Removing AudioCoding class, a.k.a the new ACM API

webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc

Index: webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
deleted file mode 100644
index c12baf3f4047acbede681e84f87b40869be0f1ee..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
+++ /dev/null
@@ -1,1004 +0,0 @@
-/*
- *  Copyright (c) 2014 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 <vector>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/md5digest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_send_test.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_checksum.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-
-namespace webrtc {
-
-const int kSampleRateHz = 16000;
-const int kNumSamples10ms = kSampleRateHz / 100;
-const int kFrameSizeMs = 10;  // Multiple of 10.
-const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms;
-const size_t kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
-const uint8_t kPayloadType = 111;
-
-class RtpUtility {
- public:
-  RtpUtility(int samples_per_packet, uint8_t payload_type)
-      : samples_per_packet_(samples_per_packet), payload_type_(payload_type) {}
-
-  virtual ~RtpUtility() {}
-
-  void Populate(WebRtcRTPHeader* rtp_header) {
-    rtp_header->header.sequenceNumber = 0xABCD;
-    rtp_header->header.timestamp = 0xABCDEF01;
-    rtp_header->header.payloadType = payload_type_;
-    rtp_header->header.markerBit = false;
-    rtp_header->header.ssrc = 0x1234;
-    rtp_header->header.numCSRCs = 0;
-    rtp_header->frameType = kAudioFrameSpeech;
-
-    rtp_header->header.payload_type_frequency = kSampleRateHz;
-    rtp_header->type.Audio.channel = 1;
-    rtp_header->type.Audio.isCNG = false;
-  }
-
-  void Forward(WebRtcRTPHeader* rtp_header) {
-    ++rtp_header->header.sequenceNumber;
-    rtp_header->header.timestamp += samples_per_packet_;
-  }
-
- private:
-  int samples_per_packet_;
-  uint8_t payload_type_;
-};
-
-class PacketizationCallbackStub : public AudioPacketizationCallback {
- public:
-  PacketizationCallbackStub()
-      : num_calls_(0),
-        crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {}
-
-  int32_t SendData(FrameType frame_type,
-                   uint8_t payload_type,
-                   uint32_t timestamp,
-                   const uint8_t* payload_data,
-                   size_t payload_len_bytes,
-                   const RTPFragmentationHeader* fragmentation) override {
-    CriticalSectionScoped lock(crit_sect_.get());
-    ++num_calls_;
-    last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
-    return 0;
-  }
-
-  int num_calls() const {
-    CriticalSectionScoped lock(crit_sect_.get());
-    return num_calls_;
-  }
-
-  int last_payload_len_bytes() const {
-    CriticalSectionScoped lock(crit_sect_.get());
-    return last_payload_vec_.size();
-  }
-
-  void SwapBuffers(std::vector<uint8_t>* payload) {
-    CriticalSectionScoped lock(crit_sect_.get());
-    last_payload_vec_.swap(*payload);
-  }
-
- private:
-  int num_calls_ GUARDED_BY(crit_sect_);
-  std::vector<uint8_t> last_payload_vec_ GUARDED_BY(crit_sect_);
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
-};
-
-class AudioCodingModuleTest : public ::testing::Test {
- protected:
-  AudioCodingModuleTest()
-      : rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)) {
-    config_.transport = &packet_cb_;
-  }
-
-  ~AudioCodingModuleTest() {}
-
-  void TearDown() override {}
-
-  void SetUp() override {
-    rtp_utility_->Populate(&rtp_header_);
-
-    input_frame_.sample_rate_hz_ = kSampleRateHz;
-    input_frame_.num_channels_ = 1;
-    input_frame_.samples_per_channel_ = kSampleRateHz * 10 / 1000;  // 10 ms.
-    static_assert(kSampleRateHz * 10 / 1000 <= AudioFrame::kMaxDataSizeSamples,
-                  "audio frame too small");
-    memset(input_frame_.data_,
-           0,
-           input_frame_.samples_per_channel_ * sizeof(input_frame_.data_[0]));
-  }
-
-  void CreateAcm() {
-    acm_.reset(AudioCoding::Create(config_));
-    ASSERT_TRUE(acm_.get() != NULL);
-    RegisterCodec();
-  }
-
-  virtual void RegisterCodec() {
-    // Register L16 codec in ACM.
