Removing AudioAllocationSettings moving functionality to AudioSendStream.
This is a no-op change that just removes the AudioAllocationSettings
helper class that was previously introduced since the field trials in it
were used in several places. Those other usages has now been removed
and AudioSendStream is now the only user. By moving the trials directly
to AudioSendStream we reduce the reader overhead when trying to follow
what a particular field trial does.
The "WebRTC-Audio-ForceNoTWCC" trial was removed as it is always set
together with "WebRTC-Audio-ABWENoTWCC".
Bug: webrtc:9883
Change-Id: Ib63589255bfe7adb155ea41279bdcd153f1536c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155366
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29371}
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index fa16bb7..ba4518d 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -82,7 +82,7 @@
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../rtc_base:safe_minmax",
- "../rtc_base/experiments:audio_allocation_settings",
+ "../rtc_base/experiments:field_trial_parser",
"../system_wrappers",
"../system_wrappers:field_trial",
"../system_wrappers:metrics",
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 1ec2766..7f14768 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -41,7 +41,6 @@
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
-namespace internal {
namespace {
// TODO(eladalon): Subsequent CL will make these values experiment-dependent.
constexpr size_t kPacketLossTrackerMaxWindowSizeMs = 15000;
@@ -79,9 +78,28 @@
event_log->Log(std::make_unique<RtcEventAudioSendStreamConfig>(
std::move(rtclog_config)));
}
-
} // namespace
+constexpr char AudioAllocationConfig::kKey[];
+
+std::unique_ptr<StructParametersParser> AudioAllocationConfig::Parser() {
+ return StructParametersParser::Create( //
+ "min", &min_bitrate, //
+ "max", &max_bitrate, //
+ "prio_rate", &priority_bitrate, //
+ "prio_rate_raw", &priority_bitrate_raw, //
+ "rate_prio", &bitrate_priority);
+}
+
+AudioAllocationConfig::AudioAllocationConfig() {
+ Parser()->Parse(field_trial::FindFullName(kKey));
+ if (priority_bitrate_raw && !priority_bitrate.IsZero()) {
+ RTC_LOG(LS_WARNING) << "'priority_bitrate' and '_raw' are mutually "
+ "exclusive but both were configured.";
+ }
+}
+
+namespace internal {
AudioSendStream::AudioSendStream(
Clock* clock,
const webrtc::AudioSendStream::Config& config,
@@ -129,6 +147,13 @@
std::unique_ptr<voe::ChannelSendInterface> channel_send)
: clock_(clock),
worker_queue_(rtp_transport->GetWorkerQueue()),
+ audio_send_side_bwe_(field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")),
+ allocate_audio_without_feedback_(
+ field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC")),
+ enable_audio_alr_probing_(
+ !field_trial::IsDisabled("WebRTC-Audio-AlrProbing")),
+ send_side_bwe_with_overhead_(
+ field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
config_(Config(/*send_transport=*/nullptr, MediaTransportConfig())),
audio_state_(audio_state),
channel_send_(std::move(channel_send)),
@@ -286,15 +311,16 @@
bool transport_seq_num_id_changed =
new_ids.transport_sequence_number != old_ids.transport_sequence_number;
if (first_time || (transport_seq_num_id_changed &&
- !stream->allocation_settings_.ForceNoAudioFeedback())) {
+ !stream->allocate_audio_without_feedback_)) {
if (!first_time) {
channel_send->ResetSenderCongestionControlObjects();
}
RtcpBandwidthObserver* bandwidth_observer = nullptr;
- if (stream->allocation_settings_.ShouldSendTransportSequenceNumber(
- new_ids.transport_sequence_number)) {
+ if (stream->audio_send_side_bwe_ &&
+ !stream->allocate_audio_without_feedback_ &&
+ new_ids.transport_sequence_number != 0) {
channel_send->EnableSendTransportSequenceNumber(
new_ids.transport_sequence_number);
// Probing in application limited region is only used in combination with
@@ -303,7 +329,7 @@
if (stream->rtp_transport_) {
// Optionally request ALR probing but do not override any existing
// request from other streams.
- if (stream->allocation_settings_.RequestAlrProbing()) {
+ if (stream->enable_audio_alr_probing_) {
stream->rtp_transport_->EnablePeriodicAlrProbing(true);
}
bandwidth_observer = stream->rtp_transport_->GetBandwidthObserver();
@@ -345,10 +371,12 @@
if (sending_) {
return;
}
-
- if (allocation_settings_.IncludeAudioInAllocationOnStart(
- config_.min_bitrate_bps, config_.max_bitrate_bps, config_.has_dscp,
- TransportSeqNumId(config_))) {
+ // TODO(srte): We should not add audio to allocation just because
+ // audio_send_side_bwe_ is false.
