Delete AudioCodingModule::LeastRequiredDelayMs and related NetEq code.
Bug: None
Change-Id: I2f68502d19415899b3694f7bf5da523da831b223
Reviewed-on: https://webrtc-review.googlesource.com/95640
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24439}diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 0a88e70..892abe5 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -60,10 +60,6 @@
return -1;
}
-int AcmReceiver::LeastRequiredDelayMs() const {
- return neteq_->LeastRequiredDelayMs();
-}
-
absl::optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
rtc::CritScope lock(&crit_sect_);
return last_packet_sample_rate_hz_;
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index f133a87..0731677 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -143,13 +143,6 @@
int SetMaximumDelay(int delay_ms);
//
- // Get least required delay computed based on channel conditions. Note that
- // this is before applying any user-defined limits (specified by calling
- // (SetMinimumDelay() and/or SetMaximumDelay()).
- //
- int LeastRequiredDelayMs() const;
-
- //
// Resets the initial delay to zero.
//
void ResetInitialDelay();
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 92a4a9d..2c96fc4 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -155,9 +155,6 @@
// Maximum playout delay.
int SetMaximumPlayoutDelay(int time_ms) override;
- // Smallest latency NetEq will maintain.
- int LeastRequiredDelayMs() const override;
-
absl::optional<uint32_t> PlayoutTimestamp() override;
int FilteredCurrentDelayMs() const override;
@@ -1207,10 +1204,6 @@
return receiver_.GetNackList(round_trip_time_ms);
}
-int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
- return receiver_.LeastRequiredDelayMs();
-}
-
void AudioCodingModuleImpl::GetDecodingCallStatistics(
AudioDecodingCallStats* call_stats) const {
receiver_.GetDecodingCallStatistics(call_stats);
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 85e340b..454e0e4 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -602,15 +602,6 @@
//
virtual int SetMaximumPlayoutDelay(int time_ms) = 0;
- // TODO(kwiberg): Consider if this is needed anymore, now that voe::Channel
- // doesn't use it.
- // The shortest latency, in milliseconds, required by jitter buffer. This
- // is computed based on inter-arrival times and playout mode of NetEq. The
- // actual delay is the maximum of least-required-delay and the minimum-delay
- // specified by SetMinumumPlayoutDelay() API.
- //
- virtual int LeastRequiredDelayMs() const = 0;
-
///////////////////////////////////////////////////////////////////////////
// int32_t PlayoutTimestamp()
// The send timestamp of an RTP packet is associated with the decoded
diff --git a/modules/audio_coding/neteq/delay_manager.cc b/modules/audio_coding/neteq/delay_manager.cc
index a945cdc..84d457c 100644
--- a/modules/audio_coding/neteq/delay_manager.cc
+++ b/modules/audio_coding/neteq/delay_manager.cc
@@ -40,7 +40,6 @@
last_seq_no_(0),
last_timestamp_(0),
minimum_delay_ms_(0),
- least_required_delay_ms_(target_level_),
maximum_delay_ms_(target_level_),
iat_cumulative_sum_(0),
max_iat_cumulative_sum_(0),
@@ -234,8 +233,6 @@
// |least_required_level_| while the above limits are applied.
// TODO(hlundin): Move this check to the buffer logistics class.
void DelayManager::LimitTargetLevel() {
- least_required_delay_ms_ = (target_level_ * packet_len_ms_) >> 8;
-
if (packet_len_ms_ > 0 && minimum_delay_ms_ > 0) {
int minimum_delay_packet_q8 = (minimum_delay_ms_ << 8) / packet_len_ms_;
target_level_ = std::max(target_level_, minimum_delay_packet_q8);
@@ -467,10 +464,6 @@
return true;
}
-int DelayManager::least_required_delay_ms() const {
- return least_required_delay_ms_;
-}
-
int DelayManager::base_target_level() const {
return base_target_level_;
}
diff --git a/modules/audio_coding/neteq/delay_manager.h b/modules/audio_coding/neteq/delay_manager.h
index d75d5b4..00e39af 100644
--- a/modules/audio_coding/neteq/delay_manager.h
+++ b/modules/audio_coding/neteq/delay_manager.h
@@ -109,7 +109,6 @@
// Assuming |delay| is in valid range.
virtual bool SetMinimumDelay(int delay_ms);
virtual bool SetMaximumDelay(int delay_ms);
- virtual int least_required_delay_ms() const;
virtual int base_target_level() const;
virtual void set_streaming_mode(bool value);
virtual int last_pack_cng_or_dtmf() const;
@@ -161,10 +160,6 @@
uint16_t last_seq_no_; // Sequence number for last received packet.
uint32_t last_timestamp_; // Timestamp for the last received packet.
int minimum_delay_ms_; // Externally set minimum delay.
