diff options
Diffstat (limited to '')
7 files changed, 397 insertions, 54 deletions
diff --git a/third_party/libwebrtc/modules/pacing/BUILD.gn b/third_party/libwebrtc/modules/pacing/BUILD.gn index ea80c8c819..87498817b6 100644 --- a/third_party/libwebrtc/modules/pacing/BUILD.gn +++ b/third_party/libwebrtc/modules/pacing/BUILD.gn @@ -63,6 +63,7 @@ rtc_library("pacing") { ] absl_deps = [ "//third_party/abseil-cpp/absl/cleanup", + "//third_party/abseil-cpp/absl/container:inlined_vector", "//third_party/abseil-cpp/absl/memory", "//third_party/abseil-cpp/absl/strings", "//third_party/abseil-cpp/absl/types:optional", diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller.cc b/third_party/libwebrtc/modules/pacing/pacing_controller.cc index 5b81207d56..41f97a37fb 100644 --- a/third_party/libwebrtc/modules/pacing/pacing_controller.cc +++ b/third_party/libwebrtc/modules/pacing/pacing_controller.cc @@ -19,11 +19,11 @@ #include "absl/strings/match.h" #include "api/units/data_size.h" #include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "modules/pacing/bitrate_prober.h" -#include "modules/pacing/interval_budget.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.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 { @@ -44,8 +44,6 @@ bool IsEnabled(const FieldTrialsView& field_trials, absl::string_view key) { } // namespace -const TimeDelta PacingController::kMaxExpectedQueueLength = - TimeDelta::Millis(2000); const TimeDelta PacingController::kPausedProcessInterval = kCongestedPacketInterval; const TimeDelta PacingController::kMinSleepTime = TimeDelta::Millis(1); @@ -57,11 +55,13 @@ const TimeDelta PacingController::kMaxEarlyProbeProcessing = PacingController::PacingController(Clock* clock, PacketSender* packet_sender, - const FieldTrialsView& field_trials) + const FieldTrialsView& field_trials, + Configuration configuration) : clock_(clock), packet_sender_(packet_sender), field_trials_(field_trials), drain_large_queues_( + configuration.drain_large_queues && !IsDisabled(field_trials_, "WebRTC-Pacer-DrainQueue")), send_padding_if_silent_( IsEnabled(field_trials_, "WebRTC-Pacer-PadInSilence")), @@ -71,9 +71,10 @@ PacingController::PacingController(Clock* clock, fast_retransmissions_( IsEnabled(field_trials_, "WebRTC-Pacer-FastRetransmissions")), keyframe_flushing_( + configuration.keyframe_flushing || IsEnabled(field_trials_, "WebRTC-Pacer-KeyframeFlushing")), transport_overhead_per_packet_(DataSize::Zero()), - send_burst_interval_(kDefaultBurstInterval), + send_burst_interval_(configuration.send_burst_interval), last_timestamp_(clock_->CurrentTime()), paused_(false), media_debt_(DataSize::Zero()), @@ -86,9 +87,11 @@ PacingController::PacingController(Clock* clock, last_process_time_(clock->CurrentTime()), last_send_time_(last_process_time_), seen_first_packet_(false), - packet_queue_(/*creation_time=*/last_process_time_), + packet_queue_(/*creation_time=*/last_process_time_, + configuration.prioritize_audio_retransmission, + configuration.packet_queue_ttl), congested_(false), - queue_time_limit_(kMaxExpectedQueueLength), + queue_time_limit_(configuration.queue_time_limit), account_for_audio_(false), include_overhead_(false), circuit_breaker_threshold_(1 << 16) { @@ -710,8 +713,7 @@ Timestamp PacingController::NextUnpacedSendTime() const { } if (fast_retransmissions_) { Timestamp leading_retransmission_send_time = - packet_queue_.LeadingPacketEnqueueTime( - RtpPacketMediaType::kRetransmission); + packet_queue_.LeadingPacketEnqueueTimeForRetransmission(); if (leading_retransmission_send_time.IsFinite()) { return leading_retransmission_send_time; } diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller.h b/third_party/libwebrtc/modules/pacing/pacing_controller.h