1 /*
2 * Copyright (c) 2014 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 // Unit tests for FilePlayer.
12
13 #include "webrtc/voice_engine/file_player.h"
14
15 #include <stdio.h>
16
17 #include <memory>
18 #include <string>
19
20 #include "gflags/gflags.h"
21 #include "webrtc/base/md5digest.h"
22 #include "webrtc/base/stringencode.h"
23 #include "webrtc/test/gtest.h"
24 #include "webrtc/test/testsupport/fileutils.h"
25
26 DEFINE_bool(file_player_output, false, "Generate reference files.");
27
28 namespace webrtc {
29
30 class FilePlayerTest : public ::testing::Test {
31 protected:
32 static const uint32_t kId = 0;
33 static const FileFormats kFileFormat = kFileFormatWavFile;
34 static const int kSampleRateHz = 8000;
35
FilePlayerTest()36 FilePlayerTest()
37 : player_(FilePlayer::CreateFilePlayer(kId, kFileFormat)),
38 output_file_(NULL) {}
39
SetUp()40 void SetUp() override {
41 if (FLAGS_file_player_output) {
42 std::string output_file =
43 webrtc::test::OutputPath() + "file_player_unittest_out.pcm";
44 output_file_ = fopen(output_file.c_str(), "wb");
45 ASSERT_TRUE(output_file_ != NULL);
46 }
47 }
48
TearDown()49 void TearDown() override {
50 if (output_file_)
51 fclose(output_file_);
52 }
53
PlayFileAndCheck(const std::string & input_file,const std::string & ref_checksum,int output_length_ms)54 void PlayFileAndCheck(const std::string& input_file,
55 const std::string& ref_checksum,
56 int output_length_ms) {
57 const float kScaling = 1;
58 ASSERT_EQ(0, player_->StartPlayingFile(input_file.c_str(), false, 0,
59 kScaling, 0, 0, NULL));
60 rtc::Md5Digest checksum;
61 for (int i = 0; i < output_length_ms / 10; ++i) {
62 int16_t out[10 * kSampleRateHz / 1000] = {0};
63 size_t num_samples;
64 EXPECT_EQ(
65 0, player_->Get10msAudioFromFile(out, &num_samples, kSampleRateHz));
66 checksum.Update(out, num_samples * sizeof(out[0]));
67 if (FLAGS_file_player_output) {
68 ASSERT_EQ(num_samples,
69 fwrite(out, sizeof(out[0]), num_samples, output_file_));
70 }
71 }
72 char checksum_result[rtc::Md5Digest::kSize];
73 EXPECT_EQ(rtc::Md5Digest::kSize,
74 checksum.Finish(checksum_result, rtc::Md5Digest::kSize));
75 EXPECT_EQ(ref_checksum,
76 rtc::hex_encode(checksum_result, sizeof(checksum_result)));
77 }
78
79 std::unique_ptr<FilePlayer> player_;
80 FILE* output_file_;
81 };
82
83 #if defined(WEBRTC_IOS)
84 #define MAYBE_PlayWavPcmuFile DISABLED_PlayWavPcmuFile
85 #else
86 #define MAYBE_PlayWavPcmuFile PlayWavPcmuFile
87 #endif
TEST_F(FilePlayerTest,MAYBE_PlayWavPcmuFile)88 TEST_F(FilePlayerTest, MAYBE_PlayWavPcmuFile) {
89 const std::string kFileName =
90 test::ResourcePath("utility/encapsulated_pcmu_8khz", "wav");
91 // The file is longer than this, but keeping the output shorter limits the
92 // runtime for the test.
93 const int kOutputLengthMs = 10000;
94 const std::string kRefChecksum = "c74e7fd432d439b1311e1c16815b3e9a";
95
96 PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs);
97 }
98
99 #if defined(WEBRTC_IOS)
100 #define MAYBE_PlayWavPcm16File DISABLED_PlayWavPcm16File
101 #else
102 #define MAYBE_PlayWavPcm16File PlayWavPcm16File
103 #endif
TEST_F(FilePlayerTest,MAYBE_PlayWavPcm16File)104 TEST_F(FilePlayerTest, MAYBE_PlayWavPcm16File) {
105 const std::string kFileName =
106 test::ResourcePath("utility/encapsulated_pcm16b_8khz", "wav");
107 // The file is longer than this, but keeping the output shorter limits the
108 // runtime for the test.
109 const int kOutputLengthMs = 10000;
110 const std::string kRefChecksum = "e41d7e1dac8aeae9f21e8e03cd7ecd71";
111
112 PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs);
113 }
114
115 } // namespace webrtc
116