NetEq: Use TickTimer in PacketBuffer

This change makes use of the TickTimer::Stopwatch in Packets. When a
packet is inserted into the PacketBuffer, a Stopwatch object is
attached to it. When the packet is extracted from the buffer, the
Stopwatch is read to know how long the packet waited in the buffer.

BUG=webrtc:5608

Review URL: https://codereview.webrtc.org/1917913002

Cr-Commit-Position: refs/heads/master@{#12508}

webrtc/modules/audio_coding/BUILD.gn

@@ -791,6 +791,8 @@ source_set("neteq") {
     "neteq/neteq_impl.h",
     "neteq/normal.cc",
     "neteq/normal.h",
+    "neteq/packet.cc",
+    "neteq/packet.h",
     "neteq/packet_buffer.cc",
     "neteq/packet_buffer.h",
     "neteq/payload_splitter.cc",

webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc

@@ -17,6 +17,7 @@
 #include "webrtc/modules/audio_coding/neteq/delay_manager.h"
 #include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
 #include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 
 namespace webrtc {
 
@@ -24,7 +25,8 @@ TEST(DecisionLogic, CreateAndDestroy) {
   int fs_hz = 8000;
   int output_size_samples = fs_hz / 100;  // Samples per 10 ms.
   DecoderDatabase decoder_database;
-  PacketBuffer packet_buffer(10);
+  TickTimer tick_timer;
+  PacketBuffer packet_buffer(10, &tick_timer);
   DelayPeakDetector delay_peak_detector;
   DelayManager delay_manager(240, &delay_peak_detector);
   BufferLevelFilter buffer_level_filter;

webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h

@@ -19,8 +19,8 @@ namespace webrtc {
 
 class MockPacketBuffer : public PacketBuffer {
  public:
-  MockPacketBuffer(size_t max_number_of_packets)
+  MockPacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer)
-      : PacketBuffer(max_number_of_packets) {}
+      : PacketBuffer(max_number_of_packets, tick_timer) {}
   virtual ~MockPacketBuffer() { Die(); }
   MOCK_METHOD0(Die, void());
   MOCK_METHOD0(Flush,

webrtc/modules/audio_coding/neteq/neteq.cc

@@ -54,7 +54,8 @@ NetEq* NetEq::Create(const NetEq::Config& config) {
   delay_manager->SetMaximumDelay(config.max_delay_ms);
   DtmfBuffer* dtmf_buffer = new DtmfBuffer(config.sample_rate_hz);
   DtmfToneGenerator* dtmf_tone_generator = new DtmfToneGenerator;
-  PacketBuffer* packet_buffer = new PacketBuffer(config.max_packets_in_buffer);
+  PacketBuffer* packet_buffer =
+      new PacketBuffer(config.max_packets_in_buffer, tick_timer.get());
   PayloadSplitter* payload_splitter = new PayloadSplitter;
   TimestampScaler* timestamp_scaler = new TimestampScaler(*decoder_database);
   AccelerateFactory* accelerate_factory = new AccelerateFactory;

webrtc/modules/audio_coding/neteq/neteq.gypi

@@ -105,6 +105,8 @@
         'statistics_calculator.h',
         'normal.cc',
         'normal.h',
+        'packet.cc',
+        'packet.h',
         'packet_buffer.cc',
         'packet_buffer.h',
         'payload_splitter.cc',

webrtc/modules/audio_coding/neteq/neteq_impl.cc

@@ -536,7 +536,8 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     packet->header.numCSRCs = 0;
     packet->payload_length = payload.size();
     packet->primary = true;
-    packet->waiting_time = 0;
+    // Waiting time will be set upon inserting the packet in the buffer.
+    RTC_DCHECK(!packet->waiting_time);
     packet->payload = new uint8_t[packet->payload_length];
     packet->sync_packet = is_sync_packet;
     if (!packet->payload) {
@@ -1002,7 +1003,6 @@ int NetEqImpl::GetDecision(Operations* operation,
   *operation = kUndefined;
 
