Revert "BWE allocation strategy"
This reverts commit a5fbc23379823d74b8cf4bc18887ff40237989e8.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> BWE allocation strategy
>
> This is reland of https://webrtc-review.googlesource.com/c/src/+/4860 with the fixed RampUpTest test
>
> Bug: webrtc:8243
> Change-Id: I4b90a449b00dd05feee974001e08fb40710b59ac
> Reviewed-on: https://webrtc-review.googlesource.com/13124
> Commit-Queue: Alex Narest <alexnarest@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20345}
TBR=stefan@webrtc.org,alexnarest@webrtc.org
Change-Id: I8ed12cd2115ef63204e384cc93c9f4473daa54d1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8243
Reviewed-on: https://webrtc-review.googlesource.com/14020
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20361}diff --git a/api/peerconnectioninterface.h b/api/peerconnectioninterface.h
index 839b0f4..dd1f87c 100644
--- a/api/peerconnectioninterface.h
+++ b/api/peerconnectioninterface.h
@@ -786,14 +786,6 @@
// to the provided value.
virtual RTCError SetBitrate(const BitrateParameters& bitrate) = 0;
- // Sets current strategy. If not set default WebRTC allocator will be used.
- // May be changed during an active session. The strategy
- // ownership is passed with std::unique_ptr
- // TODO(alexnarest): Make this pure virtual when tests will be updated
- virtual void SetBitrateAllocationStrategy(
- std::unique_ptr<rtc::BitrateAllocationStrategy>
- bitrate_allocation_strategy) {}
-
// Returns the current SignalingState.
virtual SignalingState signaling_state() = 0;
virtual IceConnectionState ice_connection_state() = 0;
diff --git a/api/peerconnectionproxy.h b/api/peerconnectionproxy.h
index a8ea3fa3..c490b8f 100644
--- a/api/peerconnectionproxy.h
+++ b/api/peerconnectionproxy.h
@@ -102,9 +102,6 @@
const std::vector<cricket::Candidate>&);
PROXY_METHOD1(void, RegisterUMAObserver, UMAObserver*)
PROXY_METHOD1(RTCError, SetBitrate, const BitrateParameters&);
- PROXY_METHOD1(void,
- SetBitrateAllocationStrategy,
- std::unique_ptr<rtc::BitrateAllocationStrategy>);
PROXY_METHOD0(SignalingState, signaling_state)
PROXY_METHOD0(IceConnectionState, ice_connection_state)
PROXY_METHOD0(IceGatheringState, ice_gathering_state)
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 7895af3..1cfc0ba 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -236,8 +236,6 @@
void AudioSendStream::Start() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1) {
- // Audio BWE is enabled.
- transport_->packet_sender()->SetAccountForAudioPackets(true);
ConfigureBitrateObserver(config_.min_bitrate_bps, config_.max_bitrate_bps);
}
diff --git a/call/bitrate_allocator.cc b/call/bitrate_allocator.cc
index e186380..b379f7f 100644
--- a/call/bitrate_allocator.cc
+++ b/call/bitrate_allocator.cc
@@ -12,7 +12,6 @@
#include "call/bitrate_allocator.h"
#include <algorithm>
-#include <memory>
#include <utility>
#include "modules/bitrate_controller/include/bitrate_controller.h"
@@ -57,8 +56,7 @@
clock_(Clock::GetRealTimeClock()),
last_bwe_log_time_(0),
total_requested_padding_bitrate_(0),
- total_requested_min_bitrate_(0),
- bitrate_allocation_strategy_(nullptr) {
+ total_requested_min_bitrate_(0) {
sequenced_checker_.Detach();
}
@@ -201,7 +199,6 @@
void BitrateAllocator::RemoveObserver(BitrateAllocatorObserver* observer) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
-
auto it = FindObserverConfig(observer);
if (it != bitrate_observer_configs_.end()) {
bitrate_observer_configs_.erase(it);
@@ -227,13 +224,6 @@
}
}
-void BitrateAllocator::SetBitrateAllocationStrategy(
- std::unique_ptr<rtc::BitrateAllocationStrategy>
- bitrate_allocation_strategy) {
- RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
- bitrate_allocation_strategy_ = std::move(bitrate_allocation_strategy);
-}
-
BitrateAllocator::ObserverConfigs::iterator
BitrateAllocator::FindObserverConfig(const BitrateAllocatorObserver* observer) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
@@ -251,25 +241,6 @@
if (bitrate_observer_configs_.empty())
return ObserverAllocation();
- if (bitrate_allocation_strategy_ != nullptr) {
- std::vector<const rtc::BitrateAllocationStrategy::TrackConfig*>
- track_configs(bitrate_observer_configs_.size());
- int i = 0;
- for (const auto& c : bitrate_observer_configs_) {
- track_configs[i++] = &c;
- }
- std::vector<uint32_t> track_allocations =
- bitrate_allocation_strategy_->AllocateBitrates(bitrate, track_configs);
- // The strategy should return allocation for all tracks.
