Commit Graph

30725 Commits

Author SHA1 Message Date
8eccf236f8 Generate new Android.bp file and correct build errors am: b6df60492c am: 1206c45de1 am: 024d59e7a7
Original change: https://android-review.googlesource.com/c/platform/external/webrtc/+/1358846

Change-Id: Ie829283b3180d93489ed4fe9703e77e7e3b25930
2020-07-25 00:10:49 +00:00
024d59e7a7 Generate new Android.bp file and correct build errors am: b6df60492c am: 1206c45de1
Original change: https://android-review.googlesource.com/c/platform/external/webrtc/+/1358846

Change-Id: I10dd254b34af35cb422b998fee23649bd9a7b1d4
2020-07-24 23:57:13 +00:00
1206c45de1 Generate new Android.bp file and correct build errors am: b6df60492c
Original change: https://android-review.googlesource.com/c/platform/external/webrtc/+/1358846

Change-Id: I39cd125b095adbd0aed6b426d61a1266545068dc
2020-07-24 23:39:22 +00:00
a1c83112e4 [automerger skipped] Merge "Merge branch 'upstream-master'" am: e41ddee6b4 -s ours am: 1af77b4142 -s ours am: 798edca709 -s ours
am skip reason: Change-Id Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802 with SHA-1 d1c647242a is in history

Original change: https://android-review.googlesource.com/c/platform/external/webrtc/+/1358845

Change-Id: Ia5ef2db93849dd927ef68bb7c715d6894c2b36a6
2020-07-23 20:54:40 +00:00
798edca709 [automerger skipped] Merge "Merge branch 'upstream-master'" am: e41ddee6b4 -s ours am: 1af77b4142 -s ours
am skip reason: Change-Id Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802 with SHA-1 d1c647242a is in history

Original change: https://android-review.googlesource.com/c/platform/external/webrtc/+/1358845

Change-Id: Id8a56a72b7c91fa60947693bec9beb2c1a290678
2020-07-23 20:35:54 +00:00
b6df60492c Generate new Android.bp file and correct build errors
The following are not yet available in their respective libraries so
attempts to use it in webrtc result in a call to abort():
* libvpx's CONSTRAINED_FROM_ABOVE_DROP constant
* libyuv's I010 buffers

The original webrtc project expects to have third party libraries
checked out in third_party/ and base/third_party/. Added some headers
in those directories with a single line including the right header from
external/<library>. Updated .gitignore to keep track of said headers.

Bug: 153469641
Test: mm, also built cuttlefish using this library and ran it locally
Change-Id: I2d596942e34093dccc65d4b7b8249b6afc14d31f
Merged-In: I2d596942e34093dccc65d4b7b8249b6afc14d31f
2020-07-23 13:34:20 -07:00
1af77b4142 [automerger skipped] Merge "Merge branch 'upstream-master'" am: e41ddee6b4 -s ours
am skip reason: Change-Id Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802 with SHA-1 d1c647242a is in history

Original change: https://android-review.googlesource.com/c/platform/external/webrtc/+/1358845

Change-Id: Iec9fe0782c52c13be1dd604aff9b8c755105d58d
2020-07-23 20:23:47 +00:00
e41ddee6b4 Merge "Merge branch 'upstream-master'" 2020-07-23 19:53:36 +00:00
0d145d1783 [automerger skipped] Merge branch 'upstream-master' am: d1c647242a -s ours am: 107719457e -s ours
am skip reason: Change-Id Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802 with SHA-1 d4cf4096a2 is in history

Original change: https://googleplex-android-review.googlesource.com/c/platform/external/webrtc/+/12197202

Change-Id: I43f03c372d6be90002d1e029c0e0c134ba1c1019
2020-07-23 09:38:04 +00:00
107719457e [automerger skipped] Merge branch 'upstream-master' am: d1c647242a -s ours
am skip reason: Change-Id Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802 with SHA-1 d4cf4096a2 is in history

Original change: https://googleplex-android-review.googlesource.com/c/platform/external/webrtc/+/12197202

