Implementing APIs to set maximum and minimum for latency.

cpplint warnning fixed

Ready for review

BUG=
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1971004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4563 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/neteq4/delay_manager.cc

@@ -34,7 +34,9 @@ DelayManager::DelayManager(int max_packets_in_buffer,
       streaming_mode_(false),
       last_seq_no_(0),
       last_timestamp_(0),
-      extra_delay_ms_(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),
       max_timer_ms_(0),
@@ -218,21 +220,34 @@ void DelayManager::UpdateHistogram(size_t iat_packets) {
   iat_factor_ += (kIatFactor_ - iat_factor_ + 3) >> 2;
 }
 
-// Enforces upper limit for |target_level_|. The limit is chosen to be
+// Enforces upper and lower limits for |target_level_|. The upper limit is
-// 75% of |max_packets_in_buffer_|, to leave some headroom for natural
+// chosen to be minimum of i) 75% of |max_packets_in_buffer_|, to leave some
-// fluctuations around the target. If an extra delay is requested, the
+// headroom for natural fluctuations around the target, and ii) equivalent of
-// cap is lowered even further. Note that in practice, this does not have
+// |maximum_delay_ms_| in packets. Note that in practice, if no
-// any impact, since the target level is far below the buffer capacity in
+// |maximum_delay_ms_| is specified, this does not have any impact, since the
-// all reasonable cases.
+// target level is far below the buffer capacity in all reasonable cases.
+// The lower limit is equivalent of |minimum_delay_ms_| in packets. We update
+// |least_required_level_| while the above limits are applied.
 // TODO(hlundin): Move this check to the buffer logistics class.
 void DelayManager::LimitTargetLevel() {
-  int max_buffer_len = max_packets_in_buffer_;
+  least_required_delay_ms_ = (target_level_ * packet_len_ms_) >> 8;
-  if (extra_delay_ms_ > 0 && packet_len_ms_ > 0) {
+
-    max_buffer_len -= extra_delay_ms_ / packet_len_ms_;
+  if (packet_len_ms_ > 0 && minimum_delay_ms_ > 0) {
-    max_buffer_len = std::max(max_buffer_len, 1);  // Sanity check.
+    int minimum_delay_packet_q8 =  (minimum_delay_ms_ << 8) / packet_len_ms_;
+    target_level_ = std::max(target_level_, minimum_delay_packet_q8);
   }
-  max_buffer_len = (3 * (max_buffer_len << 8)) / 4;  // Shift to Q8, then 75%.
+
-  target_level_ = std::min(target_level_, max_buffer_len);
+  if (maximum_delay_ms_ > 0 && packet_len_ms_ > 0) {
+    int maximum_delay_packet_q8 = (maximum_delay_ms_ << 8) / packet_len_ms_;
+    target_level_ = std::min(target_level_, maximum_delay_packet_q8);
+  }
+
+  // Shift to Q8, then 75%.;
+  int max_buffer_packets_q8 = (3 * (max_packets_in_buffer_ << 8)) / 4;
+  target_level_ = std::min(target_level_, max_buffer_packets_q8);
+
+  // Sanity check, at least 1 packet (in Q8).
+  target_level_ = std::max(target_level_, 1 << 8);
 }
 
 int DelayManager::CalculateTargetLevel(int iat_packets) {
@@ -331,6 +346,9 @@ void DelayManager::UpdateCounters(int elapsed_time_ms) {
 
 void DelayManager::ResetPacketIatCount() { packet_iat_count_ms_ = 0; }
 
+// Note that |low_limit| and |higher_limit| are not assigned to
+// |minimum_delay_ms_| and |maximum_delay_ms_| defined by the client of this
+// class. They are computed from |target_level_| and used for decision making.
 void DelayManager::BufferLimits(int* lower_limit, int* higher_limit) const {
   if (!lower_limit || !higher_limit) {
     LOG_F(LS_ERROR) << "NULL pointers supplied as input";
@@ -338,29 +356,20 @@ void DelayManager::BufferLimits(int* lower_limit, int* higher_limit) const {
     return;
   }
 
