/* * 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 "modules/audio_coding/include/audio_coding_module.h" #include #include #include #include #include #include "api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h" #include "api/audio_codecs/opus/audio_decoder_opus.h" #include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h" #include "api/audio_codecs/opus/audio_encoder_opus.h" #include "modules/audio_coding/acm2/acm_receive_test.h" #include "modules/audio_coding/acm2/acm_send_test.h" #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h" #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h" #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h" #include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h" #include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "modules/audio_coding/neteq/tools/audio_checksum.h" #include "modules/audio_coding/neteq/tools/audio_loop.h" #include "modules/audio_coding/neteq/tools/constant_pcm_packet_source.h" #include "modules/audio_coding/neteq/tools/input_audio_file.h" #include "modules/audio_coding/neteq/tools/output_audio_file.h" #include "modules/audio_coding/neteq/tools/output_wav_file.h" #include "modules/audio_coding/neteq/tools/packet.h" #include "modules/audio_coding/neteq/tools/rtp_file_source.h" #include "rtc_base/event.h" #include "rtc_base/message_digest.h" #include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/platform_thread.h" #include "rtc_base/ref_counted_object.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/system/arch.h" #include "rtc_base/thread_annotations.h" #include "system_wrappers/include/clock.h" #include "system_wrappers/include/cpu_features_wrapper.h" #include "system_wrappers/include/sleep.h" #include "test/audio_decoder_proxy_factory.h" #include "test/gtest.h" #include "test/mock_audio_decoder.h" #include "test/mock_audio_encoder.h" #include "test/testsupport/file_utils.h" #include "test/testsupport/rtc_expect_death.h" using ::testing::_; using ::testing::AtLeast; using ::testing::Invoke; namespace webrtc { namespace { const int kSampleRateHz = 16000; const int kNumSamples10ms = kSampleRateHz / 100; const int kFrameSizeMs = 10; // Multiple of 10. const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms; const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t); const uint8_t kPayloadType = 111; } // namespace class RtpData { public: RtpData(int samples_per_packet, uint8_t payload_type) : samples_per_packet_(samples_per_packet), payload_type_(payload_type) {} virtual ~RtpData() {} void Populate(RTPHeader* rtp_header) { rtp_header->sequenceNumber = 0xABCD; rtp_header->timestamp = 0xABCDEF01; rtp_header->payloadType = payload_type_; rtp_header->markerBit = false; rtp_header->ssrc = 0x1234; rtp_header->numCSRCs = 0; rtp_header->payload_type_frequency = kSampleRateHz; } void Forward(RTPHeader* rtp_header) { ++rtp_header->sequenceNumber; rtp_header->timestamp += samples_per_packet_; } private: int samples_per_packet_; uint8_t payload_type_; }; class PacketizationCallbackStubOldApi : public AudioPacketizationCallback { public: PacketizationCallbackStubOldApi() : num_calls_(0), last_frame_type_(AudioFrameType::kEmptyFrame), last_payload_type_(-1), last_timestamp_(0) {} int32_t SendData(AudioFrameType frame_type, uint8_t payload_type, uint32_t timestamp, const uint8_t* payload_data, size_t payload_len_bytes, int64_t absolute_capture_timestamp_ms) override { MutexLock lock(&mutex_); ++num_calls_; last_frame_type_ = frame_type; last_payload_type_ = payload_type; last_timestamp_ = timestamp; last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes); return 0; } int num_calls() const { MutexLock lock(&mutex_); return num_calls_; } int last_payload_len_bytes() const { MutexLock lock(&mutex_); return rtc::checked_cast(last_payload_vec_.size()); } AudioFrameType last_frame_type() const { MutexLock lock(&mutex_); return last_frame_type_; } int last_payload_type() const { MutexLock lock(&mutex_); return last_payload_type_; } uint32_t last_timestamp() const { MutexLock lock(&mutex_); return last_timestamp_; } void SwapBuffers(std::vector* payload) { MutexLock lock(&mutex_); last_payload_vec_.swap(*payload); } private: int num_calls_ RTC_GUARDED_BY(mutex_); AudioFrameType last_frame_type_ RTC_GUARDED_BY(mutex_); int last_payload_type_ RTC_GUARDED_BY(mutex_); uint32_t last_timestamp_ RTC_GUARDED_BY(mutex_); std::vector last_payload_vec_ RTC_GUARDED_BY(mutex_); mutable Mutex mutex_; }; class AudioCodingModuleTestOldApi : public ::testing::Test { protected: AudioCodingModuleTestOldApi() : rtp_utility_(new RtpData(kFrameSizeSamples, kPayloadType)), clock_(Clock::GetRealTimeClock()) {} ~AudioCodingModuleTestOldApi() {} void TearDown() {} void SetUp() { acm_.reset(AudioCodingModule::Create([this] { AudioCodingModule::Config config; config.clock = clock_; config.decoder_factory = CreateBuiltinAudioDecoderFactory(); return config; }())); 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"); input_frame_.Mute(); ASSERT_EQ(0, acm_->RegisterTransportCallback(&packet_cb_)); SetUpL16Codec(); } // Set up L16 codec. virtual void SetUpL16Codec() { audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 1); pac_size_ = 160; } virtual void RegisterCodec() { acm_->SetReceiveCodecs({{kPayloadType, *audio_format_}}); acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder( kPayloadType, *audio_format_, absl::nullopt)); } virtual void InsertPacketAndPullAudio() { InsertPacket(); PullAudio(); } virtual void InsertPacket() { const uint8_t kPayload[kPayloadSizeBytes] = {0}; ASSERT_EQ(0, acm_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_)); rtp_utility_->Forward(&rtp_header_); } virtual void PullAudio() { AudioFrame audio_frame; bool muted; ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame, &muted)); ASSERT_FALSE(muted); } virtual void InsertAudio() { ASSERT_GE(acm_->Add10MsData(input_frame_), 0); input_frame_.timestamp_ += kNumSamples10ms; } virtual void VerifyEncoding() { int last_length = packet_cb_.last_payload_len_bytes(); EXPECT_TRUE(last_length == 2 * pac_size_ || last_length == 0) << "Last encoded packet was " << last_length << " bytes."; } virtual void InsertAudioAndVerifyEncoding() { InsertAudio(); VerifyEncoding(); } std::unique_ptr rtp_utility_; std::unique_ptr acm_; PacketizationCallbackStubOldApi packet_cb_; RTPHeader rtp_header_; AudioFrame input_frame_; absl::optional audio_format_; int pac_size_ = -1; Clock* clock_; }; class AudioCodingModuleTestOldApiDeathTest : public AudioCodingModuleTestOldApi {}; TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) { AudioFrame audio_frame; const int kSampleRateHz = 32000; bool muted; EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame, &muted)); ASSERT_FALSE(muted); EXPECT_EQ(0u, audio_frame.timestamp_); EXPECT_GT(audio_frame.num_channels_, 0u); EXPECT_EQ(static_cast(kSampleRateHz / 100), audio_frame.samples_per_channel_); EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_); } // The