webrtc_m130/call/bitrate_allocator.cc

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/*
* Copyright (c) 2015 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 "call/bitrate_allocator.h"
#include <algorithm>
#include <cmath>
#include <memory>
#include <utility>
#include "absl/algorithm/container.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
using bitrate_allocator_impl::AllocatableTrack;
// Allow packets to be transmitted in up to 2 times max video bitrate if the
// bandwidth estimate allows it.
const uint8_t kTransmissionMaxBitrateMultiplier = 2;
const int kDefaultBitrateBps = 300000;
// Require a bitrate increase of max(10%, 20kbps) to resume paused streams.
const double kToggleFactor = 0.1;
const uint32_t kMinToggleBitrateBps = 20000;
const int64_t kBweLogIntervalMs = 5000;
double MediaRatio(uint32_t allocated_bitrate, uint32_t protection_bitrate) {
RTC_DCHECK_GT(allocated_bitrate, 0);
if (protection_bitrate == 0)
return 1.0;
uint32_t media_bitrate = allocated_bitrate - protection_bitrate;
return media_bitrate / static_cast<double>(allocated_bitrate);
}
bool EnoughBitrateForAllObservers(
const std::vector<AllocatableTrack>& allocatable_tracks,
uint32_t bitrate,
uint32_t sum_min_bitrates) {
if (bitrate < sum_min_bitrates)
return false;
uint32_t extra_bitrate_per_observer =
(bitrate - sum_min_bitrates) /
static_cast<uint32_t>(allocatable_tracks.size());
for (const auto& observer_config : allocatable_tracks) {
if (observer_config.config.min_bitrate_bps + extra_bitrate_per_observer <
observer_config.MinBitrateWithHysteresis()) {
return false;
}
}
return true;
}
// Splits `bitrate` evenly to observers already in `allocation`.
// `include_zero_allocations` decides if zero allocations should be part of
// the distribution or not. The allowed max bitrate is `max_multiplier` x
// observer max bitrate.
void DistributeBitrateEvenly(
const std::vector<AllocatableTrack>& allocatable_tracks,
uint32_t bitrate,
bool include_zero_allocations,
int max_multiplier,
std::map<BitrateAllocatorObserver*, int>* allocation) {
RTC_DCHECK_EQ(allocation->size(), allocatable_tracks.size());
std::multimap<uint32_t, const AllocatableTrack*> list_max_bitrates;
for (const auto& observer_config : allocatable_tracks) {
if (include_zero_allocations ||
allocation->at(observer_config.observer) != 0) {
list_max_bitrates.insert(
{observer_config.config.max_bitrate_bps, &observer_config});
}
}
auto it = list_max_bitrates.begin();
while (it != list_max_bitrates.end()) {
RTC_DCHECK_GT(bitrate, 0);
uint32_t extra_allocation =
bitrate / static_cast<uint32_t>(list_max_bitrates.size());
uint32_t total_allocation =
extra_allocation + allocation->at(it->second->observer);
bitrate -= extra_allocation;
if (total_allocation > max_multiplier * it->first) {
// There is more than we can fit for this observer, carry over to the
// remaining observers.
bitrate += total_allocation - max_multiplier * it->first;
total_allocation = max_multiplier * it->first;
}
// Finally, update the allocation for this observer.
allocation->at(it->second->observer) = total_allocation;
it = list_max_bitrates.erase(it);
}
}
// From the available `bitrate`, each observer will be allocated a
// proportional amount based upon its bitrate priority. If that amount is
// more than the observer's capacity, it will be allocated its capacity, and
// the excess bitrate is still allocated proportionally to other observers.