-    int codec_type = acm2::ACMCodecDB::kNone;
-    switch (kSampleRateHz) {
-      case 8000:
-        codec_type = acm2::ACMCodecDB::kPCM16B;
-        break;
-      case 16000:
-        codec_type = acm2::ACMCodecDB::kPCM16Bwb;
-        break;
-      case 32000:
-        codec_type = acm2::ACMCodecDB::kPCM16Bswb32kHz;
-        break;
-      default:
-        FATAL() << "Sample rate not supported in this test.";
-    }
-    ASSERT_TRUE(acm_->RegisterSendCodec(codec_type, kPayloadType));
-    ASSERT_TRUE(acm_->RegisterReceiveCodec(codec_type, kPayloadType));
-  }
-
-  virtual void InsertPacketAndPullAudio() {
-    InsertPacket();
-    PullAudio();
-  }
-
-  virtual void InsertPacket() {
-    const uint8_t kPayload[kPayloadSizeBytes] = {0};
-    ASSERT_TRUE(acm_->InsertPacket(kPayload, kPayloadSizeBytes, rtp_header_));
-    rtp_utility_->Forward(&rtp_header_);
-  }
-
-  virtual void PullAudio() {
-    AudioFrame audio_frame;
-    ASSERT_TRUE(acm_->Get10MsAudio(&audio_frame));
-  }
-
-  virtual void InsertAudio() {
-    int encoded_bytes = acm_->Add10MsAudio(input_frame_);
-    ASSERT_GE(encoded_bytes, 0);
-    input_frame_.timestamp_ += kNumSamples10ms;
-  }
-
-  AudioCoding::Config config_;
-  rtc::scoped_ptr<RtpUtility> rtp_utility_;
-  rtc::scoped_ptr<AudioCoding> acm_;
-  PacketizationCallbackStub packet_cb_;
-  WebRtcRTPHeader rtp_header_;
-  AudioFrame input_frame_;
-};
-
-// Check if the statistics are initialized correctly. Before any call to ACM
-// all fields have to be zero.
-TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(InitializedToZero)) {
-  CreateAcm();
-  AudioDecodingCallStats stats;
-  acm_->GetDecodingCallStatistics(&stats);
-  EXPECT_EQ(0, stats.calls_to_neteq);
-  EXPECT_EQ(0, stats.calls_to_silence_generator);
-  EXPECT_EQ(0, stats.decoded_normal);
-  EXPECT_EQ(0, stats.decoded_cng);
-  EXPECT_EQ(0, stats.decoded_plc);
-  EXPECT_EQ(0, stats.decoded_plc_cng);
-}
-
-// Apply an initial playout delay. Calls to AudioCodingModule::PlayoutData10ms()
-// should result in generating silence, check the associated field.
-TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(SilenceGeneratorCalled)) {
-  const int kInitialDelay = 100;
-  config_.initial_playout_delay_ms = kInitialDelay;
-  CreateAcm();
-  AudioDecodingCallStats stats;
-
-  int num_calls = 0;
-  for (int time_ms = 0; time_ms < kInitialDelay;
-       time_ms += kFrameSizeMs, ++num_calls) {
-    InsertPacketAndPullAudio();
-  }
-  acm_->GetDecodingCallStatistics(&stats);
-  EXPECT_EQ(0, stats.calls_to_neteq);
-  EXPECT_EQ(num_calls, stats.calls_to_silence_generator);
-  EXPECT_EQ(0, stats.decoded_normal);
-  EXPECT_EQ(0, stats.decoded_cng);
-  EXPECT_EQ(0, stats.decoded_plc);
-  EXPECT_EQ(0, stats.decoded_plc_cng);
-}
-
-// Insert some packets and pull audio. Check statistics are valid. Then,
-// simulate packet loss and check if PLC and PLC-to-CNG statistics are
-// correctly updated.
-TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(NetEqCalls)) {
-  CreateAcm();
-  AudioDecodingCallStats stats;
-  const int kNumNormalCalls = 10;
-
-  for (int num_calls = 0; num_calls < kNumNormalCalls; ++num_calls) {
-    InsertPacketAndPullAudio();
-  }
-  acm_->GetDecodingCallStatistics(&stats);
-  EXPECT_EQ(kNumNormalCalls, stats.calls_to_neteq);
-  EXPECT_EQ(0, stats.calls_to_silence_generator);
-  EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
-  EXPECT_EQ(0, stats.decoded_cng);
-  EXPECT_EQ(0, stats.decoded_plc);
-  EXPECT_EQ(0, stats.decoded_plc_cng);
-
-  const int kNumPlc = 3;
-  const int kNumPlcCng = 5;
-
-  // Simulate packet-loss. NetEq first performs PLC then PLC fades to CNG.