- RTC_CHECK(track_allocations.size() == bitrate_observer_configs_.size());
- ObserverAllocation allocation;
- auto track_allocations_it = track_allocations.begin();
- for (const auto& observer_config : bitrate_observer_configs_) {
- allocation[observer_config.observer] = *track_allocations_it++;
- }
- return allocation;
- }
-
if (bitrate == 0)
return ZeroRateAllocation();
diff --git a/call/bitrate_allocator.h b/call/bitrate_allocator.h
index cf518f6..d51740c 100644
--- a/call/bitrate_allocator.h
+++ b/call/bitrate_allocator.h
@@ -14,12 +14,10 @@
#include <stdint.h>
#include <map>
-#include <memory>
#include <string>
#include <utility>
#include <vector>
-#include "rtc_base/bitrateallocationstrategy.h"
#include "rtc_base/sequenced_task_checker.h"
namespace webrtc {
@@ -96,35 +94,32 @@
// the list of added observers, a best guess is returned.
int GetStartBitrate(BitrateAllocatorObserver* observer);
- // Sets external allocation strategy. If strategy is not set default WebRTC
- // allocation mechanism will be used. The strategy may be changed during call.
- // Setting NULL value will restore default WEBRTC allocation strategy.
- void SetBitrateAllocationStrategy(
- std::unique_ptr<rtc::BitrateAllocationStrategy>
- bitrate_allocation_strategy);
-
private:
// Note: All bitrates for member variables and methods are in bps.
- struct ObserverConfig : rtc::BitrateAllocationStrategy::TrackConfig {
+ struct ObserverConfig {
ObserverConfig(BitrateAllocatorObserver* observer,
uint32_t min_bitrate_bps,
uint32_t max_bitrate_bps,
uint32_t pad_up_bitrate_bps,
bool enforce_min_bitrate,
std::string track_id)
- : TrackConfig(min_bitrate_bps,
- max_bitrate_bps,
- enforce_min_bitrate,
- track_id),
- observer(observer),
+ : observer(observer),
+ min_bitrate_bps(min_bitrate_bps),
+ max_bitrate_bps(max_bitrate_bps),
pad_up_bitrate_bps(pad_up_bitrate_bps),
+ enforce_min_bitrate(enforce_min_bitrate),
allocated_bitrate_bps(-1),
- media_ratio(1.0) {}
+ media_ratio(1.0),
+ track_id(track_id) {}
BitrateAllocatorObserver* observer;
+ uint32_t min_bitrate_bps;
+ uint32_t max_bitrate_bps;
uint32_t pad_up_bitrate_bps;
+ bool enforce_min_bitrate;
int64_t allocated_bitrate_bps;
double media_ratio; // Part of the total bitrate used for media [0.0, 1.0].
+ std::string track_id;
};
// Calculates the minimum requested send bitrate and max padding bitrate and
@@ -177,8 +172,6 @@
int64_t last_bwe_log_time_ RTC_GUARDED_BY(&sequenced_checker_);
uint32_t total_requested_padding_bitrate_ RTC_GUARDED_BY(&sequenced_checker_);
uint32_t total_requested_min_bitrate_ RTC_GUARDED_BY(&sequenced_checker_);
- std::unique_ptr<rtc::BitrateAllocationStrategy> bitrate_allocation_strategy_
- RTC_GUARDED_BY(&sequenced_checker_);
};
} // namespace webrtc
#endif // CALL_BITRATE_ALLOCATOR_H_
diff --git a/call/call.cc b/call/call.cc
index 90aaba3..215c103 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -214,10 +214,6 @@
void SetBitrateConfigMask(
const webrtc::Call::Config::BitrateConfigMask& bitrate_config) override;
- void SetBitrateAllocationStrategy(
- std::unique_ptr<rtc::BitrateAllocationStrategy>
- bitrate_allocation_strategy) override;
-
void SignalChannelNetworkState(MediaType media, NetworkState state) override;
void OnTransportOverheadChanged(MediaType media,
@@ -1011,24 +1007,6 @@
config_.bitrate_config = updated;
}
-void Call::SetBitrateAllocationStrategy(
- std::unique_ptr<rtc::BitrateAllocationStrategy>
- bitrate_allocation_strategy) {
- if (!worker_queue_.IsCurrent()) {
- rtc::BitrateAllocationStrategy* strategy_raw =
- bitrate_allocation_strategy.release();
- auto functor = [this, strategy_raw]() {
- SetBitrateAllocationStrategy(
- rtc::WrapUnique<rtc::BitrateAllocationStrategy>(strategy_raw));
- };
- worker_queue_.PostTask([functor] { functor(); });
- return;
- }
- RTC_DCHECK_RUN_ON(&worker_queue_);
- bitrate_allocator_->SetBitrateAllocationStrategy(
- std::move(bitrate_allocation_strategy));
-}
-
void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
switch (media) {
diff --git a/call/call.h b/call/call.h
index 4de5b55..f26d9d6 100644
--- a/call/call.h
+++ b/call/call.h
@@ -24,7 +24,6 @@
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "common_types.h" // NOLINT(build/include)
-#include "rtc_base/bitrateallocationstrategy.h"
#include "rtc_base/networkroute.h"
#include "rtc_base/platform_file.h"
#include "rtc_base/socket.h"
@@ -184,10 +183,6 @@
virtual void SetBitrateConfigMask(
const Config::BitrateConfigMask& bitrate_mask) = 0;
- virtual void SetBitrateAllocationStrategy(
- std::unique_ptr<rtc::BitrateAllocationStrategy>
- bitrate_allocation_strategy) = 0;
-
// TODO(skvlad): When the unbundled case with multiple streams for the same
// media type going over different networks is supported, track the state
// for each stream separately. Right now it's global per media type.
