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Introduces common AudioAllocationSettings class.

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This class collects the field trial based configuration of audio
allocation and bandwidth in one place. This makes it easier
overview and prepares for future cleanup of the trials.

Bug: webrtc:9718
Change-Id: I34a441c0165b423f1e2ee63894337484684146ac
Reviewed-on: https://webrtc-review.googlesource.com/c/118282
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26370}
This commit is contained in:
Sebastian Jansson
2019-01-23 12:37:49 +01:00
committed by Commit Bot
parent 33b716f7dd
commit 470a5eae93
15 changed files with 294 additions and 125 deletions
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5
modules/pacing/paced_sender.cc
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@ -51,7 +51,7 @@ PacedSender::PacedSender(const Clock* clock,
drain_large_queues_(!field_trial::IsDisabled("WebRTC-Pacer-DrainQueue")),
send_padding_if_silent_(
field_trial::IsEnabled("WebRTC-Pacer-PadInSilence")),
video_blocks_audio_(!field_trial::IsDisabled("WebRTC-Pacer-BlockAudio")),
pace_audio_(!field_trial::IsDisabled("WebRTC-Pacer-BlockAudio")),
min_packet_limit_ms_("", kDefaultMinPacketLimitMs),
last_timestamp_ms_(clock_->TimeInMilliseconds()),
paused_(false),
@ -408,8 +408,7 @@ const RoundRobinPacketQueue::Packet* PacedSender::GetPendingPacket(
// reinsert it if send fails.
const RoundRobinPacketQueue::Packet* packet = &packets_.BeginPop();
bool audio_packet = packet->priority == kHighPriority;
bool apply_pacing =
!audio_packet || account_for_audio_ || video_blocks_audio_;
bool apply_pacing = !audio_packet || pace_audio_;
if (apply_pacing && (Congested() || (media_budget_.bytes_remaining() == 0 &&
pacing_info.probe_cluster_id ==
PacedPacketInfo::kNotAProbe))) {