// Allocating the proportional amount means an observer with twice the
// bitrate_priority of another will be allocated twice the bitrate.
void DistributeBitrateRelatively(
const std::vector<AllocatableTrack>& allocatable_tracks,
uint32_t remaining_bitrate,
const std::map<BitrateAllocatorObserver*, int>& observers_capacities,
std::map<BitrateAllocatorObserver*, int>* allocation) {
RTC_DCHECK_EQ(allocation->size(), allocatable_tracks.size());
RTC_DCHECK_EQ(observers_capacities.size(), allocatable_tracks.size());
struct PriorityRateObserverConfig {
BitrateAllocatorObserver* allocation_key;
// The amount of bitrate bps that can be allocated to this observer.
int capacity_bps;
double bitrate_priority;
};
double bitrate_priority_sum = 0;
std::vector<PriorityRateObserverConfig> priority_rate_observers;
for (const auto& observer_config : allocatable_tracks) {
priority_rate_observers.push_back(PriorityRateObserverConfig{
observer_config.observer,
observers_capacities.at(observer_config.observer),
observer_config.config.bitrate_priority});
bitrate_priority_sum += observer_config.config.bitrate_priority;
}
// Iterate in the order observers can be allocated their full capacity.
// We want to sort by which observers will be allocated their full capacity
// first. By dividing each observer's capacity by its bitrate priority we
// are "normalizing" the capacity of an observer by the rate it will be
// filled. This is because the amount allocated is based upon bitrate
// priority. We allocate twice as much bitrate to an observer with twice the
// bitrate priority of another.
absl::c_sort(priority_rate_observers, [](const auto& a, const auto& b) {
return a.capacity_bps / a.bitrate_priority <
b.capacity_bps / b.bitrate_priority;
});
size_t i;
for (i = 0; i < priority_rate_observers.size(); ++i) {
const auto& priority_rate_observer = priority_rate_observers[i];
// We allocate the full capacity to an observer only if its relative
// portion from the remaining bitrate is sufficient to allocate its full
// capacity. This means we aren't greedily allocating the full capacity, but
// that it is only done when there is also enough bitrate to allocate the
// proportional amounts to all other observers.
double observer_share =
priority_rate_observer.bitrate_priority / bitrate_priority_sum;
double allocation_bps = observer_share * remaining_bitrate;
bool enough_bitrate = allocation_bps >= priority_rate_observer.capacity_bps;
if (!enough_bitrate)
break;
allocation->at(priority_rate_observer.allocation_key) +=
priority_rate_observer.capacity_bps;
remaining_bitrate -= priority_rate_observer.capacity_bps;
bitrate_priority_sum -= priority_rate_observer.bitrate_priority;
}
// From the remaining bitrate, allocate the proportional amounts to the
// observers that aren't allocated their max capacity.
for (; i < priority_rate_observers.size(); ++i) {
const auto& priority_rate_observer = priority_rate_observers[i];
double fraction_allocated =
priority_rate_observer.bitrate_priority / bitrate_priority_sum;
allocation->at(priority_rate_observer.allocation_key) +=
fraction_allocated * remaining_bitrate;
}
}
// Allocates bitrate to observers when there isn't enough to allocate the
// minimum to all observers.
std::map<BitrateAllocatorObserver*, int> LowRateAllocation(
const std::vector<AllocatableTrack>& allocatable_tracks,
uint32_t bitrate) {
std::map<BitrateAllocatorObserver*, int> allocation;
// Start by allocating bitrate to observers enforcing a min bitrate, hence
// remaining_bitrate might turn negative.
int64_t remaining_bitrate = bitrate;
for (const auto& observer_config : allocatable_tracks) {
int32_t allocated_bitrate = 0;
if (observer_config.config.enforce_min_bitrate)
allocated_bitrate = observer_config.config.min_bitrate_bps;
allocation[observer_config.observer] = allocated_bitrate;
remaining_bitrate -= allocated_bitrate;
}
// Allocate bitrate to all previously active streams.
if (remaining_bitrate > 0) {
for (const auto& observer_config : allocatable_tracks) {
if (observer_config.config.enforce_min_bitrate ||
observer_config.LastAllocatedBitrate() == 0)
continue;
uint32_t required_bitrate = observer_config.MinBitrateWithHysteresis();
if (remaining_bitrate >= required_bitrate) {
allocation[observer_config.observer] = required_bitrate;
remaining_bitrate -= required_bitrate;
}
}
}
// Allocate bitrate to previously paused streams.
