webrtc_m130/modules/pacing/pacing_controller.cc

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/*
* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/pacing/pacing_controller.h"
#include <algorithm>
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
#include <memory>
#include <utility>
#include <vector>
#include "modules/pacing/bitrate_prober.h"
#include "modules/pacing/interval_budget.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
namespace {
// Time limit in milliseconds between packet bursts.
Use newer version of TimeDelta and TimeStamp factories in modules/ This change generated with following commands: find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I117d64a54950be040d996035c54bc0043310943a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168340 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30489}
2020-02-07 14:53:52 +01:00
constexpr TimeDelta kDefaultMinPacketLimit = TimeDelta::Millis(5);
constexpr TimeDelta kCongestedPacketInterval = TimeDelta::Millis(500);
// TODO(sprang): Consider dropping this limit.
// The maximum debt level, in terms of time, capped when sending packets.
Use newer version of TimeDelta and TimeStamp factories in modules/ This change generated with following commands: find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I117d64a54950be040d996035c54bc0043310943a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168340 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30489}
2020-02-07 14:53:52 +01:00
constexpr TimeDelta kMaxDebtInTime = TimeDelta::Millis(500);
constexpr TimeDelta kMaxElapsedTime = TimeDelta::Seconds(2);
constexpr DataSize kDefaultPaddingTarget = DataSize::Bytes(50);
// Upper cap on process interval, in case process has not been called in a long
// time.
Use newer version of TimeDelta and TimeStamp factories in modules/ This change generated with following commands: find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I117d64a54950be040d996035c54bc0043310943a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168340 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30489}
2020-02-07 14:53:52 +01:00
constexpr TimeDelta kMaxProcessingInterval = TimeDelta::Millis(30);
constexpr int kFirstPriority = 0;
bool IsDisabled(const WebRtcKeyValueConfig& field_trials,
absl::string_view key) {
return field_trials.Lookup(key).find("Disabled") == 0;
}
bool IsEnabled(const WebRtcKeyValueConfig& field_trials,
absl::string_view key) {
return field_trials.Lookup(key).find("Enabled") == 0;
}
int GetPriorityForType(RtpPacketMediaType type) {
// Lower number takes priority over higher.
switch (type) {
case RtpPacketMediaType::kAudio:
// Audio is always prioritized over other packet types.
return kFirstPriority + 1;
case RtpPacketMediaType::kRetransmission:
// Send retransmissions before new media.
return kFirstPriority + 2;
case RtpPacketMediaType::kVideo:
case RtpPacketMediaType::kForwardErrorCorrection:
// Video has "normal" priority, in the old speak.
// Send redundancy concurrently to video. If it is delayed it might have a
// lower chance of being useful.
return kFirstPriority + 3;
case RtpPacketMediaType::kPadding:
// Packets that are in themselves likely useless, only sent to keep the
// BWE high.
return kFirstPriority + 4;
}
}
} // namespace
const TimeDelta PacingController::kMaxExpectedQueueLength =
Use newer version of TimeDelta and TimeStamp factories in modules/ This change generated with following commands: find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I117d64a54950be040d996035c54bc0043310943a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168340 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30489}
2020-02-07 14:53:52 +01:00
TimeDelta::Millis(2000);
const float PacingController::kDefaultPaceMultiplier = 2.5f;
const TimeDelta PacingController::kPausedProcessInterval =
kCongestedPacketInterval;
Use newer version of TimeDelta and TimeStamp factories in modules/ This change generated with following commands: find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I117d64a54950be040d996035c54bc0043310943a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168340 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30489}
2020-02-07 14:53:52 +01:00
const TimeDelta PacingController::kMinSleepTime = TimeDelta::Millis(1);
PacingController::PacingController(Clock* clock,
PacketSender* packet_sender,
RtcEventLog* event_log,
const WebRtcKeyValueConfig* field_trials,
ProcessMode mode)
: mode_(mode),
clock_(clock),
packet_sender_(packet_sender),
fallback_field_trials_(
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
!field_trials ? std::make_unique<FieldTrialBasedConfig>() : nullptr),
field_trials_(field_trials ? field_trials : fallback_field_trials_.get()),
drain_large_queues_(
!IsDisabled(*field_trials_, "WebRTC-Pacer-DrainQueue")),
send_padding_if_silent_(
IsEnabled(*field_trials_, "WebRTC-Pacer-PadInSilence")),
pace_audio_(IsEnabled(*field_trials_, "WebRTC-Pacer-BlockAudio")),
small_first_probe_packet_(
IsEnabled(*field_trials_, "WebRTC-Pacer-SmallFirstProbePacket")),
Reland "Adds trial to use correct overhead calculation in pacer." This reverts commit 7affd9bcbb7a778408942d8afa4fe3ce29a8fc0b. Reason for revert: The perf issue has been addressed in the reland (https://webrtc-review.googlesource.com/c/src/+/167883). Original change's description: > Revert "Adds trial to use correct overhead calculation in pacer." > > This reverts commit 71a77c4b3b314a5e3b4e6b2f12d4886cff1b60d7. > > Reason for revert: https://webrtc-review.googlesource.com/c/src/+/167524 needs to be reverted and this CL causes a merge conflict. > > Original change's description: > > Adds trial to use correct overhead calculation in pacer. > > > > Bug: webrtc:9883 > > Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534 > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30399} > > TBR=sprang@webrtc.org,srte@webrtc.org > > Change-Id: I7d3efa29f70aa0363311766980acae6d88bbcaaa > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9883 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167880 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30409} TBR=mbonadei@webrtc.org,sprang@webrtc.org,srte@webrtc.org Change-Id: Iafdef81d08078000dc368e001f67bee660e2f5bc No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9883 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167861 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30414}
