diff options
Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq_unittest.cc')
| -rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq_unittest.cc | 778 |
1 files changed, 0 insertions, 778 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq_unittest.cc deleted file mode 100644 index c37f8990a8b..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq_unittest.cc +++ /dev/null @@ -1,778 +0,0 @@ -/* - * Copyright (c) 2011 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. - */ - -/* - * This file includes unit tests for NetEQ. - */ - -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" - -#include <stdlib.h> -#include <string.h> // memset - -#include <set> -#include <sstream> -#include <string> -#include <vector> - -#include "gtest/gtest.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h" -#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h" -#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h" -#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h" -#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h" -#include "webrtc/modules/audio_coding/neteq4/tools/input_audio_file.h" -#include "webrtc/test/testsupport/fileutils.h" -#include "webrtc/typedefs.h" - -namespace webrtc { - -class RefFiles { - public: - RefFiles(const std::string& input_file, const std::string& output_file); - ~RefFiles(); - template<class T> void ProcessReference(const T& test_results); - template<typename T, size_t n> void ProcessReference( - const T (&test_results)[n], - size_t length); - template<typename T, size_t n> void WriteToFile( - const T (&test_results)[n], - size_t length); - template<typename T, size_t n> void ReadFromFileAndCompare( - const T (&test_results)[n], - size_t length); - void WriteToFile(const WebRtcNetEQ_NetworkStatistics& stats); - void ReadFromFileAndCompare(const WebRtcNetEQ_NetworkStatistics& stats); - void WriteToFile(const WebRtcNetEQ_RTCPStat& stats); - void ReadFromFileAndCompare(const WebRtcNetEQ_RTCPStat& stats); - - FILE* input_fp_; - FILE* output_fp_; -}; - -RefFiles::RefFiles(const std::string &input_file, - const std::string &output_file) - : input_fp_(NULL), - output_fp_(NULL) { - if (!input_file.empty()) { - input_fp_ = fopen(input_file.c_str(), "rb"); - EXPECT_TRUE(input_fp_ != NULL); - } - if (!output_file.empty()) { - output_fp_ = fopen(output_file.c_str(), "wb"); - EXPECT_TRUE(output_fp_ != NULL); - } -} - -RefFiles::~RefFiles() { - if (input_fp_) { - EXPECT_EQ(EOF, fgetc(input_fp_)); // Make sure that we reached the end. - fclose(input_fp_); - } - if (output_fp_) fclose(output_fp_); -} - -template<class T> -void RefFiles::ProcessReference(const T& test_results) { - WriteToFile(test_results); - ReadFromFileAndCompare(test_results); -} - -template<typename T, size_t n> -void RefFiles::ProcessReference(const T (&test_results)[n], size_t length) { - WriteToFile(test_results, length); - ReadFromFileAndCompare(test_results, length); -} - -template<typename T, size_t n> -void RefFiles::WriteToFile(const T (&test_results)[n], size_t length) { - if (output_fp_) { - ASSERT_EQ(length, fwrite(&test_results, sizeof(T), length, output_fp_)); - } -} - -template<typename T, size_t n> -void RefFiles::ReadFromFileAndCompare(const T (&test_results)[n], - size_t length) { - if (input_fp_) { - // Read from