| /* |
| * 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/md5digest.h" |
| #include "webrtc/base/thread_annotations.h" |
| #include "webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h" |
| #include "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.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/constant_pcm_packet_source.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/compile_assert.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/event_wrapper.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.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 int 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()) {} |
| |
| virtual int32_t SendData( |
| FrameType frame_type, |
| uint8_t payload_type, |
| uint32_t timestamp, |
| const uint8_t* payload_data, |
| uint16_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 scoped_ptr<CriticalSectionWrapper> crit_sect_; |
| }; |
| |
| class AudioCodingModuleTestOldApi : public ::testing::Test { |
| protected: |
| AudioCodingModuleTestOldApi() |
| : id_(1), |
| rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)), |
| clock_(Clock::GetRealTimeClock()) {} |
| |
| ~AudioCodingModuleTestOldApi() {} |
| |
| void TearDown() {} |
| |
| void SetUp() { |
| acm_.reset(AudioCodingModule::Create(id_, clock_)); |
| |
| RegisterCodec(); |
| |
| 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. |
| COMPILE_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])); |
| |
| ASSERT_EQ(0, acm_->RegisterTransportCallback(&packet_cb_)); |
| } |
| |
| virtual void RegisterCodec() { |
| AudioCodingModule::Codec("L16", &codec_, kSampleRateHz, 1); |
| codec_.pltype = kPayloadType; |
| |
| // Register L16 codec in ACM. |
| ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_)); |
| ASSERT_EQ(0, acm_->RegisterSendCodec(codec_)); |
| } |
| |
| 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; |
| ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame)); |
| } |
| |
| virtual void InsertAudio() { |
| ASSERT_EQ(0, acm_->Add10MsData(input_frame_)); |
| input_frame_.timestamp_ += kNumSamples10ms; |
| } |
| |
| virtual void Encode() { |
| int32_t encoded_bytes = acm_->Process(); |
| // Expect to get one packet with two bytes per sample, or no packet at all, |
| // depending on how many 10 ms blocks go into |codec_.pacsize|. |
| EXPECT_TRUE(encoded_bytes == 2 * codec_.pacsize || encoded_bytes == 0); |
| } |
| |
| const int id_; |
| scoped_ptr<RtpUtility> rtp_utility_; |
| scoped_ptr<AudioCodingModule> acm_; |
| PacketizationCallbackStub packet_cb_; |
| WebRtcRTPHeader rtp_header_; |
| AudioFrame input_frame_; |
| CodecInst codec_; |
| Clock* clock_; |
| }; |
| |
| // Check if the statistics are initialized correctly. Before any call to ACM |
| // all fields have to be zero. |
| TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(InitializedToZero)) { |
| 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(AudioCodingModuleTestOldApi, |
| DISABLED_ON_ANDROID(SilenceGeneratorCalled)) { |
| AudioDecodingCallStats stats; |
| const int kInitialDelay = 100; |
| |
| acm_->SetInitialPlayoutDelay(kInitialDelay); |
| |
| 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(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(NetEqCalls)) { |
| 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(AudioCodingModuleTestOldApi, VerifyOutputFrame) { |
| AudioFrame audio_frame; |
| const int kSampleRateHz = 32000; |
| EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame)); |
| EXPECT_EQ(id_, audio_frame.id_); |
| EXPECT_EQ(0u, audio_frame.timestamp_); |
| EXPECT_GT(audio_frame.num_channels_, 0); |
| EXPECT_EQ(kSampleRateHz / 100, audio_frame.samples_per_channel_); |
| EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_); |
| } |
| |
| TEST_F(AudioCodingModuleTestOldApi, FailOnZeroDesiredFrequency) { |
| AudioFrame audio_frame; |
| EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame)); |