2020-01-29 18:45:00 +00:00
ignore_transport_overhead_(
IsEnabled(*field_trials_, "WebRTC-Pacer-IgnoreTransportOverhead")),
min_packet_limit_(kDefaultMinPacketLimit),
Reland "Adds trial to use correct overhead calculation in pacer." This reverts commit 7affd9bcbb7a778408942d8afa4fe3ce29a8fc0b. Reason for revert: The perf issue has been addressed in the reland (https://webrtc-review.googlesource.com/c/src/+/167883). Original change's description: > Revert "Adds trial to use correct overhead calculation in pacer." > > This reverts commit 71a77c4b3b314a5e3b4e6b2f12d4886cff1b60d7. > > Reason for revert: https://webrtc-review.googlesource.com/c/src/+/167524 needs to be reverted and this CL causes a merge conflict. > > Original change's description: > > Adds trial to use correct overhead calculation in pacer. > > > > Bug: webrtc:9883 > > Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534 > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30399} > > TBR=sprang@webrtc.org,srte@webrtc.org > > Change-Id: I7d3efa29f70aa0363311766980acae6d88bbcaaa > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9883 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167880 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30409} TBR=mbonadei@webrtc.org,sprang@webrtc.org,srte@webrtc.org Change-Id: Iafdef81d08078000dc368e001f67bee660e2f5bc No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9883 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167861 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30414}
2020-01-29 18:45:00 +00:00
transport_overhead_per_packet_(DataSize::Zero()),
last_timestamp_(clock_->CurrentTime()),
paused_(false),
media_budget_(0),
padding_budget_(0),
media_debt_(DataSize::Zero()),
padding_debt_(DataSize::Zero()),
media_rate_(DataRate::Zero()),
padding_rate_(DataRate::Zero()),
prober_(*field_trials_),
probing_send_failure_(false),
pacing_bitrate_(DataRate::Zero()),
last_process_time_(clock->CurrentTime()),
last_send_time_(last_process_time_),
packet_queue_(last_process_time_, field_trials_),
packet_counter_(0),
congestion_window_size_(DataSize::PlusInfinity()),
outstanding_data_(DataSize::Zero()),
queue_time_limit(kMaxExpectedQueueLength),
Reland "Reland "Only include overhead if using send side bandwidth estimation."" This is a reland of 086055d0fd9b9b9efe8bcf85884324a019e9bd33 ANA was accitendly disabled even when transport sequence numbers were negotiated due to a bug in how the audio send stream is configured. To solve this we simply continue to always allow enabling ANA and leave it up to the application to ensure that it's not used together with receive side estimation. Original change's description: > Reland "Only include overhead if using send side bandwidth estimation." > > This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e > > Original change's description: > > Only include overhead if using send side bandwidth estimation. > > > > Bug: webrtc:11298 > > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820 > > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Reviewed-by: Ali Tofigh <alito@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30382} > > Bug: webrtc:11298 > Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524 > Reviewed-by: Ali Tofigh <alito@webrtc.org> > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30390} Bug: webrtc:11298 Change-Id: If2ad91e17ebfc85dc51edcd9607996e18c5d1f13 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167883 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30413}
2020-01-29 17:42:52 +01:00
account_for_audio_(false),
include_overhead_(false) {
if (!drain_large_queues_) {
RTC_LOG(LS_WARNING) << "Pacer queues will not be drained,"
"pushback experiment must be enabled.";
}
FieldTrialParameter<int> min_packet_limit_ms("", min_packet_limit_.ms());
ParseFieldTrial({&min_packet_limit_ms},
field_trials_->Lookup("WebRTC-Pacer-MinPacketLimitMs"));
Use newer version of TimeDelta and TimeStamp factories in modules/ This change generated with following commands: find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I117d64a54950be040d996035c54bc0043310943a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168340 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30489}
2020-02-07 14:53:52 +01:00
min_packet_limit_ = TimeDelta::Millis(min_packet_limit_ms.Get());
UpdateBudgetWithElapsedTime(min_packet_limit_);
}
PacingController::~PacingController() = default;
void PacingController::CreateProbeCluster(DataRate bitrate, int cluster_id) {
prober_.CreateProbeCluster(bitrate, CurrentTime(), cluster_id);
}
void PacingController::Pause() {
if (!paused_)
RTC_LOG(LS_INFO) << "PacedSender paused.";
paused_ = true;
packet_queue_.SetPauseState(true, CurrentTime());
}
void PacingController::Resume() {
if (paused_)
RTC_LOG(LS_INFO) << "PacedSender resumed.";
paused_ = false;
packet_queue_.SetPauseState(false, CurrentTime());
}
bool PacingController::IsPaused() const {
return paused_;
}
void PacingController::SetCongestionWindow(DataSize congestion_window_size) {
const bool was_congested = Congested();
congestion_window_size_ = congestion_window_size;
if (was_congested && !Congested()) {
TimeDelta elapsed_time = UpdateTimeAndGetElapsed(CurrentTime());
UpdateBudgetWithElapsedTime(elapsed_time);
}
}
void PacingController::UpdateOutstandingData(DataSize outstanding_data) {
const bool was_congested = Congested();
outstanding_data_ = outstanding_data;
if (was_congested && !Congested()) {
TimeDelta elapsed_time = UpdateTimeAndGetElapsed(CurrentTime());
UpdateBudgetWithElapsedTime(elapsed_time);
}
}
bool PacingController::Congested() const {
if (congestion_window_size_.IsFinite()) {
