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/*
* 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));
}
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