index 04e0a820f9..fe6ee737a9 100644 --- a/third_party/libwebrtc/modules/pacing/pacing_controller.h +++ b/third_party/libwebrtc/modules/pacing/pacing_controller.h @@ -67,11 +67,6 @@ class PacingController { } }; - // Expected max pacer delay. If ExpectedQueueTime() is higher than - // this value, the packet producers should wait (eg drop frames rather than - // encoding them). Bitrate sent may temporarily exceed target set by - // UpdateBitrate() so that this limit will be upheld. - static const TimeDelta kMaxExpectedQueueLength; // If no media or paused, wake up at least every `kPausedProcessIntervalMs` in // order to send a keep-alive packet so we don't get stuck in a bad state due // to lack of feedback. @@ -93,14 +88,45 @@ class PacingController { // the send burst interval. // Ex: max send burst interval = 63Kb / 10Mbit/s = 50ms. static constexpr DataSize kMaxBurstSize = DataSize::Bytes(63 * 1000); - // The pacer is allowed to send enqued packets in bursts and can build up a - // packet "debt" that correspond to approximately the send rate during - // the burst interval. + + // Configuration default values. static constexpr TimeDelta kDefaultBurstInterval = TimeDelta::Millis(40); + static constexpr TimeDelta kMaxExpectedQueueLength = TimeDelta::Millis(2000); + + struct Configuration { + // If the pacer queue grows longer than the configured max queue limit, + // pacer sends at the minimum rate needed to keep the max queue limit and + // ignore the current bandwidth estimate. + bool drain_large_queues = true; + // Expected max pacer delay. If ExpectedQueueTime() is higher than + // this value, the packet producers should wait (eg drop frames rather than + // encoding them). Bitrate sent may temporarily exceed target set by + // SetPacingRates() so that this limit will be upheld if + // `drain_large_queues` is set. + TimeDelta queue_time_limit = kMaxExpectedQueueLength; + // If the first packet of a keyframe is enqueued on a RTP stream, pacer + // skips forward to that packet and drops other enqueued packets on that + // stream, unless a keyframe is already being paced. + bool keyframe_flushing = false; + // Audio retransmission is prioritized before video retransmission packets. + bool prioritize_audio_retransmission = false; + // Configure separate timeouts per priority. After a timeout, a packet of + // that sort will not be paced and instead dropped. + // Note: to set TTL on audio retransmission, + // `prioritize_audio_retransmission` must be true. + PacketQueueTTL packet_queue_ttl; + // The pacer is allowed to send enqueued packets in bursts and can build up + // a packet "debt" that correspond to approximately the send rate during the + // burst interval. + TimeDelta send_burst_interval = kDefaultBurstInterval; + }; + + static Configuration DefaultConfiguration() { return Configuration{}; } PacingController(Clock* clock, PacketSender* packet_sender, - const FieldTrialsView& field_trials); + const FieldTrialsView& field_trials, + Configuration configuration = DefaultConfiguration()); ~PacingController(); diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc b/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc index 9e6ede6dc0..2c3a71b369 100644 --- a/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc +++ b/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc @@ -2348,5 +2348,43 @@ TEST_F(PacingControllerTest, FlushesPacketsOnKeyFrames) { pacer->ProcessPackets(); } +TEST_F(PacingControllerTest, CanControlQueueSizeUsingTtl) { + const uint32_t kSsrc = 12345; + const uint32_t kAudioSsrc = 2345; + uint16_t sequence_number = 1234; + + PacingController::Configuration config; + config.drain_large_queues = false; + config.packet_queue_ttl.video = TimeDelta::Millis(500); + auto pacer = + std::make_unique<PacingController>(&clock_, &callback_, trials_, config); + pacer->SetPacingRates(DataRate::BitsPerSec(100'000), DataRate::Zero()); + + Timestamp send_time = Timestamp::Zero(); + for (int i = 0; i < 100; ++i) { + // Enqueue a new audio and video frame every 33ms. + if (clock_.CurrentTime() - send_time > TimeDelta::Millis(33)) { + for (int j = 0; j < 3; ++j) { + auto packet = BuildPacket(RtpPacketMediaType::kVideo, kSsrc, + /*sequence_number=*/++sequence_number, + /*capture_time_ms=*/2, + /*size_bytes=*/1000); + pacer->EnqueuePacket(std::move(packet)); + } + auto packet = BuildPacket(RtpPacketMediaType::kAudio, kAudioSsrc, + /*sequence_number=*/++sequence_number, + /*capture_time_ms=*/2, + /*size_bytes=*/100); + pacer->EnqueuePacket(std::move(packet)); + send_time = clock_.CurrentTime(); + } + + EXPECT_LE(clock_.CurrentTime() - pacer->OldestPacketEnqueueTime(), + TimeDelta::Millis(500)); + clock_.AdvanceTime(pacer->NextSendTime() - clock_.CurrentTime()); + pacer->ProcessPackets(); + } +} + } // namespace } // namespace webrtc diff --git a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.cc b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.cc index ea211ea683..2d0d829648 100644 --- a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.cc +++ b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.cc @@ -10,41 +10,70 @@ #include "modules/pacing/prioritized_packet_queue.h" +#include <algorithm> +#include <array> #include <utility> +#include "absl/container/inlined_vector.h" +#include "absl/types/optional.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "modules/rtp_rtcp/source/rtp_packet_to_send.h" #include "rtc_base/checks.h" +#include "rtc_base/logging.h" namespace webrtc { namespace { constexpr int kAudioPrioLevel = 0; -int GetPriorityForType(RtpPacketMediaType type) { +int GetPriorityForType( + RtpPacketMediaType type, + absl::optional<RtpPacketToSend::OriginalType> original_type) { // Lower number takes priority over higher. switch (type) { case RtpPacketMediaType::kAudio: // Audio is always prioritized over other packet types. return kAudioPrioLevel; case RtpPacketMediaType::kRetransmission: - // Send retransmissions before new media. + // Send retransmissions before new media. If original_type is set, audio + // retransmission is prioritized more than video retransmission. + if (original_type == RtpPacketToSend::OriginalType::kVideo) { + return kAudioPrioLevel + 2; + } return kAudioPrioLevel + 1; 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 kAudioPrioLevel + 2; + return kAudioPrioLevel + 3; case RtpPacketMediaType::kPadding: // Packets that are in themselves likely useless, only sent to keep the // BWE high. - return kAudioPrioLevel + 3; + return kAudioPrioLevel + 4; } RTC_CHECK_NOTREACHED(); } } // namespace +absl::InlinedVector<TimeDelta, PrioritizedPacketQueue::kNumPriorityLevels> +PrioritizedPacketQueue::ToTtlPerPrio(PacketQueueTTL packet_queue_ttl) { + absl::InlinedVector<TimeDelta, PrioritizedPacketQueue::kNumPriorityLevels> + ttl_per_prio(kNumPriorityLevels, TimeDelta::PlusInfinity()); + ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kRetransmission, + RtpPacketToSend::OriginalType::kAudio)] = + packet_queue_ttl.audio_retransmission; + ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kRetransmission, + RtpPacketToSend::OriginalType::kVideo)] = + packet_queue_ttl.video_retransmission; + ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kVideo, absl::nullopt)] = + packet_queue_ttl.video; + return ttl_per_prio; +} + DataSize PrioritizedPacketQueue::QueuedPacket::PacketSize() const { return