Reland "Prepares PacingController for simplified packet queue."
This is a reland of acdc22d7845c5dde7c23366110e54e5d26127c85
Original change's description:
> Prepares PacingController for simplified packet queue.
>
> This CL removes references to RoundRobinPacketQueue::QueuedPacket,
> other than the method to release an RtpPacketToSend. It also moves
> both the BeginPop() and FinalizePop() to within a single helper
> method.
>
> A follow-up cleanup of the packet queue will stop exposing the
> QueuedPacket struct and replaces the the pop-methods with a single
> new one that just returns an RtpPacketToSend.
>
> Bug: webrtc:10809
> Change-Id: I5208a93e12e6b56714d483cc12d2a37225ea8e5e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159889
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29820}
TBR=philipel@webrtc.org
Bug: webrtc:10809
Change-Id: Id8196d9348d7fa69a5e410367b8a88e6039ef1b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160205
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29867}
diff --git a/modules/pacing/pacing_controller.cc b/modules/pacing/pacing_controller.cc
index 16d1f36..18a4f88 100644
--- a/modules/pacing/pacing_controller.cc
+++ b/modules/pacing/pacing_controller.cc
@@ -99,6 +99,8 @@
pace_audio_(!IsDisabled(*field_trials_, "WebRTC-Pacer-BlockAudio")),
small_first_probe_packet_(
IsEnabled(*field_trials_, "WebRTC-Pacer-SmallFirstProbePacket")),
+ send_side_bwe_with_overhead_(
+ IsEnabled(*field_trials_, "WebRTC-SendSideBwe-WithOverhead")),
min_packet_limit_(kDefaultMinPacketLimit),
last_timestamp_(clock_->CurrentTime()),
paused_(false),
@@ -463,8 +465,10 @@
// Fetch the next packet, so long as queue is not empty or budget is not
// exhausted.
- auto* packet = GetPendingPacket(pacing_info, target_send_time, now);
- if (packet == nullptr) {
+ std::unique_ptr<RtpPacketToSend> rtp_packet =
+ GetPendingPacket(pacing_info, target_send_time, now);
+
+ if (rtp_packet == nullptr) {
// No packet available to send, check if we should send padding.
DataSize padding_to_add = PaddingToAdd(recommended_probe_size, data_sent);
if (padding_to_add > DataSize::Zero()) {
@@ -485,14 +489,19 @@
break;
}
- std::unique_ptr<RtpPacketToSend> rtp_packet = packet->ReleasePacket();
RTC_DCHECK(rtp_packet);
+ RTC_DCHECK(rtp_packet->packet_type().has_value());
+ const RtpPacketToSend::Type packet_type = *rtp_packet->packet_type();
+ const DataSize packet_size = DataSize::bytes(
+ send_side_bwe_with_overhead_
+ ? rtp_packet->size()
+ : rtp_packet->payload_size() + rtp_packet->padding_size());
packet_sender_->SendRtpPacket(std::move(rtp_packet), pacing_info);
- data_sent += packet->size();
- // Send succeeded, remove it from the queue and update send/process time to
- // the target send time.
- OnPacketSent(packet, target_send_time);
+ data_sent += packet_size;
+
+ // Send done, update send/process time to the target send time.
+ OnPacketSent(packet_type, packet_size, target_send_time);
if (recommended_probe_size && data_sent > *recommended_probe_size)
break;
@@ -551,7 +560,7 @@
return DataSize::Zero();
}
-RoundRobinPacketQueue::QueuedPacket* PacingController::GetPendingPacket(
+std::unique_ptr<RtpPacketToSend> PacingController::GetPendingPacket(
const PacedPacketInfo& pacing_info,
Timestamp target_send_time,
Timestamp now) {
@@ -593,23 +602,28 @@
}
}
- return packet_queue_.BeginPop();
+ auto* queued_packet = packet_queue_.BeginPop();
+ std::unique_ptr<RtpPacketToSend> rtp_packet;
+ if (queued_packet != nullptr) {
+ rtp_packet = queued_packet->ReleasePacket();
+ packet_queue_.FinalizePop();
+ }
+ return rtp_packet;
}
-void PacingController::OnPacketSent(RoundRobinPacketQueue::QueuedPacket* packet,
+void PacingController::OnPacketSent(RtpPacketToSend::Type packet_type,
+ DataSize packet_size,
Timestamp send_time) {
if (!first_sent_packet_time_) {
first_sent_packet_time_ = send_time;
}
- bool audio_packet = packet->type() == RtpPacketToSend::Type::kAudio;
+ bool audio_packet = packet_type == RtpPacketToSend::Type::kAudio;
if (!audio_packet || account_for_audio_) {
// Update media bytes sent.
- UpdateBudgetWithSentData(packet->size());
+ UpdateBudgetWithSentData(packet_size);
}
last_send_time_ = send_time;
last_process_time_ = send_time;
- // Send succeeded, remove it from the queue.
- packet_queue_.FinalizePop();
}
void PacingController::OnPaddingSent(DataSize data_sent) {