ACM: Removed the code for InitialDelayManager

webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc

Index: webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
diff --git a/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc b/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
deleted file mode 100644
index bf2850d0ec8b409ea6f8941680097c52a83fc50e..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
+++ /dev/null
@@ -1,378 +0,0 @@
-/*
- *  Copyright (c) 2013 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 <string.h>
-
-#include <memory>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
-
-namespace webrtc {
-
-namespace acm2 {
-
-namespace {
-
-const uint8_t kAudioPayloadType = 0;
-const uint8_t kCngPayloadType = 1;
-const uint8_t kAvtPayloadType = 2;
-
-const int kSamplingRateHz = 16000;
-const int kInitDelayMs = 200;
-const int kFrameSizeMs = 20;
-const uint32_t kTimestampStep = kFrameSizeMs * kSamplingRateHz / 1000;
-const int kLatePacketThreshold = 5;
-
-void InitRtpInfo(WebRtcRTPHeader* rtp_info) {
-  memset(rtp_info, 0, sizeof(*rtp_info));
-  rtp_info->header.markerBit = false;
-  rtp_info->header.payloadType = kAudioPayloadType;
-  rtp_info->header.sequenceNumber = 1234;
-  rtp_info->header.timestamp = 0xFFFFFFFD;  // Close to wrap around.
-  rtp_info->header.ssrc = 0x87654321;  // Arbitrary.
-  rtp_info->header.numCSRCs = 0;  // Arbitrary.
-  rtp_info->header.paddingLength = 0;
-  rtp_info->header.headerLength = sizeof(RTPHeader);
-  rtp_info->header.payload_type_frequency = kSamplingRateHz;
-  rtp_info->header.extension.absoluteSendTime = 0;
-  rtp_info->header.extension.transmissionTimeOffset = 0;
-  rtp_info->frameType = kAudioFrameSpeech;
-}
-
-void ForwardRtpHeader(int n,
-                      WebRtcRTPHeader* rtp_info,
-                      uint32_t* rtp_receive_timestamp) {
-  rtp_info->header.sequenceNumber += n;
-  rtp_info->header.timestamp += n * kTimestampStep;
-  *rtp_receive_timestamp += n * kTimestampStep;
-}
-
-void NextRtpHeader(WebRtcRTPHeader* rtp_info,
-                   uint32_t* rtp_receive_timestamp) {
-  ForwardRtpHeader(1, rtp_info, rtp_receive_timestamp);
-}
-
-}  // namespace
-
-class InitialDelayManagerTest : public ::testing::Test {
- protected:
-  InitialDelayManagerTest()
-      : manager_(new InitialDelayManager(kInitDelayMs, kLatePacketThreshold)),
-        rtp_receive_timestamp_(1111) { }  // Arbitrary starting point.
-
-  virtual void SetUp() {
-    ASSERT_TRUE(manager_.get() != NULL);
-    InitRtpInfo(&rtp_info_);
-  }
-
-  void GetNextRtpHeader(WebRtcRTPHeader* rtp_info,
-                        uint32_t* rtp_receive_timestamp) const {
-    memcpy(rtp_info, &rtp_info_, sizeof(*rtp_info));
-    *rtp_receive_timestamp = rtp_receive_timestamp_;
-    NextRtpHeader(rtp_info, rtp_receive_timestamp);
-  }
-
-  std::unique_ptr<InitialDelayManager> manager_;
-  WebRtcRTPHeader rtp_info_;
-  uint32_t rtp_receive_timestamp_;
-};
-
-TEST_F(InitialDelayManagerTest, Init) {
-  EXPECT_TRUE(manager_->buffering());
-  EXPECT_FALSE(manager_->PacketBuffered());
-  manager_->DisableBuffering();
-  EXPECT_FALSE(manager_->buffering());
-  InitialDelayManager::SyncStream sync_stream;
-
-  // Call before any packet inserted.
-  manager_->LatePackets(0x6789ABCD, &sync_stream);  // Arbitrary but large
-                                                    // receive timestamp.
-  EXPECT_EQ(0, sync_stream.num_sync_packets);
-
-  // Insert non-audio packets, a CNG and DTMF.
