection option to configure minimum audio jitter buffer delay.
Note that this value will override the minimum delay that is used for audio/video sync.

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

Change-Id: Ib191a11c6bd9e617abbe9dd82239b0c5b4e6b4e0
Bug: webrtc:10043
Reviewed-on: https://webrtc-review.googlesource.com/c/111922
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25802}
2018-11-27 15:10:09 +00:00
Ruslan Burakov
8af8896596 Expose jitter buffer flushes metric in new getStats api.
Origin trial experiment proposal (new statistic part):
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI/edit?ts=5bf5535c#

Bug: chromium:907113
Change-Id: I1d005291f9b47665f70c26148dbdcbb55564bef8
Reviewed-on: https://webrtc-review.googlesource.com/c/111505
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#25768}
2018-11-23 11:41:43 +00:00
Fredrik Solenberg
c69a56ef04 Remove more unneeded things from ChannelSend
- SetNACKStatus() - only affects NetEq and RTP receiver
- GetRtpTimestampRateHz() - never used.
- ResendPackets() - never used.

Bug: webrtc:9801
Change-Id: I280b620723eb6917624f30f503eb8b8c88144e6d
Reviewed-on: https://webrtc-review.googlesource.com/c/111460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25721}
2018-11-21 09:04:07 +00:00
Niels Möller
2222a80e79 Delete unneeded includes of common_types.h and gn deps on webrtc_common.
Bug: webrtc:5876
Change-Id: Iae14e5f1679067a5a5e0584ca830aee0870c8807
Reviewed-on: https://webrtc-review.googlesource.com/c/111463
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25715}
2018-11-20 16:28:39 +00:00
Florent Castelli
38332cdcb1 Add RTCP and simulcast support for RTCRtpReceiver::getParameters()
Bug: webrtc:9989
Change-Id: I1235789cd485750937a427199f9d32ed6180145e
Reviewed-on: https://webrtc-review.googlesource.com/c/110616
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25714}
2018-11-20 14:03:18 +00:00
Niels Möller
22b70ff1d4 Move VideoCodecType from common_types.h to api/video/video_codec_type.h
Bug: webrtc:7660
Change-Id: I9381364a64113dbb622b26acbf2b71228c3c4b96
Reviewed-on: https://webrtc-review.googlesource.com/c/111480
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25710}
2018-11-20 13:12:57 +00:00
Mirko Bonadei
8ef57932b1 Switch from RTC_DISABLE_VP9 to RTC_ENABLE_VP9.
RTC_ENABLE_VP9 is more natural to deal with then RTC_DISABLE_VP9.
In all the places this macro is used, WebRTC needs to do more things
so it is easier to "do more if RTC_ENABLE_VP9 is defined" than
"do more if RTC_DISABLE_VP9 is not defined".

Bug: None
Change-Id: If992e5c554173e6af3f030f6e0fd21bd82acf9eb
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/111242
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25679}
2018-11-19 08:30:55 +00:00
Sebastian Jansson
fa0aa39fba Removes templating from CompositeMediaEngine.
Usage of templates makes it harder for tooling to help the user. This
can be experienced when trying to investigate compile failures and using
editor tools to browse the code.

This CL replaces usage of templates with injection of unique pointers to
interfaces that implements the behavior that previously was assumed by
the templated implementation.

Bug: webrtc:9883
Change-Id: Ica17af9646f68a9b063988f9e85d6acc8ca37c10
Reviewed-on: https://webrtc-review.googlesource.com/c/106703
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25668}
2018-11-16 10:31:57 +00:00
Sebastian Jansson
84848f26b5 Adds interfaces for audio and video engines.
This makes the currently implicit interfaces explicit and
prepares for making CompositeMediaEngine non-templated.

Bug: webrtc:9883
Change-Id: I57452acc9ada60a801f6d624894440a942c12ded
Reviewed-on: https://webrtc-review.googlesource.com/c/106940
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25667}
2018-11-16 10:10:36 +00:00
Piotr (Peter) Slatala
cc8e8bb73f Pass the media transport from JsepTransportController to Call.
Add TargetRateObservers for media transport in the call object.