-  int extra_delay_packets_q8 = 0;
   int window_20ms = 0x7FFF;  // Default large value for legacy bit-exactness.
   if (packet_len_ms_ > 0) {
-    extra_delay_packets_q8 = (extra_delay_ms_ << 8) / packet_len_ms_;
     window_20ms = (20 << 8) / packet_len_ms_;
   }
-  // |lower_limit| is 75% of |target_level_| + extra delay.
+
   // |target_level_| is in Q8 already.
-  *lower_limit = (target_level_ * 3) / 4 + extra_delay_packets_q8;
+  *lower_limit = (target_level_ * 3) / 4;
-  // |higher_limit| is equal to |target_level_| + extra delay, but should at
+  // |higher_limit| is equal to |target_level_|, but should at
   // least be 20 ms higher than |lower_limit_|.
-  *higher_limit = std::max(target_level_ + extra_delay_packets_q8,
+  *higher_limit = std::max(target_level_, *lower_limit + window_20ms);
-                           *lower_limit + window_20ms);
 }
 
 int DelayManager::TargetLevel() const {
-  if (packet_len_ms_ > 0) {
-    // Add |extra_delay_ms_| converted to packets in Q8.
-    return target_level_ + (extra_delay_ms_ << 8) / packet_len_ms_;
-  } else {
-    // Cannot convert |extra_delay_ms_|; simply return |target_level_|.
   return target_level_;
-  }
 }
 
 void DelayManager::LastDecoderType(NetEqDecoder decoder_type) {
@@ -375,8 +384,34 @@ void DelayManager::LastDecoderType(NetEqDecoder decoder_type) {
   }
 }
 
-void DelayManager::set_extra_delay_ms(int16_t delay) {
+bool DelayManager::SetMinimumDelay(int delay_ms) {
-  extra_delay_ms_ = delay;
+  // Minimum delay shouldn't be more than maximum delay, if any maximum is set.
+  // Also, if possible check |delay| to less than 75% of
+  // |max_packets_in_buffer_|.
+  if ((maximum_delay_ms_ > 0 && delay_ms > maximum_delay_ms_) ||
+      (packet_len_ms_ > 0 &&
+          delay_ms > 3 * max_packets_in_buffer_ * packet_len_ms_ / 4)) {
+    return false;
+  }
+  minimum_delay_ms_ = delay_ms;
+  return true;
+}
+
+bool DelayManager::SetMaximumDelay(int delay_ms) {
+  if (delay_ms == 0) {
+    // Zero input unsets the maximum delay.
+    maximum_delay_ms_ = 0;
+    return true;
+  } else if (delay_ms < minimum_delay_ms_ || delay_ms < packet_len_ms_) {
+    // Maximum delay shouldn't be less than minimum delay or less than a packet.
+    return false;
+  }
+  maximum_delay_ms_ = delay_ms;
+  return true;
+}
+
+int DelayManager::least_required_delay_ms() const {
+  return least_required_delay_ms_;
 }
 
 int DelayManager::base_target_level() const { return base_target_level_; }

webrtc/modules/audio_coding/neteq4/delay_manager.h

@@ -94,7 +94,10 @@ class DelayManager {
   virtual void LastDecoderType(NetEqDecoder decoder_type);
 
   // Accessors and mutators.
-  virtual void set_extra_delay_ms(int16_t delay);
+  // 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;
@@ -135,13 +138,19 @@ class DelayManager {
   int packet_iat_count_ms_;  // Milliseconds elapsed since last packet.
   int base_target_level_;   // Currently preferred buffer level before peak
                             // detection and streaming mode (Q0).
+  // TODO(turajs) change the comment according to the implementation of
+  // minimum-delay.
   int target_level_;  // Currently preferred buffer level in (fractions)
                       // of packets (Q8), before adding any extra delay.
   int packet_len_ms_;  // Length of audio in each incoming packet [ms].
   bool streaming_mode_;
   uint16_t last_seq_no_;  // Sequence number for last received packet.
   uint32_t last_timestamp_;  // Timestamp for the last received packet.
-  int extra_delay_ms_;  // Externally set extra delay.
+  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_|.
   int max_timer_ms_;  // Time elapsed since maximum was observed.

