Replacing bandwidth adaptation trial with stable target in Opus encoder.

This also means that the NetworkEstimate::bandwidth can be deprecated
as it's currently just a copy of the target_rate.

Bug: webrtc:10981
Change-Id: I1bc57b98480bd77ce052736b19d630c775428546
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153669
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29288}
This commit is contained in:
Sebastian Jansson
2019-09-24 17:55:50 +02:00
committed by Commit Bot
parent c30bc169ea
commit f34116e356
16 changed files with 39 additions and 67 deletions

View File

@ -425,8 +425,8 @@ AudioEncoderOpusImpl::AudioEncoderOpusImpl(
: payload_type_(payload_type),
send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
use_link_capacity_for_adaptation_(webrtc::field_trial::IsEnabled(
"WebRTC-Audio-LinkCapacityAdaptation")),
use_stable_target_for_adaptation_(webrtc::field_trial::IsEnabled(
"WebRTC-Audio-StableTargetAdaptation")),
adjust_bandwidth_(
webrtc::field_trial::IsEnabled("WebRTC-AdjustOpusBandwidth")),
bitrate_changed_(true),
@ -563,26 +563,28 @@ void AudioEncoderOpusImpl::OnReceivedUplinkRecoverablePacketLossFraction(
void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
absl::optional<int64_t> bwe_period_ms,
absl::optional<int64_t> link_capacity_allocation_bps) {
absl::optional<int64_t> stable_target_bitrate_bps) {
if (audio_network_adaptor_) {
audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
// We give smoothed bitrate allocation to audio network adaptor as
// the uplink bandwidth.
// The BWE spikes should not affect the bitrate smoother more than 25%.
// To simplify the calculations we use a step response as input signal.
// The step response of an exponential filter is
// u(t) = 1 - e^(-t / time_constant).
// In order to limit the affect of a BWE spike within 25% of its value
// before
// the next BWE update, we would choose a time constant that fulfills
// 1 - e^(-bwe_period_ms / time_constant) < 0.25
// Then 4 * bwe_period_ms is a good choice.
if (bwe_period_ms)
bitrate_smoother_->SetTimeConstantMs(*bwe_period_ms * 4);
bitrate_smoother_->AddSample(target_audio_bitrate_bps);
if (link_capacity_allocation_bps)
link_capacity_allocation_bps_ = link_capacity_allocation_bps;
if (use_stable_target_for_adaptation_) {
if (stable_target_bitrate_bps)
audio_network_adaptor_->SetUplinkBandwidth(*stable_target_bitrate_bps);
} else {
// We give smoothed bitrate allocation to audio network adaptor as
// the uplink bandwidth.
// The BWE spikes should not affect the bitrate smoother more than 25%.
// To simplify the calculations we use a step response as input signal.
// The step response of an exponential filter is
// u(t) = 1 - e^(-t / time_constant).
// In order to limit the affect of a BWE spike within 25% of its value
// before
// the next BWE update, we would choose a time constant that fulfills
// 1 - e^(-bwe_period_ms / time_constant) < 0.25
// Then 4 * bwe_period_ms is a good choice.
if (bwe_period_ms)
bitrate_smoother_->SetTimeConstantMs(*bwe_period_ms * 4);
bitrate_smoother_->AddSample(target_audio_bitrate_bps);
}
ApplyAudioNetworkAdaptor();
} else if (send_side_bwe_with_overhead_) {
@ -612,7 +614,7 @@ void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
void AudioEncoderOpusImpl::OnReceivedUplinkAllocation(
BitrateAllocationUpdate update) {
OnReceivedUplinkBandwidth(update.target_bitrate.bps(), update.bwe_period.ms(),
update.link_capacity.bps());
update.stable_target_bitrate.bps());
}
void AudioEncoderOpusImpl::OnReceivedRtt(int rtt_ms) {
@ -857,21 +859,15 @@ AudioEncoderOpusImpl::DefaultAudioNetworkAdaptorCreator(
}
void AudioEncoderOpusImpl::MaybeUpdateUplinkBandwidth() {
if (audio_network_adaptor_) {
if (use_link_capacity_for_adaptation_ && link_capacity_allocation_bps_) {
audio_network_adaptor_->SetUplinkBandwidth(
*link_capacity_allocation_bps_);
} else {
int64_t now_ms = rtc::TimeMillis();
if (!bitrate_smoother_last_update_time_ ||
now_ms - *bitrate_smoother_last_update_time_ >=
config_.uplink_bandwidth_update_interval_ms) {
absl::optional<float> smoothed_bitrate =
bitrate_smoother_->GetAverage();
if (smoothed_bitrate)
audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
bitrate_smoother_last_update_time_ = now_ms;
}
if (audio_network_adaptor_ && !use_stable_target_for_adaptation_) {
int64_t now_ms = rtc::TimeMillis();
if (!bitrate_smoother_last_update_time_ ||
now_ms - *bitrate_smoother_last_update_time_ >=
config_.uplink_bandwidth_update_interval_ms) {
absl::optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
if (smoothed_bitrate)
audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
bitrate_smoother_last_update_time_ = now_ms;
}
}
}

View File

@ -174,7 +174,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
AudioEncoderOpusConfig config_;
const int payload_type_;
const bool send_side_bwe_with_overhead_;
const bool use_link_capacity_for_adaptation_;
const bool use_stable_target_for_adaptation_;
const bool adjust_bandwidth_;
bool bitrate_changed_;
float packet_loss_rate_;
@ -192,7 +192,6 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
absl::optional<size_t> overhead_bytes_per_packet_;
const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
absl::optional<int64_t> bitrate_smoother_last_update_time_;
absl::optional<int64_t> link_capacity_allocation_bps_;
int consecutive_dtx_frames_;
friend struct AudioEncoderOpus;