ref file. - T* ref = new T[length]; - ASSERT_EQ(length, fread(ref, sizeof(T), length, input_fp_)); - // Compare - ASSERT_EQ(0, memcmp(&test_results, ref, sizeof(T) * length)); - delete [] ref; - } -} - -void RefFiles::WriteToFile(const WebRtcNetEQ_NetworkStatistics& stats) { - if (output_fp_) { - ASSERT_EQ(1u, fwrite(&stats, sizeof(WebRtcNetEQ_NetworkStatistics), 1, - output_fp_)); - } -} - -void RefFiles::ReadFromFileAndCompare( - const WebRtcNetEQ_NetworkStatistics& stats) { - if (input_fp_) { - // Read from ref file. - size_t stat_size = sizeof(WebRtcNetEQ_NetworkStatistics); - WebRtcNetEQ_NetworkStatistics ref_stats; - ASSERT_EQ(1u, fread(&ref_stats, stat_size, 1, input_fp_)); - // Compare - EXPECT_EQ(0, memcmp(&stats, &ref_stats, stat_size)); - } -} - -void RefFiles::WriteToFile(const WebRtcNetEQ_RTCPStat& stats) { - if (output_fp_) { - ASSERT_EQ(1u, fwrite(&(stats.fraction_lost), sizeof(stats.fraction_lost), 1, - output_fp_)); - ASSERT_EQ(1u, fwrite(&(stats.cum_lost), sizeof(stats.cum_lost), 1, - output_fp_)); - ASSERT_EQ(1u, fwrite(&(stats.ext_max), sizeof(stats.ext_max), 1, - output_fp_)); - ASSERT_EQ(1u, fwrite(&(stats.jitter), sizeof(stats.jitter), 1, - output_fp_)); - } -} - -void RefFiles::ReadFromFileAndCompare( - const WebRtcNetEQ_RTCPStat& stats) { - if (input_fp_) { - // Read from ref file. - WebRtcNetEQ_RTCPStat ref_stats; - ASSERT_EQ(1u, fread(&(ref_stats.fraction_lost), - sizeof(ref_stats.fraction_lost), 1, input_fp_)); - ASSERT_EQ(1u, fread(&(ref_stats.cum_lost), sizeof(ref_stats.cum_lost), 1, - input_fp_)); - ASSERT_EQ(1u, fread(&(ref_stats.ext_max), sizeof(ref_stats.ext_max), 1, - input_fp_)); - ASSERT_EQ(1u, fread(&(ref_stats.jitter), sizeof(ref_stats.jitter), 1, - input_fp_)); - // Compare - EXPECT_EQ(ref_stats.fraction_lost, stats.fraction_lost); - EXPECT_EQ(ref_stats.cum_lost, stats.cum_lost); - EXPECT_EQ(ref_stats.ext_max, stats.ext_max); - EXPECT_EQ(ref_stats.jitter, stats.jitter); - } -} - -class NetEqDecodingTest : public ::testing::Test { - protected: - // NetEQ must be polled for data once every 10 ms. Thus, neither of the - // constants below can be changed. - static const int kTimeStepMs = 10; - static const int kBlockSize8kHz = kTimeStepMs * 8; - static const int kBlockSize16kHz = kTimeStepMs * 16; - static const int kBlockSize32kHz = kTimeStepMs * 32; - static const int kMaxBlockSize = kBlockSize32kHz; - - NetEqDecodingTest(); - virtual void SetUp(); - virtual void TearDown(); - void SelectDecoders(WebRtcNetEQDecoder* used_codec); - void LoadDecoders(); - void OpenInputFile(const std::string &rtp_file); - void Process(NETEQTEST_RTPpacket* rtp_ptr, int16_t* out_len); - void DecodeAndCompare(const std::string &rtp_file, - const std::string &ref_file); - void DecodeAndCheckStats(const std::string &rtp_file, - const std::string &stat_ref_file, - const std::string &rtcp_ref_file); - static void PopulateRtpInfo(int frame_index, - int timestamp, - WebRtcNetEQ_RTPInfo* rtp_info); - static void PopulateCng(int frame_index, - int timestamp, - WebRtcNetEQ_RTPInfo* rtp_info, - uint8_t* payload, - int* payload_len); - void WrapTest(uint16_t start_seq_no, uint32_t start_timestamp, - const std::set<uint16_t>& drop_seq_numbers); - - NETEQTEST_NetEQClass* neteq_inst_; - std::vector<NETEQTEST_Decoder*> dec_; - FILE* rtp_fp_; - unsigned