webrtc/modules/audio_coding/neteq4/delay_manager_unittest.cc

@@ -21,6 +21,7 @@
 namespace webrtc {
 
 using ::testing::Return;
+using ::testing::_;
 
 class DelayManagerTest : public ::testing::Test {
  protected:
@@ -193,9 +194,7 @@ TEST_F(DelayManagerTest, UpdatePeakFound) {
   EXPECT_EQ(5 << 8, higher);
 }
 
-TEST_F(DelayManagerTest, ExtraDelay) {
+TEST_F(DelayManagerTest, TargetDelay) {
-  const int kExtraDelayMs = 200;
-  dm_->set_extra_delay_ms(kExtraDelayMs);
   SetPacketAudioLength(kFrameSizeMs);
   // First packet arrival.
   InsertNextPacket();
@@ -208,24 +207,91 @@ TEST_F(DelayManagerTest, ExtraDelay) {
   EXPECT_CALL(detector_, Update(1, 1))
       .WillOnce(Return(false));
   InsertNextPacket();
-  const int kExpectedTarget = 1 + kExtraDelayMs / kFrameSizeMs;
+  const int kExpectedTarget = 1;
   EXPECT_EQ(kExpectedTarget << 8, dm_->TargetLevel());  // In Q8.
   EXPECT_EQ(1, dm_->base_target_level());
   int lower, higher;
   dm_->BufferLimits(&lower, &higher);
-  // Expect |lower| to be 75% of base target level + extra delay, and |higher|
+  // Expect |lower| to be 75% of base target level, and |higher| to be
-  // to be target level + extra delay, but at least leave 20 ms headroom from
+  // lower + 20 ms headroom.
-  // lower.
+  EXPECT_EQ((1 << 8) * 3 / 4, lower);
-  EXPECT_EQ((1 << 8) * 3 / 4 + (kExtraDelayMs << 8) / kFrameSizeMs, lower);
   EXPECT_EQ(lower + (20 << 8) / kFrameSizeMs, higher);
 }
 
+TEST_F(DelayManagerTest, MaxAndRequiredDelay) {
+  const int kExpectedTarget = 5;
+  const int kTimeIncrement = kExpectedTarget * kFrameSizeMs;
+  SetPacketAudioLength(kFrameSizeMs);
+  // First packet arrival.
+  InsertNextPacket();
+  // Second packet arrival.
+  // Expect detector update method to be called once with inter-arrival time
+  // equal to |kExpectedTarget| packet. Return true to indicate peaks found.
+  EXPECT_CALL(detector_, Update(kExpectedTarget, _))
+      .WillRepeatedly(Return(true));
+  EXPECT_CALL(detector_, MaxPeakHeight())
+      .WillRepeatedly(Return(kExpectedTarget));
+  IncreaseTime(kTimeIncrement);
+  InsertNextPacket();
+
+  // No limit is set.
+  EXPECT_EQ(kExpectedTarget << 8, dm_->TargetLevel());
+
+  int kMaxDelayPackets = kExpectedTarget - 2;
+  int kMaxDelayMs = kMaxDelayPackets * kFrameSizeMs;
+  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) {
+  const int kExpectedTarget = 5;
+  const int kTimeIncrement = kExpectedTarget * kFrameSizeMs;
+  SetPacketAudioLength(kFrameSizeMs);
+  // First packet arrival.
+  InsertNextPacket();
+  // Second packet arrival.
+  // Expect detector update method to be called once with inter-arrival time
+  // equal to |kExpectedTarget| packet. Return true to indicate peaks found.
+  EXPECT_CALL(detector_, Update(kExpectedTarget, _))
+      .WillRepeatedly(Return(true));
+  EXPECT_CALL(detector_, MaxPeakHeight())
+      .WillRepeatedly(Return(kExpectedTarget));
+  IncreaseTime(kTimeIncrement);
+  InsertNextPacket();
+
+  // No limit is applied.
+  EXPECT_EQ(kExpectedTarget << 8, dm_->TargetLevel());
+
+  int kMinDelayPackets = kExpectedTarget + 2;
+  int kMinDelayMs = kMinDelayPackets * kFrameSizeMs;
+  dm_->SetMinimumDelay(kMinDelayMs);
+  IncreaseTime(kTimeIncrement);
+  InsertNextPacket();
+  EXPECT_EQ(kExpectedTarget * kFrameSizeMs, dm_->least_required_delay_ms());
+  EXPECT_EQ(kMinDelayPackets << 8, dm_->TargetLevel());
+}
+
 TEST_F(DelayManagerTest, Failures) {
   // Wrong sample rate.
   EXPECT_EQ(-1, dm_->Update(0, 0, -1));
   // Wrong packet size.
   EXPECT_EQ(-1, dm_->SetPacketAudioLength(0));
   EXPECT_EQ(-1, dm_->SetPacketAudioLength(-1));
+
+  // Minimum delay higher than a maximum delay is not accepted.
+  EXPECT_TRUE(dm_->SetMaximumDelay(10));
+  EXPECT_FALSE(dm_->SetMinimumDelay(20));
+
+  // Maximum delay less than minimum delay is not accepted.
+  EXPECT_TRUE(dm_->SetMaximumDelay(100));
+  EXPECT_TRUE(dm_->SetMinimumDelay(80));
+  EXPECT_FALSE(dm_->SetMaximumDelay(60));
 }
 