below test is temporarily disabled on Windows due to problems // with clang debug builds. // TODO(tommi): Re-enable when we've figured out what the problem is. // http://crbug.com/615050 #if !defined(WEBRTC_WIN) && defined(__clang__) && RTC_DCHECK_IS_ON && \ GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) TEST_F(AudioCodingModuleTestOldApiDeathTest, FailOnZeroDesiredFrequency) { AudioFrame audio_frame; bool muted; RTC_EXPECT_DEATH(acm_->PlayoutData10Ms(0, &audio_frame, &muted), "dst_sample_rate_hz"); } #endif // Checks that the transport callback is invoked once for each speech packet. // Also checks that the frame type is kAudioFrameSpeech. TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) { const int k10MsBlocksPerPacket = 3; pac_size_ = k10MsBlocksPerPacket * kSampleRateHz / 100; audio_format_->parameters["ptime"] = "30"; RegisterCodec(); const int kLoops = 10; for (int i = 0; i < kLoops; ++i) { EXPECT_EQ(i / k10MsBlocksPerPacket, packet_cb_.num_calls()); if (packet_cb_.num_calls() > 0) EXPECT_EQ(AudioFrameType::kAudioFrameSpeech, packet_cb_.last_frame_type()); InsertAudioAndVerifyEncoding(); } EXPECT_EQ(kLoops / k10MsBlocksPerPacket, packet_cb_.num_calls()); EXPECT_EQ(AudioFrameType::kAudioFrameSpeech, packet_cb_.last_frame_type()); } #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX) // Verifies that the RTP timestamp series is not reset when the codec is // changed. TEST_F(AudioCodingModuleTestOldApi, TimestampSeriesContinuesWhenCodecChanges) { RegisterCodec(); // This registers the default codec. uint32_t expected_ts = input_frame_.timestamp_; int blocks_per_packet = pac_size_ / (kSampleRateHz / 100); // Encode 5 packets of the first codec type. const int kNumPackets1 = 5; for (int j = 0; j < kNumPackets1; ++j) { for (int i = 0; i < blocks_per_packet; ++i) { EXPECT_EQ(j, packet_cb_.num_calls()); InsertAudio(); } EXPECT_EQ(j + 1, packet_cb_.num_calls()); EXPECT_EQ(expected_ts, packet_cb_.last_timestamp()); expected_ts += pac_size_; } // Change codec. audio_format_ = SdpAudioFormat("ISAC", kSampleRateHz, 1); pac_size_ = 480; RegisterCodec(); blocks_per_packet = pac_size_ / (kSampleRateHz / 100); // Encode another 5 packets. const int kNumPackets2 = 5; for (int j = 0; j < kNumPackets2; ++j) { for (int i = 0; i < blocks_per_packet; ++i) { EXPECT_EQ(kNumPackets1 + j, packet_cb_.num_calls()); InsertAudio(); } EXPECT_EQ(kNumPackets1 + j + 1, packet_cb_.num_calls()); EXPECT_EQ(expected_ts, packet_cb_.last_timestamp()); expected_ts += pac_size_; } } #endif // Introduce this class to set different expectations on the number of encoded // bytes. This class expects all encoded packets to be 9 bytes (matching one // CNG SID frame) or 0 bytes. This test depends on |input_frame_| containing // (near-)zero values. It also introduces a way to register comfort noise with // a custom payload type. class AudioCodingModuleTestWithComfortNoiseOldApi : public AudioCodingModuleTestOldApi { protected: void RegisterCngCodec(int rtp_payload_type) { acm_->SetReceiveCodecs({{kPayloadType, *audio_format_}, {rtp_payload_type, {"cn", kSampleRateHz, 1}}}); acm_->ModifyEncoder([&](std::unique_ptr* enc) { AudioEncoderCngConfig config; config.speech_encoder = std::move(*enc); config.num_channels = 1; config.payload_type = rtp_payload_type; config.vad_mode = Vad::kVadNormal; *enc = CreateComfortNoiseEncoder(std::move(config)); }); } void VerifyEncoding() override { int last_length = packet_cb_.last_payload_len_bytes(); EXPECT_TRUE(last_length == 9 || last_length == 0) << "Last encoded packet was " << last_length << " bytes."; } void DoTest(int blocks_per_packet, int cng_pt) { const int kLoops = 40; // This array defines the expected frame types, and when they should arrive. // We expect a frame to arrive each time the speech encoder would have // produced a packet, and once every 100 ms the frame should be non-empty, // that is contain comfort noise. const struct { int ix; AudioFrameType type; } expectation[] = {{2, AudioFrameType::kAudioFrameCN}, {5, AudioFrameType::kEmptyFrame}, {8, AudioFrameType::kEmptyFrame}, {11, AudioFrameType::kAudioFrameCN}, {14, AudioFrameType::kEmptyFrame}, {17, AudioFrameType::kEmptyFrame}, {20, AudioFrameType::kAudioFrameCN}, {23, AudioFrameType::kEmptyFrame}, {26, AudioFrameType::kEmptyFrame}, {29, AudioFrameType::kEmptyFrame}, {32, AudioFrameType::kAudioFrameCN}, {35, AudioFrameType::kEmptyFrame}, {38, AudioFrameType::kEmptyFrame}}; for (int i = 0; i < kLoops; ++i) { int num_calls_before = packet_cb_.num_calls(); EXPECT_EQ(i / blocks_per_packet, num_calls_before); InsertAudioAndVerifyEncoding(); int num_calls = packet_cb_.num_calls(); if (num_calls == num_calls_before + 1) { EXPECT_EQ(expectation[num_calls - 1].ix, i); EXPECT_EQ(expectation[num_calls - 1].type, packet_cb_.last_frame_type()) << "Wrong frame type for lap " << i; EXPECT_EQ(cng_pt, packet_cb_.last_payload_type()); } else { EXPECT_EQ(num_calls, num_calls_before); } } } }; // Checks that the transport callback is invoked once per frame period of the // underlying speech encoder, even when comfort noise is produced. // Also checks that the frame type is kAudioFrameCN or kEmptyFrame. TEST_F(AudioCodingModuleTestWithComfortNoiseOldApi, TransportCallbackTestForComfortNoiseRegisterCngLast) { const int k10MsBlocksPerPacket = 3; pac_size_ = k10MsBlocksPerPacket * kSampleRateHz / 100; audio_format_->parameters["ptime"] = "30"; RegisterCodec(); const int kCngPayloadType = 105; RegisterCngCodec(kCngPayloadType); DoTest(k10MsBlocksPerPacket, kCngPayloadType); } // 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 AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi { protected: static const int kNumPackets = 500; static const int kNumPullCalls = 500; AudioCodingModuleMtTestOldApi() : AudioCodingModuleTestOldApi(), send_count_(0), insert_packet_count_(0), pull_audio_count_(0), next_insert_packet_time_ms_(0), fake_clock_(new SimulatedClock(0)) { clock_ = fake_clock_.get(); } void SetUp() { AudioCodingModuleTestOldApi::SetUp(); RegisterCodec(); // Must be called before the threads start below. StartThreads(); } void StartThreads() { quit_.store(false); const auto attributes = rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime); send_thread_ = rtc::PlatformThread::SpawnJoinable( [this] { while (!quit_.load()) { CbSendImpl(); } }, "send", attributes); insert_packet_thread_ = rtc::PlatformThread::SpawnJoinable( [this] { while (!quit_.load()) { CbInsertPacketImpl(); } }, "insert_packet", attributes); pull_audio_thread_ = rtc::PlatformThread::SpawnJoinable( [this] { while (!quit_.load()) { CbPullAudioImpl(); } }, "pull_audio", attributes); } void TearDown() { AudioCodingModuleTestOldApi::TearDown(); quit_.store(true); pull_audio_thread_.Finalize(); send_thread_.Finalize(); insert_packet_thread_.Finalize(); } bool RunTest() { return test_complete_.Wait(10 * 60 * 1000); // 10 minutes' timeout. } virtual bool TestDone() { if (packet_cb_.num_calls() > kNumPackets) { MutexLock