henrik.lundin@webrtc.org | 9a40081 | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | |
| 11 | // Test to verify correct operation for externally created decoders. |
| 12 | |
| 13 | #include <string> |
| 14 | #include <list> |
| 15 | |
| 16 | #include "gmock/gmock.h" |
| 17 | #include "gtest/gtest.h" |
| 18 | #include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" |
| 19 | #include "webrtc/modules/audio_coding/neteq4/mock/mock_external_decoder_pcm16b.h" |
| 20 | #include "webrtc/modules/audio_coding/neteq4/tools/input_audio_file.h" |
| 21 | #include "webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h" |
| 22 | #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| 23 | #include "webrtc/test/testsupport/fileutils.h" |
henrike@webrtc.org | 7537dde | 2013-07-08 18:53:54 +0000 | [diff] [blame] | 24 | #include "webrtc/test/testsupport/gtest_disable.h" |
henrik.lundin@webrtc.org | 9a40081 | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 25 | |
| 26 | namespace webrtc { |
| 27 | |
| 28 | using ::testing::_; |
| 29 | |
| 30 | // This test encodes a few packets of PCM16b 32 kHz data and inserts it into two |
| 31 | // different NetEq instances. The first instance uses the internal version of |
| 32 | // the decoder object, while the second one uses an externally created decoder |
| 33 | // object (ExternalPcm16B wrapped in MockExternalPcm16B, both defined above). |
| 34 | // The test verifies that the output from both instances match. |
| 35 | class NetEqExternalDecoderTest : public ::testing::Test { |
| 36 | protected: |
| 37 | static const int kTimeStepMs = 10; |
| 38 | static const int kMaxBlockSize = 480; // 10 ms @ 48 kHz. |
| 39 | static const uint8_t kPayloadType = 95; |
| 40 | static const int kSampleRateHz = 32000; |
| 41 | |
| 42 | NetEqExternalDecoderTest() |
| 43 | : sample_rate_hz_(kSampleRateHz), |
| 44 | samples_per_ms_(sample_rate_hz_ / 1000), |
| 45 | frame_size_ms_(10), |
| 46 | frame_size_samples_(frame_size_ms_ * samples_per_ms_), |
| 47 | output_size_samples_(frame_size_ms_ * samples_per_ms_), |
| 48 | neteq_external_(NetEq::Create(sample_rate_hz_)), |
| 49 | neteq_(NetEq::Create(sample_rate_hz_)), |
| 50 | external_decoder_(new MockExternalPcm16B(kDecoderPCM16Bswb32kHz)), |
| 51 | rtp_generator_(samples_per_ms_), |
| 52 | payload_size_bytes_(0), |
| 53 | last_send_time_(0), |
| 54 | last_arrival_time_(0) { |
| 55 | input_ = new int16_t[frame_size_samples_]; |
| 56 | encoded_ = new uint8_t[2 * frame_size_samples_]; |
| 57 | } |
| 58 | |
| 59 | ~NetEqExternalDecoderTest() { |
| 60 | delete neteq_external_; |
| 61 | delete neteq_; |
| 62 | // We will now delete the decoder ourselves, so expecting Die to be called. |
| 63 | EXPECT_CALL(*external_decoder_, Die()).Times(1); |
| 64 | delete external_decoder_; |
| 65 | delete [] input_; |
| 66 | delete [] encoded_; |
| 67 | } |
| 68 | |
| 69 | virtual void SetUp() { |
| 70 | const std::string file_name = |
| 71 | webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); |
| 72 | input_file_.reset(new test::InputAudioFile(file_name)); |
| 73 | assert(sample_rate_hz_ == 32000); |
| 74 | NetEqDecoder decoder = kDecoderPCM16Bswb32kHz; |
| 75 | EXPECT_CALL(*external_decoder_, Init()); |
| 76 | // NetEq is not allowed to delete the external decoder (hence Times(0)). |
| 77 | EXPECT_CALL(*external_decoder_, Die()).Times(0); |
| 78 | ASSERT_EQ(NetEq::kOK, |
| 79 | neteq_external_->RegisterExternalDecoder(external_decoder_, |
| 80 | decoder, |
| 81 | sample_rate_hz_, |
| 82 | kPayloadType)); |
| 83 | ASSERT_EQ(NetEq::kOK, |
| 84 | neteq_->RegisterPayloadType(decoder, kPayloadType)); |
| 85 | } |
| 86 | |
| 87 | virtual void TearDown() {} |
| 88 | |
| 89 | int GetNewPackets() { |
| 90 | if (!input_file_->Read(frame_size_samples_, input_)) { |
| 91 | return -1; |
| 92 | } |
| 93 | payload_size_bytes_ = WebRtcPcm16b_Encode(input_, frame_size_samples_, |
| 94 | encoded_); |
| 95 | if (frame_size_samples_ * 2 != payload_size_bytes_) { |
| 96 | return -1; |
| 97 | } |
| 98 | int next_send_time = rtp_generator_.GetRtpHeader(kPayloadType, |
| 99 | frame_size_samples_, |
| 100 | &rtp_header_); |
| 101 | return next_send_time; |
| 102 | } |
| 103 | |
| 104 | void VerifyOutput(size_t num_samples) { |
