2011-07-07 08:21:25 +00:00
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/*
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2012-03-01 18:01:48 +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_AUDIO_DEVICE_IMPL_H
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#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H
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2015-05-20 09:44:38 +02:00
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#if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
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2016-02-24 05:00:36 -08:00
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#include <memory>
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2013-07-11 13:24:38 +00:00
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#include "webrtc/modules/audio_device/audio_device_buffer.h"
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#include "webrtc/modules/audio_device/include/audio_device.h"
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2017-07-06 19:44:34 +02:00
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#include "webrtc/rtc_base/checks.h"
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#include "webrtc/rtc_base/criticalsection.h"
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2011-07-07 08:21:25 +00:00
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2015-07-14 17:04:08 +02:00
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namespace webrtc {
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2011-07-07 08:21:25 +00:00
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class AudioDeviceGeneric;
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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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class AudioManager;
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2011-07-07 08:21:25 +00:00
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2015-07-14 17:04:08 +02:00
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class AudioDeviceModuleImpl : public AudioDeviceModule {
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public:
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enum PlatformType {
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kPlatformNotSupported = 0,
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kPlatformWin32 = 1,
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kPlatformWinCe = 2,
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kPlatformLinux = 3,
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kPlatformMac = 4,
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kPlatformAndroid = 5,
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kPlatformIOS = 6
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};
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int32_t CheckPlatform();
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int32_t CreatePlatformSpecificObjects();
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int32_t AttachAudioBuffer();
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AudioDeviceModuleImpl(const int32_t id, const AudioLayer audioLayer);
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2016-08-16 00:56:09 -07:00
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~AudioDeviceModuleImpl() override;
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2015-07-14 17:04:08 +02:00
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int64_t TimeUntilNextProcess() override;
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2016-02-25 04:50:01 -08:00
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void Process() override;
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2015-07-14 17:04:08 +02:00
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// Retrieve the currently utilized audio layer
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int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
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// Error handling
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ErrorCode LastError() const override;
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int32_t RegisterEventObserver(AudioDeviceObserver* eventCallback) override;
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// Full-duplex transportation of PCM audio
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int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
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// Main initializaton and termination
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int32_t Init() override;
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int32_t Terminate() override;
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bool Initialized() const override;
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// Device enumeration
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int16_t PlayoutDevices() override;
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int16_t RecordingDevices() override;
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int32_t PlayoutDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) override;
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int32_t RecordingDeviceName(uint16_t index,
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2015-03-04 12:58:35 +00:00
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]) override;
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2015-07-14 17:04:08 +02:00
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// Device selection
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int32_t SetPlayoutDevice(uint16_t index) override;
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int32_t SetPlayoutDevice(WindowsDeviceType device) override;
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int32_t SetRecordingDevice(uint16_t index) override;
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int32_t SetRecordingDevice(WindowsDeviceType device) override;
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// Audio transport initialization
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int32_t PlayoutIsAvailable(bool* available) override;
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int32_t InitPlayout() override;
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bool PlayoutIsInitialized() const override;
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int32_t RecordingIsAvailable(bool* available) override;
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int32_t InitRecording() override;
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bool RecordingIsInitialized() const override;
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// Audio transport control
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int32_t StartPlayout() override;
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int32_t StopPlayout() override;
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bool Playing() const override;
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int32_t StartRecording() override;
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int32_t StopRecording() override;
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bool Recording() const override;
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// Microphone Automatic Gain Control (AGC)
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int32_t SetAGC(bool enable) override;
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bool AGC() const override;
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// Audio mixer initialization
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int32_t InitSpeaker() override;
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bool SpeakerIsInitialized() const override;
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int32_t InitMicrophone() override;
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bool MicrophoneIsInitialized() const override;
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// Speaker volume controls
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int32_t SpeakerVolumeIsAvailable(bool* available) override;
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int32_t SetSpeakerVolume(uint32_t volume) override;
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int32_t SpeakerVolume(uint32_t* volume) const override;
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int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override;
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int32_t MinSpeakerVolume(uint32_t* minVolume) const override;
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// Microphone volume controls
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int32_t MicrophoneVolumeIsAvailable(bool* available) override;
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int32_t SetMicrophoneVolume(uint32_t volume) override;
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int32_t MicrophoneVolume(uint32_t* volume) const override;
