This CL wires up the new SetRates() method of the video encoders, and
refactors a few things in the process:
Most notably, the VideoStreamEncoderInterface is update so that the
|target_headroom| parameter is replaced with |link_allocation|, meaning
that instead of indicating bitrate capacity in excess of the target
bitrate, it indicates to total network capacity allocated for the
stream including the target bitrate. This matches the VideoEncoder API.
The VideoEncoder::RateControlParameters struct gets a few new helper
methods.
In VideoStreamEncoder, instead of adding more fields to the
|last_observed_bitrate*| family, uses an optional struct that
inherits from VideoEncoder::RateControlParameters.
Bug: webrtc:10481
Change-Id: Iee3965531142ae9b964ed86c0d51db59b1cdd61c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131123
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27487}
The DefaultAudioQualityAnalyzer will read stats reports (temporarily
using the old PeerConnectionInterface::GetStats) and for each audio
stream it will collect some NetEq related stats.
When DefaultAudioQualityAnalyzer::Stop is invoked by the framework,
it will report the following metrics:
- expand_rate
- accelerate_rate
- preemptive_rate
- speech_expand_rate
- preferred_buffer_size_ms
Bug: webrtc:10138
Change-Id: Ie493456fcb9ed86455b12dabdab98a317387ef46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125980
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27474}
This change adds FrameDecryptorInterface::Result to the FrameDecryptorInterface
API. Result contains a Status and bytes_written. This removes requiring out
parameters from the API and provides a simpler status return code for the
function. This is in response to comments suggested here:
https://webrtc-review.googlesource.com/c/src/+/131358
int FrameDecryptorInterface::Decrypt() will be removed in a follow up CL.
Bug: webrtc:10512
Change-Id: I47f19f154d1d8430acd6e4a6f433ab24c455fd51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131362
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27470}
This change makes it possible to disable AEC3's render delay
controller and delay estimator, and instead rely on an external
delay estimator. The delay is communicated via SetAudioBufferDelay.
When the feature is enabled, no echo removal will be performed
until the first delay is provided.
The delay is
Bug: b/130016532
Change-Id: I16643109d78d770ff1d2713cf247b0b9cce1bc1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131327
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27467}
This change removes the following unused parameters from the AEC3
configuration:
- render_pre_window_size_init
- render_post_window_size_init
- nonlinear_hold
- nonlinear_release
Bug: webrtc:8671
Change-Id: I8f7a3d350387cd8ada4d507c3a9fab43b7813f5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131321
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27450}
making a step for GlobalTaskQueueFactory to be optional way
to provide TaskQueueFactory
Bug: webrtc:10284
Change-Id: Ife838b3691c256820973118bc5b3cb372dea09cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130488
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27423}
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: Ib2c29054b2ae008f5291bd3b762a504b18534326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27410}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
Some NaCl system headers live in a special directory and the
toolchain doesn't propagate the -I compiler flag [2].
A common workaround in Chromium is to use 'public_deps' in order
to propagate //native_client_sdk/src/libraries/nacl_io:nacl_io_include_dirs
one step further in the build graph.
[1] - https://cs.chromium.org/chromium/src/native_client_sdk/src/libraries/nacl_io/
[2] - -Inative_client_sdk/src/libraries/third_party/newlib-extras
Bug: chromium:925028
Change-Id: I5145b80c2ae6969f79fcbfcf93a6b05c8a122746
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27373}
Chromium requires that all code that waits on a sync primitive be
annotated with ScopedAllowBaseSyncPrimitives(ForTesting). Webrtc
already imports ScopedAllowBaseSyncPrimitives.
ScopedAllowBaseSyncPrimitivesForTesting is equivalent but can only
be used in tests and doesn't required adding a friend declaration to
thread_restrictions.h.
Previously, the code that is annotated with
ScopedAllowBaseSyncPrimitivesForTesting in this CL didn't fail because
it ran on a TaskRunner annotated with the deprecated
WithBaseSyncPrimitives() trait (cf.
https://cs.chromium.org/chromium/src/content/renderer/media/webrtc/task_queue_factory_unittest.cc?l=23&rcl=362f3723ac358d932ea2e3af65512a1243697a31).
Change-Id: Id7cfa2ea108870de86dc887458ae783c807791cc
Bug: chromium:889029
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128823
Commit-Queue: Francois Pierre Doray <fdoray@chromium.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27339}
This CL changes the API for webrtc::VideoEncoder.