diff --git a/call/rampup_tests.cc b/call/rampup_tests.cc
index 1d58419..85346b74 100644
--- a/call/rampup_tests.cc
+++ b/call/rampup_tests.cc
@@ -418,21 +418,19 @@
void RampUpDownUpTester::PollStats() {
do {
- int transmit_bitrate_bps = 0;
- bool suspended = false;
- if (num_video_streams_ > 0) {
+ if (send_stream_) {
webrtc::VideoSendStream::Stats stats = send_stream_->GetStats();
+ int transmit_bitrate_bps = 0;
for (auto it : stats.substreams) {
transmit_bitrate_bps += it.second.total_bitrate_bps;
}
- suspended = stats.suspended;
- }
- if (num_audio_streams_ > 0 && sender_call_ != nullptr) {
+ EvolveTestState(transmit_bitrate_bps, stats.suspended);
+ } else if (num_audio_streams_ > 0 && sender_call_ != nullptr) {
// An audio send stream doesn't have bitrate stats, so the call send BW is
// currently used instead.
- transmit_bitrate_bps = sender_call_->GetStats().send_bandwidth_bps;
+ int transmit_bitrate_bps = sender_call_->GetStats().send_bandwidth_bps;
+ EvolveTestState(transmit_bitrate_bps, false);
}
- EvolveTestState(transmit_bitrate_bps, suspended);
} while (!stop_event_.Wait(kPollIntervalMs));
}
diff --git a/examples/BUILD.gn b/examples/BUILD.gn
index b7474d6..0041d04 100644
--- a/examples/BUILD.gn
+++ b/examples/BUILD.gn
@@ -176,7 +176,6 @@
if (is_ios) {
deps = [
":AppRTCMobile_ios_frameworks",
- "../rtc_base:rtc_base",
]
} else {
deps = [
@@ -207,8 +206,6 @@
"objc/AppRTCMobile/ARDAppClient.m",
"objc/AppRTCMobile/ARDAppEngineClient.h",
"objc/AppRTCMobile/ARDAppEngineClient.m",
- "objc/AppRTCMobile/ARDBitrateAllocationStrategy.h",
- "objc/AppRTCMobile/ARDBitrateAllocationStrategy.mm",
"objc/AppRTCMobile/ARDBitrateTracker.h",
"objc/AppRTCMobile/ARDBitrateTracker.m",
"objc/AppRTCMobile/ARDCaptureController.h",
@@ -252,7 +249,6 @@
deps = [
":apprtc_common",
":socketrocket",
- "../rtc_base:rtc_base",
]
if (is_ios) {
deps += [ ":AppRTCMobile_ios_frameworks" ]
diff --git a/examples/objc/AppRTCMobile/ARDAppClient.m b/examples/objc/AppRTCMobile/ARDAppClient.m
index f606de0..8e933b9 100644
--- a/examples/objc/AppRTCMobile/ARDAppClient.m
+++ b/examples/objc/AppRTCMobile/ARDAppClient.m
@@ -25,7 +25,6 @@
#import "WebRTC/RTCVideoTrack.h"
#import "ARDAppEngineClient.h"
-#import "ARDBitrateAllocationStrategy.h"
#import "ARDJoinResponse.h"
#import "ARDMessageResponse.h"
#import "ARDSettingsModel.h"
@@ -51,7 +50,6 @@
static NSString * const kARDAudioTrackId = @"ARDAMSa0";
static NSString * const kARDVideoTrackId = @"ARDAMSv0";
static NSString * const kARDVideoTrackKind = @"video";
-static uint32_t const kSufficientAudioBitrate = 16000;
// TODO(tkchin): Add these as UI options.
static BOOL const kARDAppClientEnableTracing = NO;
@@ -106,7 +104,6 @@
ARDTimerProxy *_statsTimer;
ARDSettingsModel *_settings;
RTCVideoTrack *_localVideoTrack;
- ARDBitrateAllocationStrategy *_bitrateAllocationStrategy;
}
@synthesize shouldGetStats = _shouldGetStats;
@@ -309,14 +306,12 @@
_hasReceivedSdp = NO;
_messageQueue = [NSMutableArray array];
_localVideoTrack = nil;
-
#if defined(WEBRTC_IOS)
[_factory stopAecDump];
[_peerConnection stopRtcEventLog];
#endif
[_peerConnection close];
_peerConnection = nil;
- _bitrateAllocationStrategy = nil;
self.state = kARDAppClientStateDisconnected;
#if defined(WEBRTC_IOS)
if (kARDAppClientEnableTracing) {
@@ -538,14 +533,8 @@
_peerConnection = [_factory peerConnectionWithConfiguration:config
constraints:constraints
delegate:self];
- _bitrateAllocationStrategy = [ARDBitrateAllocationStrategy
- createAudioPriorityBitrateAllocationStrategyForPeerConnection:_peerConnection
- withAudioTrack:kARDAudioTrackId
- sufficientAudioBitrate:kSufficientAudioBitrate];
-
// Create AV senders.
