Remove audio_device_test_func.
This code does not work and hasn't been used in a long time. It also lacks a GN target. There's no reason to save it. BUG=none Review-Url: https://codereview.webrtc.org/2255173002 Cr-Commit-Position: refs/heads/master@{#13820}
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maxmorin
Commit bot
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3a11933a63
@ -277,24 +277,6 @@
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'test/audio_device_test_defines.h',
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],
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},
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{
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'target_name': 'audio_device_test_func',
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'type': 'executable',
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'dependencies': [
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'audio_device',
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'webrtc_utility',
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'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
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'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
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'<(webrtc_root)/test/test.gyp:test_support',
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'<(DEPTH)/testing/gtest.gyp:gtest',
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],
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'sources': [
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'test/audio_device_test_func.cc',
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'test/audio_device_test_defines.h',
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'test/func_test_manager.cc',
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'test/func_test_manager.h',
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],
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},
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], # targets
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}], # include_tests==1 and OS!=ios
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],
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@ -1,23 +0,0 @@
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INSTRUCTIONS:
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- Start with test #3 (Device enumeration) to get an overview of the available
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audio devices.
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- Next, proceed with test #4 (Device selection) to get more details about
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the supported functions for each audio device.
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- Verify two-way audio in test #5.
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Repeat this test for different selections of playout and recording devices.
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- More detailed tests (volume, mute etc.) can also be performed using #6-#11.
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NOTE:
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- Some tests requires that the user opens up the audio mixer dialog and
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verifies that a certain action (e.g. Mute ON/OFF) is executed correctly.
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- Files can be recorded during some tests to enable off-line analysis.
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- Full support of 'Default Communication' devices requires Windows 7.
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- If a test consists of several sub tests, press any key to start a new sub test.
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KNOWN ISSUES:
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- Microphone Boost control is not supported on Windows Vista or Windows 7.
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- Speaker and microphone volume controls will not work as intended on Windows
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Vista if a 'Default Communication' device is selected in any direction.
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@ -1,162 +0,0 @@
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/*
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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 <stdio.h>
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#include "webrtc/modules/audio_device/test/audio_device_test_defines.h"
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#include "webrtc/modules/audio_device/test/func_test_manager.h"
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#ifndef __GNUC__
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// Disable warning message 4996 ('scanf': This function or variable may be unsafe)
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#pragma warning( disable : 4996 )
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#endif
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using namespace webrtc;
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int func_test(int);
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// ----------------------------------------------------------------------------
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// main()
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// ----------------------------------------------------------------------------
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#if !defined(WEBRTC_IOS)
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int main(int /*argc*/, char* /*argv*/[])
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{
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func_test(0);
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}
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#endif
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// ----------------------------------------------------------------------------
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// func_test()
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// ----------------------------------------------------------------------------
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int func_test(int sel)
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{
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TEST_LOG("=========================================\n");
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TEST_LOG("Func Test of the WebRtcAudioDevice Module\n");
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TEST_LOG("=========================================\n\n");
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// Initialize the counters here to get rid of "unused variables" warnings.