Bug: webrtc:9719
Change-Id: I5448d05359cf09b8cd2a678b2ac876aa8f8970e7
Reviewed-on: https://webrtc-review.googlesource.com/c/110622
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25662}
2018-11-15 17:36:48 +00:00
Jonas Olsson
8544799cf1 Introduce DLOG to video and voiceengine.
This CL removes a handful of low-importance logging from our release builds.

Bug: webrtc:8529
Change-Id: I1043f501c16ce24a39512307e8cddccf4c4d4ab6
Reviewed-on: https://webrtc-review.googlesource.com/c/47163
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25622}
2018-11-13 15:11:47 +00:00
Jiawei Ou
55718120e6 Adding rtcp report interval into RTCConfiguration.
This is a follow up of https://webrtc-review.googlesource.com/c/src/+/43201.

Issue 43201 didn't do the job properly.
1. The audio rtcp report interval is not properly hooked up.
2. We don't need to propagate audio rtcp interval into video send stream or vice versa.
3. We don't need to propagate rtcp report interval to any receiving streams.

Bug: webrtc:8789
Change-Id: I1f637d6e5173608564ef0702d7eda6fc93b3200f
Reviewed-on: https://webrtc-review.googlesource.com/c/110105
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#25610}
2018-11-12 20:00:00 +00:00
Jiawei Ou
c2ebe21ba9 Reland "Use the factory instead of using the builtin code path in VideoCodecInitializer"
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782

This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.

Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.

One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.

Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
2018-11-08 19:10:47 +00:00
Erik Språng
d3438aa1ed Add ability to specify if rate controller of video encoder is trusted.
If rate controller is trusted, we disable the frame dropper in the
media optimization module.

This is a re-land of
https://webrtc-review.googlesource.com/c/src/+/105020

Bug: webrtc:9890
Change-Id: I418e47a43a1a98cb2fd5295c03360b954f2288f2
Reviewed-on: https://webrtc-review.googlesource.com/c/109141
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25570}
2018-11-08 16:41:12 +00:00
Erik Språng
75de46a966 Update SimulcastEncoderAdapter merging of EncoderInfo
Determining the EncoderInfo meta data is now done during InitEncode().

This implementation assums that no dynamic wrappers are wrapped in this
simulcast encoder adapter. Ie, if supports_native_handle changes,
InitEncode() must be called again for it to be reported properly.

Bug: webrtc:9722
Change-Id: I7901effe11e89ac011659a4ea862ab2a42577eb5
Reviewed-on: https://webrtc-review.googlesource.com/c/109620
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25549}
2018-11-07 15:22:38 +00:00
Niels Möller
c572ff3c71 Add default constructor for rtc::Event
Bug: webrtc:9962
Change-Id: Icaa91e657e6881fcb1553f354c07866109a0ea68
Reviewed-on: https://webrtc-review.googlesource.com/c/109500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25535}
2018-11-07 08:57:50 +00:00
philipel
ee49f7087f Remove VideoEncoder::SetChannelParameters.
The SetChannelParameters function was used when WebRTC supported decoding
with errors, which we no longer do.

This cleanup CL is related to the work tracked by 9946.

Bug: webrtc:9946
Change-Id: Id2d5ed23031388f890c42651bfbe5f79eda701e5
Reviewed-on: https://webrtc-review.googlesource.com/c/108861
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25505}
2018-11-05 17:37:07 +00:00
Niels Möller
e693381cda Delete struct rtc::PacketTime.
Replaced by a int64_t representing time in us. To aid transition of
downstream code, rtc::PacketTime is made an alias for int64_t.

Bug: webrtc:9584
Change-Id: Ic3a5ee87d6de2aad7712894906dab074f1443df9
Reviewed-on: https://webrtc-review.googlesource.com/c/91860
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25503}
2018-11-05 16:21:39 +00:00
Erik Språng
9b5b070817 Use EncoderInfo in SimulcastEncoderAdapter
Remove use of deprecated methods.