-  rtp_info_.header.payloadType = kCngPayloadType;
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kCngPacket, false,
-                                     kSamplingRateHz, &sync_stream);
-  EXPECT_EQ(0, sync_stream.num_sync_packets);
-  ForwardRtpHeader(5, &rtp_info_, &rtp_receive_timestamp_);
-  rtp_info_.header.payloadType = kAvtPayloadType;
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAvtPacket, false,
-                                     kSamplingRateHz, &sync_stream);
-  // Gap in sequence numbers but no audio received, sync-stream should be empty.
-  EXPECT_EQ(0, sync_stream.num_sync_packets);
-  manager_->LatePackets(0x45678987, &sync_stream);  // Large arbitrary receive
-                                                    // timestamp.
-  // |manager_| has no estimate of timestamp-step and has not received any
-  // audio packet.
-  EXPECT_EQ(0, sync_stream.num_sync_packets);
-
-
-  NextRtpHeader(&rtp_info_, &rtp_receive_timestamp_);
-  rtp_info_.header.payloadType = kAudioPayloadType;
-  // First packet.
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAudioPacket, true,
-                                     kSamplingRateHz, &sync_stream);
-  EXPECT_EQ(0, sync_stream.num_sync_packets);
-
-  // Call LatePAcket() after only one packet inserted.
-  manager_->LatePackets(0x6789ABCD, &sync_stream);  // Arbitrary but large
-                                                    // receive timestamp.
-  EXPECT_EQ(0, sync_stream.num_sync_packets);
-
-  // Gap in timestamp, but this packet is also flagged as "new," therefore,
-  // expecting empty sync-stream.
-  ForwardRtpHeader(5, &rtp_info_, &rtp_receive_timestamp_);
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAudioPacket, true,
-                                     kSamplingRateHz, &sync_stream);
-}
-
-TEST_F(InitialDelayManagerTest, MissingPacket) {
-  InitialDelayManager::SyncStream sync_stream;
-  // First packet.
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAudioPacket, true,
-                                     kSamplingRateHz, &sync_stream);
-  ASSERT_EQ(0, sync_stream.num_sync_packets);
-
-  // Second packet.
-  NextRtpHeader(&rtp_info_, &rtp_receive_timestamp_);
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAudioPacket, false,
-                                     kSamplingRateHz, &sync_stream);
-  ASSERT_EQ(0, sync_stream.num_sync_packets);
-
-  // Third packet, missing packets start from here.
-  NextRtpHeader(&rtp_info_, &rtp_receive_timestamp_);
-
-  // First sync-packet in sync-stream is one after the above packet.
-  WebRtcRTPHeader expected_rtp_info;
-  uint32_t expected_receive_timestamp;
-  GetNextRtpHeader(&expected_rtp_info, &expected_receive_timestamp);
-
-  const int kNumMissingPackets = 10;
-  ForwardRtpHeader(kNumMissingPackets, &rtp_info_, &rtp_receive_timestamp_);
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAudioPacket, false,
-                                     kSamplingRateHz, &sync_stream);
-  EXPECT_EQ(kNumMissingPackets - 2, sync_stream.num_sync_packets);
-  EXPECT_EQ(0, memcmp(&expected_rtp_info, &sync_stream.rtp_info,
-                      sizeof(expected_rtp_info)));
-  EXPECT_EQ(kTimestampStep, sync_stream.timestamp_step);
-  EXPECT_EQ(expected_receive_timestamp, sync_stream.receive_timestamp);
-}
-
-// There hasn't been any consecutive packets to estimate timestamp-step.
-TEST_F(InitialDelayManagerTest, MissingPacketEstimateTimestamp) {
-  InitialDelayManager::SyncStream sync_stream;
-  // First packet.
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAudioPacket, true,
-                                     kSamplingRateHz, &sync_stream);
-  ASSERT_EQ(0, sync_stream.num_sync_packets);
-
-  // Second packet, missing packets start here.
-  NextRtpHeader(&rtp_info_, &rtp_receive_timestamp_);
-
-  // First sync-packet in sync-stream is one after the above.
-  WebRtcRTPHeader expected_rtp_info;
-  uint32_t expected_receive_timestamp;
-  GetNextRtpHeader(&expected_rtp_info, &expected_receive_timestamp);
-
-  const int kNumMissingPackets = 10;
-  ForwardRtpHeader(kNumMissingPackets, &rtp_info_, &rtp_receive_timestamp_);
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAudioPacket, false,
-                                     kSamplingRateHz, &sync_stream);
-  EXPECT_EQ(kNumMissingPackets - 2, sync_stream.num_sync_packets);
-  EXPECT_EQ(0, memcmp(&expected_rtp_info, &sync_stream.rtp_info,
-                      sizeof(expected_rtp_info)));
-}
-
-TEST_F(InitialDelayManagerTest, MissingPacketWithCng) {
-  InitialDelayManager::SyncStream sync_stream;
-
-  // First packet.