[self createMediaSenders];
-
if (_isInitiator) {
// Send offer.
__weak ARDAppClient *weakSelf = self;
diff --git a/examples/objc/AppRTCMobile/ARDBitrateAllocationStrategy.h b/examples/objc/AppRTCMobile/ARDBitrateAllocationStrategy.h
deleted file mode 100644
index 18f83ea..0000000
--- a/examples/objc/AppRTCMobile/ARDBitrateAllocationStrategy.h
+++ /dev/null
@@ -1,23 +0,0 @@
-/*
- * Copyright 2017 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.
- */
-
-#import <Foundation/Foundation.h>
-#import "WebRTC/RTCPeerConnection.h"
-
-@interface ARDBitrateAllocationStrategy : NSObject
-
-+ (ARDBitrateAllocationStrategy*)
- createAudioPriorityBitrateAllocationStrategyForPeerConnection:(RTCPeerConnection*)peerConnection
- withAudioTrack:(NSString*)audioTrackID
- sufficientAudioBitrate:(uint32_t)sufficientAudioBitrate;
-
-- (instancetype)init NS_UNAVAILABLE;
-
-@end
diff --git a/examples/objc/AppRTCMobile/ARDBitrateAllocationStrategy.mm b/examples/objc/AppRTCMobile/ARDBitrateAllocationStrategy.mm
deleted file mode 100644
index 5901f8c..0000000
--- a/examples/objc/AppRTCMobile/ARDBitrateAllocationStrategy.mm
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright 2017 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.
- */
-
-#import "ARDBitrateAllocationStrategy.h"
-#import "WebRTC/RTCBitrateAllocationStrategy.h"
-
-#include "rtc_base/bitrateallocationstrategy.h"
-
-@implementation ARDBitrateAllocationStrategy
-
-+ (ARDBitrateAllocationStrategy*)
- createAudioPriorityBitrateAllocationStrategyForPeerConnection:(RTCPeerConnection*)peerConnection
- withAudioTrack:(NSString*)audioTrackID
- sufficientAudioBitrate:(uint32_t)sufficientAudioBitrate {
- return [[ARDBitrateAllocationStrategy alloc] initWithPeerCoonnection:peerConnection
- withAudioTrack:audioTrackID
- sufficientAudioBitrate:sufficientAudioBitrate];
-}
-
-- (instancetype)initWithPeerCoonnection:(RTCPeerConnection*)peerConnection
- withAudioTrack:(NSString*)audioTrackID
- sufficientAudioBitrate:(uint32_t)sufficientAudioBitrate {
- if (self = [super init]) {
- [peerConnection
- setBitrateAllocationStrategy:[[RTCBitrateAllocationStrategy alloc]
- initWith:new rtc::AudioPriorityBitrateAllocationStrategy(
- std::string(audioTrackID.UTF8String),
- sufficientAudioBitrate)]];
- }
- return self;
-}
-
-@end
diff --git a/examples/objc/AppRTCMobile/ios/ARDAppDelegate.m b/examples/objc/AppRTCMobile/ios/ARDAppDelegate.m
index 20ae170..07c83a0 100644
--- a/examples/objc/AppRTCMobile/ios/ARDAppDelegate.m
+++ b/examples/objc/AppRTCMobile/ios/ARDAppDelegate.m
@@ -26,8 +26,7 @@
- (BOOL)application:(UIApplication *)application
didFinishLaunchingWithOptions:(NSDictionary *)launchOptions {
NSDictionary *fieldTrials = @{
- kRTCFieldTrialH264HighProfileKey : kRTCFieldTrialEnabledValue,
- kRTCFieldTrialAudioSendSideBweKey : kRTCFieldTrialEnabledValue
+ kRTCFieldTrialH264HighProfileKey: kRTCFieldTrialEnabledValue,
};
RTCInitFieldTrialDictionary(fieldTrials);
RTCInitializeSSL();
diff --git a/media/engine/fakewebrtccall.cc b/media/engine/fakewebrtccall.cc
index d43c7ba..8556cc7 100644
--- a/media/engine/fakewebrtccall.cc
+++ b/media/engine/fakewebrtccall.cc
@@ -595,12 +595,6 @@
// TODO(zstein): not implemented
}
-void FakeCall::SetBitrateAllocationStrategy(
- std::unique_ptr<rtc::BitrateAllocationStrategy>
- bitrate_allocation_strategy){
- // TODO(alexnarest): not implemented
-};
-
void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,
webrtc::NetworkState state) {
switch (media) {
diff --git a/media/engine/fakewebrtccall.h b/media/engine/fakewebrtccall.h
index 3d0825d..04dc5d9 100644
--- a/media/engine/fakewebrtccall.h
+++ b/media/engine/fakewebrtccall.h
@@ -296,9 +296,6 @@
const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
void SetBitrateConfigMask(
const webrtc::Call::Config::BitrateConfigMask& mask) override;
- void SetBitrateAllocationStrategy(
- std::unique_ptr<rtc::BitrateAllocationStrategy>
- bitrate_allocation_strategy) override;
void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) override {}
void SignalChannelNetworkState(webrtc::MediaType media,
diff --git a/modules/pacing/paced_sender.cc b/modules/pacing/paced_sender.cc
index 3889631..c122f86 100644
--- a/modules/pacing/paced_sender.cc
+++ b/modules/pacing/paced_sender.cc
@@ -63,8 +63,7 @@
packets_(rtc::MakeUnique<PacketQueue>(clock)),
packet_counter_(0),
pacing_factor_(kDefaultPaceMultiplier),
- queue_time_limit(kMaxQueueLengthMs),
- account_for_audio_(false) {
+ queue_time_limit(kMaxQueueLengthMs) {
UpdateBudgetWithElapsedTime(kMinPacketLimitMs);
}
@@ -155,11 +154,6 @@
retransmission, packet_counter_++));
}
-void PacedSender::SetAccountForAudioPackets(bool account_for_audio) {
- rtc::CritScope cs(&critsect_);
- account_for_audio_ = account_for_audio;
-}
-
int64_t PacedSender::ExpectedQueueTimeMs() const {
rtc::CritScope cs(&critsect_);
RTC_DCHECK_GT(pacing_bitrate_kbps_, 0);
@@ -324,7 +318,9 @@
critsect_.Enter();
if (success) {
- if (packet.priority != kHighPriority || account_for_audio_) {
+ // TODO(holmer): High priority packets should only be accounted for if we
+ // are allocating bandwidth for audio.