Change-Id: I7c5be5191ce8a7072612e1a3a463e8917a87e6bb
2020-07-23 09:22:16 +00:00
ca1f27e182 [automerger skipped] Merge branch 'upstream-master' am: d4cf4096a2 -s ours
am skip reason: Change-Id Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802 with SHA-1 aad12c2ec0 is in history

Original change: https://googleplex-android-review.googlesource.com/c/platform/external/webrtc/+/12197404

Change-Id: I04eb77aa76e049a395c167ff0f928008bc556169
2020-07-23 00:52:19 +00:00
f8ebb49c09 Merge branch 'upstream-master'
Bug: 153469641
Test: none
Change-Id: Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802
Merged-In: Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802
2020-07-21 14:54:49 -07:00
d1c647242a Merge branch 'upstream-master'
Bug: 153469641
Test: none
Change-Id: Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802
Merged-In: Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802
2020-07-20 17:28:31 -07:00
d4cf4096a2 Merge branch 'upstream-master'
Bug: 153469641
Test: none
Change-Id: Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802
Merged-In: Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802
2020-07-20 17:26:10 -07:00
aad12c2ec0 Merge branch 'upstream-master'
Bug: 153469641
Test: none
Change-Id: Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802
Merged-In: Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802
2020-07-20 17:24:18 -07:00
547a67b290 Merge "Export include dirs" am: f92ec5a27c am: 39a3db90ff am: 973c7ccea0
Original change: https://android-review.googlesource.com/c/platform/external/webrtc/+/1347705

Change-Id: Iebab7f2dd536b7e237fddf1435134fc85db191d8
2020-06-24 01:56:36 +00:00
973c7ccea0 Merge "Export include dirs" am: f92ec5a27c am: 39a3db90ff
Original change: https://android-review.googlesource.com/c/platform/external/webrtc/+/1347705

Change-Id: I02a488ce82be251d29358124810c4e96963d73e2
2020-06-24 01:41:07 +00:00
39a3db90ff Merge "Export include dirs" am: f92ec5a27c
Original change: https://android-review.googlesource.com/c/platform/external/webrtc/+/1347705

Change-Id: I02e9d2b9eaa8d8bfa87ad9349eb6affcd8312fa2
2020-06-24 01:22:24 +00:00
f92ec5a27c Merge "Export include dirs" 2020-06-24 01:11:58 +00:00
8db44406d1 Export include dirs
Bug: 159726468
Test: m libeffects
Change-Id: I63782bfc073b64ed9b9cd5993d9f2c1a6b938672
2020-06-23 22:40:38 +00:00
cc57b935cd Make video quality analyzer compatible with real SFU in the network
Bug: webrtc:11557
Change-Id: I8ab1fb0896e267f30856a45df6099bd9aae9bc03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174801
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31216}
2020-05-11 18:54:33 +00:00
baa2c836ba Introduce ability to set peer name for PC level tests
Add peer's name to params and use it for logging and metrics naming
for whole peer related metrics.

Bug: webrtc:11479
Change-Id: Ia7e3fc4839c90a958d66910614515ac02a96e389
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174752
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31215}
2020-05-11 18:47:03 +00:00
bf46cfef22 Refactors send rate statistics in RtpSenderEgress
When FEC generation is moved to egress, we'll need to poll bitrates from
there instead of the RtpVideoSender. In preparation, refactoring some
getter methods.

