Revert 6257 "Rename neteq4 folder to neteq"
> Rename neteq4 folder to neteq
>
> BUG=2996
> R=turaj@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/12569005
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6259 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
new file mode 100644
index 0000000..f66a3cf
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
@@ -0,0 +1,1429 @@
+/*
+ * 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/neteq4/interface/neteq.h"
+
+#include <math.h>
+#include <stdlib.h>
+#include <string.h> // memset
+
+#include <algorithm>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "gflags/gflags.h"
+#include "gtest/gtest.h"
+#include "webrtc/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
+#include "webrtc/typedefs.h"
+
+DEFINE_bool(gen_ref, false, "Generate reference files.");
+
+namespace webrtc {
+
+static bool IsAllZero(const int16_t* buf, int buf_length) {
+ bool all_zero = true;
+ for (int n = 0; n < buf_length && all_zero; ++n)
+ all_zero = buf[n] == 0;
+ return all_zero;
+}
+
+static bool IsAllNonZero(const int16_t* buf, int buf_length) {
+ bool all_non_zero = true;
+ for (int n = 0; n < buf_length && all_non_zero; ++n)
+ all_non_zero = buf[n] != 0;
+ return all_non_zero;
+}
+
+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 NetEqNetworkStatistics& stats);
+ void ReadFromFileAndCompare(const NetEqNetworkStatistics& stats);
+ void WriteToFile(const RtcpStatistics& stats);
+ void ReadFromFileAndCompare(const RtcpStatistics& 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 NetEqNetworkStatistics& stats) {
+ if (output_fp_) {
+ ASSERT_EQ(1u, fwrite(&stats, sizeof(NetEqNetworkStatistics), 1,
+ output_fp_));
+ }
+}
+
+void RefFiles::ReadFromFileAndCompare(
+ const NetEqNetworkStatistics& stats) {
+ if (input_fp_) {
+ // Read from ref file.
+ size_t stat_size = sizeof(NetEqNetworkStatistics);
+ NetEqNetworkStatistics ref_stats;
+ ASSERT_EQ(1u, fread(&ref_stats, stat_size, 1, input_fp_));
+ // Compare
+ ASSERT_EQ(0, memcmp(&stats, &ref_stats, stat_size));
+ }
+}
+
+void RefFiles::WriteToFile(const RtcpStatistics& stats) {
+ if (output_fp_) {
+ ASSERT_EQ(1u, fwrite(&(stats.fraction_lost), sizeof(stats.fraction_lost), 1,
+ output_fp_));
+ ASSERT_EQ(1u, fwrite(&(stats.cumulative_lost),
+ sizeof(stats.cumulative_lost), 1, output_fp_));
+ ASSERT_EQ(1u, fwrite(&(stats.extended_max_sequence_number),
+ sizeof(stats.extended_max_sequence_number), 1,
+ output_fp_));
+ ASSERT_EQ(1u, fwrite(&(stats.jitter), sizeof(stats.jitter), 1,
+ output_fp_));
+ }
+}
+
+void RefFiles::ReadFromFileAndCompare(
+ const RtcpStatistics& stats) {
+ if (input_fp_) {
+ // Read from ref file.
+ RtcpStatistics ref_stats;
+ ASSERT_EQ(1u, fread(&(ref_stats.fraction_lost),
+ sizeof(ref_stats.fraction_lost), 1, input_fp_));
+ ASSERT_EQ(1u, fread(&(ref_stats.cumulative_lost),
+ sizeof(ref_stats.cumulative_lost), 1, input_fp_));
+ ASSERT_EQ(1u, fread(&(ref_stats.extended_max_sequence_number),
+ sizeof(ref_stats.extended_max_sequence_number), 1,
+ input_fp_));
+ ASSERT_EQ(1u, fread(&(ref_stats.jitter), sizeof(ref_stats.jitter), 1,
+ input_fp_));
+ // Compare
+ ASSERT_EQ(ref_stats.fraction_lost, stats.fraction_lost);
+ ASSERT_EQ(ref_stats.cumulative_lost, stats.cumulative_lost);
+ ASSERT_EQ(ref_stats.extended_max_sequence_number,
+ stats.extended_max_sequence_number);
+ ASSERT_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;
+ static const int kInitSampleRateHz = 8000;
+
+ NetEqDecodingTest();
+ virtual void SetUp();
+ virtual void TearDown();
+ void SelectDecoders(NetEqDecoder* used_codec);
+ void LoadDecoders();
+ void OpenInputFile(const std::string &rtp_file);
+ void Process(NETEQTEST_RTPpacket* rtp_ptr, int* 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,
+ WebRtcRTPHeader* rtp_info);
+ static void PopulateCng(int frame_index,
+ int timestamp,
+ WebRtcRTPHeader* rtp_info,
+ uint8_t* payload,
+ int* payload_len);
+
+ void CheckBgnOff(int sampling_rate, NetEqBackgroundNoiseMode bgn_mode);
+
+ void WrapTest(uint16_t start_seq_no, uint32_t start_timestamp,
+ const std::set<uint16_t>& drop_seq_numbers,
+ bool expect_seq_no_wrap, bool expect_timestamp_wrap);
+
+ void LongCngWithClockDrift(double drift_factor,
+ double network_freeze_ms,
+ bool pull_audio_during_freeze,
+ int delay_tolerance_ms,
+ int max_time_to_speech_ms);
+
+ void DuplicateCng();
+
+ NetEq* neteq_;
+ FILE* rtp_fp_;
+ unsigned int sim_clock_;
+ int16_t out_data_[kMaxBlockSize];
+ int output_sample_rate_;
+ int algorithmic_delay_ms_;
+};
+
+// Allocating the static const so that it can be passed by reference.