- int least_required_delay_ms_; // Smallest preferred buffer level (same unit
- // as |target_level_|), before applying
- // |minimum_delay_ms_| and/or
- // |maximum_delay_ms_|.
int maximum_delay_ms_; // Externally set maximum allowed delay.
int iat_cumulative_sum_; // Cumulative sum of delta inter-arrival times.
int max_iat_cumulative_sum_; // Max of |iat_cumulative_sum_|.
diff --git a/modules/audio_coding/neteq/delay_manager_unittest.cc b/modules/audio_coding/neteq/delay_manager_unittest.cc
index f9c5680..6afed66 100644
--- a/modules/audio_coding/neteq/delay_manager_unittest.cc
+++ b/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -200,7 +200,7 @@
EXPECT_EQ(lower + (20 << 8) / kFrameSizeMs, higher);
}
-TEST_F(DelayManagerTest, MaxAndRequiredDelay) {
+TEST_F(DelayManagerTest, MaxDelay) {
const int kExpectedTarget = 5;
const int kTimeIncrement = kExpectedTarget * kFrameSizeMs;
SetPacketAudioLength(kFrameSizeMs);
@@ -224,14 +224,13 @@
EXPECT_TRUE(dm_->SetMaximumDelay(kMaxDelayMs));
IncreaseTime(kTimeIncrement);
InsertNextPacket();
- EXPECT_EQ(kExpectedTarget * kFrameSizeMs, dm_->least_required_delay_ms());
EXPECT_EQ(kMaxDelayPackets << 8, dm_->TargetLevel());
// Target level at least should be one packet.
EXPECT_FALSE(dm_->SetMaximumDelay(kFrameSizeMs - 1));
}
-TEST_F(DelayManagerTest, MinAndRequiredDelay) {
+TEST_F(DelayManagerTest, MinDelay) {
const int kExpectedTarget = 5;
const int kTimeIncrement = kExpectedTarget * kFrameSizeMs;
SetPacketAudioLength(kFrameSizeMs);
@@ -255,7 +254,6 @@
dm_->SetMinimumDelay(kMinDelayMs);
IncreaseTime(kTimeIncrement);
InsertNextPacket();
- EXPECT_EQ(kExpectedTarget * kFrameSizeMs, dm_->least_required_delay_ms());
EXPECT_EQ(kMinDelayPackets << 8, dm_->TargetLevel());
}
diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
index 731dbe6..ce5e7ca 100644
--- a/modules/audio_coding/neteq/include/neteq.h
+++ b/modules/audio_coding/neteq/include/neteq.h
@@ -180,12 +180,6 @@
// the |max_delay_ms| value in the NetEq::Config struct.
virtual bool SetMaximumDelay(int delay_ms) = 0;
- // The smallest latency required. This is computed bases on inter-arrival
- // time and internal NetEq logic. Note that in computing this latency none of
- // the user defined limits (applied by calling setMinimumDelay() and/or
- // SetMaximumDelay()) are applied.
- virtual int LeastRequiredDelayMs() const = 0;
-
// Not implemented.
virtual int SetTargetDelay() = 0;
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 0ef3263..c9edbce 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -307,12 +307,6 @@
return false;
}
-int NetEqImpl::LeastRequiredDelayMs() const {
- rtc::CritScope lock(&crit_sect_);
- assert(delay_manager_.get());
- return delay_manager_->least_required_delay_ms();
-}
-
int NetEqImpl::SetTargetDelay() {
return kNotImplemented;
}
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index e0e0c13..32af294 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -157,8 +157,6 @@
bool SetMaximumDelay(int delay_ms) override;
- int LeastRequiredDelayMs() const override;
-
int SetTargetDelay() override;
int TargetDelayMs() const override;
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 3b129ea..071a6d8 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -57,18 +57,6 @@
EXPECT_EQ(-1, SetMinimumDelay(10001));
}
- void NoTargetDelayBufferSizeChanges() {
- for (int n = 0; n < 30; ++n) // Run enough iterations.