For context, see https://webrtc-review.googlesource.com/c/src/+/173708

Bug: webrtc:11340
Change-Id: Ibc27362361ee9640d9fce676fc8e1093a579344f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174202
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31214}
2020-05-11 17:14:33 +00:00
8e321cd690 [Adaptation] Make QuailtyScalerResourse to report underuse if quality scaling is off
Bug: chromium:1080789
Change-Id: I3aefb746fd6f4adae4b32db322af6b787e8ede1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174804
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31213}
2020-05-11 14:10:08 +00:00
a270250426 [Adaptation] Disable inital frame drop for simuclast/svc
Bug: chromium:1080789
Change-Id: I72bbee4ac21302d15b6c54abea48d665e8ef6922
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174808
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31212}
2020-05-11 13:38:42 +00:00
6efc14b33d VideoTrackSourceInterface: make some newly introduced methods pure virtual.
Bug: webrtc:11114
Change-Id: Ic4d3835ae84b6a652c49f30a9c275870bbf3dacf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174440
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31211}
2020-05-11 12:28:32 +00:00
a54ba4c02e Make video_loopback work with av1
Bug: webrtc:11404
Change-Id: Id4fb4ac7e545df2e4f0a0d91b3531074ff77c9f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172340
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31210}
2020-05-11 12:20:09 +00:00
3a65dba926 Revert "Removes lock release in PacedSender callback."
This reverts commit 6b9c60b06d04bc519195fca1f621b10accfeb46b.

Reason for revert: Breaks downstream test

Original change's description:
> Removes lock release in PacedSender callback.
> 
> The PacedSender currently has logic to temporarily release its internal
> lock while sending or asking for padding.
> This creates some tricky situations in the pacing controller where we
> need to consider if some thread can enter while we the process thread is
> actually processing, just temporarily busy sending.
> 
> Since the pacing call stack is no longer cyclic, we can actually remove
> this lock-release now.
> 
> Bug: webrtc:10809
> Change-Id: Ic59c605252bed1f96a03406c908a30cd1012f995
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173592
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31206}

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

Change-Id: Ic84eee6097528d0792e3b1f90f36bc78447a0d81
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10809
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174820
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31209}
2020-05-11 11:37:57 +00:00
ffd0a844b2 Handle OnRttUpdate in ReceiveStatisticsProxy.
Bug: webrtc:11490
Change-Id: Iba76f77ac1d73350810508f52293e4848f2f6f46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174300
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31208}
2020-05-11 10:55:52 +00:00
09e9a83d91 Change the way that AecDumps are created in APM
This CL changes the way that AecDumps are created in APM. Instead
of being injected, they are now created via the API.

This removes the AecDumpFactory from the API surface of APM and
makes the API more explicit.

The CL will be followed by one more CL that deprecates the usage
of the AttachAecDump API also within the audio_processing
and the fuzzer folders.

The CL also moves the aec_dump.* files from the include folder
to the aec_dump folder and changes the build files. The reasons
for this are that
1) The content of aec_dump.h is not really part of the API
   surface of APM.
2) Those files anyway needed to be moved to a separate build-
   target to avoid a circular build-file dependency caused by
   the other changes in this CL

Bug: webrtc:5298
Change-Id: I7dd6b49de76eb44158472874e1d4ae17dca9be54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174750
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31207}
2020-05-11 10:33:00 +00:00
6b9c60b06d Removes lock release in PacedSender callback.
The PacedSender currently has logic to temporarily release its internal
lock while sending or asking for padding.
This creates some tricky situations in the pacing controller where we
need to consider if some thread can enter while we the process thread is
actually processing, just temporarily busy sending.

Since the pacing call stack is no longer cyclic, we can actually remove
this lock-release now.

Bug: webrtc:10809
Change-Id: Ic59c605252bed1f96a03406c908a30cd1012f995
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173592
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31206}
2020-05-11 09:14:37 +00:00
cc8c07895d Disable PipeWire on Chromecast builds.
For various reasons is_desktop_linux is true on Chromecast builds though
arguably it should not be. This means that the detection logic previously
used is incorrect for Chromecast builds. Since Chromecast needs to
start enabling use_sysroot, this logic needs to explicitly exclude
is_chromecast.

