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Remove unlimited retransmission for screenshare experiment code

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Bug: webrtc:9659
Change-Id: I29d8f0d20b0faee5ec2e8e196581338770b1a74d
Reviewed-on: https://webrtc-review.googlesource.com/c/105001
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25103}
This commit is contained in:
Ilya Nikolaevskiy
2018-10-10 15:17:39 +02:00
committed by Commit Bot
parent cc21e61e07
commit 23b2a25675
3 changed files with 2 additions and 27 deletions
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1
modules/rtp_rtcp/BUILD.gn
View File

@ -194,7 +194,6 @@ rtc_static_library("rtp_rtcp") {
"../../api/audio_codecs:audio_codecs_api",
"../../api/video:video_bitrate_allocation",
"../../api/video:video_bitrate_allocator",
"../../api/video:video_frame",
"../../api/video_codecs:video_codecs_api",
"../../call:rtp_interfaces",
"../../common_video",

22
modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -168,9 +168,7 @@ RTPSender::RTPSender(
overhead_observer_(overhead_observer),
populate_network2_timestamp_(populate_network2_timestamp),
send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
unlimited_retransmission_experiment_(
field_trial::IsEnabled("WebRTC-UnlimitedScreenshareRetransmission")) {
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
// This random initialization is not intended to be cryptographic strong.
timestamp_offset_ = random_.Rand<uint32_t>();
// Random start, 16 bits. Can't be 0.
@ -427,11 +425,6 @@ bool RTPSender::SendOutgoingData(FrameType frame_type,
*transport_frame_id_out = rtp_timestamp;
if (!sending_media_)
return true;
// Cache video content type.
if (!audio_configured_ && rtp_header) {
video_content_type_ = rtp_header->content_type;
}
}
VideoCodecType video_type = kVideoCodecGeneric;
if (CheckPayloadType(payload_type, &video_type) != 0) {
@ -671,20 +664,9 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id) {
// Skip retransmission rate check if not configured.
if (retransmission_rate_limiter_) {
// Skip retransmission rate check if sending screenshare and the experiment
// is on.
bool skip_retransmission_rate_limit = false;
if (unlimited_retransmission_experiment_) {
rtc::CritScope lock(&send_critsect_);
skip_retransmission_rate_limit =
video_content_type_ &&
videocontenttypehelpers::IsScreenshare(*video_content_type_);
}
// Check if we're overusing retransmission bitrate.
// TODO(sprang): Add histograms for nack success or failure reasons.
if (!skip_retransmission_rate_limit &&
!retransmission_rate_limiter_->TryUseRate(packet_size)) {
if (!retransmission_rate_limiter_->TryUseRate(packet_size)) {
return -1;
}
}

6
modules/rtp_rtcp/source/rtp_sender.h
View File

@ -20,7 +20,6 @@
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/video/video_content_type.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/include/flexfec_sender.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
@ -345,11 +344,6 @@ class RTPSender {
const bool send_side_bwe_with_overhead_;
const bool unlimited_retransmission_experiment_;
absl::optional<VideoContentType> video_content_type_
RTC_GUARDED_BY(send_critsect_);
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
};