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAudioPacket, true,
-                                     kSamplingRateHz, &sync_stream);
-  ASSERT_EQ(0, sync_stream.num_sync_packets);
-
-  // Second packet as CNG.
-  NextRtpHeader(&rtp_info_, &rtp_receive_timestamp_);
-  rtp_info_.header.payloadType = kCngPayloadType;
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kCngPacket, false,
-                                     kSamplingRateHz, &sync_stream);
-  ASSERT_EQ(0, sync_stream.num_sync_packets);
-
-  // Audio packet after CNG. Missing packets start from this packet.
-  rtp_info_.header.payloadType = kAudioPayloadType;
-  NextRtpHeader(&rtp_info_, &rtp_receive_timestamp_);
-
-  // Timestamps are increased higher than regular packet.
-  const uint32_t kCngTimestampStep = 5 * kTimestampStep;
-  rtp_info_.header.timestamp += kCngTimestampStep;
-  rtp_receive_timestamp_ += kCngTimestampStep;
-
-  // First sync-packet in sync-stream is the one after the above packet.
-  WebRtcRTPHeader expected_rtp_info;
-  uint32_t expected_receive_timestamp;
-  GetNextRtpHeader(&expected_rtp_info, &expected_receive_timestamp);
-
-  const int kNumMissingPackets = 10;
-  ForwardRtpHeader(kNumMissingPackets, &rtp_info_, &rtp_receive_timestamp_);
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAudioPacket, false,
-                                     kSamplingRateHz, &sync_stream);
-  EXPECT_EQ(kNumMissingPackets - 2, sync_stream.num_sync_packets);
-  EXPECT_EQ(0, memcmp(&expected_rtp_info, &sync_stream.rtp_info,
-                      sizeof(expected_rtp_info)));
-  EXPECT_EQ(kTimestampStep, sync_stream.timestamp_step);
-  EXPECT_EQ(expected_receive_timestamp, sync_stream.receive_timestamp);
-}
-
-TEST_F(InitialDelayManagerTest, LatePacket) {
-  InitialDelayManager::SyncStream sync_stream;
-  // First packet.
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAudioPacket, true,
-                                     kSamplingRateHz, &sync_stream);
-  ASSERT_EQ(0, sync_stream.num_sync_packets);
-
-  // Second packet.
-  NextRtpHeader(&rtp_info_, &rtp_receive_timestamp_);
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAudioPacket, false,
-                                     kSamplingRateHz, &sync_stream);
-  ASSERT_EQ(0, sync_stream.num_sync_packets);
-
-  // Timestamp increment for 10ms;
-  const uint32_t kTimestampStep10Ms = kSamplingRateHz / 100;
-
-  // 10 ms after the second packet is inserted.
-  uint32_t timestamp_now = rtp_receive_timestamp_ + kTimestampStep10Ms;
-
-  // Third packet, late packets start from this packet.
-  NextRtpHeader(&rtp_info_, &rtp_receive_timestamp_);
-
-  // First sync-packet in sync-stream, which is one after the above packet.
-  WebRtcRTPHeader expected_rtp_info;
-  uint32_t expected_receive_timestamp;
-  GetNextRtpHeader(&expected_rtp_info, &expected_receive_timestamp);
-
-  const int kLatePacketThreshold = 5;
-
-  int expected_num_late_packets = kLatePacketThreshold - 1;
-  for (int k = 0; k < 2; ++k) {
-    for (int n = 1; n < kLatePacketThreshold * kFrameSizeMs / 10; ++n) {
-      manager_->LatePackets(timestamp_now, &sync_stream);
-      EXPECT_EQ(0, sync_stream.num_sync_packets) <<
-          "try " << k << " loop number " << n;
-      timestamp_now += kTimestampStep10Ms;
-    }
-    manager_->LatePackets(timestamp_now, &sync_stream);
-
-    EXPECT_EQ(expected_num_late_packets, sync_stream.num_sync_packets) <<
-        "try " << k;
-    EXPECT_EQ(kTimestampStep, sync_stream.timestamp_step) <<
-        "try " << k;
-    EXPECT_EQ(expected_receive_timestamp, sync_stream.receive_timestamp) <<
-        "try " << k;
-    EXPECT_EQ(0, memcmp(&expected_rtp_info, &sync_stream.rtp_info,
-                        sizeof(expected_rtp_info)));
-
-    timestamp_now += kTimestampStep10Ms;
-
-    // |manger_| assumes the |sync_stream| obtained by LatePacket() is fully
-    // injected. The last injected packet is sync-packet, therefore, there will
-    // not be any gap between sync stream of this and the next iteration.