Bug: b/154635846
Change-Id: I6ced6f7e4c78f9d8d7055018e68090883b9e21bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174620
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31205}
2020-05-11 08:44:47 +00:00
1c33075257 Trigger bots again.
TBR=mbonadei@webrtc.org,tommi@webrtc.org

Change-Id: Ia2bf9447c5352ef5999eeab973a23aed8c77d854
Bug: None
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174800
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31204}
2020-05-11 05:38:59 +00:00
822a874463 Switch CallStats to TQ interface + callbacks on the worker thread.
Bug: webrtc:11489
Change-Id: I08c4cd42dfa28d88ed9f0aa8c8b2cfb606bf00df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174240
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31203}
2020-05-10 23:24:35 +00:00
674b0c8111 Move ReceiveStatisticsProxy stats variables to the worker thread.
This reduces locking on the decoder thread and moves all stats
management to the worker thread, which also avoids contention between
querying for these stats and the threads where the media processing happens..

Bug: webrtc:11489,webrtc:11490
Change-Id: I802577eab6b48edcbe124c02a1b793a640b74181
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174205
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31202}
2020-05-10 20:43:40 +00:00
d93bf127cd Call OnDecodedFrame asynchronously on the worker thread.
This offloads the decoder thread with managing histograms,
moves the management over to the thread on which they're queried.
This will allow us to remove more locking from the decoder threads
and avoid contention when querying for stats.

Bug: webrtc:11489
Change-Id: I563c90a0ed01e0b3598ee314d8118622216a2e0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174201
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31201}
2020-05-10 19:35:40 +00:00
ad84d0254a Remove locking from RtpStreamsSynchronizer.
Remove dependency on ProcessThread.

Instead RtpStreamsSynchronizer uses the worker thread
and makes callbacks on the same thread. That in turn
simplifies locking for VideoReceiveStream2, which we'll
take advantage of later.

Bug: webrtc:11489
Change-Id: Id9a5a7977771b92e420a09cc472cfb43de5627cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174221
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31200}
2020-05-10 18:11:44 +00:00
d7e08c8cf8 Move processing of frame meta data for OnFrame/OnRenderedFrame to the worker thread
Bug: webrtc:11489
Change-Id: I9f88fec0aef449fd8923c5eec81cddf9ee42316b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174220
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31199}
2020-05-10 11:47:52 +00:00
67ecb68fba Trigger bots
No-Try: True
Bug: None
Change-Id: Ic86f9063e7f82ab781e463face3647dbd3c2a9ce
Tbr: mbonadei@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174761
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31198}
2020-05-10 09:34:42 +00:00
6a871d3487 Revert "Remove playout delay lock."
This reverts commit c623495fd1ff90aada0eb625af91ec17843fefd0.

Reason for revert: Need to look into failure in remoting_unittests in Chrome (Webrtc/ConnectionTest.SecondCaptureFailed/0). It looks like the order FrameBuffer2 calls into VCMTiming while receiving frames and updating playout delay values, needs to be synchronized better.

Original change's description:
> Remove playout delay lock.
> Now update the playout delay and related stats on the worker thread.
> 
> This was previously reviewed here:
> https://webrtc-review.googlesource.com/c/src/+/172929/
> 
> With the exception of reducing unnecessarily broad
> lock scope in one function in rtp_rtcp_impl.cc
> and added comments in rtp_streams_synchronizer.h
> 
> Bug: webrtc:11489
> Change-Id: I77807b5da2accfe774255d9409542d358f288993
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174200
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31193}

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

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

Bug: webrtc:11489
Change-Id: I9149025d2fc10686314e6d4e89d1b92125650c36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174757
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31197}
2020-05-09 21:30:40 +00:00
3580706684 Add a RunLoop to RtpReplayer to fix fuzzers
Bug: chromium:1080852
Change-Id: Ia02511cde09994deee222e4f1267d5265d0364ca
Tbr: mbonadei@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174756
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31196}
2020-05-09 06:45:14 +00:00
fc11519c92 Cleanup mocks in api/test
Modernise functions to unified MOCK_METHOD macro,
delete few deprecated functions on the way.
add one missing function (in MockEncodedImageCallback)
Remove proxy mock function (in MockVideoBitrateAllocatorFactory)

Remove default constructors and destructors

Bug: None
Change-Id: Ibebb0d9e3c9be5877649af7bde8b87222ddf04fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174751
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31195}
2020-05-08 20:01:03 +00:00
74ef940d79 Stop pulling binutils from WebRTC DEPS.
TBR: titovartem@webrtc.org
Bug: None
Change-Id: If417a7c9dc952325076a5d75f38ac8e984285f9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174755
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31194}
2020-05-08 19:06:36 +00:00
c623495fd1 Remove playout delay lock.
Now update the playout delay and related stats on the worker thread.

