Remove voe_auto_test cases for VoEFile.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2786083004
Cr-Original-Commit-Position: refs/heads/master@{#17484}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: e6a8009417571bd904c74ba1eaabffa9323ce725
diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
index 233d129..cb604c6 100644
--- a/voice_engine/BUILD.gn
+++ b/voice_engine/BUILD.gn
@@ -271,8 +271,6 @@
"test/auto_test/standard/codec_before_streaming_test.cc",
"test/auto_test/standard/codec_test.cc",
"test/auto_test/standard/dtmf_test.cc",
- "test/auto_test/standard/file_before_streaming_test.cc",
- "test/auto_test/standard/file_test.cc",
"test/auto_test/standard/rtp_rtcp_before_streaming_test.cc",
"test/auto_test/standard/rtp_rtcp_extensions.cc",
"test/auto_test/standard/rtp_rtcp_test.cc",
diff --git a/voice_engine/test/auto_test/standard/file_before_streaming_test.cc b/voice_engine/test/auto_test/standard/file_before_streaming_test.cc
deleted file mode 100644
index 96ac03c..0000000
--- a/voice_engine/test/auto_test/standard/file_before_streaming_test.cc
+++ /dev/null
@@ -1,132 +0,0 @@
-/*
- * 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 "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h"
-
-namespace {
-
-const int kSampleRateHz = 16000;
-const int kTestDurationMs = 1000;
-const int kSkipOutputMs = 50;
-const int16_t kInputValue = 15000;
-const int16_t kSilenceValue = 0;
-
-} // namespace
-
-class FileBeforeStreamingTest : public AfterInitializationFixture {
- protected:
- FileBeforeStreamingTest()
- : input_filename_(webrtc::test::OutputPath() + "file_test_input.pcm"),
- output_filename_(webrtc::test::OutputPath() + "file_test_output.pcm") {
- }
-
- void SetUp() {
- channel_ = voe_base_->CreateChannel();
- }
-
- void TearDown() {
- voe_base_->DeleteChannel(channel_);
- }
-
- // TODO(andrew): consolidate below methods in a shared place?
-
- // Generate input file with constant values as |kInputValue|. The file
- // will be one second longer than the duration of the test.
- void GenerateInputFile() {
- FILE* input_file = fopen(input_filename_.c_str(), "wb");
- ASSERT_TRUE(input_file != NULL);
- for (int i = 0; i < kSampleRateHz / 1000 * (kTestDurationMs + 1000); i++) {
- ASSERT_EQ(1u, fwrite(&kInputValue, sizeof(kInputValue), 1, input_file));
- }
- ASSERT_EQ(0, fclose(input_file));
- }
-
- void RecordOutput() {
- // Start recording the mixed output for |kTestDurationMs| long.
- EXPECT_EQ(0, voe_file_->StartRecordingPlayout(-1,
- output_filename_.c_str()));
- Sleep(kTestDurationMs);
- EXPECT_EQ(0, voe_file_->StopRecordingPlayout(-1));
- }
-
- void VerifyOutput(int16_t target_value) {
- FILE* output_file = fopen(output_filename_.c_str(), "rb");
- ASSERT_TRUE(output_file != NULL);
- int16_t output_value = 0;
- int samples_read = 0;
-
- // Skip the first segment to avoid initialization and ramping-in effects.
- EXPECT_EQ(0, fseek(output_file, sizeof(output_value) *
- kSampleRateHz / 1000 * kSkipOutputMs, SEEK_SET));
- while (fread(&output_value, sizeof(output_value), 1, output_file) == 1) {
- samples_read++;
- EXPECT_EQ(output_value, target_value);
- }
-
- // Ensure that a reasonable amount was recorded. We use a loose
- // tolerance to avoid flaky bot failures.
- ASSERT_GE((samples_read * 1000.0) / kSampleRateHz, 0.4 * kTestDurationMs);
-
- // Ensure we read the entire file.
- ASSERT_NE(0, feof(output_file));
- ASSERT_EQ(0, fclose(output_file));
- }
-
-void VerifyEmptyOutput() {
- FILE* output_file = fopen(output_filename_.c_str(), "rb");
- ASSERT_TRUE(output_file != NULL);
- ASSERT_EQ(0, fseek(output_file, 0, SEEK_END));
- EXPECT_EQ(0, ftell(output_file));
- ASSERT_EQ(0, fclose(output_file));
-}
-
- int channel_;
- const std::string input_filename_;
- const std::string output_filename_;
-};
-
-// This test case is to ensure that StartPlayingFileLocally() and
-// StartPlayout() can be called in any order.
-// A DC signal is used as input. And the output of mixer is supposed to be:
-// 1. the same DC signal if file is played out,
-// 2. total silence if file is not played out,
-// 3. no output if playout is not started.