There is a legacy method called SetRates(). This is indicated as being
deprecated, but there seem to be a number of usages still left.
Then there is the new SetRateAllocation() method which takes a
VideoBitrateAllocation instance instead of a single target bitrate.
This CL adds a new version of SetRates() which moves all the existing
parameters in a RateControlParameters struct, and adds a bandwidth
allocation signal. The intent of this signal is to allow the encoder
to know how close to the target it needs to stay. If the encoder rate
is network restricted, it will need to be more aggressive in keep the
rate down and possibly drop frames. If there is some margin for
overshoot, it might do different trade-offs.
Furthermore, the frame rate signal is changes from an integer to a
floating point type in order to get more precise rates at low frame
rates and the return type has been changed to void since the only use
of the previous values to log it and that is better done inside encoder
where the error condition originates.
The intent is to properly deprecate the "old" SetRates() /
SetRateAllocation() methods, send out a PSA and then remove them in two
weeks. Changes required by users should be trivial, as using the new
headroom signal is optional.
Bug: webrtc:10155, webrtc:10481
Change-Id: I4f797b0b0c73086111869ef4ee5f42bf530f5fde
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129724
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27314}
- Add GetFrameStatistics API:
This is useful for downstream test users that want to read frame-level stats.
- Remove other APIs that are not used by downstream tests:
* AddFrame
* GetFrame
* GetFrameWithTimestamp
* SliceAndCalcAggregatedVideoStatistic
* PrintFrameStatistics
* Size
* Clear
The implementations, which are used by the fixture implementation, are kept.
Bug: webrtc:10349
Change-Id: Id2f6fa5a36b8341a5ccb365725f71ebe0c0f1570
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128779
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27306}
Bug: None
Change-Id: I87439a234d7018757eb61e99d5c6f9c7be4ab357
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128825
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27272}
Reinterpret will only allow conversions when the underlying types are
fundamental and have the same size.
Bug: None
Change-Id: Id6a4e9784998fe65fb26ab3fd398710c892c4a67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128228
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27249}
This CL paves the way to making FrameBufferController injectable.
LibvpxVp8Encoder can manage multiple streams. Prior to this CL,
each stream had its own frame buffer controller, all of them held
in a vector by LibvpxVp8Encoder. This complicated the code and
produced some code duplication (cf. SetupTemporalLayers).
This CL:
1. Replaces CreateVp8TemporalLayers() by a factory. (Later CLs
will make this factory injectable.)
2. Makes LibvpxVp8Encoder use a single controller. This single
controller will, in the case of multiple streams, delegate
its work to multiple controllers, but that fact is not visible
to LibvpxVp8Encoder.
This CL also squashes CL #126046 (Send notifications of RTT and
PLR changes to Vp8FrameBufferController) into it.
Bug: webrtc:10382
Change-Id: Id9b55734bebb457acc276f34a7a9e52cc19c8eb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126483
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27206}
Previously only reading from the filesystem was supported, this CL
allows parsing an event log from a string.
Bug: webrtc:10337
Change-Id: Iadde3319eb8fb4175625f510201fac9c01c80ed9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127296
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27202}
The target should contain rtp_headers.{cc,h}, but downstream
dependencies must be adjusted before moving the files into the new
target.
Bug: None
Change-Id: Ie8a37c43200463762e2fdaa99d7b49d880298602
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128570
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27200}
WebRTC video engine now configures bitrate on media transport
correctly.
Bug: webrtc:9719
Change-Id: I85884cd76644b7eca3763cec8ce9e31b5b64db27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127941
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27167}
No-Try: True
Bug: None
Change-Id: I411cdb48c9da23e72c575c59ede8f5d140a437e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128223
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27165}
This CL removes parameters for AEC3 which are no longer used. To reflect
that change, one of the parameters also is renamed
Bug: chromium:941949,webrtc:8671
Change-Id: I26609b396fa14ecb7523eebfe531a1338718103b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127780
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27128}
This change reduces the risk of echo due to noise in the headroom
of the linear filter.
Changes:
- Use shorter delay headroom
- Delay headroom is specified in samples (not blocks)
- No hysteresis limit when delay is reduced
Bug: chromium:119942,webrtc:10341
Change-Id: I708e80e26d541dff8ca04b6da2d346a1d59cbfcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126420
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27126}
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.
Bug: webrtc:10410
Change-Id: I20190c01559ff315422be1b3f980853cbc5afbcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127625
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27117}