int sim_clock_; - int16_t out_data_[kMaxBlockSize]; -}; - -NetEqDecodingTest::NetEqDecodingTest() - : neteq_inst_(NULL), - rtp_fp_(NULL), - sim_clock_(0) { - memset(out_data_, 0, sizeof(out_data_)); -} - -void NetEqDecodingTest::SetUp() { - WebRtcNetEQDecoder usedCodec[kDecoderReservedEnd - 1]; - - SelectDecoders(usedCodec); - neteq_inst_ = new NETEQTEST_NetEQClass(usedCodec, dec_.size(), 8000, - kTCPLargeJitter); - ASSERT_TRUE(neteq_inst_); - LoadDecoders(); -} - -void NetEqDecodingTest::TearDown() { - if (neteq_inst_) - delete neteq_inst_; - for (size_t i = 0; i < dec_.size(); ++i) { - if (dec_[i]) - delete dec_[i]; - } - if (rtp_fp_) - fclose(rtp_fp_); -} - -void NetEqDecodingTest::SelectDecoders(WebRtcNetEQDecoder* used_codec) { - *used_codec++ = kDecoderPCMu; - dec_.push_back(new decoder_PCMU(0)); - *used_codec++ = kDecoderPCMa; - dec_.push_back(new decoder_PCMA(8)); - *used_codec++ = kDecoderILBC; - dec_.push_back(new decoder_ILBC(102)); - *used_codec++ = kDecoderISAC; - dec_.push_back(new decoder_iSAC(103)); - *used_codec++ = kDecoderISACswb; - dec_.push_back(new decoder_iSACSWB(104)); - *used_codec++ = kDecoderISACfb; - dec_.push_back(new decoder_iSACFB(105)); - *used_codec++ = kDecoderPCM16B; - dec_.push_back(new decoder_PCM16B_NB(93)); - *used_codec++ = kDecoderPCM16Bwb; - dec_.push_back(new decoder_PCM16B_WB(94)); - *used_codec++ = kDecoderPCM16Bswb32kHz; - dec_.push_back(new decoder_PCM16B_SWB32(95)); - *used_codec++ = kDecoderCNG; - dec_.push_back(new decoder_CNG(13, 8000)); - *used_codec++ = kDecoderCNG; - dec_.push_back(new decoder_CNG(98, 16000)); -} - -void NetEqDecodingTest::LoadDecoders() { - for (size_t i = 0; i < dec_.size(); ++i) { - ASSERT_EQ(0, dec_[i]->loadToNetEQ(*neteq_inst_)); - } -} - -void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) { - rtp_fp_ = fopen(rtp_file.c_str(), "rb"); - ASSERT_TRUE(rtp_fp_ != NULL); - ASSERT_EQ(0, NETEQTEST_RTPpacket::skipFileHeader(rtp_fp_)); -} - -void NetEqDecodingTest::Process(NETEQTEST_RTPpacket* rtp, int16_t* out_len) { - // Check if time to receive. - while ((sim_clock_ >= rtp->time()) && - (rtp->dataLen() >= 0)) { - if (rtp->dataLen() > 0) { - ASSERT_EQ(0, neteq_inst_->recIn(*rtp)); - } - // Get next packet. - ASSERT_NE(-1, rtp->readFromFile(rtp_fp_)); - } - - // RecOut - *out_len = neteq_inst_->recOut(out_data_); - ASSERT_TRUE((*out_len == kBlockSize8kHz) || - (*out_len == kBlockSize16kHz) || - (*out_len == kBlockSize32kHz)); - - // Increase time. - sim_clock_ += kTimeStepMs; -} - -void NetEqDecodingTest::DecodeAndCompare(const std::string &rtp_file, - const std::string &ref_file) { - OpenInputFile(rtp_file); - - std::string ref_out_file = ""; - if (ref_file.empty()) { - ref_out_file = webrtc::test::OutputPath() + "neteq_out.pcm"; - } - RefFiles ref_files(ref_file, ref_out_file); - - NETEQTEST_RTPpacket rtp; - ASSERT_GT(rtp.readFromFile(rtp_fp_), 0); - int i = 0; - while (rtp.dataLen() >= 0) { - std::ostringstream ss; - ss << "Lap number " << i++ << " in DecodeAndCompare while loop"; - SCOPED_TRACE(ss.str()); // Print out the parameter values on failure. - int16_t out_len; - ASSERT_NO_FATAL_FAILURE(Process(&rtp, &out_len)); - ASSERT_NO_FATAL_FAILURE(ref_files.ProcessReference(out_data_, out_len)); - } -} - -void