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}
Any new capture implementation for windows should use VideoSourceInterface,
like current capturers for android, mac and ios.
Bug: None
Change-Id: I0dffa9fde5ba069aab6273bfef826d17f03877d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131337
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27483}
rtcp_receiver_fuzzer was running over inputs of unreasonable
length, leading to timeouts. RTCP typically runs over UDP.
This CL limits the inputs to a bit over the max UDP payload length.
Bug: chromium:948469
Change-Id: I669a5b24c265bb3b6da2503da109efed32c25182
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131393
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27482}
A reasonable amount of incoming packets could generate feedback
for millions of packets.
Bug: chromium:949020
Change-Id: I7f3e6b75b683af5b2732c472cc92c6788540486b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131333
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27481}
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 is to avoid inconsistent/flaky behavior on mobile bots.
Bug: webrtc:10365
Change-Id: I52ab4f9ef92b10329c1eac502adfcf2886058114
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131329
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27465}
This change introduces new logic to allow the injection of the FrameDecryptor
into an arbitrary already running VideoReceiveStream without resetting it. It
does this by taking advantage of the BufferedFrameDecryptor which will
forcefully be created regardless of whether a FrameDecryptor is passed in
during construction of the VideoReceiver if the
crypto_option.require_frame_encryption is true. By allowing the
BufferedFrameDecryptor to swap out which FrameDecryptor it uses this allows the
Receiver to switch decryptors without resetting the stream.
This is intended to mostly be used when you set your FrameDecryptor at a point
post creation for the first time.
Bug: webrtc:10416
Change-Id: If656b2acc447e2e77537cfa394729e5c3a8b660a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130361
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27458}
Extracting the work that's thread dependent from the work that will
also be done when using task queue.
Bug: webrtc:10365
Change-Id: I648796fe016c966c731c9b7f85d2a871c1f2a349
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131241
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27454}
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}
The video encoder should be informed of incoming LossNotification
RTCP messages. Since it is unaware of RTP sequence numbers,
or anything else RTP-related, the sequence numbers mentioned
in the RTCP message must first be mapped to timestamps,
since those are meaningful to the encoder.
This CL introduces RtpSequenceNumberMap, which maps RTP sequence
numbers to timestamps, while providing:
1. Capping the number of entries.
2. Wrap-around handling.
RtpSequenceNumberMap also remembers which packets were first
and/or last in the frame.
Later CLs will wire this up.
Bug: webrtc:10501
Change-Id: Ie0662cdb5706a3bcf63aa2934816a9df88439357
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130497
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27448}
This CL extends, and partly corrects, the benchmarking
code in audioproc_f to provide statistics for the API
call durations in audioproc_f
Bug: chromium:939791
Change-Id: I4c26c4bb3782335f13dd3e21e6f861842539ea62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129260
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27443}
When ALR was made default-on we removed the ability to use field trials
to configure alternative ALR detector values. This CL just restores
the ability to force them, defaults are unaffected.
Bug: webrtc:10509
Change-Id: Ibc09e27f1f7b72513de1482d280683802e962497
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131145
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27442}