DataSize::Bytes(packet->payload_size() + packet->padding_size()); } @@ -109,8 +138,13 @@ PrioritizedPacketQueue::StreamQueue::DequeueAll() { return packets_by_prio; } -PrioritizedPacketQueue::PrioritizedPacketQueue(Timestamp creation_time) - : queue_time_sum_(TimeDelta::Zero()), +PrioritizedPacketQueue::PrioritizedPacketQueue( + Timestamp creation_time, + bool prioritize_audio_retransmission, + PacketQueueTTL packet_queue_ttl) + : prioritize_audio_retransmission_(prioritize_audio_retransmission), + time_to_live_per_prio_(ToTtlPerPrio(packet_queue_ttl)), + queue_time_sum_(TimeDelta::Zero()), pause_time_sum_(TimeDelta::Zero()), size_packets_(0), size_packets_per_media_type_({}), @@ -133,7 +167,11 @@ void PrioritizedPacketQueue::Push(Timestamp enqueue_time, enqueue_times_.insert(enqueue_times_.end(), enqueue_time); RTC_DCHECK(packet->packet_type().has_value()); RtpPacketMediaType packet_type = packet->packet_type().value(); - int prio_level = GetPriorityForType(packet_type); + int prio_level = + GetPriorityForType(packet_type, prioritize_audio_retransmission_ + ? packet->original_packet_type() + : absl::nullopt); + PurgeOldPacketsAtPriorityLevel(prio_level, enqueue_time); RTC_DCHECK_GE(prio_level, 0); RTC_DCHECK_LT(prio_level, kNumPriorityLevels); QueuedPacket queued_packed = {.packet = std::move(packet), @@ -214,7 +252,8 @@ PrioritizedPacketQueue::SizeInPacketsPerRtpPacketMediaType() const { Timestamp PrioritizedPacketQueue::LeadingPacketEnqueueTime( RtpPacketMediaType type) const { - const int priority_level = GetPriorityForType(type); + RTC_DCHECK(type != RtpPacketMediaType::kRetransmission); + const int priority_level = GetPriorityForType(type, absl::nullopt); if (streams_by_prio_[priority_level].empty()) { return Timestamp::MinusInfinity(); } @@ -222,6 +261,39 @@ Timestamp PrioritizedPacketQueue::LeadingPacketEnqueueTime( priority_level); } +Timestamp PrioritizedPacketQueue::LeadingPacketEnqueueTimeForRetransmission() + const { + if (!prioritize_audio_retransmission_) { + const int priority_level = + GetPriorityForType(RtpPacketMediaType::kRetransmission, absl::nullopt); + if (streams_by_prio_[priority_level].empty()) { + return Timestamp::PlusInfinity(); + } + return streams_by_prio_[priority_level].front()->LeadingPacketEnqueueTime( + priority_level); + } + const int audio_priority_level = + GetPriorityForType(RtpPacketMediaType::kRetransmission, + RtpPacketToSend::OriginalType::kAudio); + const int video_priority_level = + GetPriorityForType(RtpPacketMediaType::kRetransmission, + RtpPacketToSend::OriginalType::kVideo); + + Timestamp next_audio = + streams_by_prio_[audio_priority_level].empty() + ? Timestamp::PlusInfinity() + : streams_by_prio_[audio_priority_level] + .front() + ->LeadingPacketEnqueueTime(audio_priority_level); + Timestamp next_video = + streams_by_prio_[video_priority_level].empty() + ? Timestamp::PlusInfinity() + : streams_by_prio_[video_priority_level] + .front() + ->LeadingPacketEnqueueTime(video_priority_level); + return std::min(next_audio, next_video); +} + Timestamp PrioritizedPacketQueue::OldestEnqueueTime() const { return enqueue_times_.empty() ? Timestamp::MinusInfinity() : enqueue_times_.front(); @@ -283,9 +355,6 @@ void PrioritizedPacketQueue::RemovePacketsForSsrc(uint32_t ssrc) { // Update the global top prio level if neccessary. RTC_DCHECK(streams_by_prio_[i].front() == &queue); streams_by_prio_[i].pop_front(); - if (i == top_active_prio_level_) { - MaybeUpdateTopPrioLevel(); - } } else { // More than stream had packets at this prio level, filter this one out. std::deque<StreamQueue*> filtered_queue; @@ -298,6 +367,7 @@ void PrioritizedPacketQueue::RemovePacketsForSsrc(uint32_t ssrc) { } } } + MaybeUpdateTopPrioLevel(); } bool