NetEqDecodingTest::DecodeAndCheckStats(const std::string &rtp_file, - const std::string &stat_ref_file, - const std::string &rtcp_ref_file) { - OpenInputFile(rtp_file); - std::string stat_out_file = ""; - if (stat_ref_file.empty()) { - stat_out_file = webrtc::test::OutputPath() + - "neteq_network_stats.dat"; - } - RefFiles network_stat_files(stat_ref_file, stat_out_file); - - std::string rtcp_out_file = ""; - if (rtcp_ref_file.empty()) { - rtcp_out_file = webrtc::test::OutputPath() + - "neteq_rtcp_stats.dat"; - } - RefFiles rtcp_stat_files(rtcp_ref_file, rtcp_out_file); - - NETEQTEST_RTPpacket rtp; - ASSERT_GT(rtp.readFromFile(rtp_fp_), 0); - while (rtp.dataLen() >= 0) { - int16_t out_len; - Process(&rtp, &out_len); - - // Query the network statistics API once per second - if (sim_clock_ % 1000 == 0) { - // Process NetworkStatistics. - WebRtcNetEQ_NetworkStatistics network_stats; - ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(), - &network_stats)); - network_stat_files.ProcessReference(network_stats); - - // Process RTCPstat. - WebRtcNetEQ_RTCPStat rtcp_stats; - ASSERT_EQ(0, WebRtcNetEQ_GetRTCPStats(neteq_inst_->instance(), - &rtcp_stats)); - rtcp_stat_files.ProcessReference(rtcp_stats); - } - } -} - -void NetEqDecodingTest::PopulateRtpInfo(int frame_index, - int timestamp, - WebRtcNetEQ_RTPInfo* rtp_info) { - rtp_info->sequenceNumber = frame_index; - rtp_info->timeStamp = timestamp; - rtp_info->SSRC = 0x1234; // Just an arbitrary SSRC. - rtp_info->payloadType = 94; // PCM16b WB codec. - rtp_info->markerBit = 0; -} - -void NetEqDecodingTest::PopulateCng(int frame_index, - int timestamp, - WebRtcNetEQ_RTPInfo* rtp_info, - uint8_t* payload, - int* payload_len) { - rtp_info->sequenceNumber = frame_index; - rtp_info->timeStamp = timestamp; - rtp_info->SSRC = 0x1234; // Just an arbitrary SSRC. - rtp_info->payloadType = 98; // WB CNG. - rtp_info->markerBit = 0; - payload[0] = 64; // Noise level -64 dBov, quite arbitrarily chosen. - *payload_len = 1; // Only noise level, no spectral parameters. -} - -#if (defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS)) || defined(WEBRTC_ANDROID) -// Disabled for Windows 64-bit until webrtc:1460 is fixed. -#define MAYBE_TestBitExactness DISABLED_TestBitExactness -#else -#define MAYBE_TestBitExactness TestBitExactness -#endif - -TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) { - const std::string kInputRtpFile = webrtc::test::ProjectRootPath() + - "resources/audio_coding/neteq_universal.rtp"; -#if defined(_MSC_VER) && (_MSC_VER >= 1700) - // For Visual Studio 2012 and later, we will have to use the generic reference - // file, rather than the windows-specific one. - const std::string kInputRefFile = webrtc::test::ProjectRootPath() + - "resources/audio_coding/neteq_universal_ref.pcm"; -#else - const std::string kInputRefFile = - webrtc::test::ResourcePath("audio_coding/neteq_universal_ref", "pcm"); -#endif - DecodeAndCompare(kInputRtpFile, kInputRefFile); -} - -TEST_F(NetEqDecodingTest, TestNetworkStatistics) { - const std::string kInputRtpFile = webrtc::test::ProjectRootPath() + - "resources/audio_coding/neteq_universal.rtp"; -#if defined(_MSC_VER) && (_MSC_VER >= 1700) - // For Visual Studio 2012 and later, we will have to use the generic reference - // file, rather than the windows-specific