| 105 | for (size_t i = 0; i < num_samples; ++i) { |
| 106 | ASSERT_EQ(output_[i], output_external_[i]) << |
| 107 | "Diff in sample " << i << "."; |
| 108 | } |
| 109 | } |
| 110 | |
| 111 | virtual int GetArrivalTime(int send_time) { |
| 112 | int arrival_time = last_arrival_time_ + (send_time - last_send_time_); |
| 113 | last_send_time_ = send_time; |
| 114 | last_arrival_time_ = arrival_time; |
| 115 | return arrival_time; |
| 116 | } |
| 117 | |
| 118 | virtual bool Lost() { return false; } |
| 119 | |
| 120 | void RunTest(int num_loops) { |
| 121 | // Get next input packets (mono and multi-channel). |
| 122 | int next_send_time; |
| 123 | int next_arrival_time; |
| 124 | do { |
| 125 | next_send_time = GetNewPackets(); |
| 126 | ASSERT_NE(-1, next_send_time); |
| 127 | next_arrival_time = GetArrivalTime(next_send_time); |
| 128 | } while (Lost()); // If lost, immediately read the next packet. |
| 129 | |
| 130 | EXPECT_CALL(*external_decoder_, Decode(_, payload_size_bytes_, _, _)) |
| 131 | .Times(num_loops); |
| 132 | |
| 133 | int time_now = 0; |
| 134 | for (int k = 0; k < num_loops; ++k) { |
| 135 | while (time_now >= next_arrival_time) { |
| 136 | // Insert packet in regular instance. |
| 137 | ASSERT_EQ(NetEq::kOK, |
| 138 | neteq_->InsertPacket(rtp_header_, encoded_, |
| 139 | payload_size_bytes_, |
| 140 | next_arrival_time)); |
| 141 | // Insert packet in external decoder instance. |
| 142 | EXPECT_CALL(*external_decoder_, |
| 143 | IncomingPacket(_, payload_size_bytes_, |
| 144 | rtp_header_.header.sequenceNumber, |
| 145 | rtp_header_.header.timestamp, |
| 146 | next_arrival_time)); |
| 147 | ASSERT_EQ(NetEq::kOK, |
| 148 | neteq_external_->InsertPacket(rtp_header_, encoded_, |
| 149 | payload_size_bytes_, |
| 150 | next_arrival_time)); |
| 151 | // Get next input packet. |
| 152 | do { |
| 153 | next_send_time = GetNewPackets(); |
| 154 | ASSERT_NE(-1, next_send_time); |
| 155 | next_arrival_time = GetArrivalTime(next_send_time); |
| 156 | } while (Lost()); // If lost, immediately read the next packet. |
| 157 | } |
| 158 | NetEqOutputType output_type; |
| 159 | // Get audio from regular instance. |
| 160 | int samples_per_channel; |
| 161 | int num_channels; |
| 162 | EXPECT_EQ(NetEq::kOK, |
| 163 | neteq_->GetAudio(kMaxBlockSize, output_, |
| 164 | &samples_per_channel, &num_channels, |
| 165 | &output_type)); |
| 166 | EXPECT_EQ(1, num_channels); |
| 167 | EXPECT_EQ(output_size_samples_, samples_per_channel); |
| 168 | // Get audio from external decoder instance. |
| 169 | ASSERT_EQ(NetEq::kOK, |
| 170 | neteq_external_->GetAudio(kMaxBlockSize, output_external_, |
| 171 | &samples_per_channel, &num_channels, |
| 172 | &output_type)); |
| 173 | EXPECT_EQ(1, num_channels); |
| 174 | EXPECT_EQ(output_size_samples_, samples_per_channel); |
| 175 | std::ostringstream ss; |
| 176 | ss << "Lap number " << k << "."; |
| 177 | SCOPED_TRACE(ss.str()); // Print out the parameter values on failure. |
| 178 | // Compare mono and multi-channel. |
| 179 | ASSERT_NO_FATAL_FAILURE(VerifyOutput(output_size_samples_)); |
| 180 | |
| 181 | time_now += kTimeStepMs; |
| 182 | } |
| 183 | } |
| 184 | |
| 185 | const int sample_rate_hz_; |
| 186 | const int samples_per_ms_; |
| 187 | const int frame_size_ms_; |
| 188 | const int frame_size_samples_; |
| 189 | const int output_size_samples_; |
| 190 | NetEq* neteq_external_; |
| 191 | NetEq* neteq_; |
| 192 | MockExternalPcm16B* external_decoder_; |
| 193 | test::RtpGenerator rtp_generator_; |
| 194 | int16_t* input_; |
| 195 | uint8_t* encoded_; |
| 196 | int16_t output_[kMaxBlockSize]; |
| 197 | int16_t output_external_[kMaxBlockSize]; |
| 198 | WebRtcRTPHeader rtp_header_; |
| 199 | int payload_size_bytes_; |
| 200 | int last_send_time_; |
| 201 | int last_arrival_time_; |
| 202 | scoped_ptr<test::InputAudioFile> input_file_; |
| 203 | }; |
| 204 | |
henrike@webrtc.org | 7537dde | 2013-07-08 18:53:54 +0000 | [diff] [blame] | 205 | TEST_F(NetEqExternalDecoderTest, DISABLED_ON_ANDROID(RunTest)) { |
henrik.lundin@webrtc.org | 9a40081 | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 206 | RunTest(100); // Run 100 laps @ 10 ms each in the test loop. |
| 207 | } |
| 208 | |
| 209 | } // namespace webrtc |