diff --git a/modules/audio_coding/neteq/delay_manager.cc b/modules/audio_coding/neteq/delay_manager.cc index 0df55c67cd..0003d3243b 100644 --- a/modules/audio_coding/neteq/delay_manager.cc +++ b/modules/audio_coding/neteq/delay_manager.cc @@ -78,23 +78,6 @@ DelayHistogramConfig GetDelayHistogramConfig() { return config; } -absl::optional GetExtraDelayMs() { - constexpr char kExtraDelayFieldTrial[] = "WebRTC-Audio-NetEqExtraDelay"; - if (!webrtc::field_trial::IsEnabled(kExtraDelayFieldTrial)) { - return absl::nullopt; - } - - const auto field_trial_string = - webrtc::field_trial::FindFullName(kExtraDelayFieldTrial); - int extra_delay_ms = -1; - sscanf(field_trial_string.c_str(), "Enabled-%d", &extra_delay_ms); - if (extra_delay_ms >= 0) { - RTC_LOG(LS_INFO) << "NetEq extra delay in milliseconds: " << extra_delay_ms; - return extra_delay_ms; - } - return absl::nullopt; -} - } // namespace namespace webrtc { @@ -120,8 +103,7 @@ DelayManager::DelayManager(size_t max_packets_in_buffer, minimum_delay_ms_(0), maximum_delay_ms_(0), last_pack_cng_or_dtmf_(1), - enable_rtx_handling_(enable_rtx_handling), - extra_delay_ms_(GetExtraDelayMs()) { + enable_rtx_handling_(enable_rtx_handling) { RTC_CHECK(histogram_); RTC_DCHECK_GE(base_minimum_delay_ms_, 0); @@ -299,10 +281,6 @@ int DelayManager::CalculateTargetLevel() { target_level = std::max(target_level, 1); // Scale to Q8 and assign to member variable. target_level_ = target_level << 8; - if (extra_delay_ms_ && packet_len_ms_ > 0) { - int extra_delay = (extra_delay_ms_.value() << 8) / packet_len_ms_; - target_level_ += extra_delay; - } return target_level_; } diff --git a/modules/audio_coding/neteq/delay_manager.h b/modules/audio_coding/neteq/delay_manager.h index d7bea9e168..ab9ba34167 100644 --- a/modules/audio_coding/neteq/delay_manager.h +++ b/modules/audio_coding/neteq/delay_manager.h @@ -131,8 +131,7 @@ class DelayManager { void UpdateEffectiveMinimumDelay(); // Makes sure that |target_level_| is not too large, taking - // |max_packets_in_buffer_| and |extra_delay_ms_| into account. This method is - // called by Update(). + // |max_packets_in_buffer_| into account. This method is called by Update(). void LimitTargetLevel(); // Makes sure that |delay_ms| is less than maximum delay, if any maximum @@ -175,8 +174,6 @@ class DelayManager { }; std::deque delay_history_; - const absl::optional extra_delay_ms_; - RTC_DISALLOW_COPY_AND_ASSIGN(DelayManager); }; diff --git a/modules/audio_coding/neteq/delay_manager_unittest.cc b/modules/audio_coding/neteq/delay_manager_unittest.cc index d60dbeb755..4a118f765f 100644 --- a/modules/audio_coding/neteq/delay_manager_unittest.cc +++ b/modules/audio_coding/neteq/delay_manager_unittest.cc @@ -594,27 +594,4 @@ TEST_F(DelayManagerTest, DecelerationTargetLevelOffset) { } } -TEST_F(DelayManagerTest, ExtraDelay) { - { - // Default behavior. Insert two packets so that a new target level is - // calculated. - SetPacketAudioLength(kFrameSizeMs); - InsertNextPacket(); - IncreaseTime(kFrameSizeMs); - InsertNextPacket(); - EXPECT_EQ(dm_->TargetLevel(), 1 << 8); - } - { - // Add 80 ms extra delay and calculate a new target level. - test::ScopedFieldTrials field_trial( - "WebRTC-Audio-NetEqExtraDelay/Enabled-80/"); - RecreateDelayManager(); - SetPacketAudioLength(kFrameSizeMs); - InsertNextPacket(); - IncreaseTime(kFrameSizeMs); - InsertNextPacket(); - EXPECT_EQ(dm_->TargetLevel(), 5 << 8); - } -} - } // namespace webrtc