one. - const std::string kNetworkStatRefFile = webrtc::test::ProjectRootPath() + - "resources/audio_coding/neteq_network_stats.dat"; -#else - const std::string kNetworkStatRefFile = - webrtc::test::ResourcePath("audio_coding/neteq_network_stats", "dat"); -#endif - const std::string kRtcpStatRefFile = - webrtc::test::ResourcePath("audio_coding/neteq_rtcp_stats", "dat"); - DecodeAndCheckStats(kInputRtpFile, kNetworkStatRefFile, kRtcpStatRefFile); -} - -TEST_F(NetEqDecodingTest, TestFrameWaitingTimeStatistics) { - // Use fax mode to avoid time-scaling. This is to simplify the testing of - // packet waiting times in the packet buffer. - ASSERT_EQ(0, - WebRtcNetEQ_SetPlayoutMode(neteq_inst_->instance(), kPlayoutFax)); - // Insert 30 dummy packets at once. Each packet contains 10 ms 16 kHz audio. - int num_frames = 30; - const int kSamples = 10 * 16; - const int kPayloadBytes = kSamples * 2; - for (int i = 0; i < num_frames; ++i) { - uint16_t payload[kSamples] = {0}; - WebRtcNetEQ_RTPInfo rtp_info; - rtp_info.sequenceNumber = i; - rtp_info.timeStamp = i * kSamples; - rtp_info.SSRC = 0x1234; // Just an arbitrary SSRC. - rtp_info.payloadType = 94; // PCM16b WB codec. - rtp_info.markerBit = 0; - ASSERT_EQ(0, WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), &rtp_info, - reinterpret_cast<uint8_t*>(payload), - kPayloadBytes, 0)); - } - // Pull out all data. - for (int i = 0; i < num_frames; ++i) { - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - } - const int kVecLen = 110; // More than kLenWaitingTimes in mcu.h. - int waiting_times[kVecLen]; - int len = WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(), - kVecLen, waiting_times); - EXPECT_EQ(num_frames, len); - // Since all frames are dumped into NetEQ at once, but pulled out with 10 ms - // spacing (per definition), we expect the delay to increase with 10 ms for - // each packet. - for (int i = 0; i < len; ++i) { - EXPECT_EQ((i + 1) * 10, waiting_times[i]); - } - - // Check statistics again and make sure it's been reset. - EXPECT_EQ(0, WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(), - kVecLen, waiting_times)); - - // Process > 100 frames, and make sure that that we get statistics - // only for 100 frames. Note the new SSRC, causing NetEQ to reset. - num_frames = 110; - for (int i = 0; i < num_frames; ++i) { - uint16_t payload[kSamples] = {0}; - WebRtcNetEQ_RTPInfo rtp_info; - rtp_info.sequenceNumber = i; - rtp_info.timeStamp = i * kSamples; - rtp_info.SSRC = 0x1235; // Just an arbitrary SSRC. - rtp_info.payloadType = 94; // PCM16b WB codec. - rtp_info.markerBit = 0; - ASSERT_EQ(0, WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), &rtp_info, - reinterpret_cast<uint8_t*>(payload), - kPayloadBytes, 0)); - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - } - - len = WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(), - kVecLen, waiting_times); - EXPECT_EQ(100, len); -} - -TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) { - const int kNumFrames = 3000; // Needed for convergence. - int frame_index = 0; - const int kSamples = 10 * 16; - const int kPayloadBytes = kSamples * 2; - while (frame_index < kNumFrames) { - // Insert one packet each time, except every 10th time where we insert two - // packets at once. This will create a negative clock-drift of approx. 