| } |
| |
| // 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_thread_(ThreadWrapper::CreateThread(CbSendThread, |
| this, |
| kRealtimePriority, |
| "send")), |
| insert_packet_thread_(ThreadWrapper::CreateThread(CbInsertPacketThread, |
| this, |
| kRealtimePriority, |
| "insert_packet")), |
| pull_audio_thread_(ThreadWrapper::CreateThread(CbPullAudioThread, |
| this, |
| kRealtimePriority, |
| "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)) { |
| clock_ = fake_clock_.get(); |
| } |
| |
| void SetUp() { |
| AudioCodingModuleTestOldApi::SetUp(); |
| StartThreads(); |
| } |
| |
| void StartThreads() { |
| unsigned int thread_id = 0; |
| ASSERT_TRUE(send_thread_->Start(thread_id)); |
| ASSERT_TRUE(insert_packet_thread_->Start(thread_id)); |
| ASSERT_TRUE(pull_audio_thread_->Start(thread_id)); |
| } |
| |
| void TearDown() { |
| AudioCodingModuleTestOldApi::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<AudioCodingModuleMtTestOldApi*>(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(); |
| Encode(); |
| if (TestDone()) { |
| test_complete_->Set(); |
| } |
| return true; |
| } |
| |
| static bool CbInsertPacketThread(void* context) { |
| return reinterpret_cast<AudioCodingModuleMtTestOldApi*>(context) |
| ->CbInsertPacketImpl(); |
| } |
| |
| bool CbInsertPacketImpl() { |
| SleepMs(1); |
| { |
| CriticalSectionScoped lock(crit_sect_.get()); |
| if (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<AudioCodingModuleMtTestOldApi*>(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_ < 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; |
| } |
| |
| scoped_ptr<ThreadWrapper> send_thread_; |
| scoped_ptr<ThreadWrapper> insert_packet_thread_; |
| scoped_ptr<ThreadWrapper> pull_audio_thread_; |
| const scoped_ptr<EventWrapper> test_complete_; |
| int send_count_; |
| int insert_packet_count_; |
| int pull_audio_count_ GUARDED_BY(crit_sect_); |
| const scoped_ptr<CriticalSectionWrapper> crit_sect_; |
| int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_); |
| scoped_ptr<SimulatedClock> fake_clock_; |
| }; |
| |
| TEST_F(AudioCodingModuleMtTestOldApi, 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 AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi { |
| protected: |
| static const int kNumPackets = 500; |
| static const int kNumPullCalls = 500; |
| |
| AcmIsacMtTestOldApi() |
| : AudioCodingModuleMtTestOldApi(), last_packet_number_(0) {} |
| |
| ~AcmIsacMtTestOldApi() {} |
| |
| void SetUp() { |
| 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); |
| |
| // Generate one packet to have something to insert. |
| int loop_counter = 0; |
| while (packet_cb_.last_payload_len_bytes() == 0) { |
| InsertAudio(); |
| Encode(); |
| 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(); |
| } |
| |
| virtual void RegisterCodec() { |
| COMPILE_ASSERT(kSampleRateHz == 16000, test_designed_for_isac_16khz); |
| AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1); |
| codec_.pltype = kPayloadType; |
| |
| // Register iSAC codec in ACM, effectively unregistering the PCM16B codec |
| // registered in AudioCodingModuleTestOldApi::SetUp(); |
| ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_)); |
| ASSERT_EQ(0, acm_->RegisterSendCodec(codec_)); |
| } |
| |
| void InsertPacket() { |
| 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() { |
| memcpy(input_frame_.data_, audio_loop_.GetNextBlock(), kNumSamples10ms); |
| AudioCodingModuleTestOldApi::InsertAudio(); |
| } |
| |
| void Encode() { ASSERT_GE(acm_->Process(), 0); } |
| |
| // This method is the same as AudioCodingModuleMtTestOldApi::TestDone(), but |
| // here it is using the constants defined in this class (i.e., shorter test |
| // run). |
| 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; |
| } |
| |
| int last_packet_number_; |
| std::vector<uint8_t> last_payload_vec_; |
| test::AudioLoop audio_loop_; |
| }; |
| |