PrioritizedPacketQueue::HasKeyframePackets(uint32_t ssrc) const { @@ -340,18 +410,53 @@ void PrioritizedPacketQueue::DequeuePacketInternal(QueuedPacket& packet) { } void PrioritizedPacketQueue::MaybeUpdateTopPrioLevel() { - if (streams_by_prio_[top_active_prio_level_].empty()) { - // No stream queues have packets at this prio level, find top priority - // that is not empty. - if (size_packets_ == 0) { - top_active_prio_level_ = -1; + if (top_active_prio_level_ != -1 && + !streams_by_prio_[top_active_prio_level_].empty()) { + return; + } + // No stream queues have packets at top_active_prio_level_, find top priority + // that is not empty. + for (int i = 0; i < kNumPriorityLevels; ++i) { + PurgeOldPacketsAtPriorityLevel(i, last_update_time_); + if (!streams_by_prio_[i].empty()) { + top_active_prio_level_ = i; + break; + } + } + if (size_packets_ == 0) { + // There are no packets left to send. Last packet may have been purged. Prio + // will change when a new packet is pushed. + top_active_prio_level_ = -1; + } +} + +void PrioritizedPacketQueue::PurgeOldPacketsAtPriorityLevel(int prio_level, + Timestamp now) { + RTC_DCHECK(prio_level >= 0 && prio_level < kNumPriorityLevels); + TimeDelta time_to_live = time_to_live_per_prio_[prio_level]; + if (time_to_live.IsInfinite()) { + return; + } + + std::deque<StreamQueue*>& queues = streams_by_prio_[prio_level]; + auto iter = queues.begin(); + while (iter != queues.end()) { + StreamQueue* queue_ptr = *iter; + while (queue_ptr->HasPacketsAtPrio(prio_level) && + (now - queue_ptr->LeadingPacketEnqueueTime(prio_level)) > + time_to_live) { + QueuedPacket packet = queue_ptr->DequeuePacket(prio_level); + RTC_LOG(LS_INFO) << "Dropping old packet on SSRC: " + << packet.packet->Ssrc() + << " seq:" << packet.packet->SequenceNumber() + << " time in queue:" << (now - packet.enqueue_time).ms() + << " ms"; + DequeuePacketInternal(packet); + } + if (!queue_ptr->HasPacketsAtPrio(prio_level)) { + iter = queues.erase(iter); } else { - for (int i = 0; i < kNumPriorityLevels; ++i) { - if (!streams_by_prio_[i].empty()) { - top_active_prio_level_ = i; - break; - } - } + ++iter; } } } diff --git a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.h b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.h index 935c530027..179ef104fe 100644 --- a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.h +++ b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.h @@ -18,8 +18,8 @@ #include <list> #include <memory> #include <unordered_map> -#include <vector> +#include "absl/container/inlined_vector.h" #include "api/units/data_size.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" @@ -27,9 +27,19 @@ namespace webrtc { +// Describes how long time a packet may stay in the queue before being dropped. +struct PacketQueueTTL { + TimeDelta audio_retransmission = TimeDelta::PlusInfinity(); + TimeDelta video_retransmission = TimeDelta::PlusInfinity(); + TimeDelta video = TimeDelta::PlusInfinity(); +}; + class PrioritizedPacketQueue { public: - explicit PrioritizedPacketQueue(Timestamp creation_time); + explicit PrioritizedPacketQueue( + Timestamp creation_time, + bool prioritize_audio_retransmission = false, + PacketQueueTTL packet_queue_ttl = PacketQueueTTL()); PrioritizedPacketQueue(const PrioritizedPacketQueue&) = delete; PrioritizedPacketQueue& operator=(const PrioritizedPacketQueue&) = delete; @@ -63,6 +73,7 @@ class PrioritizedPacketQueue { // method, for the given packet type. If queue has no packets, of that type, // returns Timestamp::MinusInfinity(). Timestamp LeadingPacketEnqueueTime(RtpPacketMediaType type) const; + Timestamp LeadingPacketEnqueueTimeForRetransmission() const; // Enqueue time of the oldest