+const int NetEqDecodingTest::kTimeStepMs;
+const int NetEqDecodingTest::kBlockSize8kHz;
+const int NetEqDecodingTest::kBlockSize16kHz;
+const int NetEqDecodingTest::kBlockSize32kHz;
+const int NetEqDecodingTest::kMaxBlockSize;
+const int NetEqDecodingTest::kInitSampleRateHz;
+
+NetEqDecodingTest::NetEqDecodingTest()
+ : neteq_(NULL),
+ rtp_fp_(NULL),
+ sim_clock_(0),
+ output_sample_rate_(kInitSampleRateHz),
+ algorithmic_delay_ms_(0) {
+ memset(out_data_, 0, sizeof(out_data_));
+}
+
+void NetEqDecodingTest::SetUp() {
+ NetEq::Config config;
+ config.sample_rate_hz = kInitSampleRateHz;
+ neteq_ = NetEq::Create(config);
+ NetEqNetworkStatistics stat;
+ ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
+ algorithmic_delay_ms_ = stat.current_buffer_size_ms;
+ ASSERT_TRUE(neteq_);
+ LoadDecoders();
+}
+
+void NetEqDecodingTest::TearDown() {
+ delete neteq_;
+ if (rtp_fp_)
+ fclose(rtp_fp_);
+}
+
+void NetEqDecodingTest::LoadDecoders() {
+ // Load PCMu.
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMu, 0));
+ // Load PCMa.
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMa, 8));
+#ifndef WEBRTC_ANDROID
+ // Load iLBC.
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderILBC, 102));
+#endif // WEBRTC_ANDROID
+ // Load iSAC.
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISAC, 103));
+#ifndef WEBRTC_ANDROID
+ // Load iSAC SWB.
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACswb, 104));
+ // Load iSAC FB.
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACfb, 105));
+#endif // WEBRTC_ANDROID
+ // Load PCM16B nb.
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16B, 93));
+ // Load PCM16B wb.
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16Bwb, 94));
+ // Load PCM16B swb32.
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16Bswb32kHz, 95));
+ // Load CNG 8 kHz.
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGnb, 13));
+ // Load CNG 16 kHz.
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGwb, 98));
+}
+
+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, int* out_len) {
+ // Check if time to receive.
+ while ((sim_clock_ >= rtp->time()) &&
+ (rtp->dataLen() >= 0)) {
+ if (rtp->dataLen() > 0) {
+ WebRtcRTPHeader rtpInfo;
+ rtp->parseHeader(&rtpInfo);
+ ASSERT_EQ(0, neteq_->InsertPacket(
+ rtpInfo,
+ rtp->payload(),
+ rtp->payloadLen(),
+ rtp->time() * (output_sample_rate_ / 1000)));
+ }
+ // Get next packet.
+ ASSERT_NE(-1, rtp->readFromFile(rtp_fp_));
+ }
+
+ // Get audio from NetEq.
+ NetEqOutputType type;
+ int num_channels;
+ ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, out_len,
+ &num_channels, &type));
+ ASSERT_TRUE((*out_len == kBlockSize8kHz) ||
+ (*out_len == kBlockSize16kHz) ||
+ (*out_len == kBlockSize32kHz));
+ output_sample_rate_ = *out_len / 10 * 1000;
+
+ // 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_universal_ref.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.
+ int out_len = 0;
+ 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) {
+ int out_len;
+ ASSERT_NO_FATAL_FAILURE(Process(&rtp, &out_len));
+
+ // Query the network statistics API once per second
+ if (sim_clock_ % 1000 == 0) {
+ // Process NetworkStatistics.
+ NetEqNetworkStatistics network_stats;
+ ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
+ ASSERT_NO_FATAL_FAILURE(
+ network_stat_files.ProcessReference(network_stats));
+
+ // Process RTCPstat.
+ RtcpStatistics rtcp_stats;
+ neteq_->GetRtcpStatistics(&rtcp_stats);
+ ASSERT_NO_FATAL_FAILURE(rtcp_stat_files.ProcessReference(rtcp_stats));
+ }
+ }
+}
+
+void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
+ int timestamp,
+ WebRtcRTPHeader* rtp_info) {
+ rtp_info->header.sequenceNumber = frame_index;
+ rtp_info->header.timestamp = timestamp;
+ rtp_info->header.ssrc = 0x1234; // Just an arbitrary SSRC.
+ rtp_info->header.payloadType = 94; // PCM16b WB codec.
+ rtp_info->header.markerBit = 0;
+}
+
+void NetEqDecodingTest::PopulateCng(int frame_index,
+ int timestamp,
+ WebRtcRTPHeader* rtp_info,
+ uint8_t* payload,
+ int* payload_len) {
+ rtp_info->header.sequenceNumber = frame_index;
+ rtp_info->header.timestamp = timestamp;
+ rtp_info->header.ssrc = 0x1234; // Just an arbitrary SSRC.
+ rtp_info->header.payloadType = 98; // WB CNG.
+ rtp_info->header.markerBit = 0;
+ payload[0] = 64; // Noise level -64 dBov, quite arbitrarily chosen.
+ *payload_len = 1; // Only noise level, no spectral parameters.
+}
+
+void NetEqDecodingTest::CheckBgnOff(int sampling_rate_hz,
+ NetEqBackgroundNoiseMode bgn_mode) {
+ int expected_samples_per_channel = 0;
+ uint8_t payload_type = 0xFF; // Invalid.
+ if (sampling_rate_hz == 8000) {
+ expected_samples_per_channel = kBlockSize8kHz;
+ payload_type = 93; // PCM 16, 8 kHz.
+ } else if (sampling_rate_hz == 16000) {
+ expected_samples_per_channel = kBlockSize16kHz;
+ payload_type = 94; // PCM 16, 16 kHZ.
+ } else if (sampling_rate_hz == 32000) {
+ expected_samples_per_channel = kBlockSize32kHz;
+ payload_type = 95; // PCM 16, 32 kHz.
+ } else {
+ ASSERT_TRUE(false); // Unsupported test case.
+ }
+
+ NetEqOutputType type;
+ int16_t output[kBlockSize32kHz]; // Maximum size is chosen.
+ int16_t input[kBlockSize32kHz]; // Maximum size is chosen.
+
+ // Payload of 10 ms of PCM16 32 kHz.