   // Increment time counters.
-  packet_buffer_->IncrementWaitingTimes();
   stats_.IncreaseCounter(output_size_samples_, fs_hz_);
 
   assert(sync_buffer_.get());
@@ -1931,8 +1931,7 @@ int NetEqImpl::ExtractPackets(size_t required_samples,
       return -1;
     }
     stats_.PacketsDiscarded(discard_count);
-    // Store waiting time in ms; packets->waiting_time is in "output blocks".
+    stats_.StoreWaitingTime(packet->waiting_time->ElapsedMs());
-    stats_.StoreWaitingTime(packet->waiting_time * kOutputSizeMs);
     assert(packet->payload_length > 0);
     packet_list->push_back(packet);  // Store packet in list.
 

webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc

@@ -130,10 +130,12 @@ class NetEqImplTest : public ::testing::Test {
       dtmf_tone_generator_ = new DtmfToneGenerator;
     }
     if (use_mock_packet_buffer_) {
-      mock_packet_buffer_ = new MockPacketBuffer(config_.max_packets_in_buffer);
+      mock_packet_buffer_ =
+          new MockPacketBuffer(config_.max_packets_in_buffer, tick_timer_);
       packet_buffer_ = mock_packet_buffer_;
     } else {
-      packet_buffer_ = new PacketBuffer(config_.max_packets_in_buffer);
+      packet_buffer_ =
+          new PacketBuffer(config_.max_packets_in_buffer, tick_timer_);
     }
     if (use_mock_payload_splitter_) {
       mock_payload_splitter_ = new MockPayloadSplitter;

webrtc/modules/audio_coding/neteq/packet.cc

@@ -0,0 +1,19 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/neteq/packet.h"
+
+namespace webrtc {
+
+Packet::Packet() = default;
+
+Packet::~Packet() = default;
+
+}  // namespace webrtc

webrtc/modules/audio_coding/neteq/packet.h

@@ -12,7 +12,9 @@
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_H_
 
 #include <list>
+#include <memory>
 
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
@@ -21,20 +23,15 @@ namespace webrtc {
 // Struct for holding RTP packets.
 struct Packet {
   RTPHeader header;
-  uint8_t* payload;  // Datagram excluding RTP header and header extension.
+  // Datagram excluding RTP header and header extension.
-  size_t payload_length;
+  uint8_t* payload = nullptr;
-  bool primary;  // Primary, i.e., not redundant payload.
+  size_t payload_length = 0;
-  int waiting_time;
+  bool primary = true;  // Primary, i.e., not redundant payload.
-  bool sync_packet;
+  bool sync_packet = false;
+  std::unique_ptr<TickTimer::Stopwatch> waiting_time;
 
-  // Constructor.
+  Packet();
-  Packet()
+  ~Packet();
-      : payload(NULL),
-        payload_length(0),
-        primary(true),
-        waiting_time(0),
-        sync_packet(false) {
-  }
 
   // Comparison operators. Establish a packet ordering based on (1) timestamp,
   // (2) sequence number, (3) regular packet vs sync-packet and (4) redundancy.

webrtc/modules/audio_coding/neteq/packet_buffer.cc

@@ -19,6 +19,7 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 
 namespace webrtc {
 
@@ -37,8 +38,9 @@ class NewTimestampIsLarger {
   const Packet* new_packet_;
 };
 
-PacketBuffer::PacketBuffer(size_t max_number_of_packets)
+PacketBuffer::PacketBuffer(size_t max_number_of_packets,
-    : max_number_of_packets_(max_number_of_packets) {}
+                           const TickTimer* tick_timer)
+    : max_number_of_packets_(max_number_of_packets), tick_timer_(tick_timer) {}
 