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warningCount = 0;
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FuncTestManager funcMgr;
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funcMgr.Init();
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bool quit(false);
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while (!quit)
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{
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TEST_LOG("---------------------------------------\n");
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TEST_LOG("Select type of test\n\n");
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TEST_LOG(" (0) Quit\n");
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TEST_LOG(" (1) All\n");
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TEST_LOG("- - - - - - - - - - - - - - - - - - - -\n");
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TEST_LOG(" (2) Audio-layer selection\n");
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TEST_LOG(" (3) Device enumeration\n");
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TEST_LOG(" (4) Device selection\n");
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TEST_LOG(" (5) Audio transport\n");
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TEST_LOG(" (6) Speaker volume\n");
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TEST_LOG(" (7) Microphone volume\n");
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TEST_LOG(" (8) Speaker mute\n");
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TEST_LOG(" (9) Microphone mute\n");
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TEST_LOG(" (10) Microphone boost\n");
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TEST_LOG(" (11) Microphone AGC\n");
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TEST_LOG(" (12) Loopback measurements\n");
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TEST_LOG(" (13) Device removal\n");
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TEST_LOG(" (14) Advanced mobile device API\n");
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TEST_LOG(" (66) XTEST\n");
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TEST_LOG("- - - - - - - - - - - - - - - - - - - -\n");
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TEST_LOG("\n: ");
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int selection(0);
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enum TestType testType(TTInvalid);
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SHOW_MENU:
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if (sel > 0)
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{
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selection = sel;
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}
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else
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{
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if (scanf("%d", &selection) < 0) {
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perror("Failed to get selection.");
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}
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}
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switch (selection)
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{
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case 0:
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quit = true;
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break;
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case 1:
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testType = TTAll;
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break;
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case 2:
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testType = TTAudioLayerSelection;
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break;
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case 3:
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testType = TTDeviceEnumeration;
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break;
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case 4:
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testType = TTDeviceSelection;
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break;
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case 5:
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testType = TTAudioTransport;
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break;
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case 6:
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testType = TTSpeakerVolume;
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break;
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case 7:
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testType = TTMicrophoneVolume;
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break;
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case 8:
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testType = TTSpeakerMute;
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break;
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case 9:
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testType = TTMicrophoneMute;
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break;
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case 10:
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testType = TTMicrophoneBoost;
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break;
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case 11:
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testType = TTMicrophoneAGC;
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break;
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case 12:
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testType = TTLoopback;
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break;
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case 13:
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testType = TTDeviceRemoval;
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break;
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case 14:
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testType = TTMobileAPI;
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break;
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case 66:
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testType = TTTest;
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break;
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default:
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testType = TTInvalid;
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TEST_LOG(": ");
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goto SHOW_MENU;
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break;
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}
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funcMgr.DoTest(testType);
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if (sel > 0)
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{
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quit = true;
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}
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}
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funcMgr.Close();
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return 0;
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}
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File diff suppressed because it is too large
Load Diff
@ -1,233 +0,0 @@
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/*
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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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#ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
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#define WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
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#include <list>
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#include <memory>
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#include <string>
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#include "webrtc/common_audio/resampler/include/resampler.h"
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#include "webrtc/modules/audio_device/include/audio_device.h"
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#include "webrtc/modules/audio_device/test/audio_device_test_defines.h"
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#include "webrtc/system_wrappers/include/file_wrapper.h"
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#include "webrtc/typedefs.h"
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#define ADM_AUDIO_LAYER AudioDeviceModule::kPlatformDefaultAudio
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//#define ADM_AUDIO_LAYER AudioDeviceModule::kLinuxPulseAudio
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enum TestType
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{
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TTInvalid = -1,
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TTAll = 0,
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TTAudioLayerSelection = 1,
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TTDeviceEnumeration = 2,
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TTDeviceSelection = 3,
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TTAudioTransport = 4,
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TTSpeakerVolume = 5,
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TTMicrophoneVolume = 6,
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TTSpeakerMute = 7,
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TTMicrophoneMute = 8,
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TTMicrophoneBoost = 9,
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TTMicrophoneAGC = 10,
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TTLoopback = 11,
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TTDeviceRemoval = 13,
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TTMobileAPI = 14,
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TTTest = 66,
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};
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struct AudioPacket
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{
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uint8_t dataBuffer[4 * 960];
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size_t nSamples;
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size_t nBytesPerSample;
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size_t nChannels;
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uint32_t samplesPerSec;
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};
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class ProcessThread;
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namespace webrtc
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{
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class AudioDeviceModule;
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class AudioEventObserver;
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class AudioTransport;
