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webrtc_m130/modules/audio_device/include/audio_device.h

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Removes deprecated ADM APIs (reland) Usage should now be removed and this change can be relanded. It was reverted here: https://webrtc-review.googlesource.com/c/src/+/27200 NOTRY=TRUE TBR=solenberg Bug: webrtc:7306 Change-Id: I5191263e6cfd48952b59ff8f9af2e59c3e9eadef Reviewed-on: https://webrtc-review.googlesource.com/29682 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21080}
2017-12-05 11:23:00 +01:00
/*
Factory method for the ADM in the interface file. BUG=N/A TEST=no Review URL: https://webrtc-codereview.appspot.com/396017 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1753 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-23 17:45:33 +00:00
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-07 08:21:25 +00:00
*
* 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.
*/
The CL broke the builds is : http://code.google.com/p/webrtc/source/detail?r=2804 Review URL: https://webrtc-codereview.appspot.com/873012 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2911 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-11 15:41:55 +00:00
#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-07 08:21:25 +00:00
Fixing WebRTC after moving from src/webrtc to src/ 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}
2017-09-15 06:47:31 +02:00
#include "modules/audio_device/include/audio_device_defines.h"
Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." This is a reland of 34cdd2d402b08aee4e17a6fd38c87e0e5cd7aa30 Original change's description: > Remove AudioDeviceObserver and make ADM not inherit from the Module interface. > > (Re-upload of https://codereview.webrtc.org/3020493002/) > > Bug: webrtc:4690, webrtc:7306 > Change-Id: I67fb9ebca1296aabc08eae8a292a5c69832dc35e > Reviewed-on: https://webrtc-review.googlesource.com/5360 > Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20083} Bug: webrtc:4690, webrtc:7306 Change-Id: Ib019439fe6ab0e6b759819e1e9bd320ba1d983bd Reviewed-on: https://webrtc-review.googlesource.com/6300 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20137}
2017-10-04 13:27:21 +02:00
#include "rtc_base/refcount.h"
Fixing WebRTC after moving from src/webrtc to src/ 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}
2017-09-15 06:47:31 +02:00
#include "rtc_base/scoped_ref_ptr.h"
git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-07 08:21:25 +00:00
namespace webrtc {
Adds stream-switch support in new Windows ADM. Second round of the new Windows ADM is now ready for review. Main changes are: Supports internal (automatic) restart of audio streams when an active audio stream disconnects (happens when a device is removed). Adds support for IAudioClient3 and IAudioClient2 for platforms which supports it (>Win8 and >Win10). Modifies the threading model to support restart "from the inside" on the native audio thread. Adds two new test methods for the ADM to emulate restart events or stream-switch events. Adds two new test methods to support rate conversion to ensure that audio can be tested in loopback even if devices runs at different sample rates. Added initial components for low-latency support. Verified that it works but disabled it with a flag for now. Bug: webrtc:9265 Change-Id: Ia8e577daabea6b433f2c2eabab4e46ce8added6a Reviewed-on: https://webrtc-review.googlesource.com/86020 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24578}
2018-09-05 14:34:40 +02:00
class AudioDeviceModuleForTest;
Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." This is a reland of 34cdd2d402b08aee4e17a6fd38c87e0e5cd7aa30 Original change's description: > Remove AudioDeviceObserver and make ADM not inherit from the Module interface. > > (Re-upload of https://codereview.webrtc.org/3020493002/) > > Bug: webrtc:4690, webrtc:7306 > Change-Id: I67fb9ebca1296aabc08eae8a292a5c69832dc35e > Reviewed-on: https://webrtc-review.googlesource.com/5360 > Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20083} Bug: webrtc:4690, webrtc:7306 Change-Id: Ib019439fe6ab0e6b759819e1e9bd320ba1d983bd Reviewed-on: https://webrtc-review.googlesource.com/6300 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20137}
2017-10-04 13:27:21 +02:00
class AudioDeviceModule : public rtc::RefCountInterface {
Adds reference counting to the ADM. This CL modifies the ADM interface to ensure that an external ADM can't call Create and Destroy any longer. It also contains some minor style nits to conform better with the Chromium style guide. Review URL: http://webrtc-codereview.appspot.com/133014 git-svn-id: http://webrtc.googlecode.com/svn/trunk@552 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-07 15:11:18 +00:00
public:
Removes more unused ADM APIs: - RecordingDelay() - LastError() Bug: webrtc:7306 Change-Id: I3bb9cd243a1464f0ba612787c854eeb6602c7e38 Reviewed-on: https://webrtc-review.googlesource.com/12060 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20326}
2017-10-17 14:47:44 +02:00
// Deprecated.