 // Destructor. All packets in the buffer will be destroyed.
 PacketBuffer::~PacketBuffer() {
@@ -65,6 +67,8 @@ int PacketBuffer::InsertPacket(Packet* packet) {
 
   int return_val = kOK;
 
+  packet->waiting_time = tick_timer_->GetNewStopwatch();
+
   if (buffer_.size() >= max_number_of_packets_) {
     // Buffer is full. Flush it.
     Flush();
@@ -268,13 +272,6 @@ size_t PacketBuffer::NumSamplesInBuffer(DecoderDatabase* decoder_database,
   return num_samples;
 }
 
-void PacketBuffer::IncrementWaitingTimes(int inc) {
-  PacketList::iterator it;
-  for (it = buffer_.begin(); it != buffer_.end(); ++it) {
-    (*it)->waiting_time += inc;
-  }
-}
-
 bool PacketBuffer::DeleteFirstPacket(PacketList* packet_list) {
   if (packet_list->empty()) {
     return false;

webrtc/modules/audio_coding/neteq/packet_buffer.h

@@ -17,8 +17,8 @@
 
 namespace webrtc {
 
-// Forward declaration.
 class DecoderDatabase;
+class TickTimer;
 
 // This is the actual buffer holding the packets before decoding.
 class PacketBuffer {
@@ -34,7 +34,7 @@ class PacketBuffer {
 
   // Constructor creates a buffer which can hold a maximum of
   // |max_number_of_packets| packets.
-  PacketBuffer(size_t max_number_of_packets);
+  PacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer);
 
   // Deletes all packets in the buffer before destroying the buffer.
   virtual ~PacketBuffer();
@@ -116,10 +116,6 @@ class PacketBuffer {
   virtual size_t NumSamplesInBuffer(DecoderDatabase* decoder_database,
                                     size_t last_decoded_length) const;
 
-  // Increase the waiting time counter for every packet in the buffer by |inc|.
-  // The default value for |inc| is 1.
-  virtual void IncrementWaitingTimes(int inc = 1);
-
   virtual void BufferStat(int* num_packets, int* max_num_packets) const;
 
   // Static method that properly deletes the first packet, and its payload
@@ -148,6 +144,7 @@ class PacketBuffer {
  private:
   size_t max_number_of_packets_;
   PacketList buffer_;
+  const TickTimer* tick_timer_;
   RTC_DISALLOW_COPY_AND_ASSIGN(PacketBuffer);
 };
 

webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc

@@ -16,6 +16,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 
 using ::testing::Return;
 using ::testing::_;
@@ -80,13 +81,15 @@ struct PacketsToInsert {
 // Start of test definitions.
 
 TEST(PacketBuffer, CreateAndDestroy) {
-  PacketBuffer* buffer = new PacketBuffer(10);  // 10 packets.
+  TickTimer tick_timer;
+  PacketBuffer* buffer = new PacketBuffer(10, &tick_timer);  // 10 packets.
   EXPECT_TRUE(buffer->Empty());
   delete buffer;
 }
 
 TEST(PacketBuffer, InsertPacket) {
-  PacketBuffer buffer(10);  // 10 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
   PacketGenerator gen(17u, 4711u, 0, 10);
 
   const int payload_len = 100;
@@ -107,7 +110,8 @@ TEST(PacketBuffer, InsertPacket) {
 
 // Test to flush buffer.
 TEST(PacketBuffer, FlushBuffer) {
-  PacketBuffer buffer(10);  // 10 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
   PacketGenerator gen(0, 0, 0, 10);
   const int payload_len = 10;
 
@@ -127,7 +131,8 @@ TEST(PacketBuffer, FlushBuffer) {
 
 // Test to fill the buffer over the limits, and verify that it flushes.
 TEST(PacketBuffer, OverfillBuffer) {
-  PacketBuffer buffer(10);  // 10 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
   PacketGenerator gen(0, 0, 0, 10);
 