-  for (int n = 0; n < kNumPlc + kNumPlcCng; ++n) {
-    PullAudio();
-  }
-  acm_->GetDecodingCallStatistics(&stats);
-  EXPECT_EQ(kNumNormalCalls + kNumPlc + kNumPlcCng, stats.calls_to_neteq);
-  EXPECT_EQ(0, stats.calls_to_silence_generator);
-  EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
-  EXPECT_EQ(0, stats.decoded_cng);
-  EXPECT_EQ(kNumPlc, stats.decoded_plc);
-  EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng);
-}
-
-TEST_F(AudioCodingModuleTest, VerifyOutputFrame) {
-  CreateAcm();
-  AudioFrame audio_frame;
-  const int kSampleRateHz = 32000;
-  EXPECT_TRUE(acm_->Get10MsAudio(&audio_frame));
-  EXPECT_EQ(0u, audio_frame.timestamp_);
-  EXPECT_GT(audio_frame.num_channels_, 0);
-  EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
-            audio_frame.samples_per_channel_);
-  EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
-}
-
-// A multi-threaded test for ACM. This base class is using the PCM16b 16 kHz
-// codec, while the derive class AcmIsacMtTest is using iSAC.
-class AudioCodingModuleMtTest : public AudioCodingModuleTest {
- protected:
-  static const int kNumPackets = 500;
-  static const int kNumPullCalls = 500;
-
-  AudioCodingModuleMtTest()
-      : AudioCodingModuleTest(),
-        send_thread_(ThreadWrapper::CreateThread(CbSendThread, this, "send")),
-        insert_packet_thread_(ThreadWrapper::CreateThread(
-            CbInsertPacketThread, this, "insert_packet")),
-        pull_audio_thread_(ThreadWrapper::CreateThread(
-            CbPullAudioThread, this, "pull_audio")),
-        test_complete_(EventWrapper::Create()),
-        send_count_(0),
-        insert_packet_count_(0),
-        pull_audio_count_(0),
-        crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-        next_insert_packet_time_ms_(0),
-        fake_clock_(new SimulatedClock(0)) {
-    config_.clock = fake_clock_.get();
-  }
-
-  void SetUp() override {
-    AudioCodingModuleTest::SetUp();
-    CreateAcm();
-    StartThreads();
-  }
-
-  void StartThreads() {
-    ASSERT_TRUE(send_thread_->Start());
-    send_thread_->SetPriority(kRealtimePriority);
-    ASSERT_TRUE(insert_packet_thread_->Start());
-    insert_packet_thread_->SetPriority(kRealtimePriority);
-    ASSERT_TRUE(pull_audio_thread_->Start());
-    pull_audio_thread_->SetPriority(kRealtimePriority);
-  }
-
-  void TearDown() override {
-    AudioCodingModuleTest::TearDown();
-    pull_audio_thread_->Stop();
-    send_thread_->Stop();
-    insert_packet_thread_->Stop();
-  }
-
-  EventTypeWrapper RunTest() {
-    return test_complete_->Wait(10 * 60 * 1000);  // 10 minutes' timeout.
-  }
-
-  virtual bool TestDone() {
-    if (packet_cb_.num_calls() > kNumPackets) {
-      CriticalSectionScoped lock(crit_sect_.get());
-      if (pull_audio_count_ > kNumPullCalls) {
-        // Both conditions for completion are met. End the test.
-        return true;
-      }
-    }
-    return false;
-  }
-
-  static bool CbSendThread(void* context) {
-    return reinterpret_cast<AudioCodingModuleMtTest*>(context)->CbSendImpl();
-  }
-
-  // The send thread doesn't have to care about the current simulated time,
-  // since only the AcmReceiver is using the clock.
-  bool CbSendImpl() {
-    SleepMs(1);
-    if (HasFatalFailure()) {
-      // End the test early if a fatal failure (ASSERT_*) has occurred.
-      test_complete_->Set();
-    }
-    ++send_count_;
-    InsertAudio();
-    if (TestDone()) {
-      test_complete_->Set();
-    }
-    return true;
-  }
-
-  static bool CbInsertPacketThread(void* context) {
-    return reinterpret_cast<AudioCodingModuleMtTest*>(context)
-        ->CbInsertPacketImpl();
-  }
-
-  bool CbInsertPacketImpl() {
-    SleepMs(1);
-    {
-      CriticalSectionScoped lock(crit_sect_.get());
-      if (fake_clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
-        return true;
-      }
-      next_insert_packet_time_ms_ += 10;
-    }
-    // Now we're not holding the crit sect when calling ACM.
-    ++insert_packet_count_;
-    InsertPacket();
-    return true;
-  }
-
-  static bool CbPullAudioThread(void* context) {
-    return reinterpret_cast<AudioCodingModuleMtTest*>(context)
-        ->CbPullAudioImpl();
-  }
-
-  bool CbPullAudioImpl() {
-    SleepMs(1);
-    {
-      CriticalSectionScoped lock(crit_sect_.get());
-      // Don't let the insert thread fall behind.
-      if (next_insert_packet_time_ms_ < fake_clock_->TimeInMilliseconds()) {
-        return true;
-      }
-      ++pull_audio_count_;
-    }
-    // Now we're not holding the crit sect when calling ACM.
