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Add unlimited retransmission experiment for screenshare

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Bug: webrtc:9659
Change-Id: Idcdc647c112ed2c7c027a7a0056b145ce8f45788
Reviewed-on: https://webrtc-review.googlesource.com/95724
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24422}
This commit is contained in:
Ilya Nikolaevskiy
2018-08-23 17:07:29 +02:00
committed by Commit Bot
parent cd634ce933
commit 523b4c4330
3 changed files with 27 additions and 2 deletions
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1
modules/rtp_rtcp/BUILD.gn
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@ -203,6 +203,7 @@ 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",
"../../common_video",
"../../logging:rtc_event_audio",

22
modules/rtp_rtcp/source/rtp_sender.cc
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@ -160,7 +160,9 @@ RTPSender::RTPSender(
overhead_observer_(overhead_observer),
populate_network2_timestamp_(populate_network2_timestamp),
send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
unlimited_retransmission_experiment_(
field_trial::IsEnabled("WebRTC-UnlimitedScreenshareRetransmission")) {
// This random initialization is not intended to be cryptographic strong.
timestamp_offset_ = random_.Rand<uint32_t>();
// Random start, 16 bits. Can't be 0.
@ -412,6 +414,11 @@ 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) {
@ -643,10 +650,21 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id) {
const int32_t packet_size = static_cast<int32_t>(stored_packet->payload_size);
// Skip retransmission rate check if sending screenshare and the experiment
// is on.
bool skip_retransmission_rate_limit;
{
rtc::CritScope lock(&send_critsect_);
skip_retransmission_rate_limit =
unlimited_retransmission_experiment_ && video_content_type_ &&
videocontenttypehelpers::IsScreenshare(*video_content_type_);
}
RTC_DCHECK(retransmission_rate_limiter_);
// Check if we're overusing retransmission bitrate.
// TODO(sprang): Add histograms for nack success or failure reasons.
if (!retransmission_rate_limiter_->TryUseRate(packet_size)) {
if (!skip_retransmission_rate_limit &&
!retransmission_rate_limiter_->TryUseRate(packet_size)) {
return -1;
}

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

@ -20,6 +20,7 @@
#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"
@ -338,6 +339,11 @@ 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);
};