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author | kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2014-07-17 08:11:32 +0000 |
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committer | kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2014-07-17 08:11:32 +0000 |
commit | bde2bcb7526de9c7b9f4edf7f0f25b257200f648 (patch) | |
tree | fba4bf233fe1b8401b69ccc23c1c9392272cd99a /modules | |
parent | fbdd355e5f6b9f7e9571c482fcca2ba5d7ba3446 (diff) | |
download | webrtc-bde2bcb7526de9c7b9f4edf7f0f25b257200f648.tar.gz |
Add unit test for MediaFile WAV file writing
R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6713 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'modules')
-rw-r--r-- | modules/media_file/source/media_file_unittest.cc | 48 |
1 files changed, 48 insertions, 0 deletions
diff --git a/modules/media_file/source/media_file_unittest.cc b/modules/media_file/source/media_file_unittest.cc index d658dc2c..56d3544c 100644 --- a/modules/media_file/source/media_file_unittest.cc +++ b/modules/media_file/source/media_file_unittest.cc @@ -10,6 +10,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/modules/media_file/interface/media_file.h" +#include "webrtc/system_wrappers/interface/compile_assert.h" #include "webrtc/system_wrappers/interface/sleep.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/gtest_disable.h" @@ -45,3 +46,50 @@ TEST_F(MediaFileTest, DISABLED_ON_ANDROID(StartPlayingAudioFileWithoutError)) { ASSERT_EQ(0, media_file_->StopPlaying()); } + +TEST_F(MediaFileTest, WriteWavFile) { + // Write file. + static const int kHeaderSize = 44; + static const int kPayloadSize = 320; + webrtc::CodecInst codec = {0, "L16", 16000, kPayloadSize, 1}; + std::string outfile = webrtc::test::OutputPath() + "wavtest.wav"; + ASSERT_EQ(0, + media_file_->StartRecordingAudioFile( + outfile.c_str(), webrtc::kFileFormatWavFile, codec)); + static const int8_t kFakeData[kPayloadSize] = {0}; + ASSERT_EQ(0, media_file_->IncomingAudioData(kFakeData, kPayloadSize)); + ASSERT_EQ(0, media_file_->StopRecording()); + + // Check the file we just wrote. + static const uint8_t kExpectedHeader[] = { + 'R', 'I', 'F', 'F', + 0x64, 0x1, 0, 0, // size of whole file - 8: 320 + 44 - 8 + 'W', 'A', 'V', 'E', + 'f', 'm', 't', ' ', + 0x10, 0, 0, 0, // size of fmt block - 8: 24 - 8 + 0x1, 0, // format: PCM (1) + 0x1, 0, // channels: 1 + 0x80, 0x3e, 0, 0, // sample rate: 16000 + 0, 0x7d, 0, 0, // byte rate: 2 * 16000 + 0x2, 0, // block align: NumChannels * BytesPerSample + 0x10, 0, // bits per sample: 2 * 8 + 'd', 'a', 't', 'a', + 0x40, 0x1, 0, 0, // size of payload: 320 + }; + COMPILE_ASSERT(sizeof(kExpectedHeader) == kHeaderSize, header_size); + + EXPECT_EQ(size_t(kHeaderSize + kPayloadSize), + webrtc::test::GetFileSize(outfile)); + FILE* f = fopen(outfile.c_str(), "rb"); + ASSERT_TRUE(f); + + uint8_t header[kHeaderSize]; + ASSERT_EQ(1u, fread(header, kHeaderSize, 1, f)); + EXPECT_EQ(0, memcmp(kExpectedHeader, header, kHeaderSize)); + + uint8_t payload[kPayloadSize]; + ASSERT_EQ(1u, fread(payload, kPayloadSize, 1, f)); + EXPECT_EQ(0, memcmp(kFakeData, payload, kPayloadSize)); + + EXPECT_EQ(0, fclose(f)); +} |