Reland "Reland "Only include overhead if using send side bandwidth estimation.""

This is a reland of 086055d0fd9b9b9efe8bcf85884324a019e9bd33

ANA was accitendly disabled even when transport sequence numbers were
negotiated due to a bug in how the audio send stream is configured. To
solve this we simply continue to always allow enabling ANA and leave it
up to the application to ensure that it's not used together with receive
side estimation.

Original change's description:
> Reland "Only include overhead if using send side bandwidth estimation."
>
> This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e
>
> Original change's description:
> > Only include overhead if using send side bandwidth estimation.
> >
> > Bug: webrtc:11298
> > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30382}
>
> Bug: webrtc:11298
> Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30390}

Bug: webrtc:11298
Change-Id: If2ad91e17ebfc85dc51edcd9607996e18c5d1f13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167883
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30413}
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 5e3b9ff..ba13fcb 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -342,6 +342,8 @@
       config_.max_bitrate_bps != -1 &&
       (allocate_audio_without_feedback_ || TransportSeqNumId(config_) != 0)) {
     rtp_transport_->AccountForAudioPacketsInPacedSender(true);
+    if (send_side_bwe_with_overhead_)
+      rtp_transport_->IncludeOverheadInPacedSender();
     rtp_rtcp_module_->SetAsPartOfAllocation(true);
     rtc::Event thread_sync_event;
     worker_queue_->PostTask([&] {
@@ -765,6 +767,8 @@
   if (!new_config.has_dscp && new_config.min_bitrate_bps != -1 &&
       new_config.max_bitrate_bps != -1 && TransportSeqNumId(new_config) != 0) {
     rtp_transport_->AccountForAudioPacketsInPacedSender(true);
+    if (send_side_bwe_with_overhead_)
+      rtp_transport_->IncludeOverheadInPacedSender();
     rtc::Event thread_sync_event;
     worker_queue_->PostTask([&] {
       RTC_DCHECK_RUN_ON(worker_queue_);