packet in the queue, // Timestamp::MinusInfinity() if queue is empty. @@ -90,7 +101,7 @@ class PrioritizedPacketQueue { bool HasKeyframePackets(uint32_t ssrc) const; private: - static constexpr int kNumPriorityLevels = 4; + static constexpr int kNumPriorityLevels = 5; class QueuedPacket { public: @@ -139,6 +150,15 @@ class PrioritizedPacketQueue { // if so move it to the lowest non-empty index. void MaybeUpdateTopPrioLevel(); + void PurgeOldPacketsAtPriorityLevel(int prio_level, Timestamp now); + + static absl::InlinedVector<TimeDelta, kNumPriorityLevels> ToTtlPerPrio( + PacketQueueTTL); + + const bool prioritize_audio_retransmission_; + const absl::InlinedVector<TimeDelta, kNumPriorityLevels> + time_to_live_per_prio_; + // Cumulative sum, over all packets, of time spent in the queue. TimeDelta queue_time_sum_; // Cumulative sum of time the queue has spent in a paused state. diff --git a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue_unittest.cc b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue_unittest.cc index 9ed19642c7..76c31036b3 100644 --- a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue_unittest.cc +++ b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue_unittest.cc @@ -10,6 +10,7 @@ #include "modules/pacing/prioritized_packet_queue.h" +#include <memory> #include <utility> #include "api/units/time_delta.h" @@ -26,18 +27,39 @@ constexpr uint32_t kDefaultSsrc = 123; constexpr int kDefaultPayloadSize = 789; std::unique_ptr<RtpPacketToSend> CreatePacket(RtpPacketMediaType type, - uint16_t sequence_number, + uint16_t seq, uint32_t ssrc = kDefaultSsrc, bool is_key_frame = false) { auto packet = std::make_unique<RtpPacketToSend>(/*extensions=*/nullptr); packet->set_packet_type(type); packet->SetSsrc(ssrc); - packet->SetSequenceNumber(sequence_number); + packet->SetSequenceNumber(seq); packet->SetPayloadSize(kDefaultPayloadSize); packet->set_is_key_frame(is_key_frame); return packet; } +std::unique_ptr<RtpPacketToSend> CreateRetransmissionPacket( + RtpPacketMediaType original_type, + uint16_t seq, + uint32_t ssrc = kDefaultSsrc) { + auto packet = std::make_unique<RtpPacketToSend>(/*extensions=*/nullptr); + packet->set_packet_type(original_type); + packet->set_packet_type(RtpPacketMediaType::kRetransmission); + RTC_DCHECK(packet->packet_type() == RtpPacketMediaType::kRetransmission); + if (original_type == RtpPacketMediaType::kVideo) { + RTC_DCHECK(packet->original_packet_type() == + RtpPacketToSend::OriginalType::kVideo); + } else { + RTC_DCHECK(packet->original_packet_type() == + RtpPacketToSend::OriginalType::kAudio); + } + packet->SetSsrc(ssrc); + packet->SetSequenceNumber(seq); + packet->SetPayloadSize(kDefaultPayloadSize); + return packet; +} + } // namespace TEST(PrioritizedPacketQueue, ReturnsPacketsInPrioritizedOrder) { @@ -49,18 +71,42 @@ TEST(PrioritizedPacketQueue, ReturnsPacketsInPrioritizedOrder) { queue.Push(now, CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2)); queue.Push(now, CreatePacket(RtpPacketMediaType::kForwardErrorCorrection, /*seq=*/3)); - queue.Push(now, CreatePacket(RtpPacketMediaType::kRetransmission, /*seq=*/4)); - queue.Push(now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/5)); + queue.Push(now, + CreateRetransmissionPacket(RtpPacketMediaType::kVideo, /*seq=*/4)); + queue.Push(now, + CreateRetransmissionPacket(RtpPacketMediaType::kAudio, /*seq=*/5)); + queue.Push(now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/6)); // Packets should be returned in high to low order. - EXPECT_EQ(queue.Pop()->SequenceNumber(), 5); + EXPECT_EQ(queue.Pop()->SequenceNumber(), 6); + // Audio and video retransmission has same prio, but video was enqueued first. EXPECT_EQ(queue.Pop()->SequenceNumber(), 