3776 Commits

Author SHA1 Message Date
Qingsi Wang
1c1b99e30f Revert "Delete RtpPayloadParams::SetDependenciesVp8Deprecated as unused"
This reverts commit dc7fe40f497179721e53af1b3ece37c741bb757e.

Reason for revert: speculative revert for breaking downstream projects

Original change's description:
> Delete RtpPayloadParams::SetDependenciesVp8Deprecated as unused
> 
> Bug: webrtc:10242
> Change-Id: Iddad086d8ce3652bd9f0fb12788d5c73b5ebda76
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161945
> Reviewed-by: Niels Moller <nisse@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@{#30159}

TBR=danilchap@webrtc.org,eladalon@webrtc.org,nisse@webrtc.org,philipel@webrtc.org

Change-Id: Ie7f875291610a7b676539a5ccc4bac9a08011f42
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10242
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165240
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30173}
2020-01-07 19:16:48 +00:00
Niels Möller
0aa7e37363 Add include of <cstdlib>
Needed since opus_interface.cc uses the C functions calloc and free.

Bug: None
Change-Id: Iad30be533d7f6d8234c8e49efd461dc6ce0e2442
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164534
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30168}
2020-01-07 14:46:03 +00:00
Danil Chapovalov
27f4d325ad Add VideoRtpDepacketizerGeneric
Bug: webrtc:11152
Change-Id: I27d6a62093d36a4e77dd35d4c115af8cdcc0178a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162202
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30160}
2020-01-07 09:27:34 +00:00
Danil Chapovalov
dc7fe40f49 Delete RtpPayloadParams::SetDependenciesVp8Deprecated as unused
Bug: webrtc:10242
Change-Id: Iddad086d8ce3652bd9f0fb12788d5c73b5ebda76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161945
Reviewed-by: Niels Moller <nisse@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@{#30159}
2020-01-07 09:13:29 +00:00
Yves Gerey
eb3beb8504 Revert "Replace the ExperimentalAgc config with the new config format"
This reverts commit f3aa6326b8e21f627b9fba72040122723251999b.

Reason for revert: Breaks downstream project.

Original change's description:
> Replace the ExperimentalAgc config with the new config format
> 
> This CL replaces the use of the ExperimentalAgc config with
> using the new config format.
> 
> Beyond that, some further changes were made to how the analog
> and digital AGCs are initialized/called. While these can be
> made in a separate CL, I believe the code changes becomes more
> clear by bundling those with the replacement of the
> ExperimentalAgc config.
> 
> TBR: saza@webrtc.org
> Bug: webrtc:5298
> Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30149}

TBR=saza@webrtc.org,peah@webrtc.org

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

Bug: webrtc:5298
Change-Id: I794d2ab4b8caa5330c5ad490ba604646a249a1c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164530
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30153}
2020-01-07 05:22:01 +00:00
Per Åhgren
f3aa6326b8 Replace the ExperimentalAgc config with the new config format
This CL replaces the use of the ExperimentalAgc config with
using the new config format.

Beyond that, some further changes were made to how the analog
and digital AGCs are initialized/called. While these can be
made in a separate CL, I believe the code changes becomes more
clear by bundling those with the replacement of the
ExperimentalAgc config.

TBR: saza@webrtc.org
Bug: webrtc:5298
Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30149}
2020-01-03 23:14:13 +00:00
Sam Zackrisson
12e319aafe Merge the preambles of the ProcessStream implementations
The two functions have a lot of shared logic and locking. This CL consolidates that into a single function.

Bug: webrtc:111235
Change-Id: Ib1c32165dbf0e212c7d4b0753bcbb5ffd05eb6fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163022
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30144}
2020-01-03 14:37:41 +00:00
Per Åhgren
0f14db22de Reduce for reallocations the pre-amplifier and high-pass filter
This CL ensures that the pre-amplifier and the high-pass filter
submodules are not reallocated more than needed.