return outstanding_data_ >= congestion_window_size_;
}
return false;
}
Timestamp PacingController::CurrentTime() const {
Timestamp time = clock_->CurrentTime();
if (time < last_timestamp_) {
RTC_LOG(LS_WARNING)
<< "Non-monotonic clock behavior observed. Previous timestamp: "
<< last_timestamp_.ms() << ", new timestamp: " << time.ms();
RTC_DCHECK_GE(time, last_timestamp_);
time = last_timestamp_;
}
last_timestamp_ = time;
return time;
}
void PacingController::SetProbingEnabled(bool enabled) {
RTC_CHECK_EQ(0, packet_counter_);
prober_.SetEnabled(enabled);
}
void PacingController::SetPacingRates(DataRate pacing_rate,
DataRate padding_rate) {
RTC_DCHECK_GT(pacing_rate, DataRate::Zero());
media_rate_ = pacing_rate;
padding_rate_ = padding_rate;
pacing_bitrate_ = pacing_rate;
padding_budget_.set_target_rate_kbps(padding_rate.kbps());
RTC_LOG(LS_VERBOSE) << "bwe:pacer_updated pacing_kbps="
<< pacing_bitrate_.kbps()
<< " padding_budget_kbps=" << padding_rate.kbps();
}
void PacingController::EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) {
RTC_DCHECK(pacing_bitrate_ > DataRate::Zero())
<< "SetPacingRate must be called before InsertPacket.";
RTC_CHECK(packet->packet_type());
// Get priority first and store in temporary, to avoid chance of object being
// moved before GetPriorityForType() being called.
const int priority = GetPriorityForType(*packet->packet_type());
EnqueuePacketInternal(std::move(packet), priority);
}
void PacingController::SetAccountForAudioPackets(bool account_for_audio) {
account_for_audio_ = account_for_audio;
}
Reland "Reland "Only include overhead if using send side bandwidth estimation."" This is a reland of 086055d0fd9b9b9efe8bcf85884324a019e9bd33 ANA was accitendly disabled even when transport sequence numbers were negotiated due to a bug in how the audio send stream is configured. To solve this we simply continue to always allow enabling ANA and leave it up to the application to ensure that it's not used together with receive side estimation. Original change's description: > Reland "Only include overhead if using send side bandwidth estimation." > > This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e > > Original change's description: > > Only include overhead if using send side bandwidth estimation. > > > > Bug: webrtc:11298 > > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820 > > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Reviewed-by: Ali Tofigh <alito@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30382} > > Bug: webrtc:11298 > Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524 > Reviewed-by: Ali Tofigh <alito@webrtc.org> > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30390} Bug: webrtc:11298 Change-Id: If2ad91e17ebfc85dc51edcd9607996e18c5d1f13 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167883 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30413}
2020-01-29 17:42:52 +01:00
void PacingController::SetIncludeOverhead() {
include_overhead_ = true;
packet_queue_.SetIncludeOverhead();
}
Reland "Adds trial to use correct overhead calculation in pacer." This reverts commit 7affd9bcbb7a778408942d8afa4fe3ce29a8fc0b. Reason for revert: The perf issue has been addressed in the reland (https://webrtc-review.googlesource.com/c/src/+/167883). Original change's description: > Revert "Adds trial to use correct overhead calculation in pacer." > > This reverts commit 71a77c4b3b314a5e3b4e6b2f12d4886cff1b60d7. > > Reason for revert: https://webrtc-review.googlesource.com/c/src/+/167524 needs to be reverted and this CL causes a merge conflict. > > Original change's description: > > Adds trial to use correct overhead calculation in pacer. > > > > Bug: webrtc:9883 > > Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534 > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30399} > > TBR=sprang@webrtc.org,srte@webrtc.org > > Change-Id: I7d3efa29f70aa0363311766980acae6d88bbcaaa > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9883 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167880 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30409} TBR=mbonadei@webrtc.org,sprang@webrtc.org,srte@webrtc.org Change-Id: Iafdef81d08078000dc368e001f67bee660e2f5bc No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9883 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167861 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30414}
2020-01-29 18:45:00 +00:00
void PacingController::SetTransportOverhead(DataSize overhead_per_packet) {
if (ignore_transport_overhead_)
return;
transport_overhead_per_packet_ = overhead_per_packet;
packet_queue_.SetTransportOverhead(overhead_per_packet);
}
TimeDelta PacingController::ExpectedQueueTime() const {
RTC_DCHECK_GT(pacing_bitrate_, DataRate::Zero());
Use newer version of TimeDelta and TimeStamp factories in modules/ This change generated with following commands: find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I117d64a54950be040d996035c54bc0043310943a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168340 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30489}
2020-02-07 14:53:52 +01:00
return TimeDelta::Millis(
(QueueSizeData().bytes() * 8 * rtc::kNumMillisecsPerSec) /
pacing_bitrate_.bps());
}
size_t PacingController::QueueSizePackets() const {
return packet_queue_.SizeInPackets();
}
DataSize PacingController::QueueSizeData() const {
return packet_queue_.Size();
}
DataSize PacingController::CurrentBufferLevel() const {
return std::max(media_debt_, padding_debt_);
}
absl::optional<Timestamp> PacingController::FirstSentPacketTime() const {
return first_sent_packet_time_;
}
TimeDelta PacingController::OldestPacketWaitTime() const {
Timestamp oldest_packet = packet_queue_.OldestEnqueueTime();
if (oldest_packet.IsInfinite()) {
return TimeDelta::Zero();
}
return CurrentTime() - oldest_packet;
}
void PacingController::EnqueuePacketInternal(
std::unique_ptr<RtpPacketToSend> packet,
int priority) {
prober_.OnIncomingPacket(packet->payload_size());
// TODO(sprang): Make sure tests respect this, replace with DCHECK.
