1 /*
2 * Copyright (c) 2011 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 #include "modules/audio_coding/neteq/test/result_sink.h"
12
13 #include <vector>
14
15 #include "rtc_base/ignore_wundef.h"
16 #include "rtc_base/message_digest.h"
17 #include "rtc_base/string_encode.h"
18 #include "test/gtest.h"
19
20 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
21 RTC_PUSH_IGNORING_WUNDEF()
22 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
23 #include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
24 #else
25 #include "modules/audio_coding/neteq/neteq_unittest.pb.h"
26 #endif
27 RTC_POP_IGNORING_WUNDEF()
28 #endif
29
30 namespace webrtc {
31
32 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
Convert(const webrtc::NetEqNetworkStatistics & stats_raw,webrtc::neteq_unittest::NetEqNetworkStatistics * stats)33 void Convert(const webrtc::NetEqNetworkStatistics& stats_raw,
34 webrtc::neteq_unittest::NetEqNetworkStatistics* stats) {
35 stats->set_current_buffer_size_ms(stats_raw.current_buffer_size_ms);
36 stats->set_preferred_buffer_size_ms(stats_raw.preferred_buffer_size_ms);
37 stats->set_jitter_peaks_found(stats_raw.jitter_peaks_found);
38 stats->set_packet_loss_rate(stats_raw.packet_loss_rate);
39 stats->set_expand_rate(stats_raw.expand_rate);
40 stats->set_speech_expand_rate(stats_raw.speech_expand_rate);
41 stats->set_preemptive_rate(stats_raw.preemptive_rate);
42 stats->set_accelerate_rate(stats_raw.accelerate_rate);
43 stats->set_secondary_decoded_rate(stats_raw.secondary_decoded_rate);
44 stats->set_secondary_discarded_rate(stats_raw.secondary_discarded_rate);
45 stats->set_added_zero_samples(stats_raw.added_zero_samples);
46 stats->set_mean_waiting_time_ms(stats_raw.mean_waiting_time_ms);
47 stats->set_median_waiting_time_ms(stats_raw.median_waiting_time_ms);
48 stats->set_min_waiting_time_ms(stats_raw.min_waiting_time_ms);
49 stats->set_max_waiting_time_ms(stats_raw.max_waiting_time_ms);
50 }
51
Convert(const webrtc::RtcpStatistics & stats_raw,webrtc::neteq_unittest::RtcpStatistics * stats)52 void Convert(const webrtc::RtcpStatistics& stats_raw,
53 webrtc::neteq_unittest::RtcpStatistics* stats) {
54 stats->set_fraction_lost(stats_raw.fraction_lost);
55 stats->set_cumulative_lost(stats_raw.packets_lost);
56 stats->set_extended_max_sequence_number(
57 stats_raw.extended_highest_sequence_number);
58 stats->set_jitter(stats_raw.jitter);
59 }
60
AddMessage(FILE * file,rtc::MessageDigest * digest,const std::string & message)61 void AddMessage(FILE* file,
62 rtc::MessageDigest* digest,
63 const std::string& message) {
64 int32_t size = message.length();
65 if (file)
66 ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
67 digest->Update(&size, sizeof(size));
68
69 if (file)
70 ASSERT_EQ(static_cast<size_t>(size),
71 fwrite(message.data(), sizeof(char), size, file));
72 digest->Update(message.data(), sizeof(char) * size);
73 }
74
75 #endif // WEBRTC_NETEQ_UNITTEST_BITEXACT
76
ResultSink(const std::string & output_file)77 ResultSink::ResultSink(const std::string& output_file)
78 : output_fp_(nullptr),
79 digest_(rtc::MessageDigestFactory::Create(rtc::DIGEST_SHA_1)) {
80 if (!output_file.empty()) {
81 output_fp_ = fopen(output_file.c_str(), "wb");
82 EXPECT_TRUE(output_fp_ != NULL);
83 }
84 }
85
~ResultSink()86 ResultSink::~ResultSink() {
87 if (output_fp_)
88 fclose(output_fp_);
89 }
90
AddResult(const NetEqNetworkStatistics & stats_raw)91 void ResultSink::AddResult(const NetEqNetworkStatistics& stats_raw) {
92 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
93 neteq_unittest::NetEqNetworkStatistics stats;
94 Convert(stats_raw, &stats);
95
96 std::string stats_string;
97 ASSERT_TRUE(stats.SerializeToString(&stats_string));
98 AddMessage(output_fp_, digest_.get(), stats_string);
99 #else
100 FAIL() << "Writing to reference file requires Proto Buffer.";
101 #endif // WEBRTC_NETEQ_UNITTEST_BITEXACT
102 }
103
AddResult(const RtcpStatistics & stats_raw)104 void ResultSink::AddResult(const RtcpStatistics& stats_raw) {
105 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
106 neteq_unittest::RtcpStatistics stats;
107 Convert(stats_raw, &stats);
108
109 std::string stats_string;
110 ASSERT_TRUE(stats.SerializeToString(&stats_string));
111 AddMessage(output_fp_, digest_.get(), stats_string);
112 #else
113 FAIL() << "Writing to reference file requires Proto Buffer.";
114 #endif // WEBRTC_NETEQ_UNITTEST_BITEXACT
115 }
116
VerifyChecksum(const std::string & checksum)117 void ResultSink::VerifyChecksum(const std::string& checksum) {
118 std::vector<char> buffer;
119 buffer.resize(digest_->Size());
120 digest_->Finish(&buffer[0], buffer.size());
121 const std::string result = rtc::hex_encode(&buffer[0], digest_->Size());
122 if (checksum.size() == result.size()) {
123 EXPECT_EQ(checksum, result);
124 } else {
125 // Check result is one the '|'-separated checksums.
126 EXPECT_NE(checksum.find(result), std::string::npos)
127 << result << " should be one of these:\n"
128 << checksum;
129 }
130 }
131
132 } // namespace webrtc
133