Bug: webrtc:5298
Change-Id: I7ed23807d4d2d9fef0eda2e7dca9de9b0b1a4649
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163988
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30143}
2020-01-03 14:10:21 +00:00
Sam Zackrisson
308bc646e0 Remove one acquisition of capture lock in APM AudioFrame API
This brings the two ProcessStream functions closer in implementation.
Additionally, the error checking that is currently done in the period of not holding the lock seems cheaper than releasing and reacquiring the capture lock.

Bug: webrtc:11235
Change-Id: Ib4afc68afb419fcabbb8cf08a3a2e61d2c12acda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163021
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30140}
2020-01-03 10:56:24 +00:00
Per Åhgren
2bd85ab039 Avoid AGC2 runtime allocation and activate it on demand
This CL ensures that the AGC2 is created and initialized only when
needed.

Apart from that, the CL also avoids a runtime-reallocation that happens
each time the setting is applied.

Bug: webrtc:5298
Change-Id: Iad9eaa05a3d0baa0788cd11b2aa17ddd8e0c509b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163987
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30139}
2020-01-03 10:47:14 +00:00
Per Åhgren
c0734715d1 APM: Move the TransientSuppression activation to the apm_config
This CL moves the activation of the transient suppression to the APM
config.

Bug: webrtc:5298
Change-Id: Iba7975bec4654c3df8834fd5b7d1082ff53641dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163985
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30137}
2020-01-03 08:48:54 +00:00
Natalie Chouinard
65bbcabe2f [Android] Replace java_files with sources
Replace all usages of java_files with sources in gn files, and
automatically format.

This is in preparation for java_files being completely removed upstream
in favor of sources.

NOPRESUBMIT=true

Bug: chromium:1035074
Change-Id: Ib9a698740b7ad26a127031d90321c7ae2feb59bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163161
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Natalie Chouinard <chouinard@google.com>
Cr-Commit-Position: refs/heads/master@{#30135}
2020-01-02 20:08:20 +00:00
Per Åhgren
29fec66c77 AEC3: Remove metrics that are not used for analysis
Bug: webrtc:8671
Change-Id: I12a6584a70e2b56e0926c07999c919272499c255
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163981
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30134}
2020-01-02 16:23:43 +00:00
Per Åhgren
cf4c872dbd APM: Make the GetStatistics call independent of the locks in APM
This CL changes the GetStatistics call in the audio processing module
(APM) to not aquire the render or capture locks in APM, while still
being thread-safe.
This change eliminates the risk of thread-priority inversion due to the
GetStatistics call.

Apart from the above the CL:
-Corrects the GetStatistics to not be const (it was const even though it
 aquired locks).
-Slightly changes the statistics reporting, so that the stats received
may be older than the most recent stats reported.

Bug: webrtc:11241
Change-Id: I00deb5507e004cbe6e4a19a8bad357491f86f4ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163982
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30131}
2020-01-02 15:45:14 +00:00
Jiwon Kim
077ee35774 Remove unused parameter in RtpFragmentize
Bug: None
Change-Id: Ic110e3561bc93cb2156240193bc2077e2646ed87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161560
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30118}
2019-12-20 11:22:33 +00:00
Per Åhgren
2e8e1c699e Open up for do the noise suppressor analysis on the linear AEC output
This CL allows the noise suppressor to use the linear AEC output
for analysis whenever that is available. This will potentially
lower the risk that the noise suppressor estimates the noise
based on echo.
The feature is off by default.

Bug: webrtc:5298,b:132164318
Change-Id: Idc6c8e197d96209d213819d87a8fb2533b7303ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162900
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30116}
2019-12-20 09:28:01 +00:00
Per Åhgren
9136abb45a AEC3: Ensure that the data size in the reverb computer is not fixed
This CL ensures that the no data vectors in the reverb computer code
are fixed. This allows arbitrary long filters to be used, and ensures
that a minimum required heap size is used.