-    PullAudio();
-    fake_clock_->AdvanceTimeMilliseconds(10);
-    return true;
-  }
-
-  rtc::scoped_ptr<ThreadWrapper> send_thread_;
-  rtc::scoped_ptr<ThreadWrapper> insert_packet_thread_;
-  rtc::scoped_ptr<ThreadWrapper> pull_audio_thread_;
-  const rtc::scoped_ptr<EventWrapper> test_complete_;
-  int send_count_;
-  int insert_packet_count_;
-  int pull_audio_count_ GUARDED_BY(crit_sect_);
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
-  int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
-  rtc::scoped_ptr<SimulatedClock> fake_clock_;
-};
-
-TEST_F(AudioCodingModuleMtTest, DoTest) {
-  EXPECT_EQ(kEventSignaled, RunTest());
-}
-
-// This is a multi-threaded ACM test using iSAC. The test encodes audio
-// from a PCM file. The most recent encoded frame is used as input to the
-// receiving part. Depending on timing, it may happen that the same RTP packet
-// is inserted into the receiver multiple times, but this is a valid use-case,
-// and simplifies the test code a lot.
-class AcmIsacMtTest : public AudioCodingModuleMtTest {
- protected:
-  static const int kNumPackets = 500;
-  static const int kNumPullCalls = 500;
-
-  AcmIsacMtTest()
-      : AudioCodingModuleMtTest(),
-        last_packet_number_(0) {}
-
-  ~AcmIsacMtTest() {}
-
-  void SetUp() override {
-    AudioCodingModuleTest::SetUp();
-    CreateAcm();
-
-    // Set up input audio source to read from specified file, loop after 5
-    // seconds, and deliver blocks of 10 ms.
-    const std::string input_file_name =
-        webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm");
-    audio_loop_.Init(input_file_name, 5 * kSampleRateHz, kNumSamples10ms);
-
-    // Generate one packet to have something to insert.
-    int loop_counter = 0;
-    while (packet_cb_.last_payload_len_bytes() == 0) {
-      InsertAudio();
-      ASSERT_LT(loop_counter++, 10);
-    }
-    // Set |last_packet_number_| to one less that |num_calls| so that the packet
-    // will be fetched in the next InsertPacket() call.
-    last_packet_number_ = packet_cb_.num_calls() - 1;
-
-    StartThreads();
-  }
-
-  void RegisterCodec() override {
-    static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
-
-    // Register iSAC codec in ACM, effectively unregistering the PCM16B codec
-    // registered in AudioCodingModuleTest::SetUp();
-    ASSERT_TRUE(acm_->RegisterSendCodec(acm2::ACMCodecDB::kISAC, kPayloadType));
-    ASSERT_TRUE(
-        acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, kPayloadType));
-  }
-
-  void InsertPacket() override {
-    int num_calls = packet_cb_.num_calls();  // Store locally for thread safety.
-    if (num_calls > last_packet_number_) {
-      // Get the new payload out from the callback handler.
-      // Note that since we swap buffers here instead of directly inserting
-      // a pointer to the data in |packet_cb_|, we avoid locking the callback
-      // for the duration of the IncomingPacket() call.
-      packet_cb_.SwapBuffers(&last_payload_vec_);
-      ASSERT_GT(last_payload_vec_.size(), 0u);
-      rtp_utility_->Forward(&rtp_header_);
-      last_packet_number_ = num_calls;
-    }
-    ASSERT_GT(last_payload_vec_.size(), 0u);
-    ASSERT_TRUE(acm_->InsertPacket(
-        &last_payload_vec_[0], last_payload_vec_.size(), rtp_header_));
-  }
-
-  void InsertAudio() override {
-    memcpy(input_frame_.data_, audio_loop_.GetNextBlock(), kNumSamples10ms);
-    AudioCodingModuleTest::InsertAudio();
-  }
-
-  // This method is the same as AudioCodingModuleMtTest::TestDone(), but here
-  // it is using the constants defined in this class (i.e., shorter test run).
-  bool TestDone() override {
-    if (packet_cb_.num_calls() > kNumPackets) {
-      CriticalSectionScoped lock(crit_sect_.get());
-      if (pull_audio_count_ > kNumPullCalls) {
-        // Both conditions for completion are met. End the test.
-        return true;
-      }
-    }
-    return false;
-  }
-
-  int last_packet_number_;
-  std::vector<uint8_t> last_payload_vec_;
-  test::AudioLoop audio_loop_;
-};
-
-#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-#define IF_ISAC(x) x
-#else
-#define IF_ISAC(x) DISABLED_##x
-#endif
-
-TEST_F(AcmIsacMtTest, IF_ISAC(DoTest)) {
-  EXPECT_EQ(kEventSignaled, RunTest());
-}
-
-// Disabling all of these tests on iOS for now.
