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| 1 /* |
| 2 * Copyright (c) 2015 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 #ifdef RTC_AUDIOCODING_DEBUG_DUMP |
| 12 |
| 13 #include <stdio.h> |
| 14 #include <string> |
| 15 #include <vector> |
| 16 |
| 17 #include "testing/gtest/include/gtest/gtest.h" |
| 18 #include "webrtc/base/scoped_ptr.h" |
| 19 #include "webrtc/modules/audio_coding/main/acm2/acm_dump.h" |
| 20 #include "webrtc/system_wrappers/interface/clock.h" |
| 21 #include "webrtc/test/test_suite.h" |
| 22 #include "webrtc/test/testsupport/fileutils.h" |
| 23 #include "webrtc/test/testsupport/gtest_disable.h" |
| 24 |
| 25 // Files generated at build-time by the protobuf compiler. |
| 26 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| 27 #include "external/webrtc/webrtc/modules/audio_coding/dump.pb.h" |
| 28 #else |
| 29 #include "webrtc/audio_coding/dump.pb.h" |
| 30 #endif |
| 31 |
| 32 namespace webrtc { |
| 33 |
| 34 // Test for the acm dump class. Dumps some RTP packets to disk, then reads them |
| 35 // back to see if they match. |
| 36 class AcmDumpTest : public ::testing::Test { |
| 37 public: |
| 38 void VerifyResults(const ACMDumpEventStream& parsed_stream, |
| 39 size_t packet_size) { |
| 40 // Verify the result. |
| 41 EXPECT_EQ(5, parsed_stream.stream_size()); |
| 42 const ACMDumpEvent& start_event = parsed_stream.stream(2); |
| 43 ASSERT_TRUE(start_event.has_type()); |
| 44 EXPECT_EQ(ACMDumpEvent::DEBUG_EVENT, start_event.type()); |
| 45 EXPECT_TRUE(start_event.has_timestamp_us()); |
| 46 EXPECT_FALSE(start_event.has_packet()); |
| 47 ASSERT_TRUE(start_event.has_debug_event()); |
| 48 auto start_debug_event = start_event.debug_event(); |
| 49 ASSERT_TRUE(start_debug_event.has_type()); |
| 50 EXPECT_EQ(ACMDumpDebugEvent::LOG_START, start_debug_event.type()); |
| 51 ASSERT_TRUE(start_debug_event.has_message()); |
| 52 |
| 53 for (int i = 0; i < parsed_stream.stream_size(); i++) { |
| 54 if (i == 2) { |
| 55 // This is the LOG_START packet that was already verified. |
| 56 continue; |
| 57 } |
| 58 const ACMDumpEvent& test_event = parsed_stream.stream(i); |
| 59 ASSERT_TRUE(test_event.has_type()); |
| 60 EXPECT_EQ(ACMDumpEvent::RTP_EVENT, test_event.type()); |
| 61 EXPECT_TRUE(test_event.has_timestamp_us()); |
| 62 EXPECT_FALSE(test_event.has_debug_event()); |
| 63 ASSERT_TRUE(test_event.has_packet()); |
| 64 const ACMDumpRTPPacket& test_packet = test_event.packet(); |
| 65 ASSERT_TRUE(test_packet.has_direction()); |
| 66 if (i <= 1) { |
| 67 EXPECT_EQ(ACMDumpRTPPacket::INCOMING, test_packet.direction()); |
| 68 } else if (i >= 3) { |
| 69 EXPECT_EQ(ACMDumpRTPPacket::OUTGOING, test_packet.direction()); |
| 70 } |
| 71 ASSERT_TRUE(test_packet.has_rtp_data()); |
| 72 ASSERT_EQ(packet_size, test_packet.rtp_data().size()); |
| 73 for (size_t i = 0; i < packet_size; i++) { |
| 74 EXPECT_EQ(rtp_packet_[i], |
| 75 static_cast<uint8_t>(test_packet.rtp_data()[i])); |
| 76 } |
| 77 } |
| 78 } |
| 79 |
| 80 void Run(int packet_size, int random_seed) { |
| 81 rtp_packet_.clear(); |
| 82 rtp_packet_.reserve(packet_size); |
| 83 srand(random_seed); |
| 84 // Fill the packet vector with random data. |
| 85 for (int i = 0; i < packet_size; i++) { |
| 86 rtp_packet_.push_back(rand()); |
| 87 } |
| 88 // Find the name of the current test, in order to use it as a temporary |
| 89 // filename. |
| 90 auto test_info = ::testing::UnitTest::GetInstance()->current_test_info(); |
| 91 const std::string temp_filename = |
| 92 test::OutputPath() + test_info->test_case_name() + test_info->name(); |
| 93 |
| 94 // When log_dumper goes out of scope, it causes the log file to be flushed |
| 95 // to disk. |
| 96 { |
| 97 rtc::scoped_ptr<AcmDump> log_dumper(AcmDump::Create()); |
| 98 log_dumper->LogRtpPacket(true, rtp_packet_.data(), rtp_packet_.size()); |
| 99 log_dumper->LogRtpPacket(true, rtp_packet_.data(), rtp_packet_.size()); |
| 100 log_dumper->StartLogging(temp_filename, 10000000); |
| 101 log_dumper->LogRtpPacket(false, rtp_packet_.data(), rtp_packet_.size()); |
| 102 log_dumper->LogRtpPacket(false, rtp_packet_.data(), rtp_packet_.size()); |
| 103 } |
| 104 |
| 105 // Read the generated file from disk. |
| 106 ACMDumpEventStream parsed_stream; |
| 107 |
| 108 ASSERT_EQ(true, AcmDump::ParseAcmDump(temp_filename, &parsed_stream)); |
| 109 |
| 110 VerifyResults(parsed_stream, packet_size); |
| 111 |
| 112 // Clean up temporary file - can be pretty slow. |
| 113 remove(temp_filename.c_str()); |
| 114 } |
| 115 std::vector<uint8_t> rtp_packet_; |
| 116 }; |
| 117 |
| 118 TEST_F(AcmDumpTest, DumpAndRead) { |
| 119 Run(256, 321); |
| 120 } |
| 121 |
| 122 } // namespace webrtc |
| 123 |
| 124 #endif // RTC_AUDIOCODING_DEBUG_DUMP |
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