Remove include dirs from modules/{media_file,pacing}

webrtc/modules/media_file/source/media_file_unittest.cc

-/*
- *  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/include/media_file.h"
-#include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.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_;
-};
-
-TEST_F(MediaFileTest, DISABLED_ON_IOS(
-       DISABLED_ON_ANDROID(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());
-}
-
-TEST_F(MediaFileTest, DISABLED_ON_IOS(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));
-}