-    ForwardRtpHeader(sync_stream.num_sync_packets, &expected_rtp_info,
-        &expected_receive_timestamp);
-    expected_num_late_packets = kLatePacketThreshold;
-  }
-
-  // Test "no-gap" for missing packet after late packet.
-  // |expected_rtp_info| is the expected sync-packet if any packet is missing.
-  memcpy(&rtp_info_, &expected_rtp_info, sizeof(rtp_info_));
-  rtp_receive_timestamp_ = expected_receive_timestamp;
-
-  int kNumMissingPackets = 3;  // Arbitrary.
-  ForwardRtpHeader(kNumMissingPackets, &rtp_info_, &rtp_receive_timestamp_);
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAudioPacket, false,
-                                     kSamplingRateHz, &sync_stream);
-
-  // Note that there is one packet gap between the last sync-packet and the
-  // latest inserted packet.
-  EXPECT_EQ(kNumMissingPackets - 1, sync_stream.num_sync_packets);
-  EXPECT_EQ(kTimestampStep, sync_stream.timestamp_step);
-  EXPECT_EQ(expected_receive_timestamp, sync_stream.receive_timestamp);
-  EXPECT_EQ(0, memcmp(&expected_rtp_info, &sync_stream.rtp_info,
-                      sizeof(expected_rtp_info)));
-}
-
-TEST_F(InitialDelayManagerTest, NoLatePacketAfterCng) {
-  InitialDelayManager::SyncStream sync_stream;
-
-  // First packet.
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAudioPacket, true,
-                                     kSamplingRateHz, &sync_stream);
-  ASSERT_EQ(0, sync_stream.num_sync_packets);
-
-  // Second packet as CNG.
-  NextRtpHeader(&rtp_info_, &rtp_receive_timestamp_);
-  rtp_info_.header.payloadType = kCngPayloadType;
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kCngPacket, false,
-                                     kSamplingRateHz, &sync_stream);
-  ASSERT_EQ(0, sync_stream.num_sync_packets);
-
-  // Forward the time more then |kLatePacketThreshold| packets.
-  uint32_t timestamp_now = rtp_receive_timestamp_ + kTimestampStep * (3 +
-      kLatePacketThreshold);
-
-  manager_->LatePackets(timestamp_now, &sync_stream);
-  EXPECT_EQ(0, sync_stream.num_sync_packets);
-}
-
-TEST_F(InitialDelayManagerTest, BufferingAudio) {
-  InitialDelayManager::SyncStream sync_stream;
-
-  // Very first packet is not counted in calculation of buffered audio.
-  for (int n = 0; n < kInitDelayMs / kFrameSizeMs; ++n) {
-    manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                       InitialDelayManager::kAudioPacket,
-                                       n == 0, kSamplingRateHz, &sync_stream);
-    EXPECT_EQ(0, sync_stream.num_sync_packets);
-    EXPECT_TRUE(manager_->buffering());
-    const uint32_t expected_playout_timestamp = rtp_info_.header.timestamp -
-        kInitDelayMs * kSamplingRateHz / 1000;
-    uint32_t actual_playout_timestamp = 0;
-    EXPECT_TRUE(manager_->GetPlayoutTimestamp(&actual_playout_timestamp));
-    EXPECT_EQ(expected_playout_timestamp, actual_playout_timestamp);
-    NextRtpHeader(&rtp_info_, &rtp_receive_timestamp_);
-  }
-
-  manager_->UpdateLastReceivedPacket(rtp_info_, rtp_receive_timestamp_,
-                                     InitialDelayManager::kAudioPacket,
-                                     false, kSamplingRateHz, &sync_stream);
-  EXPECT_EQ(0, sync_stream.num_sync_packets);
-  EXPECT_FALSE(manager_->buffering());
-}
-
-}  // namespace acm2
-
-}  // namespace webrtc