Bug: webrtc:8671
Change-Id: I7085ed262a3f5965d796270434b6578f4030606e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162661
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30115}
2019-12-19 16:35:56 +00:00
Rasmus Brandt
5cad55b240 Signal requested resolution alignment requirements from sinks to sources.
Bug: webrtc:11218
Change-Id: I593b0515ea389bece472234a3c4082ccc5321ea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162400
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30113}
2019-12-19 10:39:04 +00:00
Per Åhgren
c04242548c Make the high-pass filter operate in full-band
This CL moves the high-pass filter to run in the full-band domain
instead of the split-band domain.

Bug: webrtc:11193
Change-Id: Ie61f4a80afda11236ecbb1ad544bbd0350c7bbfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161453
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30112}
2019-12-18 16:01:24 +00:00
Erik Språng
ae10029bff Prevents probing while paused.
The pacing controller allowed sending bitrate probes, despite it being
paused. This CL adresses that, and makes sure the task-queue based mode
also properly repsects pausing.

Bug: webrtc:10809
Change-Id: I79643c9a24666110d7583fce3ed1bfd6865e9e10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162520
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30109}
2019-12-17 17:57:38 +00:00
Ilya Nikolaevskiy
00a1bcb441 Ensure that unset capture timestamp wouldn't cause incorrect SR rtp timestamps
If for some reason capture timestamp is unset, the default value of 0 would be
passed to RtcpSender. This will cause rtp timestamps to grow at double the rate
in Sender Reports because it has time since the last frame capture as a term.

Bug: none
Change-Id: I2fe09dabef6b0957fb504deaa06393dedc4a9e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162481
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30105}
2019-12-17 12:03:24 +00:00
Björn Terelius
3a8df884d1 Add field trial to avoid extra backoffs in AIMD rate control.
Bug: None
Change-Id: Iaa7dd0ffd6cfabb933e8e68a002b5432d13b9aab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161946
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30103}
2019-12-16 18:01:20 +00:00
Danil Chapovalov
32fe4ef967 Move vp9 rtp depacketization to VideoRtpDepacketizerVp9
Bug: webrtc:11152
Change-Id: I560d4cd62fabae093e3df592f0e7cc4001c10657
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162420
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30102}
2019-12-16 17:11:13 +00:00
Artem Titov
8525a8028a Add ability to resize buffers pool in decoder and use it in IVF generator
Bug: webrtc:10138
Change-Id: I452f08f1d9af57de789bd947a1fcb95536845f80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162183
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30098}
2019-12-16 14:51:16 +00:00
Raman Budny
5331079132 Protect against assigning current_offset_ negative value.
Bug: webrtc:11176
Change-Id: Ic3937da6f1ee9cd118372693cb71d70beb43159c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161329
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30096}
2019-12-16 13:06:52 +00:00
Patrik Höglund
73eb784676 Don't crash the test process when X11 isn't available.
It's not great to use asserts in util functions like this because it
breaks the arrange-act-assert rule, but using checks is worse because
they will crash the test process on failure (= no other tests get run
after that).

Bug: b/143587130
Change-Id: If4d085311de0792b9fca1584db299fd24199e72e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162360
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30093}
2019-12-16 09:57:59 +00:00
Per Åhgren
95059e0779 Moved the legacy noise suppressor to a separate build target
Bug: webrtc:5298
Change-Id: Ia1c5eb9d0f7b4ba578acb646e73229de63ae91fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161441
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30089}
2019-12-13 15:26:16 +00:00
Danil Chapovalov
eae6896f76 Move vp8 rtp depacketization to VideoRtpDepacketizerVp8
Bug: webrtc:11152
Change-Id: Ic2b7fd091cb4d095ce29fbe06196f6424c08fce1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161451
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30088}
2019-12-13 15:10:46 +00:00
Sebastian Jansson
41466b7bef Revert "Extracts ssrc based feedback tracking from feedback adapter."
This reverts commit 08c46adc1e9f9a8d74357fe132a68906ae6e6974.

