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Side by Side Diff: webrtc/modules/media_file/media_file_unittest.cc

Issue 1547343002: Remove DISABLED_ON_ macros. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: MAYBE_ yo Created 4 years, 11 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #include "testing/gtest/include/gtest/gtest.h" 11 #include "testing/gtest/include/gtest/gtest.h"
12 #include "webrtc/modules/media_file/media_file.h" 12 #include "webrtc/modules/media_file/media_file.h"
13 #include "webrtc/system_wrappers/include/sleep.h" 13 #include "webrtc/system_wrappers/include/sleep.h"
14 #include "webrtc/test/testsupport/fileutils.h" 14 #include "webrtc/test/testsupport/fileutils.h"
15 #include "webrtc/test/testsupport/gtest_disable.h"
16 15
17 class MediaFileTest : public testing::Test { 16 class MediaFileTest : public testing::Test {
18 protected: 17 protected:
19 void SetUp() { 18 void SetUp() {
20 // Use number 0 as the the identifier and pass to CreateMediaFile. 19 // Use number 0 as the the identifier and pass to CreateMediaFile.
21 media_file_ = webrtc::MediaFile::CreateMediaFile(0); 20 media_file_ = webrtc::MediaFile::CreateMediaFile(0);
22 ASSERT_TRUE(media_file_ != NULL); 21 ASSERT_TRUE(media_file_ != NULL);
23 } 22 }
24 void TearDown() { 23 void TearDown() {
25 webrtc::MediaFile::DestroyMediaFile(media_file_); 24 webrtc::MediaFile::DestroyMediaFile(media_file_);
26 media_file_ = NULL; 25 media_file_ = NULL;
27 } 26 }
28 webrtc::MediaFile* media_file_; 27 webrtc::MediaFile* media_file_;
29 }; 28 };
30 29
31 TEST_F(MediaFileTest, DISABLED_ON_IOS( 30 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
32 DISABLED_ON_ANDROID(StartPlayingAudioFileWithoutError))) { 31 #define MAYBE_StartPlayingAudioFileWithoutError \
32 DISABLED_StartPlayingAudioFileWithoutError
33 #else
34 #define MAYBE_StartPlayingAudioFileWithoutError \
35 StartPlayingAudioFileWithoutError
36 #endif
37 TEST_F(MediaFileTest, MAYBE_StartPlayingAudioFileWithoutError) {
33 // TODO(leozwang): Use hard coded filename here, we want to 38 // TODO(leozwang): Use hard coded filename here, we want to
34 // loop through all audio files in future 39 // loop through all audio files in future
35 const std::string audio_file = webrtc::test::ProjectRootPath() + 40 const std::string audio_file = webrtc::test::ProjectRootPath() +
36 "data/voice_engine/audio_tiny48.wav"; 41 "data/voice_engine/audio_tiny48.wav";
37 ASSERT_EQ(0, media_file_->StartPlayingAudioFile( 42 ASSERT_EQ(0, media_file_->StartPlayingAudioFile(
38 audio_file.c_str(), 43 audio_file.c_str(),
39 0, 44 0,
40 false, 45 false,
41 webrtc::kFileFormatWavFile)); 46 webrtc::kFileFormatWavFile));
42 47
43 ASSERT_EQ(true, media_file_->IsPlaying()); 48 ASSERT_EQ(true, media_file_->IsPlaying());
44 49
45 webrtc::SleepMs(1); 50 webrtc::SleepMs(1);
46 51
47 ASSERT_EQ(0, media_file_->StopPlaying()); 52 ASSERT_EQ(0, media_file_->StopPlaying());
48 } 53 }
49 54
50 TEST_F(MediaFileTest, DISABLED_ON_IOS(WriteWavFile)) { 55 #if defined(WEBRTC_IOS)
56 #define MAYBE_WriteWavFile DISABLED_WriteWavFile
57 #else
58 #define MAYBE_WriteWavFile WriteWavFile
59 #endif
60 TEST_F(MediaFileTest, MAYBE_WriteWavFile) {
51 // Write file. 61 // Write file.
52 static const size_t kHeaderSize = 44; 62 static const size_t kHeaderSize = 44;
53 static const size_t kPayloadSize = 320; 63 static const size_t kPayloadSize = 320;
54 webrtc::CodecInst codec = { 64 webrtc::CodecInst codec = {
55 0, "L16", 16000, static_cast<int>(kPayloadSize), 1 65 0, "L16", 16000, static_cast<int>(kPayloadSize), 1
56 }; 66 };
57 std::string outfile = webrtc::test::OutputPath() + "wavtest.wav"; 67 std::string outfile = webrtc::test::OutputPath() + "wavtest.wav";
58 ASSERT_EQ(0, 68 ASSERT_EQ(0,
59 media_file_->StartRecordingAudioFile( 69 media_file_->StartRecordingAudioFile(
60 outfile.c_str(), webrtc::kFileFormatWavFile, codec)); 70 outfile.c_str(), webrtc::kFileFormatWavFile, codec));
(...skipping 26 matching lines...) Expand all
87 uint8_t header[kHeaderSize]; 97 uint8_t header[kHeaderSize];
88 ASSERT_EQ(1u, fread(header, kHeaderSize, 1, f)); 98 ASSERT_EQ(1u, fread(header, kHeaderSize, 1, f));
89 EXPECT_EQ(0, memcmp(kExpectedHeader, header, kHeaderSize)); 99 EXPECT_EQ(0, memcmp(kExpectedHeader, header, kHeaderSize));
90 100
91 uint8_t payload[kPayloadSize]; 101 uint8_t payload[kPayloadSize];
92 ASSERT_EQ(1u, fread(payload, kPayloadSize, 1, f)); 102 ASSERT_EQ(1u, fread(payload, kPayloadSize, 1, f));
93 EXPECT_EQ(0, memcmp(kFakeData, payload, kPayloadSize)); 103 EXPECT_EQ(0, memcmp(kFakeData, payload, kPayloadSize));
94 104
95 EXPECT_EQ(0, fclose(f)); 105 EXPECT_EQ(0, fclose(f));
96 } 106 }
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