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Remove packet loss rate optimization and minimum field trial.

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Bug: webrtc:11664
Change-Id: I63fab70e5ae85e2971bed4998ab3b15f61f9e1c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176752
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31480}
This commit is contained in:
Jakob Ivarsson
2020-06-09 16:05:42 +02:00
committed by Commit Bot
parent 2d27b1ab0c
commit 7649006692
3 changed files with 12 additions and 179 deletions
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74
modules/audio_coding/codecs/opus/audio_encoder_opus.cc
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@ -66,46 +66,7 @@ constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60};
// PacketLossFractionSmoother uses an exponential filter with a time constant
// of -1.0 / ln(0.9999) = 10000 ms.
constexpr float kAlphaForPacketLossFractionSmoother = 0.9999f;
// Optimize the loss rate to configure Opus. Basically, optimized loss rate is
// the input loss rate rounded down to various levels, because a robustly good
// audio quality is achieved by lowering the packet loss down.
// Additionally, to prevent toggling, margins are used, i.e., when jumping to
// a loss rate from below, a higher threshold is used than jumping to the same
// level from above.
float OptimizePacketLossRate(float new_loss_rate, float old_loss_rate) {
RTC_DCHECK_GE(new_loss_rate, 0.0f);
RTC_DCHECK_LE(new_loss_rate, 1.0f);
RTC_DCHECK_GE(old_loss_rate, 0.0f);
RTC_DCHECK_LE(old_loss_rate, 1.0f);
constexpr float kPacketLossRate20 = 0.20f;
constexpr float kPacketLossRate10 = 0.10f;
constexpr float kPacketLossRate5 = 0.05f;
constexpr float kPacketLossRate1 = 0.01f;
constexpr float kLossRate20Margin = 0.02f;
constexpr float kLossRate10Margin = 0.01f;
constexpr float kLossRate5Margin = 0.01f;
if (new_loss_rate >=
kPacketLossRate20 +
kLossRate20Margin *
(kPacketLossRate20 - old_loss_rate > 0 ? 1 : -1)) {
return kPacketLossRate20;
} else if (new_loss_rate >=
kPacketLossRate10 +
kLossRate10Margin *
(kPacketLossRate10 - old_loss_rate > 0 ? 1 : -1)) {
return kPacketLossRate10;
} else if (new_loss_rate >=
kPacketLossRate5 +
kLossRate5Margin *
(kPacketLossRate5 - old_loss_rate > 0 ? 1 : -1)) {
return kPacketLossRate5;
} else if (new_loss_rate >= kPacketLossRate1) {
return kPacketLossRate1;
} else {
return 0.0f;
}
}
constexpr float kMaxPacketLossFraction = 0.2f;
int CalculateDefaultBitrate(int max_playback_rate, size_t num_channels) {
const int bitrate = [&] {
@ -201,35 +162,6 @@ int GetBitrateBps(const AudioEncoderOpusConfig& config) {
return *config.bitrate_bps;
}
bool IsValidPacketLossRate(int value) {
return value >= 0 && value <= 100;
}
float ToFraction(int percent) {
return static_cast<float>(percent) / 100;
}
float GetMinPacketLossRate() {
constexpr char kPacketLossFieldTrial[] = "WebRTC-Audio-OpusMinPacketLossRate";
const bool use_opus_min_packet_loss_rate =
webrtc::field_trial::IsEnabled(kPacketLossFieldTrial);
if (use_opus_min_packet_loss_rate) {
const std::string field_trial_string =
webrtc::field_trial::FindFullName(kPacketLossFieldTrial);
constexpr int kDefaultMinPacketLossRate = 1;
int value = kDefaultMinPacketLossRate;
if (sscanf(field_trial_string.c_str(), "Enabled-%d", &value) == 1 &&
!IsValidPacketLossRate(value)) {
RTC_LOG(LS_WARNING) << "Invalid parameter for " << kPacketLossFieldTrial
<< ", using default value: "
<< kDefaultMinPacketLossRate;
value = kDefaultMinPacketLossRate;
}
return ToFraction(value);
}
return 0.0;
}
std::vector<float> GetBitrateMultipliers() {
constexpr char kBitrateMultipliersName[] =
"WebRTC-Audio-OpusBitrateMultipliers";
@ -432,7 +364,6 @@ AudioEncoderOpusImpl::AudioEncoderOpusImpl(
bitrate_changed_(true),
bitrate_multipliers_(GetBitrateMultipliers()),
packet_loss_rate_(0.0),
min_packet_loss_rate_(GetMinPacketLossRate()),
inst_(nullptr),
packet_loss_fraction_smoother_(new PacketLossFractionSmoother()),
audio_network_adaptor_creator_(audio_network_adaptor_creator),
@ -789,8 +720,7 @@ void AudioEncoderOpusImpl::SetNumChannelsToEncode(
}
void AudioEncoderOpusImpl::SetProjectedPacketLossRate(float fraction) {
fraction = OptimizePacketLossRate(fraction, packet_loss_rate_);
fraction = std::max(fraction, min_packet_loss_rate_);
fraction = std::min(std::max(fraction, 0.0f), kMaxPacketLossFraction);
if (packet_loss_rate_ != fraction) {
packet_loss_rate_ = fraction;
RTC_CHECK_EQ(