+ if (packet.priority != kHighPriority) {
// Update media bytes sent.
// TODO(eladalon): TimeToSendPacket() can also return |true| in some
// situations where nothing actually ended up being sent to the network,
diff --git a/modules/pacing/paced_sender.h b/modules/pacing/paced_sender.h
index 8e47161..3c5ad60 100644
--- a/modules/pacing/paced_sender.h
+++ b/modules/pacing/paced_sender.h
@@ -107,12 +107,6 @@
size_t bytes,
bool retransmission) override;
- // Currently audio traffic is not accounted by pacer and passed through.
- // With the introduction of audio BWE audio traffic will be accounted for
- // the pacer budget calculation. The audio traffic still will be injected
- // at high priority.
- void SetAccountForAudioPackets(bool account_for_audio) override;
-
// Returns the time since the oldest queued packet was enqueued.
virtual int64_t QueueInMs() const;
@@ -198,7 +192,6 @@
float pacing_factor_ RTC_GUARDED_BY(critsect_);
int64_t queue_time_limit RTC_GUARDED_BY(critsect_);
- bool account_for_audio_ RTC_GUARDED_BY(critsect_);
};
} // namespace webrtc
#endif // MODULES_PACING_PACED_SENDER_H_
diff --git a/modules/remote_bitrate_estimator/test/bbr_paced_sender.h b/modules/remote_bitrate_estimator/test/bbr_paced_sender.h
index 096f9d6..6cb45e3 100644
--- a/modules/remote_bitrate_estimator/test/bbr_paced_sender.h
+++ b/modules/remote_bitrate_estimator/test/bbr_paced_sender.h
@@ -68,7 +68,6 @@
int64_t capture_time_ms,
size_t bytes,
bool retransmission) override;
- void SetAccountForAudioPackets(bool account_for_audio) override {}
int64_t TimeUntilNextProcess() override;
void OnBytesAcked(size_t bytes) override;
void Process() override;
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 9023d68..1e25f28 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -498,14 +498,6 @@
int64_t capture_time_ms,
size_t bytes,
bool retransmission) = 0;
-
- // Currently audio traffic is not accounted by pacer and passed through.
- // With the introduction of audio BWE audio traffic will be accounted for
- // the pacer budget calculation. The audio traffic still will be injected
- // at high priority.
- // TODO(alexnarest): Make it pure virtual after rtp_sender_unittest will be
- // updated to support it
- virtual void SetAccountForAudioPackets(bool account_for_audio) {}
};
class TransportSequenceNumberAllocator {
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index 9eb12d6..a78c290 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -1293,24 +1293,6 @@
return RTCError::OK();
}
-void PeerConnection::SetBitrateAllocationStrategy(
- std::unique_ptr<rtc::BitrateAllocationStrategy>
- bitrate_allocation_strategy) {
- rtc::Thread* worker_thread = factory_->worker_thread();
- if (!worker_thread->IsCurrent()) {
- rtc::BitrateAllocationStrategy* strategy_raw =
- bitrate_allocation_strategy.release();
- auto functor = [this, strategy_raw]() {
- call_->SetBitrateAllocationStrategy(
- rtc::WrapUnique<rtc::BitrateAllocationStrategy>(strategy_raw));
- };
- worker_thread->Invoke<void>(RTC_FROM_HERE, functor);
- return;
- }
- RTC_DCHECK(call_.get());
- call_->SetBitrateAllocationStrategy(std::move(bitrate_allocation_strategy));
-}
-
std::unique_ptr<rtc::SSLCertificate>
PeerConnection::GetRemoteAudioSSLCertificate() {
if (!session_) {
diff --git a/pc/peerconnection.h b/pc/peerconnection.h
index 116b249..7d0be4d 100644
--- a/pc/peerconnection.h
+++ b/pc/peerconnection.h
@@ -145,10 +145,6 @@
RTCError SetBitrate(const BitrateParameters& bitrate) override;
- void SetBitrateAllocationStrategy(
- std::unique_ptr<rtc::BitrateAllocationStrategy>
- bitrate_allocation_strategy) override;
-
RTC_DEPRECATED bool StartRtcEventLog(rtc::PlatformFile file,
int64_t max_size_bytes) override;
bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output) override;
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 443d32e..eddef51 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -113,8 +113,6 @@
"bind.h",
"bitbuffer.cc",
"bitbuffer.h",
- "bitrateallocationstrategy.cc",
- "bitrateallocationstrategy.h",
"buffer.h",
"bufferqueue.cc",
"bufferqueue.h",
diff --git a/rtc_base/bitrateallocationstrategy.cc b/rtc_base/bitrateallocationstrategy.cc
deleted file mode 100644
index 66528d75..0000000
--- a/rtc_base/bitrateallocationstrategy.cc
+++ /dev/null