-// See https://code.google.com/p/webrtc/issues/detail?id=4768 for details.
-#if !defined(WEBRTC_IOS)
-
-class AcmReceiverBitExactness : public ::testing::Test {
- public:
-  static std::string PlatformChecksum(std::string win64,
-                                      std::string android,
-                                      std::string others) {
-#if defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS)
-    return win64;
-#elif defined(WEBRTC_ANDROID)
-    return android;
-#else
-    return others;
-#endif
-  }
-
- protected:
-  void Run(int output_freq_hz, const std::string& checksum_ref) {
-    const std::string input_file_name =
-        webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
-    rtc::scoped_ptr<test::RtpFileSource> packet_source(
-        test::RtpFileSource::Create(input_file_name));
-#ifdef WEBRTC_ANDROID
-    // Filter out iLBC and iSAC-swb since they are not supported on Android.
-    packet_source->FilterOutPayloadType(102);  // iLBC.
-    packet_source->FilterOutPayloadType(104);  // iSAC-swb.
-#endif
-
-    test::AudioChecksum checksum;
-    const std::string output_file_name =
-        webrtc::test::OutputPath() +
-        ::testing::UnitTest::GetInstance()
-            ->current_test_info()
-            ->test_case_name() +
-        "_" + ::testing::UnitTest::GetInstance()->current_test_info()->name() +
-        "_output.pcm";
-    test::OutputAudioFile output_file(output_file_name);
-    test::AudioSinkFork output(&checksum, &output_file);
-
-    test::AcmReceiveTest test(packet_source.get(), &output, output_freq_hz,
-                              test::AcmReceiveTest::kArbitraryChannels);
-    ASSERT_NO_FATAL_FAILURE(test.RegisterNetEqTestCodecs());
-    test.Run();
-
-    std::string checksum_string = checksum.Finish();
-    EXPECT_EQ(checksum_ref, checksum_string);
-  }
-};
-
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISAC)) && \
-    defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
-#define IF_ALL_CODECS(x) x
-#else
-#define IF_ALL_CODECS(x) DISABLED_##x
-#endif
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_8kHzOutput DISABLED_8kHzOutput
-#else
-#define MAYBE_8kHzOutput 8kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactness, IF_ALL_CODECS(MAYBE_8kHzOutput)) {
-  Run(8000,
-      PlatformChecksum("dcee98c623b147ebe1b40dd30efa896e",
-                       "adc92e173f908f93b96ba5844209815a",
-                       "908002dc01fc4eb1d2be24eb1d3f354b"));
-}
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_16kHzOutput DISABLED_16kHzOutput
-#else
-#define MAYBE_16kHzOutput 16kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactness, IF_ALL_CODECS(MAYBE_16kHzOutput)) {
-  Run(16000,
-      PlatformChecksum("f790e7a8cce4e2c8b7bb5e0e4c5dac0d",
-                       "8cffa6abcb3e18e33b9d857666dff66a",
-                       "a909560b5ca49fa472b17b7b277195e9"));
-}
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_32kHzOutput DISABLED_32kHzOutput
-#else
-#define MAYBE_32kHzOutput 32kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactness, IF_ALL_CODECS(MAYBE_32kHzOutput)) {
-  Run(32000,
-      PlatformChecksum("306e0d990ee6e92de3fbecc0123ece37",
-                       "3e126fe894720c3f85edadcc91964ba5",
-                       "441aab4b347fb3db4e9244337aca8d8e"));
-}
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_48kHzOutput DISABLED_48kHzOutput
-#else
-#define MAYBE_48kHzOutput 48kHzOutput
-#endif
-TEST_F(AcmReceiverBitExactness, IF_ALL_CODECS(MAYBE_48kHzOutput)) {
-  Run(48000,
-      PlatformChecksum("aa7c232f63a67b2a72703593bdd172e0",
-                       "0155665e93067c4e89256b944dd11999",
-                       "4ee2730fa1daae755e8a8fd3abd779ec"));
-}
-
-// This test verifies bit exactness for the send-side of ACM. The test setup is
-// a chain of three different test classes:
-//
-// test::AcmSendTest -> AcmSenderBitExactness -> test::AcmReceiveTest
-//
-// The receiver side is driving the test by requesting new packets from
-// AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
-// packet from test::AcmSendTest::NextPacket, which inserts audio from the
-// input file until one packet is produced. (The input file loops indefinitely.)
-// Before passing the packet to the receiver, this test class verifies the
-// packet header and updates a payload checksum with the new payload. The
-// decoded output from the receiver is also verified with a (separate) checksum.