if (remaining_bitrate > 0) {
for (const auto& observer_config : allocatable_tracks) {
if (observer_config.LastAllocatedBitrate() != 0)
continue;
// Add a hysteresis to avoid toggling.
uint32_t required_bitrate = observer_config.MinBitrateWithHysteresis();
if (remaining_bitrate >= required_bitrate) {
allocation[observer_config.observer] = required_bitrate;
remaining_bitrate -= required_bitrate;
}
}
}
// Split a possible remainder evenly on all streams with an allocation.
if (remaining_bitrate > 0)
DistributeBitrateEvenly(allocatable_tracks, remaining_bitrate, false, 1,
&allocation);
RTC_DCHECK_EQ(allocation.size(), allocatable_tracks.size());
return allocation;
}
// Allocates bitrate to all observers when the available bandwidth is enough
// to allocate the minimum to all observers but not enough to allocate the
// max bitrate of each observer.
// Allocates the bitrate based on the bitrate priority of each observer. This
// bitrate priority defines the priority for bitrate to be allocated to that
// observer in relation to other observers. For example with two observers, if
// observer 1 had a bitrate_priority = 1.0, and observer 2 has a
// bitrate_priority = 2.0, the expected behavior is that observer 2 will be
// allocated twice the bitrate as observer 1 above the each observer's
// min_bitrate_bps values, until one of the observers hits its max_bitrate_bps.
std::map<BitrateAllocatorObserver*, int> NormalRateAllocation(
const std::vector<AllocatableTrack>& allocatable_tracks,
uint32_t bitrate,
uint32_t sum_min_bitrates) {
std::map<BitrateAllocatorObserver*, int> allocation;
std::map<BitrateAllocatorObserver*, int> observers_capacities;
for (const auto& observer_config : allocatable_tracks) {
allocation[observer_config.observer] =
observer_config.config.min_bitrate_bps;
observers_capacities[observer_config.observer] =
observer_config.config.max_bitrate_bps -
observer_config.config.min_bitrate_bps;
}
bitrate -= sum_min_bitrates;
// TODO(srte): Implement fair sharing between prioritized streams, currently
// they are treated on a first come first serve basis.
for (const auto& observer_config : allocatable_tracks) {
int64_t priority_margin = observer_config.config.priority_bitrate_bps -
allocation[observer_config.observer];
if (priority_margin > 0 && bitrate > 0) {
int64_t extra_bitrate = std::min<int64_t>(priority_margin, bitrate);
allocation[observer_config.observer] +=
rtc::dchecked_cast<int>(extra_bitrate);
observers_capacities[observer_config.observer] -= extra_bitrate;
bitrate -= extra_bitrate;
}
}
// From the remaining bitrate, allocate a proportional amount to each observer
// above the min bitrate already allocated.
if (bitrate > 0)
DistributeBitrateRelatively(allocatable_tracks, bitrate,
observers_capacities, &allocation);
return allocation;
}
// Allocates bitrate to observers when there is enough available bandwidth
// for all observers to be allocated their max bitrate.
std::map<BitrateAllocatorObserver*, int> MaxRateAllocation(
const std::vector<AllocatableTrack>& allocatable_tracks,
uint32_t bitrate,
uint32_t sum_max_bitrates) {
std::map<BitrateAllocatorObserver*, int> allocation;
for (const auto& observer_config : allocatable_tracks) {
allocation[observer_config.observer] =
observer_config.config.max_bitrate_bps;
bitrate -= observer_config.config.max_bitrate_bps;
}
DistributeBitrateEvenly(allocatable_tracks, bitrate, true,
kTransmissionMaxBitrateMultiplier, &allocation);
return allocation;
}
// Allocates zero bitrate to all observers.