+ uint8_t payload[kBlockSize32kHz * sizeof(int16_t)];
+
+ // Random payload.
+ for (int n = 0; n < expected_samples_per_channel; ++n) {
+ input[n] = (rand() & ((1 << 10) - 1)) - ((1 << 5) - 1);
+ }
+ int enc_len_bytes = WebRtcPcm16b_EncodeW16(
+ input, expected_samples_per_channel, reinterpret_cast<int16_t*>(payload));
+ ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
+
+ WebRtcRTPHeader rtp_info;
+ PopulateRtpInfo(0, 0, &rtp_info);
+ rtp_info.header.payloadType = payload_type;
+
+ int number_channels = 0;
+ int samples_per_channel = 0;
+
+ uint32_t receive_timestamp = 0;
+ for (int n = 0; n < 10; ++n) { // Insert few packets and get audio.
+ number_channels = 0;
+ samples_per_channel = 0;
+ ASSERT_EQ(0, neteq_->InsertPacket(
+ rtp_info, payload, enc_len_bytes, receive_timestamp));
+ ASSERT_EQ(0, neteq_->GetAudio(kBlockSize32kHz, output, &samples_per_channel,
+ &number_channels, &type));
+ ASSERT_EQ(1, number_channels);
+ ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
+ ASSERT_EQ(kOutputNormal, type);
+
+ // Next packet.
+ rtp_info.header.timestamp += expected_samples_per_channel;
+ rtp_info.header.sequenceNumber++;
+ receive_timestamp += expected_samples_per_channel;
+ }
+
+ number_channels = 0;
+ samples_per_channel = 0;
+
+ // Get audio without inserting packets, expecting PLC and PLC-to-CNG. Pull one
+ // frame without checking speech-type. This is the first frame pulled without
+ // inserting any packet, and might not be labeled as PCL.
+ ASSERT_EQ(0, neteq_->GetAudio(kBlockSize32kHz, output, &samples_per_channel,
+ &number_channels, &type));
+ ASSERT_EQ(1, number_channels);
+ ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
+
+ // To be able to test the fading of background noise we need at lease to pull
+ // 611 frames.
+ const int kFadingThreshold = 611;
+
+ // Test several CNG-to-PLC packet for the expected behavior. The number 20 is
+ // arbitrary, but sufficiently large to test enough number of frames.
+ const int kNumPlcToCngTestFrames = 20;
+ bool plc_to_cng = false;
+ for (int n = 0; n < kFadingThreshold + kNumPlcToCngTestFrames; ++n) {
+ number_channels = 0;
+ samples_per_channel = 0;
+ memset(output, 1, sizeof(output)); // Set to non-zero.
+ ASSERT_EQ(0, neteq_->GetAudio(kBlockSize32kHz, output, &samples_per_channel,
+ &number_channels, &type));
+ ASSERT_EQ(1, number_channels);
+ ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
+ if (type == kOutputPLCtoCNG) {
+ plc_to_cng = true;
+ double sum_squared = 0;
+ for (int k = 0; k < number_channels * samples_per_channel; ++k)
+ sum_squared += output[k] * output[k];
+ if (bgn_mode == kBgnOn) {
+ EXPECT_NE(0, sum_squared);
+ } else if (bgn_mode == kBgnOff || n > kFadingThreshold) {
+ EXPECT_EQ(0, sum_squared);
+ }
+ } else {
+ EXPECT_EQ(kOutputPLC, type);
+ }
+ }
+ EXPECT_TRUE(plc_to_cng); // Just to be sure that PLC-to-CNG has occurred.
+}
+
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestBitExactness)) {
+ const std::string input_rtp_file = webrtc::test::ProjectRootPath() +
+ "resources/audio_coding/neteq_universal_new.rtp";
+ // Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm
+ // are identical. The latter could have been removed, but if clients still
+ // have a copy of the file, the test will fail.
+ const std::string input_ref_file =
+ webrtc::test::ResourcePath("audio_coding/neteq4_universal_ref", "pcm");
+
+ if (FLAGS_gen_ref) {
+ DecodeAndCompare(input_rtp_file, "");
+ } else {
+ DecodeAndCompare(input_rtp_file, input_ref_file);
+ }
+}
+
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestNetworkStatistics)) {
+ const std::string input_rtp_file = webrtc::test::ProjectRootPath() +
+ "resources/audio_coding/neteq_universal_new.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 network_stat_ref_file = webrtc::test::ProjectRootPath() +
+ "resources/audio_coding/neteq4_network_stats.dat";
+#else
+ const std::string network_stat_ref_file =
+ webrtc::test::ResourcePath("audio_coding/neteq4_network_stats", "dat");
+#endif
+ const std::string rtcp_stat_ref_file =
+ webrtc::test::ResourcePath("audio_coding/neteq4_rtcp_stats", "dat");
+ if (FLAGS_gen_ref) {
+ DecodeAndCheckStats(input_rtp_file, "", "");
+ } else {
+ DecodeAndCheckStats(input_rtp_file, network_stat_ref_file,
+ rtcp_stat_ref_file);
+ }
+}
+
+// TODO(hlundin): Re-enable test once the statistics interface is up and again.
+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.
+ neteq_->SetPlayoutMode(kPlayoutFax);
+ ASSERT_EQ(kPlayoutFax, neteq_->PlayoutMode());
+ // Insert 30 dummy packets at once. Each packet contains 10 ms 16 kHz audio.
+ size_t num_frames = 30;
+ const int kSamples = 10 * 16;
+ const int kPayloadBytes = kSamples * 2;
+ for (size_t i = 0; i < num_frames; ++i) {
+ uint16_t payload[kSamples] = {0};
+ WebRtcRTPHeader rtp_info;
+ rtp_info.header.sequenceNumber = i;
+ rtp_info.header.timestamp = i * kSamples;
+ rtp_info.header.ssrc = 0x1234; // Just an arbitrary SSRC.
+ rtp_info.header.payloadType = 94; // PCM16b WB codec.