lock(&mutex_); if (pull_audio_count_ > kNumPullCalls) { // Both conditions for completion are met. End the test. return true; } } return false; } // The send thread doesn't have to care about the current simulated time, // since only the AcmReceiver is using the clock. void CbSendImpl() { SleepMs(1); if (HasFatalFailure()) { // End the test early if a fatal failure (ASSERT_*) has occurred. test_complete_.Set(); } ++send_count_; InsertAudioAndVerifyEncoding(); if (TestDone()) { test_complete_.Set(); } } void CbInsertPacketImpl() { SleepMs(1); { MutexLock lock(&mutex_); if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) { return; } next_insert_packet_time_ms_ += 10; } // Now we're not holding the crit sect when calling ACM. ++insert_packet_count_; InsertPacket(); } void CbPullAudioImpl() { SleepMs(1); { MutexLock lock(&mutex_); // Don't let the insert thread fall behind. if (next_insert_packet_time_ms_ < clock_->TimeInMilliseconds()) { return; } ++pull_audio_count_; } // Now we're not holding the crit sect when calling ACM. PullAudio(); fake_clock_->AdvanceTimeMilliseconds(10); } rtc::PlatformThread send_thread_; rtc::PlatformThread insert_packet_thread_; rtc::PlatformThread pull_audio_thread_; // Used to force worker threads to stop looping. std::atomic quit_; rtc::Event test_complete_; int send_count_; int insert_packet_count_; int pull_audio_count_ RTC_GUARDED_BY(mutex_); Mutex mutex_; int64_t next_insert_packet_time_ms_ RTC_GUARDED_BY(mutex_); std::unique_ptr fake_clock_; }; #if defined(WEBRTC_IOS) #define MAYBE_DoTest DISABLED_DoTest #else #define MAYBE_DoTest DoTest #endif TEST_F(AudioCodingModuleMtTestOldApi, MAYBE_DoTest) { EXPECT_TRUE(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 AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi { protected: static const int kNumPackets = 500; static const int kNumPullCalls = 500; AcmIsacMtTestOldApi() : AudioCodingModuleMtTestOldApi(), last_packet_number_(0) {} ~AcmIsacMtTestOldApi() {} void SetUp() override { AudioCodingModuleTestOldApi::SetUp(); RegisterCodec(); // Must be called before the threads start below. // 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"); audio_format_ = SdpAudioFormat("isac", kSampleRateHz, 1); pac_size_ = 480; // Register iSAC codec in ACM, effectively unregistering the PCM16B codec // registered in AudioCodingModuleTestOldApi::SetUp(); acm_->SetReceiveCodecs({{kPayloadType, *audio_format_}}); acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder( kPayloadType, *audio_format_, absl::nullopt)); } 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_EQ(0, acm_->IncomingPacket(&last_payload_vec_[0], last_payload_vec_.size(), rtp_header_)); } void InsertAudio() override { // TODO(kwiberg): Use std::copy here. Might be complications because AFAICS // this call confuses the number of samples with the number of bytes, and // ends up copying only half of what it should. memcpy(input_frame_.mutable_data(), audio_loop_.GetNextBlock().data(), kNumSamples10ms); AudioCodingModuleTestOldApi::InsertAudio(); } // Override the verification function with no-op, since iSAC produces variable // payload sizes. void VerifyEncoding() override {} // This method is the same as AudioCodingModuleMtTestOldApi::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) { MutexLock lock(&mutex_); if (pull_audio_count_ > kNumPullCalls) { // Both conditions for completion are met. End the test. return true; } } return false; } int last_packet_number_; std::vector last_payload_vec_; test::AudioLoop audio_loop_; }; #if defined(WEBRTC_IOS) #define MAYBE_DoTest DISABLED_DoTest #else #define MAYBE_DoTest DoTest #endif #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX) TEST_F(AcmIsacMtTestOldApi, MAYBE_DoTest) { EXPECT_TRUE(RunTest()); } #endif class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi { protected: static const int kRegisterAfterNumPackets = 5; static const int kNumPackets = 10; static const int kPacketSizeMs = 30; static const int kPacketSizeSamples = kPacketSizeMs * 16; AcmReRegisterIsacMtTestOldApi() : AudioCodingModuleTestOldApi(), codec_registered_(false), receive_packet_count_(0), next_insert_packet_time_ms_(0), fake_clock_(new SimulatedClock(0)) { AudioEncoderIsacFloatImpl::Config config; config.payload_type = kPayloadType; isac_encoder_.reset(new AudioEncoderIsacFloatImpl(config)); clock_ = fake_clock_.get(); } void SetUp() override { AudioCodingModuleTestOldApi::SetUp(); // 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); RegisterCodec(); // Must be called before the threads start below. StartThreads(); } void RegisterCodec() override { // Register iSAC codec in ACM, effectively unregistering the PCM16B codec // registered in AudioCodingModuleTestOldApi::SetUp(); // Only register the decoder for now. The encoder is registered later. static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz"); acm_->SetReceiveCodecs({{kPayloadType, {"ISAC", kSampleRateHz, 1}}}); } void StartThreads() { quit_.store(false); const auto attributes = rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime); receive_thread_ = rtc::PlatformThread::SpawnJoinable( [this] { while (!quit_.load() && CbReceiveImpl()) { } }, "receive", attributes); codec_registration_thread_ = rtc::PlatformThread::SpawnJoinable( [this] { while (!quit_.load()) { CbCodecRegistrationImpl(); } }, "codec_registration", attributes); } void TearDown() override { AudioCodingModuleTestOldApi::TearDown(); quit_.store(true); receive_thread_.Finalize(); codec_registration_thread_.Finalize(); } bool RunTest() { return test_complete_.Wait(10 * 60 * 1000); // 10 minutes' timeout. } bool CbReceiveImpl() { SleepMs(1); rtc::Buffer encoded; AudioEncoder::EncodedInfo info; { MutexLock lock(&mutex_); if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) { return true; } next_insert_packet_time_ms_ += kPacketSizeMs; ++receive_packet_count_; // Encode new frame. uint32_t input_timestamp = rtp_header_.timestamp; while (info.encoded_bytes == 0) { info = isac_encoder_->Encode(input_timestamp, audio_loop_.GetNextBlock(), &encoded); input_timestamp += 160; // 10 ms at 16 kHz. } EXPECT_EQ(rtp_header_.timestamp + kPacketSizeSamples, input_timestamp); EXPECT_EQ(rtp_header_.timestamp, info.encoded_timestamp); EXPECT_EQ(rtp_header_.payloadType, info.payload_type); } // Now we're not holding the crit sect when calling ACM. // Insert into ACM. EXPECT_EQ(0, acm_->IncomingPacket(encoded.data(), info.encoded_bytes, rtp_header_)); // Pull audio. for (int i = 0; i < rtc::CheckedDivExact(kPacketSizeMs, 10); ++i) { AudioFrame audio_frame; bool muted; EXPECT_EQ(0, acm_->PlayoutData10Ms(-1 /* default output frequency */, &audio_frame, &muted)); if (muted) { ADD_FAILURE(); return false; } fake_clock_->AdvanceTimeMilliseconds(10); } rtp_utility_->Forward(&rtp_header_); return true; } void CbCodecRegistrationImpl() { SleepMs(1); if (HasFatalFailure()) { // End the test early if a fatal failure (ASSERT_*) has