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Adds trial to use link capacity estimate in Opus encoder.

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Since the link capacity is designed to be a more stable value, we don't
need the smoothing. This allows us to react faster to changes in link
capacity while still avoiding to react to changes in target bitrate due
to normal control behavior.

Bug: webrtc:9718
Change-Id: I2fbf6bb882f312a7b28ea43d27057886d035ac45
Reviewed-on: https://webrtc-review.googlesource.com/c/111511
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25745}
This commit is contained in:
Sebastian Jansson
2018-11-21 19:26:12 +01:00
committed by Commit Bot
parent 2ff3f49700
commit 8ac05ccaa7
2 changed files with 41 additions and 9 deletions
Download Patch File Download Diff File Expand all files Collapse all files

42
modules/audio_coding/codecs/opus/audio_encoder_opus.cc
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@ -471,6 +471,8 @@ 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")),
adjust_bandwidth_(
webrtc::field_trial::IsEnabled("WebRTC-AdjustOpusBandwidth")),
bitrate_changed_(true),
@ -605,7 +607,8 @@ void AudioEncoderOpusImpl::OnReceivedUplinkRecoverablePacketLossFraction(
void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
absl::optional<int64_t> bwe_period_ms) {
absl::optional<int64_t> bwe_period_ms,
absl::optional<int64_t> link_capacity_allocation_bps) {
if (audio_network_adaptor_) {
audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
// We give smoothed bitrate allocation to audio network adaptor as
@ -623,6 +626,9 @@ void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
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;
ApplyAudioNetworkAdaptor();
} else if (send_side_bwe_with_overhead_) {
if (!overhead_bytes_per_packet_) {
@ -641,6 +647,18 @@ void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
SetTargetBitrate(target_audio_bitrate_bps);
}
}
void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
absl::optional<int64_t> bwe_period_ms) {
OnReceivedUplinkBandwidth(target_audio_bitrate_bps, bwe_period_ms,
absl::nullopt);
}
void AudioEncoderOpusImpl::OnReceivedUplinkAllocation(
BitrateAllocationUpdate update) {
OnReceivedUplinkBandwidth(update.target_bitrate.bps(), update.bwe_period.ms(),
update.link_capacity.bps());
}
void AudioEncoderOpusImpl::OnReceivedRtt(int rtt_ms) {
if (!audio_network_adaptor_)
@ -875,14 +893,20 @@ AudioEncoderOpusImpl::DefaultAudioNetworkAdaptorCreator(
void AudioEncoderOpusImpl::MaybeUpdateUplinkBandwidth() {
if (audio_network_adaptor_) {
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 (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;
}
}
}
}