+ rtp_info.header.markerBit = 0;
+ ASSERT_EQ(0, neteq_->InsertPacket(
+ rtp_info,
+ reinterpret_cast<uint8_t*>(payload),
+ kPayloadBytes, 0));
+ }
+ // Pull out all data.
+ for (size_t i = 0; i < num_frames; ++i) {
+ int out_len;
+ int num_channels;
+ NetEqOutputType type;
+ ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
+ &num_channels, &type));
+ ASSERT_EQ(kBlockSize16kHz, out_len);
+ }
+
+ std::vector<int> waiting_times;
+ neteq_->WaitingTimes(&waiting_times);
+ EXPECT_EQ(num_frames, waiting_times.size());
+ // 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 (size_t i = 0; i < waiting_times.size(); ++i) {
+ EXPECT_EQ(static_cast<int>(i + 1) * 10, waiting_times[i]);
+ }
+
+ // Check statistics again and make sure it's been reset.
+ neteq_->WaitingTimes(&waiting_times);
+ int len = waiting_times.size();
+ EXPECT_EQ(0, len);
+
+ // 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 (size_t i = 0; i < num_frames; ++i) {
+ uint16_t payload[kSamples] = {0};
+ WebRtcRTPHeader rtp_info;
+ rtp_info.header.sequenceNumber = i;
+ rtp_info.header.timestamp = i * kSamples;
+ rtp_info.header.ssrc = 0x1235; // Just an arbitrary SSRC.
+ rtp_info.header.payloadType = 94; // PCM16b WB codec.
+ rtp_info.header.markerBit = 0;
+ ASSERT_EQ(0, neteq_->InsertPacket(
+ rtp_info,
+ reinterpret_cast<uint8_t*>(payload),
+ kPayloadBytes, 0));
+ int out_len;
+ int num_channels;
+ NetEqOutputType type;
+ ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
+ &num_channels, &type));
+ ASSERT_EQ(kBlockSize16kHz, out_len);
+ }
+
+ neteq_->WaitingTimes(&waiting_times);
+ EXPECT_EQ(100u, waiting_times.size());
+}
+
+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};
+ WebRtcRTPHeader rtp_info;
+ PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+ ++frame_index;
+ }
+
+ // Pull out data once.
+ int out_len;
+ int num_channels;
+ NetEqOutputType type;
+ ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
+ &num_channels, &type));
+ ASSERT_EQ(kBlockSize16kHz, out_len);
+ }
+
+ NetEqNetworkStatistics network_stats;
+ ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
+ EXPECT_EQ(-103196, network_stats.clockdrift_ppm);
+}
+
+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};
+ WebRtcRTPHeader rtp_info;
+ PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+ ++frame_index;
+ }
+
+ // Pull out data once.
+ int out_len;
+ int num_channels;
+ NetEqOutputType type;
+ ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
+ &num_channels, &type));
+ ASSERT_EQ(kBlockSize16kHz, out_len);
+ }
+
+ NetEqNetworkStatistics network_stats;
+ ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
+ EXPECT_EQ(110946, network_stats.clockdrift_ppm);
+}
+
+void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
+ double network_freeze_ms,
+ bool pull_audio_during_freeze,
+ int delay_tolerance_ms,
+ int max_time_to_speech_ms) {
+ uint16_t seq_no = 0;
+ uint32_t timestamp = 0;
+ const int kFrameSizeMs = 30;
+ const int kSamples = kFrameSizeMs * 16;
+ const int kPayloadBytes = kSamples * 2;
+ double next_input_time_ms = 0.0;
+ double t_ms;
+ int out_len;
+ int num_channels;
+ NetEqOutputType type;
+
+ // 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};
+ WebRtcRTPHeader rtp_info;
+ PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+ ++seq_no;
+ timestamp += kSamples;
+ next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
+ }
+ // Pull out data once.
+ ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
+ &num_channels, &type));
+ ASSERT_EQ(kBlockSize16kHz, out_len);
+ }
+
+ EXPECT_EQ(kOutputNormal, type);
+ int32_t delay_before = timestamp - neteq_->PlayoutTimestamp();
+
+ // 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;
+ WebRtcRTPHeader rtp_info;
+ PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
+ ++seq_no;
+ timestamp += kCngPeriodSamples;
+ next_input_time_ms += static_cast<double>(kCngPeriodMs) * drift_factor;
+ }
+ // Pull out data once.
+ ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
+ &num_channels, &type));
+ ASSERT_EQ(kBlockSize16kHz, out_len);
+ }
+
+ EXPECT_EQ(kOutputCNG, type);
+
+ if (network_freeze_ms > 0) {
+ // First keep pulling audio for |network_freeze_ms| without inserting
+ // any data, then insert CNG data corresponding to |network_freeze_ms|
+ // without pulling any output audio.
+ const double loop_end_time = t_ms + network_freeze_ms;
+ for (; t_ms < loop_end_time; t_ms += 10) {
+ // Pull out data once.
+ ASSERT_EQ(0,
+ neteq_->GetAudio(
+ kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
+ ASSERT_EQ(kBlockSize16kHz, out_len);
+ EXPECT_EQ(kOutputCNG, type);
+ }
+ bool pull_once = pull_audio_during_freeze;
+ // If |pull_once| is true, GetAudio will be called once half-way through
+ // the network recovery period.
+ double pull_time_ms = (t_ms + next_input_time_ms) / 2;
+ while (next_input_time_ms <= t_ms) {
+ if (pull_once && next_input_time_ms >= pull_time_ms) {
+ pull_once = false;
+ // Pull out data once.
+ ASSERT_EQ(
+ 0,
+ neteq_->GetAudio(
+ kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
+ ASSERT_EQ(kBlockSize16kHz, out_len);
+ EXPECT_EQ(kOutputCNG, type);
+ t_ms += 10;
+ }
+ // Insert one CNG frame each 100 ms.