occurred. test_complete_.Set(); } MutexLock lock(&mutex_); if (!codec_registered_ && receive_packet_count_ > kRegisterAfterNumPackets) { // Register the iSAC encoder. acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder( kPayloadType, *audio_format_, absl::nullopt)); codec_registered_ = true; } if (codec_registered_ && receive_packet_count_ > kNumPackets) { test_complete_.Set(); } } rtc::PlatformThread receive_thread_; rtc::PlatformThread codec_registration_thread_; // Used to force worker threads to stop looping. std::atomic quit_; rtc::Event test_complete_; Mutex mutex_; bool codec_registered_ RTC_GUARDED_BY(mutex_); int receive_packet_count_ RTC_GUARDED_BY(mutex_); int64_t next_insert_packet_time_ms_ RTC_GUARDED_BY(mutex_); std::unique_ptr isac_encoder_; std::unique_ptr fake_clock_; test::AudioLoop audio_loop_; }; #if defined(WEBRTC_IOS) #define MAYBE_DoTest DISABLED_DoTest #else #define MAYBE_DoTest DoTest #endif #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX) TEST_F(AcmReRegisterIsacMtTestOldApi, MAYBE_DoTest) { EXPECT_TRUE(RunTest()); } #endif // Disabling all of these tests on iOS until file support has been added. // See https://code.google.com/p/webrtc/issues/detail?id=4752 for details. #if !defined(WEBRTC_IOS) class AcmReceiverBitExactnessOldApi : public ::testing::Test { public: static std::string PlatformChecksum(std::string others, std::string win64, std::string android_arm32, std::string android_arm64, std::string android_arm64_clang) { #if defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS) return win64; #elif defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM) return android_arm32; #elif defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64) #if defined(__clang__) // Android ARM64 with Clang compiler return android_arm64_clang; #else // Android ARM64 with non-Clang compiler return android_arm64; #endif // __clang__ #else return others; #endif } protected: struct ExternalDecoder { int rtp_payload_type; AudioDecoder* external_decoder; int sample_rate_hz; int num_channels; std::string name; }; void Run(int output_freq_hz, const std::string& checksum_ref) { Run(output_freq_hz, checksum_ref, CreateBuiltinAudioDecoderFactory(), [](AudioCodingModule*) {}); } void Run(int output_freq_hz, const std::string& checksum_ref, rtc::scoped_refptr decoder_factory, rtc::FunctionView decoder_reg) { const std::string input_file_name = webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"); std::unique_ptr 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.wav"; test::OutputWavFile output_file(output_file_name, output_freq_hz, 1); test::AudioSinkFork output(&checksum, &output_file); test::AcmReceiveTestOldApi test( packet_source.get(), &output, output_freq_hz, test::AcmReceiveTestOldApi::kArbitraryChannels, std::move(decoder_factory)); ASSERT_NO_FATAL_FAILURE(test.RegisterNetEqTestCodecs()); decoder_reg(test.get_acm()); test.Run(); std::string checksum_string = checksum.Finish(); EXPECT_EQ(checksum_ref, checksum_string); // Delete the output file. remove(output_file_name.c_str()); } }; #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \ defined(WEBRTC_CODEC_ILBC) TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) { std::string others_checksum_reference = GetCPUInfo(kAVX2) != 0 ? "e0c966d7b8c36ff60167988fa35d33e0" : "7d8f6b84abd1e57ec010a53bc2130652"; std::string win64_checksum_reference = GetCPUInfo(kAVX2) != 0 ? "405a50f0bcb8827e20aa944299fc59f6" : "0ed5830930f5527a01bbec0ba11f8541"; Run(8000, PlatformChecksum(others_checksum_reference, win64_checksum_reference, "b892ed69c38b21b16c132ec2ce03aa7b", "4598140b5e4f7ee66c5adad609e65a3e", "5fec8d770778ef7969ec98c56d9eb10f")); } TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) { std::string others_checksum_reference = GetCPUInfo(kAVX2) != 0 ? "a63c578e1195c8420f453962c6d8519c" : "6bac83762c1306b932cd25a560155681"; std::string win64_checksum_reference = GetCPUInfo(kAVX2) != 0 ? "58fd62a5c49ee513f9fa6fe7dbf62c97" : "0509cf0672f543efb4b050e8cffefb1d"; Run(16000, PlatformChecksum(others_checksum_reference, win64_checksum_reference, "3cea9abbeabbdea9a79719941b241af5", "f2aad418af974a3b1694d5ae5cc2c3c7", "9d4b92c31c00e321a4cff29ad002d6a2")); } TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) { std::string others_checksum_reference = GetCPUInfo(kAVX2) != 0 ? "8775ce387f44dc5ff4a26da295d5ee7c" : "e319222ca47733709f90fdf33c8574db"; std::string win64_checksum_reference = GetCPUInfo(kAVX2) != 0 ? "04ce6a1dac5ffdd8438d804623d0132f" : "39a4a7a1c455b35baeffb9fd193d7858"; Run(32000, PlatformChecksum(others_checksum_reference, win64_checksum_reference, "4df55b3b62bcbf4328786d474ae87f61", "100869c8dcde51346c2073e52a272d98", "ff58d3153d2780a3df6bc2068844cb2d")); } TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) { std::string others_checksum_reference = GetCPUInfo(kAVX2) != 0 ? "7a55700b7ca9aa60237db58b33e55606" : "57d1d316c88279f4f3da3511665069a9"; std::string win64_checksum_reference = GetCPUInfo(kAVX2) != 0 ? "f59833d9b0924f4b0704707dd3589f80" : "74cbe7345e2b6b45c1e455a5d1e921ca"; Run(48000, PlatformChecksum(others_checksum_reference, win64_checksum_reference, "f52bc7bf0f499c9da25932fdf176c4ec", "bd44bf97e7899186532f91235cef444d", "364d403dae55d73cd69e6dbd6b723a4d")); } TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) { class ADFactory : public AudioDecoderFactory { public: ADFactory() : mock_decoder_(new MockAudioDecoder()), pcmu_decoder_(1), decode_forwarder_(&pcmu_decoder_), fact_(CreateBuiltinAudioDecoderFactory()) { // Set expectations on the mock decoder and also delegate the calls to // the real decoder. EXPECT_CALL(*mock_decoder_, SampleRateHz()) .Times(AtLeast(1)) .WillRepeatedly( Invoke(&pcmu_decoder_, &AudioDecoderPcmU::SampleRateHz)); EXPECT_CALL(*mock_decoder_, Channels()) .Times(AtLeast(1)) .WillRepeatedly(Invoke(&pcmu_decoder_, &AudioDecoderPcmU::Channels)); EXPECT_CALL(*mock_decoder_, DecodeInternal(_, _, _, _, _)) .Times(AtLeast(1)) .WillRepeatedly(Invoke(&decode_forwarder_, &DecodeForwarder::Decode)); EXPECT_CALL(*mock_decoder_, HasDecodePlc()) .Times(AtLeast(1)) .WillRepeatedly( Invoke(&pcmu_decoder_, &AudioDecoderPcmU::HasDecodePlc)); EXPECT_CALL(*mock_decoder_, PacketDuration(_, _)) .Times(AtLeast(1)) .WillRepeatedly( Invoke(&pcmu_decoder_, &AudioDecoderPcmU::PacketDuration)); EXPECT_CALL(*mock_decoder_, Die()); } std::vector GetSupportedDecoders() override { return fact_->GetSupportedDecoders(); } bool IsSupportedDecoder(const SdpAudioFormat& format) override { return format.name == "MockPCMu" ? true : fact_->IsSupportedDecoder(format); } std::unique_ptr MakeAudioDecoder( const SdpAudioFormat& format, absl::optional codec_pair_id) override { return format.name == "MockPCMu" ? std::move(mock_decoder_) : fact_->MakeAudioDecoder(format, codec_pair_id); } private: // Class intended to forward a call from a mock DecodeInternal to Decode on // the real decoder's Decode. DecodeInternal for the real decoder isn't // public. class DecodeForwarder { public: explicit DecodeForwarder(AudioDecoder* decoder) : decoder_(decoder) {} int Decode(const uint8_t* encoded, size_t encoded_len, int sample_rate_hz, int16_t* decoded, AudioDecoder::SpeechType* speech_type) { return decoder_->Decode(encoded, encoded_len, sample_rate_hz, decoder_->PacketDuration(encoded, encoded_len) * decoder_->Channels() * sizeof(int16_t), decoded, speech_type); } private: AudioDecoder* const decoder_; }; std::unique_ptr mock_decoder_; AudioDecoderPcmU pcmu_decoder_; DecodeForwarder decode_forwarder_; rtc::scoped_refptr fact_; // Fallback factory. }; rtc::scoped_refptr> factory( new rtc::RefCountedObject); std::string others_checksum_reference = GetCPUInfo(kAVX2) != 0 ? "7a55700b7ca9aa60237db58b33e55606" : "57d1d316c88279f4f3da3511665069a9"; std::string win64_checksum_reference = GetCPUInfo(kAVX2) != 0 ? "f59833d9b0924f4b0704707dd3589f80" : "74cbe7345e2b6b45c1e455a5d1e921ca"; Run(48000, PlatformChecksum(others_checksum_reference, win64_checksum_reference, "f52bc7bf0f499c9da25932fdf176c4ec", "bd44bf97e7899186532f91235cef444d", "364d403dae55d73cd69e6dbd6b723a4d"), factory, [](AudioCodingModule* acm) { acm->SetReceiveCodecs({{0, {"MockPCMu", 8000, 1}}, {103, {"ISAC", 16000, 1}}, {104, {"ISAC", 32000, 1}}, {93, {"L16", 8000, 1}}, {94, {"L16", 16000, 1}}, {95, {"L16", 32000, 1}}, {8, {"PCMA", 8000, 1}}, {102, {"ILBC", 8000, 1}}, {13, {"CN", 8000, 1}}, {98, {"CN", 16000, 1}}, {99, {"CN", 32000, 1}}}); }); } #endif // 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 AcmSenderBitExactnessOldApi : public ::testing::Test, public test::PacketSource { protected: static const int kTestDurationMs = 1000; AcmSenderBitExactnessOldApi() : frame_size_rtp_timestamps_(0), packet_count_(0), payload_type_(0), last_sequence_number_(0), last_timestamp_(0), payload_checksum_(rtc::MessageDigestFactory::Create(rtc::DIGEST_MD5)) {} // Sets up the test::AcmSendTest object. Returns true on success, otherwise // false. bool SetUpSender(std::string input_file_name, int source_rate) { // Note that |audio_source_| will loop forever. The test duration is set // explicitly by |kTestDurationMs|. audio_source_.reset(new test::InputAudioFile(input_file_name)); send_test_.reset(new test::AcmSendTestOldApi(audio_source_.get(), source_rate, kTestDurationMs)); return send_test_.get() != NULL; } // Registers a send codec in the test::AcmSendTest object. Returns true on // success, false on failure. bool RegisterSendCodec(const char* payload_name, int sampling_freq_hz, 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(payload_name, sampling_freq_hz, channels, payload_type, frame_size_samples); } void RegisterExternalSendCodec( std::unique_ptr external_speech_encoder, int payload_type) { payload_type_ = payload_type; frame_size_rtp_timestamps_ = rtc::checked_cast( external_speech_encoder->Num10MsFramesInNextPacket() * external_speech_encoder->RtpTimestampRateHz() / 100); send_test_->RegisterExternalCodec(std::move(external_speech_encoder)); } // 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::AcmReceiveTestOldApi::NumOutputChannels expected_channels, rtc::scoped_refptr decoder_factory = nullptr) { if (!decoder_factory) { decoder_factory = CreateBuiltinAudioDecoderFactory(); } // 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.wav"; const int kOutputFreqHz = 8000; test::OutputWavFile output_file(output_file_name, kOutputFreqHz, expected_channels); // Have the output audio sent both to file and to the checksum calculator. test::AudioSinkFork output(&audio_checksum, &output_file); test::AcmReceiveTestOldApi receive_test(this, &output, kOutputFreqHz, expected_channels, decoder_factory); 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(); ExpectChecksumEq(audio_checksum_ref, checksum_string); // Extract and verify the payload checksum. rtc::Buffer checksum_result(payload_checksum_->Size()); payload_checksum_->Finish(checksum_result.data(), checksum_result.size()); checksum_string = rtc::hex_encode(checksum_result.data(), checksum_result.size()); ExpectChecksumEq(payload_checksum_ref, checksum_string); // Verify number of packets produced. EXPECT_EQ(expected_packets, packet_count_); // Delete the output file. remove(output_file_name.c_str()); } // Helper: result must be one the "|"-separated checksums. void ExpectChecksumEq(std::string ref, std::string result) { if (ref.size() == result.size()) { // Only one checksum: clearer message. EXPECT_EQ(ref, result); } else { EXPECT_NE(ref.find(result), std::string::npos) << result << " must be one of these:\n" << ref; } } // Inherited from test::PacketSource. std::unique_ptr NextPacket() override { auto packet = send_test_->NextPacket(); if (!packet) return NULL; VerifyPacket(packet.get()); // 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(const char* codec_name, int codec_sample_rate_hz, int channels, int payload_type, int codec_frame_size_samples, int codec_frame_size_rtp_timestamps) { ASSERT_TRUE(SetUpSender( channels == 1 ? kTestFileMono32kHz : kTestFileFakeStereo32kHz, 32000)); ASSERT_TRUE(RegisterSendCodec(codec_name, codec_sample_rate_hz, channels, payload_type, codec_frame_size_samples, codec_frame_size_rtp_timestamps)); } void SetUpTestExternalEncoder( std::unique_ptr external_speech_encoder, int payload_type) { ASSERT_TRUE(send_test_); RegisterExternalSendCodec(std::move(external_speech_encoder), payload_type); } std::unique_ptr send_test_; std::unique_ptr audio_source_; uint32_t frame_size_rtp_timestamps_; int packet_count_; uint8_t payload_type_; uint16_t last_sequence_number_; uint32_t last_timestamp_; std::unique_ptr payload_checksum_; const std::string kTestFileMono32kHz = webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); const std::string kTestFileFakeStereo32kHz = webrtc::test::ResourcePath("audio_coding/testfile_fake_stereo_32kHz", "pcm"); const std::string kTestFileQuad48kHz = webrtc::test::ResourcePath( "audio_coding/speech_4_channels_48k_one_second", "wav"); }; class AcmSenderBitExactnessNewApi : public AcmSenderBitExactnessOldApi {}; #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX) TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480)); Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( "2c9cb15d4ed55b5a0cadd04883bc73b0", "9336a9b993cbd8a751f0e8958e66c89c", "5c2eb46199994506236f68b2c8e51b0d", "343f1f42be0607c61e6516aece424609", "2c9cb15d4ed55b5a0cadd04883bc73b0"), AcmReceiverBitExactnessOldApi::PlatformChecksum( "3c79f16f34218271f3dca4e2b1dfe1bb", "d42cb5195463da26c8129bbfe73a22e6", "83de248aea9c3c2bd680b6952401b4ca", "3c79f16f34218271f3dca4e2b1dfe1bb", "3c79f16f34218271f3dca4e2b1dfe1bb"), 33, test::AcmReceiveTestOldApi::kMonoOutput); } TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960)); Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( "1ad29139a04782a33daad8c2b9b35875", "14d63c5f08127d280e722e3191b73bdd", "9a81e467eb1485f84aca796f8ea65011", "ef75e900e6f375e3061163c53fd09a63", "1ad29139a04782a33daad8c2b9b35875"), AcmReceiverBitExactnessOldApi::PlatformChecksum( "9e0a0ab743ad987b55b8e14802769c56", "ebe04a819d3a9d83a83a17f271e1139a", "97aeef98553b5a4b5a68f8b716e8eaf0", "9e0a0ab743ad987b55b8e14802769c56", "9e0a0ab743ad987b55b8e14802769c56"), 