@@ -1,125 +0,0 @@
-/*
- * Copyright 2017 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 "rtc_base/bitrateallocationstrategy.h"
-#include <algorithm>
-#include <utility>
-
-namespace rtc {
-
-std::vector<uint32_t> BitrateAllocationStrategy::SetAllBitratesToMinimum(
- const ArrayView<const TrackConfig*> track_configs) {
- std::vector<uint32_t> track_allocations;
- for (const auto* track_config : track_configs) {
- track_allocations.push_back(track_config->min_bitrate_bps);
- }
- return track_allocations;
-}
-
-std::vector<uint32_t> BitrateAllocationStrategy::DistributeBitratesEvenly(
- const ArrayView<const TrackConfig*> track_configs,
- uint32_t available_bitrate) {
- std::vector<uint32_t> track_allocations =
- SetAllBitratesToMinimum(track_configs);
- uint32_t sum_min_bitrates = 0;
- uint32_t sum_max_bitrates = 0;
- for (const auto* track_config : track_configs) {
- sum_min_bitrates += track_config->min_bitrate_bps;
- sum_max_bitrates += track_config->max_bitrate_bps;
- }
- if (sum_min_bitrates >= available_bitrate) {
- return track_allocations;
- } else if (available_bitrate >= sum_max_bitrates) {
- auto track_allocations_it = track_allocations.begin();
- for (const auto* track_config : track_configs) {
- *track_allocations_it++ = track_config->max_bitrate_bps;
- }
- return track_allocations;
- } else {
- // If sum_min_bitrates < available_bitrate < sum_max_bitrates allocate
- // bitrates evenly up to max_bitrate_bps starting from the track with the
- // lowest max_bitrate_bps. Remainder of available bitrate split evenly among
- // remaining tracks.
- std::multimap<uint32_t, size_t> max_bitrate_sorted_configs;
- for (const TrackConfig** track_configs_it = track_configs.begin();
- track_configs_it != track_configs.end(); ++track_configs_it) {
- max_bitrate_sorted_configs.insert(
- std::make_pair((*track_configs_it)->max_bitrate_bps,
- track_configs_it - track_configs.begin()));
- }
- uint32_t total_available_increase = available_bitrate - sum_min_bitrates;
- int processed_configs = 0;
- for (const auto& track_config_pair : max_bitrate_sorted_configs) {
- uint32_t available_increase =
- total_available_increase /
- (static_cast<uint32_t>(track_configs.size() - processed_configs));
- uint32_t consumed_increase =
- std::min(track_configs[track_config_pair.second]->max_bitrate_bps -
- track_configs[track_config_pair.second]->min_bitrate_bps,
- available_increase);
- track_allocations[track_config_pair.second] += consumed_increase;
- total_available_increase -= consumed_increase;
- ++processed_configs;
- }
- return track_allocations;
- }
-}
-
-AudioPriorityBitrateAllocationStrategy::AudioPriorityBitrateAllocationStrategy(
- std::string audio_track_id,
- uint32_t sufficient_audio_bitrate)
- : audio_track_id_(audio_track_id),
- sufficient_audio_bitrate_(sufficient_audio_bitrate) {}
-
-std::vector<uint32_t> AudioPriorityBitrateAllocationStrategy::AllocateBitrates(
- uint32_t available_bitrate,
- const ArrayView<const TrackConfig*> track_configs) {
- const TrackConfig* audio_track_config = NULL;
- size_t audio_config_index = 0;
- uint32_t sum_min_bitrates = 0;
-
- for (const auto*& track_config : track_configs) {
- sum_min_bitrates += track_config->min_bitrate_bps;
- if (track_config->track_id == audio_track_id_) {
- audio_track_config = track_config;
- audio_config_index = &track_config - &track_configs[0];
- }
- }
- if (audio_track_config == nullptr) {
- return DistributeBitratesEvenly(track_configs, available_bitrate);
- }
- auto safe_sufficient_audio_bitrate = std::min(
- std::max(audio_track_config->min_bitrate_bps, sufficient_audio_bitrate_),
- audio_track_config->max_bitrate_bps);
- if (available_bitrate <= sum_min_bitrates) {
- return SetAllBitratesToMinimum(track_configs);
- } else {
- if (available_bitrate <= sum_min_bitrates + safe_sufficient_audio_bitrate -
- audio_track_config->min_bitrate_bps) {
- std::vector<uint32_t> track_allocations =
- SetAllBitratesToMinimum(track_configs);
- track_allocations[audio_config_index] +=
- available_bitrate - sum_min_bitrates;
- return track_allocations;
- } else {
- // Setting audio track minimum to safe_sufficient_audio_bitrate will
- // allow using DistributeBitratesEvenly to allocate at least sufficient
- // bitrate for audio and the rest evenly.