+ uint8_t payload[kPayloadBytes];
+ int payload_len;
+ WebRtcRTPHeader rtp_info;
+ PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
+ ++seq_no;
+ timestamp += kCngPeriodSamples;
+ next_input_time_ms += kCngPeriodMs * drift_factor;
+ }
+ }
+
+ // Insert speech again until output type is speech.
+ double speech_restart_time_ms = t_ms;
+ while (type != 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};
+ WebRtcRTPHeader rtp_info;
+ PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+ ++seq_no;
+ timestamp += kSamples;
+ next_input_time_ms += kFrameSizeMs * drift_factor;
+ }
+ // Pull out data once.
+ ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
+ &num_channels, &type));
+ ASSERT_EQ(kBlockSize16kHz, out_len);
+ // Increase clock.
+ t_ms += 10;
+ }
+
+ // Check that the speech starts again within reasonable time.
+ double time_until_speech_returns_ms = t_ms - speech_restart_time_ms;
+ EXPECT_LT(time_until_speech_returns_ms, max_time_to_speech_ms);
+ int32_t delay_after = timestamp - neteq_->PlayoutTimestamp();
+ // Compare delay before and after, and make sure it differs less than 20 ms.
+ EXPECT_LE(delay_after, delay_before + delay_tolerance_ms * 16);
+ EXPECT_GE(delay_after, delay_before - delay_tolerance_ms * 16);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDrift) {
+ // Apply a clock drift of -25 ms / s (sender faster than receiver).
+ const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
+ const double kNetworkFreezeTimeMs = 0.0;
+ const bool kGetAudioDuringFreezeRecovery = false;
+ const int kDelayToleranceMs = 20;
+ const int kMaxTimeToSpeechMs = 100;
+ LongCngWithClockDrift(kDriftFactor,
+ kNetworkFreezeTimeMs,
+ kGetAudioDuringFreezeRecovery,
+ kDelayToleranceMs,
+ kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDrift) {
+ // Apply a clock drift of +25 ms / s (sender slower than receiver).
+ const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
+ const double kNetworkFreezeTimeMs = 0.0;
+ const bool kGetAudioDuringFreezeRecovery = false;
+ const int kDelayToleranceMs = 20;
+ const int kMaxTimeToSpeechMs = 100;
+ LongCngWithClockDrift(kDriftFactor,
+ kNetworkFreezeTimeMs,
+ kGetAudioDuringFreezeRecovery,
+ kDelayToleranceMs,
+ kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDriftNetworkFreeze) {
+ // Apply a clock drift of -25 ms / s (sender faster than receiver).
+ const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
+ const double kNetworkFreezeTimeMs = 5000.0;
+ const bool kGetAudioDuringFreezeRecovery = false;
+ const int kDelayToleranceMs = 50;
+ const int kMaxTimeToSpeechMs = 200;
+ LongCngWithClockDrift(kDriftFactor,
+ kNetworkFreezeTimeMs,
+ kGetAudioDuringFreezeRecovery,
+ kDelayToleranceMs,
+ kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDriftNetworkFreeze) {
+ // Apply a clock drift of +25 ms / s (sender slower than receiver).
+ const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
+ const double kNetworkFreezeTimeMs = 5000.0;
+ const bool kGetAudioDuringFreezeRecovery = false;
+ const int kDelayToleranceMs = 20;
+ const int kMaxTimeToSpeechMs = 100;
+ LongCngWithClockDrift(kDriftFactor,
+ kNetworkFreezeTimeMs,
+ kGetAudioDuringFreezeRecovery,
+ kDelayToleranceMs,
+ kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDriftNetworkFreezeExtraPull) {
+ // Apply a clock drift of +25 ms / s (sender slower than receiver).
+ const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
+ const double kNetworkFreezeTimeMs = 5000.0;
+ const bool kGetAudioDuringFreezeRecovery = true;
+ const int kDelayToleranceMs = 20;
+ const int kMaxTimeToSpeechMs = 100;
+ LongCngWithClockDrift(kDriftFactor,
+ kNetworkFreezeTimeMs,
+ kGetAudioDuringFreezeRecovery,
+ kDelayToleranceMs,
+ kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithoutClockDrift) {
+ const double kDriftFactor = 1.0; // No drift.
+ const double kNetworkFreezeTimeMs = 0.0;
+ const bool kGetAudioDuringFreezeRecovery = false;
+ const int kDelayToleranceMs = 10;
+ const int kMaxTimeToSpeechMs = 50;
+ LongCngWithClockDrift(kDriftFactor,
+ kNetworkFreezeTimeMs,
+ kGetAudioDuringFreezeRecovery,
+ kDelayToleranceMs,
+ kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, UnknownPayloadType) {
+ const int kPayloadBytes = 100;
+ uint8_t payload[kPayloadBytes] = {0};
+ WebRtcRTPHeader rtp_info;
+ PopulateRtpInfo(0, 0, &rtp_info);
+ rtp_info.header.payloadType = 1; // Not registered as a decoder.
+ EXPECT_EQ(NetEq::kFail,
+ neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+ EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
+}
+
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) {
+ const int kPayloadBytes = 100;
+ uint8_t payload[kPayloadBytes] = {0};
+ WebRtcRTPHeader rtp_info;
+ PopulateRtpInfo(0, 0, &rtp_info);
+ rtp_info.header.payloadType = 103; // iSAC, but the payload is invalid.
+ EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+ NetEqOutputType type;
+ // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
+ // to GetAudio.
+ for (int i = 0; i < kMaxBlockSize; ++i) {
+ out_data_[i] = 1;
+ }
+ int num_channels;
+ int samples_per_channel;
+ EXPECT_EQ(NetEq::kFail,
+ neteq_->GetAudio(kMaxBlockSize, out_data_,
+ &samples_per_channel, &num_channels, &type));
+ // Verify that there is a decoder error to check.
+ EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
+ // Code 6730 is an iSAC error code.
+ EXPECT_EQ(6730, neteq_->LastDecoderError());
+ // Verify that the first 160 samples are set to 0, and that the remaining
+ // samples are left unmodified.
+ static const int kExpectedOutputLength = 160; // 10 ms at 16 kHz sample rate.