Timestamp now = CurrentTime();
if (packet->capture_time_ms() < 0) {
packet->set_capture_time_ms(now.ms());
}
if (mode_ == ProcessMode::kDynamic && packet_queue_.Empty() &&
media_debt_ == DataSize::Zero()) {
last_process_time_ = CurrentTime();
}
packet_queue_.Push(priority, now, packet_counter_++, std::move(packet));
}
TimeDelta PacingController::UpdateTimeAndGetElapsed(Timestamp now) {
if (last_process_time_.IsMinusInfinity()) {
return TimeDelta::Zero();
}
Reland "Fixes dynamic mode pacing issues." This is a reland of 72e6cb0b3f548900fd3b548b4b6966e3f5ee854f Was not the cause of perf alert, relanding. TBR=ilnik@webrtc.org Original change's description: > Fixes dynamic mode pacing issues. > > This CL fixes a few issues in the (default-disabled) dynamic pacing > mode: > * Slight update to sleep timing to avoid short spin loops > * Removed support for early execution as that lead to time-travel > contradictions that were difficult to solve. > * Makes sure we schedule a process call when a packet is due to be > drained even if the queue is empty, so that padding will start at > the correct time. > * While paused or empty, sleep relative last send time if we send > padding while silent - otherwise just relative to last process > time. > * If target send time shifts so far back that packet should have > been sent prior to the last process, make sure we don't let the > buffer level remain. > * Update the PacedSender test to _actually_ use dynamic processing > when the param says so. > > Bug: webrtc:10809 > Change-Id: Iebfde9769647d2390fd192a40bbe2d5bf1f6cc62 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160407 > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29911} Bug: webrtc:10809 Change-Id: Ie7b307e574c2057bb05af87b6718a132d639a416 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160786 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29928}
2019-11-25 18:22:09 +01:00
RTC_DCHECK_GE(now, last_process_time_);
TimeDelta elapsed_time = now - last_process_time_;
last_process_time_ = now;
if (elapsed_time > kMaxElapsedTime) {
RTC_LOG(LS_WARNING) << "Elapsed time (" << elapsed_time.ms()
<< " ms) longer than expected, limiting to "
<< kMaxElapsedTime.ms();
elapsed_time = kMaxElapsedTime;
}
return elapsed_time;
}
bool PacingController::ShouldSendKeepalive(Timestamp now) const {
if (send_padding_if_silent_ || paused_ || Congested() ||
packet_counter_ == 0) {
// We send a padding packet every 500 ms to ensure we won't get stuck in
// congested state due to no feedback being received.
TimeDelta elapsed_since_last_send = now - last_send_time_;
if (elapsed_since_last_send >= kCongestedPacketInterval) {
return true;
}
}
return false;
}
Timestamp PacingController::NextSendTime() const {
Timestamp now = CurrentTime();
if (paused_) {
return last_send_time_ + kPausedProcessInterval;
}
// If probing is active, that always takes priority.
if (prober_.is_probing()) {
Timestamp probe_time = prober_.NextProbeTime(now);
// |probe_time| == PlusInfinity indicates no probe scheduled.
if (probe_time != Timestamp::PlusInfinity() && !probing_send_failure_) {
return probe_time;
}
}
if (mode_ == ProcessMode::kPeriodic) {
// In periodic non-probing mode, we just have a fixed interval.
return last_process_time_ + min_packet_limit_;
}
// In dynamic mode, figure out when the next packet should be sent,
// given the current conditions.
if (!pace_audio_ && packet_queue_.NextPacketIsAudio()) {
return now;
}
if (Congested() || packet_counter_ == 0) {
// We need to at least send keep-alive packets with some interval.