// TODO(henrika): to be removed.
Adds reference counting to the ADM. This CL modifies the ADM interface to ensure that an external ADM can't call Create and Destroy any longer. It also contains some minor style nits to conform better with the Chromium style guide. Review URL: http://webrtc-codereview.appspot.com/133014 git-svn-id: http://webrtc.googlecode.com/svn/trunk@552 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-07 15:11:18 +00:00
enum ErrorCode { kAdmErrNone = 0, kAdmErrArgument = 1 };
enum AudioLayer {
kPlatformDefaultAudio = 0,
Adds support for new Windows ADM with limited API support. Summary of what this CL does: Existing users can keep using the old ADM for Windows as before. A new ADM for Windows is created and a dedicated factory method is used to create it. The old way (using AudioDeviceImpl) is not utilized. The new ADM is based on a structure where most of the "action" takes place in new AudioInput/AudioOutput implementations. This is inline with our mobile platforms and also makes it easier to break out common parts into a base class. The AudioDevice unittest has always mainly focused on the "Start/Stop"- parts of the ADM and not the complete ADM interface. This new ADM supports all tests in AudioDeviceTest and is therefore tested in combination with the old version. A value-parametrized test us added for Windows builds. Improved readability, threading model and makes the code easier to maintain. Uses the previously landed methods in webrtc::webrtc_win::core_audio_utility. Bug: webrtc:9265 Change-Id: If2894b44528e74a181cf7ad1216f57386ee3a24d Reviewed-on: https://webrtc-review.googlesource.com/78060 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23554}
2018-06-08 16:10:03 +02:00
kWindowsCoreAudio,
kWindowsCoreAudio2, // experimental
kLinuxAlsaAudio,
kLinuxPulseAudio,
kAndroidJavaAudio,
kAndroidOpenSLESAudio,
kAndroidJavaInputAndOpenSLESOutputAudio,
kAndroidAAudioAudio,
kAndroidJavaInputAndAAudioOutputAudio,
kDummyAudio,
Adds reference counting to the ADM. This CL modifies the ADM interface to ensure that an external ADM can't call Create and Destroy any longer. It also contains some minor style nits to conform better with the Chromium style guide. Review URL: http://webrtc-codereview.appspot.com/133014 git-svn-id: http://webrtc.googlecode.com/svn/trunk@552 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-07 15:11:18 +00:00
};
enum WindowsDeviceType {
kDefaultCommunicationDevice = -1,
kDefaultDevice = -2
};
Removes Set/GetRecordingChannel() from the ADM These two unused APIs are removed: SetRecordingChannel(const ChannelType channel) RecordingChannel(ChannelType* channel) const Bug: webrtc:7306 Change-Id: I3289c4b9a5eebb64cc0aa3a1c1144e9c4d6a661d Reviewed-on: https://webrtc-review.googlesource.com/22681 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20667}
2017-11-13 16:57:58 +01:00
// TODO(bugs.webrtc.org/7306): deprecated.
Adds reference counting to the ADM. This CL modifies the ADM interface to ensure that an external ADM can't call Create and Destroy any longer. It also contains some minor style nits to conform better with the Chromium style guide. Review URL: http://webrtc-codereview.appspot.com/133014 git-svn-id: http://webrtc.googlecode.com/svn/trunk@552 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-07 15:11:18 +00:00
enum ChannelType { kChannelLeft = 0, kChannelRight = 1, kChannelBoth = 2 };
public:
Adds stream-switch support in new Windows ADM. Second round of the new Windows ADM is now ready for review. Main changes are: Supports internal (automatic) restart of audio streams when an active audio stream disconnects (happens when a device is removed). Adds support for IAudioClient3 and IAudioClient2 for platforms which supports it (>Win8 and >Win10). Modifies the threading model to support restart "from the inside" on the native audio thread. Adds two new test methods for the ADM to emulate restart events or stream-switch events. Adds two new test methods to support rate conversion to ensure that audio can be tested in loopback even if devices runs at different sample rates. Added initial components for low-latency support. Verified that it works but disabled it with a flag for now. Bug: webrtc:9265 Change-Id: Ia8e577daabea6b433f2c2eabab4e46ce8added6a Reviewed-on: https://webrtc-review.googlesource.com/86020 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24578}
2018-09-05 14:34:40 +02:00
// Creates a default ADM for usage in production code.