+ for (int i = 0; i < kExpectedOutputLength; ++i) {
+ std::ostringstream ss;
+ ss << "i = " << i;
+ SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
+ EXPECT_EQ(0, out_data_[i]);
+ }
+ for (int i = kExpectedOutputLength; i < kMaxBlockSize; ++i) {
+ std::ostringstream ss;
+ ss << "i = " << i;
+ SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
+ EXPECT_EQ(1, out_data_[i]);
+ }
+}
+
+TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
+ NetEqOutputType type;
+ // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
+ // to GetAudio.
+ for (int i = 0; i < kMaxBlockSize; ++i) {
+ out_data_[i] = 1;
+ }
+ int num_channels;
+ int samples_per_channel;
+ EXPECT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_,
+ &samples_per_channel,
+ &num_channels, &type));
+ // Verify that the first block of samples is set to 0.
+ static const int kExpectedOutputLength =
+ kInitSampleRateHz / 100; // 10 ms at initial sample rate.
+ for (int i = 0; i < kExpectedOutputLength; ++i) {
+ std::ostringstream ss;
+ ss << "i = " << i;
+ SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
+ EXPECT_EQ(0, out_data_[i]);
+ }
+}
+
+TEST_F(NetEqDecodingTest, BackgroundNoise) {
+ neteq_->SetBackgroundNoiseMode(kBgnOn);
+ CheckBgnOff(8000, kBgnOn);
+ CheckBgnOff(16000, kBgnOn);
+ CheckBgnOff(32000, kBgnOn);
+ EXPECT_EQ(kBgnOn, neteq_->BackgroundNoiseMode());
+
+ neteq_->SetBackgroundNoiseMode(kBgnOff);
+ CheckBgnOff(8000, kBgnOff);
+ CheckBgnOff(16000, kBgnOff);
+ CheckBgnOff(32000, kBgnOff);
+ EXPECT_EQ(kBgnOff, neteq_->BackgroundNoiseMode());
+
+ neteq_->SetBackgroundNoiseMode(kBgnFade);
+ CheckBgnOff(8000, kBgnFade);
+ CheckBgnOff(16000, kBgnFade);
+ CheckBgnOff(32000, kBgnFade);
+ EXPECT_EQ(kBgnFade, neteq_->BackgroundNoiseMode());
+}
+
+TEST_F(NetEqDecodingTest, SyncPacketInsert) {
+ WebRtcRTPHeader rtp_info;
+ uint32_t receive_timestamp = 0;
+ // For the readability use the following payloads instead of the defaults of
+ // this test.
+ uint8_t kPcm16WbPayloadType = 1;
+ uint8_t kCngNbPayloadType = 2;
+ uint8_t kCngWbPayloadType = 3;
+ uint8_t kCngSwb32PayloadType = 4;
+ uint8_t kCngSwb48PayloadType = 5;
+ uint8_t kAvtPayloadType = 6;
+ uint8_t kRedPayloadType = 7;
+ uint8_t kIsacPayloadType = 9; // Payload type 8 is already registered.
+
+ // Register decoders.
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16Bwb,
+ kPcm16WbPayloadType));
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGnb, kCngNbPayloadType));
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGwb, kCngWbPayloadType));
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGswb32kHz,
+ kCngSwb32PayloadType));
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGswb48kHz,
+ kCngSwb48PayloadType));
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderAVT, kAvtPayloadType));
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderRED, kRedPayloadType));
+ ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISAC, kIsacPayloadType));
+
+ PopulateRtpInfo(0, 0, &rtp_info);
+ rtp_info.header.payloadType = kPcm16WbPayloadType;
+
+ // The first packet injected cannot be sync-packet.
+ EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
+
+ // Payload length of 10 ms PCM16 16 kHz.
+ const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
+ uint8_t payload[kPayloadBytes] = {0};
+ ASSERT_EQ(0, neteq_->InsertPacket(
+ rtp_info, payload, kPayloadBytes, receive_timestamp));
+
+ // Next packet. Last packet contained 10 ms audio.
+ rtp_info.header.sequenceNumber++;
+ rtp_info.header.timestamp += kBlockSize16kHz;
+ receive_timestamp += kBlockSize16kHz;
+
+ // Unacceptable payload types CNG, AVT (DTMF), RED.
+ rtp_info.header.payloadType = kCngNbPayloadType;
+ EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
+
+ rtp_info.header.payloadType = kCngWbPayloadType;
+ EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
+
+ rtp_info.header.payloadType = kCngSwb32PayloadType;
+ EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
+
+ rtp_info.header.payloadType = kCngSwb48PayloadType;
+ EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
+
+ rtp_info.header.payloadType = kAvtPayloadType;
+ EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
+
+ rtp_info.header.payloadType = kRedPayloadType;
+ EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
+
+ // Change of codec cannot be initiated with a sync packet.
+ rtp_info.header.payloadType = kIsacPayloadType;
+ EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
+
+ // Change of SSRC is not allowed with a sync packet.
+ rtp_info.header.payloadType = kPcm16WbPayloadType;
+ ++rtp_info.header.ssrc;
+ EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
+
+ --rtp_info.header.ssrc;
+ EXPECT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
+}
+
+// First insert several noise like packets, then sync-packets. Decoding all
+// packets should not produce error, statistics should not show any packet loss
+// and sync-packets should decode to zero.
+// TODO(turajs) we will have a better test if we have a referece NetEq, and
+// when Sync packets are inserted in "test" NetEq we insert all-zero payload
+// in reference NetEq and compare the output of those two.
+TEST_F(NetEqDecodingTest, SyncPacketDecode) {
+ WebRtcRTPHeader rtp_info;
+ PopulateRtpInfo(0, 0, &rtp_info);
+ const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
+ uint8_t payload[kPayloadBytes];
+ int16_t decoded[kBlockSize16kHz];
+ int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
+ for (int n = 0; n < kPayloadBytes; ++n) {
+ payload[n] = (rand() & 0xF0) + 1; // Non-zero random sequence.