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// ----------------------------------------------------------------------------
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// AudioEventObserver
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// ----------------------------------------------------------------------------
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class AudioEventObserver: public AudioDeviceObserver
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{
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public:
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virtual void OnErrorIsReported(const ErrorCode error);
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virtual void OnWarningIsReported(const WarningCode warning);
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AudioEventObserver(AudioDeviceModule* audioDevice);
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~AudioEventObserver();
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public:
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ErrorCode _error;
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WarningCode _warning;
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};
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// ----------------------------------------------------------------------------
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// AudioTransport
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// ----------------------------------------------------------------------------
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class AudioTransportImpl: public AudioTransport
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{
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public:
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int32_t RecordedDataIsAvailable(const void* audioSamples,
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const size_t nSamples,
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const size_t nBytesPerSample,
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const size_t nChannels,
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const uint32_t samplesPerSec,
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const uint32_t totalDelayMS,
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const int32_t clockDrift,
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const uint32_t currentMicLevel,
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const bool keyPressed,
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uint32_t& newMicLevel) override;
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int32_t NeedMorePlayData(const size_t nSamples,
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const size_t nBytesPerSample,
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const size_t nChannels,
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const uint32_t samplesPerSec,
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void* audioSamples,
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size_t& nSamplesOut,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) override;
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void PushCaptureData(int voe_channel,
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const void* audio_data,
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int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames) override;
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void PullRenderData(int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames,
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void* audio_data,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) override;
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AudioTransportImpl(AudioDeviceModule* audioDevice);
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~AudioTransportImpl();
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public:
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int32_t SetFilePlayout(bool enable, const char* fileName = NULL);
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void SetFullDuplex(bool enable);
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void SetSpeakerVolume(bool enable)
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{
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_speakerVolume = enable;
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}
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;
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void SetSpeakerMute(bool enable)
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{
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_speakerMute = enable;
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}
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;
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void SetMicrophoneMute(bool enable)
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{
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_microphoneMute = enable;
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}
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;
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void SetMicrophoneVolume(bool enable)
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{
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_microphoneVolume = enable;
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}
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;
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void SetMicrophoneBoost(bool enable)
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{
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_microphoneBoost = enable;
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}
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;
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void SetLoopbackMeasurements(bool enable)
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{
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_loopBackMeasurements = enable;
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}
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;
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void SetMicrophoneAGC(bool enable)
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{
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_microphoneAGC = enable;
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}
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;
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private:
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typedef std::list<AudioPacket*> AudioPacketList;
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AudioDeviceModule* _audioDevice;
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bool _playFromFile;
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bool _fullDuplex;
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bool _speakerVolume;
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bool _speakerMute;
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bool _microphoneVolume;
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bool _microphoneMute;
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bool _microphoneBoost;
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bool _microphoneAGC;
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bool _loopBackMeasurements;
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FileWrapper& _playFile;
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uint32_t _recCount;
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uint32_t _playCount;
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AudioPacketList _audioList;
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Resampler _resampler;
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};
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// ----------------------------------------------------------------------------
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// FuncTestManager
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// ----------------------------------------------------------------------------
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class FuncTestManager
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{
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public:
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FuncTestManager();
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~FuncTestManager();
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int32_t Init();
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int32_t Close();
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int32_t DoTest(const TestType testType);
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private:
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int32_t TestAudioLayerSelection();
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int32_t TestDeviceEnumeration();
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int32_t TestDeviceSelection();
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int32_t TestAudioTransport();
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int32_t TestSpeakerVolume();
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int32_t TestMicrophoneVolume();
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int32_t TestSpeakerMute();
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int32_t TestMicrophoneMute();
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int32_t TestMicrophoneBoost();
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int32_t TestLoopback();
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int32_t TestDeviceRemoval();
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int32_t TestExtra();
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int32_t TestMicrophoneAGC();
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int32_t SelectPlayoutDevice();
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int32_t SelectRecordingDevice();
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int32_t TestAdvancedMBAPI();
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private:
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// Paths to where the resource files to be used for this test are located.
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std::string _playoutFile48;
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std::string _playoutFile44;
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std::string _playoutFile16;
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std::string _playoutFile8;
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std::unique_ptr<ProcessThread> _processThread;
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AudioDeviceModule* _audioDevice;
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AudioEventObserver* _audioEventObserver;
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AudioTransportImpl* _audioTransport;
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};
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} // namespace webrtc
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#endif // #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
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