- Added field trial to force issuing of key frame on deactivation of
spatial layer. This fixes video corruptions in VP9 K-SVC tests where
layers can be activated/deactivated on-fly due to bandwidth change.
- Added 100ms network delay to the test with restricted link capacity.
This fixes rapid drop of available bandwidth which happens when
bandwidth overuse is detected in the very beginning of call and several
feedback packets arrive without any delay. Also, this makes the test
more realistic.
- Disabled filtering of spatial layer in the test with restricted
link capacity. 1) We don't really need filtering in this test.
2) It appeared that in video quality tests filtering is done before
sending packets to network simulator. Filtering of high layers causes
channel underuse which is compensated by increase of sent bitrate.
This is why we got sent/media bitrates about 2Mbps in test where link
was limited to 1Mbps.
Bug: chromium:889017
Change-Id: I33ffcee0274523f6183c3bbd27d3d29395417d52
Reviewed-on: https://webrtc-review.googlesource.com/c/103520
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24988}
Some important NetEq information was not available in NetEqState, which
meant it was not available on the API. This CL adds additional
information.
Bug: webrtc:9667
Change-Id: I702707c7d60472f488047d48fb286f839c5608dc
Reviewed-on: https://webrtc-review.googlesource.com/c/102300
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24985}
It will soon lose the ability to do so.
Bug: webrtc:8396
Change-Id: If120afa37325c00ae2c3e9a9bd75bf89c8897f8c
Reviewed-on: https://webrtc-review.googlesource.com/c/103441
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24979}
UBSan will trigger when time_stretched_samples overflows using a
big number. This change avoids this problem by storing the
intermediate result into a int64_t.
Bug: chromium:886904
Change-Id: Id09dc4b468f841f03b523d5f21763f610b163a42
Reviewed-on: https://webrtc-review.googlesource.com/c/103123
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Pablo Barrera González <barrerap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24977}
After the removal of field_trial_default, metrics_default and
runtime_enabled_features_default, this build target doesn't build
anything and can be safely removed.
Bug: webrtc:9631
Change-Id: Iee1111e065ffefe0b4b9a695ee67a594e6d82caa
Reviewed-on: https://webrtc-review.googlesource.com/c/103702
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24976}
This reverts commit 9e24dcff167c4eb3555bf0ce6eaba090c10fbe53.
Reason for revert: Breaks chromium.webrtc.fyi bots.
Original change's description:
> Export symbols needed by the Chromium component build (part 1).
>
> This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
> to mark WebRTC symbols as visible from a shared library, this doesn't
> mean these symbols are part of the public API (please continue to refer
> to [1] for info about what is considered public WebRTC API).
>
> [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
>
> Bug: webrtc:9419
> Change-Id: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99
> Reviewed-on: https://webrtc-review.googlesource.com/c/103505
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24969}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
Change-Id: I01f6e18f0d2c0f0309cdaa6c943c3927e1f1f49f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/103720
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24974}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
Bug: webrtc:9419
Change-Id: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99
Reviewed-on: https://webrtc-review.googlesource.com/c/103505
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24969}
The RtpReceiver class is no longer used. Together with it, delete
RTPPayloadRegistry, RtpReceiverStrategy, and the tests under
modules/rtp_rtcp/test/testAPI/.
Bug: webrtc:8995
Change-Id: Ia9924d2f0f4315914a0dce6b7375ebb3601a6f96
Reviewed-on: https://webrtc-review.googlesource.com/c/103503
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24968}
Rename to better match what it does,
Adjust to support two-byte header extension
Bug: webrtc:7990
Change-Id: I2786d70e7cf9cd3d722f54fb1d07c9cfaafab947
Reviewed-on: https://webrtc-review.googlesource.com/103201
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24958}
This refactoring merged PopulateCodecSpecific and FrameEncoded into a
single callback method. It also removes the FrameConfig parameter and
instead relies on the temporal layer to remember it internally.
Bug: webrtc:9012
Change-Id: I489b76821b534398ad452643f1322f411d3455b1
Reviewed-on: https://webrtc-review.googlesource.com/95681
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24957}
This CL adds the ability to toggle the AGC2 adaptive digital mode in
audioproc_f
Bug: webrtc:5298
Change-Id: If1567d8c87f88992dff89253edb293a56cee0a73
Reviewed-on: https://webrtc-review.googlesource.com/c/103361
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24954}
../../third_party/webrtc/modules/pacing/round_robin_packet_queue.h:70:5:
warning: explicitly defaulted default constructor is implicitly deleted
[-Wdefaulted-function-deleted]
StreamPrioKey() = default;
^
../../third_party/webrtc/modules/pacing/round_robin_packet_queue.h:80:37: note:
default constructor of 'StreamPrioKey' is implicitly deleted because field
'priority' of const-qualified type 'const RtpPacketSender::Priority' would not
be initialized
const RtpPacketSender::Priority priority;
^
Bug: chromium:890307
Change-Id: I58f21121fc9083a60ba1ad26492fdca6285d0447
Reviewed-on: https://webrtc-review.googlesource.com/c/103181
Commit-Queue: Nico Weber <thakis@chromium.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24952}
It will soon lose the ability to do so.