Bug: webrtc:9890
Change-Id: I96cce2fc94cb4c4ac07ffc882f5d2b84e279e3b6
Reviewed-on: https://webrtc-review.googlesource.com/c/108123
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25474}
2018-11-01 16:05:57 +00:00
Erik Språng
449afd9374 Updated ScopedVideoEncoder to use GetEncoderInfo()
Bug: webrtc:9890
Change-Id: Icca1cc1df6a227a30a5f54228fa33a9e63e702e0
Reviewed-on: https://webrtc-review.googlesource.com/c/109007
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25471}
2018-11-01 15:43:15 +00:00
Niels Möller
15ca5a9533 Add implicit conversion between rtc:PacketTime and int64_t.
This is a preparation for deleting rtc::PacketTime. Next step, after
downstream code has been updated to not access the |timestamp| member,
is to make rtc::PacketTime an alias for int64_t.

Also delete the unused member rtc::PacketTime::not_before.

Bug: webrtc:9584
Change-Id: Iba9d2d55047d69565ad62b1beb525591fd432ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/108860
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25468}
2018-11-01 14:28:24 +00:00
Qingsi Wang
59844ce57e Revert "Use the factory instead of using the builtin code path in VideoCodecInitializer."
This reverts commit be142178aaf6ab4089b4d81c88c3d59c12cca567.

Reason for revert: breaking internal projects

Original change's description:
> Use the factory instead of using the builtin code path in `VideoCodecInitializer`.
> 
> Bug: webrtc:9513
> Change-Id: Ia299ae1044a3ff4c91e208200938cba540bdcea6
> Reviewed-on: https://webrtc-review.googlesource.com/c/94782
> Commit-Queue: Jiawei Ou <ouj@fb.com>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25456}

TBR=brandtr@webrtc.org,magjed@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,andersc@webrtc.org,tommi@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,tkchin@webrtc.org,shampson@webrtc.org,glaznev@webrtc.org,ouj@fb.com,qingsi@webrtc.org

Change-Id: I8040ccabe3ae6464d72c7696adb663c1dd275b63
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9513
Reviewed-on: https://webrtc-review.googlesource.com/c/108980
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25459}
2018-11-01 04:46:02 +00:00
Jiawei Ou
be142178aa Use the factory instead of using the builtin code path in VideoCodecInitializer.
Bug: webrtc:9513
Change-Id: Ia299ae1044a3ff4c91e208200938cba540bdcea6
Reviewed-on: https://webrtc-review.googlesource.com/c/94782
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25456}
2018-10-31 22:47:02 +00:00
Johannes Kron
9190b82660 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
Bug: webrtc:7990
Change-Id: I662595f90b9d0be39f7e14752e13b2bb7a1746ee
Reviewed-on: https://webrtc-review.googlesource.com/c/106020
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25421}
2018-10-30 08:06:49 +00:00
Tim Haloun
436ebcaec1 Fix extra setdscp call introduced by bad merge.
Bug: webrtc:5008
Change-Id: I29b0debf0468c8c0ab5120e77dc774b566f5b446
Reviewed-on: https://webrtc-review.googlesource.com/c/108003
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Tim Haloun <thaloun@chromium.org>
Cr-Commit-Position: refs/heads/master@{#25394}
2018-10-26 17:33:16 +00:00
Danil Chapovalov
99b71dfd4a Use function_video_(en|de)coder_factory from api
Remove them from test.
It is completion of the move started with
https://webrtc-review.googlesource.com/c/src/+/107705

Bug: None
Change-Id: Ib0b26db04a1ee814322851280ba1e59b4b3f7ce6
Reviewed-on: https://webrtc-review.googlesource.com/c/107891
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25392}
2018-10-26 14:54:19 +00:00
Niels Möller
7d76a31f3d Use MediaTransportInterface, for audio streams.
Bug: webrtc:9719
Change-Id: I6d3db66b781173b207de51d84193fbd34a7f3239
Reviewed-on: https://webrtc-review.googlesource.com/c/104642
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25385}
2018-10-26 11:40:57 +00:00
Jonathan Yu
327b7535f9 Split out a separate target for VP8EncoderSimulcastProxy
This will allow downstream projects to use it to construct their own
injected codecs without pulling in dependencies on software codecs that
they don't need.