std::map<BitrateAllocatorObserver*, int> ZeroRateAllocation(
const std::vector<AllocatableTrack>& allocatable_tracks) {
std::map<BitrateAllocatorObserver*, int> allocation;
for (const auto& observer_config : allocatable_tracks)
allocation[observer_config.observer] = 0;
return allocation;
}
std::map<BitrateAllocatorObserver*, int> AllocateBitrates(
const std::vector<AllocatableTrack>& allocatable_tracks,
uint32_t bitrate) {
if (allocatable_tracks.empty())
return std::map<BitrateAllocatorObserver*, int>();
if (bitrate == 0)
return ZeroRateAllocation(allocatable_tracks);
uint32_t sum_min_bitrates = 0;
uint32_t sum_max_bitrates = 0;
for (const auto& observer_config : allocatable_tracks) {
sum_min_bitrates += observer_config.config.min_bitrate_bps;
sum_max_bitrates += observer_config.config.max_bitrate_bps;
}
// Not enough for all observers to get an allocation, allocate according to:
// enforced min bitrate -> allocated bitrate previous round -> restart paused
// streams.
if (!EnoughBitrateForAllObservers(allocatable_tracks, bitrate,
sum_min_bitrates))
return LowRateAllocation(allocatable_tracks, bitrate);
// All observers will get their min bitrate plus a share of the rest. This
// share is allocated to each observer based on its bitrate_priority.
if (bitrate <= sum_max_bitrates)
return NormalRateAllocation(allocatable_tracks, bitrate, sum_min_bitrates);
// All observers will get up to transmission_max_bitrate_multiplier_ x max.
return MaxRateAllocation(allocatable_tracks, bitrate, sum_max_bitrates);
}
} // namespace
BitrateAllocator::BitrateAllocator(LimitObserver* limit_observer)
: limit_observer_(limit_observer),
last_target_bps_(0),
last_stable_target_bps_(0),
last_non_zero_bitrate_bps_(kDefaultBitrateBps),
last_fraction_loss_(0),
last_rtt_(0),
last_bwe_period_ms_(1000),
num_pause_events_(0),
last_bwe_log_time_(0) {
sequenced_checker_.Detach();
}
BitrateAllocator::~BitrateAllocator() {
RTC_HISTOGRAM_COUNTS_100("WebRTC.Call.NumberOfPauseEvents",
num_pause_events_);
}
void BitrateAllocator::UpdateStartRate(uint32_t start_rate_bps) {
RTC_DCHECK_RUN_ON(&sequenced_checker_);
last_non_zero_bitrate_bps_ = start_rate_bps;
}
void BitrateAllocator::OnNetworkEstimateChanged(TargetTransferRate msg) {
RTC_DCHECK_RUN_ON(&sequenced_checker_);
last_target_bps_ = msg.target_rate.bps();
last_stable_target_bps_ = msg.stable_target_rate.bps();
last_non_zero_bitrate_bps_ =
last_target_bps_ > 0 ? last_target_bps_ : last_non_zero_bitrate_bps_;
int loss_ratio_255 = msg.network_estimate.loss_rate_ratio * 255;
last_fraction_loss_ =
rtc::dchecked_cast<uint8_t>(rtc::SafeClamp(loss_ratio_255, 0, 255));
last_rtt_ = msg.network_estimate.round_trip_time.ms();
last_bwe_period_ms_ = msg.network_estimate.bwe_period.ms();
// Periodically log the incoming BWE.