Bug: webrtc:8396
Change-Id: I1d7c62fbc2585233cf1656fdcc4bb5380c2f41a5
Reviewed-on: https://webrtc-review.googlesource.com/c/100980
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24947}
This is a reland of 2fbb83b16b4c2c1712cbe898ca3ba42d6da3e96f
Original change's description:
> Remove APM-internal usage of EchoControlMobile
>
> This is a sibling CL to a similar one for EchoCancellation:
> https://webrtc-review.googlesource.com/c/src/+/97603
>
> - EchoControlMobileImpl will no longer inherit EchoControlMobile.
> - Removes usage of AudioProcessing::echo_control_mobile() inside most of
> the audio processing module and unit tests.
>
> The CL breaks audioproc_f backwards compatibility: It can no longer
> use all recorded settings (comfort noise, routing mode), but prints an
> error message when unsupported settings are encountered.
>
> Tested: audioproc_f with .wav and aecdump inputs.
> Bug: webrtc:9535
> Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
> Reviewed-on: https://webrtc-review.googlesource.com/101621
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24888}
Bug: webrtc:9535
Change-Id: I172706c6729cac4eb6afde1ebd6fc8f3a289d6c7
Reviewed-on: https://webrtc-review.googlesource.com/c/102881
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24943}
This CL corrects the incorrectly implemented check to avoid that AEC3
reacts on the initial pre-amp gain setting.
TBR: devicentepena@webrtc.org
Bug: webrtc:9805
Change-Id: I5decbf00a80457f24b8cd499c35720805ff9ccbc
Reviewed-on: https://webrtc-review.googlesource.com/c/103360
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24938}
In this CL the ERLE estimator is reset after a pre-amplifier gain change is communicated to APM.
Bug: webrtc:9805
Change-Id: I040f344e4607e862240250f9478d06de0d58a096
Reviewed-on: https://webrtc-review.googlesource.com/103222
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24933}
This change does three things:
- Move rtc_json into rtc_base/strings/, a non-API directory more fitting to
its purpose.
- Make a target for the currently unused json_unittest.
- Make the code available for use in non-test code again.
Bug: webrtc:9802
Change-Id: Id964a8a4b47b732a962a364894a4dbd3e7f4650f
Reviewed-on: https://webrtc-review.googlesource.com/103126
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24932}
This CL makes the RtpGenericFrameDescriptor available in
RTPSenderVideo::SendVideo for encryption and in
RtpVideoStreamReceiver::OnReceivedFrame for decryption.
Bug: webrtc:9361
Change-Id: I5b6d10138c0874657862f103c8c9a2328e6d4a66
Reviewed-on: https://webrtc-review.googlesource.com/102720
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24929}
Use in VideoQualityTest replaced by creating a wrapper for the decoder,
similarly to https://webrtc-review.googlesource.com/94152 which
deleted the corresponding method on VideoSendStream.
Bug: webrtc:9106
Change-Id: I0a7798bc44704af8b36017655b9ffa34fa1423e6
Reviewed-on: https://webrtc-review.googlesource.com/97580
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24926}
to extract byte representation of a built extension without rebuilding it.
Bug: webrtc:9361
Change-Id: I5e2a5caeb8ff28dcb58dc25d53407c449c86df44
Reviewed-on: https://webrtc-review.googlesource.com/102940
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24925}
Hypothetical scenario: short weak speech at start of call, then high
noise. The digital adaptive AGC2 would pick a high gain, and then
continue to apply it on the noise. Unless the noise is detected by the
noise estimator, the gain would never be reduced.
This CL addresses the issue by sending limiter gain info to the
adaptive digital AGC2.
Bug: webrtc:7494
Change-Id: Idf5c2686af0f5e5bad981d39a95b8efc9ffb9d64
Reviewed-on: https://webrtc-review.googlesource.com/102641
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24922}
It will soon lose the ability to do so.
Bug: webrtc:8396
Change-Id: I06dce417bba855b57130bd1a052988b2f235dcbd
Reviewed-on: https://webrtc-review.googlesource.com/102882
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24921}
This CL is a step towards making the TemporalLayers landable in api/ :
* It splits TemporalLayers from TemporalLayersChecker
* It initially renames temporal_layer.h to vp8_temporal_layers.h and
moved it into the include/ folder
* It removes the dependency on VideoCodec, which was essentially only
used to determine if screenshare_layers or default_temporal_layers
should be used, and the number of temporal temporal layers to use.
Subsequent CLs will make further cleanup before attempting a move to api
Bug: webrtc:9012
Change-Id: I87ea7aac66d39284eaebd86aa9d015aba2eaaaea
Reviewed-on: https://webrtc-review.googlesource.com/94156
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24920}
In the case a frame_object is kept for some time before it is deleted,
it may happend that a new frame is received with overlapping sequence
numbers. If the old frame_object is removed while receiving the new
frame there used to be a crash.