Reason for revert: Incomplete.

Original change's description:
> Extracts ssrc based feedback tracking from feedback adapter.
> 
> This prepares for moving TransportFeedbackAdapter to TaskQueue.
> 
> Bug: webrtc:9883
> Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30076}

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

Change-Id: I6a79e7627f9de2d8c876d6a13ca36f3ac06fde7f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162200
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30087}
2019-12-13 14:47:48 +00:00
Danil Chapovalov
c9e532a7eb Fix PacketBuffer::LastReceivedKeyframePacketMs
to return time of the last receieved packet of a key frame rather than
last received first packet of a key frame.

To match VideoReceiveStream expectation and prevent requesting
a new key frame if a large key frame is currently on the way.

Bug: None
Change-Id: I443a60872a3580d324f050080a9868f7b90d71a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161730
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@{#30084}
2019-12-13 11:36:24 +00:00
Sebastian Jansson
5e9cac984f Don't try to resend packets that were removed out of order.
Bug: webrtc:11206
Change-Id: Iae05e1db80afd871d37aca203e17bad40dbc9522
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162041
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30083}
2019-12-13 10:29:49 +00:00
Jerome Humbert
9338bbcd90 Replace assert() with RTC_DCHECK
Remove some uses of assert() breaking MSVC compiling, use RTC_DCHECK
instead.

Bug: webrtc:11201
Change-Id: Ie6c3607e422ea17d3393352b4915da3fa24779f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161949
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30081}
2019-12-13 10:06:07 +00:00
Patrik Höglund
0808a8c690 Explicitly set and use WEBRTC_USE_X11 instead of USE_X11.
USE_X11 is magically set by the toolchain. Let's tie this to
whether X11 extensions is on or not. WEBRTC_USE_X11 is used
in the audio device module so let's be consistent with that.

Bug: b/143587130
Change-Id: I15b1744c9acb534dbdf2119deb2dc5d17aa04184
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161901
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30080}
2019-12-13 09:36:25 +00:00
Harald Alvestrand
977b265702 Reduce some logging at INFO level by moving log statements
from LS_INFO to LS_VERBOSE.

By default, unit tests run with logging at info level.
A random run today produced more than 70.000 lines of
output. This CL would reduce that by approximately 15.000.

Bug: none
Change-Id: Ie62708cebf109510a2443aa5ab5c4e645ffc6707
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161950
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30077}
2019-12-12 21:54:06 +00:00
Sebastian Jansson
08c46adc1e Extracts ssrc based feedback tracking from feedback adapter.
This prepares for moving TransportFeedbackAdapter to TaskQueue.

Bug: webrtc:9883
Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30076}
2019-12-12 18:25:25 +00:00
Erik Språng
1e51a388bc Makes padding prefer video SSRCs instead of audio.
Some clients will not count audio packets into the bandwidth estimate
despite negotiating e.g. abs-send-time for that SSRC.
If padding is sent on such an RTP module, we might get stuck in a low
resolution.

This CL works around that by preferring to send padding on video SSRCs.

Bug: webrtc:11196
Change-Id: I1ff503a31a85bc32315006a4f15f8b08e5d4e883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161941
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30066}
2019-12-11 16:32:14 +00:00
Sam Zackrisson
184da528a7 Fix ASAN bug in iSAC codec
There is a value missing in an array, causing indexing to sometimes be off by one.
The result is never used anyway, so the "solution" is to insert an extra zero at the end.

Details:

In an old refactoring CL, the corresponding array in the floating point version of iSAC is declared with 578 elements, the fixed point version has 577.
See WebRtcIsac_kQKltCodeLenShape:
https://webrtc-codereview.appspot.com/548004/patch/10002/11005
Perhaps the entire array was not carried over to fixed point.