-TEST_F(FileBeforeStreamingTest, TestStartPlayingFileLocallyWithStartPlayout) {
- GenerateInputFile();
-
- TEST_LOG("Playout is not started. File will not be played out.\n");
- EXPECT_EQ(0, voe_file_->StartPlayingFileLocally(
- channel_, input_filename_.c_str(), true));
- EXPECT_EQ(1, voe_file_->IsPlayingFileLocally(channel_));
- RecordOutput();
- VerifyEmptyOutput();
-
- TEST_LOG("Playout is now started. File will be played out.\n");
- EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
- RecordOutput();
- VerifyOutput(kInputValue);
-
- TEST_LOG("Stop playing file. Only silence will be played out.\n");
- EXPECT_EQ(0, voe_file_->StopPlayingFileLocally(channel_));
- EXPECT_EQ(0, voe_file_->IsPlayingFileLocally(channel_));
- RecordOutput();
- VerifyOutput(kSilenceValue);
-
- TEST_LOG("Start playing file again. File will be played out.\n");
- EXPECT_EQ(0, voe_file_->StartPlayingFileLocally(
- channel_, input_filename_.c_str(), true));
- EXPECT_EQ(1, voe_file_->IsPlayingFileLocally(channel_));
- RecordOutput();
- VerifyOutput(kInputValue);
-
- EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
- EXPECT_EQ(0, voe_file_->StopPlayingFileLocally(channel_));
-}
diff --git a/voice_engine/test/auto_test/standard/file_test.cc b/voice_engine/test/auto_test/standard/file_test.cc
deleted file mode 100644
index 8b7ffb7..0000000
--- a/voice_engine/test/auto_test/standard/file_test.cc
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- * 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 "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
-#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
-
-
-class FileTest : public AfterStreamingFixture {
- protected:
- // Creates the string åäö.pcm.
-// TODO(henrika): enable this test once CreateTrickyFilenameInUtf8 no longer
-// prevents compilation on Windows. Likely webrtc/base can be used here.
-#if 0
- std::string CreateTrickyFilenameInUtf8() {
- char filename[16] = { (char)0xc3, (char)0xa5,
- (char)0xc3, (char)0xa4,
- (char)0xc3, (char)0xb6,
- static_cast<char>(0) };
- return std::string(filename) + ".pcm";
- }
-#endif // 0
-};
-
-// TODO(henrika): enable this test once CreateTrickyFilenameInUtf8 no longer
-// prevents compilation on Windows. Likely webrtc/base can be used here.
-#if 0
-TEST_F(FileTest, ManualRecordToFileForThreeSecondsAndPlayback) {
- if (!FLAGS_include_timing_dependent_tests) {
- TEST_LOG("Skipping test - running in slow execution environment...\n");
- return;
- }
-
- SwitchToManualMicrophone();
-
- std::string recording_filename =
- webrtc::test::OutputPath() + CreateTrickyFilenameInUtf8();
-
- TEST_LOG("Recording to %s for 3 seconds.\n", recording_filename.c_str());
- EXPECT_EQ(0, voe_file_->StartRecordingMicrophone(recording_filename.c_str()));
- Sleep(3000);
- EXPECT_EQ(0, voe_file_->StopRecordingMicrophone());
-
- TEST_LOG("Playing back %s.\n", recording_filename.c_str());
- EXPECT_EQ(0, voe_file_->StartPlayingFileLocally(
- channel_, recording_filename.c_str()));
-
- // Play the file to the user and ensure the is-playing-locally.
- // The clip is 3 seconds long.
- Sleep(250);
- EXPECT_EQ(1, voe_file_->IsPlayingFileLocally(channel_));
- Sleep(1500);
-}
-#endif // 0
-
-TEST_F(FileTest, ManualRecordPlayoutToWavFileForThreeSecondsAndPlayback) {
- webrtc::CodecInst send_codec;
- voe_codec_->GetSendCodec(channel_, send_codec);
-
- std::string recording_filename =
- webrtc::test::OutputPath() + "playout.wav";
-
- TEST_LOG("Recording playout to %s.\n", recording_filename.c_str());
- EXPECT_EQ(0, voe_file_->StartRecordingPlayout(
- channel_, recording_filename.c_str(), &send_codec));
- Sleep(3000);
- EXPECT_EQ(0, voe_file_->StopRecordingPlayout(channel_));
-
- TEST_LOG("Playing back the recording in looping mode.\n");
- EXPECT_EQ(0, voe_file_->StartPlayingFileAsMicrophone(
- channel_, recording_filename.c_str(), true, false,
- webrtc::kFileFormatWavFile));
-
- Sleep(2000);
- EXPECT_EQ(1, voe_file_->IsPlayingFileAsMicrophone(channel_));
- Sleep(2000);
- // We should still be playing since we're looping.
- EXPECT_EQ(1, voe_file_->IsPlayingFileAsMicrophone(channel_));
-}