| TEST_F(AcmIsacMtTestOldApi, DoTest) { |
| EXPECT_EQ(kEventSignaled, RunTest()); |
| } |
| |
| class AcmReceiverBitExactnessOldApi : 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"); |
| 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::AcmReceiveTestOldApi test( |
| packet_source.get(), |
| &output, |
| output_freq_hz, |
| test::AcmReceiveTestOldApi::kArbitraryChannels); |
| ASSERT_NO_FATAL_FAILURE(test.RegisterNetEqTestCodecs()); |
| test.Run(); |
| |
| std::string checksum_string = checksum.Finish(); |
| EXPECT_EQ(checksum_ref, checksum_string); |
| } |
| }; |
| |
| TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) { |
| Run(8000, |
| PlatformChecksum("bd6f8d9602cd82444ea2539e674df747", |
| "6ac89c7145072c26bfeba602cd661afb", |
| "8a8440f5511eb729221b9aac25cda3a0")); |
| } |
| |
| TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) { |
| Run(16000, |
| PlatformChecksum("a39bc6ee0c4eb15f3ad2f43cebcc571d", |
| "3e888eb04f57db2c6ef952fe64f17fe6", |
| "7be583092c5adbcb0f6cd66eca20ea63")); |
| } |
| |
| TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) { |
| Run(32000, |
| PlatformChecksum("80964572aaa2dc92f9e34896dd3802b3", |
| "aeca37e963310f5b6552b7edea23c2f1", |
| "3a84188abe9fca25fedd6034760f3e22")); |
| } |
| |
| TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) { |
| Run(48000, |
| PlatformChecksum("8aacde91f390e0d5a9c2ed571a25fd37", |
| "76b9e99e0a3998aa28355e7a2bd836f7", |
| "89b4b19bdb4de40f1d88302ef8cb9f9b")); |
| } |
| |
| // 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) {} |
| |
| // 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() != 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); |
| } |
| |
| // 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) { |
| // 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::AcmReceiveTestOldApi 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(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)); |
| } |
| |
| scoped_ptr<test::AcmSendTestOldApi> send_test_; |
| 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_; |
| }; |
| |
| TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) { |
| ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480)); |
| Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( |
| "c7e5bdadfa2871df95639fcc297cf23d", |
| "0499ca260390769b3172136faad925b9", |
| "0b58f9eeee43d5891f5f6c75e77984a3"), |
| AcmReceiverBitExactnessOldApi::PlatformChecksum( |
| "d42cb5195463da26c8129bbfe73a22e6", |
| "83de248aea9c3c2bd680b6952401b4ca", |
| "3c79f16f34218271f3dca4e2b1dfe1bb"), |
| 33, |
| test::AcmReceiveTestOldApi::kMonoOutput); |
| } |
| |
| TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) { |
| ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960)); |
| Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( |
| "14d63c5f08127d280e722e3191b73bdd", |
| "8da003e16c5371af2dc2be79a50f9076", |
| "1ad29139a04782a33daad8c2b9b35875"), |
| AcmReceiverBitExactnessOldApi::PlatformChecksum( |
| "ebe04a819d3a9d83a83a17f271e1139a", |
| "97aeef98553b5a4b5a68f8b716e8eaf0", |
| "9e0a0ab743ad987b55b8e14802769c56"), |
| 16, |
| test::AcmReceiveTestOldApi::kMonoOutput); |
| } |
| |
| TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IsacSwb30ms)) { |
| ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960)); |
| Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( |
| "98d960600eb4ddb3fcbe11f5057ddfd7", |
| "", |
| "2f6dfe142f735f1d96f6bd86d2526f42"), |
| AcmReceiverBitExactnessOldApi::PlatformChecksum( |
| "cc9d2d86a71d6f99f97680a5c27e2762", |
| "", |
| "7b214fc3a5e33d68bf30e77969371f31"), |
| 33, |
| test::AcmReceiveTestOldApi::kMonoOutput); |
| } |
| |
| TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) { |
| ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80)); |
| Run("de4a98e1406f8b798d99cd0704e862e2", |
| "c1edd36339ce0326cc4550041ad719a0", |
| 100, |
| test::AcmReceiveTestOldApi::kMonoOutput); |
| } |