 }  // namespace webrtc

webrtc/modules/audio_coding/neteq4/interface/neteq.h

@@ -151,10 +151,22 @@ class NetEq {
   // -1 on failure.
   virtual int RemovePayloadType(uint8_t rtp_payload_type) = 0;
 
-  // Sets the desired extra delay on top of what NetEq already applies due to
+  // Sets a minimum delay in millisecond for packet buffer. The minimum is
-  // current network situation. Used for synchronization with video. Returns
+  // maintained unless a higher latency is dictated by channel condition.
-  // true if successful, otherwise false.
+  // Returns true if the minimum is successfully applied, otherwise false is
-  virtual bool SetExtraDelay(int extra_delay_ms) = 0;
+  // returned.
+  virtual bool SetMinimumDelay(int delay_ms) = 0;
+
+  // Sets a maximum delay in milliseconds for packet buffer. The latency will
+  // not exceed the given value, even required delay (given the channel
+  // conditions) is higher.
+  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;

webrtc/modules/audio_coding/neteq4/neteq_impl.cc

@@ -236,16 +236,30 @@ int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) {
   return kFail;
 }
 
-bool NetEqImpl::SetExtraDelay(int extra_delay_ms) {
+bool NetEqImpl::SetMinimumDelay(int delay_ms) {
   CriticalSectionScoped lock(crit_sect_);
-  if (extra_delay_ms >= 0 && extra_delay_ms < 10000) {
+  if (delay_ms >= 0 && delay_ms < 10000) {
     assert(delay_manager_.get());
-    delay_manager_->set_extra_delay_ms(extra_delay_ms);
+    return delay_manager_->SetMinimumDelay(delay_ms);
-    return true;
   }
   return false;
 }
 
+bool NetEqImpl::SetMaximumDelay(int delay_ms) {
+  CriticalSectionScoped lock(crit_sect_);
+  if (delay_ms >= 0 && delay_ms < 10000) {
+    assert(delay_manager_.get());
+    return delay_manager_->SetMaximumDelay(delay_ms);
+  }
+  return false;
+}
+
+int NetEqImpl::LeastRequiredDelayMs() const {
+  CriticalSectionScoped lock(crit_sect_);
+  assert(delay_manager_.get());
+  return delay_manager_->least_required_delay_ms();
+}
+
 void NetEqImpl::SetPlayoutMode(NetEqPlayoutMode mode) {
   CriticalSectionScoped lock(crit_sect_);
   if (!decision_logic_.get() || mode != decision_logic_->playout_mode()) {

webrtc/modules/audio_coding/neteq4/neteq_impl.h

@@ -102,10 +102,11 @@ class NetEqImpl : public webrtc::NetEq {
   // -1 on failure.
   virtual int RemovePayloadType(uint8_t rtp_payload_type);
 
-  // Sets the desired extra delay on top of what NetEq already applies due to
+  virtual bool SetMinimumDelay(int delay_ms);
-  // current network situation. Used for synchronization with video. Returns
+
-  // true if successful, otherwise false.
+  virtual bool SetMaximumDelay(int delay_ms);
-  virtual bool SetExtraDelay(int extra_delay_ms);
+
+  virtual int LeastRequiredDelayMs() const;
 
   virtual int SetTargetDelay() { return kNotImplemented; }