return last_send_time_ + kCongestedPacketInterval;
}
Reland "Fixes dynamic mode pacing issues." This is a reland of 72e6cb0b3f548900fd3b548b4b6966e3f5ee854f Was not the cause of perf alert, relanding. TBR=ilnik@webrtc.org Original change's description: > Fixes dynamic mode pacing issues. > > This CL fixes a few issues in the (default-disabled) dynamic pacing > mode: > * Slight update to sleep timing to avoid short spin loops > * Removed support for early execution as that lead to time-travel > contradictions that were difficult to solve. > * Makes sure we schedule a process call when a packet is due to be > drained even if the queue is empty, so that padding will start at > the correct time. > * While paused or empty, sleep relative last send time if we send > padding while silent - otherwise just relative to last process > time. > * If target send time shifts so far back that packet should have > been sent prior to the last process, make sure we don't let the > buffer level remain. > * Update the PacedSender test to _actually_ use dynamic processing > when the param says so. > > Bug: webrtc:10809 > Change-Id: Iebfde9769647d2390fd192a40bbe2d5bf1f6cc62 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160407 > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29911} Bug: webrtc:10809 Change-Id: Ie7b307e574c2057bb05af87b6718a132d639a416 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160786 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29928}
2019-11-25 18:22:09 +01:00
// Check how long until media buffer has drained. We schedule a call
// for when the last packet in the queue drains as otherwise we may
// be late in starting padding.
if (media_rate_ > DataRate::Zero() &&
(!packet_queue_.Empty() || !media_debt_.IsZero())) {
return std::min(last_send_time_ + kPausedProcessInterval,
last_process_time_ + media_debt_ / media_rate_);
}
// If we _don't_ have pending packets, check how long until we have
// bandwidth for padding packets.
if (padding_rate_ > DataRate::Zero() && packet_queue_.Empty()) {
return std::min(last_send_time_ + kPausedProcessInterval,
last_process_time_ + padding_debt_ / padding_rate_);
}
Reland "Fixes dynamic mode pacing issues." This is a reland of 72e6cb0b3f548900fd3b548b4b6966e3f5ee854f Was not the cause of perf alert, relanding. TBR=ilnik@webrtc.org Original change's description: > Fixes dynamic mode pacing issues. > > This CL fixes a few issues in the (default-disabled) dynamic pacing > mode: > * Slight update to sleep timing to avoid short spin loops > * Removed support for early execution as that lead to time-travel > contradictions that were difficult to solve. > * Makes sure we schedule a process call when a packet is due to be > drained even if the queue is empty, so that padding will start at > the correct time. > * While paused or empty, sleep relative last send time if we send > padding while silent - otherwise just relative to last process > time. > * If target send time shifts so far back that packet should have > been sent prior to the last process, make sure we don't let the > buffer level remain. > * Update the PacedSender test to _actually_ use dynamic processing > when the param says so. > > Bug: webrtc:10809 > Change-Id: Iebfde9769647d2390fd192a40bbe2d5bf1f6cc62 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160407 > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29911} Bug: webrtc:10809 Change-Id: Ie7b307e574c2057bb05af87b6718a132d639a416 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160786 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29928}
2019-11-25 18:22:09 +01:00
if (send_padding_if_silent_) {
return last_send_time_ + kPausedProcessInterval;
}
return last_process_time_ + kPausedProcessInterval;
}
void PacingController::ProcessPackets() {
Timestamp now = CurrentTime();
Timestamp target_send_time = now;
if (mode_ == ProcessMode::kDynamic) {
target_send_time = NextSendTime();
if (target_send_time.IsMinusInfinity()) {
target_send_time = now;
Reland "Fixes dynamic mode pacing issues." This is a reland of 72e6cb0b3f548900fd3b548b4b6966e3f5ee854f Was not the cause of perf alert, relanding. TBR=ilnik@webrtc.org Original change's description: > Fixes dynamic mode pacing issues. > > This CL fixes a few issues in the (default-disabled) dynamic pacing > mode: > * Slight update to sleep timing to avoid short spin loops > * Removed support for early execution as that lead to time-travel > contradictions that were difficult to solve. > * Makes sure we schedule a process call when a packet is due to be > drained even if the queue is empty, so that padding will start at > the correct time. > * While paused or empty, sleep relative last send time if we send > padding while silent - otherwise just relative to last process > time. > * If target send time shifts so far back that packet should have > been sent prior to the last process, make sure we don't let the > buffer level remain. > * Update the PacedSender test to _actually_ use dynamic processing > when the param says so. > > Bug: webrtc:10809 > Change-Id: Iebfde9769647d2390fd192a40bbe2d5bf1f6cc62 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160407 > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29911} Bug: webrtc:10809 Change-Id: Ie7b307e574c2057bb05af87b6718a132d639a416 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160786 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29928}
2019-11-25 18:22:09 +01:00
} else if (now < target_send_time) {
// We are too early, abort and regroup!
