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Reland "Call: Deduplicate SentPacket notifications"

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This is a reland of Ib121d5af07abe208bd7d36715a234f48cdabb032

In order to be backward compatible with bandwidth estimation behavior,
pass all packets without a |packet_id| to downstream.

Original change's description:
> Call: Deduplicate SentPacket notifications
>
> When bundling is in effect, multiple senders may be sharing the same
> transport. It means every |sent_packet| will be multiply notified from
> different channels, WebRtcVoiceMediaChannel or WebRtcVideoChannel.
> Record |last_sent_packet_| to deduplicate redundant notifications to
> downstream objects.
>
> This CL reduces 50% PostTask/Wakeup of Dynamic Mode Pacer.
>
> [1] https://datatracker.ietf.org/doc/html/rfc8829#section-4.1.1
> [2] https://datatracker.ietf.org/doc/html/rfc8843
>
> Bug: webrtc:13417
> Change-Id: Ib121d5af07abe208bd7d36715a234f48cdabb032
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238720
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35417}

Bug: webrtc:13417, webrtc:13437
Change-Id: Ia5e9fbe5e4f47fe851935ca2484125411114cb68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239437
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35492}
This commit is contained in:
Jianhui Dai
2021-12-01 19:23:31 +08:00
committed by WebRTC LUCI CQ
parent 5c198e100d
commit f349e53ca5
1 changed files with 19 additions and 0 deletions
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19
call/call.cc
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@ -466,6 +466,10 @@ class Call final : public webrtc::Call,
bool is_started_ RTC_GUARDED_BY(worker_thread_) = false;
RTC_NO_UNIQUE_ADDRESS SequenceChecker sent_packet_sequence_checker_;
absl::optional<rtc::SentPacket> last_sent_packet_
RTC_GUARDED_BY(sent_packet_sequence_checker_);
RTC_DISALLOW_COPY_AND_ASSIGN(Call);
};
} // namespace internal
@ -814,6 +818,7 @@ Call::Call(Clock* clock,
RTC_DCHECK(worker_thread_->IsCurrent());
send_transport_sequence_checker_.Detach();
sent_packet_sequence_checker_.Detach();
// Do not remove this call; it is here to convince the compiler that the
// WebRTC source timestamp string needs to be in the final binary.
@ -1382,6 +1387,20 @@ void Call::OnUpdateSyncGroup(webrtc::AudioReceiveStream& stream,
}
void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
RTC_DCHECK_RUN_ON(&sent_packet_sequence_checker_);
// When bundling is in effect, multiple senders may be sharing the same
// transport. It means every |sent_packet| will be multiply notified from
// different channels, WebRtcVoiceMediaChannel or WebRtcVideoChannel. Record
// |last_sent_packet_| to deduplicate redundant notifications to downstream.
// (https://crbug.com/webrtc/13437): Pass all packets without a |packet_id| to
// downstream.
if (last_sent_packet_.has_value() && last_sent_packet_->packet_id != -1 &&
last_sent_packet_->packet_id == sent_packet.packet_id &&
last_sent_packet_->send_time_ms == sent_packet.send_time_ms) {
return;
}
last_sent_packet_ = sent_packet;
// In production and with most tests, this method will be called on the
// network thread. However some test classes such as DirectTransport don't
// incorporate a network thread. This means that tests for RtpSenderEgress