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int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override;
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int32_t MinMicrophoneVolume(uint32_t* minVolume) const override;
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// Speaker mute control
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int32_t SpeakerMuteIsAvailable(bool* available) override;
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int32_t SetSpeakerMute(bool enable) override;
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int32_t SpeakerMute(bool* enabled) const override;
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// Microphone mute control
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int32_t MicrophoneMuteIsAvailable(bool* available) override;
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int32_t SetMicrophoneMute(bool enable) override;
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int32_t MicrophoneMute(bool* enabled) const override;
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// Stereo support
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int32_t StereoPlayoutIsAvailable(bool* available) const override;
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int32_t SetStereoPlayout(bool enable) override;
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int32_t StereoPlayout(bool* enabled) const override;
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int32_t StereoRecordingIsAvailable(bool* available) const override;
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int32_t SetStereoRecording(bool enable) override;
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int32_t StereoRecording(bool* enabled) const override;
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int32_t SetRecordingChannel(const ChannelType channel) override;
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int32_t RecordingChannel(ChannelType* channel) const override;
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// Delay information and control
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int32_t PlayoutDelay(uint16_t* delayMS) const override;
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int32_t RecordingDelay(uint16_t* delayMS) const override;
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// Native sample rate controls (samples/sec)
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int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) override;
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int32_t RecordingSampleRate(uint32_t* samplesPerSec) const override;
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int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) override;
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int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const override;
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// Mobile device specific functions
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int32_t SetLoudspeakerStatus(bool enable) override;
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int32_t GetLoudspeakerStatus(bool* enabled) const override;
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bool BuiltInAECIsAvailable() const override;
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int32_t EnableBuiltInAEC(bool enable) override;
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2015-09-23 14:08:33 +02:00
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bool BuiltInAGCIsAvailable() const override;
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int32_t EnableBuiltInAGC(bool enable) override;
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bool BuiltInNSIsAvailable() const override;
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int32_t EnableBuiltInNS(bool enable) override;
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2015-07-14 17:04:08 +02:00
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2016-08-16 00:56:09 -07:00
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#if defined(WEBRTC_IOS)
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2015-07-14 17:04:08 +02:00
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int GetPlayoutAudioParameters(AudioParameters* params) const override;
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int GetRecordAudioParameters(AudioParameters* params) const override;
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2016-08-16 00:56:09 -07:00
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#endif // WEBRTC_IOS
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2015-07-14 17:04:08 +02:00
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int32_t Id() { return _id; }
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2015-05-20 09:44:38 +02:00
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#if defined(WEBRTC_ANDROID)
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2015-07-14 17:04:08 +02:00
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// Only use this acccessor for test purposes on Android.
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AudioManager* GetAndroidAudioManagerForTest() {
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return _audioManagerAndroid.get();
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}
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2015-05-19 11:48:51 +02:00
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#endif
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2015-07-14 17:04:08 +02:00
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AudioDeviceBuffer* GetAudioDeviceBuffer() { return &_audioDeviceBuffer; }
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2015-03-09 12:39:53 +00:00
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2015-07-14 17:04:08 +02:00
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private:
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PlatformType Platform() const;
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AudioLayer PlatformAudioLayer() const;
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2011-07-07 08:21:25 +00:00
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2017-03-30 01:14:41 -07:00
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rtc::CriticalSection _critSect;
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rtc::CriticalSection _critSectEventCb;
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rtc::CriticalSection _critSectAudioCb;
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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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2015-07-14 17:04:08 +02:00
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AudioDeviceObserver* _ptrCbAudioDeviceObserver;
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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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2015-07-14 17:04:08 +02:00
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AudioDeviceGeneric* _ptrAudioDevice;
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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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2015-07-14 17:04:08 +02:00
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AudioDeviceBuffer _audioDeviceBuffer;
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2015-05-20 09:44:38 +02:00
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#if defined(WEBRTC_ANDROID)
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2016-02-24 05:00:36 -08:00
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std::unique_ptr<AudioManager> _audioManagerAndroid;
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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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#endif
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2015-07-14 17:04:08 +02:00
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int32_t _id;
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AudioLayer _platformAudioLayer;
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int64_t _lastProcessTime;
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PlatformType _platformType;
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bool _initialized;
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mutable ErrorCode _lastError;
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2011-07-07 08:21:25 +00:00
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};
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} // namespace webrtc
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2015-05-20 09:44:38 +02:00
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#endif // defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
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2011-07-07 08:21:25 +00:00
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#endif // WEBRTC_MODULES_INTERFACE_AUDIO_DEVICE_IMPL_H_
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