Removes id parameter in ADM factory method (reland) Second attempt to land https://webrtc-review.googlesource.com/c/src/+/21980 Bug: webrtc:7306 Change-Id: Idbbb6d5371b154247ea3db55c28e5c0618c4fae1 Reviewed-on: https://webrtc-review.googlesource.com/22480 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20653}
2017-11-13 12:47:59 +01:00
static rtc::scoped_refptr<AudioDeviceModule> Create(
const AudioLayer audio_layer);
Adds stream-switch support in new Windows ADM. Second round of the new Windows ADM is now ready for review. Main changes are: Supports internal (automatic) restart of audio streams when an active audio stream disconnects (happens when a device is removed). Adds support for IAudioClient3 and IAudioClient2 for platforms which supports it (>Win8 and >Win10). Modifies the threading model to support restart "from the inside" on the native audio thread. Adds two new test methods for the ADM to emulate restart events or stream-switch events. Adds two new test methods to support rate conversion to ensure that audio can be tested in loopback even if devices runs at different sample rates. Added initial components for low-latency support. Verified that it works but disabled it with a flag for now. Bug: webrtc:9265 Change-Id: Ia8e577daabea6b433f2c2eabab4e46ce8added6a Reviewed-on: https://webrtc-review.googlesource.com/86020 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24578}
2018-09-05 14:34:40 +02:00
// Creates an ADM with support for extra test methods. Don't use this factory
// in production code.
static rtc::scoped_refptr<AudioDeviceModuleForTest> CreateForTest(
const AudioLayer audio_layer);
Removes Set/GetLoudspeakerStatus APIs from the ADM. int32_t SetLoudspeakerStatus(bool enable) int32_t GetLoudspeakerStatus(bool* enabled) const These APIs are only implemented on iOS and they do not belong in the native audio layer since the client can achieve the same functionality by using the shared audio session in sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h. It also gives the client a better flexibility in how the audio routing is done. Bug: webrtc:7306 Change-Id: I853e2f57e0f5ae0a0f9fc4729ce961d81f92588b Reviewed-on: https://webrtc-review.googlesource.com/23740 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20721}
2017-11-16 16:15:49 +01:00
// TODO(bugs.webrtc.org/7306): deprecated (to be removed).
Move ADM Create() method to public interface. ADMs were previously created by CreateAudioDeviceModule which was removed in previous refactoring without a replacement added. BUG= R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1944883002 . Cr-Commit-Position: refs/heads/master@{#12613}
2016-05-03 15:51:26 -04:00
static rtc::scoped_refptr<AudioDeviceModule> Create(
const int32_t id,
const AudioLayer audio_layer);
Adds reference counting to the ADM. This CL modifies the ADM interface to ensure that an external ADM can't call Create and Destroy any longer. It also contains some minor style nits to conform better with the Chromium style guide. Review URL: http://webrtc-codereview.appspot.com/133014 git-svn-id: http://webrtc.googlecode.com/svn/trunk@552 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-07 15:11:18 +00:00
// Retrieve the currently utilized audio layer
virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
// Full-duplex transportation of PCM audio
virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) = 0;
// Main initialization and termination
virtual int32_t Init() = 0;
virtual int32_t Terminate() = 0;
virtual bool Initialized() const = 0;
// Device enumeration
virtual int16_t PlayoutDevices() = 0;
virtual int16_t RecordingDevices() = 0;
virtual int32_t PlayoutDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) = 0;
virtual int32_t RecordingDeviceName(uint16_t index,
git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-07 08:21:25 +00:00
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) = 0;