This was previously reviewed here:
https://webrtc-review.googlesource.com/c/src/+/172929/

With the exception of reducing unnecessarily broad
lock scope in one function in rtp_rtcp_impl.cc
and added comments in rtp_streams_synchronizer.h

Bug: webrtc:11489
Change-Id: I77807b5da2accfe774255d9409542d358f288993
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174200
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31193}
2020-05-08 19:02:36 +00:00
33d81a05eb Keep OpenH264 iMaxBitrate unspecified.
Max encoder bitrate in WebRTC and OpenH264 are different settings. In
WebRTC it is a cap for encoder target bitrate whilst in OpenH264 it is
a peak bitrate. I.e. OpenH264 is allowed to produce bitrate up to
iMaxBitrate for short time interval. That is not what WebRTC expects.

https://webrtc.googlesource.com/src/+/5ee6967c4edc667688d736c27db6f2e7be00dd0a
disabled encoders re-initialization on min/max bitrate change. Reinit of
some HW encoders takes hundreds of milliseconds and causes video freeze.
I missed that max bitrate is used by OpenH264. This caused regression
described in webrtc:11543.

This change sets iMaxBitrate=UNSPECIFIED_BIT_RATE (which is the default
value). Settings iMaxBitrate=UNSPECIFIED_BIT_RATE disables the frame
dropping logic based on that parameter. But the encoder still will drop
frames based on buffer fullness, https://source.chromium.org/chromium/chromium/src/+/master:third_party/openh264/src/codec/encoder/core/src/ratectl.cpp;l=806-807

Bug: webrtc:10773, webrtc:11543
Change-Id: I728be49e0df8a0d9a8f4438299e4c7b4c1497a78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174745
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31192}
2020-05-08 15:10:26 +00:00
2454d85bb6 Cleanup rtp_rtcp mocks
Modernise function to unified MOCK_METHOD macro, delete few deprecated functions on the way.
Remove default constructors to stress they do nothing special

Bug: None
Change-Id: Ie126f38f0589acb65886f25f754ca575c17af29b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174583
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31191}
2020-05-08 13:43:15 +00:00
804393b369 Removing incorrect DCHECK - breaks android
Bug: webrtc:11489
Change-Id: Ied9ea3095ebe6e42b2be05902b23be306037abbb
NoTry: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174749
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31190}
2020-05-08 12:29:03 +00:00
28da36a6ea Add unittest for av1 wrappers to test Encode and Decode functions
while helpful by itself, it is also a preparation
for adding unittests for (to be added) svc features of the encoder.

Bug: webrtc:11404
Change-Id: I62b0645f44579f21f228d406a206b4c01d80dd02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174580
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31189}
2020-05-08 11:57:27 +00:00
dcde85c912 Pass PeerConfigurerImpl directly into CreateTestPeer
Bug: webrtc:11479
Change-Id: Ib514d264bfd94d648d90a053554537880bd9ebe5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174747
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31188}
2020-05-08 10:56:40 +00:00
435fb9ad06 Remove screen_share_config from the VideoConfig.
After the migration of the pc framework tests (https://webrtc-review.googlesource.com/c/src/+/174023), having "absl::optional<ScreenShareConfig> screen_share_config" field in VideoConfig became redundant. Replaced it with VideoTrackInterface::ContentHint content_hint field.

Bug: webrtc:11534
Change-Id: Ibf4b1c8daed95ef02111fe952171f11e290905d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174702
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31187}
2020-05-08 08:56:13 +00:00