int64_t now = msg.at_time.ms();
if (now > last_bwe_log_time_ + kBweLogIntervalMs) {
RTC_LOG(LS_INFO) << "Current BWE " << last_target_bps_;
last_bwe_log_time_ = now;
}
auto allocation = AllocateBitrates(allocatable_tracks_, last_target_bps_);
auto stable_bitrate_allocation =
AllocateBitrates(allocatable_tracks_, last_stable_target_bps_);
for (auto& config : allocatable_tracks_) {
uint32_t allocated_bitrate = allocation[config.observer];
uint32_t allocated_stable_target_rate =
stable_bitrate_allocation[config.observer];
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::BitsPerSec(allocated_bitrate);
update.stable_target_bitrate =
DataRate::BitsPerSec(allocated_stable_target_rate);
update.packet_loss_ratio = last_fraction_loss_ / 256.0;
Use newer version of TimeDelta and TimeStamp factories in webrtc find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 11:16:00 +01:00
update.round_trip_time = TimeDelta::Millis(last_rtt_);
update.bwe_period = TimeDelta::Millis(last_bwe_period_ms_);
update.cwnd_reduce_ratio = msg.cwnd_reduce_ratio;
uint32_t protection_bitrate = config.observer->OnBitrateUpdated(update);
if (allocated_bitrate == 0 && config.allocated_bitrate_bps > 0) {
if (last_target_bps_ > 0)
++num_pause_events_;
// The protection bitrate is an estimate based on the ratio between media
// and protection used before this observer was muted.
uint32_t predicted_protection_bps =
(1.0 - config.media_ratio) * config.config.min_bitrate_bps;
RTC_LOG(LS_INFO) << "Pausing observer " << config.observer
<< " with configured min bitrate "
<< config.config.min_bitrate_bps
<< " and current estimate of " << last_target_bps_
<< " and protection bitrate "
<< predicted_protection_bps;
} else if (allocated_bitrate > 0 && config.allocated_bitrate_bps == 0) {
if (last_target_bps_ > 0)
++num_pause_events_;
RTC_LOG(LS_INFO) << "Resuming observer " << config.observer
<< ", configured min bitrate "
<< config.config.min_bitrate_bps
<< ", current allocation " << allocated_bitrate
<< " and protection bitrate " << protection_bitrate;
}
// Only update the media ratio if the observer got an allocation.
if (allocated_bitrate > 0)
config.media_ratio = MediaRatio(allocated_bitrate, protection_bitrate);
config.allocated_bitrate_bps = allocated_bitrate;
}
Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ ) Reason for revert: Downstream fixed Original issue's description: > Revert of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #8 id:120001 of https://codereview.webrtc.org/2705603002/ ) > > Reason for revert: > Breaks downstream > > Original issue's description: > > Fixes a bug where a video stream can get stuck in the suspended state. > > > > This happens if a lot of FEC is allocated when the stream becomes suspended. The required bitrate to unsuspend can then be too high so that the padding bitrate we are allowed to generate is not enough. > > > > This CL also switches the tests from using ISAC to OPUS as RampUpTest.UpDownUpAudioVideoTransportSequenceNumberRtx relies on audio BWE to work (which is only compatible with OPUS). I don't know why it didn't fail before. > > > > BUG=webrtc:7178 > > > > Review-Url: https://codereview.webrtc.org/2705603002 > > Cr-Commit-Position: refs/heads/master@{#16739} > > Committed: https://chromium.googlesource.com/external/webrtc/+/a518a39963d34616d8f0e94991c7f5fbb5affb38 > > TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7178 > > Review-Url: https://codereview.webrtc.org/2703393002 > Cr-Commit-Position: refs/heads/master@{#16751} > Committed: https://chromium.googlesource.com/external/webrtc/+/b80bdcafed6c529be140da7d9f3e95a00b94219e TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7178 Review-Url: https://codereview.webrtc.org/2704323003 Cr-Commit-Position: refs/heads/master@{#16753}
2017-02-21 07:28:31 -08:00
UpdateAllocationLimits();
}
void BitrateAllocator::AddObserver(BitrateAllocatorObserver* observer,
MediaStreamAllocationConfig config) {
RTC_DCHECK_RUN_ON(&sequenced_checker_);
RTC_DCHECK_GT(config.bitrate_priority, 0);
RTC_DCHECK(std::isnormal(config.bitrate_priority));
auto it = absl::c_find_if(
allocatable_tracks_,
[observer](const auto& config) { return config.observer == observer; });
// Update settings if the observer already exists, create a new one otherwise.