TBR=shampson@webrtc.org

Bug: webrtc:7925
Change-Id: Ie5c246bbf8e2ef1b27562887f717af9e719a1edf
Reviewed-on: https://webrtc-review.googlesource.com/c/107698
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25379}
2018-10-25 21:44:15 +00:00
Niels Möller
039743e066 Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
This is a reland of 80cd25bcfb2264fa0f1192de942a6f063879dd42

Original change's description:
> Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase
>
> Bug: None
> Change-Id: I225fe1e16a3c96e5a03e3ae8fe975f368be7e6ad
> Reviewed-on: https://webrtc-review.googlesource.com/c/107303
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25312}

Tbr: kwiberg@webrtc.org
Bug: None
Change-Id: Id43a93bada9d6d66a4d0f0286f583066156aa2fc
Reviewed-on: https://webrtc-review.googlesource.com/c/107716
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25368}
2018-10-25 14:13:44 +00:00
Erik Språng
e2fd86a79c Move encoder metadata into EncoderInfo struct.
This deprecates the following methods in VideoEncoder:
  virtual ScalingSettings GetScalingSettings() const;
  virtual bool SupportsNativeHandle() const;
  virtual const char* ImplementationName() const;

Though they are not marked RTC_DEPRECATED since we still want to call
them from within the default GetEncoderInfo() until downstream
projects have been updated.

Furthmore, implementation name is changed from const char* to
std:string, which prevents some lifetime issues with dynamic encoder
names, and CodecSpecificInfo.codec_name is removed in favor of getting
the implementation name via GetEncoderInfo().

This CL removes calls to these deprecated methods, follow-ups will also
remove implementations of the methods and replace them with new
GetEncoderInfo() substitutions.

Bug: webrtc:9890
Change-Id: I6fd6e531480c0b952f53dbd5105e0b0adc3e3b0c
Reviewed-on: https://webrtc-review.googlesource.com/c/106905
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25351}
2018-10-25 08:51:53 +00:00
Johannes Kron
d38a2b860b Increase the UDP receive buffer for video
Lost packets have been seen in high-bitrate applications and increasing
the UDP receive buffer reduced the problems.

Bug: b/115713113
Change-Id: I671f528afeaea525150fdc2013f2b245778e5d16
Reviewed-on: https://webrtc-review.googlesource.com/c/107580
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25328}
2018-10-24 07:54:12 +00:00
Oleh Prypin
6e8e2993dd Revert "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
This reverts commit 80cd25bcfb2264fa0f1192de942a6f063879dd42.

Reason for revert: Breaks downstream project

Original change's description:
> Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase
>
> Bug: None
> Change-Id: I225fe1e16a3c96e5a03e3ae8fe975f368be7e6ad
> Reviewed-on: https://webrtc-review.googlesource.com/c/107303
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25312}

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

No-Try: true
Bug: None
Change-Id: I77b66bc032e2d95d1bd408c6cdeceb4dcd511699
Reviewed-on: https://webrtc-review.googlesource.com/c/107643
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25317}
2018-10-23 13:21:27 +00:00
Niels Möller
80cd25bcfb Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase
Bug: None
Change-Id: I225fe1e16a3c96e5a03e3ae8fe975f368be7e6ad
Reviewed-on: https://webrtc-review.googlesource.com/c/107303
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25312}
2018-10-23 12:13:02 +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
Sam Zackrisson
b0ab2ce256 Reland "Remove the HighPassFilter interface"
Downstream Chromium dependencies fixed here:
https://chromium-review.googlesource.com/c/chromium/src/+/1286449

This is a reland of e2405c1a823f3baf90a9c72f2e058f91eb659c20

Original change's description:
> Remove the HighPassFilter interface
>
> The functionality remains unaffected.
> Filter toggling is still available via webrtc::AudioProcessing::Config.
> Example:
> webrtc::AudioProcessing::Config config = apm.GetConfig();
> // Read settings
> if (config.high_pass_filter.enabled) { ... }
> // Apply setting
> config.high_pass_filter.enabled = true;
> apm.ApplyConfig();
>
> Bug: webrtc:9535
> Change-Id: Ib4c4b04078bbb490ebdab9721b8c7811d73777a8
> Reviewed-on: https://webrtc-review.googlesource.com/c/102541
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25198}

Bug: webrtc:9535
Change-Id: I0017193ad3ca1762e186f3ad79f29d33ef468202
Reviewed-on: https://webrtc-review.googlesource.com/c/106681
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25300}
2018-10-23 07:44:09 +00:00