return;
}
Reland "Fixes dynamic mode pacing issues." This is a reland of 72e6cb0b3f548900fd3b548b4b6966e3f5ee854f Was not the cause of perf alert, relanding. TBR=ilnik@webrtc.org Original change's description: > Fixes dynamic mode pacing issues. > > This CL fixes a few issues in the (default-disabled) dynamic pacing > mode: > * Slight update to sleep timing to avoid short spin loops > * Removed support for early execution as that lead to time-travel > contradictions that were difficult to solve. > * Makes sure we schedule a process call when a packet is due to be > drained even if the queue is empty, so that padding will start at > the correct time. > * While paused or empty, sleep relative last send time if we send > padding while silent - otherwise just relative to last process > time. > * If target send time shifts so far back that packet should have > been sent prior to the last process, make sure we don't let the > buffer level remain. > * Update the PacedSender test to _actually_ use dynamic processing > when the param says so. > > Bug: webrtc:10809 > Change-Id: Iebfde9769647d2390fd192a40bbe2d5bf1f6cc62 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160407 > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29911} Bug: webrtc:10809 Change-Id: Ie7b307e574c2057bb05af87b6718a132d639a416 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160786 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29928}
2019-11-25 18:22:09 +01:00
if (target_send_time < last_process_time_) {
// After the last process call, at time X, the target send time
// shifted to be earlier than X. This should normally not happen
// but we want to make sure rounding errors or erratic behavior
// of NextSendTime() does not cause issue. In particular, if the
// buffer reduction of
// rate * (target_send_time - previous_process_time)
// in the main loop doesn't clean up the existing debt we may not
// be able to send again. We don't want to check this reordering
// there as it is the normal exit condtion when the buffer is
// exhausted and there are packets in the queue.
UpdateBudgetWithElapsedTime(last_process_time_ - target_send_time);
target_send_time = last_process_time_;
}
}
Timestamp previous_process_time = last_process_time_;
TimeDelta elapsed_time = UpdateTimeAndGetElapsed(now);
if (ShouldSendKeepalive(now)) {
// We can not send padding unless a normal packet has first been sent. If
// we do, timestamps get messed up.
if (packet_counter_ == 0) {
last_send_time_ = now;
} else {
DataSize keepalive_data_sent = DataSize::Zero();
std::vector<std::unique_ptr<RtpPacketToSend>> keepalive_packets =
packet_sender_->GeneratePadding(DataSize::Bytes(1));
for (auto& packet : keepalive_packets) {
keepalive_data_sent +=
DataSize::Bytes(packet->payload_size() + packet->padding_size());
packet_sender_->SendRtpPacket(std::move(packet), PacedPacketInfo());
}
OnPaddingSent(keepalive_data_sent);
}
}
if (paused_) {
return;
}
if (elapsed_time > TimeDelta::Zero()) {
DataRate target_rate = pacing_bitrate_;
DataSize queue_size_data = packet_queue_.Size();
if (queue_size_data > DataSize::Zero()) {
// Assuming equal size packets and input/output rate, the average packet
// has avg_time_left_ms left to get queue_size_bytes out of the queue, if
// time constraint shall be met. Determine bitrate needed for that.
packet_queue_.UpdateQueueTime(now);
if (drain_large_queues_) {
TimeDelta avg_time_left =
Use newer version of TimeDelta and TimeStamp factories in modules/ This change generated with following commands: find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I117d64a54950be040d996035c54bc0043310943a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168340 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30489}
2020-02-07 14:53:52 +01:00
std::max(TimeDelta::Millis(1),
queue_time_limit - packet_queue_.AverageQueueTime());
DataRate min_rate_needed = queue_size_data / avg_time_left;
if (min_rate_needed > target_rate) {
target_rate = min_rate_needed;
RTC_LOG(LS_VERBOSE) << "bwe:large_pacing_queue pacing_rate_kbps="
<< target_rate.kbps();
}
}
}
if (mode_ == ProcessMode::kPeriodic) {
// In periodic processing mode, the IntevalBudget allows positive budget
// up to (process interval duration) * (target rate), so we only need to
// update it once before the packet sending loop.
media_budget_.set_target_rate_kbps(target_rate.kbps());
UpdateBudgetWithElapsedTime(elapsed_time);
} else {
media_rate_ = target_rate;
}
}
bool first_packet_in_probe = false;
bool is_probing = prober_.is_probing();
PacedPacketInfo pacing_info;
absl::optional<DataSize> recommended_probe_size;
if (is_probing) {
pacing_info = prober_.CurrentCluster();
first_packet_in_probe = pacing_info.probe_cluster_bytes_sent == 0;
recommended_probe_size = DataSize::Bytes(prober_.RecommendedMinProbeSize());
}
DataSize data_sent = DataSize::Zero();
// The paused state is checked in the loop since it leaves the critical
// section allowing the paused state to be changed from other code.
while (!paused_) {
if (small_first_probe_packet_ && first_packet_in_probe) {
// If first packet in probe, insert a small padding packet so we have a
// more reliable start window for the rate estimation.
auto padding = packet_sender_->GeneratePadding(DataSize::Bytes(1));
// If no RTP modules sending media are registered, we may not get a
// padding packet back.
if (!padding.empty()) {
// Insert with high priority so larger media packets don't preempt it.
EnqueuePacketInternal(std::move(padding[0]), kFirstPriority);
// We should never get more than one padding packets with a requested
// size of 1 byte.
