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Added field trials to disable video resizing

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Bug: webrtc:11812
Change-Id: If4d270c1c9abb4b0809fad579697faf63b9015cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180540
Commit-Queue: Razvan Surdulescu <razvans@google.com>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31876}
This commit is contained in:
Razvan Surdulescu
2020-08-07 00:08:17 -07:00
committed by Commit Bot
parent 5184219b6f
commit c55e24acc7
2 changed files with 14 additions and 5 deletions
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12
media/engine/webrtc_video_engine.cc
View File

@ -427,10 +427,12 @@ WebRtcVideoChannel::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
const VideoCodec& codec) { const VideoCodec& codec) {
RTC_DCHECK_RUN_ON(&thread_checker_); RTC_DCHECK_RUN_ON(&thread_checker_);
bool is_screencast = parameters_.options.is_screencast.value_or(false); bool is_screencast = parameters_.options.is_screencast.value_or(false);
// No automatic resizing when using simulcast or screencast. // No automatic resizing when using simulcast or screencast, or when
bool automatic_resize = // disabled by field trial flag.
!is_screencast && (parameters_.config.rtp.ssrcs.size() == 1 || bool automatic_resize = !disable_automatic_resize_ && !is_screencast &&
(parameters_.config.rtp.ssrcs.size() == 1 ||
NumActiveStreams(rtp_parameters_) == 1); NumActiveStreams(rtp_parameters_) == 1);
bool frame_dropping = !is_screencast; bool frame_dropping = !is_screencast;
bool denoising; bool denoising;
bool codec_default_denoising = false; bool codec_default_denoising = false;
@ -1946,7 +1948,9 @@ WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
encoder_sink_(nullptr), encoder_sink_(nullptr),
parameters_(std::move(config), options, max_bitrate_bps, codec_settings), parameters_(std::move(config), options, max_bitrate_bps, codec_settings),
rtp_parameters_(CreateRtpParametersWithEncodings(sp)), rtp_parameters_(CreateRtpParametersWithEncodings(sp)),
sending_(false) { sending_(false),
disable_automatic_resize_(webrtc::field_trial::IsEnabled(
"WebRTC-Video-DisableAutomaticResize")) {
// Maximum packet size may come in RtpConfig from external transport, for // Maximum packet size may come in RtpConfig from external transport, for
// example from QuicTransportInterface implementation, so do not exceed // example from QuicTransportInterface implementation, so do not exceed
// given max_packet_size. // given max_packet_size.

5
media/engine/webrtc_video_engine.h
View File

@ -433,6 +433,11 @@ class WebRtcVideoChannel : public VideoMediaChannel,
// destructed as they are used in asynchronous tasks it has to be destructed // destructed as they are used in asynchronous tasks it has to be destructed
// first. // first.
rtc::AsyncInvoker invoker_; rtc::AsyncInvoker invoker_;
// TODO(asapersson): investigate why setting
// DegrationPreferences::MAINTAIN_RESOLUTION isn't sufficient to disable
// downscaling everywhere in the pipeline.
const bool disable_automatic_resize_;
}; };
// Wrapper for the receiver part, contains configs etc. that are needed to // Wrapper for the receiver part, contains configs etc. that are needed to