+ }
+ // Insert some packets which decode to noise. We are not interested in
+ // actual decoded values.
+ NetEqOutputType output_type;
+ int num_channels;
+ int samples_per_channel;
+ uint32_t receive_timestamp = 0;
+ for (int n = 0; n < 100; ++n) {
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
+ receive_timestamp));
+ ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
+ &samples_per_channel, &num_channels,
+ &output_type));
+ ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
+ ASSERT_EQ(1, num_channels);
+
+ rtp_info.header.sequenceNumber++;
+ rtp_info.header.timestamp += kBlockSize16kHz;
+ receive_timestamp += kBlockSize16kHz;
+ }
+ const int kNumSyncPackets = 10;
+
+ // Make sure sufficient number of sync packets are inserted that we can
+ // conduct a test.
+ ASSERT_GT(kNumSyncPackets, algorithmic_frame_delay);
+ // Insert sync-packets, the decoded sequence should be all-zero.
+ for (int n = 0; n < kNumSyncPackets; ++n) {
+ ASSERT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
+ ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
+ &samples_per_channel, &num_channels,
+ &output_type));
+ ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
+ ASSERT_EQ(1, num_channels);
+ if (n > algorithmic_frame_delay) {
+ EXPECT_TRUE(IsAllZero(decoded, samples_per_channel * num_channels));
+ }
+ rtp_info.header.sequenceNumber++;
+ rtp_info.header.timestamp += kBlockSize16kHz;
+ receive_timestamp += kBlockSize16kHz;
+ }
+
+ // We insert regular packets, if sync packet are not correctly buffered then
+ // network statistics would show some packet loss.
+ for (int n = 0; n <= algorithmic_frame_delay + 10; ++n) {
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
+ receive_timestamp));
+ ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
+ &samples_per_channel, &num_channels,
+ &output_type));
+ if (n >= algorithmic_frame_delay + 1) {
+ // Expect that this frame contain samples from regular RTP.
+ EXPECT_TRUE(IsAllNonZero(decoded, samples_per_channel * num_channels));
+ }
+ rtp_info.header.sequenceNumber++;
+ rtp_info.header.timestamp += kBlockSize16kHz;
+ receive_timestamp += kBlockSize16kHz;
+ }
+ NetEqNetworkStatistics network_stats;
+ ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
+ // Expecting a "clean" network.
+ EXPECT_EQ(0, network_stats.packet_loss_rate);
+ EXPECT_EQ(0, network_stats.expand_rate);
+ EXPECT_EQ(0, network_stats.accelerate_rate);
+ EXPECT_LE(network_stats.preemptive_rate, 150);
+}
+
+// Test if the size of the packet buffer reported correctly when containing
+// sync packets. Also, test if network packets override sync packets. That is to
+// prefer decoding a network packet to a sync packet, if both have same sequence
+// number and timestamp.
+TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
+ WebRtcRTPHeader rtp_info;
+ PopulateRtpInfo(0, 0, &rtp_info);
+ const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
+ uint8_t payload[kPayloadBytes];
+ int16_t decoded[kBlockSize16kHz];
+ for (int n = 0; n < kPayloadBytes; ++n) {
+ payload[n] = (rand() & 0xF0) + 1; // Non-zero random sequence.
+ }
+ // Insert some packets which decode to noise. We are not interested in
+ // actual decoded values.
+ NetEqOutputType output_type;
+ int num_channels;
+ int samples_per_channel;
+ uint32_t receive_timestamp = 0;
+ int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
+ for (int n = 0; n < algorithmic_frame_delay; ++n) {
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
+ receive_timestamp));
+ ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
+ &samples_per_channel, &num_channels,
+ &output_type));
+ ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
+ ASSERT_EQ(1, num_channels);
+ rtp_info.header.sequenceNumber++;
+ rtp_info.header.timestamp += kBlockSize16kHz;
+ receive_timestamp += kBlockSize16kHz;
+ }
+ const int kNumSyncPackets = 10;
+
+ WebRtcRTPHeader first_sync_packet_rtp_info;
+ memcpy(&first_sync_packet_rtp_info, &rtp_info, sizeof(rtp_info));
+
+ // Insert sync-packets, but no decoding.
+ for (int n = 0; n < kNumSyncPackets; ++n) {
+ ASSERT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
+ rtp_info.header.sequenceNumber++;
+ rtp_info.header.timestamp += kBlockSize16kHz;
+ receive_timestamp += kBlockSize16kHz;
+ }
+ NetEqNetworkStatistics network_stats;
+ ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
+ EXPECT_EQ(kNumSyncPackets * 10 + algorithmic_delay_ms_,
+ network_stats.current_buffer_size_ms);
+
+ // Rewind |rtp_info| to that of the first sync packet.
+ memcpy(&rtp_info, &first_sync_packet_rtp_info, sizeof(rtp_info));
+
+ // Insert.
+ for (int n = 0; n < kNumSyncPackets; ++n) {
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
+ receive_timestamp));
+ rtp_info.header.sequenceNumber++;
+ rtp_info.header.timestamp += kBlockSize16kHz;
+ receive_timestamp += kBlockSize16kHz;
+ }
+
+ // Decode.
+ for (int n = 0; n < kNumSyncPackets; ++n) {
+ ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
+ &samples_per_channel, &num_channels,
+ &output_type));
+ ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
+ ASSERT_EQ(1, num_channels);
+ EXPECT_TRUE(IsAllNonZero(decoded, samples_per_channel * num_channels));
+ }
+}
+
+void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
+ uint32_t start_timestamp,
+ const std::set<uint16_t>& drop_seq_numbers,
+ bool expect_seq_no_wrap,
+ bool expect_timestamp_wrap) {
+ uint16_t seq_no = start_seq_no;
+ uint32_t timestamp = start_timestamp;
+ const int kBlocksPerFrame = 3; // Number of 10 ms blocks per frame.