RTC_DCHECK_EQ(padding.size(), 1u);
}
first_packet_in_probe = false;
}
if (mode_ == ProcessMode::kDynamic &&
previous_process_time < target_send_time) {
// Reduce buffer levels with amount corresponding to time between last
// process and target send time for the next packet.
// If the process call is late, that may be the time between the optimal
// send times for two packets we should already have sent.
UpdateBudgetWithElapsedTime(target_send_time - previous_process_time);
previous_process_time = target_send_time;
}
// Fetch the next packet, so long as queue is not empty or budget is not
// exhausted.
std::unique_ptr<RtpPacketToSend> rtp_packet =
GetPendingPacket(pacing_info, target_send_time, now);
if (rtp_packet == nullptr) {
// No packet available to send, check if we should send padding.
DataSize padding_to_add = PaddingToAdd(recommended_probe_size, data_sent);
if (padding_to_add > DataSize::Zero()) {
std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
packet_sender_->GeneratePadding(padding_to_add);
if (padding_packets.empty()) {
// No padding packets were generated, quite send loop.
break;
}
for (auto& packet : padding_packets) {
EnqueuePacket(std::move(packet));
}
// Continue loop to send the padding that was just added.
continue;
}
// Can't fetch new packet and no padding to send, exit send loop.
break;
}
RTC_DCHECK(rtp_packet);
RTC_DCHECK(rtp_packet->packet_type().has_value());
const RtpPacketMediaType packet_type = *rtp_packet->packet_type();
DataSize packet_size = DataSize::Bytes(rtp_packet->payload_size() +
Reland "Adds trial to use correct overhead calculation in pacer." This reverts commit 7affd9bcbb7a778408942d8afa4fe3ce29a8fc0b. Reason for revert: The perf issue has been addressed in the reland (https://webrtc-review.googlesource.com/c/src/+/167883). Original change's description: > Revert "Adds trial to use correct overhead calculation in pacer." > > This reverts commit 71a77c4b3b314a5e3b4e6b2f12d4886cff1b60d7. > > Reason for revert: https://webrtc-review.googlesource.com/c/src/+/167524 needs to be reverted and this CL causes a merge conflict. > > Original change's description: > > Adds trial to use correct overhead calculation in pacer. > > > > Bug: webrtc:9883 > > Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534 > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30399} > > TBR=sprang@webrtc.org,srte@webrtc.org > > Change-Id: I7d3efa29f70aa0363311766980acae6d88bbcaaa > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9883 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167880 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30409} TBR=mbonadei@webrtc.org,sprang@webrtc.org,srte@webrtc.org Change-Id: Iafdef81d08078000dc368e001f67bee660e2f5bc No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9883 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167861 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30414}
2020-01-29 18:45:00 +00:00
rtp_packet->padding_size());
if (include_overhead_) {
packet_size += DataSize::Bytes(rtp_packet->headers_size()) +
Reland "Adds trial to use correct overhead calculation in pacer." This reverts commit 7affd9bcbb7a778408942d8afa4fe3ce29a8fc0b. Reason for revert: The perf issue has been addressed in the reland (https://webrtc-review.googlesource.com/c/src/+/167883). Original change's description: > Revert "Adds trial to use correct overhead calculation in pacer." > > This reverts commit 71a77c4b3b314a5e3b4e6b2f12d4886cff1b60d7. > > Reason for revert: https://webrtc-review.googlesource.com/c/src/+/167524 needs to be reverted and this CL causes a merge conflict. > > Original change's description: > > Adds trial to use correct overhead calculation in pacer. > > > > Bug: webrtc:9883 > > Change-Id: I1f25a235468678bf823ee1399ba31d94acf33be9 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166534 > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30399} > > TBR=sprang@webrtc.org,srte@webrtc.org > > Change-Id: I7d3efa29f70aa0363311766980acae6d88bbcaaa > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9883 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167880 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30409} TBR=mbonadei@webrtc.org,sprang@webrtc.org,srte@webrtc.org Change-Id: Iafdef81d08078000dc368e001f67bee660e2f5bc No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9883 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167861 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30414}
2020-01-29 18:45:00 +00:00
transport_overhead_per_packet_;
}
packet_sender_->SendRtpPacket(std::move(rtp_packet), pacing_info);
data_sent += packet_size;
// Send done, update send/process time to the target send time.
OnPacketSent(packet_type, packet_size, target_send_time);
if (recommended_probe_size && data_sent > *recommended_probe_size)
break;
if (mode_ == ProcessMode::kDynamic) {
// Update target send time in case that are more packets that we are late
// in processing.
