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authorkwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2014-07-17 08:11:32 +0000
committerkwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2014-07-17 08:11:32 +0000
commitbde2bcb7526de9c7b9f4edf7f0f25b257200f648 (patch)
treefba4bf233fe1b8401b69ccc23c1c9392272cd99a /modules
parentfbdd355e5f6b9f7e9571c482fcca2ba5d7ba3446 (diff)
downloadwebrtc-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.cc48
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));
+}