2011-07-07 08:21:25 +00:00
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/*
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2012-03-05 16:07:51 +00:00
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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2011-07-07 08:21:25 +00:00
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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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2014-01-13 15:21:30 +00:00
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#include <list>
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2016-02-24 05:00:36 -08:00
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#include <memory>
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2011-10-25 08:24:20 +00:00
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#include <string>
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2013-07-11 13:24:38 +00:00
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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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2015-10-28 18:17:40 +01:00
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#include "webrtc/system_wrappers/include/file_wrapper.h"
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2013-07-11 13:24:38 +00:00
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#include "webrtc/typedefs.h"
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2011-07-07 08:21:25 +00:00
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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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2014-01-13 15:21:30 +00:00
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struct AudioPacket
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{
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uint8_t dataBuffer[4 * 960];
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Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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size_t nSamples;
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size_t nBytesPerSample;
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Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-12 16:26:35 -08:00
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size_t nChannels;
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2014-01-13 15:21:30 +00:00
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uint32_t samplesPerSec;
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};
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2011-07-07 08:21:25 +00:00
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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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Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
2015-06-11 14:31:38 -07:00
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int32_t RecordedDataIsAvailable(const void* audioSamples,
|
Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
|
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const size_t nSamples,
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const size_t nBytesPerSample,
|
Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-12 16:26:35 -08:00
|
|
|
const size_t nChannels,
|
Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
2015-06-11 14:31:38 -07:00
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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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|
|
Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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int32_t NeedMorePlayData(const size_t nSamples,
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const size_t nBytesPerSample,
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Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-12 16:26:35 -08:00
|
|
|
const size_t nChannels,
|
Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
2015-06-11 14:31:38 -07:00
|
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const uint32_t samplesPerSec,
|
|
|
|
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void* audioSamples,
|
Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
|
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size_t& nSamplesOut,
|
Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
2015-06-11 14:31:38 -07:00
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) override;
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2011-07-07 08:21:25 +00:00
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AudioTransportImpl(AudioDeviceModule* audioDevice);
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~AudioTransportImpl();
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public:
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2013-04-09 10:30:35 +00:00
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int32_t SetFilePlayout(bool enable, const char* fileName = NULL);
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2011-07-07 08:21:25 +00:00
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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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|
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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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;
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|
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void SetMicrophoneAGC(bool enable)
|
|
|
|
|
{
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|
|
|
|
_microphoneAGC = enable;
|
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|
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}
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;
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private:
|
2014-01-13 15:21:30 +00:00
|
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typedef std::list<AudioPacket*> AudioPacketList;
|
2011-07-07 08:21:25 +00:00
|
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|
AudioDeviceModule* _audioDevice;
|
|
|
|
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|
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|
|
bool _playFromFile;
|
|
|
|
|
bool _fullDuplex;
|
|
|
|
|
bool _speakerVolume;
|
|
|
|
|
bool _speakerMute;
|
|
|
|
|
bool _microphoneVolume;
|
|
|
|
|
bool _microphoneMute;
|
|
|
|
|
bool _microphoneBoost;
|
|
|
|
|
bool _microphoneAGC;
|
|
|
|
|
bool _loopBackMeasurements;
|
|
|
|
|
|
|
|
|
|
FileWrapper& _playFile;
|
|
|
|
|
|
2013-04-09 10:30:35 +00:00
|
|
|
uint32_t _recCount;
|
|
|
|
|
uint32_t _playCount;
|
2014-01-13 15:21:30 +00:00
|
|
|
AudioPacketList _audioList;
|
2011-07-07 08:21:25 +00:00
|
|
|
|
|
|
|
|
Resampler _resampler;
|
|
|
|
|
};
|
|
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|
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|
// ----------------------------------------------------------------------------
|
|
|
|
|
// FuncTestManager
|
|
|
|
|
// ----------------------------------------------------------------------------
|
|
|
|
|
|
|
|
|
|
class FuncTestManager
|
|
|
|
|
{
|
|
|
|
|
public:
|
|
|
|
|
FuncTestManager();
|
|
|
|
|
~FuncTestManager();
|
2013-04-09 10:30:35 +00:00
|
|
|
int32_t Init();
|
|
|
|
|
int32_t Close();
|
|
|
|
|
int32_t DoTest(const TestType testType);
|
2011-07-07 08:21:25 +00:00
|
|
|
private:
|
2013-04-09 10:30:35 +00:00
|
|
|
int32_t TestAudioLayerSelection();
|
|
|
|
|
int32_t TestDeviceEnumeration();
|
|
|
|
|
int32_t TestDeviceSelection();
|
|
|
|
|
int32_t TestAudioTransport();
|
|
|
|
|
int32_t TestSpeakerVolume();
|
|
|
|
|
int32_t TestMicrophoneVolume();
|
|
|
|
|
int32_t TestSpeakerMute();
|
|
|
|
|
int32_t TestMicrophoneMute();
|
|
|
|
|
int32_t TestMicrophoneBoost();
|
|
|
|
|
int32_t TestLoopback();
|
|
|
|
|
int32_t TestDeviceRemoval();
|
|
|
|
|
int32_t TestExtra();
|
|
|
|
|
int32_t TestMicrophoneAGC();
|
|
|
|
|
int32_t SelectPlayoutDevice();
|
|
|
|
|
int32_t SelectRecordingDevice();
|
|
|
|
|
int32_t TestAdvancedMBAPI();
|
2011-07-07 08:21:25 +00:00
|
|
|
private:
|
2011-10-25 08:24:20 +00:00
|
|
|
// Paths to where the resource files to be used for this test are located.
|
|
|
|
|
std::string _playoutFile48;
|
|
|
|
|
std::string _playoutFile44;
|
|
|
|
|
std::string _playoutFile16;
|
|
|
|
|
std::string _playoutFile8;
|
|
|
|
|
|
2016-02-24 05:00:36 -08:00
|
|
|
std::unique_ptr<ProcessThread> _processThread;
|
2011-07-07 08:21:25 +00:00
|
|
|
AudioDeviceModule* _audioDevice;
|
|
|
|
|
AudioEventObserver* _audioEventObserver;
|
|
|
|
|
AudioTransportImpl* _audioTransport;
|
|
|
|
|
};
|
|
|
|
|
|
2013-07-03 15:12:26 +00:00
|
|
|
} // namespace webrtc
|
2011-07-07 08:21:25 +00:00
|
|
|
|
|
|
|
|
#endif // #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
|