- TrackConfig sufficient_track_config(*track_configs[audio_config_index]);
- sufficient_track_config.min_bitrate_bps = safe_sufficient_audio_bitrate;
- track_configs[audio_config_index] = &sufficient_track_config;
- std::vector<uint32_t> track_allocations =
- DistributeBitratesEvenly(track_configs, available_bitrate);
- return track_allocations;
- }
- }
-}
-
-} // namespace rtc
diff --git a/rtc_base/bitrateallocationstrategy.h b/rtc_base/bitrateallocationstrategy.h
deleted file mode 100644
index f711d1f..0000000
--- a/rtc_base/bitrateallocationstrategy.h
+++ /dev/null
@@ -1,101 +0,0 @@
-/*
- * Copyright 2017 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.
- */
-
-#ifndef RTC_BASE_BITRATEALLOCATIONSTRATEGY_H_
-#define RTC_BASE_BITRATEALLOCATIONSTRATEGY_H_
-
-#include <map>
-#include <memory>
-#include <string>
-#include <vector>
-#include "api/array_view.h"
-#include "rtc_base/checks.h"
-
-namespace rtc {
-
-// Pluggable strategy allows configuration of bitrate allocation per media
-// track.
-//
-// The strategy should provide allocation for every track passed with
-// track_configs in AllocateBitrates. The allocations are constrained by
-// max_bitrate_bps, min_bitrate_bps defining the track supported range and
-// enforce_min_bitrate indicating if the track my be paused by allocating 0
-// bitrate.
-class BitrateAllocationStrategy {
- public:
- struct TrackConfig {
- TrackConfig(uint32_t min_bitrate_bps,
- uint32_t max_bitrate_bps,
- bool enforce_min_bitrate,
- std::string track_id)
- : min_bitrate_bps(min_bitrate_bps),
- max_bitrate_bps(max_bitrate_bps),
- enforce_min_bitrate(enforce_min_bitrate),
- track_id(track_id) {}
- TrackConfig(const TrackConfig& track_config) = default;
- virtual ~TrackConfig() = default;
- TrackConfig() {}
-
- // Minimum bitrate supported by track.
- uint32_t min_bitrate_bps;
-
- // Maximum bitrate supported by track.
- uint32_t max_bitrate_bps;
-
- // True means track may not be paused by allocating 0 bitrate.
- bool enforce_min_bitrate;
-
- // MediaStreamTrack ID as defined by application. May be empty.
- std::string track_id;
- };
-
- static std::vector<uint32_t> SetAllBitratesToMinimum(
- const ArrayView<const TrackConfig*> track_configs);
- static std::vector<uint32_t> DistributeBitratesEvenly(
- const ArrayView<const TrackConfig*> track_configs,
- uint32_t available_bitrate);
-
- // Strategy is expected to allocate all available_bitrate up to the sum of
- // max_bitrate_bps of all tracks. If available_bitrate is less than the sum of
- // min_bitrate_bps of all tracks, tracks having enforce_min_bitrate set to
- // false may get 0 allocation and are suppoused to pause, tracks with
- // enforce_min_bitrate set to true are expecting to get min_bitrate_bps.
- //
- // If the strategy will allocate more than available_bitrate it may cause
- // overuse of the currently available network capacity and may cause increase
- // in RTT and packet loss. Allocating less than available bitrate may cause
- // available_bitrate decrease.
- virtual std::vector<uint32_t> AllocateBitrates(
- uint32_t available_bitrate,
- const ArrayView<const TrackConfig*> track_configs) = 0;
-
- virtual ~BitrateAllocationStrategy() = default;
-};
-
-// Simple allocation strategy giving priority to audio until
-// sufficient_audio_bitrate is reached. Bitrate is distributed evenly between
-// the tracks after sufficient_audio_bitrate is reached. This implementation
-// does not pause tracks even if enforce_min_bitrate is false.