4); + EXPECT_EQ(queue.Pop()->SequenceNumber(), 5); // Video and FEC prioritized equally - but video was enqueued first. EXPECT_EQ(queue.Pop()->SequenceNumber(), 2); EXPECT_EQ(queue.Pop()->SequenceNumber(), 3); EXPECT_EQ(queue.Pop()->SequenceNumber(), 1); } +TEST(PrioritizedPacketQueue, + PrioritizeAudioRetransmissionBeforeVideoRetransmissionIfConfigured) { + Timestamp now = Timestamp::Zero(); + PrioritizedPacketQueue queue(now, /*prioritize_audio_retransmission=*/true); + + // Add packets in low to high packet order. + queue.Push(now, CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/3)); + queue.Push(now, + CreateRetransmissionPacket(RtpPacketMediaType::kVideo, /*seq=*/4)); + queue.Push(now, + CreateRetransmissionPacket(RtpPacketMediaType::kAudio, /*seq=*/5)); + queue.Push(now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/6)); + + // Packets should be returned in high to low order. + EXPECT_EQ(queue.Pop()->SequenceNumber(), 6); + EXPECT_EQ(queue.Pop()->SequenceNumber(), 5); + EXPECT_EQ(queue.Pop()->SequenceNumber(), 4); +} + TEST(PrioritizedPacketQueue, ReturnsEqualPrioPacketsInRoundRobinOrder) { Timestamp now = Timestamp::Zero(); PrioritizedPacketQueue queue(now); @@ -251,6 +297,26 @@ TEST(PrioritizedPacketQueue, ReportsLeadingPacketEnqueueTime) { Timestamp::MinusInfinity()); } +TEST(PrioritizedPacketQueue, ReportsLeadingPacketEnqueueTimeForRetransmission) { + PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero(), + /*prioritize_audio_retransmission=*/true); + EXPECT_EQ(queue.LeadingPacketEnqueueTimeForRetransmission(), + Timestamp::PlusInfinity()); + + queue.Push(Timestamp::Millis(10), + CreateRetransmissionPacket(RtpPacketMediaType::kVideo, /*seq=*/1)); + queue.Push(Timestamp::Millis(11), + CreateRetransmissionPacket(RtpPacketMediaType::kAudio, /*seq=*/2)); + EXPECT_EQ(queue.LeadingPacketEnqueueTimeForRetransmission(), + Timestamp::Millis(10)); + queue.Pop(); // Pop audio retransmission since it has higher prio. + EXPECT_EQ(queue.LeadingPacketEnqueueTimeForRetransmission(), + Timestamp::Millis(10)); + queue.Pop(); // Pop video retransmission. + EXPECT_EQ(queue.LeadingPacketEnqueueTimeForRetransmission(), + Timestamp::PlusInfinity()); +} + TEST(PrioritizedPacketQueue, PushAndPopUpdatesSizeInPacketsPerRtpPacketMediaType) { Timestamp now = Timestamp::Zero(); @@ -272,7 +338,7 @@ TEST(PrioritizedPacketQueue, RtpPacketMediaType::kVideo)], 1); - queue.Push(now, CreatePacket(RtpPacketMediaType::kRetransmission, 3)); + queue.Push(now, CreateRetransmissionPacket(RtpPacketMediaType::kVideo, 3)); EXPECT_EQ(queue.SizeInPacketsPerRtpPacketMediaType()[static_cast<size_t>( RtpPacketMediaType::kRetransmission)], 1); @@ -326,6 +392,8 @@ TEST(PrioritizedPacketQueue, ClearsPackets) { // Remove all of them. queue.RemovePacketsForSsrc(kSsrc); EXPECT_TRUE(queue.Empty()); + queue.RemovePacketsForSsrc(kSsrc); + EXPECT_TRUE(queue.Empty()); } TEST(PrioritizedPacketQueue, ClearPacketsAffectsOnlySpecifiedSsrc) { @@ -338,16 +406,16 @@ TEST(PrioritizedPacketQueue, ClearPacketsAffectsOnlySpecifiedSsrc) { // ensuring they are first in line. queue.Push( now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/1, kRemovingSsrc)); - queue.Push(now, CreatePacket(RtpPacketMediaType::kRetransmission, /*seq=*/2, - kRemovingSsrc)); + queue.Push(now, CreateRetransmissionPacket(RtpPacketMediaType::kVideo, + /*seq=*/2, kRemovingSsrc)); // Add a video packet and a retransmission for the SSRC that will remain. // The retransmission packets now both have pointers to their respective qeues // from the same prio