+ if (!config_.has_dscp && config_.min_bitrate_bps != -1 &&
+ config_.max_bitrate_bps != -1 &&
+ (allocate_audio_without_feedback_ || TransportSeqNumId(config_) != 0 ||
+ !audio_send_side_bwe_)) {
rtp_transport_->AccountForAudioPacketsInPacedSender(true);
rtp_rtcp_module_->SetAsPartOfAllocation(true);
rtc::Event thread_sync_event;
@@ -812,13 +840,15 @@
stream->config_.max_bitrate_bps == new_config.max_bitrate_bps &&
stream->config_.bitrate_priority == new_config.bitrate_priority &&
(TransportSeqNumId(stream->config_) == TransportSeqNumId(new_config) ||
- stream->allocation_settings_.IgnoreSeqNumIdChange())) {
+ !stream->audio_send_side_bwe_)) {
return;
}
- if (stream->allocation_settings_.IncludeAudioInAllocationOnReconfigure(
- new_config.min_bitrate_bps, new_config.max_bitrate_bps,
- new_config.has_dscp, TransportSeqNumId(new_config))) {
+ // TODO(srte): We should not add audio to allocation just because
+ // audio_send_side_bwe_ is false.
+ if (!new_config.has_dscp && new_config.min_bitrate_bps != -1 &&
+ new_config.max_bitrate_bps != -1 &&
+ (TransportSeqNumId(new_config) != 0 || !stream->audio_send_side_bwe_)) {
stream->rtp_transport_->AccountForAudioPacketsInPacedSender(true);
rtc::Event thread_sync_event;
stream->worker_queue_->PostTask([&] {
@@ -847,12 +877,26 @@
// TODO(srte): Add overhead compensation here.
auto constraints = GetMinMaxBitrateConstraints();
+ DataRate max_overhead = DataRate::Zero();
+ if (send_side_bwe_with_overhead_) {
+ // TODO(srte): Respect |use_legacy_overhead_calculation_| here as well.
+ // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
+ constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
+ const TimeDelta kMinPacketDuration = TimeDelta::ms(20);
+ max_overhead = DataSize::bytes(kOverheadPerPacket) / kMinPacketDuration;
+ }
+ DataRate priority_bitrate =
+ allocation_settings_.priority_bitrate + max_overhead;
+
+ if (allocation_settings_.priority_bitrate_raw)
+ priority_bitrate = *allocation_settings_.priority_bitrate_raw;
+
bitrate_allocator_->AddObserver(
this,
MediaStreamAllocationConfig{
constraints.min.bps<uint32_t>(), constraints.max.bps<uint32_t>(), 0,
- allocation_settings_.DefaultPriorityBitrate().bps(), true,
- allocation_settings_.BitratePriority().value_or(
+ priority_bitrate.bps(), true,
+ allocation_settings_.bitrate_priority.value_or(
config_.bitrate_priority)});
}
@@ -875,15 +919,15 @@
DataRate::bps(config_.max_bitrate_bps)};
// If bitrates were explicitly overriden via field trial, use those values.
- if (allocation_settings_.MinBitrate())
- constraints.min = *allocation_settings_.MinBitrate();
- if (allocation_settings_.MaxBitrate())
- constraints.max = *allocation_settings_.MaxBitrate();
+ if (allocation_settings_.min_bitrate)
+ constraints.min = *allocation_settings_.min_bitrate;
+ if (allocation_settings_.max_bitrate)
+ constraints.max = *allocation_settings_.max_bitrate;
RTC_DCHECK_GE(constraints.min, DataRate::Zero());
RTC_DCHECK_GE(constraints.max, DataRate::Zero());
RTC_DCHECK_GE(constraints.max, constraints.min);
- if (allocation_settings_.IncludeOverheadInAudioAllocation()) {
+ if (send_side_bwe_with_overhead_) {
if (use_legacy_overhead_calculation_) {
// OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
const DataSize kOverheadPerPacket = DataSize::bytes(20 + 8 + 10 + 12);
diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index 80b5087..3cd3a6c 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -23,7 +23,7 @@
#include "call/bitrate_allocator.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "rtc_base/constructor_magic.h"
-#include "rtc_base/experiments/audio_allocation_settings.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/thread_checker.h"
@@ -34,6 +34,21 @@
class RtcpRttStats;
class RtpTransportControllerSendInterface;
+struct AudioAllocationConfig {
+ static constexpr char kKey[] = "WebRTC-Audio-Allocation";
+ // Field Trial configured bitrates to use as overrides over default/user
+ // configured bitrate range when audio bitrate allocation is enabled.
+ absl::optional<DataRate> min_bitrate;
+ absl::optional<DataRate> max_bitrate;
+ DataRate priority_bitrate = DataRate::Zero();
+ // By default the priority_bitrate is compensated for packet overhead.
+ // Use this flag to configure a raw value instead.
+ absl::optional<DataRate> priority_bitrate_raw;
+ absl::optional<double> bitrate_priority;
+
+ std::unique_ptr<StructParametersParser> Parser();
+ AudioAllocationConfig();
+};
namespace internal {
class AudioState;
@@ -153,7 +168,14 @@
rtc::ThreadChecker pacer_thread_checker_;
rtc::RaceChecker audio_capture_race_checker_;
rtc::TaskQueue* worker_queue_;
- const AudioAllocationSettings allocation_settings_;
+
+ const bool audio_send_side_bwe_;
+ const bool allocate_audio_without_feedback_;
+ const bool force_no_audio_feedback_ = allocate_audio_without_feedback_;
+ const bool enable_audio_alr_probing_;
+ const bool send_side_bwe_with_overhead_;
+ const AudioAllocationConfig allocation_settings_;
+
rtc::CriticalSection config_cs_;
webrtc::AudioSendStream::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;