   // Insert 10 small packets; should be ok.
@@ -156,7 +161,8 @@ TEST(PacketBuffer, OverfillBuffer) {
 
 // Test inserting a list of packets.
 TEST(PacketBuffer, InsertPacketList) {
-  PacketBuffer buffer(10);  // 10 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
   PacketGenerator gen(0, 0, 0, 10);
   PacketList list;
   const int payload_len = 10;
@@ -192,7 +198,8 @@ TEST(PacketBuffer, InsertPacketList) {
 // Expecting the buffer to flush.
 // TODO(hlundin): Remove this test when legacy operation is no longer needed.
 TEST(PacketBuffer, InsertPacketListChangePayloadType) {
-  PacketBuffer buffer(10);  // 10 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
   PacketGenerator gen(0, 0, 0, 10);
   PacketList list;
   const int payload_len = 10;
@@ -230,7 +237,8 @@ TEST(PacketBuffer, InsertPacketListChangePayloadType) {
 }
 
 TEST(PacketBuffer, ExtractOrderRedundancy) {
-  PacketBuffer buffer(100);  // 100 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(100, &tick_timer);  // 100 packets.
   const int kPackets = 18;
   const int kFrameSize = 10;
   const int kPayloadLength = 10;
@@ -289,7 +297,8 @@ TEST(PacketBuffer, ExtractOrderRedundancy) {
 }
 
 TEST(PacketBuffer, DiscardPackets) {
-  PacketBuffer buffer(100);  // 100 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(100, &tick_timer);  // 100 packets.
   const uint16_t start_seq_no = 17;
   const uint32_t start_ts = 4711;
   const uint32_t ts_increment = 10;
@@ -318,7 +327,8 @@ TEST(PacketBuffer, DiscardPackets) {
 }
 
 TEST(PacketBuffer, Reordering) {
-  PacketBuffer buffer(100);  // 100 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(100, &tick_timer);  // 100 packets.
   const uint16_t start_seq_no = 17;
   const uint32_t start_ts = 4711;
   const uint32_t ts_increment = 10;
@@ -373,8 +383,9 @@ TEST(PacketBuffer, Failures) {
   const uint32_t ts_increment = 10;
   int payload_len = 100;
   PacketGenerator gen(start_seq_no, start_ts, 0, ts_increment);
+  TickTimer tick_timer;
 
-  PacketBuffer* buffer = new PacketBuffer(100);  // 100 packets.
+  PacketBuffer* buffer = new PacketBuffer(100, &tick_timer);  // 100 packets.
   Packet* packet = NULL;
   EXPECT_EQ(PacketBuffer::kInvalidPacket, buffer->InsertPacket(packet));
   packet = gen.NextPacket(payload_len);
@@ -404,7 +415,7 @@ TEST(PacketBuffer, Failures) {
   // Insert packet list of three packets, where the second packet has an invalid
   // payload.  Expect first packet to be inserted, and the remaining two to be
   // discarded.
-  buffer = new PacketBuffer(100);  // 100 packets.
+  buffer = new PacketBuffer(100, &tick_timer);  // 100 packets.
   PacketList list;
   list.push_back(gen.NextPacket(payload_len));  // Valid packet.
   packet = gen.NextPacket(payload_len);

webrtc/modules/audio_coding/neteq/payload_splitter.cc

@@ -12,6 +12,7 @@
 
 #include <assert.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 
@@ -168,8 +169,9 @@ int PayloadSplitter::SplitFec(PacketList* packet_list,
         memcpy(new_packet->payload, packet->payload, packet->payload_length);
         new_packet->payload_length = packet->payload_length;
         new_packet->primary = false;
-        new_packet->waiting_time = packet->waiting_time;
         new_packet->sync_packet = packet->sync_packet;
+        // Waiting time should not be set here.
+        RTC_DCHECK(!packet->waiting_time);
 
         packet_list->insert(it, new_packet);
         break;