10%. - int num_packets = (frame_index % 10 == 0 ? 2 : 1); - for (int n = 0; n < num_packets; ++n) { - uint8_t payload[kPayloadBytes] = {0}; - WebRtcNetEQ_RTPInfo rtp_info; - PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info); - ASSERT_EQ(0, - WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), - &rtp_info, - payload, - kPayloadBytes, 0)); - ++frame_index; - } - - // Pull out data once. - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - } - - WebRtcNetEQ_NetworkStatistics network_stats; - ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(), - &network_stats)); - EXPECT_EQ(-103196, network_stats.clockDriftPPM); -} - -TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) { - const int kNumFrames = 5000; // Needed for convergence. - int frame_index = 0; - const int kSamples = 10 * 16; - const int kPayloadBytes = kSamples * 2; - for (int i = 0; i < kNumFrames; ++i) { - // Insert one packet each time, except every 10th time where we don't insert - // any packet. This will create a positive clock-drift of approx. 11%. - int num_packets = (i % 10 == 9 ? 0 : 1); - for (int n = 0; n < num_packets; ++n) { - uint8_t payload[kPayloadBytes] = {0}; - WebRtcNetEQ_RTPInfo rtp_info; - PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info); - ASSERT_EQ(0, - WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), - &rtp_info, - payload, - kPayloadBytes, 0)); - ++frame_index; - } - - // Pull out data once. - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - } - - WebRtcNetEQ_NetworkStatistics network_stats; - ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(), - &network_stats)); - EXPECT_EQ(110946, network_stats.clockDriftPPM); -} - -TEST_F(NetEqDecodingTest, LongCngWithClockDrift) { - uint16_t seq_no = 0; - uint32_t timestamp = 0; - const int kFrameSizeMs = 30; - const int kSamples = kFrameSizeMs * 16; - const int kPayloadBytes = kSamples * 2; - // Apply a clock drift of -25 ms / s (sender faster than receiver). - const double kDriftFactor = 1000.0 / (1000.0 + 25.0); - double next_input_time_ms = 0.0; - double t_ms; - - // Insert speech for 5 seconds. - const int kSpeechDurationMs = 5000; - for (t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) { - // Each turn in this for loop is 10 ms. - while (next_input_time_ms <= t_ms) { - // Insert one 30 ms speech frame. - uint8_t payload[kPayloadBytes] = {0}; - WebRtcNetEQ_RTPInfo rtp_info; - PopulateRtpInfo(seq_no, timestamp, &rtp_info); - ASSERT_EQ(0, - WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), - &rtp_info, - payload, - kPayloadBytes, 0)); - ++seq_no; - timestamp += kSamples; - next_input_time_ms += static_cast<double>(kFrameSizeMs) * kDriftFactor; - } - // Pull out data once. - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - } - - EXPECT_EQ(kOutputNormal, neteq_inst_->getOutputType()); - int32_t delay_before = timestamp - neteq_inst_->getSpeechTimeStamp(); - - // Insert CNG for 1 minute (= 60000 ms). - const int kCngPeriodMs = 100; - const int kCngPeriodSamples = kCngPeriodMs * 16; // Period in 16 kHz samples. - const int kCngDurationMs = 60000; - for (; t_ms < kSpeechDurationMs + kCngDurationMs; t_ms += 10) { - // Each turn in this for loop is 10 ms. - while (next_input_time_ms <= t_ms) { - // Insert one CNG frame