+ const int kFrameSizeMs = kBlocksPerFrame * kTimeStepMs;
+ const int kSamples = kBlockSize16kHz * kBlocksPerFrame;
+ const int kPayloadBytes = kSamples * sizeof(int16_t);
+ double next_input_time_ms = 0.0;
+ int16_t decoded[kBlockSize16kHz];
+ int num_channels;
+ int samples_per_channel;
+ NetEqOutputType output_type;
+ uint32_t receive_timestamp = 0;
+
+ // Insert speech for 2 seconds.
+ const int kSpeechDurationMs = 2000;
+ int packets_inserted = 0;
+ uint16_t last_seq_no;
+ uint32_t last_timestamp;
+ bool timestamp_wrapped = false;
+ bool seq_no_wrapped = false;
+ 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};
+ WebRtcRTPHeader 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,
+ neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
+ receive_timestamp));
+ ++packets_inserted;
+ }
+ NetEqNetworkStatistics network_stats;
+ ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
+
+ // Due to internal NetEq logic, preferred buffer-size is about 4 times the
+ // packet size for first few packets. Therefore we refrain from checking
+ // the criteria.
+ if (packets_inserted > 4) {
+ // Expect preferred and actual buffer size to be no more than 2 frames.
+ EXPECT_LE(network_stats.preferred_buffer_size_ms, kFrameSizeMs * 2);
+ EXPECT_LE(network_stats.current_buffer_size_ms, kFrameSizeMs * 2 +
+ algorithmic_delay_ms_);
+ }
+ last_seq_no = seq_no;
+ last_timestamp = timestamp;
+
+ ++seq_no;
+ timestamp += kSamples;
+ receive_timestamp += kSamples;
+ next_input_time_ms += static_cast<double>(kFrameSizeMs);
+
+ seq_no_wrapped |= seq_no < last_seq_no;
+ timestamp_wrapped |= timestamp < last_timestamp;
+ }
+ // Pull out data once.
+ ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
+ &samples_per_channel, &num_channels,
+ &output_type));
+ ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
+ ASSERT_EQ(1, num_channels);
+
+ // Expect delay (in samples) to be less than 2 packets.
+ EXPECT_LE(timestamp - neteq_->PlayoutTimestamp(),
+ static_cast<uint32_t>(kSamples * 2));
+ }
+ // Make sure we have actually tested wrap-around.
+ ASSERT_EQ(expect_seq_no_wrap, seq_no_wrapped);
+ ASSERT_EQ(expect_timestamp_wrap, timestamp_wrapped);
+}
+
+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 - 10, 0, drop_seq_numbers, true, false);
+}
+
+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 - 10, 0, drop_seq_numbers, true, false);
+}
+
+TEST_F(NetEqDecodingTest, TimestampWrap) {
+ // Start with a timestamp that will soon wrap.
+ std::set<uint16_t> drop_seq_numbers;
+ WrapTest(0, 0xFFFFFFFF - 3000, drop_seq_numbers, false, true);
+}
+
+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 - 10, 0xFFFFFFFF - 5000, drop_seq_numbers, true, true);
+}
+
+void NetEqDecodingTest::DuplicateCng() {
+ uint16_t seq_no = 0;
+ uint32_t timestamp = 0;
+ const int kFrameSizeMs = 10;
+ const int kSampleRateKhz = 16;
+ const int kSamples = kFrameSizeMs * kSampleRateKhz;
+ const int kPayloadBytes = kSamples * 2;
+
+ const int algorithmic_delay_samples = std::max(
+ algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
+ // Insert three speech packet. Three are needed to get the frame length
+ // correct.
+ int out_len;
+ int num_channels;
+ NetEqOutputType type;
+ uint8_t payload[kPayloadBytes] = {0};
+ WebRtcRTPHeader rtp_info;
+ for (int i = 0; i < 3; ++i) {
+ PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+ ++seq_no;
+ timestamp += kSamples;
+
+ // Pull audio once.
+ ASSERT_EQ(0,
+ neteq_->GetAudio(
+ kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
+ ASSERT_EQ(kBlockSize16kHz, out_len);
+ }
+ // Verify speech output.
+ EXPECT_EQ(kOutputNormal, type);
+
+ // Insert same CNG packet twice.
+ const int kCngPeriodMs = 100;
+ const int kCngPeriodSamples = kCngPeriodMs * kSampleRateKhz;
+ int payload_len;
+ PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+ // This is the first time this CNG packet is inserted.
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
+
+ // Pull audio once and make sure CNG is played.
+ ASSERT_EQ(0,
+ neteq_->GetAudio(
+ kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
+ ASSERT_EQ(kBlockSize16kHz, out_len);
+ EXPECT_EQ(kOutputCNG, type);
+ EXPECT_EQ(timestamp - algorithmic_delay_samples, neteq_->PlayoutTimestamp());
+
+ // Insert the same CNG packet again. Note that at this point it is old, since
+ // we have already decoded the first copy of it.
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
+
+ // Pull audio until we have played |kCngPeriodMs| of CNG. Start at 10 ms since
+ // we have already pulled out CNG once.
+ for (int cng_time_ms = 10; cng_time_ms < kCngPeriodMs; cng_time_ms += 10) {
+ ASSERT_EQ(0,
+ neteq_->GetAudio(
+ kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
+ ASSERT_EQ(kBlockSize16kHz, out_len);
+ EXPECT_EQ(kOutputCNG, type);
+ EXPECT_EQ(timestamp - algorithmic_delay_samples,
+ neteq_->PlayoutTimestamp());
+ }
+
+ // Insert speech again.
+ ++seq_no;
+ timestamp += kCngPeriodSamples;
+ PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
+
+ // Pull audio once and verify that the output is speech again.
+ ASSERT_EQ(0,
+ neteq_->GetAudio(
+ kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
+ ASSERT_EQ(kBlockSize16kHz, out_len);
+ EXPECT_EQ(kOutputNormal, type);
+ EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples,
+ neteq_->PlayoutTimestamp());
+}
+
+TEST_F(NetEqDecodingTest, DiscardDuplicateCng) { DuplicateCng(); }
+} // namespace webrtc