| |
| TEST_F(AcmSenderBitExactnessOldApi, Pcm16_16000khz_10ms) { |
| ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 1, 108, 160, 160)); |
| Run("ae646d7b68384a1269cc080dd4501916", |
| "ad786526383178b08d80d6eee06e9bad", |
| 100, |
| test::AcmReceiveTestOldApi::kMonoOutput); |
| } |
| |
| TEST_F(AcmSenderBitExactnessOldApi, Pcm16_32000khz_10ms) { |
| ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 1, 109, 320, 320)); |
| Run("7fe325e8fbaf755e3c5df0b11a4774fb", |
| "5ef82ea885e922263606c6fdbc49f651", |
| 100, |
| test::AcmReceiveTestOldApi::kMonoOutput); |
| } |
| |
| TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_8000khz_10ms) { |
| ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 2, 111, 80, 80)); |
| Run("fb263b74e7ac3de915474d77e4744ceb", |
| "62ce5adb0d4965d0a52ec98ae7f98974", |
| 100, |
| test::AcmReceiveTestOldApi::kStereoOutput); |
| } |
| |
| TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_16000khz_10ms) { |
| ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 2, 112, 160, 160)); |
| Run("d09e9239553649d7ac93e19d304281fd", |
| "41ca8edac4b8c71cd54fd9f25ec14870", |
| 100, |
| test::AcmReceiveTestOldApi::kStereoOutput); |
| } |
| |
| TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_32000khz_10ms) { |
| ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 2, 113, 320, 320)); |
| Run("5f025d4f390982cc26b3d92fe02e3044", |
| "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); |
| } |
| |
| TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(Ilbc_30ms)) { |
| ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240)); |
| Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( |
| "7b6ec10910debd9af08011d3ed5249f7", |
| "android_audio", |
| "7b6ec10910debd9af08011d3ed5249f7"), |
| AcmReceiverBitExactnessOldApi::PlatformChecksum( |
| "cfae2e9f6aba96e145f2bcdd5050ce78", |
| "android_payload", |
| "cfae2e9f6aba96e145f2bcdd5050ce78"), |
| 33, |
| test::AcmReceiveTestOldApi::kMonoOutput); |
| } |
| |
| TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(G722_20ms)) { |
| ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160)); |
| Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( |
| "7d759436f2533582950d148b5161a36c", |
| "android_audio", |
| "7d759436f2533582950d148b5161a36c"), |
| AcmReceiverBitExactnessOldApi::PlatformChecksum( |
| "fc68a87e1380614e658087cb35d5ca10", |
| "android_payload", |
| "fc68a87e1380614e658087cb35d5ca10"), |
| 50, |
| test::AcmReceiveTestOldApi::kMonoOutput); |
| } |
| |
| TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(G722_stereo_20ms)) { |
| ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160)); |
| Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( |
| "7190ee718ab3d80eca181e5f7140c210", |
| "android_audio", |
| "7190ee718ab3d80eca181e5f7140c210"), |
| AcmReceiverBitExactnessOldApi::PlatformChecksum( |
| "66516152eeaa1e650ad94ff85f668dac", |
| "android_payload", |
| "66516152eeaa1e650ad94ff85f668dac"), |
| 50, |
| test::AcmReceiveTestOldApi::kStereoOutput); |
| } |
| |
| TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) { |
| ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960)); |
| Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( |
| "855041f2490b887302bce9d544731849", |
| "1e1a0fce893fef2d66886a7f09e2ebce", |
| "855041f2490b887302bce9d544731849"), |
| AcmReceiverBitExactnessOldApi::PlatformChecksum( |
| "d781cce1ab986b618d0da87226cdde30", |
| "1a1fe04dd12e755949987c8d729fb3e0", |
| "d781cce1ab986b618d0da87226cdde30"), |
| 50, |
| test::AcmReceiveTestOldApi::kStereoOutput); |
| } |
| |
| // 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(); |
| } |
| |
| // Inherited from test::PacketSource. |
| test::Packet* 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) { |
| // 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<size_t>(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); |
| } |
| } // namespace webrtc |
