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in RtpRtcp configuration delete unused remote bitrate estimator

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No code sets that configuration field.

Bug: None
Change-Id: Idd611d15ec54b3bd9115eac77d2222b97620d675
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267180
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37382}
This commit is contained in:
Danil Chapovalov
2022-06-29 16:45:31 +02:00
committed by WebRTC LUCI CQ
parent de2ac5a6f3
commit ed665521e4
5 changed files with 0 additions and 24 deletions
Download Patch File Download Diff File Expand all files Collapse all files

12
modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -73,7 +73,6 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
packet_overhead_(28), // IPV4 UDP.
nack_last_time_sent_full_ms_(0),
nack_last_seq_number_sent_(0),
remote_bitrate_(configuration.remote_bitrate_estimator),
rtt_stats_(configuration.rtt_stats),
rtt_ms_(0) {
if (!configuration.receiver_only) {
@ -141,17 +140,6 @@ void ModuleRtpRtcpImpl::Process() {
RTC_LOG_F(LS_WARNING) << "Timeout: No increase in RTCP RR extended "
"highest sequence number.";
}
if (remote_bitrate_ && rtcp_sender_.TMMBR()) {
unsigned int target_bitrate = 0;
std::vector<unsigned int> ssrcs;
if (remote_bitrate_->LatestEstimate(&ssrcs, &target_bitrate)) {
if (!ssrcs.empty()) {
target_bitrate = target_bitrate / ssrcs.size();
}
rtcp_sender_.SetTargetBitrate(target_bitrate);
}
}
} else {
// Report rtt from receiver.
if (process_rtt) {

3
modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -24,7 +24,6 @@
#include "api/rtp_headers.h"
#include "api/video/video_bitrate_allocation.h"
#include "modules/include/module_fec_types.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" // RTCPPacketType
#include "modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h"
@ -311,8 +310,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
int64_t nack_last_time_sent_full_ms_;
uint16_t nack_last_seq_number_sent_;
RemoteBitrateEstimator* const remote_bitrate_;
RtcpRttStats* const rtt_stats_;
// The processed RTT from RtcpRttStats.

1
modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
View File

@ -87,7 +87,6 @@ ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(const Configuration& configuration)
packet_overhead_(28), // IPV4 UDP.
nack_last_time_sent_full_ms_(0),
nack_last_seq_number_sent_(0),
remote_bitrate_(configuration.remote_bitrate_estimator),
rtt_stats_(configuration.rtt_stats),
rtt_ms_(0) {
RTC_DCHECK(worker_queue_);

3
modules/rtp_rtcp/source/rtp_rtcp_impl2.h
View File

@ -29,7 +29,6 @@
#include "api/units/time_delta.h"
#include "api/video/video_bitrate_allocation.h"
#include "modules/include/module_fec_types.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" // RTCPPacketType
#include "modules/rtp_rtcp/source/packet_sequencer.h"
#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
@ -322,8 +321,6 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
int64_t nack_last_time_sent_full_ms_;
uint16_t nack_last_seq_number_sent_;
RemoteBitrateEstimator* const remote_bitrate_;
RtcpRttStats* const rtt_stats_;
RepeatingTaskHandle rtt_update_task_ RTC_GUARDED_BY(worker_queue_);

5
modules/rtp_rtcp/source/rtp_rtcp_interface.h
View File

@ -35,7 +35,6 @@ namespace webrtc {
// Forward declarations.
class FrameEncryptorInterface;
class RateLimiter;
class RemoteBitrateEstimator;
class RtcEventLog;
class RTPSender;
class Transport;
@ -87,10 +86,6 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
RtcpCnameCallback* rtcp_cname_callback = nullptr;
ReportBlockDataObserver* report_block_data_observer = nullptr;
// Estimates the bandwidth available for a set of streams from the same
// client.
RemoteBitrateEstimator* remote_bitrate_estimator = nullptr;
// Spread any bursts of packets into smaller bursts to minimize packet loss.
RtpPacketSender* paced_sender = nullptr;