level. queue.Push(now, CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/3, kStayingSsrc)); - queue.Push(now, CreatePacket(RtpPacketMediaType::kRetransmission, /*seq=*/4, - kStayingSsrc)); + queue.Push(now, CreateRetransmissionPacket(RtpPacketMediaType::kVideo, + /*seq=*/4, kStayingSsrc)); EXPECT_EQ(queue.SizeInPackets(), 4); @@ -413,4 +481,87 @@ TEST(PrioritizedPacketQueue, ReportsKeyframePackets) { EXPECT_FALSE(queue.HasKeyframePackets(kVideoSsrc2)); } +TEST(PrioritizedPacketQueue, PacketsDroppedIfNotPulledWithinTttl) { + Timestamp now = Timestamp::Zero(); + PacketQueueTTL ttls; + ttls.audio_retransmission = TimeDelta::Millis(200); + PrioritizedPacketQueue queue(now, /*prioritize_audio_retransmission=*/true, + ttls); + + queue.Push(now, + CreateRetransmissionPacket(RtpPacketMediaType::kAudio, /*seq=*/1)); + now += ttls.audio_retransmission + TimeDelta::Millis(1); + EXPECT_EQ(queue.SizeInPackets(), 1); + queue.Push(now, + CreateRetransmissionPacket(RtpPacketMediaType::kAudio, /*seq=*/2)); + EXPECT_EQ(queue.SizeInPackets(), 1); + EXPECT_EQ(queue.Pop()->SequenceNumber(), 2); +} + +TEST(PrioritizedPacketQueue, DontSendPacketsAfterTttl) { + Timestamp now = Timestamp::Zero(); + PacketQueueTTL ttls; + ttls.audio_retransmission = TimeDelta::Millis(200); + PrioritizedPacketQueue queue(now, /*prioritize_audio_retransmission=*/true, + ttls); + + queue.Push(now, + CreateRetransmissionPacket(RtpPacketMediaType::kAudio, /*seq=*/1)); + now += ttls.audio_retransmission + TimeDelta::Millis(1); + EXPECT_EQ(queue.SizeInPackets(), 1); + queue.Push(now, CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2)); + queue.Push(now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/3)); + // Expect the old packet to have been removed since it was not popped in time. + EXPECT_EQ(queue.SizeInPackets(), 3); + EXPECT_EQ(queue.Pop()->SequenceNumber(), 3); + EXPECT_EQ(queue.SizeInPackets(), 1); + EXPECT_EQ(queue.Pop()->SequenceNumber(), 2); + EXPECT_EQ(queue.SizeInPackets(), 0); +} + +TEST(PrioritizedPacketQueue, SendsNewVideoPacketAfterPurgingLastOldRtxPacket) { + Timestamp now = Timestamp::Zero(); + PacketQueueTTL ttls; + ttls.video_retransmission = TimeDelta::Millis(400); + PrioritizedPacketQueue queue(now, /*prioritize_audio_retransmission=*/true, + ttls); + + queue.Push(now, + CreateRetransmissionPacket(RtpPacketMediaType::kVideo, /*seq=*/1)); + now += ttls.video_retransmission + TimeDelta::Millis(1); + queue.Push(now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/2)); + EXPECT_EQ(queue.SizeInPackets(), 2); + // Expect the audio packet to be send and the video retransmission packet to + // be dropped since it is old. + EXPECT_EQ(queue.Pop()->SequenceNumber(), 2); + EXPECT_EQ(queue.SizeInPackets(), 0); + + queue.Push(now, CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/3)); + EXPECT_EQ(queue.SizeInPackets(), 1); + EXPECT_EQ(queue.Pop()->SequenceNumber(), 3); + EXPECT_EQ(queue.SizeInPackets(), 0); +} + +TEST(PrioritizedPacketQueue, + SendsPacketsAfterTttlIfPrioHigherThanPushedPackets) { + Timestamp now = Timestamp::Zero(); + PacketQueueTTL ttls; + ttls.audio_retransmission = TimeDelta::Millis(200); + PrioritizedPacketQueue queue(now, /*prioritize_audio_retransmission=*/true, + ttls); + + queue.Push(now, + CreateRetransmissionPacket(RtpPacketMediaType::kAudio, /*seq=*/1)); + now += ttls.audio_retransmission + TimeDelta::Millis(1); + EXPECT_EQ(queue.SizeInPackets(), 1); + queue.Push(now, CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2)); + + // This test just show that TTL is not enforced strictly. If a new audio + // packet had been queued before a packet was popped, the audio retransmission + // packet would have been dropped. + EXPECT_EQ(queue.SizeInPackets(), 2); + EXPECT_EQ(queue.Pop()->SequenceNumber(), 1); + EXPECT_EQ(queue.SizeInPackets(), 1); +} + } // namespace webrtc |