16, test::AcmReceiveTestOldApi::kMonoOutput); } #endif #if defined(WEBRTC_ANDROID) #define MAYBE_IsacSwb30ms DISABLED_IsacSwb30ms #else #define MAYBE_IsacSwb30ms IsacSwb30ms #endif #if defined(WEBRTC_CODEC_ISAC) TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacSwb30ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960)); Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( "5683b58da0fbf2063c7adc2e6bfb3fb8", "2b3c387d06f00b7b7aad4c9be56fb83d", "android_arm32_audio", "android_arm64_audio", "android_arm64_clang_audio"), AcmReceiverBitExactnessOldApi::PlatformChecksum( "ce86106a93419aefb063097108ec94ab", "bcc2041e7744c7ebd9f701866856849c", "android_arm32_payload", "android_arm64_payload", "android_arm64_clang_payload"), 33, test::AcmReceiveTestOldApi::kMonoOutput); } #endif TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80)); Run("15396f66b5b0ab6842e151c807395e4c", "c1edd36339ce0326cc4550041ad719a0", 100, test::AcmReceiveTestOldApi::kMonoOutput); } TEST_F(AcmSenderBitExactnessOldApi, Pcm16_16000khz_10ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 1, 108, 160, 160)); Run("54ae004529874c2b362c7f0ccd19cb99", "ad786526383178b08d80d6eee06e9bad", 100, test::AcmReceiveTestOldApi::kMonoOutput); } TEST_F(AcmSenderBitExactnessOldApi, Pcm16_32000khz_10ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 1, 109, 320, 320)); Run("d6a4a68b8c838dcc1e7ae7136467cdf0", "5ef82ea885e922263606c6fdbc49f651", 100, test::AcmReceiveTestOldApi::kMonoOutput); } TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_8000khz_10ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 2, 111, 80, 80)); Run("6b011dab43e3a8a46ccff7e4412ed8a2", "62ce5adb0d4965d0a52ec98ae7f98974", 100, test::AcmReceiveTestOldApi::kStereoOutput); } TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_16000khz_10ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 2, 112, 160, 160)); Run("17fc9854358bfe0419408290664bd78e", "41ca8edac4b8c71cd54fd9f25ec14870", 100, test::AcmReceiveTestOldApi::kStereoOutput); } TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_32000khz_10ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 2, 113, 320, 320)); Run("9ac9a1f64d55da2fc9f3167181cc511d", "50e58502fb04421bf5b857dda4c96879", 100, test::AcmReceiveTestOldApi::kStereoOutput); } TEST_F(AcmSenderBitExactnessOldApi, Pcmu_20ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 1, 0, 160, 160)); Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9", 50, test::AcmReceiveTestOldApi::kMonoOutput); } TEST_F(AcmSenderBitExactnessOldApi, Pcma_20ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 1, 8, 160, 160)); Run("39611f798969053925a49dc06d08de29", "6ad745e55aa48981bfc790d0eeef2dd1", 50, test::AcmReceiveTestOldApi::kMonoOutput); } TEST_F(AcmSenderBitExactnessOldApi, Pcmu_stereo_20ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 2, 110, 160, 160)); Run("437bec032fdc5cbaa0d5175430af7b18", "60b6f25e8d1e74cb679cfe756dd9bca5", 50, test::AcmReceiveTestOldApi::kStereoOutput); } TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 2, 118, 160, 160)); Run("a5c6d83c5b7cedbeff734238220a4b0c", "92b282c83efd20e7eeef52ba40842cf7", 50, test::AcmReceiveTestOldApi::kStereoOutput); } #if defined(WEBRTC_ANDROID) #define MAYBE_Ilbc_30ms DISABLED_Ilbc_30ms #else #define MAYBE_Ilbc_30ms Ilbc_30ms #endif #if defined(WEBRTC_CODEC_ILBC) TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Ilbc_30ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240)); Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( "7b6ec10910debd9af08011d3ed5249f7", "7b6ec10910debd9af08011d3ed5249f7", "android_arm32_audio", "android_arm64_audio", "android_arm64_clang_audio"), AcmReceiverBitExactnessOldApi::PlatformChecksum( "cfae2e9f6aba96e145f2bcdd5050ce78", "cfae2e9f6aba96e145f2bcdd5050ce78", "android_arm32_payload", "android_arm64_payload", "android_arm64_clang_payload"), 33, test::AcmReceiveTestOldApi::kMonoOutput); } #endif #if defined(WEBRTC_ANDROID) #define MAYBE_G722_20ms DISABLED_G722_20ms #else #define MAYBE_G722_20ms G722_20ms #endif TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_20ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160)); Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( "e99c89be49a46325d03c0d990c292d68", "e99c89be49a46325d03c0d990c292d68", "android_arm32_audio", "android_arm64_audio", "android_arm64_clang_audio"), AcmReceiverBitExactnessOldApi::PlatformChecksum( "fc68a87e1380614e658087cb35d5ca10", "fc68a87e1380614e658087cb35d5ca10", "android_arm32_payload", "android_arm64_payload", "android_arm64_clang_payload"), 50, test::AcmReceiveTestOldApi::kMonoOutput); } #if defined(WEBRTC_ANDROID) #define MAYBE_G722_stereo_20ms DISABLED_G722_stereo_20ms #else #define MAYBE_G722_stereo_20ms G722_stereo_20ms #endif TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160)); Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( "e280aed283e499d37091b481ca094807", "e280aed283e499d37091b481ca094807", "android_arm32_audio", "android_arm64_audio", "android_arm64_clang_audio"), AcmReceiverBitExactnessOldApi::PlatformChecksum( "66516152eeaa1e650ad94ff85f668dac", "66516152eeaa1e650ad94ff85f668dac", "android_arm32_payload", "android_arm64_payload", "android_arm64_clang_payload"), 50, test::AcmReceiveTestOldApi::kStereoOutput); } namespace { // Checksum depends on libopus being compiled with or without SSE. const std::string audio_maybe_sse = "e0ddf36854059151cdb7a0c4af3d282a" "|32574e78db4eab0c467d3c0785e3b484"; const std::string payload_maybe_sse = "b43bdf7638b2bc2a5a6f30bdc640b9ed" "|c30d463e7ed10bdd1da9045f80561f27"; // Common checksums. const std::string audio_checksum = AcmReceiverBitExactnessOldApi::PlatformChecksum( audio_maybe_sse, audio_maybe_sse, "6fcceb83acf427730570bc13eeac920c", "fd96f15d547c4e155daeeef4253b174e", "fd96f15d547c4e155daeeef4253b174e"); const std::string payload_checksum = AcmReceiverBitExactnessOldApi::PlatformChecksum( payload_maybe_sse, payload_maybe_sse, "4bd846d0aa5656ecd5dfd85701a1b78c", "7efbfc9f8e3b4b2933ae2d01ab919028", "7efbfc9f8e3b4b2933ae2d01ab919028"); } // namespace // TODO(http://bugs.webrtc.org/12518): Enable the test after Opus has been // updated. TEST_F(AcmSenderBitExactnessOldApi, DISABLED_Opus_stereo_20ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960)); Run(audio_checksum, payload_checksum, 50, test::AcmReceiveTestOldApi::kStereoOutput); } // TODO(http://bugs.webrtc.org/12518): Enable the test after Opus has been // updated. TEST_F(AcmSenderBitExactnessNewApi, DISABLED_OpusFromFormat_stereo_20ms) { const auto config = AudioEncoderOpus::SdpToConfig( SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}})); ASSERT_TRUE(SetUpSender(kTestFileFakeStereo32kHz, 32000)); ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder( AudioEncoderOpus::MakeAudioEncoder(*config, 