1 files changed, 106 insertions, 0 deletions
diff --git a/webrtc/modules/media_file/media_file_unittest.cc b/webrtc/modules/media_file/media_file_unittest.cc
new file mode 100644
index 0000000000..6541a8fb7c
--- /dev/null
+++ b/webrtc/modules/media_file/media_file_unittest.cc
@@ -0,0 +1,106 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/media_file/media_file.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+class MediaFileTest : public testing::Test {
+ protected:
+  void SetUp() {
+    // Use number 0 as the the identifier and pass to CreateMediaFile.
+    media_file_ = webrtc::MediaFile::CreateMediaFile(0);
+    ASSERT_TRUE(media_file_ != NULL);
+  }
+  void TearDown() {
+    webrtc::MediaFile::DestroyMediaFile(media_file_);
+    media_file_ = NULL;
+  }
+  webrtc::MediaFile* media_file_;
+};
+
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+#define MAYBE_StartPlayingAudioFileWithoutError \
+  DISABLED_StartPlayingAudioFileWithoutError
+#else
+#define MAYBE_StartPlayingAudioFileWithoutError \
+  StartPlayingAudioFileWithoutError
+#endif
+TEST_F(MediaFileTest, MAYBE_StartPlayingAudioFileWithoutError) {
+  // TODO(leozwang): Use hard coded filename here, we want to
+  // loop through all audio files in future
+  const std::string audio_file = webrtc::test::ProjectRootPath() +
+      "data/voice_engine/audio_tiny48.wav";
+  ASSERT_EQ(0, media_file_->StartPlayingAudioFile(
+      audio_file.c_str(),
+      0,
+      false,
+      webrtc::kFileFormatWavFile));
+
+  ASSERT_EQ(true, media_file_->IsPlaying());
+
+  webrtc::SleepMs(1);
+
+  ASSERT_EQ(0, media_file_->StopPlaying());
+}
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteWavFile DISABLED_WriteWavFile
+#else
+#define MAYBE_WriteWavFile WriteWavFile
+#endif
+TEST_F(MediaFileTest, MAYBE_WriteWavFile) {
+  // Write file.
+  static const size_t kHeaderSize = 44;
+  static const size_t kPayloadSize = 320;
+  webrtc::CodecInst codec = {
+    0, "L16", 16000, static_cast<int>(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
+  };
+  static_assert(sizeof(kExpectedHeader) == kHeaderSize, "header size");
+
+  EXPECT_EQ(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));
+}