Adds reference counting to the ADM. This CL modifies the ADM interface to ensure that an external ADM can't call Create and Destroy any longer. It also contains some minor style nits to conform better with the Chromium style guide. Review URL: http://webrtc-codereview.appspot.com/133014 git-svn-id: http://webrtc.googlecode.com/svn/trunk@552 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-07 15:11:18 +00:00
// Device selection
virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
virtual int32_t SetPlayoutDevice(WindowsDeviceType device) = 0;
virtual int32_t SetRecordingDevice(uint16_t index) = 0;
virtual int32_t SetRecordingDevice(WindowsDeviceType device) = 0;
// Audio transport initialization
virtual int32_t PlayoutIsAvailable(bool* available) = 0;
virtual int32_t InitPlayout() = 0;
virtual bool PlayoutIsInitialized() const = 0;
virtual int32_t RecordingIsAvailable(bool* available) = 0;
virtual int32_t InitRecording() = 0;
virtual bool RecordingIsInitialized() const = 0;
// Audio transport control
virtual int32_t StartPlayout() = 0;
virtual int32_t StopPlayout() = 0;
virtual bool Playing() const = 0;
virtual int32_t StartRecording() = 0;
virtual int32_t StopRecording() = 0;
virtual bool Recording() const = 0;
// Audio mixer initialization
virtual int32_t InitSpeaker() = 0;
virtual bool SpeakerIsInitialized() const = 0;
virtual int32_t InitMicrophone() = 0;
virtual bool MicrophoneIsInitialized() const = 0;
// Speaker volume controls
virtual int32_t SpeakerVolumeIsAvailable(bool* available) = 0;
virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;
virtual int32_t SpeakerVolume(uint32_t* volume) const = 0;
virtual int32_t MaxSpeakerVolume(uint32_t* maxVolume) const = 0;
virtual int32_t MinSpeakerVolume(uint32_t* minVolume) const = 0;
// Microphone volume controls
virtual int32_t MicrophoneVolumeIsAvailable(bool* available) = 0;
virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;
virtual int32_t MicrophoneVolume(uint32_t* volume) const = 0;
virtual int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const = 0;
virtual int32_t MinMicrophoneVolume(uint32_t* minVolume) const = 0;
// Speaker mute control
virtual int32_t SpeakerMuteIsAvailable(bool* available) = 0;
virtual int32_t SetSpeakerMute(bool enable) = 0;
virtual int32_t SpeakerMute(bool* enabled) const = 0;
// Microphone mute control
virtual int32_t MicrophoneMuteIsAvailable(bool* available) = 0;
virtual int32_t SetMicrophoneMute(bool enable) = 0;
virtual int32_t MicrophoneMute(bool* enabled) const = 0;
// Stereo support
virtual int32_t StereoPlayoutIsAvailable(bool* available) const = 0;
virtual int32_t SetStereoPlayout(bool enable) = 0;
virtual int32_t StereoPlayout(bool* enabled) const = 0;
virtual int32_t StereoRecordingIsAvailable(bool* available) const = 0;
virtual int32_t SetStereoRecording(bool enable) = 0;
virtual int32_t StereoRecording(bool* enabled) const = 0;
Removes more unused ADM APIs: - RecordingDelay() - LastError() Bug: webrtc:7306 Change-Id: I3bb9cd243a1464f0ba612787c854eeb6602c7e38 Reviewed-on: https://webrtc-review.googlesource.com/12060 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20326}
2017-10-17 14:47:44 +02:00
// Playout delay
Adds reference counting to the ADM. This CL modifies the ADM interface to ensure that an external ADM can't call Create and Destroy any longer. It also contains some minor style nits to conform better with the Chromium style guide. Review URL: http://webrtc-codereview.appspot.com/133014 git-svn-id: http://webrtc.googlecode.com/svn/trunk@552 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-07 15:11:18 +00:00
virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
Removes more unused ADM APIs: - RecordingDelay() - LastError() Bug: webrtc:7306 Change-Id: I3bb9cd243a1464f0ba612787c854eeb6602c7e38 Reviewed-on: https://webrtc-review.googlesource.com/12060 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20326}
2017-10-17 14:47:44 +02:00
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer BUG=4034 R=andrew@webrtc.org, perkj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 16:22:09 +00:00
// Only supported on Android.