However, the floating point version does not actually have 578 elements in its definition, it has 568. The remaining 10 elements are initialized to 0. So there is no value to use from there, and using a zero instead will mimic the behavior of the floating point code.


Use of the value:
WebRtcIsacfix_kCodeLenShapeQ11 is only accessed in one place [0].
The value sumQQ is set to is only used to compute the BitsQQ,
which is only used to populate the sizeQ11 param of EstCodeLpcCoef,
which is never used anywhere. Not within EstCodeLpcCoef, not within the calling function WebRtcIsacfix_EncodeLpc [1], not in the calling function's calling function [2]. Then it falls out of scope.

[0] https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c?l=1223&rcl=3c4fda2ed8140a330634c0a2704a3cf0311a89de
[1] https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c?l=1468&rcl=3c4fda2ed8140a330634c0a2704a3cf0311a89de
[2] https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c?l=214&rcl=3c4fda2ed8140a330634c0a2704a3cf0311a89de

Tested: Instructions in linked bug. No longer repro after applying this CL.
Bug: webrtc:10584
Change-Id: I87a90e8cf58d82f615217e05ea401d3259887769
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161942
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30065}
2019-12-11 15:59:14 +00:00
Jonas Olsson
c907d4f223 Revert "Ensure loss-based controller is always enabled."
This reverts commit 60ec3703cd1f87081c0e4becde5d9ef210a6d44a.

Reason for revert: Needs back-end test before always enabling.

Original change's description:
> Ensure loss-based controller is always enabled.
> 
> The new default parameters are the ones that were used in the Chrome
> Finch trial. The deleted unit test is invalidated by these changes.
> 
> Bug: chromium:941413
> Change-Id: I597f4b0defaebe5bb3a6710b071fae2ee5c6f461
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160652
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30049}

TBR=srte@webrtc.org,crodbro@webrtc.org,jonasolsson@webrtc.org

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

Bug: chromium:941413
Change-Id: I5da4676ad8be2569ad7eed99e954e0d0b624110b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161902
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30061}
2019-12-11 14:09:20 +00:00
Danil Chapovalov
a3ecb7a656 Migrate tests from RtpDepacketizer to VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: I1b1c5183d35b791c09c14c9d1f0ca240c1749d9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161455
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30055}
2019-12-10 17:37:46 +00:00
Mirko Bonadei
f18f9206e5 Revert "Moves TransportFeedbackAdapter to TaskQueue."
This reverts commit 62d01cde6f6ec1fa91b1e5234a7922ad1a4ce036.

Reason for revert: Causes SIGSEGV in webrtc::RTPSender::BuildRtxPacket.

Original change's description:
> Moves TransportFeedbackAdapter to TaskQueue.
>
> Bug: webrtc:9883
> Change-Id: Id87e281751d98043f4470df5a71d458f4cd654c1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158793
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30037}

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

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

No-Try: True
Bug: webrtc:9883
Change-Id: If54bdb8694144fae3fafbabd72d1ac1198e51aa6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161726
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30052}
2019-12-10 13:51:29 +00:00
Daniel Johansson
89aaedac12 Move audioproc_f to rtc_tools.
The motivation in https://webrtc-review.googlesource.com/c/src/+/32340/3 applies here as well. We
would like to use this tool downstream.

Bug: None
Change-Id: Id5b23f792679ab9c07294bfb8e53119c423044b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161681
Commit-Queue: Daniel Johansson <dajo@webrtc.org>
Reviewed-by: Daniel Johansson <dajo@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30051}
2019-12-10 12:08:56 +00:00
Per Åhgren
ded86c1ad8 Remove remaining settings for using legacy AEC
This CL removes the remaining settings for using the legacy AEC.

It also adds a missing printout of the enforce_high_pass_filtering
parameter in the ToString method.

