3795 Commits

Author SHA1 Message Date
Sebastian Jansson
d61338fa6e Reland "Extracts ssrc based feedback tracking from feedback adapter."
This is a reland of 08c46adc1e9f9a8d74357fe132a68906ae6e6974

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}

Bug: webrtc:9883
Change-Id: Ia74a3b1fba4d83eece9b0eb6475d6e6aecb65700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162201
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30266}
2020-01-15 12:51:16 +00:00
Danil Chapovalov
61d6471912 Change H264 depacketizer to implement VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: If5169f47d85918356fa66e2bf3422d722044aa1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165581
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30264}
2020-01-15 12:26:55 +00:00
Evan Shrubsole
6ef59d1ced Don't pace audio by default
After experimentation, not pacing audio is better. This is controlled
by the field trial WebRTC-Pacer-BlockAudio. This change keeps the flag,
but changes the behaviour such that it defaults to Disabled. However,
audio can still be paced if one chooses by enabling the field trial.

Bug: webrtc:11257
Change-Id: I5b23a82bb6708c007cf8dfb40065c821eefdc4e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165381
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30262}
2020-01-15 11:21:14 +00:00
Danil Chapovalov
d06588a758 Change Av1 depacketizer to implement VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: I322115263f60439bee36277157a0acef9bd28e3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165343
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30260}
2020-01-15 10:16:03 +00:00
Jonas Olsson
b2b2031457 Concatenate string literals at compile time.
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format

After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.

This primary benefit of this change is a small reduction in binary size.

Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
2020-01-14 14:47:48 +00:00
Sam Zackrisson
ecc5b93b13 AEC3: Restrict default logging of some delay changes to VERBOSE
It leads to overly verbose test output. Example:
https://chromium-swarm.appspot.com/task?id=49bc386e0545ef10

Bug: webrtc:11278
Change-Id: I4a1c565f3aab94d98910722b23dcadc5fcde602a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165962
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30249}
2020-01-14 12:52:47 +00:00
Sebastian Jansson
3e66a498c3 Use RTX SSRCs in scenario test framework.
Using RTX SSRCs and payload type for retransmission of video. This
corresponds to the behavior when using the peer connection API.

Bug: webrtc:9883
Change-Id: Ic0e3964d097f42219ca225513a4bc771d70ccaa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164460
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30248}
2020-01-14 12:04:56 +00:00
Jerome Humbert
3e3c551ac6 Suppress C5041 constexpr warning for MSVC 2019
Disable the C5041 warning which makes the build fail. This is a
C++17-only change and WebRTC doesn't support C++17 yet, so the code is
technically correct, but fails to build on MSVC 2019 and
warning-as-error active.

Also fix another warning-as-error build error with MSVC 2019 due to
ignoring the result of a [[nodiscard]] function.

No-Presubmit: True
Bug: webrtc:11275,webrtc:11276
Change-Id: I891a894ee87252f96e84fd8d282576f46907256f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30244}
2020-01-14 07:44:35 +00:00
Danil Chapovalov
7d43801a07 Delete RtpGenericDepacketizer as no longer used
Bug: webrtc:11152
Change-Id: I275765e1aa013d8188d43e2911e8ab022563d1d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165394
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30234}
2020-01-13 13:45:37 +00:00
Sam Zackrisson
48148dc840 Change log level of AEC3 buffer info to VERBOSE
Otherwise, test logs become very verbose:
https://chrome-swarming.appspot.com/task?id=49b6fa6ac93e2310
See linked issue.

Bug: webrtc:11278
Change-Id: I778ee4826de6c1b23d47a5d5ce302d074900ce6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165786
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30233}
2020-01-13 13:31:27 +00:00
Danil Chapovalov
b42aeaa3fb Move RtpDepacketizerH264 into own files
Bug: webrtc:11152
Change-Id: Iab4975e9f378b177a2abf34559f9b74752e69843
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165582
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30212}
2020-01-10 15:33:54 +00:00
Jonas Oreland
350a82aec3 Reland "Add field trial to base stable target rate on loss based target rate"
This is a reland of 63db77007bea78487af05d46b1b46106761556a1 that
was broken as I flipped != and == :(

Luckily this made a test flaky, and hence was the original change reverted.

Original change's description:
> Add field trial to base stable target rate on loss based target rate
>
> I.e not the pushback_rate that includes the congestion window pushback
> (if enabled).
>
> Bug: None
> Change-Id: I413d011004a95da03dd62f5b423abc3c8b66b333
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165383
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30189}

Bug: None
Change-Id: Ia637d0498e6c0c2708eba659e2a30f3235944d4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165391
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30196}
2020-01-09 14:21:07 +00:00
Danil Chapovalov
5c35f2fb1b Delete RtpDepacketizerVp9 in favor of VideoRtpDepacketizerVp9
Bug: webrtc:11152
Change-Id: Ic50f2dc49ca420b3406d4dea11ed20328aa59136
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165382
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30195}
2020-01-09 13:07:44 +00:00
Jonas Oreland
b93a7d7e05 Revert "Add field trial to base stable target rate on loss based target rate"
This reverts commit 63db77007bea78487af05d46b1b46106761556a1.

Reason for revert: Flipped !=which should have been == makes tests

Original change's description:
> Add field trial to base stable target rate on loss based target rate
> 
> I.e not the pushback_rate that includes the congestion window pushback
> (if enabled).
> 
> Bug: None
> Change-Id: I413d011004a95da03dd62f5b423abc3c8b66b333
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165383
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30189}

TBR=brandtr@webrtc.org,srte@webrtc.org,jonaso@webrtc.org

Change-Id: I883edb8a74f1ae2a4d783b9825cc08c6a5228aa9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165388
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30193}
2020-01-09 12:52:06 +00:00
Danil Chapovalov
26e1b7ac01 Delete RtpDepacketizerVp8 in favor of VideoRtpDepacketizerVp8
Bug: webrtc:11152
Change-Id: I1a6225701ecd6f7a34c946d7296f0ab0cbb5eaef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165342
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30190}
2020-01-09 12:10:19 +00:00
Jonas Oreland
63db77007b Add field trial to base stable target rate on loss based target rate
I.e not the pushback_rate that includes the congestion window pushback
(if enabled).

Bug: None
Change-Id: I413d011004a95da03dd62f5b423abc3c8b66b333
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165383
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30189}
2020-01-09 11:32:25 +00:00
Jakob Ivarsson
2ee15eb4fa Remove extra delay field trial.
Bug: webrtc:10817
Change-Id: I704a8ea0dc774f242f8d5d88b140f850cf23d518
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164539
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30182}
2020-01-08 14:39:27 +00:00
Jakob Ivarsson
bd5874accf Remove inter-arrival delay mode from DelayManager.
Also remove the delay peak detector which is no longer used.

This should be a no-op since relative arrival delay mode is used by default.

Bug: webrtc:10333
Change-Id: Ifa326b762d52f16f9dc5f3da2874139faf1022da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164462
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30179}
2020-01-08 13:20:36 +00:00
Danil Chapovalov
57218b4e22 Delete RtpDepacketizer::Create factory
Bug: webrtc:11152
Change-Id: I09824b97506a11f917cd71f2f0d30306538eee13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163023
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30178}
2020-01-08 11:41:06 +00:00
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