-class AcmSenderBitExactness : public ::testing::Test,
-                              public test::PacketSource {
- protected:
-  static const int kTestDurationMs = 1000;
-
-  AcmSenderBitExactness()
-      : frame_size_rtp_timestamps_(0),
-        packet_count_(0),
-        payload_type_(0),
-        last_sequence_number_(0),
-        last_timestamp_(0) {}
-
-  // Sets up the test::AcmSendTest object. Returns true on success, otherwise
-  // false.
-  bool SetUpSender() {
-    const std::string input_file_name =
-        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
-    // Note that |audio_source_| will loop forever. The test duration is set
-    // explicitly by |kTestDurationMs|.
-    audio_source_.reset(new test::InputAudioFile(input_file_name));
-    static const int kSourceRateHz = 32000;
-    send_test_.reset(new test::AcmSendTest(
-        audio_source_.get(), kSourceRateHz, kTestDurationMs));
-    return send_test_.get() != NULL;
-  }
-
-  // Registers a send codec in the test::AcmSendTest object. Returns true on
-  // success, false on failure.
-  bool RegisterSendCodec(int codec_type,
-                         int channels,
-                         int payload_type,
-                         int frame_size_samples,
-                         int frame_size_rtp_timestamps) {
-    payload_type_ = payload_type;
-    frame_size_rtp_timestamps_ = frame_size_rtp_timestamps;
-    return send_test_->RegisterCodec(
-        codec_type, channels, payload_type, frame_size_samples);
-  }
-
-  // Runs the test. SetUpSender() and RegisterSendCodec() must have been called
-  // before calling this method.
-  void Run(const std::string& audio_checksum_ref,
-           const std::string& payload_checksum_ref,
-           int expected_packets,
-           test::AcmReceiveTest::NumOutputChannels expected_channels) {
-    // Set up the receiver used to decode the packets and verify the decoded
-    // output.
-    test::AudioChecksum audio_checksum;
-    const std::string output_file_name =
-        webrtc::test::OutputPath() +
-        ::testing::UnitTest::GetInstance()
-            ->current_test_info()
-            ->test_case_name() +
-        "_" +
-        ::testing::UnitTest::GetInstance()->current_test_info()->name() +
-        "_output.pcm";
-    test::OutputAudioFile output_file(output_file_name);
-    // Have the output audio sent both to file and to the checksum calculator.
-    test::AudioSinkFork output(&audio_checksum, &output_file);
-    const int kOutputFreqHz = 8000;
-    test::AcmReceiveTest receive_test(
-        this, &output, kOutputFreqHz, expected_channels);
-    ASSERT_NO_FATAL_FAILURE(receive_test.RegisterDefaultCodecs());
-
-    // This is where the actual test is executed.
-    receive_test.Run();
-
-    // Extract and verify the audio checksum.
-    std::string checksum_string = audio_checksum.Finish();
-    EXPECT_EQ(audio_checksum_ref, checksum_string);
-
-    // Extract and verify the payload checksum.
-    char checksum_result[rtc::Md5Digest::kSize];
-    payload_checksum_.Finish(checksum_result, rtc::Md5Digest::kSize);
-    checksum_string = rtc::hex_encode(checksum_result, rtc::Md5Digest::kSize);
-    EXPECT_EQ(payload_checksum_ref, checksum_string);
-
-    // Verify number of packets produced.
-    EXPECT_EQ(expected_packets, packet_count_);
-  }
-
-  // Returns a pointer to the next packet. Returns NULL if the source is
-  // depleted (i.e., the test duration is exceeded), or if an error occurred.
-  // Inherited from test::PacketSource.
-  test::Packet* NextPacket() override {
-    // Get the next packet from AcmSendTest. Ownership of |packet| is
-    // transferred to this method.
-    test::Packet* packet = send_test_->NextPacket();
-    if (!packet)
-      return NULL;
-
-    VerifyPacket(packet);
-    // TODO(henrik.lundin) Save the packet to file as well.
-
-    // Pass it on to the caller. The caller becomes the owner of |packet|.
-    return packet;
-  }
-
-  // Verifies the packet.
-  void VerifyPacket(const test::Packet* packet) {
-    EXPECT_TRUE(packet->valid_header());
-    // (We can check the header fields even if valid_header() is false.)
-    EXPECT_EQ(payload_type_, packet->header().payloadType);
-    if (packet_count_ > 0) {
-      // This is not the first packet.
-      uint16_t sequence_number_diff =
-          packet->header().sequenceNumber - last_sequence_number_;
-      EXPECT_EQ(1, sequence_number_diff);
-      uint32_t timestamp_diff = packet->header().timestamp - last_timestamp_;
-      EXPECT_EQ(frame_size_rtp_timestamps_, timestamp_diff);
-    }
-    ++packet_count_;
-    last_sequence_number_ = packet->header().sequenceNumber;
-    last_timestamp_ = packet->header().timestamp;
-    // Update the checksum.
