2015-03-27 10:56:23 +01:00
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/*
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* Copyright (c) 2015 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_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
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#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
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2016-02-24 05:00:36 -08:00
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#include <memory>
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2015-03-27 10:56:23 +01:00
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#include <jni.h>
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/modules/audio_device/android/audio_common.h"
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2015-07-14 17:04:08 +02:00
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#include "webrtc/modules/audio_device/audio_device_config.h"
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2015-03-27 10:56:23 +01:00
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#include "webrtc/modules/audio_device/include/audio_device_defines.h"
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#include "webrtc/modules/audio_device/audio_device_generic.h"
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2015-11-04 08:31:52 +01:00
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#include "webrtc/modules/utility/include/helpers_android.h"
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#include "webrtc/modules/utility/include/jvm_android.h"
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2015-03-27 10:56:23 +01:00
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namespace webrtc {
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// Implements support for functions in the WebRTC audio stack for Android that
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// relies on the AudioManager in android.media. It also populates an
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// AudioParameter structure with native audio parameters detected at
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// construction. This class does not make any audio-related modifications
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// unless Init() is called. Caching audio parameters makes no changes but only
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// reads data from the Java side.
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class AudioManager {
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public:
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Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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// Wraps the Java specific parts of the AudioManager into one helper class.
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// Stores method IDs for all supported methods at construction and then
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// allows calls like JavaAudioManager::Close() while hiding the Java/JNI
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// parts that are associated with this call.
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class JavaAudioManager {
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public:
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JavaAudioManager(NativeRegistration* native_registration,
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2016-02-24 05:00:36 -08:00
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std::unique_ptr<GlobalRef> audio_manager);
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2015-05-28 14:18:33 +02:00
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~JavaAudioManager();
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Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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2015-05-28 14:18:33 +02:00
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bool Init();
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void Close();
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2015-06-05 11:45:56 +02:00
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bool IsCommunicationModeEnabled();
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2015-06-09 10:45:09 +02:00
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bool IsDeviceBlacklistedForOpenSLESUsage();
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Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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2015-05-28 14:18:33 +02:00
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private:
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2016-02-24 05:00:36 -08:00
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std::unique_ptr<GlobalRef> audio_manager_;
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2015-05-28 14:18:33 +02:00
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jmethodID init_;
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jmethodID dispose_;
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2015-06-05 11:45:56 +02:00
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jmethodID is_communication_mode_enabled_;
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2015-06-09 10:45:09 +02:00
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jmethodID is_device_blacklisted_for_open_sles_usage_;
|
Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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};
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2015-03-27 10:56:23 +01:00
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AudioManager();
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~AudioManager();
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Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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// Sets the currently active audio layer combination. Must be called before
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// Init().
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void SetActiveAudioLayer(AudioDeviceModule::AudioLayer audio_layer);
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2015-04-10 11:46:55 +02:00
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// Initializes the audio manager and stores the current audio mode.
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2015-03-27 10:56:23 +01:00
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bool Init();
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// Revert any setting done by Init().
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bool Close();
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2015-06-05 11:45:56 +02:00
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// Returns true if current audio mode is AudioManager.MODE_IN_COMMUNICATION.
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bool IsCommunicationModeEnabled() const;
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2015-04-10 11:46:55 +02:00
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2015-03-27 10:56:23 +01:00
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// Native audio parameters stored during construction.
|
Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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const AudioParameters& GetPlayoutAudioParameters();
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const AudioParameters& GetRecordAudioParameters();
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2015-09-23 14:08:33 +02:00
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// Returns true if the device supports built-in audio effects for AEC, AGC
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// and NS. Some devices can also be blacklisted for use in combination with
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// platform effects and these devices will return false.
|
Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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// Can currently only be used in combination with a Java based audio backend
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// for the recoring side (i.e. using the android.media.AudioRecord API).