if (it != allocatable_tracks_.end()) {
it->config = config;
} else {
allocatable_tracks_.push_back(AllocatableTrack(observer, config));
}
if (last_target_bps_ > 0) {
// Calculate a new allocation and update all observers.
auto allocation = AllocateBitrates(allocatable_tracks_, last_target_bps_);
auto stable_bitrate_allocation =
AllocateBitrates(allocatable_tracks_, last_stable_target_bps_);
for (auto& config : allocatable_tracks_) {
uint32_t allocated_bitrate = allocation[config.observer];
uint32_t allocated_stable_bitrate =
stable_bitrate_allocation[config.observer];
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::BitsPerSec(allocated_bitrate);
update.stable_target_bitrate =
DataRate::BitsPerSec(allocated_stable_bitrate);
update.packet_loss_ratio = last_fraction_loss_ / 256.0;
Use newer version of TimeDelta and TimeStamp factories in webrtc find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 11:16:00 +01:00
update.round_trip_time = TimeDelta::Millis(last_rtt_);
update.bwe_period = TimeDelta::Millis(last_bwe_period_ms_);
uint32_t protection_bitrate = config.observer->OnBitrateUpdated(update);
config.allocated_bitrate_bps = allocated_bitrate;
if (allocated_bitrate > 0)
config.media_ratio = MediaRatio(allocated_bitrate, protection_bitrate);
}
} else {
// Currently, an encoder is not allowed to produce frames.
// But we still have to return the initial config bitrate + let the
// observer know that it can not produce frames.
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::Zero();
update.stable_target_bitrate = DataRate::Zero();
update.packet_loss_ratio = last_fraction_loss_ / 256.0;
Use newer version of TimeDelta and TimeStamp factories in webrtc find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 11:16:00 +01:00
update.round_trip_time = TimeDelta::Millis(last_rtt_);
update.bwe_period = TimeDelta::Millis(last_bwe_period_ms_);
observer->OnBitrateUpdated(update);
}
UpdateAllocationLimits();
}
void BitrateAllocator::UpdateAllocationLimits() {
BitrateAllocationLimits limits;
for (const auto& config : allocatable_tracks_) {
uint32_t stream_padding = config.config.pad_up_bitrate_bps;
if (config.config.enforce_min_bitrate) {
limits.min_allocatable_rate +=
DataRate::BitsPerSec(config.config.min_bitrate_bps);
Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ ) Reason for revert: Downstream fixed Original issue's description: > Revert of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #8 id:120001 of https://codereview.webrtc.org/2705603002/ ) > > Reason for revert: > Breaks downstream > > Original issue's description: > > Fixes a bug where a video stream can get stuck in the suspended state. > > > > This happens if a lot of FEC is allocated when the stream becomes suspended. The required bitrate to unsuspend can then be too high so that the padding bitrate we are allowed to generate is not enough. > > > > This CL also switches the tests from using ISAC to OPUS as RampUpTest.UpDownUpAudioVideoTransportSequenceNumberRtx relies on audio BWE to work (which is only compatible with OPUS). I don't know why it didn't fail before. > > > > BUG=webrtc:7178 > > > > Review-Url: https://codereview.webrtc.org/2705603002 > > Cr-Commit-Position: refs/heads/master@{#16739} > > Committed: https://chromium.googlesource.com/external/webrtc/+/a518a39963d34616d8f0e94991c7f5fbb5affb38 > > TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7178 > > Review-Url: https://codereview.webrtc.org/2703393002 > Cr-Commit-Position: refs/heads/master@{#16751} > Committed: https://chromium.googlesource.com/external/webrtc/+/b80bdcafed6c529be140da7d9f3e95a00b94219e TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7178 Review-Url: https://codereview.webrtc.org/2704323003 Cr-Commit-Position: refs/heads/master@{#16753}
2017-02-21 07:28:31 -08:00