-    payload_checksum_.Update(packet->payload(), packet->payload_length_bytes());
-  }
-
-  void SetUpTest(int codec_type,
-                 int channels,
-                 int payload_type,
-                 int codec_frame_size_samples,
-                 int codec_frame_size_rtp_timestamps) {
-    ASSERT_TRUE(SetUpSender());
-    ASSERT_TRUE(RegisterSendCodec(codec_type,
-                                  channels,
-                                  payload_type,
-                                  codec_frame_size_samples,
-                                  codec_frame_size_rtp_timestamps));
-  }
-
-  rtc::scoped_ptr<test::AcmSendTest> send_test_;
-  rtc::scoped_ptr<test::InputAudioFile> audio_source_;
-  uint32_t frame_size_rtp_timestamps_;
-  int packet_count_;
-  uint8_t payload_type_;
-  uint16_t last_sequence_number_;
-  uint32_t last_timestamp_;
-  rtc::Md5Digest payload_checksum_;
-};
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_IsacWb30ms DISABLED_IsacWb30ms
-#else
-#define MAYBE_IsacWb30ms IsacWb30ms
-#endif
-TEST_F(AcmSenderBitExactness, IF_ISAC(MAYBE_IsacWb30ms)) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kISAC, 1, 103, 480, 480));
-  Run(AcmReceiverBitExactness::PlatformChecksum(
-          "c7e5bdadfa2871df95639fcc297cf23d",
-          "0499ca260390769b3172136faad925b9",
-          "0b58f9eeee43d5891f5f6c75e77984a3"),
-      AcmReceiverBitExactness::PlatformChecksum(
-          "d42cb5195463da26c8129bbfe73a22e6",
-          "83de248aea9c3c2bd680b6952401b4ca",
-          "3c79f16f34218271f3dca4e2b1dfe1bb"),
-      33,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_IsacWb60ms DISABLED_IsacWb60ms
-#else
-#define MAYBE_IsacWb60ms IsacWb60ms
-#endif
-TEST_F(AcmSenderBitExactness, IF_ISAC(MAYBE_IsacWb60ms)) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kISAC, 1, 103, 960, 960));
-  Run(AcmReceiverBitExactness::PlatformChecksum(
-          "14d63c5f08127d280e722e3191b73bdd",
-          "8da003e16c5371af2dc2be79a50f9076",
-          "1ad29139a04782a33daad8c2b9b35875"),
-      AcmReceiverBitExactness::PlatformChecksum(
-          "ebe04a819d3a9d83a83a17f271e1139a",
-          "97aeef98553b5a4b5a68f8b716e8eaf0",
-          "9e0a0ab743ad987b55b8e14802769c56"),
-      16,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-#ifdef WEBRTC_CODEC_ISAC
-#define IF_ISAC_FLOAT(x) x
-#else
-#define IF_ISAC_FLOAT(x) DISABLED_##x
-#endif
-
-TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IF_ISAC_FLOAT(IsacSwb30ms))) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kISACSWB, 1, 104, 960, 960));
-  Run(AcmReceiverBitExactness::PlatformChecksum(
-          "2b3c387d06f00b7b7aad4c9be56fb83d",
-          "",
-          "5683b58da0fbf2063c7adc2e6bfb3fb8"),
-      AcmReceiverBitExactness::PlatformChecksum(
-          "bcc2041e7744c7ebd9f701866856849c",
-          "",
-          "ce86106a93419aefb063097108ec94ab"),
-      33, test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcm16_8000khz_10ms) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kPCM16B, 1, 107, 80, 80));
-  Run("de4a98e1406f8b798d99cd0704e862e2",
-      "c1edd36339ce0326cc4550041ad719a0",
-      100,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcm16_16000khz_10ms) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kPCM16Bwb, 1, 108, 160, 160));
-  Run("ae646d7b68384a1269cc080dd4501916",
-      "ad786526383178b08d80d6eee06e9bad",
-      100,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcm16_32000khz_10ms) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kPCM16Bswb32kHz, 1, 109, 320, 320));
-  Run("7fe325e8fbaf755e3c5df0b11a4774fb",
-      "5ef82ea885e922263606c6fdbc49f651",
-      100,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcm16_stereo_8000khz_10ms) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kPCM16B_2ch, 2, 111, 80, 80));
-  Run("fb263b74e7ac3de915474d77e4744ceb",
-      "62ce5adb0d4965d0a52ec98ae7f98974",
-      100,
-      test::AcmReceiveTest::kStereoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcm16_stereo_16000khz_10ms) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kPCM16Bwb_2ch, 2, 112, 160, 160));
-  Run("d09e9239553649d7ac93e19d304281fd",