120), 120)); Run(audio_checksum, payload_checksum, 50, test::AcmReceiveTestOldApi::kStereoOutput); } // TODO(webrtc:8649): Disabled until the Encoder counterpart of // https://webrtc-review.googlesource.com/c/src/+/129768 lands. TEST_F(AcmSenderBitExactnessNewApi, DISABLED_OpusManyChannels) { constexpr int kNumChannels = 4; constexpr int kOpusPayloadType = 120; // Read a 4 channel file at 48kHz. ASSERT_TRUE(SetUpSender(kTestFileQuad48kHz, 48000)); const auto sdp_format = SdpAudioFormat("multiopus", 48000, kNumChannels, {{"channel_mapping", "0,1,2,3"}, {"coupled_streams", "2"}, {"num_streams", "2"}}); const auto encoder_config = AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format); ASSERT_TRUE(encoder_config.has_value()); ASSERT_NO_FATAL_FAILURE( SetUpTestExternalEncoder(AudioEncoderMultiChannelOpus::MakeAudioEncoder( *encoder_config, kOpusPayloadType), kOpusPayloadType)); const auto decoder_config = AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format); const auto opus_decoder = AudioDecoderMultiChannelOpus::MakeAudioDecoder(*decoder_config); rtc::scoped_refptr decoder_factory = new rtc::RefCountedObject( opus_decoder.get()); // Set up an EXTERNAL DECODER to parse 4 channels. Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( // audio checksum "audio checksum check downstream|8051617907766bec5f4e4a4f7c6d5291", "8051617907766bec5f4e4a4f7c6d5291", "6183752a62dc1368f959eb3a8c93b846", "android arm64 audio checksum", "48bf1f3ca0b72f3c9cdfbe79956122b1"), // payload_checksum, AcmReceiverBitExactnessOldApi::PlatformChecksum( // payload checksum "payload checksum check downstream|b09c52e44b2bdd9a0809e3a5b1623a76", "b09c52e44b2bdd9a0809e3a5b1623a76", "2ea535ef60f7d0c9d89e3002d4c2124f", "android arm64 payload checksum", "e87995a80f50a0a735a230ca8b04a67d"), 50, test::AcmReceiveTestOldApi::kQuadOutput, decoder_factory); } // TODO(http://bugs.webrtc.org/12518): Enable the test after Opus has been // updated. TEST_F(AcmSenderBitExactnessNewApi, DISABLED_OpusFromFormat_stereo_20ms_voip) { auto config = AudioEncoderOpus::SdpToConfig( SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}})); // If not set, default will be kAudio in case of stereo. config->application = AudioEncoderOpusConfig::ApplicationMode::kVoip; ASSERT_TRUE(SetUpSender(kTestFileFakeStereo32kHz, 32000)); ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder( AudioEncoderOpus::MakeAudioEncoder(*config, 120), 120)); const std::string audio_maybe_sse = "2d7e5797444f75e5bfeaffbd8c25176b" "|408d4bdc05a8c23e46c6ac06c5b917ee"; const std::string payload_maybe_sse = "b38b5584cfa7b6999b2e8e996c950c88" "|eb0752ce1b6f2436fefc2e19bd084fb5"; Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( audio_maybe_sse, audio_maybe_sse, "f1cefe107ffdced7694d7f735342adf3", "3b1bfe5dd8ed16ee5b04b93a5b5e7e48", "3b1bfe5dd8ed16ee5b04b93a5b5e7e48"), AcmReceiverBitExactnessOldApi::PlatformChecksum( payload_maybe_sse, payload_maybe_sse, "5e79a2f51c633fe145b6c10ae198d1aa", "e730050cb304d54d853fd285ab0424fa", "e730050cb304d54d853fd285ab0424fa"), 50, test::AcmReceiveTestOldApi::kStereoOutput); } // This test is for verifying the SetBitRate function. The bitrate is changed at // the beginning, and the number of generated bytes are checked. class AcmSetBitRateTest : public ::testing::Test { protected: static const int kTestDurationMs = 1000; // 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::AcmSendTestOldApi( audio_source_.get(), kSourceRateHz, kTestDurationMs)); return send_test_.get(); } // Registers a send codec in the test::AcmSendTest object. Returns true on // success, false on failure. virtual bool RegisterSendCodec(const char* payload_name, int sampling_freq_hz, int channels, int payload_type, int frame_size_samples, int frame_size_rtp_timestamps) { return send_test_->RegisterCodec(payload_name, sampling_freq_hz, channels, payload_type, frame_size_samples); } void RegisterExternalSendCodec( std::unique_ptr external_speech_encoder, int payload_type) { send_test_->RegisterExternalCodec(std::move(external_speech_encoder)); } void RunInner(int min_expected_total_bits, int max_expected_total_bits) { int nr_bytes = 0; while (std::unique_ptr next_packet = send_test_->NextPacket()) { nr_bytes += rtc::checked_cast(next_packet->payload_length_bytes()); } EXPECT_LE(min_expected_total_bits, nr_bytes * 8); EXPECT_GE(max_expected_total_bits, nr_bytes * 8); } void SetUpTest(const char* codec_name, int codec_sample_rate_hz, int channels, int payload_type, int codec_frame_size_samples, int codec_frame_size_rtp_timestamps) { ASSERT_TRUE(SetUpSender()); ASSERT_TRUE(RegisterSendCodec(codec_name, codec_sample_rate_hz, channels, payload_type, codec_frame_size_samples, codec_frame_size_rtp_timestamps)); } std::unique_ptr send_test_; std::unique_ptr audio_source_; }; class AcmSetBitRateNewApi : public AcmSetBitRateTest { protected: // Runs the test. SetUpSender() must have been called and a codec must be set // up before calling this method. void Run(int min_expected_total_bits, int max_expected_total_bits) { RunInner(min_expected_total_bits, max_expected_total_bits); } }; TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_10kbps) { const auto config = AudioEncoderOpus::SdpToConfig( SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "10000"}})); ASSERT_TRUE(SetUpSender()); RegisterExternalSendCodec(AudioEncoderOpus::MakeAudioEncoder(*config, 107), 107); RunInner(7000, 12000); } TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) { const auto config = AudioEncoderOpus::SdpToConfig( SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "50000"}})); ASSERT_TRUE(SetUpSender()); RegisterExternalSendCodec(AudioEncoderOpus::MakeAudioEncoder(*config, 107), 107); RunInner(40000, 60000); } // Verify that it works when the data to send is mono and the encoder is set to // send surround audio. TEST_F(AudioCodingModuleTestOldApi, SendingMultiChannelForMonoInput) { constexpr int kSampleRateHz = 48000; constexpr int kSamplesPerChannel = kSampleRateHz * 10 / 1000; audio_format_ = SdpAudioFormat({"multiopus", kSampleRateHz, 6, {{"minptime", "10"}, {"useinbandfec", "1"}, {"channel_mapping", "0,4,1,2,3,5"}, {"num_streams", "4"}, {"coupled_streams", "2"}}}); RegisterCodec(); input_frame_.sample_rate_hz_ = kSampleRateHz; input_frame_.num_channels_ = 1; input_frame_.samples_per_channel_ = kSamplesPerChannel; for (size_t k = 0; k < 10; ++k) { ASSERT_GE(acm_->Add10MsData(input_frame_), 0); input_frame_.timestamp_ += kSamplesPerChannel; } } // Verify that it works when the data to send is stereo and the encoder is set // to send surround audio. TEST_F(AudioCodingModuleTestOldApi, SendingMultiChannelForStereoInput) { constexpr int kSampleRateHz = 48000; constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000; audio_format_ = SdpAudioFormat({"multiopus", kSampleRateHz, 6, {{"minptime", "10"}, {"useinbandfec", "1"}, {"channel_mapping", "0,4,1,2,3,5"}, {"num_streams", "4"}, {"coupled_streams", "2"}}}); RegisterCodec(); input_frame_.sample_rate_hz_ = kSampleRateHz; input_frame_.num_channels_ = 2; input_frame_.samples_per_channel_ = kSamplesPerChannel; for (size_t k = 0; k < 10; ++k) { ASSERT_GE(acm_->Add10MsData(input_frame_), 0); input_frame_.timestamp_ += kSamplesPerChannel; } } // Verify that it works when the data to send is mono and the encoder is set to // send stereo audio. TEST_F(AudioCodingModuleTestOldApi, SendingStereoForMonoInput) { constexpr int kSampleRateHz = 48000; constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000; audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 2); RegisterCodec(); input_frame_.sample_rate_hz_ = kSampleRateHz; input_frame_.num_channels_ = 1; input_frame_.samples_per_channel_ = kSamplesPerChannel; for (size_t k = 0; k < 10; ++k) { ASSERT_GE(acm_->Add10MsData(input_frame_), 0); input_frame_.timestamp_ += kSamplesPerChannel; } } // Verify that it works when the data to send is stereo and the encoder is set // to send mono audio. TEST_F(AudioCodingModuleTestOldApi, SendingMonoForStereoInput) { constexpr int kSampleRateHz = 48000; constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000; audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 1); RegisterCodec(); input_frame_.sample_rate_hz_ = kSampleRateHz; input_frame_.num_channels_ = 1; input_frame_.samples_per_channel_ = kSamplesPerChannel; for (size_t k = 0; k < 10; ++k) { ASSERT_GE(acm_->Add10MsData(input_frame_), 0); input_frame_.timestamp_ += kSamplesPerChannel; } } // The result on the Android platforms is inconsistent for this test case. // On android_rel the result is different from android and android arm64 rel. #if defined(WEBRTC_ANDROID) #define MAYBE_OpusFromFormat_48khz_20ms_100kbps \ DISABLED_OpusFromFormat_48khz_20ms_100kbps #else #define MAYBE_OpusFromFormat_48khz_20ms_100kbps \ OpusFromFormat_48khz_20ms_100kbps #endif TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_100kbps) { const auto config = AudioEncoderOpus::SdpToConfig( SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "100000"}})); ASSERT_TRUE(SetUpSender()); RegisterExternalSendCodec(AudioEncoderOpus::MakeAudioEncoder(*config, 107), 107); RunInner(80000, 120000); } TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) { AudioEncoderPcmU::Config config; config.frame_size_ms = 20; config.num_channels = 1; config.payload_type = 0; AudioEncoderPcmU encoder(config); auto mock_encoder = std::make_unique(); // Set expectations on the mock encoder and also delegate the calls to the // real encoder. EXPECT_CALL(*mock_encoder, SampleRateHz()) .Times(AtLeast(1)) .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SampleRateHz)); EXPECT_CALL(*mock_encoder, NumChannels()) .Times(AtLeast(1)) .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::NumChannels)); EXPECT_CALL(*mock_encoder, RtpTimestampRateHz()) .Times(AtLeast(1)) .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::RtpTimestampRateHz)); EXPECT_CALL(*mock_encoder, Num10MsFramesInNextPacket()) .Times(AtLeast(1)) .WillRepeatedly( Invoke(&encoder, &AudioEncoderPcmU::Num10MsFramesInNextPacket)); EXPECT_CALL(*mock_encoder, GetTargetBitrate()) .Times(AtLeast(1)) .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::GetTargetBitrate)); EXPECT_CALL(*mock_encoder, EncodeImpl(_, _, _)) .Times(AtLeast(1)) .WillRepeatedly(Invoke( &encoder, static_cast, rtc::Buffer*)>( &AudioEncoderPcmU::Encode))); ASSERT_TRUE(SetUpSender(kTestFileMono32kHz, 32000)); ASSERT_NO_FATAL_FAILURE( SetUpTestExternalEncoder(std::move(mock_encoder), config.payload_type)); Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9", 50, test::AcmReceiveTestOldApi::kMonoOutput); } // This test fixture is implemented to run ACM and change the desired output // frequency during the call. The input packets are simply PCM16b-wb encoded // payloads with a constant value of |kSampleValue|. The test fixture itself // acts as PacketSource in between the receive test class and the constant- // payload packet source class. The output is both written to file, and analyzed // in this test fixture. class AcmSwitchingOutputFrequencyOldApi : public ::testing::Test, public test::PacketSource, public test::AudioSink { protected: static const size_t kTestNumPackets = 50; static const int kEncodedSampleRateHz = 16000; static const size_t kPayloadLenSamples = 30 * kEncodedSampleRateHz / 1000; static const int kPayloadType = 108; // Default payload type for PCM16b-wb. AcmSwitchingOutputFrequencyOldApi() : first_output_(true), num_packets_(0), packet_source_(kPayloadLenSamples, kSampleValue, kEncodedSampleRateHz, kPayloadType), output_freq_2_(0), has_toggled_(false) {} void Run(int output_freq_1, int output_freq_2, int toggle_period_ms) { // Set up the receiver used to decode the packets and verify the decoded // output. 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 WriteArray method in // this class. test::AudioSinkFork output(this, &output_file); test::AcmReceiveTestToggleOutputFreqOldApi receive_test( this, &output, output_freq_1, output_freq_2, toggle_period_ms, test::AcmReceiveTestOldApi::kMonoOutput); ASSERT_NO_FATAL_FAILURE(receive_test.RegisterDefaultCodecs()); output_freq_2_ = output_freq_2; // This is where the actual test is executed. receive_test.Run(); // Delete output file. remove(output_file_name.c_str()); } // Inherited from test::PacketSource. std::unique_ptr NextPacket() override { // Check if it is time to terminate the test. The packet source is of type // ConstantPcmPacketSource, which is infinite, so we must end the test // "manually". if (num_packets_++ > kTestNumPackets) { EXPECT_TRUE(has_toggled_); return NULL; // Test ended. } // Get the next packet from the source. return packet_source_.NextPacket(); } // Inherited from test::AudioSink. bool WriteArray(const int16_t* audio, size_t num_samples) override { // Skip checking the first output frame, since it has a number of zeros // due to how NetEq is initialized. if (first_output_) { first_output_ = false; return true; } for (size_t i = 0; i < num_samples; ++i) { EXPECT_EQ(kSampleValue, audio[i]); } if (num_samples == static_cast(output_freq_2_ / 100)) // Size of 10 ms frame. has_toggled_ = true; // The return value does not say if the values match the expectation, just // that the method could process the samples. return true; } const int16_t kSampleValue = 1000; bool first_output_; size_t num_packets_; test::ConstantPcmPacketSource packet_source_; int output_freq_2_; bool has_toggled_; }; TEST_F(AcmSwitchingOutputFrequencyOldApi, TestWithoutToggling) { Run(16000, 16000, 1000); } TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle16KhzTo32Khz) { Run(16000, 32000, 1000); } TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle32KhzTo16Khz) { Run(32000, 16000, 1000); } TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle16KhzTo8Khz) { Run(16000, 8000, 1000); } TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) { Run(8000, 16000, 1000); } #endif } // namespace webrtc