Timestamp next_send_time = NextSendTime();
if (next_send_time.IsMinusInfinity()) {
target_send_time = now;
} else {
target_send_time = std::min(now, next_send_time);
}
}
}
if (is_probing) {
probing_send_failure_ = data_sent == DataSize::Zero();
if (!probing_send_failure_) {
prober_.ProbeSent(CurrentTime(), data_sent.bytes());
}
}
}
DataSize PacingController::PaddingToAdd(
absl::optional<DataSize> recommended_probe_size,
DataSize data_sent) const {
if (!packet_queue_.Empty()) {
// Actual payload available, no need to add padding.
return DataSize::Zero();
}
if (Congested()) {
// Don't add padding if congested, even if requested for probing.
return DataSize::Zero();
}
if (packet_counter_ == 0) {
// We can not send padding unless a normal packet has first been sent. If we
// do, timestamps get messed up.
return DataSize::Zero();
}
if (recommended_probe_size) {
if (*recommended_probe_size > data_sent) {
return *recommended_probe_size - data_sent;
}
return DataSize::Zero();
}
if (mode_ == ProcessMode::kPeriodic) {
return DataSize::Bytes(padding_budget_.bytes_remaining());
} else if (padding_rate_ > DataRate::Zero() &&
padding_debt_ == DataSize::Zero()) {
return kDefaultPaddingTarget;
}
return DataSize::Zero();
}
std::unique_ptr<RtpPacketToSend> PacingController::GetPendingPacket(
const PacedPacketInfo& pacing_info,
Timestamp target_send_time,
Timestamp now) {
if (packet_queue_.Empty()) {
return nullptr;
}
// First, check if there is any reason _not_ to send the next queued packet.
// Unpaced audio packets and probes are exempted from send checks.
bool unpaced_audio_packet = !pace_audio_ && packet_queue_.NextPacketIsAudio();
bool is_probe = pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe;
if (!unpaced_audio_packet && !is_probe) {
if (Congested()) {
// Don't send anything if congested.
return nullptr;
}
if (mode_ == ProcessMode::kPeriodic) {
if (media_budget_.bytes_remaining() <= 0) {
// Not enough budget.
return nullptr;
}
} else {
Reland "Fixes dynamic mode pacing issues." This is a reland of 72e6cb0b3f548900fd3b548b4b6966e3f5ee854f Was not the cause of perf alert, relanding. TBR=ilnik@webrtc.org Original change's description: > Fixes dynamic mode pacing issues. > > This CL fixes a few issues in the (default-disabled) dynamic pacing > mode: > * Slight update to sleep timing to avoid short spin loops > * Removed support for early execution as that lead to time-travel > contradictions that were difficult to solve. > * Makes sure we schedule a process call when a packet is due to be > drained even if the queue is empty, so that padding will start at > the correct time. > * While paused or empty, sleep relative last send time if we send > padding while silent - otherwise just relative to last process > time. > * If target send time shifts so far back that packet should have > been sent prior to the last process, make sure we don't let the > buffer level remain. > * Update the PacedSender test to _actually_ use dynamic processing > when the param says so. > > Bug: webrtc:10809 > Change-Id: Iebfde9769647d2390fd192a40bbe2d5bf1f6cc62 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160407 > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29911} Bug: webrtc:10809 Change-Id: Ie7b307e574c2057bb05af87b6718a132d639a416 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160786 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29928}
2019-11-25 18:22:09 +01:00
// Dynamic processing mode.
if (now <= target_send_time) {
// We allow sending slightly early if we think that we would actually
// had been able to, had we been right on time - i.e. the current debt
// is not more than would be reduced to zero at the target sent time.
TimeDelta flush_time = media_debt_ / media_rate_;
if (now + flush_time > target_send_time) {
return nullptr;
}
}
}
}
return packet_queue_.Pop();
}
void PacingController::OnPacketSent(RtpPacketMediaType packet_type,
DataSize packet_size,
Timestamp send_time) {
if (!first_sent_packet_time_) {
first_sent_packet_time_ = send_time;
}
bool audio_packet = packet_type == RtpPacketMediaType::kAudio;
if (!audio_packet || account_for_audio_) {
// Update media bytes sent.
UpdateBudgetWithSentData(packet_size);
}
last_send_time_ = send_time;
last_process_time_ = send_time;
}
void PacingController::OnPaddingSent(DataSize data_sent) {
if (data_sent > DataSize::Zero()) {
UpdateBudgetWithSentData(data_sent);
}
last_send_time_ = CurrentTime();
last_process_time_ = CurrentTime();
}
void PacingController::UpdateBudgetWithElapsedTime(TimeDelta delta) {
if (mode_ == ProcessMode::kPeriodic) {
delta = std::min(kMaxProcessingInterval, delta);
media_budget_.IncreaseBudget(delta.ms());
padding_budget_.IncreaseBudget(delta.ms());
} else {
media_debt_ -= std::min(media_debt_, media_rate_ * delta);
padding_debt_ -= std::min(padding_debt_, padding_rate_ * delta);
}
}
void PacingController::UpdateBudgetWithSentData(DataSize size) {
outstanding_data_ += size;
if (mode_ == ProcessMode::kPeriodic) {
media_budget_.UseBudget(size.bytes());
padding_budget_.UseBudget(size.bytes());
} else {
media_debt_ += size;
media_debt_ = std::min(media_debt_, media_rate_ * kMaxDebtInTime);
padding_debt_ += size;
padding_debt_ = std::min(padding_debt_, padding_rate_ * kMaxDebtInTime);
}
}
void PacingController::SetQueueTimeLimit(TimeDelta limit) {
queue_time_limit = limit;
}
} // namespace webrtc