-      "41ca8edac4b8c71cd54fd9f25ec14870",
-      100,
-      test::AcmReceiveTest::kStereoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcm16_stereo_32000khz_10ms) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch, 2, 113, 320, 320));
-  Run("5f025d4f390982cc26b3d92fe02e3044",
-      "50e58502fb04421bf5b857dda4c96879",
-      100,
-      test::AcmReceiveTest::kStereoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcmu_20ms) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kPCMU, 1, 0, 160, 160));
-  Run("81a9d4c0bb72e9becc43aef124c981e9",
-      "8f9b8750bd80fe26b6cbf6659b89f0f9",
-      50,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcma_20ms) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kPCMA, 1, 8, 160, 160));
-  Run("39611f798969053925a49dc06d08de29",
-      "6ad745e55aa48981bfc790d0eeef2dd1",
-      50,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcmu_stereo_20ms) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kPCMU_2ch, 2, 110, 160, 160));
-  Run("437bec032fdc5cbaa0d5175430af7b18",
-      "60b6f25e8d1e74cb679cfe756dd9bca5",
-      50,
-      test::AcmReceiveTest::kStereoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, Pcma_stereo_20ms) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kPCMA_2ch, 2, 118, 160, 160));
-  Run("a5c6d83c5b7cedbeff734238220a4b0c",
-      "92b282c83efd20e7eeef52ba40842cf7",
-      50,
-      test::AcmReceiveTest::kStereoOutput);
-}
-
-#ifdef WEBRTC_CODEC_ILBC
-#define IF_ILBC(x) x
-#else
-#define IF_ILBC(x) DISABLED_##x
-#endif
-
-TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IF_ILBC(Ilbc_30ms))) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kILBC, 1, 102, 240, 240));
-  Run(AcmReceiverBitExactness::PlatformChecksum(
-          "7b6ec10910debd9af08011d3ed5249f7",
-          "android_audio",
-          "7b6ec10910debd9af08011d3ed5249f7"),
-      AcmReceiverBitExactness::PlatformChecksum(
-          "cfae2e9f6aba96e145f2bcdd5050ce78",
-          "android_payload",
-          "cfae2e9f6aba96e145f2bcdd5050ce78"),
-      33,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-#ifdef WEBRTC_CODEC_G722
-#define IF_G722(x) x
-#else
-#define IF_G722(x) DISABLED_##x
-#endif
-
-TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IF_G722(G722_20ms))) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kG722, 1, 9, 320, 160));
-  Run(AcmReceiverBitExactness::PlatformChecksum(
-          "7d759436f2533582950d148b5161a36c",
-          "android_audio",
-          "7d759436f2533582950d148b5161a36c"),
-      AcmReceiverBitExactness::PlatformChecksum(
-          "fc68a87e1380614e658087cb35d5ca10",
-          "android_payload",
-          "fc68a87e1380614e658087cb35d5ca10"),
-      50,
-      test::AcmReceiveTest::kMonoOutput);
-}
-
-TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IF_G722(G722_stereo_20ms))) {
-  ASSERT_NO_FATAL_FAILURE(
-      SetUpTest(acm2::ACMCodecDB::kG722_2ch, 2, 119, 320, 160));
-  Run(AcmReceiverBitExactness::PlatformChecksum(
-          "7190ee718ab3d80eca181e5f7140c210",
-          "android_audio",
-          "7190ee718ab3d80eca181e5f7140c210"),
-      AcmReceiverBitExactness::PlatformChecksum(
-          "66516152eeaa1e650ad94ff85f668dac",
-          "android_payload",
-          "66516152eeaa1e650ad94ff85f668dac"),
-      50,
-      test::AcmReceiveTest::kStereoOutput);
-}
-
-// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
-#define MAYBE_Opus_stereo_20ms DISABLED_Opus_stereo_20ms
-#else
-#define MAYBE_Opus_stereo_20ms Opus_stereo_20ms
-#endif
-TEST_F(AcmSenderBitExactness, MAYBE_Opus_stereo_20ms) {
-  ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kOpus, 2, 120, 960, 960));
-  Run(AcmReceiverBitExactness::PlatformChecksum(
-          "855041f2490b887302bce9d544731849",
-          "1e1a0fce893fef2d66886a7f09e2ebce",
-          "855041f2490b887302bce9d544731849"),
-      AcmReceiverBitExactness::PlatformChecksum(
-          "d781cce1ab986b618d0da87226cdde30",
-          "1a1fe04dd12e755949987c8d729fb3e0",
-          "d781cce1ab986b618d0da87226cdde30"),
-      50,
-      test::AcmReceiveTest::kStereoOutput);
-}
-
-#endif
-
-}  // namespace webrtc