- Run(true);
- int clean_optimal_delay = GetCurrentOptimalDelayMs();
- Run(false); // Run with jitter.
- int jittery_optimal_delay = GetCurrentOptimalDelayMs();
- EXPECT_GT(jittery_optimal_delay, clean_optimal_delay);
- int required_delay = RequiredDelay();
- EXPECT_GT(required_delay, 0);
- EXPECT_NEAR(required_delay, jittery_optimal_delay, 1);
- }
-
void WithTargetDelayBufferNotChanging() {
// A target delay that is one packet larger than jitter.
const int kTargetDelayMs =
@@ -83,37 +71,6 @@
EXPECT_EQ(jittery_optimal_delay, clean_optimal_delay);
}
- void RequiredDelayAtCorrectRange() {
- for (int n = 0; n < 30; ++n) // Run clean and store delay.
- Run(true);
- int clean_optimal_delay = GetCurrentOptimalDelayMs();
-
- // A relatively large delay.
- const int kTargetDelayMs =
- (kInterarrivalJitterPacket + 10) * kNum10msPerFrame * 10;
- ASSERT_EQ(0, SetMinimumDelay(kTargetDelayMs));
- for (int n = 0; n < 300; ++n) // Run enough iterations to fill the buffer.
- Run(true);
- Run(false); // Run with jitter.
-
- int jittery_optimal_delay = GetCurrentOptimalDelayMs();
- EXPECT_EQ(kTargetDelayMs, jittery_optimal_delay);
-
- int required_delay = RequiredDelay();
-
- // Checking |required_delay| is in correct range.
- EXPECT_GT(required_delay, 0);
- EXPECT_GT(jittery_optimal_delay, required_delay);
- EXPECT_GT(required_delay, clean_optimal_delay);
-
- // A tighter check for the value of |required_delay|.
- // The jitter forces a delay of
- // |kInterarrivalJitterPacket * kNum10msPerFrame * 10| milliseconds. So we
- // expect |required_delay| be close to that.
- EXPECT_NEAR(kInterarrivalJitterPacket * kNum10msPerFrame * 10,
- required_delay, 1);
- }
-
void TargetDelayBufferMinMax() {
const int kTargetMinDelayMs = kNum10msPerFrame * 10;
ASSERT_EQ(0, SetMinimumDelay(kTargetMinDelayMs));
@@ -193,8 +150,6 @@
return stats.preferredBufferSize;
}
- int RequiredDelay() { return acm_->LeastRequiredDelayMs(); }
-
std::unique_ptr<AudioCodingModule> acm_;
WebRtcRTPHeader rtp_info_;
uint8_t payload_[kPayloadLenBytes];
@@ -212,17 +167,6 @@
// Flaky on iOS: webrtc:7057.
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
-#define MAYBE_NoTargetDelayBufferSizeChanges \
- DISABLED_NoTargetDelayBufferSizeChanges
-#else
-#define MAYBE_NoTargetDelayBufferSizeChanges NoTargetDelayBufferSizeChanges
-#endif
-TEST_F(TargetDelayTest, MAYBE_NoTargetDelayBufferSizeChanges) {
- NoTargetDelayBufferSizeChanges();
-}
-
-// Flaky on iOS: webrtc:7057.
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
#define MAYBE_WithTargetDelayBufferNotChanging \
DISABLED_WithTargetDelayBufferNotChanging
#else
@@ -234,16 +178,6 @@
// Flaky on iOS: webrtc:7057.
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
-#define MAYBE_RequiredDelayAtCorrectRange DISABLED_RequiredDelayAtCorrectRange
-#else
-#define MAYBE_RequiredDelayAtCorrectRange RequiredDelayAtCorrectRange
-#endif
-TEST_F(TargetDelayTest, MAYBE_RequiredDelayAtCorrectRange) {
- RequiredDelayAtCorrectRange();
-}
-
-// Flaky on iOS: webrtc:7057.
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
#define MAYBE_TargetDelayBufferMinMax DISABLED_TargetDelayBufferMinMax
#else
#define MAYBE_TargetDelayBufferMinMax TargetDelayBufferMinMax