Bug: webrtc:11165
Change-Id: I58f0861bf1c6cd24bd83f4d3e394653b2fab3d71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161683
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30050}
2019-12-10 10:31:07 +00:00
Jonas Olsson
60ec3703cd Ensure loss-based controller is always enabled.
The new default parameters are the ones that were used in the Chrome
Finch trial. The deleted unit test is invalidated by these changes.

Bug: chromium:941413
Change-Id: I597f4b0defaebe5bb3a6710b071fae2ee5c6f461
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160652
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30049}
2019-12-10 10:30:02 +00:00
Bjorn A Mellem
951e289853 Add VideoTimingExtension to kFecOrPaddingExtensionSizes.
As of https://webrtc-review.googlesource.com/c/src/+/158899, FEC may be
used on packets with VideoTimingExtension.  This may result in creation
of FEC packets that exceed the maximum configured RTP packet size.

This problem occurs most frequently with datagram transports that define a
smaller maximum packet size.

Bug: webrtc:9719
Change-Id: I842216a6696a695f0a3f01a221e538605fc5b9bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161557
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30045}
2019-12-09 18:46:53 +00:00
Ying Wang
ef3998ffd1 Add directive to make webrtc metrics optional.
Bug: webrtc:11144
Change-Id: I4e75e6aec033784685de3670e880bb9f2b6ee8d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30040}
2019-12-09 13:55:50 +00:00
Guido Urdaneta
00d0f178c2 Revert "Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."""
This reverts commit af51be7869994a299451e22e6382ae641767b26d.

Reason for revert: Causes failure of Linxu CFI Chromium bot.
See https://crbug.com/1031930

Original change's description:
> Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5.""
> 
> This is a reland of a0adf3d4409036d095480e9bfa0fc06990362f84
> 
> Original change's description:
> > Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."
> > 
> > This is a reland of e7153012682ccd3d1eacc18f802cab7820e3bad3
> > 
> > Original change's description:
> > > Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5.
> > >
> > > Bug: chromium:396091
> > > Change-Id: Ia1b36c771632c536bb8d15322461b479fabc409e
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148768
> > > Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
> > > Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
> > > Cr-Commit-Position: refs/heads/master@{#29083}
> > 
> > Bug: chromium:396091
> > Change-Id: I0d9171ae5f340e0489e4b45ce5d97bc52b0a4904
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156067
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29655}
> 
> Bug: chromium:396091
> Change-Id: I47525911095fabc6cee613d03b0d83134b95b084
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158900
> Reviewed-by: Tomas Gunnarsson <tommi@chromium.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30032}

TBR=zijiehe@chromium.org,jamiewalch@chromium.org,tommi@webrtc.org,julien.isorce@chromium.org,sergeyu@chromium.org,tommi@chromium.org,trevor.axiom@gmail.com,jonringle@gmail.com,justin.franco@arterys.com

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

Bug: chromium:396091
Change-Id: Ibd7b21ade1547d96f42b3c24860e9f901fc71065
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161458
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30039}
2019-12-09 11:26:20 +00:00
Christoffer Rodbro
034f767a91 Allow setting the initial congestion window size by config.
Bug: webrtc:11148
Change-Id: I4700a261661dca51d769e0a277704e1f9316e83d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161089
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30038}
2019-12-09 11:00:10 +00:00
Sebastian Jansson
62d01cde6f Moves TransportFeedbackAdapter to TaskQueue.
Bug: webrtc:9883
Change-Id: Id87e281751d98043f4470df5a71d458f4cd654c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158793
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30037}
2019-12-09 10:38:54 +00:00
Per Åhgren
62ea0aaea0 Remove deprecated legacy AEC code
This CL removes the deprecated legacy AEC code.

Note that this CL should not be landed before the M80 release has been cut.

Bug: webrtc:11165
Change-Id: I59ee94526e62f702bb9fa9fa2d38c4e48f44753c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161238
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30036}
2019-12-09 10:37:49 +00:00
Artem Titov
33f9d2b383 Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class
Bug: webrtc:10138
Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30033}
2019-12-07 00:54:26 +00:00