-class AudioPriorityBitrateAllocationStrategy
- : public BitrateAllocationStrategy {
- public:
- AudioPriorityBitrateAllocationStrategy(std::string audio_track_id,
- uint32_t sufficient_audio_bitrate);
- std::vector<uint32_t> AllocateBitrates(
- uint32_t available_bitrate,
- const ArrayView<const TrackConfig*> track_configs) override;
-
- private:
- std::string audio_track_id_;
- uint32_t sufficient_audio_bitrate_;
-};
-} // namespace rtc
-
-#endif // RTC_BASE_BITRATEALLOCATIONSTRATEGY_H_
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index d415158..1401132 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -386,7 +386,6 @@
"objc/Framework/Classes/PeerConnection/RTCAudioSource.mm",
"objc/Framework/Classes/PeerConnection/RTCAudioTrack+Private.h",
"objc/Framework/Classes/PeerConnection/RTCAudioTrack.mm",
- "objc/Framework/Classes/PeerConnection/RTCBitrateAllocationStrategy.mm",
"objc/Framework/Classes/PeerConnection/RTCConfiguration+Private.h",
"objc/Framework/Classes/PeerConnection/RTCConfiguration.mm",
"objc/Framework/Classes/PeerConnection/RTCDataChannel+Private.h",
@@ -454,7 +453,6 @@
"objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h",
"objc/Framework/Headers/WebRTC/RTCAudioSource.h",
"objc/Framework/Headers/WebRTC/RTCAudioTrack.h",
- "objc/Framework/Headers/WebRTC/RTCBitrateAllocationStrategy.h",
"objc/Framework/Headers/WebRTC/RTCConfiguration.h",
"objc/Framework/Headers/WebRTC/RTCDataChannel.h",
"objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h",
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCBitrateAllocationStrategy.mm b/sdk/objc/Framework/Classes/PeerConnection/RTCBitrateAllocationStrategy.mm
deleted file mode 100644
index bc99767..0000000
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCBitrateAllocationStrategy.mm
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * Copyright 2017 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.
- */
-
-#import "WebRTC/RTCBitrateAllocationStrategy.h"
-
-#include "rtc_base/bitrateallocationstrategy.h"
-#include "rtc_base/checks.h"
-
-@implementation RTCBitrateAllocationStrategy
-
-@synthesize strategy = _strategy;
-
-- (instancetype)initWith:(rtc::BitrateAllocationStrategy*)strategy {
- RTC_DCHECK(strategy);
- if (self = [super init]) {
- _strategy = strategy;
- }
- return self;
-}
-
-@end
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration.mm b/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration.mm
index 17aed85..b9a9e4b 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration.mm
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration.mm
@@ -178,6 +178,7 @@
nativeConfig->ice_regather_interval_range =
rtc::Optional<rtc::IntervalRange>(*nativeIntervalRange);
}
+
return nativeConfig.release();
}
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm b/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm
index 4fcc634..e443e85 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm
@@ -21,7 +21,6 @@
#import "RTCRtpReceiver+Private.h"
#import "RTCRtpSender+Private.h"
#import "RTCSessionDescription+Private.h"
-#import "WebRTC/RTCBitrateAllocationStrategy.h"
#import "WebRTC/RTCLogging.h"
#include <memory>
@@ -386,15 +385,6 @@
return _peerConnection->SetBitrate(params).ok();
}
-- (void)setBitrateAllocationStrategy:
- (RTCBitrateAllocationStrategy *_Nullable)bitrateAllocationStrategy {
- if (bitrateAllocationStrategy)
- _peerConnection->SetBitrateAllocationStrategy(
- std::unique_ptr<rtc::BitrateAllocationStrategy>(bitrateAllocationStrategy.strategy));
- else
- _peerConnection->SetBitrateAllocationStrategy(nullptr);
-}
-
- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath
maxSizeInBytes:(int64_t)maxSizeInBytes {
RTC_DCHECK(filePath.length);
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCBitrateAllocationStrategy.h b/sdk/objc/Framework/Headers/WebRTC/RTCBitrateAllocationStrategy.h
deleted file mode 100644
index c510008..0000000
--- a/sdk/objc/Framework/Headers/WebRTC/RTCBitrateAllocationStrategy.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright 2017 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.
- */
-
-#import <WebRTC/RTCMacros.h>
-#import <WebRTC/RTCMediaStreamTrack.h>
-
-NS_ASSUME_NONNULL_BEGIN
-
-namespace rtc {
-
-class BitrateAllocationStrategy;
-}
-
-RTC_EXPORT
-@interface RTCBitrateAllocationStrategy : NSObject
-
-- (instancetype)init NS_UNAVAILABLE;
-- (instancetype)initWith:(rtc::BitrateAllocationStrategy*)strategy;
-
-/** Native bitrate allocation strategy. */
-@property(nonatomic, readonly) rtc::BitrateAllocationStrategy* strategy;
-
-@end
-
-NS_ASSUME_NONNULL_END
diff --git a/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h b/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
index c74af13..7b0c449 100644
--- a/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
+++ b/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
@@ -24,7 +24,6 @@
@class RTCRtpSender;
@class RTCSessionDescription;
@class RTCLegacyStatsReport;
-@class RTCBitrateAllocationStrategy;
NS_ASSUME_NONNULL_BEGIN
@@ -194,13 +193,6 @@
currentBitrateBps:(nullable NSNumber *)currentBitrateBps
maxBitrateBps:(nullable NSNumber *)maxBitrateBps;
-/** Sets current strategy. If not set default WebRTC allocator will be used.
- * May be changed during an active session. The strategy
- * ownership is passed with std::unique_ptr
- */
-- (void)setBitrateAllocationStrategy:
- (RTCBitrateAllocationStrategy *_Nullable)bitrateAllocationStrategy;
-
/** Start or stop recording an Rtc EventLog. */
- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath
maxSizeInBytes:(int64_t)maxSizeInBytes;
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 01aa965..b9d785d 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -215,10 +215,6 @@
}
}
- void SetAccountForAudioPackets(bool account_for_audio) override {
- RTC_NOTREACHED();
- }
-
private:
rtc::ThreadChecker thread_checker_;
rtc::CriticalSection crit_;