Fix warnings, simplify ADM. This is in preparation for adding a gn target for audio_device_tests. BUG=webrtc:6170,webrtc:163 NOTRY=True Review-Url: https://codereview.webrtc.org/2222563002 Cr-Commit-Position: refs/heads/master@{#13768}
2016-08-16 00:56:09 -07:00
virtual bool BuiltInAECIsAvailable() const = 0;
virtual bool BuiltInAGCIsAvailable() const = 0;
virtual bool BuiltInNSIsAvailable() const = 0;
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer BUG=4034 R=andrew@webrtc.org, perkj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 16:22:09 +00:00
Improving support for Android Audio Effects in WebRTC. Now also supports AGC and NS effects and adds the possibility to override default settings. R=magjed@webrtc.org, pbos@webrtc.org, sophiechang@chromium.org TBR=perkj BUG=NONE Review URL: https://codereview.webrtc.org/1344563002 . Cr-Commit-Position: refs/heads/master@{#10030}
2015-09-23 14:08:33 +02:00
// Enables the built-in audio effects. Only supported on Android.
Fix warnings, simplify ADM. This is in preparation for adding a gn target for audio_device_tests. BUG=webrtc:6170,webrtc:163 NOTRY=True Review-Url: https://codereview.webrtc.org/2222563002 Cr-Commit-Position: refs/heads/master@{#13768}
2016-08-16 00:56:09 -07:00
virtual int32_t EnableBuiltInAEC(bool enable) = 0;
virtual int32_t EnableBuiltInAGC(bool enable) = 0;
virtual int32_t EnableBuiltInNS(bool enable) = 0;
// Only supported on iOS.
#if defined(WEBRTC_IOS)
virtual int GetPlayoutAudioParameters(AudioParameters* params) const = 0;
virtual int GetRecordAudioParameters(AudioParameters* params) const = 0;
#endif // WEBRTC_IOS
Cleanup of iOS AudioDevice implementation TBR=tkchin BUG=webrtc:4789 TEST=modules_unittests --gtest_filter=AudioDeviceTest* and AppRTCDemo Review URL: https://codereview.webrtc.org/1206783002 . Cr-Commit-Position: refs/heads/master@{#9578}
2015-07-14 17:04:08 +02:00
Adds reference counting to the ADM. This CL modifies the ADM interface to ensure that an external ADM can't call Create and Destroy any longer. It also contains some minor style nits to conform better with the Chromium style guide. Review URL: http://webrtc-codereview.appspot.com/133014 git-svn-id: http://webrtc.googlecode.com/svn/trunk@552 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-07 15:11:18 +00:00
protected:
Fix warnings, simplify ADM. This is in preparation for adding a gn target for audio_device_tests. BUG=webrtc:6170,webrtc:163 NOTRY=True Review-Url: https://codereview.webrtc.org/2222563002 Cr-Commit-Position: refs/heads/master@{#13768}
2016-08-16 00:56:09 -07:00
~AudioDeviceModule() override {}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-07 08:21:25 +00:00
};
Adds stream-switch support in new Windows ADM. Second round of the new Windows ADM is now ready for review. Main changes are: Supports internal (automatic) restart of audio streams when an active audio stream disconnects (happens when a device is removed). Adds support for IAudioClient3 and IAudioClient2 for platforms which supports it (>Win8 and >Win10). Modifies the threading model to support restart "from the inside" on the native audio thread. Adds two new test methods for the ADM to emulate restart events or stream-switch events. Adds two new test methods to support rate conversion to ensure that audio can be tested in loopback even if devices runs at different sample rates. Added initial components for low-latency support. Verified that it works but disabled it with a flag for now. Bug: webrtc:9265 Change-Id: Ia8e577daabea6b433f2c2eabab4e46ce8added6a Reviewed-on: https://webrtc-review.googlesource.com/86020 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24578}
2018-09-05 14:34:40 +02:00
// Extends the default ADM interface with some extra test methods.
// Intended for usage in tests only and requires a unique factory method.
class AudioDeviceModuleForTest : public AudioDeviceModule {
public:
// Triggers internal restart sequences of audio streaming. Can be used by
// tests to emulate events corresponding to e.g. removal of an active audio
// device or other actions which causes the stream to be disconnected.
virtual int RestartPlayoutInternally() = 0;
virtual int RestartRecordingInternally() = 0;
virtual int SetPlayoutSampleRate(uint32_t sample_rate) = 0;
virtual int SetRecordingSampleRate(uint32_t sample_rate) = 0;
};
git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-07 08:21:25 +00:00
} // namespace webrtc
The CL broke the builds is : http://code.google.com/p/webrtc/source/detail?r=2804 Review URL: https://webrtc-codereview.appspot.com/873012 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2911 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-11 15:41:55 +00:00
#endif // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
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