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bool IsAcousticEchoCancelerSupported() const;
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2015-09-23 14:08:33 +02:00
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bool IsAutomaticGainControlSupported() const;
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bool IsNoiseSuppressorSupported() const;
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Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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// Returns true if the device supports the low-latency audio paths in
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// combination with OpenSL ES.
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bool IsLowLatencyPlayoutSupported() const;
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// Returns the estimated total delay of this device. Unit is in milliseconds.
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// The vaule is set once at construction and never changes after that.
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// Possible values are webrtc::kLowLatencyModeDelayEstimateInMilliseconds and
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// webrtc::kHighLatencyModeDelayEstimateInMilliseconds.
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int GetDelayEstimateInMilliseconds() const;
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2015-03-27 10:56:23 +01:00
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private:
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// Called from Java side so we can cache the native audio parameters.
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// This method will be called by the WebRtcAudioManager constructor, i.e.
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// on the same thread that this object is created on.
|
Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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static void JNICALL CacheAudioParameters(JNIEnv* env,
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jobject obj,
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jint sample_rate,
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jint channels,
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jboolean hardware_aec,
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2015-09-23 14:08:33 +02:00
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jboolean hardware_agc,
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jboolean hardware_ns,
|
Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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jboolean low_latency_output,
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jint output_buffer_size,
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jint input_buffer_size,
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jlong native_audio_manager);
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void OnCacheAudioParameters(JNIEnv* env,
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jint sample_rate,
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jint channels,
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jboolean hardware_aec,
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2015-09-23 14:08:33 +02:00
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jboolean hardware_agc,
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jboolean hardware_ns,
|
Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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jboolean low_latency_output,
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jint output_buffer_size,
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2015-11-03 04:27:58 -08:00
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jint input_buffer_size);
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2015-03-27 10:56:23 +01:00
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// Stores thread ID in the constructor.
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// We can then use ThreadChecker::CalledOnValidThread() to ensure that
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// other methods are called from the same thread.
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rtc::ThreadChecker thread_checker_;
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|
Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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// Calls AttachCurrentThread() if this thread is not attached at construction.
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// Also ensures that DetachCurrentThread() is called at destruction.
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AttachCurrentThreadIfNeeded attach_thread_if_needed_;
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2015-05-25 10:11:27 +02:00
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// Wraps the JNI interface pointer and methods associated with it.
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2016-02-24 05:00:36 -08:00
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std::unique_ptr<JNIEnvironment> j_environment_;
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Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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2015-05-25 10:11:27 +02:00
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// Contains factory method for creating the Java object.
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2016-02-24 05:00:36 -08:00
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std::unique_ptr<NativeRegistration> j_native_registration_;
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Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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2015-05-25 10:11:27 +02:00
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// Wraps the Java specific parts of the AudioManager.
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2016-02-24 05:00:36 -08:00
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std::unique_ptr<AudioManager::JavaAudioManager> j_audio_manager_;
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Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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AudioDeviceModule::AudioLayer audio_layer_;
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2015-03-27 10:56:23 +01:00
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// Set to true by Init() and false by Close().
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bool initialized_;
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Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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// True if device supports hardware (or built-in) AEC.
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bool hardware_aec_;
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2015-09-23 14:08:33 +02:00
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// True if device supports hardware (or built-in) AGC.
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bool hardware_agc_;
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// True if device supports hardware (or built-in) NS.
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bool hardware_ns_;
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Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 16:49:16 +02:00
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// True if device supports the low-latency OpenSL ES audio path.
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bool low_latency_playout_;
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// The delay estimate can take one of two fixed values depending on if the
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// device supports low-latency output or not.
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int delay_estimate_in_milliseconds_;
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2015-03-27 10:56:23 +01:00
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// Contains native parameters (e.g. sample rate, channel configuration).
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// Set at construction in OnCacheAudioParameters() which is called from
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// Java on the same thread as this object is created on.
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AudioParameters playout_parameters_;
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AudioParameters record_parameters_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
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