1 /* 2 * Copyright (c) 2012 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 "testing/gtest/include/gtest/gtest.h" 12 #include "webrtc/modules/media_file/media_file.h" 13 #include "webrtc/system_wrappers/include/sleep.h" 14 #include "webrtc/test/testsupport/fileutils.h" 15 16 class MediaFileTest : public testing::Test { 17 protected: 18 void SetUp() { 19 // Use number 0 as the the identifier and pass to CreateMediaFile. 20 media_file_ = webrtc::MediaFile::CreateMediaFile(0); 21 ASSERT_TRUE(media_file_ != NULL); 22 } 23 void TearDown() { 24 webrtc::MediaFile::DestroyMediaFile(media_file_); 25 media_file_ = NULL; 26 } 27 webrtc::MediaFile* media_file_; 28 }; 29 30 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) 31 #define MAYBE_StartPlayingAudioFileWithoutError \ 32 DISABLED_StartPlayingAudioFileWithoutError 33 #else 34 #define MAYBE_StartPlayingAudioFileWithoutError \ 35 StartPlayingAudioFileWithoutError 36 #endif 37 TEST_F(MediaFileTest, MAYBE_StartPlayingAudioFileWithoutError) { 38 // TODO(leozwang): Use hard coded filename here, we want to 39 // loop through all audio files in future 40 const std::string audio_file = webrtc::test::ProjectRootPath() + 41 "data/voice_engine/audio_tiny48.wav"; 42 ASSERT_EQ(0, media_file_->StartPlayingAudioFile( 43 audio_file.c_str(), 44 0, 45 false, 46 webrtc::kFileFormatWavFile)); 47 48 ASSERT_EQ(true, media_file_->IsPlaying()); 49 50 webrtc::SleepMs(1); 51 52 ASSERT_EQ(0, media_file_->StopPlaying()); 53 } 54 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) { 61 // Write file. 62 static const size_t kHeaderSize = 44; 63 static const size_t kPayloadSize = 320; 64 webrtc::CodecInst codec = { 65 0, "L16", 16000, static_cast<int>(kPayloadSize), 1 66 }; 67 std::string outfile = webrtc::test::OutputPath() + "wavtest.wav"; 68 ASSERT_EQ(0, 69 media_file_->StartRecordingAudioFile( 70 outfile.c_str(), webrtc::kFileFormatWavFile, codec)); 71 static const int8_t kFakeData[kPayloadSize] = {0}; 72 ASSERT_EQ(0, media_file_->IncomingAudioData(kFakeData, kPayloadSize)); 73 ASSERT_EQ(0, media_file_->StopRecording()); 74 75 // Check the file we just wrote. 76 static const uint8_t kExpectedHeader[] = { 77 'R', 'I', 'F', 'F', 78 0x64, 0x1, 0, 0, // size of whole file - 8: 320 + 44 - 8 79 'W', 'A', 'V', 'E', 80 'f', 'm', 't', ' ', 81 0x10, 0, 0, 0, // size of fmt block - 8: 24 - 8 82 0x1, 0, // format: PCM (1) 83 0x1, 0, // channels: 1 84 0x80, 0x3e, 0, 0, // sample rate: 16000 85 0, 0x7d, 0, 0, // byte rate: 2 * 16000 86 0x2, 0, // block align: NumChannels * BytesPerSample 87 0x10, 0, // bits per sample: 2 * 8 88 'd', 'a', 't', 'a', 89 0x40, 0x1, 0, 0, // size of payload: 320 90 }; 91 static_assert(sizeof(kExpectedHeader) == kHeaderSize, "header size"); 92 93 EXPECT_EQ(kHeaderSize + kPayloadSize, webrtc::test::GetFileSize(outfile)); 94 FILE* f = fopen(outfile.c_str(), "rb"); 95 ASSERT_TRUE(f); 96 97 uint8_t header[kHeaderSize]; 98 ASSERT_EQ(1u, fread(header, kHeaderSize, 1, f)); 99 EXPECT_EQ(0, memcmp(kExpectedHeader, header, kHeaderSize)); 100 101 uint8_t payload[kPayloadSize]; 102 ASSERT_EQ(1u, fread(payload, kPayloadSize, 1, f)); 103 EXPECT_EQ(0, memcmp(kFakeData, payload, kPayloadSize)); 104 105 EXPECT_EQ(0, fclose(f)); 106 } 107
