ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
108 lines
3.5 KiB
C++
108 lines
3.5 KiB
C++
/*
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* Copyright (c) 2014 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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// Unit tests for FilePlayer.
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#include "webrtc/modules/utility/include/file_player.h"
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#include <stdio.h>
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#include <string>
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#include "gflags/gflags.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/base/md5digest.h"
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#include "webrtc/base/stringencode.h"
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#include "webrtc/test/testsupport/fileutils.h"
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#include "webrtc/test/testsupport/gtest_disable.h"
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DEFINE_bool(file_player_output, false, "Generate reference files.");
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namespace webrtc {
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class FilePlayerTest : public ::testing::Test {
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protected:
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static const uint32_t kId = 0;
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static const FileFormats kFileFormat = kFileFormatWavFile;
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static const int kSampleRateHz = 8000;
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FilePlayerTest()
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: player_(FilePlayer::CreateFilePlayer(kId, kFileFormat)),
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output_file_(NULL) {}
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void SetUp() override {
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if (FLAGS_file_player_output) {
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std::string output_file =
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webrtc::test::OutputPath() + "file_player_unittest_out.pcm";
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output_file_ = fopen(output_file.c_str(), "wb");
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ASSERT_TRUE(output_file_ != NULL);
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}
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}
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void TearDown() override {
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if (output_file_)
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fclose(output_file_);
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}
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~FilePlayerTest() { FilePlayer::DestroyFilePlayer(player_); }
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void PlayFileAndCheck(const std::string& input_file,
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const std::string& ref_checksum,
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int output_length_ms) {
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const float kScaling = 1;
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ASSERT_EQ(0,
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player_->StartPlayingFile(
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input_file.c_str(), false, 0, kScaling, 0, 0, NULL));
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rtc::Md5Digest checksum;
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for (int i = 0; i < output_length_ms / 10; ++i) {
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int16_t out[10 * kSampleRateHz / 1000] = {0};
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size_t num_samples;
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EXPECT_EQ(0,
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player_->Get10msAudioFromFile(out, num_samples, kSampleRateHz));
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checksum.Update(out, num_samples * sizeof(out[0]));
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if (FLAGS_file_player_output) {
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ASSERT_EQ(num_samples,
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fwrite(out, sizeof(out[0]), num_samples, output_file_));
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}
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}
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char checksum_result[rtc::Md5Digest::kSize];
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EXPECT_EQ(rtc::Md5Digest::kSize,
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checksum.Finish(checksum_result, rtc::Md5Digest::kSize));
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EXPECT_EQ(ref_checksum,
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rtc::hex_encode(checksum_result, sizeof(checksum_result)));
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}
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FilePlayer* player_;
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FILE* output_file_;
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};
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TEST_F(FilePlayerTest, DISABLED_ON_IOS(PlayWavPcmuFile)) {
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const std::string kFileName =
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test::ResourcePath("utility/encapsulated_pcmu_8khz", "wav");
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// The file is longer than this, but keeping the output shorter limits the
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// runtime for the test.
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const int kOutputLengthMs = 10000;
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const std::string kRefChecksum = "c74e7fd432d439b1311e1c16815b3e9a";
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PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs);
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}
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TEST_F(FilePlayerTest, DISABLED_ON_IOS(PlayWavPcm16File)) {
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const std::string kFileName =
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test::ResourcePath("utility/encapsulated_pcm16b_8khz", "wav");
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// The file is longer than this, but keeping the output shorter limits the
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// runtime for the test.
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const int kOutputLengthMs = 10000;
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const std::string kRefChecksum = "e41d7e1dac8aeae9f21e8e03cd7ecd71";
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PlayFileAndCheck(kFileName, kRefChecksum, kOutputLengthMs);
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}
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} // namespace webrtc
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