each 100 ms. - uint8_t payload[kPayloadBytes]; - int payload_len; - WebRtcNetEQ_RTPInfo rtp_info; - PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len); - ASSERT_EQ(0, - WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), - &rtp_info, - payload, - payload_len, 0)); - ++seq_no; - timestamp += kCngPeriodSamples; - next_input_time_ms += static_cast<double>(kCngPeriodMs) * kDriftFactor; - } - // Pull out data once. - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - } - - EXPECT_EQ(kOutputCNG, neteq_inst_->getOutputType()); - - // Insert speech again until output type is speech. - while (neteq_inst_->getOutputType() != kOutputNormal) { - // Each turn in this for loop is 10 ms. - while (next_input_time_ms <= t_ms) { - // Insert one 30 ms speech frame. - uint8_t payload[kPayloadBytes] = {0}; - WebRtcNetEQ_RTPInfo rtp_info; - PopulateRtpInfo(seq_no, timestamp, &rtp_info); - ASSERT_EQ(0, - WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), - &rtp_info, - payload, - kPayloadBytes, 0)); - ++seq_no; - timestamp += kSamples; - next_input_time_ms += static_cast<double>(kFrameSizeMs) * kDriftFactor; - } - // Pull out data once. - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - // Increase clock. - t_ms += 10; - } - - int32_t delay_after = timestamp - neteq_inst_->getSpeechTimeStamp(); - // Compare delay before and after, and make sure it differs less than 20 ms. - EXPECT_LE(delay_after, delay_before + 20 * 16); - EXPECT_GE(delay_after, delay_before - 20 * 16); -} - -TEST_F(NetEqDecodingTest, NoInputDataStereo) { - void *ms_info; - ms_info = malloc(WebRtcNetEQ_GetMasterSlaveInfoSize()); - neteq_inst_->setMaster(); - - // Slave instance without decoders (because it is easier). - WebRtcNetEQDecoder usedCodec[kDecoderReservedEnd - 1]; - usedCodec[0] = kDecoderPCMu; - NETEQTEST_NetEQClass* slave_inst = - new NETEQTEST_NetEQClass(usedCodec, 1, 8000, kTCPLargeJitter); - ASSERT_TRUE(slave_inst); - NETEQTEST_Decoder* dec = new decoder_PCMU(0); - ASSERT_TRUE(dec != NULL); - dec->loadToNetEQ(*slave_inst); - slave_inst->setSlave(); - - // Pull out data. - const int kNumFrames = 100; - for (int i = 0; i < kNumFrames; ++i) { - ASSERT_TRUE(kBlockSize8kHz == neteq_inst_->recOut(out_data_, ms_info)); - ASSERT_TRUE(kBlockSize8kHz == slave_inst->recOut(out_data_, ms_info)); - } - - delete dec; - delete slave_inst; - free(ms_info); -} - -TEST_F(NetEqDecodingTest, TestExtraDelay) { - static const int kNumFrames = 120000; // Needed for convergence. - int frame_index = 0; - static const int kFrameSizeSamples = 30 * 16; - static const int kPayloadBytes = kFrameSizeSamples * 2; - test::InputAudioFile input_file( - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm")); - int16_t input[kFrameSizeSamples]; - // Buffers of NetEq cannot accommodate larger delays for PCM16. - static const int kExtraDelayMs = 3200; - ASSERT_EQ(0, WebRtcNetEQ_SetExtraDelay(neteq_inst_->instance(), - kExtraDelayMs)); - for (int i = 0; i < kNumFrames; ++i) { - ASSERT_TRUE(input_file.Read(kFrameSizeSamples, input)); - WebRtcNetEQ_RTPInfo rtp_info; - PopulateRtpInfo(frame_index, frame_index * kFrameSizeSamples, &rtp_info); - uint8_t* payload = reinterpret_cast<uint8_t*>(input); - ASSERT_EQ(0, - WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), - &rtp_info, - payload, - kPayloadBytes, 