Bug: webrtc:9629
Change-Id: I270a8caa2b58b73c000542aa504c0ebe277d49c4
Reviewed-on: https://webrtc-review.googlesource.com/102683
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24914}
This CL removes kNumMicrosecsPerSec and kNumMillisecsPerSec from
modules/audio_device/win/core_audio_utility_win.h.
kNumMillisecsPerSec was unused, while kNumMicrosecsPerSec has been
replaced by rtc::kNumMicrosecsPerSec.
Bug: webrtc:9693
Change-Id: I560aa9dad2bfb94a9bf67d3b9941700f1948086b
Reviewed-on: https://webrtc-review.googlesource.com/102860
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24906}
When bitrate is allocated to streams that does not have packet feedback,
the allocated bitrate should be included in the estimate. This was
previously only implemented for the old congestion controller and not
for the new task queue based version.
To make the behavior more robust, the responsibility for tracking this
is moved to BitrateAllocator where it's handled consistently for
multiple streams without feedback.
Bug: webrtc:9586, webrtc:8243
Change-Id: I8af7fec23e1bdc08cc61cf1b4ff10461c3711fb0
Reviewed-on: https://webrtc-review.googlesource.com/102681
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24905}
This features is not needed anymore, with this CL it is also possible
to address two issues:
- The need to pick a default implementation.
- The need to use -Wno-global-constructors.
Bug: webrtc:9631, webrtc:9693
Change-Id: Id3daf34179fbc8db26969fc701ccbfa7182c6a9b
Reviewed-on: https://webrtc-review.googlesource.com/102543
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24904}
This reverts commit 2fbb83b16b4c2c1712cbe898ca3ba42d6da3e96f.
Reason for revert: Speculative revert over failing Chromium bot:
https://ci.chromium.org/p/chromium/builders/luci.chromium.webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28M%20Nexus5X%29/117
Original change's description:
> Remove APM-internal usage of EchoControlMobile
>
> This is a sibling CL to a similar one for EchoCancellation:
> https://webrtc-review.googlesource.com/c/src/+/97603
>
> - EchoControlMobileImpl will no longer inherit EchoControlMobile.
> - Removes usage of AudioProcessing::echo_control_mobile() inside most of
> the audio processing module and unit tests.
>
> The CL breaks audioproc_f backwards compatibility: It can no longer
> use all recorded settings (comfort noise, routing mode), but prints an
> error message when unsupported settings are encountered.
>
> Tested: audioproc_f with .wav and aecdump inputs.
> Bug: webrtc:9535
> Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
> Reviewed-on: https://webrtc-review.googlesource.com/101621
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24888}
TBR=saza@webrtc.org,aleloi@webrtc.org
Change-Id: I1f8a27ac291f2cdc16c8daa32e399b74d489dbb9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/102642
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24895}
These submodules implicitly rely on low cut filtering being enabled.
This CL clarifies a distinction:
High pass filtering is a feature that users can enable, according to the WebRTC standard.
Low cut filtering is a processing effect that is applied when any of the following is active:
- high pass filter
- noise suppression
- builtin echo cancellation
Bug: webrtc:9535
Change-Id: I9474276fb11354ea3b01e65a0699f6c29263770b
Reviewed-on: https://webrtc-review.googlesource.com/102600
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24892}
Adds SimulatedTimeClient, a class that simulates time so congestion
controllers can be tested using the Scenario test framework without
running in real time.
This allows using simplified scenario tests as unit tests, narrowing
the gap between end to end tests and unit tests.
Bug: webrtc:9510
Change-Id: I61ab388bd610f636b926675b1f14b8d85e3c1114
Reviewed-on: https://webrtc-review.googlesource.com/99801
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24890}
Add new method OnRtpPacket, but leave
ReceiveStatisticsImpl::IncomingPacket and most of the implementation
unchanged. Deleting the old method and converting implementation from
RTPHeader to RtpPacketreceived is planned for a followup, after
downstream code is updated.
Bug: webrtc:7135, webrtc:8016
Change-Id: I697ec12804618859f8d69415622d1b957e1d0847
Reviewed-on: https://webrtc-review.googlesource.com/100104
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24889}
This is a sibling CL to a similar one for EchoCancellation:
https://webrtc-review.googlesource.com/c/src/+/97603
- EchoControlMobileImpl will no longer inherit EchoControlMobile.
- Removes usage of AudioProcessing::echo_control_mobile() inside most of
the audio processing module and unit tests.
The CL breaks audioproc_f backwards compatibility: It can no longer
use all recorded settings (comfort noise, routing mode), but prints an
error message when unsupported settings are encountered.
Tested: audioproc_f with .wav and aecdump inputs.
Bug: webrtc:9535
Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
Reviewed-on: https://webrtc-review.googlesource.com/101621
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24888}