} else if (config.allocated_bitrate_bps == 0) {
stream_padding =
std::max(config.MinBitrateWithHysteresis(), stream_padding);
}
limits.max_padding_rate += DataRate::BitsPerSec(stream_padding);
limits.max_allocatable_rate +=
DataRate::BitsPerSec(config.config.max_bitrate_bps);
Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ ) Reason for revert: Downstream fixed Original issue's description: > Revert of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #8 id:120001 of https://codereview.webrtc.org/2705603002/ ) > > Reason for revert: > Breaks downstream > > Original issue's description: > > Fixes a bug where a video stream can get stuck in the suspended state. > > > > This happens if a lot of FEC is allocated when the stream becomes suspended. The required bitrate to unsuspend can then be too high so that the padding bitrate we are allowed to generate is not enough. > > > > This CL also switches the tests from using ISAC to OPUS as RampUpTest.UpDownUpAudioVideoTransportSequenceNumberRtx relies on audio BWE to work (which is only compatible with OPUS). I don't know why it didn't fail before. > > > > BUG=webrtc:7178 > > > > Review-Url: https://codereview.webrtc.org/2705603002 > > Cr-Commit-Position: refs/heads/master@{#16739} > > Committed: https://chromium.googlesource.com/external/webrtc/+/a518a39963d34616d8f0e94991c7f5fbb5affb38 > > TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7178 > > Review-Url: https://codereview.webrtc.org/2703393002 > Cr-Commit-Position: refs/heads/master@{#16751} > Committed: https://chromium.googlesource.com/external/webrtc/+/b80bdcafed6c529be140da7d9f3e95a00b94219e TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7178 Review-Url: https://codereview.webrtc.org/2704323003 Cr-Commit-Position: refs/heads/master@{#16753}
2017-02-21 07:28:31 -08:00
}
if (limits.min_allocatable_rate == current_limits_.min_allocatable_rate &&
limits.max_allocatable_rate == current_limits_.max_allocatable_rate &&
limits.max_padding_rate == current_limits_.max_padding_rate) {
Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ ) Reason for revert: Downstream fixed Original issue's description: > Revert of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #8 id:120001 of https://codereview.webrtc.org/2705603002/ ) > > Reason for revert: > Breaks downstream > > Original issue's description: > > Fixes a bug where a video stream can get stuck in the suspended state. > > > > This happens if a lot of FEC is allocated when the stream becomes suspended. The required bitrate to unsuspend can then be too high so that the padding bitrate we are allowed to generate is not enough. > > > > This CL also switches the tests from using ISAC to OPUS as RampUpTest.UpDownUpAudioVideoTransportSequenceNumberRtx relies on audio BWE to work (which is only compatible with OPUS). I don't know why it didn't fail before. > > > > BUG=webrtc:7178 > > > > Review-Url: https://codereview.webrtc.org/2705603002 > > Cr-Commit-Position: refs/heads/master@{#16739} > > Committed: https://chromium.googlesource.com/external/webrtc/+/a518a39963d34616d8f0e94991c7f5fbb5affb38 > > TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7178 > > Review-Url: https://codereview.webrtc.org/2703393002 > Cr-Commit-Position: refs/heads/master@{#16751} > Committed: https://chromium.googlesource.com/external/webrtc/+/b80bdcafed6c529be140da7d9f3e95a00b94219e TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7178 Review-Url: https://codereview.webrtc.org/2704323003 Cr-Commit-Position: refs/heads/master@{#16753}
2017-02-21 07:28:31 -08:00
return;
}
current_limits_ = limits;
Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ ) Reason for revert: Downstream fixed Original issue's description: > Revert of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #8 id:120001 of https://codereview.webrtc.org/2705603002/ ) > > Reason for revert: > Breaks downstream > > Original issue's description: > > Fixes a bug where a video stream can get stuck in the suspended state. > > > > This happens if a lot of FEC is allocated when the stream becomes suspended. The required bitrate to unsuspend can then be too high so that the padding bitrate we are allowed to generate is not enough. > > > > This CL also switches the tests from using ISAC to OPUS as RampUpTest.UpDownUpAudioVideoTransportSequenceNumberRtx relies on audio BWE to work (which is only compatible with OPUS). I don't know why it didn't fail before. > > > > BUG=webrtc:7178 > > > > Review-Url: https://codereview.webrtc.org/2705603002 > > Cr-Commit-Position: refs/heads/master@{#16739} > > Committed: https://chromium.googlesource.com/external/webrtc/+/a518a39963d34616d8f0e94991c7f5fbb5affb38 > > TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7178 > > Review-Url: https://codereview.webrtc.org/2703393002 > Cr-Commit-Position: refs/heads/master@{#16751} > Committed: https://chromium.googlesource.com/external/webrtc/+/b80bdcafed6c529be140da7d9f3e95a00b94219e TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7178 Review-Url: https://codereview.webrtc.org/2704323003 Cr-Commit-Position: refs/heads/master@{#16753}
2017-02-21 07:28:31 -08:00
RTC_LOG(LS_INFO) << "UpdateAllocationLimits : total_requested_min_bitrate: "
<< ToString(limits.min_allocatable_rate)
<< ", total_requested_padding_bitrate: "
<< ToString(limits.max_padding_rate)
<< ", total_requested_max_bitrate: "
<< ToString(limits.max_allocatable_rate);
limit_observer_->OnAllocationLimitsChanged(limits);
}
void BitrateAllocator::RemoveObserver(BitrateAllocatorObserver* observer) {
RTC_DCHECK_RUN_ON(&sequenced_checker_);
for (auto it = allocatable_tracks_.begin(); it != allocatable_tracks_.end();
++it) {
if (it->observer == observer) {
allocatable_tracks_.erase(it);
break;
}
}
UpdateAllocationLimits();
}
int BitrateAllocator::GetStartBitrate(
BitrateAllocatorObserver* observer) const {
RTC_DCHECK_RUN_ON(&sequenced_checker_);
auto it = absl::c_find_if(
allocatable_tracks_,
[observer](const auto& config) { return config.observer == observer; });
if (it == allocatable_tracks_.end()) {
// This observer hasn't been added yet, just give it its fair share.
return last_non_zero_bitrate_bps_ /
static_cast<int>((allocatable_tracks_.size() + 1));
} else if (it->allocated_bitrate_bps == -1) {
// This observer hasn't received an allocation yet, so do the same.
return last_non_zero_bitrate_bps_ /
static_cast<int>(allocatable_tracks_.size());
} else {
// This observer already has an allocation.
return it->allocated_bitrate_bps;
}
}
uint32_t bitrate_allocator_impl::AllocatableTrack::LastAllocatedBitrate()
const {
// Return the configured minimum bitrate for newly added observers, to avoid
// requiring an extra high bitrate for the observer to get an allocated
// bitrate.
return allocated_bitrate_bps == -1 ? config.min_bitrate_bps
: allocated_bitrate_bps;
}
uint32_t bitrate_allocator_impl::AllocatableTrack::MinBitrateWithHysteresis()
const {
uint32_t min_bitrate = config.min_bitrate_bps;
if (LastAllocatedBitrate() == 0) {
min_bitrate += std::max(static_cast<uint32_t>(kToggleFactor * min_bitrate),
kMinToggleBitrateBps);
}
// Account for protection bitrate used by this observer in the previous
// allocation.
// Note: the ratio will only be updated when the stream is active, meaning a
// paused stream won't get any ratio updates. This might lead to waiting a bit
// longer than necessary if the network condition improves, but this is to
// avoid too much toggling.
if (media_ratio > 0.0 && media_ratio < 1.0)
min_bitrate += min_bitrate * (1.0 - media_ratio);
return min_bitrate;
}
} // namespace webrtc