0)); - ++frame_index; - // Pull out data. - for (int j = 0; j < 3; ++j) { - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - } - if (i % 100 == 0) { - WebRtcNetEQ_NetworkStatistics network_stats; - ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(), - &network_stats)); - const int expected_lower_limit = - std::min(i * 0.083 - 210, 0.9 * network_stats.preferredBufferSize); - EXPECT_GE(network_stats.currentBufferSize, expected_lower_limit); - const int expected_upper_limit = - std::min(i * 0.083 + 255, 1.2 * network_stats.preferredBufferSize); - EXPECT_LE(network_stats.currentBufferSize, expected_upper_limit); - } - } -} - -void NetEqDecodingTest::WrapTest(uint16_t start_seq_no, - uint32_t start_timestamp, - const std::set<uint16_t>& drop_seq_numbers) { - uint16_t seq_no = start_seq_no; - uint32_t timestamp = start_timestamp; - const int kFrameSizeMs = 30; - const int kSamples = kFrameSizeMs * 16; - const int kPayloadBytes = kSamples * 2; - double next_input_time_ms = 0.0; - - // Insert speech for 1 second. - const int kSpeechDurationMs = 1000; - for (double t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) { - // Each turn in this for loop is 10 ms. - while (next_input_time_ms <= t_ms) { - // Insert one 30 ms speech frame. - uint8_t payload[kPayloadBytes] = {0}; - WebRtcNetEQ_RTPInfo rtp_info; - PopulateRtpInfo(seq_no, timestamp, &rtp_info); - if (drop_seq_numbers.find(seq_no) == drop_seq_numbers.end()) { - // This sequence number was not in the set to drop. Insert it. - ASSERT_EQ(0, - WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), - &rtp_info, - payload, - kPayloadBytes, 0)); - } - ++seq_no; - timestamp += kSamples; - next_input_time_ms += static_cast<double>(kFrameSizeMs); - WebRtcNetEQ_NetworkStatistics network_stats; - ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(), - &network_stats)); - // Expect preferred and actual buffer size to be no more than 2 frames. - EXPECT_LE(network_stats.preferredBufferSize, kFrameSizeMs * 2); - EXPECT_LE(network_stats.currentBufferSize, kFrameSizeMs * 2); - } - // Pull out data once. - ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_)); - // Expect delay (in samples) to be less than 2 packets. - EXPECT_LE(timestamp - neteq_inst_->getSpeechTimeStamp(), - static_cast<uint32_t>(kSamples * 2)); - } -} - -TEST_F(NetEqDecodingTest, SequenceNumberWrap) { - // Start with a sequence number that will soon wrap. - std::set<uint16_t> drop_seq_numbers; // Don't drop any packets. - WrapTest(0xFFFF - 5, 0, drop_seq_numbers); -} - -TEST_F(NetEqDecodingTest, SequenceNumberWrapAndDrop) { - // Start with a sequence number that will soon wrap. - std::set<uint16_t> drop_seq_numbers; - drop_seq_numbers.insert(0xFFFF); - drop_seq_numbers.insert(0x0); - WrapTest(0xFFFF - 5, 0, drop_seq_numbers); -} - -TEST_F(NetEqDecodingTest, TimestampWrap) { - // Start with a timestamp that will soon wrap. - std::set<uint16_t> drop_seq_numbers; - WrapTest(0, 0xFFFFFFFF - 1000, drop_seq_numbers); -} - -TEST_F(NetEqDecodingTest, TimestampAndSequenceNumberWrap) { - // Start with a timestamp and a sequence number that will wrap at the same - // time. - std::set<uint16_t> drop_seq_numbers; - WrapTest(0xFFFF - 2, 0xFFFFFFFF - 1000, drop_seq_numbers); -} - -} // namespace |