92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
107 lines
3.5 KiB
C++
107 lines
3.5 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/media_file/media_file.h"
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#include "system_wrappers/include/sleep.h"
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#include "test/gtest.h"
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#include "test/testsupport/fileutils.h"
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class MediaFileTest : public testing::Test {
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protected:
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void SetUp() {
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// Use number 0 as the the identifier and pass to CreateMediaFile.
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media_file_ = webrtc::MediaFile::CreateMediaFile(0);
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ASSERT_TRUE(media_file_ != NULL);
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}
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void TearDown() {
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webrtc::MediaFile::DestroyMediaFile(media_file_);
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media_file_ = NULL;
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}
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webrtc::MediaFile* media_file_;
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};
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#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
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#define MAYBE_StartPlayingAudioFileWithoutError \
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DISABLED_StartPlayingAudioFileWithoutError
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#else
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#define MAYBE_StartPlayingAudioFileWithoutError \
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StartPlayingAudioFileWithoutError
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#endif
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TEST_F(MediaFileTest, MAYBE_StartPlayingAudioFileWithoutError) {
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// TODO(leozwang): Use hard coded filename here, we want to
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// loop through all audio files in future
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const std::string audio_file =
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webrtc::test::ResourcePath("voice_engine/audio_tiny48", "wav");
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ASSERT_EQ(0, media_file_->StartPlayingAudioFile(
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audio_file.c_str(),
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0,
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false,
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webrtc::kFileFormatWavFile));
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ASSERT_EQ(true, media_file_->IsPlaying());
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webrtc::SleepMs(1);
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ASSERT_EQ(0, media_file_->StopPlaying());
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}
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#if defined(WEBRTC_IOS)
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#define MAYBE_WriteWavFile DISABLED_WriteWavFile
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#else
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#define MAYBE_WriteWavFile WriteWavFile
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#endif
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TEST_F(MediaFileTest, MAYBE_WriteWavFile) {
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// Write file.
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static const size_t kHeaderSize = 44;
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static const size_t kPayloadSize = 320;
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webrtc::CodecInst codec = {
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0, "L16", 16000, static_cast<int>(kPayloadSize), 1
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};
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std::string outfile = webrtc::test::OutputPath() + "wavtest.wav";
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ASSERT_EQ(0,
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media_file_->StartRecordingAudioFile(
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outfile.c_str(), webrtc::kFileFormatWavFile, codec));
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static const int8_t kFakeData[kPayloadSize] = {0};
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ASSERT_EQ(0, media_file_->IncomingAudioData(kFakeData, kPayloadSize));
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ASSERT_EQ(0, media_file_->StopRecording());
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// Check the file we just wrote.
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static const uint8_t kExpectedHeader[] = {
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'R', 'I', 'F', 'F',
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0x64, 0x1, 0, 0, // size of whole file - 8: 320 + 44 - 8
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'W', 'A', 'V', 'E',
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'f', 'm', 't', ' ',
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0x10, 0, 0, 0, // size of fmt block - 8: 24 - 8
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0x1, 0, // format: PCM (1)
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0x1, 0, // channels: 1
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0x80, 0x3e, 0, 0, // sample rate: 16000
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0, 0x7d, 0, 0, // byte rate: 2 * 16000
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0x2, 0, // block align: NumChannels * BytesPerSample
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0x10, 0, // bits per sample: 2 * 8
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'd', 'a', 't', 'a',
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0x40, 0x1, 0, 0, // size of payload: 320
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};
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static_assert(sizeof(kExpectedHeader) == kHeaderSize, "header size");
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EXPECT_EQ(kHeaderSize + kPayloadSize, webrtc::test::GetFileSize(outfile));
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FILE* f = fopen(outfile.c_str(), "rb");
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ASSERT_TRUE(f);
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uint8_t header[kHeaderSize];
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ASSERT_EQ(1u, fread(header, kHeaderSize, 1, f));
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EXPECT_EQ(0, memcmp(kExpectedHeader, header, kHeaderSize));
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uint8_t payload[kPayloadSize];
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ASSERT_EQ(1u, fread(payload, kPayloadSize, 1, f));
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EXPECT_EQ(0, memcmp(kFakeData, payload, kPayloadSize));
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EXPECT_EQ(0, fclose(f));
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}
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